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From: "Hisham Khartabil" <hisham.khartabil@hotsip.com>
To: "Jean-Francois Mule" <jfmule@clarent.com>, <archow@hss.hns.com>
Cc: <sip@lists.bell-labs.com>
Subject: RE: [SIP] SIP proxies and CSeq numbers
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Date: Wed, 1 Nov 2000 11:20:34 +0200
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I don't think this is a correct behaviour of the proxy.  The proxy should
wait for the ACK and then the new INVITE.  It forwards the second INVITE
with Cseq=2.  If the proxy was confident that the authentication is ok, why
would it challenge the client in the first place?

Regards,
Hisham
Hotsip Finland

-----Original Message-----
From: sip-admin@lists.bell-labs.com [mailto:sip-admin@lists.bell-labs.com]On
Behalf Of Jean-Francois Mule
Sent: Tuesday, 31 October 2000 7:39 PM
To: archow@hss.hns.com
Cc: 'sip@lists.bell-labs.com '
Subject: RE: [SIP] SIP proxies and CSeq numbers

(A record-route will be added in the call flow to close this discussion on
the sip t.38 side).

But I still believe we have a bug based on what you correctly outlined
Arjun.  CSeq numbers are truly relevant in the context of a sip transaction
and can easily get out of sequence (sick), especially for statefull proxies.
Unless the record-route header is added, we have a recipe for guaranteeing
call failures.

Take this other scenario where the proxy is seating between a public net & a
private trusted net and the proxy requires authentication on one leg only.
        A -> P: authentication is required, A is in the public net.  A
initiates an invite (F1), P challenges A (F2) but P also sends the request
to B (F3) while waiting for the credentials (proxies may act like this: the
proxy knows that A comes from a known but untrusted network and has
confidence the authentication should be ok so it sends the request F3 before
receiving F4).  Another justification for proxies to act this way may be
that to check the credentials on F4, the proxy has check with a remote
authentication server.
        P -> B: no authentication required

If the UA A behaves as you stated in this scenario and sends the ACK
directly to B, the CSeq=2 in the ACK: game over.
Any suggestions on how do we fix this?

Jean-Francois Mule'

SIP UA: A                 Proxy                       SIP UA: B

 |                          |                          |
 |   F1 INVITE (Cseq=1)     |                          |
 |------------------------->|                          |
 |                          |                          |
 |   F2  407 Proxy auth. req|                          |
 |<-------------------------|   F3 INVITE  (Cseq=1)    |
 |                          |------------------------->|
 |   F4 INVITE (Cseq=2)     |                          |
 |------------------------->|                          |
 |                          |   F5  Trying (Cseq=1)    |
 |                          |<-------------------------|
 |   F6  Trying (Cseq=2)    |                          |
 |<-------------------------|                          |
 |                          |   F7  200 OK (Cseq=1)    |
 |                          |<-------------------------|
 |   F8  200 OK (Cseq=2)    |                          |
 |<-------------------------|                          |
 |                                                     |
 |                      F9  ACK (Cseq=2) !             |
 |---------------------------------------------------->|
 |                          |                          |


> -----Original Message-----
> From: archow@hss.hns.com [mailto:archow@hss.hns.com]
> Sent: Monday, October 30, 2000 10:06 PM
> To: Jean-Francois Mule
> Cc: archow@hss.hns.com; 'sip@lists.bell-labs.com '
> Subject: RE: [SIP] SIP proxies and CSeq numbers (was: SIP and T.38 fax
> cal ls - Internet Draft)
>
>
>
>
> Jean,
> Consider this situation:
>
> A sends an INVITE which goes through your Proxy to B. B sends
> a 200 OK with
> Contact as userb@here.com.
> This is routed through your proxy and goes to A.
> Now since A has a direct path to B (and the proxy did not do a
> Record-Route),
> A now sends the next request (ACK) directly to B ( protocol behaviour)
>
> The ACK must have the same CSeq as the INVITE although is a
> transaction of
> its own. But in this case,
> the ACK will have the CSeq of A - P which is different from
> the CSeq space
> of P-B. Since ACK went directly
> B would think this ACK is not valid.
> The same problems will arise in any future requests within
> the same call
> which is forwarded directly.
>
> This , as you would notice induces incorrect behaviour.
>
> jean> PS: As for the Record-Route, it was omitted (while it
> is key to proxy
> jean> functions, isn't it optional after all? -- 6.34.1).
> I'll add the
> jean> Record-route to the next I-D revision - thanks.
>
> Yes, its not mandatory. However in your scenario, if you do not do
> Record-Route
> future request can skip your proxy who then cannot change
> CSeq (if thats
> needed in
> the first place)
>
> Regds
> Arjun
>
> --
> Arjun Roychowdhury @ Hughes Software Systems

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From sip-admin@lists.bell-labs.com  Wed Nov  1 05:27:02 2000
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To: Hisham Khartabil <hisham.khartabil@hotsip.com>
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        sip@lists.bell-labs.com
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Hisham Khartabil wrote:
> 
> I don't think this is a correct behaviour of the proxy.  The proxy should
> wait for the ACK and then the new INVITE.  It forwards the second INVITE
> with Cseq=2.  If the proxy was confident that the authentication is ok, why
> would it challenge the client in the first place?

It certainly isn't correct behaviour. Proxy 1 must not proxy 
on an INVITE (F3) after sending back a final response 407 (F2). 
It also must not change CSeq values in 200 responses (F7 + F8). 

Cheers,
Neil.
-- 
Ubiquity Software Corporation, UK        http://www.ubiquity.net

> 
> Regards,
> Hisham
> Hotsip Finland
> 
> -----Original Message-----
> From: sip-admin@lists.bell-labs.com [mailto:sip-admin@lists.bell-labs.com]On
> Behalf Of Jean-Francois Mule
> Sent: Tuesday, 31 October 2000 7:39 PM
> To: archow@hss.hns.com
> Cc: 'sip@lists.bell-labs.com '
> Subject: RE: [SIP] SIP proxies and CSeq numbers
> 
> (A record-route will be added in the call flow to close this discussion on
> the sip t.38 side).
> 
> But I still believe we have a bug based on what you correctly outlined
> Arjun.  CSeq numbers are truly relevant in the context of a sip transaction
> and can easily get out of sequence (sick), especially for statefull proxies.
> Unless the record-route header is added, we have a recipe for guaranteeing
> call failures.
> 
> Take this other scenario where the proxy is seating between a public net & a
> private trusted net and the proxy requires authentication on one leg only.
>         A -> P: authentication is required, A is in the public net.  A
> initiates an invite (F1), P challenges A (F2) but P also sends the request
> to B (F3) while waiting for the credentials (proxies may act like this: the
> proxy knows that A comes from a known but untrusted network and has
> confidence the authentication should be ok so it sends the request F3 before
> receiving F4).  Another justification for proxies to act this way may be
> that to check the credentials on F4, the proxy has check with a remote
> authentication server.
>         P -> B: no authentication required
> 
> If the UA A behaves as you stated in this scenario and sends the ACK
> directly to B, the CSeq=2 in the ACK: game over.
> Any suggestions on how do we fix this?
> 
> Jean-Francois Mule'
> 
> SIP UA: A                 Proxy                       SIP UA: B
> 
>  |                          |                          |
>  |   F1 INVITE (Cseq=1)     |                          |
>  |------------------------->|                          |
>  |                          |                          |
>  |   F2  407 Proxy auth. req|                          |
>  |<-------------------------|   F3 INVITE  (Cseq=1)    |
>  |                          |------------------------->|
>  |   F4 INVITE (Cseq=2)     |                          |
>  |------------------------->|                          |
>  |                          |   F5  Trying (Cseq=1)    |
>  |                          |<-------------------------|
>  |   F6  Trying (Cseq=2)    |                          |
>  |<-------------------------|                          |
>  |                          |   F7  200 OK (Cseq=1)    |
>  |                          |<-------------------------|
>  |   F8  200 OK (Cseq=2)    |                          |
>  |<-------------------------|                          |
>  |                                                     |
>  |                      F9  ACK (Cseq=2) !             |
>  |---------------------------------------------------->|
>  |                          |                          |

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Subject: RE: [SIP] SIP proxies and CSeq numbers
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Date: Wed, 1 Nov 2000 17:25:31 +0530



Jean,
responses inline and prefixed by ARC>

Take this other scenario where the proxy is seating between a public net &
a
private trusted net and the proxy requires authentication on one leg only.
     A -> P: authentication is required, A is in the public net.  A
initiates an invite (F1), P challenges A (F2) but P also sends the request
to B (F3) while waiting for the credentials (proxies may act like this: the
proxy knows that A comes from a known but untrusted network and has
confidence the authentication should be ok so it sends the request F3
before
receiving F4).  Another justification for proxies to act this way may be
that to check the credentials on F4, the proxy has check with a remote
authentication server.
     P -> B: no authentication required

If the UA A behaves as you stated in this scenario and sends the ACK
directly to B, the CSeq=2 in the ACK: game over.


Any suggestions on how do we fix this?



ARC> Yes. The proxy should not forward the request (F3) to B in the first
place.
ARC> Since you say the proxy wants to challenge A, it should not go about
forwarding
ARC> the request to B. Let it wait for the challenge to be met. When agreed
upon
ARC> it should then forward the correct messge to B (in your case F4).


Jean-Francois Mule'

SIP UA: A                 Proxy                       SIP UA: B

 |                          |                          |
 |   F1 INVITE (Cseq=1)     |                          |
 |------------------------->|                          |
 |                          |                          |
 |   F2  407 Proxy auth. req|                          |
 |<-------------------------|   F3 INVITE  (Cseq=1)    |
 |                          |------------------------->|
 |   F4 INVITE (Cseq=2)     |                          |
 |------------------------->|                          |
 |                          |   F5  Trying (Cseq=1)    |
 |                          |<-------------------------|
 |   F6  Trying (Cseq=2)    |                          |
 |<-------------------------|                          |
 |                          |   F7  200 OK (Cseq=1)    |
 |                          |<-------------------------|
 |   F8  200 OK (Cseq=2)    |                          |
 |<-------------------------|                          |
 |                                                     |
 |                      F9  ACK (Cseq=2) !             |
 |---------------------------------------------------->|
 |                          |                          |


> -----Original Message-----
> From: archow@hss.hns.com [mailto:archow@hss.hns.com]
> Sent: Monday, October 30, 2000 10:06 PM
> To: Jean-Francois Mule
> Cc: archow@hss.hns.com; 'sip@lists.bell-labs.com '
> Subject: RE: [SIP] SIP proxies and CSeq numbers (was: SIP and T.38 fax




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Date: Wed, 1 Nov 2000 17:44:39 +0530



Robert,
if you have the new ordering/syntax done already,
would you mind posting them right away so we could plough in the correct
version right away ?

Thanks
Arjun

--
Arjun Roychowdhury @ Hughes Software Systems





"Robert Sparks" <rsparks@dynamicsoft.com> on 10/31/2000 11:11:29 PM

To:   "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>, Ashok I Roy/HSSBLR,
      sip@lists.bell-labs.com
cc:

Subject:  RE: referred-by syntax (was RE: [SIP] (no subject))




nod - we agreed to that (and altering the syntax to remove
the ordering dependency between "scheme" and "ref") in an
earlier thread.

I plan to get -02 out (which reflects that suggestion) this
week.

RjS


> > URL parameters are _definitely_ allowed. If your URL has a
> > semicolon in it,
> > it is required to be bracketed with <>, so you don't have the
> > ambiguity you
> > suspected.
>
> I agree thats what you should do, but thats not what the grammar says.
The
> grammar says these are just URLs. To allow for the brackets, they
> need to be
> name-addr or addr-spec.
>
> -Jonathan R.
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>


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From sip-admin@lists.bell-labs.com  Wed Nov  1 07:29:05 2000
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Subject: [SIP] Errors handling on session description in ACK
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Hello,

When UAC send session description in INVITE request, UAS send 200 OK
if it supports them or 606 Not Acceptable if not. Clear.
(I assume ideal situation: no 1xx, etc.)

When UAC doesn't send session description in INVITE (what means
that it's going to do that in ACK), UAS' 200 OK contains description.
Next, UAC sees that it will not be able to support capabilities
of UAS. Does it send session descr. as if it would send in INVITE,
regardless of this failed handshake ? This scenario will fail,
because there is no way of negotiation after UAC has sent ACK,
since ACK doesn't generate response.


More generaly:
1. When UAC has an error and doesn't send session descr. in ACK (
   INVITE did NOT contain session descr. as well), Should
   UAS just ignore such ACK and treat it as *not received* ?

2. Should garbage in body of type application/sdp generate
   4xx response (which one ?) or 606 with appropriate code in 
   Warning header (which one ?)


By the way,

I've compared bis-02 with draft-ietf-mmusic-sip-new-00.ps (August '99).

Section 10.5.1 Unreliable Transport Protocols (bis-02):

"(...)Response retransmissions cease when (...) the response has been
 RETRANSMITTED seven times."

Section 10.5.1 (draft-ietf-mmusic-sip-new-00.ps):

"(...)Response retransmissions cease when (...) the response has been
 TRANSMITTED seven times."

Is it Typo or it must have been realy changed ?
If it's only typo, bis-02' the section should be like in old draft or:

"(...)Response retransmissions cease when (...) the response has been
 RETRANSMITTED SIX times."


Comment?

Best regards,
Piotr


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RFC 2543 (SIP) references the session announcement protocol (SAP), which exists only as a 1996 Internet Draft, that is, a "work in progress".  So is SAP actually a "work in progress"?  If so, why hasn't anything been published on it since the 1996 Internet Draft?  Or is it considered obsolete?  

Thanks for any info, and apologies in advance if this is a dumb question.

Duncan Thomson


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From sip-admin@lists.bell-labs.com  Wed Nov  1 13:22:06 2000
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Date: Wed, 1 Nov 2000 10:18:38 -0800

Arjun:
Your suggestion would be the recommended way of handling this scenario and
this is how most of us are doing it.  But in any case, I do not think we can
enforce this at all.  Again, my proxy may want to buy some time when sending
F3 so that as soon as get F5, I have the F5 right in the queue and send the
F6 right away.

Jean-Francois.

> ARC> Yes. The proxy should not forward the request (F3) to B 
> in the first
> place.
> ARC> Since you say the proxy wants to challenge A, it should 
> not go about
> forwarding
> ARC> the request to B. Let it wait for the challenge to be 
> met. When agreed
> upon
> ARC> it should then forward the correct messge to B (in your case F4).


> 
> 
> Jean-Francois Mule'
> 
> SIP UA: A                 Proxy                       SIP UA: B
> 
>  |                          |                          |
>  |   F1 INVITE (Cseq=1)     |                          |
>  |------------------------->|                          |
>  |                          |                          |
>  |   F2  407 Proxy auth. req|                          |
>  |<-------------------------|   F3 INVITE  (Cseq=1)    |
>  |                          |------------------------->|
>  |   F4 INVITE (Cseq=2)     |                          |
>  |------------------------->|                          |
>  |                          |   F5  Trying (Cseq=1)    |
>  |                          |<-------------------------|
>  |   F6  Trying (Cseq=2)    |                          |
>  |<-------------------------|                          |
>  |                          |   F7  200 OK (Cseq=1)    |
>  |                          |<-------------------------|
>  |   F8  200 OK (Cseq=2)    |                          |
>  |<-------------------------|                          |
>  |                                                     |
>  |                      F9  ACK (Cseq=2) !             |
>  |---------------------------------------------------->|
>  |                          |                          |

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Date: Wed, 01 Nov 2000 17:37:13 -0500

The draft of T.38 Annex-D I've read is here (can anyone point me to a more 
up-to-date draft?):

ftp://standards.nortelnetworks.com/itu_to_ietf/SG8/February00/

Based on that reading, the setup procedure of TCP is inextricably bridged 
between two protocols: SDP and SIP.  It says that SDP is used to declare 
that (a) TCP will be used, and (b) which port number to connect to.  But 
the direction of the call determines the direction of the TCP connection 
(callee must connect to caller, and caller must accept this connection), as 
this is not declared in SDP.  So now instead of having a complete 
description of the TCP behavior solely in the SDP packets, the context of 
the call must be known in order to correctly predict the behavior of the 
endpoints.

The I-D listed below addresses this problem, such that the behavior of the 
endpoints is fully described by the SDP exchange.

That said, T.38 Annex-D and the TCP/SDP I-D do not conflict in any major 
way.  The I-D defines that if the "direction" attribute is omitted, both 
sides are to attempt to establish connections.  If T.38 opts to set up the 
TCP connection based on call context in lieu of the "direction" attribute, 
that's probably not a huge discrepancy.  Anyone disagree?

What is notably missing from my copy if T.38 Annex-D is an example of a 
T.38 SIP exchange that specifies TCP.  The question I have is, what does 
the receiver of the call put into its return SDP packet if it is agreeing 
to use TCP?  Does it say "TCP"?  If so, what port number does it 
specify?  Presumably the sender knows to ignore this port number, yes?


At 04:46 PM 11/1/00 -0500, James Rafferty wrote:
>To all -
>
>The registration from ITU-T SG8 for IANA as related to SDP was related to
>the
>Approval last February of Annex D to T.38, which defines how to use SIP/SDP
>with T.38 for facsimile session control.   T.38 needs to support both UDP
>and TCP modes of operation, hence the need for the SDP extension to support
>TCP (Glenn Parsons of Nortel was the editor).
>
>So, the methods for using TCP within SDP for T.38 session control purposes
>have been  described in detail in the T.38 Annex D.
>
>So, I'd suggest that people who want to write ID's related to this, take a
>look at what is in the T.38 Annex D (TIES users can read it as TD262rev2 in
>the SG8 informal FTP area).   There is also a small amendment to T.38 annex
>D (see Com8-114) which will be reviewed at the SG16 meeting during the week
>of Nov 13-17.
>
>I just checked on the ITU site and notice that T.38 Amendment 2 is not
>published yet, so the TD262rev2 above is the only access to that text at the
>moment.  Hopefully, this will change soon.
>
>James
>
>* -----------------------------------
>James Rafferty
>Senior Product Manager, IP Telephony
>jraff@brooktrout.com
>
>Brooktrout Technology
>410 First Avenue
>Needham, MA 02494 USA
>Phone:  +1-781-433-9462
>Fax: 781-433-9268
>www.brooktrout.com
>Your Hook into the New Network(SM)
>
> > -----Original Message-----
> > From: David Yon [mailto:yon@dialout.net]
> > Sent: Monday, October 30, 2000 3:54 PM
> > To: Jean-Francois Mule; ietf-fax@imc.org
> > Cc: sip@lists.bell-labs.com
> > Subject: Re: [SIP] RE: Request for information on using T.30/T.37/T.38
> > with SIP/SDP/ RTP
> >
> >
> > I know that the I-D specifically defers the discussion of
> > media transport
> > using TCP rather than UDPTL, but at the same time it also
> > refers to a new
> > token "TCP" to be registered with IANA for use in SDP.  I am
> > of the opinion
> > that simply specifying "TCP" is insufficient to describe how
> > TCP-based
> > Media Transport is to be negotiated between two endpoints.
> >
> > I've submitted an Internet Draft for the MMUSIC group which
> > will hopefully
> > be available in the IETF I-D area soon, but for the moment it
> > can be read here:
> >
>ftp://ftp.dialout.net/drafts/draft-ietf-mmusic-sdp-tcpmedia-00.txt
>
>So, in regards to the SIP/T.38 I-D, my only comment would be that it take
>into account the I-D on SDP/TCP.  Or at the very least, participate with
>better defining the SDP/TCP I-D such that is meets the needs of SIP/T.38
>when using TCP for Media Transport.  As such, any comments on the I-D above
>would be greatly appreciated.
>
>Again, hopefully the draft will be in the IETF area soon, I don't know why
>it is taking so long to work through the queue.
>
>At 02:26 PM 10/27/00 -0700, Jean-Francois Mule wrote:
> >There is an Internet-Draft on SIP and T.38 fax calls at
> >http://search.ietf.org/internet-drafts/draft-mule-sip-t38callflows-00.txt
> >It deals with how to negotiate a t.38 fax calls using sip as a signaling
> >protocol (including switch over from audio/rtp to data/t38), the SDP
> >attributes that itu-t sg-8 has registered with IANA.  Comments are
> >appreciated on this Internet-Draft.
> >
> >
>
>
>David Yon
>Chief Technical Officer
>Dialout.Net, Inc.
>yon@dialout.net
>
>


David Yon
Chief Technical Officer
Dialout.Net, Inc.
yon@dialout.net


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Subject: [SIP] Sip- Message OPTION
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Date: Wed, 1 Nov 2000 15:43:30 -0800

Hi,



Can any one tell me what is the sip-message for OPTION look like.

I was trying to find the same in the call flow diagram document , but I
cannot see an example for an OPTION sip message.

So, please let me an example of sip-message for OPTION.

Thanks

Sandeep Sahai
Ph:- 408-571-3388 (O)
       510-487-2121 (R)
Journey of thousand miles starts with single step.


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Date: Wed, 1 Nov 2000 16:47:36 -0800

Hi, 
    Pardon me if this has already been discussed earlier, but can
somebody verify the following behavior :

1. Unknown method
   Stateless Proxy :           Forward transparently
   Trans - Stateful proxy:     Forward if possible, else 405/501  
   Call - stateful proxy:      Forward if possible, else 405/501
   Standalone Registrar server:Return 405/501 
   Redirect server:            Return 405/501
   UAS:                        Return 405/501

   The decision to send 405 or 501 is dependent upon whether the
   received message is recognized or not. If the method is recognized
   and supported but not allowed, 405 is the appropriate response, else
   501 should be sent.

2. Unknown SIP header(s)
   Stateless Proxy :           Forward transparently
   Trans - Stateful proxy:     Ignore and if possible forward  
   Call - stateful proxy:      Ignore and if possible forward
   Standalone Registrar server:Ignore
   Redirect server:            Ignore
   UAS:                        Ignore

-regards,
aseem


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Subject: Re: [SIP] Sip- Message OPTION
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Date: Thu, 2 Nov 2000 10:38:47 +0530

--0__=JygTJOpFvQSO2POw4aQSzXxtz15DQHr36OhF6mMLN7NToAlJLmRH2wXL
Content-type: text/plain; charset=us-ascii
Content-Disposition: inline



Maybe the one in the bis-draft would be of help:

A caller Alice can use an OPTIONS request to find out the capabilities of a
 potential callee Bob, with-out

--0__=JygTJOpFvQSO2POw4aQSzXxtz15DQHr36OhF6mMLN7NToAlJLmRH2wXL
Content-type: text/plain; charset=windows-1257
Content-Disposition: inline
Content-Transfer-Encoding: quoted-printable


=93ringing=94 the designated address. Bob returns a description indicat=
ing that
 he is capable of receiving
audio encodings PCM mu-law (RTP payload type 0), 1016 (payload type 1),=
 GSM
 (payload type 3), and
SX7300/8000 (dynamic payload type 99), and video encodings H.261 (paylo=
ad
type 31) and H.263 (payload
type 34).

C->S: OPTIONS sip:bob@example.com SIP/2.0
Via: SIP/2.0/UDP cat.wonderland.com
From: Alice <sip:alice@wonderland.com>
To: Bob <sip:bob@example.com>
Call-ID: 6378@cat.wonderland.com
CSeq: 1 OPTIONS
Accept: application/sdp

S->C: SIP/2.0 200 OK
From: Alice <sip:alice@wonderland.com>
To: Bob <sip:bob@example.com> ;tag=3D376364382
Call-ID: 6378@cat.wonderland.com
Content-Length: 81
Content-Type: application/sdp
v=3D0
o=3Dalice 3149329138 3149329165 IN IP4 24.124.37.3
s=3DSecurity problems
t=3D3149328650 0
c=3DIN IP4 24.124.37.3
m=3Daudio 0 RTP/AVP 0 1 3 99
a=3Drtpmap:0 PCMU/8000
a=3Drtpmap:1 1016/8000
a=3Drtpmap:3 GSM/8000
a=3Drtpmap:99 SX7300/8000
m=3Dvideo 0 RTP/AVP 31 34
a=3Drtpmap:31 H261/90000
a=3Drtpmap:34 H263/90000

Regds
Arjun
--
Arjun Roychowdhury @ Hughes Software Systems






Sandeep Sahai <ssahai@lboard.com> on 11/02/2000 05:13:30 AM

To:   "'sip@lists.bell-labs.com'" <sip@lists.bell-labs.com>
cc:

Subject:  [SIP] Sip- Message OPTION


=

--0__=JygTJOpFvQSO2POw4aQSzXxtz15DQHr36OhF6mMLN7NToAlJLmRH2wXL
Content-type: text/plain; charset=us-ascii
Content-Disposition: inline


Hi,



Can any one tell me what is the sip-message for OPTION look like.

I was trying to find the same in the call flow diagram document , but I
cannot see an example for an OPTION sip message.

So, please let me an example of sip-message for OPTION.

Thanks

Sandeep Sahai
Ph:- 408-571-3388 (O)
       510-487-2121 (R)
Journey of thousand miles starts with single step.


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--0__=JygTJOpFvQSO2POw4aQSzXxtz15DQHr36OhF6mMLN7NToAlJLmRH2wXL--


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Date: Wed, 1 Nov 2000 21:40:38 -0800 (PST)

Dear All:
    I have one question to discuss. In order to 
detect loop, RFC2543bis02 page 81 says"A request has
been looped if the server finds its own address in the
Via header field and the hash computation over the
fields enumerated in Section 6.46.6 yields the same
value as the hash part of the ¡°branch¡± parameter in
the Via entry containing the proxy server¡¯s
address.",
Page 59"In order to be able to both detect
loops and associate responses with the corresponding
request, the parameter SHOULD consist of two parts
separable by the implementation. One part, used for
loop detection (Section 12.3.1), MAY be computed
as a cryptographic hash of the To, From, Call-ID
header fields, the Request-URI of the request received
(before translation) and the sequence number from the
CSeq header field. The algorithm used to compute the
hash is implementation-dependent, but MD5 [38],
expressed in hexadecimal, is a reasonable choice.".
According to my knowledge, it is necessary to detect
the Via to find whether loop or not, why use these
parameters to detect loop, Is the hash is easy to
implement? Could anyone give me advice on this?
   Thanks a lot!
   Best regards!
                        Yours: James


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Date: Thu, 2 Nov 2000 00:44:58 -0500



 

> -----Original Message-----
> From: Jean-Francois Mule [mailto:jfmule@clarent.com]
> Sent: Wednesday, November 01, 2000 1:19 PM
> To: archow@hss.hns.com
> Cc: 'sip@lists.bell-labs.com '
> Subject: RE: [SIP] SIP proxies and CSeq numbers
> 
> 
> Arjun:
> Your suggestion would be the recommended way of handling this 
> scenario and
> this is how most of us are doing it.  But in any case, I do 
> not think we can
> enforce this at all.

Well, we can't enforce people building elements compliant to the spec. The
behavior you are advocating is simply not compliant. It will not function
correctly with anyone elses equipment. It will result in bizarre, incorrect
behavior (things like the called party gets rung even though the credentials
fail, resulting in a dangling half call at the called party). 

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Date: Thu, 2 Nov 2000 00:46:02 -0500

It is now RFC2974.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Duncan Thomson [mailto:duncant@mitre.org]
> Sent: Wednesday, November 01, 2000 12:04 PM
> To: sip@lists.bell-labs.com
> Subject: [SIP] What is the status of SAP?
> 
> 
> RFC 2543 (SIP) references the session announcement protocol 
> (SAP), which exists only as a 1996 Internet Draft, that is, a 
> "work in progress".  So is SAP actually a "work in progress"? 
>  If so, why hasn't anything been published on it since the 
> 1996 Internet Draft?  Or is it considered obsolete?  
> 
> Thanks for any info, and apologies in advance if this is a 
> dumb question.
> 
> Duncan Thomson
> 
> 
> _______________________________________________
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> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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Date: Thu, 2 Nov 2000 00:51:44 -0500



 

> -----Original Message-----
> From: james jack [mailto:sipjames@yahoo.com]
> Sent: Friday, October 20, 2000 2:50 AM
> To: SIP@lists.bell-labs.com
> Subject: [SIP] Discussion
> 
> 
> Hi,All:
>   I have one question to consult, the question is that
> about the 481 Call Leg/Transaction Does Not Exist, on
> RFC2543bis02, which says "This status is returned
> under three conditions: The server received a BYE
> request that does not match any existing call leg, the
> server received a CANCEL request that does not match
> any existing transaction or the server received an
> INVITE with a To tag that does not match the local tag
> value. (A server simply discards an ACK referring to
> an unknown transaction.)" . I can't understand the
> words "or the server received an INVITE with a To tag
> that does not match the local tag value", could
> someone give me some example? According to my known,To
> tag is inserted by UAS or redirect server. which means
> To tag is used in response, how can a INVITE request
> with a To tag?

Subsequent INVITE requests from the caller to called party for the same call
leg carry that tag in the To field. The 481 is sent if a UAS receives a
request with a tag in the To field that it did not previously insert.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Date: Thu, 2 Nov 2000 00:09:15 -0600

Agarwal, Aseem (aseem@trillium.com):

>     Pardon me if this has already been discussed earlier, but can
> somebody verify the following behavior :

  You are almost correct.  I think your table is slightly misleading.

> 1. Unknown method
>    Trans - Stateful proxy:     Forward if possible, else 405/501  
>    Call - stateful proxy:      Forward if possible, else 405/501

  A proxy should proxy the request.  Only if the request is destined for
the proxy (the Request-URI was 'sip:my.proxy.com') should the proxy
return a 501.

>    Redirect server:            Return 405/501

  Again, if the request is destined for a user of the redirect server,
the server should happily return a 300-class response to the unknown
request.  It should only return a 501 if the request was destined for
the redirect server (the Request-URI was 'sip:my.redirect.com') and the
server didn't understand it.

> 2. Unknown SIP header(s)
>    Trans - Stateful proxy:     Ignore and if possible forward  
>    Call - stateful proxy:      Ignore and if possible forward

  Why do you say 'if possible'?  It would be very, very bad if a proxy
stripped out headers it didn't understand.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Subject: Re: [SIP] Sip- Message OPTION
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Date: Thu, 2 Nov 2000 00:16:42 -0600

  Note that this should be cleaned up.  There is no Via or CSeq in the
response, the CSeq in the request starts at 1, the From has no tag, and there
is no Content-Length.

> C->S: OPTIONS sip:bob@example.com SIP/2.0
> Via: SIP/2.0/UDP cat.wonderland.com
> From: Alice <sip:alice@wonderland.com>
> To: Bob <sip:bob@example.com>
> Call-ID: 6378@cat.wonderland.com
> CSeq: 1 OPTIONS
> Accept: application/sdp
> 
> S->C: SIP/2.0 200 OK
> From: Alice <sip:alice@wonderland.com>
> To: Bob <sip:bob@example.com> ;tag=376364382
> Call-ID: 6378@cat.wonderland.com
> Content-Length: 81
> Content-Type: application/sdp
>
> v=0
> o=alice 3149329138 3149329165 IN IP4 24.124.37.3
> s=Security problems
> t=3149328650 0
> c=IN IP4 24.124.37.3
> m=audio 0 RTP/AVP 0 1 3 99
> a=rtpmap:0 PCMU/8000
> a=rtpmap:1 1016/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:99 SX7300/8000
> m=video 0 RTP/AVP 31 34
> a=rtpmap:31 H261/90000
> a=rtpmap:34 H263/90000

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Well,
Its in the examples section and the intro text for Section 16 says:

"In the following examples, we often omit the message body and the
corresponding Content-Length and
Content-Type headers for brevity." (BTW, responses has a C-Len).

And I dont think the From needs a tag to be correct, does it ?

Further, I dont see the problem of using CSeq 1. This OPTIONS message may
well be sent from alice to bob
even if they are not in a call. So 1 seems ok.

So I guess only Via needs to be added, in this list.

Regds
Arjun

--
Arjun Roychowdhury @ Hughes Software Systems












Billy Biggs <Billy_Biggs@3com.com> on 11/02/2000 11:46:42 AM

To:   archow
cc:   Sandeep Sahai <ssahai@lboard.com>, SIP List <sip@lists.bell-labs.com>

Subject:  Re: [SIP] Sip- Message OPTION




  Note that this should be cleaned up.  There is no Via or CSeq in the
response, the CSeq in the request starts at 1, the From has no tag, and
there
is no Content-Length.

> C->S: OPTIONS sip:bob@example.com SIP/2.0
> Via: SIP/2.0/UDP cat.wonderland.com
> From: Alice <sip:alice@wonderland.com>
> To: Bob <sip:bob@example.com>
> Call-ID: 6378@cat.wonderland.com
> CSeq: 1 OPTIONS
> Accept: application/sdp
>
> S->C: SIP/2.0 200 OK
> From: Alice <sip:alice@wonderland.com>
> To: Bob <sip:bob@example.com> ;tag=376364382
> Call-ID: 6378@cat.wonderland.com
> Content-Length: 81
> Content-Type: application/sdp
>
> v=0
> o=alice 3149329138 3149329165 IN IP4 24.124.37.3
> s=Security problems
> t=3149328650 0
> c=IN IP4 24.124.37.3
> m=audio 0 RTP/AVP 0 1 3 99
> a=rtpmap:0 PCMU/8000
> a=rtpmap:1 1016/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:99 SX7300/8000
> m=video 0 RTP/AVP 31 34
> a=rtpmap:31 H261/90000
> a=rtpmap:34 H263/90000

--
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com





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From: "Thys Nel" <thys@parsec.co.za>
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Date: Thu, 2 Nov 2000 08:47:44 +0200
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Does anyone have examples of messages flows with OPTIONS processed by a
registrar server? The only example in the RFC is between 2 user agents, but
the RFC states that a registrar must also support it. I can't seem to find
any description of when a registrar should respond to an OPTIONS request.

Thys

> -----Original Message-----
> From: sip-admin@lists.bell-labs.com
> [mailto:sip-admin@lists.bell-labs.com]On Behalf Of Billy Biggs
> Sent: 02 November 2000 8:17
> To: archow@hss.hns.com
> Cc: Sandeep Sahai; SIP List
> Subject: Re: [SIP] Sip- Message OPTION
>
>
>   Note that this should be cleaned up.  There is no Via or CSeq in the
> response, the CSeq in the request starts at 1, the From has
> no tag, and there
> is no Content-Length.
>
> > C->S: OPTIONS sip:bob@example.com SIP/2.0
> > Via: SIP/2.0/UDP cat.wonderland.com
> > From: Alice <sip:alice@wonderland.com>
> > To: Bob <sip:bob@example.com>
> > Call-ID: 6378@cat.wonderland.com
> > CSeq: 1 OPTIONS
> > Accept: application/sdp
> >
> > S->C: SIP/2.0 200 OK
> > From: Alice <sip:alice@wonderland.com>
> > To: Bob <sip:bob@example.com> ;tag=376364382
> > Call-ID: 6378@cat.wonderland.com
> > Content-Length: 81
> > Content-Type: application/sdp
> >
> > v=0
> > o=alice 3149329138 3149329165 IN IP4 24.124.37.3
> > s=Security problems
> > t=3149328650 0
> > c=IN IP4 24.124.37.3
> > m=audio 0 RTP/AVP 0 1 3 99
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:1 1016/8000
> > a=rtpmap:3 GSM/8000
> > a=rtpmap:99 SX7300/8000
> > m=video 0 RTP/AVP 31 34
> > a=rtpmap:31 H261/90000
> > a=rtpmap:34 H263/90000
>
> --
> Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com
>
> _______________________________________________
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> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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Date: Thu, 2 Nov 2000 13:16:35 +0530



Just looking at the Via header host list to see if this is a loop is not
enough.
You need to look into the branchid to make sure that the current request
URI is the same as the original one (which
is reflected in the branchid computation). If a same message reaches the
same proxy, but with a different request-URI,
it should be processed normally by the proxy and not considered as a loop.
(this is called a sprial loop)

For an example of when a spiral might occur, Im attaching a small except
from the "via loop detection and proxies" thread
that was discussed a while ago:


jdrosen@dynamicsoft.com wrote:
"Its quite problematic for forwarding services. Lets say I forward calls
from jdrosen@lucent.com to my secretary, secretary@lucent.com. You
actually want the proxy to loop the request to itself in this case. As
you say, the request URI is different, so its not really a loop; its a
"spiral"."


james> Is the hash is easy to
james> implement? Could anyone give me advice on this?

MD5 alogrithm is simple to implement and I think MD5 is available in Public
domain too (?).
one way to make a hash is to simply strcat From+To+CallId+CSeq+ReqUri and
pass it to MD5Encode
which will return you a digest value that you can use as the one of the
branchi components.


Regds
Arjun

--
Arjun Roychowdhury @ Hughes Software Systems






james jack <sipjames@yahoo.com> on 11/02/2000 11:10:38 AM

To:   SIP@lists.bell-labs.com
cc:

Subject:  [SIP] Discussion




Dear All:
    I have one question to discuss. In order to
detect loop, RFC2543bis02 page 81 says"A request has
been looped if the server finds its own address in the
Via header field and the hash computation over the
fields enumerated in Section 6.46.6 yields the same
value as the hash part of the ??branch?? parameter in
the Via entry containing the proxy server??s
address.",
Page 59"In order to be able to both detect
loops and associate responses with the corresponding
request, the parameter SHOULD consist of two parts
separable by the implementation. One part, used for
loop detection (Section 12.3.1), MAY be computed
as a cryptographic hash of the To, From, Call-ID
header fields, the Request-URI of the request received
(before translation) and the sequence number from the
CSeq header field. The algorithm used to compute the
hash is implementation-dependent, but MD5 [38],
expressed in hexadecimal, is a reasonable choice.".
According to my knowledge, it is necessary to detect
the Via to find whether loop or not, why use these
parameters to detect loop, Is the hash is easy to
implement? Could anyone give me advice on this?
   Thanks a lot!
   Best regards!
                        Yours: James


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Sandeep Sahai wrote:
> 
> Hi,
> 
> Can any one tell me what is the sip-message for OPTION look like.
> 
> I was trying to find the same in the call flow diagram document , but 
> I cannot see an example for an OPTION sip message.
> 
> So, please let me an example of sip-message for OPTION.

It is in the bis (Section 16, I think).

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
Internet Software & eServices Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-413
Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184

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Date: Wed, 01 Nov 2000 12:09:17 -0500

--> Duncan Thomson writes:
>RFC 2543 (SIP) references the session announcement protocol (SAP), which exists only as a 1996 Int
>ernet Draft, that is, a "work in progress".  So is SAP actually a "work in progress"?  If so, why 
>hasn't anything been published on it since the 1996 Internet Draft?  Or is it considered obsolete?

It was recently published as RFC 2974.

Colin

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Date: Wed, 1 Nov 2000 16:46:33 -0500

To all - 

The registration from ITU-T SG8 for IANA as related to SDP was related to
the 
Approval last February of Annex D to T.38, which defines how to use SIP/SDP 
with T.38 for facsimile session control.   T.38 needs to support both UDP
and TCP modes of operation, hence the need for the SDP extension to support
TCP (Glenn Parsons of Nortel was the editor).    

So, the methods for using TCP within SDP for T.38 session control purposes
have been  described in detail in the T.38 Annex D.   

So, I'd suggest that people who want to write ID's related to this, take a
look at what is in the T.38 Annex D (TIES users can read it as TD262rev2 in
the SG8 informal FTP area).   There is also a small amendment to T.38 annex
D (see Com8-114) which will be reviewed at the SG16 meeting during the week
of Nov 13-17. 

I just checked on the ITU site and notice that T.38 Amendment 2 is not
published yet, so the TD262rev2 above is the only access to that text at the
moment.  Hopefully, this will change soon.   

James

* -----------------------------------
James Rafferty
Senior Product Manager, IP Telephony
jraff@brooktrout.com

Brooktrout Technology
410 First Avenue
Needham, MA 02494 USA
Phone:  +1-781-433-9462
Fax: 781-433-9268
www.brooktrout.com
Your Hook into the New Network(SM)    

> -----Original Message-----
> From: David Yon [mailto:yon@dialout.net]
> Sent: Monday, October 30, 2000 3:54 PM
> To: Jean-Francois Mule; ietf-fax@imc.org
> Cc: sip@lists.bell-labs.com
> Subject: Re: [SIP] RE: Request for information on using T.30/T.37/T.38
> with SIP/SDP/ RTP
> 
> 
> I know that the I-D specifically defers the discussion of 
> media transport 
> using TCP rather than UDPTL, but at the same time it also 
> refers to a new 
> token "TCP" to be registered with IANA for use in SDP.  I am 
> of the opinion 
> that simply specifying "TCP" is insufficient to describe how 
> TCP-based 
> Media Transport is to be negotiated between two endpoints.
> 
> I've submitted an Internet Draft for the MMUSIC group which 
> will hopefully 
> be available in the IETF I-D area soon, but for the moment it 
> can be read here:
> 
ftp://ftp.dialout.net/drafts/draft-ietf-mmusic-sdp-tcpmedia-00.txt

So, in regards to the SIP/T.38 I-D, my only comment would be that it take 
into account the I-D on SDP/TCP.  Or at the very least, participate with 
better defining the SDP/TCP I-D such that is meets the needs of SIP/T.38 
when using TCP for Media Transport.  As such, any comments on the I-D above 
would be greatly appreciated.

Again, hopefully the draft will be in the IETF area soon, I don't know why 
it is taking so long to work through the queue.

At 02:26 PM 10/27/00 -0700, Jean-Francois Mule wrote:
>There is an Internet-Draft on SIP and T.38 fax calls at
>http://search.ietf.org/internet-drafts/draft-mule-sip-t38callflows-00.txt
>It deals with how to negotiate a t.38 fax calls using sip as a signaling
>protocol (including switch over from audio/rtp to data/t38), the SDP
>attributes that itu-t sg-8 has registered with IANA.  Comments are
>appreciated on this Internet-Draft.
>
>


David Yon
Chief Technical Officer
Dialout.Net, Inc.
yon@dialout.net

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Subject: [SIP] Working Group Last Call on draft-ietf-sip-session-timer-03.txt
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Date: Wed, 1 Nov 2000 17:05:11 -0500

We would like to announce a working group last call on
      draft-ietf-sip-session-timer-03.txt
Last Call will end in two weeks.  If you have any comments or 
concerns on this draft, or do not believe it is ready to be 
sent to the IESG for promotion to a standards track RFC, please
send your comments to the list now

Thank you
  Brian Rosen
  Dean Willis
  Joerg Ott

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Subject: [SIP] Working Group Last Call on draft-ietf-sip-session-timer-03.txt
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Date: Thu, 2 Nov 2000 09:03:58 -0500

We would like to announce a working group last call on
      draft-ietf-sip-session-timer-03.txt
Last Call will end in two weeks.  If you have any comments or 
concerns on this draft, or do not believe it is ready to be 
sent to the IESG for promotion to a standards track RFC, please
send your comments to the list now

Thank you
  Brian Rosen
  Dean Willis
  Joerg Ott

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From: "Robin Escolme" <Robin.Escolme@marconi.com>
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Date: Thu, 2 Nov 2000 15:44:48 +0000



Hi,

Some comments and questions :

   Section 5.2, CASE I - UAC supports session timers. The text states "In this
   case the proxy remembers that it inserted a Session-Expires header into the
   request and that the UAC supported sesion timer". The proxy may not have
   inserted a Session-Expires header - it would only have done so if the there
   wasn't one in the request from the UAC.
   CASE II - UAC Doesn't support, UAS doesn't support. There is, presumably,
   nothing to stop the proxy, if its policy dictates, from running a local
   session timer, the expiration of which would result in it de-allocating any
   resources - and in particular, closing any pin-holes opened across a
   firewall, with no indication at all to the end points. Whilst this isn't very
   nice for the end points, since their RTP packets would suddenly cease, the
   proxy is trying to protect it's network resources from unreliable end-points.
   The session timer used be longer that normal, given there is no opportunity
   to refresh. Alternatively, cannot the proxy issue a re-INVITE to each UA,
   using the same SDP used orginally?
   Section 6, Para 2 states "If the the UAS places a Session-Expires header into
   the response, and the request contained a Supported header with the value
   "timer",.......". This implies that the UAS may place a Session-Espires
   header into the response when the request did not contain a Supported header
   with the value "timer". However,  the table further down says that if the
   request contained neither a Supported header with the value "timer", nor a
   Session-Expires header, there there will be no timers. Surely, if the UAS
   supports session timers it can place a Session-Expires header in the
   response, so that it indicates to proxies that it is running a session timer.
   If this is the case, then the behavoir in the foirst row of this table should
   indicate that the UAS MAY add a session-Expires header to the response. This
   then equates to CASE III in section 5.2.
   Section 6, last para suggests that a 200 OK in response to a re-INVITE may
   not contain a Session-Expire header. How can this be, unless a proxy or the
   UAC has removed it, which I thought was not allowed?
   Section 8.5 - call flow for re-INVITE. The Calling UA send the INVITE to the
   SPS with a Session-Expires header. The note against the next message, INVITE
   from SPS to Called UA says the SPS adds the S-E header, but it was already
   there.
   General - ther terms UAC and UAS I have taken to be synonymous with Calling
   UAC and Called UAC, respectively. The document makes use of both terms at
   varying times. Are these terms interchangeable or is there some implied
   difference.
   Section 9 of RFC2543bis02 lists abbreviated forms for common header fields.
   The session timer feature is valuable and may be insisted upon by many
   networks, bringing the Session Expires header into common use. You should
   consider an abbreviated form.

Hope these make sense.

Robin Escolme

Marconi Communications



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Date: Thu, 02 Nov 2000 13:48:54 -0500
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Here is a straightforward question regarding the examples in the RFC and
sip call flows documents.

 Are the  Content-Length values computed on the basis of CR-LF  (2
characters) for  newline? (i.e. DOS vs UNIX convention)?

Thank you  in advance for your replies.

--
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
Tel: 301 975 3664 Fax: 301 590 0932


ÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÿÒ ùšŠX§‚X¬µ"şX¬¶ÏÛzYÿ•¦ìıÊ&†ÛiÿÿåŠËlı·¥—ùZnÏÜ¢oæj)fjåŠËbú?


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Date: Thu, 02 Nov 2000 10:59:52 -0800

This is a multi-part message in MIME format.
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Greetings:

"M. Ranganathan" wrote:

> Here is a straightforward question regarding the examples in the RFC and
> sip call flows documents.
>
>  Are the  Content-Length values computed on the basis of CR-LF  (2
> characters) for  newline? (i.e. DOS vs UNIX convention)?

Yes.

Regards,

Bobby.Sardana@mobilerain.com

>
>
> Thank you  in advance for your replies.
>
> --
> M.Ranganathan
> NIST Advanced Networking Technologies Group,
> 100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
> Tel: 301 975 3664 Fax: 301 590 0932
>
> Hfæj)b? b²Ô^>X¬¶ÆŞ-YZnÇ(sm§ÿåSËlmée*¦ìr?¿?¨¥?©ÿ-+-SwèşÈ©

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begin:vcard 
n:Sardana;Bobby
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--------------A89B8CF0BA7020851A790200--


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Date: Thu, 2 Nov 2000 13:57:54 -0500

> Here is a straightforward question regarding the examples in 
> the RFC and
> sip call flows documents.
> 
>  Are the  Content-Length values computed on the basis of CR-LF  (2
> characters) for  newline? (i.e. DOS vs UNIX convention)?

No. Content-Length is computed based on the real length of the payload,
whether it uses CRLF, LF, CR or anything else to terminate lines.

Thank you,
Igor Slepchin

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One very basic comment is that the current RFC 2806 does not follow the
normal URI syntax description rules. This is very confusing, as it leads
to duplicating the description of escaping, with different rules and, as
witnessed on this list a few months ago, weeks of confusion as the two
"worlds" are mentally merged. I would strongly recommend that the ABNF
be revised to conform to the standard way of specifying URL content.



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IANA has agreed to register SIP header field names and compact forms to
minimize the chance for collisions. I've put together an initial list
for RFC2543 and draft-ietf-sip-rfc2543bis, session timers and 100rel,
but headers from other drafts need to be added. The current list is at
http://www.cs.columbia.edu/sip/headers.txt.

Please send me corrections and additions.

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Date: Thu, 2 Nov 2000 12:17:08 -0800

Hi,

Sectoin 4, UAC behavior states :

"Note that it is possible that the calling UA is generating
refreshes, and then it receives a re-INVITE. After following
the rules for UAS described below, the calling UA now determines
it is not supposed to generate refreshes..........."

Is their a specific reason for allowing the master switchover
for generating the re-INVITES? After the initial negotation, if
the UAC was responsible for generating re-INVITES, why would UAS
in the middle of a session like to take control? 

Can we add an Example flow for this scenario in section 8 of the
draft. 

-Thanks,
Manish Singh

-----Original Message-----
From: Robin Escolme [mailto:Robin.Escolme@marconi.com]
Sent: Thursday, November 02, 2000 7:45 AM
To: Jonathan Rosenberg
Cc: sip@lists.bell-labs.com
Subject: Re: [SIP] I-D ACTION:draft-ietf-sip-session-timer-03.txt




Hi,

Some comments and questions :

   Section 5.2, CASE I - UAC supports session timers. The text states "In
this
   case the proxy remembers that it inserted a Session-Expires header into
the
   request and that the UAC supported sesion timer". The proxy may not have
   inserted a Session-Expires header - it would only have done so if the
there
   wasn't one in the request from the UAC.
   CASE II - UAC Doesn't support, UAS doesn't support. There is, presumably,
   nothing to stop the proxy, if its policy dictates, from running a local
   session timer, the expiration of which would result in it de-allocating
any
   resources - and in particular, closing any pin-holes opened across a
   firewall, with no indication at all to the end points. Whilst this isn't
very
   nice for the end points, since their RTP packets would suddenly cease,
the
   proxy is trying to protect it's network resources from unreliable
end-points.
   The session timer used be longer that normal, given there is no
opportunity
   to refresh. Alternatively, cannot the proxy issue a re-INVITE to each UA,
   using the same SDP used orginally?
   Section 6, Para 2 states "If the the UAS places a Session-Expires header
into
   the response, and the request contained a Supported header with the value
   "timer",.......". This implies that the UAS may place a Session-Espires
   header into the response when the request did not contain a Supported
header
   with the value "timer". However,  the table further down says that if the
   request contained neither a Supported header with the value "timer", nor
a
   Session-Expires header, there there will be no timers. Surely, if the UAS
   supports session timers it can place a Session-Expires header in the
   response, so that it indicates to proxies that it is running a session
timer.
   If this is the case, then the behavoir in the foirst row of this table
should
   indicate that the UAS MAY add a session-Expires header to the response.
This
   then equates to CASE III in section 5.2.
   Section 6, last para suggests that a 200 OK in response to a re-INVITE
may
   not contain a Session-Expire header. How can this be, unless a proxy or
the
   UAC has removed it, which I thought was not allowed?
   Section 8.5 - call flow for re-INVITE. The Calling UA send the INVITE to
the
   SPS with a Session-Expires header. The note against the next message,
INVITE
   from SPS to Called UA says the SPS adds the S-E header, but it was
already
   there.
   General - ther terms UAC and UAS I have taken to be synonymous with
Calling
   UAC and Called UAC, respectively. The document makes use of both terms at
   varying times. Are these terms interchangeable or is there some implied
   difference.
   Section 9 of RFC2543bis02 lists abbreviated forms for common header
fields.
   The session timer feature is valuable and may be insisted upon by many
   networks, bringing the Session Expires header into common use. You should
   consider an abbreviated form.

Hope these make sense.

Robin Escolme

Marconi Communications



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Hi,

  Just wanted to know whether you need to have root access in order to install the sip server ?

  Any help will be greatly appreciated.

Thanks
T.Singh


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Use SMTP to deliver e-mail to the UA (push), and use SIP REGISTER to update the UA's
address in the final MTA.

Send a REGISTER message with

To: mailto:simon@firetalk.com
Contact: mailto:a.workstation.firetalk.com:2000

Simon.


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Date: Fri, 03 Nov 2000 08:35:47 +0100
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Does this mean open season on compact forms? I really hope not.
In fact, I wish they would go away altogether. I don't want to re-open
this debate, but I would like to dam the flood of new compact forms.

Am I the only one who feels this way?
/sean

Henning Schulzrinne wrote:

> IANA has agreed to register SIP header field names and compact forms to
> minimize the chance for collisions. I've put together an initial list
> for RFC2543 and draft-ietf-sip-rfc2543bis, session timers and 100rel,
> but headers from other drafts need to be added. The current list is at
> http://www.cs.columbia.edu/sip/headers.txt.
>
> Please send me corrections and additions.
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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        archow@hss.hns.com
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Date: Fri, 3 Nov 2000 02:39:21 -0500

All servers (UAS, registrar) should answer a OPTIONS with a 200 OK with the
following headers added to the response: Allow, listing the supported
methods, Accept, listing the support MIME types, and Supported, listing the
supported extensions. A UAS could also insert SDP listing its codecs. How
such SDP is formatted has never been defined, and to my knowledge, no one is
yet doing this. We'll define something in bis.

As an example for a registrar response to OPTIONS:

SIP/2.0 200 OK
From: sip:querier@company.com
To: sip:registardomain.com;tag=9378asd0asd7asd
Call-ID: a90sda08shd@1.2.3.4
CSeq: 1 REGISTER
Via: SIP/2.0/UDP host.company.com
Accept: application/cpl+xml
Allow: OPTIONS, REGISTER, CANCEL
Supported: pref
Content-Length: 0


This registar supports uploads of CPL, and also the caller preferences
extension.

-Jonathan R.



---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Thys Nel [mailto:thys@parsec.co.za]
> Sent: Thursday, November 02, 2000 1:48 AM
> To: 'Billy Biggs'; archow@hss.hns.com
> Cc: 'Sandeep Sahai'; 'SIP List'
> Subject: RE: [SIP] Sip- Message OPTION
> 
> 
> Does anyone have examples of messages flows with OPTIONS 
> processed by a
> registrar server? The only example in the RFC is between 2 
> user agents, but
> the RFC states that a registrar must also support it. I can't 
> seem to find
> any description of when a registrar should respond to an 
> OPTIONS request.
> 
> Thys
> 
> > -----Original Message-----
> > From: sip-admin@lists.bell-labs.com
> > [mailto:sip-admin@lists.bell-labs.com]On Behalf Of Billy Biggs
> > Sent: 02 November 2000 8:17
> > To: archow@hss.hns.com
> > Cc: Sandeep Sahai; SIP List
> > Subject: Re: [SIP] Sip- Message OPTION
> >
> >
> >   Note that this should be cleaned up.  There is no Via or 
> CSeq in the
> > response, the CSeq in the request starts at 1, the From has
> > no tag, and there
> > is no Content-Length.
> >
> > > C->S: OPTIONS sip:bob@example.com SIP/2.0
> > > Via: SIP/2.0/UDP cat.wonderland.com
> > > From: Alice <sip:alice@wonderland.com>
> > > To: Bob <sip:bob@example.com>
> > > Call-ID: 6378@cat.wonderland.com
> > > CSeq: 1 OPTIONS
> > > Accept: application/sdp
> > >
> > > S->C: SIP/2.0 200 OK
> > > From: Alice <sip:alice@wonderland.com>
> > > To: Bob <sip:bob@example.com> ;tag=376364382
> > > Call-ID: 6378@cat.wonderland.com
> > > Content-Length: 81
> > > Content-Type: application/sdp
> > >
> > > v=0
> > > o=alice 3149329138 3149329165 IN IP4 24.124.37.3
> > > s=Security problems
> > > t=3149328650 0
> > > c=IN IP4 24.124.37.3
> > > m=audio 0 RTP/AVP 0 1 3 99
> > > a=rtpmap:0 PCMU/8000
> > > a=rtpmap:1 1016/8000
> > > a=rtpmap:3 GSM/8000
> > > a=rtpmap:99 SX7300/8000
> > > m=video 0 RTP/AVP 31 34
> > > a=rtpmap:31 H261/90000
> > > a=rtpmap:34 H263/90000
> >
> > --
> > Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> > http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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Date: Fri, 3 Nov 2000 02:43:52 -0500



 

> -----Original Message-----
> From: Igor Slepchin [mailto:ISlepchin@dynamicsoft.com]
> Sent: Thursday, November 02, 2000 1:58 PM
> To: 'mranga@nist.gov'; sip@lists.bell-labs.com
> Subject: RE: [SIP] Content-Length values in RFC and call 
> flows examples
> 
> 
> > Here is a straightforward question regarding the examples in 
> > the RFC and
> > sip call flows documents.
> > 
> >  Are the  Content-Length values computed on the basis of CR-LF  (2
> > characters) for  newline? (i.e. DOS vs UNIX convention)?
> 
> No. Content-Length is computed based on the real length of 
> the payload,
> whether it uses CRLF, LF, CR or anything else to terminate lines.

Correct. Whatever the number of bytes are in the body, thats the
content-length. The computation is independent of what the bytes represent.
So, CR-LF is two bytes, CR is one, LF is one.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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Date: Fri, 3 Nov 2000 02:51:53 -0500



 

> -----Original Message-----
> From: Singh, Manish [mailto:manishs@trillium.com]
> Sent: Thursday, November 02, 2000 3:17 PM
> To: 'Robin Escolme'; Jonathan Rosenberg
> Cc: sip@lists.bell-labs.com
> Subject: RE: [SIP] I-D ACTION:draft-ietf-sip-session-timer-03.txt
> 
> 
> Hi,
> 
> Sectoin 4, UAC behavior states :
> 
> "Note that it is possible that the calling UA is generating
> refreshes, and then it receives a re-INVITE. After following
> the rules for UAS described below, the calling UA now determines
> it is not supposed to generate refreshes..........."
> 
> Is their a specific reason for allowing the master switchover
> for generating the re-INVITES? After the initial negotation, if
> the UAC was responsible for generating re-INVITES, why would UAS
> in the middle of a session like to take control? 

The idea was just to make the whole process stateless, so that processing of
a re-INVITE is absolutely the same as processing the initial INVITE. As a
result of this choice, the roles will change over on re-INVITEs depending on
what level of support each side has. If each side both supports session
timer, the user sending the re-INVITE will become the one performing
refreshes if the re-invite completes successfully.

The stateless operation is needed, for example, to recover from crash-reboot
cycles.

I think this actually makes the implementation simpler, not more complex.

> 
> Can we add an Example flow for this scenario in section 8 of the
> draft. 

I will add text explaining the above, and also an example flow.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Subject: [SIP] More Comments on 2806bis-00
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Date: Fri, 03 Nov 2000 13:52:25 +0100
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Another minor comment on RFC2806. The private-prefix syntax
is defined as below.

private-prefix        = (%x21-22 / %x24-27 / %x2C / %x2F / %x3A /
                        %x3C-40 / %x45-4F / %x51-56 / %x58-60 /
                        %x65-6F / %x71-76 / %x78-7E)
                        *(%x21-3A / %x3C-7E)
                        ; Characters in URLs must follow escaping rules
                        ; as explained in [RFC2396]


Missing from this private-prefix are the following US-ASCII letters.

I am assuming these are missing because they are valid
dtmf-digit characters and you want to avoid ambiguity with the
local-prefix (the global-prefix is unambiguous because of the
leading '+')

%x41 = 'A'
%x42 = 'B'
%x43 = 'C'
%x44 = 'D'
%x61 = 'a'
%x62 = 'b'
%x63 = 'c'
%x64 = 'd'

These likewise would have been omitted because of the
conflict with the 'pause' and 'wait-for-dial-tone'
that can also occur in the local-prefix.

%x50 = 'P'
%x57 = 'W'
%x70 = 'p'
%x77 = 'w'

This unfortunately is a bit limiting for the private prefix.
My suggestion is that rather than omitting these from the
private-prefix, that you require the private-prefix to
start with a special prefix as with the global-prefix.
For this purpose I would recommend "_" (%x5F).

The ABNF for private-prefix would then become:

private-prefix = "_" 1*(token-char)

This would be unambiguos and more flexible.
The exact prefix to the private-prefix is not important.
It could, for instance, also be "X-"|"x-". What is important
is that the letters a-d, p, and w are not excluded.

Regards,
Sean

--
Sean Olson <sean.olson@ericsson.com>




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Date: Fri, 3 Nov 2000 08:16:34 -0500 (EST)

Following are the headers, methods, and option tags defined
in the dcsgroup drafts.  The draft names reflect the
new names under which they will be submitted in the next
week or so.

Bill Marshall
wtm@research.att.com

---Headers---
Anonymity			draft-ietf-sip-privacy-00
Dcs-Billing-ID			draft-dcsgroup-sip-proxy-proxy-03
Dcs-Billing-Info		draft-dcsgroup-sip-proxy-proxy-03
Dcs-Gate			draft-dcsgroup-sip-proxy-proxy-03
Dcs-Laes			draft-dcsgroup-sip-proxy-proxy-03
Dcs-OSPS			draft-dcsgroup-sip-proxy-proxy-03
Dcs-Redirect			draft-dcsgroup-sip-proxy-proxy-03
Dcs-Trace-Party-ID		draft-dcsgroup-sip-proxy-proxy-03
Media-Authorization		draft-ietf-sip-call-auth-00
Remote-Party-ID			draft-ietf-sip-privacy-00
State				draft-ietf-sip-state-00

---Methods---
COMET				draft-ietf-sip-manyfolks-resource-00

---Option tags---
dcs				draft-dcsgroup-sip-proxy-proxy-03
privacy				draft-ietf-sip-privacy-00
state				draft-ietf-sip-state-00

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Date: Fri, 03 Nov 2000 13:33:45 +0000
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The way the UAC builds the Route on a final 200 response 
in 3.1.2 of the call flows draft doesn't look quite 
right ...

3.1.2 Successful SIP to SIP through two proxies

   User A          Proxy 1          Proxy 2          User B
     |                |                |                |
     |   INVITE F1    |                |                |
     |--------------->|                |                |
     |     407 F2     |                |                |
     |<---------------|                |                |
     |     ACK F3     |                |                |
     |--------------->|                |                |
     |   INVITE F4    |                |                |
     |--------------->|   INVITE F5    |                |
     |    (100) F6    |--------------->|   INVITE F7    |
     |<---------------|    (100) F8    |--------------->|
     |                |<---------------|                |
     |                |                |     180 F9     |
     |                |    180 F10     |<---------------|
     |     180 F11    |<---------------|                |
     |<---------------|                |     200 F12    |
     |                |    200 F13     |<---------------|
     |     200 F14    |<---------------|                |
     |<---------------|                |                |
     |     ACK F15    |                |                |
     |--------------->|    ACK F16     |                |
     |                |--------------->|     ACK F17    |
     |                |                |--------------->|

F14 200 OK Proxy 1 -> A

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP here.com:5060
   Record-Route: <sip:UserB@there.com;maddr=ss2.wcom.com>,
    <sip:UserB@there.com;maddr=ss1.wcom.com>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: 134

   v=0
   o=UserB 2890844527 2890844527 IN IP4 there.com
   s=Session SDP
   c=IN IP4 110.111.112.113
   t=0 0
   m=audio 3456 RTP/AVP 0
   a=rtpmap:0 PCMU/8000


   F15 ACK A -> Proxy 1

   ACK sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP here.com:5060
   Route: <sip:UserB@there.com;maddr=ss2.wcom.com>,
    <sip:UserB@there.com;maddr=110.111.112.113>
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345601@here.com
   CSeq: 1 ACK
   Content-Length: 0

The UAC should build the Route header in the ACK by 
copying the 200 response Record-Route header, 
reversing the order and then adding the Contact as 
the last entry. (Note the 1st entry in the Route is
subsequently removed when the client actually sends F15).
But in this case the last entry has become an IP address. 
Shouldn't the UAC just copy the Contact and not try 
and do any resolution of it?

Cheers,
Neil.
-- 
Ubiquity Software Corporation, UK        http://www.ubiquity.net

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From sip-admin@lists.bell-labs.com  Fri Nov  3 09:37:04 2000
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Definitely not, I would also like them to disappear.
Atleast not creating new ones. The gain compared to
the rest of the message contents are not that much.

To write 't:' instead of 'to:' or 'v:' instead of 'via:'
doesn't really help much. I don't want to restart this 
discussion either only give indication that you are
not the only one. I wonder if there are anyone at all
using the compact form. 

/Bertil

Sean Olson wrote:
> 
> Does this mean open season on compact forms? I really hope not.
> In fact, I wish they would go away altogether. I don't want to re-open
> this debate, but I would like to dam the flood of new compact forms.
> 
> Am I the only one who feels this way?
> /sean
> 
> Henning Schulzrinne wrote:
> 
> > IANA has agreed to register SIP header field names and compact forms to
> > minimize the chance for collisions. I've put together an initial list
> > for RFC2543 and draft-ietf-sip-rfc2543bis, session timers and 100rel,
> > but headers from other drafts need to be added. The current list is at
> > http://www.cs.columbia.edu/sip/headers.txt.
> >
> > Please send me corrections and additions.
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Bertil Engelholm
AXE Research and Development        voice : +46 8 727 3499
SIP Security                        Fax   : +46 8 647 8276
S-126 25 Stockholm Sweden           E-mail:
Bertil.Engelholm@uab.ericsson.se

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From sip-admin@lists.bell-labs.com  Fri Nov  3 10:28:04 2000
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From: "Robert Sparks" <rsparks@dynamicsoft.com>
To: <SIP@lists.bell-labs.com>
Subject: compact headers (was RE: [SIP] IANA SIP header registration)
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Currently, draft-ietf-sip-guidelines-00 says

   Extensions that define new headers SHOULD define a compact form
   representation if the non-compact header is more than four
   characters. Header names SHOULD use ASCII characters. Header names
   are always case insensitive. Header values are generally case
   sensitive, with the exception of domain names which MUST be case
   insensitive.

RjS

> -----Original Message-----
> From: sip-admin@lists.bell-labs.com
> [mailto:sip-admin@lists.bell-labs.com]On Behalf Of Bertil Engelholm
> Sent: Friday, November 03, 2000 2:33 AM
> To: SIP@lists.bell-labs.com
> Subject: Re: [SIP] IANA SIP header registration
>
>
> Definitely not, I would also like them to disappear.
> Atleast not creating new ones. The gain compared to
> the rest of the message contents are not that much.
>
> To write 't:' instead of 'to:' or 'v:' instead of 'via:'
> doesn't really help much. I don't want to restart this
> discussion either only give indication that you are
> not the only one. I wonder if there are anyone at all
> using the compact form.
>
> /Bertil
>
> Sean Olson wrote:
> >
> > Does this mean open season on compact forms? I really hope not.
> > In fact, I wish they would go away altogether. I don't want to re-open
> > this debate, but I would like to dam the flood of new compact forms.
> >
> > Am I the only one who feels this way?
> > /sean
> >
> > Henning Schulzrinne wrote:
> >
> > > IANA has agreed to register SIP header field names and
> compact forms to
> > > minimize the chance for collisions. I've put together an initial list
> > > for RFC2543 and draft-ietf-sip-rfc2543bis, session timers and 100rel,
> > > but headers from other drafts need to be added. The current list is at
> > > http://www.cs.columbia.edu/sip/headers.txt.
> > >
> > > Please send me corrections and additions.
> > >
> > > _______________________________________________
> > > SIP mailing list
> > > SIP@lists.bell-labs.com
> > > http://lists.bell-labs.com/mailman/listinfo/sip
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
>
> --
> Bertil Engelholm
> AXE Research and Development        voice : +46 8 727 3499
> SIP Security                        Fax   : +46 8 647 8276
> S-126 25 Stockholm Sweden           E-mail:
> Bertil.Engelholm@uab.ericsson.se
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>


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From sip-admin@lists.bell-labs.com  Fri Nov  3 10:46:02 2000
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Thanks for pointing this out. What is the guideline
for when we exhaust the 26 single-letter forms?
Do we go to two-letter forms, three-letter forms, etc.?
What is the point of diminishing returns here?

/sean

Robert Sparks wrote:

> Currently, draft-ietf-sip-guidelines-00 says
>
>    Extensions that define new headers SHOULD define a compact form
>    representation if the non-compact header is more than four
>    characters. Header names SHOULD use ASCII characters. Header names
>    are always case insensitive. Header values are generally case
>    sensitive, with the exception of domain names which MUST be case
>    insensitive.
>
> RjS
>
> > -----Original Message-----
> > From: sip-admin@lists.bell-labs.com
> > [mailto:sip-admin@lists.bell-labs.com]On Behalf Of Bertil Engelholm
> > Sent: Friday, November 03, 2000 2:33 AM
> > To: SIP@lists.bell-labs.com
> > Subject: Re: [SIP] IANA SIP header registration
> >
> >
> > Definitely not, I would also like them to disappear.
> > Atleast not creating new ones. The gain compared to
> > the rest of the message contents are not that much.
> >
> > To write 't:' instead of 'to:' or 'v:' instead of 'via:'
> > doesn't really help much. I don't want to restart this
> > discussion either only give indication that you are
> > not the only one. I wonder if there are anyone at all
> > using the compact form.
> >
> > /Bertil
> >
> > Sean Olson wrote:
> > >
> > > Does this mean open season on compact forms? I really hope not.
> > > In fact, I wish they would go away altogether. I don't want to re-open
> > > this debate, but I would like to dam the flood of new compact forms.
> > >
> > > Am I the only one who feels this way?
> > > /sean
> > >
> > > Henning Schulzrinne wrote:
> > >
> > > > IANA has agreed to register SIP header field names and
> > compact forms to
> > > > minimize the chance for collisions. I've put together an initial list
> > > > for RFC2543 and draft-ietf-sip-rfc2543bis, session timers and 100rel,
> > > > but headers from other drafts need to be added. The current list is at
> > > > http://www.cs.columbia.edu/sip/headers.txt.
> > > >
> > > > Please send me corrections and additions.
> > > >
> > > > _______________________________________________
> > > > SIP mailing list
> > > > SIP@lists.bell-labs.com
> > > > http://lists.bell-labs.com/mailman/listinfo/sip
> > >
> > > _______________________________________________
> > > SIP mailing list
> > > SIP@lists.bell-labs.com
> > > http://lists.bell-labs.com/mailman/listinfo/sip
> >
> > --
> > Bertil Engelholm
> > AXE Research and Development        voice : +46 8 727 3499
> > SIP Security                        Fax   : +46 8 647 8276
> > S-126 25 Stockholm Sweden           E-mail:
> > Bertil.Engelholm@uab.ericsson.se
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> >
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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Hi,

Sofar noone have responded to the problem below so this
third time I send it I try a new subject. I was hoping 
that some SIP responsible people would come up with a 
solution to the problem or atleast say if it is regarded 
as a problem or not.

Since noone have given any solutions I'll give my view 
of what I beleave could be done.

The first solution is to remove the text where it says
that a UA should fill all Routes with values from
CONTACT or FROM. Instead the original addresses should
be kept. Personally I find it strange when you combine
addresses (from CONTACT/FROM) with a port number from
RECORD-ROUTE. This means that the total address now is 
completely useless. It looks to me that you try to 
avoid syntax changes in the protocol at all costs. I
don't understand why a change in semantics is regarded 
as more backwards compatible than a change in the syntax ?

I would have prefered to add a port number in the
maddr parameter because that's where I beleave it belongs. 

The second solution is to make it possible for a server
to add it's own parameter in RECORD-ROUTE and then get
it back in the REQUEST. Today a UAS only supposed to 
copy maddr and port value from RECORD-ROUTE to ROUTE.
If all parameters was copied instead it would be possible
for a server to add anything it likes in RECORD-ROUTE
and get it back again. I beleave this kind of solution 
have been discussed for states ?

/Bertil

Bertil Engelholm wrote:
> 
> Hi again,
> 
> I'm sorry to bring this up again but I just can't get things
> to work in the correct way. I have read through the old threads
> regarding this matter and still don't understand how it is
> suppose to work.
> 
> It's this example again :
> 
>   ---------                  --------                  --------
>   | a@c1  | -- INV u1@p1 --> | (s1) | -- INV u2@p2 --> |      |
>   ---------  (contact a@c1)  |      |                  |      |
>                              |  p1  |                  |  p2  |
>   ---------                  |      |                  |      |
>   | b@c2  | <- INV b@c2 ---- | (s2) | <- INV u3@p1 --- |      |
>   ---------                  --------                  --------
> 
> If P1 is stateful it might allocate one "call" state machine for
> each path the "call" takes in P1 (s1 and s2 in the picture).
> These two state machines are identified by the REQUEST-URI
> that came in the two INVITEs since that's the only thing that
> differs the two from each other. (topmost VIA also differs,
> more about that later)
> 
> P1 will now include the REQUEST-URI in the branch value it
> generates so that the responses can be directed to the right
> state machine when they arrive. And when the ACK is sent the
> original REQUEST-URI from ROUTE is used. So no problems here.
> 
> The problem I have is when b@C2 sends a new request (eg. BYE).
> According to the specification b should/must/may (or whatever)
> overwrite the REQUEST-URI in the ROUTE headers using the
> REQUEST-URI from CONTACT or FROM.
> 
> So how should P1 now be able to differ the two (BYE) requests
> from each other ? The Original REQUEST-URI was the only thing
> that P1 had to identify the two state machines.
> 
> When reading the old threads regarding this matter I saw
> something about using topmost VIA in some way, but how ?
> 
> By using the topmost VIA you can see that they are two different
> requests but how do you identify which state machine should receive
> the first request. (Someone might argue that it doesn't matter but if
> P1 controls a firewall it DO matter. Maybe not for BYE but for a
> RE-INVITE it matters).
> 
> --
> Bertil Engelholm
> AXE Research and Development        voice : +46 8 727 3499
> SIP Security                        Fax   : +46 8 647 8276
> S-126 25 Stockholm Sweden           E-mail:
> Bertil.Engelholm@uab.ericsson.se
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Bertil Engelholm
AXE Research and Development        voice : +46 8 727 3499
SIP Security                        Fax   : +46 8 647 8276
S-126 25 Stockholm Sweden           E-mail:
Bertil.Engelholm@uab.ericsson.se

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Subject: Re: [SIP] Identification problem
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Bertil Engelholm wrote:

<snip>

>
> The second solution is to make it possible for a server
> to add it's own parameter in RECORD-ROUTE and then get
> it back in the REQUEST. Today a UAS only supposed to
> copy maddr and port value from RECORD-ROUTE to ROUTE.
> If all parameters was copied instead it would be possible
> for a server to add anything it likes in RECORD-ROUTE
> and get it back again. I beleave this kind of solution
> have been discussed for states ?
>
> /Bertil

I was under the impression that the Record-Route was supposed
to be copied in its entirety to Route: (with some modifications, but
no exclusion of parameters).  I don't see anything in section 6.34
about the exclusion of parameters other than the fact that proxies
should not include a transport parameter.

On the topic of states, I would prefer to see the use of the
State: header for this purpose since it is more generic.
You do point out one case where state/context information could
be useful.

/sean


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Subject: Re: [SIP] Identification problem
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I am afraid I don't see a problem with the spec. Unless I 
am missing something what you are talking about is a call 
stateful proxy not just a plain transaction stateful one. 
The transaction stateful one is OK as the bis-02 change to 
make the Via branch param globally unique differentiates 
the otherwise isomorphic BYEs. A call stateful proxy needs 
to do accounting beyond proxy transaction state, e.g.To/From 
tags. This accounting will also need to cover matching the 
right call state machine to the right BYE. For example on a 
reverse leg BYE the call state machine created 2nd will get 
the BYE 1st.

BTW the issues raised in the solutions have been raised 
before see the threads "Record-Route and maddr" and 
"transaction identification". Introducing the contemplated 
State header is a cleaner way of achieving the 2nd solution.

Cheers,
Neil
-- 
Ubiquity Software Corporation, UK        http://www.ubiquity.net

Bertil Engelholm wrote:
> 
> Hi,
> 
> Sofar noone have responded to the problem below so this
> third time I send it I try a new subject. I was hoping
> that some SIP responsible people would come up with a
> solution to the problem or atleast say if it is regarded
> as a problem or not.
> 
> Since noone have given any solutions I'll give my view
> of what I beleave could be done.
> 
> The first solution is to remove the text where it says
> that a UA should fill all Routes with values from
> CONTACT or FROM. Instead the original addresses should
> be kept. Personally I find it strange when you combine
> addresses (from CONTACT/FROM) with a port number from
> RECORD-ROUTE. This means that the total address now is
> completely useless. It looks to me that you try to
> avoid syntax changes in the protocol at all costs. I
> don't understand why a change in semantics is regarded
> as more backwards compatible than a change in the syntax ?
> 
> I would have prefered to add a port number in the
> maddr parameter because that's where I beleave it belongs.
> 
> The second solution is to make it possible for a server
> to add it's own parameter in RECORD-ROUTE and then get
> it back in the REQUEST. Today a UAS only supposed to
> copy maddr and port value from RECORD-ROUTE to ROUTE.
> If all parameters was copied instead it would be possible
> for a server to add anything it likes in RECORD-ROUTE
> and get it back again. I beleave this kind of solution
> have been discussed for states ?
> 
> /Bertil
> 
> Bertil Engelholm wrote:
> >
> > Hi again,
> >
> > I'm sorry to bring this up again but I just can't get things
> > to work in the correct way. I have read through the old threads
> > regarding this matter and still don't understand how it is
> > suppose to work.
> >
> > It's this example again :
> >
> >   ---------                  --------                  --------
> >   | a@c1  | -- INV u1@p1 --> | (s1) | -- INV u2@p2 --> |      |
> >   ---------  (contact a@c1)  |      |                  |      |
> >                              |  p1  |                  |  p2  |
> >   ---------                  |      |                  |      |
> >   | b@c2  | <- INV b@c2 ---- | (s2) | <- INV u3@p1 --- |      |
> >   ---------                  --------                  --------
> >
> > If P1 is stateful it might allocate one "call" state machine for
> > each path the "call" takes in P1 (s1 and s2 in the picture).
> > These two state machines are identified by the REQUEST-URI
> > that came in the two INVITEs since that's the only thing that
> > differs the two from each other. (topmost VIA also differs,
> > more about that later)
> >
> > P1 will now include the REQUEST-URI in the branch value it
> > generates so that the responses can be directed to the right
> > state machine when they arrive. And when the ACK is sent the
> > original REQUEST-URI from ROUTE is used. So no problems here.
> >
> > The problem I have is when b@C2 sends a new request (eg. BYE).
> > According to the specification b should/must/may (or whatever)
> > overwrite the REQUEST-URI in the ROUTE headers using the
> > REQUEST-URI from CONTACT or FROM.
> >
> > So how should P1 now be able to differ the two (BYE) requests
> > from each other ? The Original REQUEST-URI was the only thing
> > that P1 had to identify the two state machines.
> >
> > When reading the old threads regarding this matter I saw
> > something about using topmost VIA in some way, but how ?
> >
> > By using the topmost VIA you can see that they are two different
> > requests but how do you identify which state machine should receive
> > the first request. (Someone might argue that it doesn't matter but if
> > P1 controls a firewall it DO matter. Maybe not for BYE but for a
> > RE-INVITE it matters).
> >
> > --
> > Bertil Engelholm
> > AXE Research and Development        voice : +46 8 727 3499
> > SIP Security                        Fax   : +46 8 647 8276
> > S-126 25 Stockholm Sweden           E-mail:
> > Bertil.Engelholm@uab.ericsson.se
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 
> --
> Bertil Engelholm
> AXE Research and Development        voice : +46 8 727 3499
> SIP Security                        Fax   : +46 8 647 8276
> S-126 25 Stockholm Sweden           E-mail:
> Bertil.Engelholm@uab.ericsson.se
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

_______________________________________________
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Date: Fri, 3 Nov 2000 12:34:03 -0500



 

> -----Original Message-----
> From: Bertil Engelholm [mailto:Bertil.Engelholm@uab.ericsson.se]
> Sent: Friday, November 03, 2000 10:54 AM
> To: SIP@lists.bell-labs.com
> Subject: [SIP] Identification problem
> 
> 
> Hi,
> 
> Sofar noone have responded to the problem below so this
> third time I send it I try a new subject. I was hoping 
> that some SIP responsible people would come up with a 
> solution to the problem or atleast say if it is regarded 
> as a problem or not.

Well, back in the good old days, the SIP list generated just a few messages
a day. I used to be able to respond almost immediately to every question.
Now, as we approach the average of 30 messages per day, it is impossible for
me, and many of the other SIP experts, to respond to every question on the
list.

Anyway, to answer your question, I can think of a few additional
possibiltiies:

1. P1 use a different IP address (i.e., different maddr) as the request
cycles through each time. This requires detection of the spiral and
selection of a different address each time. This can be done statelessly,
since the previous maddr inserted will appear somewhere in the Record-Route
list when the request arrives again. This has the drawback of requiring a
server to be configured with some number of IP addresses.

2. The difference between the requests coming to P1 the first time and
second time is not just the top Via, but also the contents of the Route
header. What you can do is this: when the response to the initial INVITE
comes, the response carries the complete RR trace. You should be able to
infer from the request and the response what the Route header set will look
like by the time the messages reach P1, and this will be different for both
times the request arrives.

3. As Sean has pointed out, the State header is another possibility. There
is the unfortunate problem that this won't work if both UAs don't support
the State header, so you pretty much need a non-State based solution anyway.

Note that according to the spec now, the UAS does copy ONLY the maddr and
port from the RR header; everything else comes from the Contact/From.

> The first solution is to remove the text where it says
> that a UA should fill all Routes with values from
> CONTACT or FROM. Instead the original addresses should
> be kept. Personally I find it strange when you combine
> addresses (from CONTACT/FROM) with a port number from
> RECORD-ROUTE. This means that the total address now is 
> completely useless. It looks to me that you try to 
> avoid syntax changes in the protocol at all costs. I
> don't understand why a change in semantics is regarded 
> as more backwards compatible than a change in the syntax ?

Because internal semantic processing does not require communications between
devices. Thus, you can maintain interoperability. Its the "black box"
paradigm; you have a hard time changing the interface between components in
a system, but you can easily change the internal implementation and
processing logic.

That aside, the motivation for doing this oddball combination was to try and
preserve a "fundamental invariant" of SIP, which was that the request URI
should always refer to an entity that the request is actually addressed to.
If the UAS simply uses the Record-Routes as is for the Route header, this
won't be true. The result is a certain amount of brittleness.

Now, I'm going to open a huge can of worms here.

First, it seems I am in the minority of folks who think this approach
(mixing the RR and From/Contact) is sensible. I'm willing to entertain the
notion that preservation of a protocol invariant is less important than real
protocol operational problems. Since the maintenance of this invariant is
causing problems with spirals, perhaps it is not worth the pain, and we
should return to the original notion of just using the list in the order you
got it. 

Now, to make things more complex, we have a slight problem that
record-routing has troubles when the request URI in the incoming request is
not a SIP URL. Thats because, as you may recall, the RR process in a proxy
is to COPY the request URI into the RR, and then modify the maddr and port.
Now, maddr and port are SIP URL parameters. There are no such things for the
tel URL, for example. Thus, record routing a request with a tel URL is a bit
problematic. You basically need to convert it to a SIP URL first, or do
something else. Converting everything to SIP URLs may not always be possible
down the road. One can make the argument, and I would agree with it, that
the IP address and port of the proxy are not URI parameters, but
record-route header parameters. We did the hack of putting them in the URI
maddr and port since this is backwards compatible.

So, perhaps we should consider defining address and port record-route
parameters. These would only be set when the request URI contained a non-SIP
URL. Proxies would use these parameters if present for routing instead of
the maddr and port in the URL. 

COmments? (I'm sure there are).

-Jonathan R.


> 
> I would have prefered to add a port number in the
> maddr parameter because that's where I beleave it belongs. 
> 
> The second solution is to make it possible for a server
> to add it's own parameter in RECORD-ROUTE and then get
> it back in the REQUEST. Today a UAS only supposed to 
> copy maddr and port value from RECORD-ROUTE to ROUTE.
> If all parameters was copied instead it would be possible
> for a server to add anything it likes in RECORD-ROUTE
> and get it back again. I beleave this kind of solution 
> have been discussed for states ?
> 
> /Bertil
> 
> Bertil Engelholm wrote:
> > 
> > Hi again,
> > 
> > I'm sorry to bring this up again but I just can't get things
> > to work in the correct way. I have read through the old threads
> > regarding this matter and still don't understand how it is
> > suppose to work.
> > 
> > It's this example again :
> > 
> >   ---------                  --------                  --------
> >   | a@c1  | -- INV u1@p1 --> | (s1) | -- INV u2@p2 --> |      |
> >   ---------  (contact a@c1)  |      |                  |      |
> >                              |  p1  |                  |  p2  |
> >   ---------                  |      |                  |      |
> >   | b@c2  | <- INV b@c2 ---- | (s2) | <- INV u3@p1 --- |      |
> >   ---------                  --------                  --------
> > 
> > If P1 is stateful it might allocate one "call" state machine for
> > each path the "call" takes in P1 (s1 and s2 in the picture).
> > These two state machines are identified by the REQUEST-URI
> > that came in the two INVITEs since that's the only thing that
> > differs the two from each other. (topmost VIA also differs,
> > more about that later)
> > 
> > P1 will now include the REQUEST-URI in the branch value it
> > generates so that the responses can be directed to the right
> > state machine when they arrive. And when the ACK is sent the
> > original REQUEST-URI from ROUTE is used. So no problems here.
> > 
> > The problem I have is when b@C2 sends a new request (eg. BYE).
> > According to the specification b should/must/may (or whatever)
> > overwrite the REQUEST-URI in the ROUTE headers using the
> > REQUEST-URI from CONTACT or FROM.
> > 
> > So how should P1 now be able to differ the two (BYE) requests
> > from each other ? The Original REQUEST-URI was the only thing
> > that P1 had to identify the two state machines.
> > 
> > When reading the old threads regarding this matter I saw
> > something about using topmost VIA in some way, but how ?
> > 
> > By using the topmost VIA you can see that they are two different
> > requests but how do you identify which state machine should receive
> > the first request. (Someone might argue that it doesn't 
> matter but if
> > P1 controls a firewall it DO matter. Maybe not for BYE but for a
> > RE-INVITE it matters).
> > 
> > --
> > Bertil Engelholm
> > AXE Research and Development        voice : +46 8 727 3499
> > SIP Security                        Fax   : +46 8 647 8276
> > S-126 25 Stockholm Sweden           E-mail:
> > Bertil.Engelholm@uab.ericsson.se
> > 
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 
> -- 
> Bertil Engelholm
> AXE Research and Development        voice : +46 8 727 3499
> SIP Security                        Fax   : +46 8 647 8276
> S-126 25 Stockholm Sweden           E-mail:
> Bertil.Engelholm@uab.ericsson.se
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Date: Fri, 3 Nov 2000 12:38:20 -0500



 

> -----Original Message-----
> From: Sean Olson [mailto:sean.olson@ericsson.com]
> Sent: Friday, November 03, 2000 10:45 AM
> To: Robert Sparks
> Cc: SIP@lists.bell-labs.com
> Subject: Re: compact headers (was RE: [SIP] IANA SIP header
> registration)
> 
> 
> Thanks for pointing this out. What is the guideline
> for when we exhaust the 26 single-letter forms?
> Do we go to two-letter forms, three-letter forms, etc.?
> What is the point of diminishing returns here?

We will end at 26. I suspect there are implementations that assume short
form is one letter. Doing more defeats the purpose.

We've had very mixed reactions on short form. There have been posters who
wanted nothing but, and even insisted on it for things like CSeq, which is
pretty short already. Others think its a waste. 

From an implementation perspective, its really a trivial amount of work
either way. Its there; its not going away (there are implementations and
there are people who send it). So, arguing about the merits of short form is
moot.

So, in an attempt to please some of the people some of the time, the
guidelines draft has this little tidbit about when to define short form.
I'll update it to indicate that the short form should always be a single
character.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From sip-admin@lists.bell-labs.com  Fri Nov  3 13:08:03 2000
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From: Neil Deason <ndeason@ubiquity.net>
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Aplogies for a rushed post but I thought we had this
covered already with the bis-02 introduction of global
unique via branch param.

http://lists.bell-labs.com/pipermail/sip/2000q3/002635.html

Am I missing the problem? The other RR suggestions are 
certainly interesting ;)

Cheers,
Neil.
-- 
Ubiquity Software Corporation, UK        http://www.ubiquity.net

Jonathan Rosenberg wrote:
> 
> 
> 
> > -----Original Message-----
> > From: Bertil Engelholm [mailto:Bertil.Engelholm@uab.ericsson.se]
> > Sent: Friday, November 03, 2000 10:54 AM
> > To: SIP@lists.bell-labs.com
> > Subject: [SIP] Identification problem
> >
> >
> > Hi,
> >
> > Sofar noone have responded to the problem below so this
> > third time I send it I try a new subject. I was hoping
> > that some SIP responsible people would come up with a
> > solution to the problem or atleast say if it is regarded
> > as a problem or not.
> 
> Well, back in the good old days, the SIP list generated just a few messages
> a day. I used to be able to respond almost immediately to every question.
> Now, as we approach the average of 30 messages per day, it is impossible for
> me, and many of the other SIP experts, to respond to every question on the
> list.
> 
> Anyway, to answer your question, I can think of a few additional
> possibiltiies:
> 
> 1. P1 use a different IP address (i.e., different maddr) as the request
> cycles through each time. This requires detection of the spiral and
> selection of a different address each time. This can be done statelessly,
> since the previous maddr inserted will appear somewhere in the Record-Route
> list when the request arrives again. This has the drawback of requiring a
> server to be configured with some number of IP addresses.
> 
> 2. The difference between the requests coming to P1 the first time and
> second time is not just the top Via, but also the contents of the Route
> header. What you can do is this: when the response to the initial INVITE
> comes, the response carries the complete RR trace. You should be able to
> infer from the request and the response what the Route header set will look
> like by the time the messages reach P1, and this will be different for both
> times the request arrives.
> 
> 3. As Sean has pointed out, the State header is another possibility. There
> is the unfortunate problem that this won't work if both UAs don't support
> the State header, so you pretty much need a non-State based solution anyway.
> 
> Note that according to the spec now, the UAS does copy ONLY the maddr and
> port from the RR header; everything else comes from the Contact/From.
> 
> > The first solution is to remove the text where it says
> > that a UA should fill all Routes with values from
> > CONTACT or FROM. Instead the original addresses should
> > be kept. Personally I find it strange when you combine
> > addresses (from CONTACT/FROM) with a port number from
> > RECORD-ROUTE. This means that the total address now is
> > completely useless. It looks to me that you try to
> > avoid syntax changes in the protocol at all costs. I
> > don't understand why a change in semantics is regarded
> > as more backwards compatible than a change in the syntax ?
> 
> Because internal semantic processing does not require communications between
> devices. Thus, you can maintain interoperability. Its the "black box"
> paradigm; you have a hard time changing the interface between components in
> a system, but you can easily change the internal implementation and
> processing logic.
> 
> That aside, the motivation for doing this oddball combination was to try and
> preserve a "fundamental invariant" of SIP, which was that the request URI
> should always refer to an entity that the request is actually addressed to.
> If the UAS simply uses the Record-Routes as is for the Route header, this
> won't be true. The result is a certain amount of brittleness.
> 
> Now, I'm going to open a huge can of worms here.
> 
> First, it seems I am in the minority of folks who think this approach
> (mixing the RR and From/Contact) is sensible. I'm willing to entertain the
> notion that preservation of a protocol invariant is less important than real
> protocol operational problems. Since the maintenance of this invariant is
> causing problems with spirals, perhaps it is not worth the pain, and we
> should return to the original notion of just using the list in the order you
> got it.
> 
> Now, to make things more complex, we have a slight problem that
> record-routing has troubles when the request URI in the incoming request is
> not a SIP URL. Thats because, as you may recall, the RR process in a proxy
> is to COPY the request URI into the RR, and then modify the maddr and port.
> Now, maddr and port are SIP URL parameters. There are no such things for the
> tel URL, for example. Thus, record routing a request with a tel URL is a bit
> problematic. You basically need to convert it to a SIP URL first, or do
> something else. Converting everything to SIP URLs may not always be possible
> down the road. One can make the argument, and I would agree with it, that
> the IP address and port of the proxy are not URI parameters, but
> record-route header parameters. We did the hack of putting them in the URI
> maddr and port since this is backwards compatible.
> 
> So, perhaps we should consider defining address and port record-route
> parameters. These would only be set when the request URI contained a non-SIP
> URL. Proxies would use these parameters if present for routing instead of
> the maddr and port in the URL.
> 
> COmments? (I'm sure there are).
> 
> -Jonathan R.
> 
> >
> > I would have prefered to add a port number in the
> > maddr parameter because that's where I beleave it belongs.
> >
> > The second solution is to make it possible for a server
> > to add it's own parameter in RECORD-ROUTE and then get
> > it back in the REQUEST. Today a UAS only supposed to
> > copy maddr and port value from RECORD-ROUTE to ROUTE.
> > If all parameters was copied instead it would be possible
> > for a server to add anything it likes in RECORD-ROUTE
> > and get it back again. I beleave this kind of solution
> > have been discussed for states ?
> >
> > /Bertil
> >
> > Bertil Engelholm wrote:
> > >
> > > Hi again,
> > >
> > > I'm sorry to bring this up again but I just can't get things
> > > to work in the correct way. I have read through the old threads
> > > regarding this matter and still don't understand how it is
> > > suppose to work.
> > >
> > > It's this example again :
> > >
> > >   ---------                  --------                  --------
> > >   | a@c1  | -- INV u1@p1 --> | (s1) | -- INV u2@p2 --> |      |
> > >   ---------  (contact a@c1)  |      |                  |      |
> > >                              |  p1  |                  |  p2  |
> > >   ---------                  |      |                  |      |
> > >   | b@c2  | <- INV b@c2 ---- | (s2) | <- INV u3@p1 --- |      |
> > >   ---------                  --------                  --------
> > >
> > > If P1 is stateful it might allocate one "call" state machine for
> > > each path the "call" takes in P1 (s1 and s2 in the picture).
> > > These two state machines are identified by the REQUEST-URI
> > > that came in the two INVITEs since that's the only thing that
> > > differs the two from each other. (topmost VIA also differs,
> > > more about that later)
> > >
> > > P1 will now include the REQUEST-URI in the branch value it
> > > generates so that the responses can be directed to the right
> > > state machine when they arrive. And when the ACK is sent the
> > > original REQUEST-URI from ROUTE is used. So no problems here.
> > >
> > > The problem I have is when b@C2 sends a new request (eg. BYE).
> > > According to the specification b should/must/may (or whatever)
> > > overwrite the REQUEST-URI in the ROUTE headers using the
> > > REQUEST-URI from CONTACT or FROM.
> > >
> > > So how should P1 now be able to differ the two (BYE) requests
> > > from each other ? The Original REQUEST-URI was the only thing
> > > that P1 had to identify the two state machines.
> > >
> > > When reading the old threads regarding this matter I saw
> > > something about using topmost VIA in some way, but how ?
> > >
> > > By using the topmost VIA you can see that they are two different
> > > requests but how do you identify which state machine should receive
> > > the first request. (Someone might argue that it doesn't
> > matter but if
> > > P1 controls a firewall it DO matter. Maybe not for BYE but for a
> > > RE-INVITE it matters).
> > >
> > > --
> > > Bertil Engelholm
> > > AXE Research and Development        voice : +46 8 727 3499
> > > SIP Security                        Fax   : +46 8 647 8276
> > > S-126 25 Stockholm Sweden           E-mail:
> > > Bertil.Engelholm@uab.ericsson.se
> > >
> > > _______________________________________________
> > > SIP mailing list
> > > SIP@lists.bell-labs.com
> > > http://lists.bell-labs.com/mailman/listinfo/sip
> >
> > --
> > Bertil Engelholm
> > AXE Research and Development        voice : +46 8 727 3499
> > SIP Security                        Fax   : +46 8 647 8276
> > S-126 25 Stockholm Sweden           E-mail:
> > Bertil.Engelholm@uab.ericsson.se
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> >
> 
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

_______________________________________________
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
To: "'Neil Deason'" <ndeason@ubiquity.net>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Date: Fri, 3 Nov 2000 13:36:36 -0500



 

> -----Original Message-----
> From: Neil Deason [mailto:ndeason@ubiquity.net]
> Sent: Friday, November 03, 2000 1:06 PM
> To: Jonathan Rosenberg
> Cc: 'Bertil Engelholm'; SIP@lists.bell-labs.com
> Subject: Re: [SIP] Identification problem
> 
> 
> Aplogies for a rushed post but I thought we had this
> covered already with the bis-02 introduction of global
> unique via branch param.
> 
> http://lists.bell-labs.com/pipermail/sip/2000q3/002635.html
> 
> Am I missing the problem? The other RR suggestions are 
> certainly interesting ;)

The problem is that the unique via branch param allows you to
*differentiate* between each request, but there is no easy way to know that
a request corresponds to a specific state machine. Thats because (1) the
proxy immediately before and/or after P1 may not actually record-route, so
the branch param in the original INVITE could be totally different from the
one in the re-INVITE or BYE, and (2) proxy P1 has no way to know what the
branch param for requests in the REVERSE direction will look like after the
original INVITE.

-Jonathan R.

> 
> Cheers,
> Neil.
> -- 
> Ubiquity Software Corporation, UK        http://www.ubiquity.net
> 
> Jonathan Rosenberg wrote:
> > 
> > 
> > 
> > > -----Original Message-----
> > > From: Bertil Engelholm [mailto:Bertil.Engelholm@uab.ericsson.se]
> > > Sent: Friday, November 03, 2000 10:54 AM
> > > To: SIP@lists.bell-labs.com
> > > Subject: [SIP] Identification problem
> > >
> > >
> > > Hi,
> > >
> > > Sofar noone have responded to the problem below so this
> > > third time I send it I try a new subject. I was hoping
> > > that some SIP responsible people would come up with a
> > > solution to the problem or atleast say if it is regarded
> > > as a problem or not.
> > 
> > Well, back in the good old days, the SIP list generated 
> just a few messages
> > a day. I used to be able to respond almost immediately to 
> every question.
> > Now, as we approach the average of 30 messages per day, it 
> is impossible for
> > me, and many of the other SIP experts, to respond to every 
> question on the
> > list.
> > 
> > Anyway, to answer your question, I can think of a few additional
> > possibiltiies:
> > 
> > 1. P1 use a different IP address (i.e., different maddr) as 
> the request
> > cycles through each time. This requires detection of the spiral and
> > selection of a different address each time. This can be 
> done statelessly,
> > since the previous maddr inserted will appear somewhere in 
> the Record-Route
> > list when the request arrives again. This has the drawback 
> of requiring a
> > server to be configured with some number of IP addresses.
> > 
> > 2. The difference between the requests coming to P1 the 
> first time and
> > second time is not just the top Via, but also the contents 
> of the Route
> > header. What you can do is this: when the response to the 
> initial INVITE
> > comes, the response carries the complete RR trace. You 
> should be able to
> > infer from the request and the response what the Route 
> header set will look
> > like by the time the messages reach P1, and this will be 
> different for both
> > times the request arrives.
> > 
> > 3. As Sean has pointed out, the State header is another 
> possibility. There
> > is the unfortunate problem that this won't work if both UAs 
> don't support
> > the State header, so you pretty much need a non-State based 
> solution anyway.
> > 
> > Note that according to the spec now, the UAS does copy ONLY 
> the maddr and
> > port from the RR header; everything else comes from the 
> Contact/From.
> > 
> > > The first solution is to remove the text where it says
> > > that a UA should fill all Routes with values from
> > > CONTACT or FROM. Instead the original addresses should
> > > be kept. Personally I find it strange when you combine
> > > addresses (from CONTACT/FROM) with a port number from
> > > RECORD-ROUTE. This means that the total address now is
> > > completely useless. It looks to me that you try to
> > > avoid syntax changes in the protocol at all costs. I
> > > don't understand why a change in semantics is regarded
> > > as more backwards compatible than a change in the syntax ?
> > 
> > Because internal semantic processing does not require 
> communications between
> > devices. Thus, you can maintain interoperability. Its the 
> "black box"
> > paradigm; you have a hard time changing the interface 
> between components in
> > a system, but you can easily change the internal implementation and
> > processing logic.
> > 
> > That aside, the motivation for doing this oddball 
> combination was to try and
> > preserve a "fundamental invariant" of SIP, which was that 
> the request URI
> > should always refer to an entity that the request is 
> actually addressed to.
> > If the UAS simply uses the Record-Routes as is for the 
> Route header, this
> > won't be true. The result is a certain amount of brittleness.
> > 
> > Now, I'm going to open a huge can of worms here.
> > 
> > First, it seems I am in the minority of folks who think 
> this approach
> > (mixing the RR and From/Contact) is sensible. I'm willing 
> to entertain the
> > notion that preservation of a protocol invariant is less 
> important than real
> > protocol operational problems. Since the maintenance of 
> this invariant is
> > causing problems with spirals, perhaps it is not worth the 
> pain, and we
> > should return to the original notion of just using the list 
> in the order you
> > got it.
> > 
> > Now, to make things more complex, we have a slight problem that
> > record-routing has troubles when the request URI in the 
> incoming request is
> > not a SIP URL. Thats because, as you may recall, the RR 
> process in a proxy
> > is to COPY the request URI into the RR, and then modify the 
> maddr and port.
> > Now, maddr and port are SIP URL parameters. There are no 
> such things for the
> > tel URL, for example. Thus, record routing a request with a 
> tel URL is a bit
> > problematic. You basically need to convert it to a SIP URL 
> first, or do
> > something else. Converting everything to SIP URLs may not 
> always be possible
> > down the road. One can make the argument, and I would agree 
> with it, that
> > the IP address and port of the proxy are not URI parameters, but
> > record-route header parameters. We did the hack of putting 
> them in the URI
> > maddr and port since this is backwards compatible.
> > 
> > So, perhaps we should consider defining address and port 
> record-route
> > parameters. These would only be set when the request URI 
> contained a non-SIP
> > URL. Proxies would use these parameters if present for 
> routing instead of
> > the maddr and port in the URL.
> > 
> > COmments? (I'm sure there are).
> > 
> > -Jonathan R.
> > 
> > >
> > > I would have prefered to add a port number in the
> > > maddr parameter because that's where I beleave it belongs.
> > >
> > > The second solution is to make it possible for a server
> > > to add it's own parameter in RECORD-ROUTE and then get
> > > it back in the REQUEST. Today a UAS only supposed to
> > > copy maddr and port value from RECORD-ROUTE to ROUTE.
> > > If all parameters was copied instead it would be possible
> > > for a server to add anything it likes in RECORD-ROUTE
> > > and get it back again. I beleave this kind of solution
> > > have been discussed for states ?
> > >
> > > /Bertil
> > >
> > > Bertil Engelholm wrote:
> > > >
> > > > Hi again,
> > > >
> > > > I'm sorry to bring this up again but I just can't get things
> > > > to work in the correct way. I have read through the old threads
> > > > regarding this matter and still don't understand how it is
> > > > suppose to work.
> > > >
> > > > It's this example again :
> > > >
> > > >   ---------                  --------                  --------
> > > >   | a@c1  | -- INV u1@p1 --> | (s1) | -- INV u2@p2 --> |      |
> > > >   ---------  (contact a@c1)  |      |                  |      |
> > > >                              |  p1  |                  |  p2  |
> > > >   ---------                  |      |                  |      |
> > > >   | b@c2  | <- INV b@c2 ---- | (s2) | <- INV u3@p1 --- |      |
> > > >   ---------                  --------                  --------
> > > >
> > > > If P1 is stateful it might allocate one "call" state machine for
> > > > each path the "call" takes in P1 (s1 and s2 in the picture).
> > > > These two state machines are identified by the REQUEST-URI
> > > > that came in the two INVITEs since that's the only thing that
> > > > differs the two from each other. (topmost VIA also differs,
> > > > more about that later)
> > > >
> > > > P1 will now include the REQUEST-URI in the branch value it
> > > > generates so that the responses can be directed to the right
> > > > state machine when they arrive. And when the ACK is sent the
> > > > original REQUEST-URI from ROUTE is used. So no problems here.
> > > >
> > > > The problem I have is when b@C2 sends a new request (eg. BYE).
> > > > According to the specification b should/must/may (or whatever)
> > > > overwrite the REQUEST-URI in the ROUTE headers using the
> > > > REQUEST-URI from CONTACT or FROM.
> > > >
> > > > So how should P1 now be able to differ the two (BYE) requests
> > > > from each other ? The Original REQUEST-URI was the only thing
> > > > that P1 had to identify the two state machines.
> > > >
> > > > When reading the old threads regarding this matter I saw
> > > > something about using topmost VIA in some way, but how ?
> > > >
> > > > By using the topmost VIA you can see that they are two different
> > > > requests but how do you identify which state machine 
> should receive
> > > > the first request. (Someone might argue that it doesn't
> > > matter but if
> > > > P1 controls a firewall it DO matter. Maybe not for BYE but for a
> > > > RE-INVITE it matters).
> > > >
> > > > --
> > > > Bertil Engelholm
> > > > AXE Research and Development        voice : +46 8 727 3499
> > > > SIP Security                        Fax   : +46 8 647 8276
> > > > S-126 25 Stockholm Sweden           E-mail:
> > > > Bertil.Engelholm@uab.ericsson.se
> > > >
> > > > _______________________________________________
> > > > SIP mailing list
> > > > SIP@lists.bell-labs.com
> > > > http://lists.bell-labs.com/mailman/listinfo/sip
> > >
> > > --
> > > Bertil Engelholm
> > > AXE Research and Development        voice : +46 8 727 3499
> > > SIP Security                        Fax   : +46 8 647 8276
> > > S-126 25 Stockholm Sweden           E-mail:
> > > Bertil.Engelholm@uab.ericsson.se
> > >
> > > _______________________________________________
> > > SIP mailing list
> > > SIP@lists.bell-labs.com
> > > http://lists.bell-labs.com/mailman/listinfo/sip
> > >
> > 
> > ---
> > Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> > Chief Scientist                             First Floor
> > dynamicsoft                                 East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> > http://www.dynamicsoft.com
> > 
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From sip-admin@lists.bell-labs.com  Fri Nov  3 13:46:07 2000
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The IANA registration simply registers existing formats and makes no
statement as to whether compact forms are good, bad or indifferent. I
agree that anything but single-letter abbreviations are useless and that
those should be reserved to standards-track documents. 

Sean Olson wrote:
> 
> Does this mean open season on compact forms? I really hope not.
> In fact, I wish they would go away altogether. I don't want to re-open
> this debate, but I would like to dam the flood of new compact forms.
> 
> Am I the only one who feels this way?
> /sean

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From sip-admin@lists.bell-labs.com  Fri Nov  3 13:48:37 2000
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Date: Fri, 3 Nov 2000 13:46:37 -0500 (EST)
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I've been pondering details of the concept of "pre-loaded" Route headers --
Route headers that go in the initial transaction of a call, used to direct a
call to go through an outbound proxy server.

My question is, if you pre-load a Route header into an initial INVITE, do
those Route headers have to persist for the rest of the call?  Or do
subsequent transactions just build their Route headers based on the
Record-Routes that were inserted in that first transaction?

I think it would have to be the latter, since the destination UAS doesn't
see any of the pre-loaded Route headers.  The UAS is going to all its
routing based on those initial Record-Routes -- all the pre-loaded Routes
are stripped by the time the request gets to the UAS.

I was going to suggest that if you get a request with your own Route header
in it you be required to insert yourself as a Record-Route -- until I
remembered that you don't see Route headers for yourself.  (This causes a
lot of headaches, it seems.)

The problem is that this seems to be asymmetric with how Route headers
persist *after* the first transaction -- 2543bis says they remain even if
subsequent transactions don't put Record-Routes in for them.  This may be
the correct behavior, but it should probably be called out explicitly.

Thoughts?

-- 
Jonathan Lennox
lennox@cs.columbia.edu

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Subject: RE: [SIP] Sip- Message OPTION
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Date: Fri, 3 Nov 2000 10:19:48 -0800

Hi Jonathan,
  The draft (bis-2 section 4.2.3) says:
"The response MAY contain a message body indicating the capabilities
of the end system ..." 

  This gives an idea that OPTIONS should be relevant only for a UAS,
since only a UAS can respond with the capabilities of the logged in
users. The example given in section 16.8, also seems to suggest the
same. 
  However, theoretically, it is possible for a user to register
his media capabilities as well in the REGISTER message with the
Registrar. But I see this as a more dynamic thing which can 
potentially change from a call to call. So, what is the usage of
supporting OPTIONS at a Registrar server ? Kindly clarify.

-regards,
aseem
 


-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Thursday, November 02, 2000 11:39 PM
To: 'Thys Nel'; 'Billy Biggs'; archow@hss.hns.com
Cc: 'Sandeep Sahai'; 'SIP List'
Subject: RE: [SIP] Sip- Message OPTION


All servers (UAS, registrar) should answer a OPTIONS with a 200 OK with the
following headers added to the response: Allow, listing the supported
methods, Accept, listing the support MIME types, and Supported, listing the
supported extensions. A UAS could also insert SDP listing its codecs. How
such SDP is formatted has never been defined, and to my knowledge, no one is
yet doing this. We'll define something in bis.

As an example for a registrar response to OPTIONS:

SIP/2.0 200 OK
From: sip:querier@company.com
To: sip:registardomain.com;tag=9378asd0asd7asd
Call-ID: a90sda08shd@1.2.3.4
CSeq: 1 REGISTER
Via: SIP/2.0/UDP host.company.com
Accept: application/cpl+xml
Allow: OPTIONS, REGISTER, CANCEL
Supported: pref
Content-Length: 0


This registar supports uploads of CPL, and also the caller preferences
extension.

-Jonathan R.



---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Thys Nel [mailto:thys@parsec.co.za]
> Sent: Thursday, November 02, 2000 1:48 AM
> To: 'Billy Biggs'; archow@hss.hns.com
> Cc: 'Sandeep Sahai'; 'SIP List'
> Subject: RE: [SIP] Sip- Message OPTION
> 
> 
> Does anyone have examples of messages flows with OPTIONS 
> processed by a
> registrar server? The only example in the RFC is between 2 
> user agents, but
> the RFC states that a registrar must also support it. I can't 
> seem to find
> any description of when a registrar should respond to an 
> OPTIONS request.
> 
> Thys
> 
> > -----Original Message-----
> > From: sip-admin@lists.bell-labs.com
> > [mailto:sip-admin@lists.bell-labs.com]On Behalf Of Billy Biggs
> > Sent: 02 November 2000 8:17
> > To: archow@hss.hns.com
> > Cc: Sandeep Sahai; SIP List
> > Subject: Re: [SIP] Sip- Message OPTION
> >
> >
> >   Note that this should be cleaned up.  There is no Via or 
> CSeq in the
> > response, the CSeq in the request starts at 1, the From has
> > no tag, and there
> > is no Content-Length.
> >
> > > C->S: OPTIONS sip:bob@example.com SIP/2.0
> > > Via: SIP/2.0/UDP cat.wonderland.com
> > > From: Alice <sip:alice@wonderland.com>
> > > To: Bob <sip:bob@example.com>
> > > Call-ID: 6378@cat.wonderland.com
> > > CSeq: 1 OPTIONS
> > > Accept: application/sdp
> > >
> > > S->C: SIP/2.0 200 OK
> > > From: Alice <sip:alice@wonderland.com>
> > > To: Bob <sip:bob@example.com> ;tag=376364382
> > > Call-ID: 6378@cat.wonderland.com
> > > Content-Length: 81
> > > Content-Type: application/sdp
> > >
> > > v=0
> > > o=alice 3149329138 3149329165 IN IP4 24.124.37.3
> > > s=Security problems
> > > t=3149328650 0
> > > c=IN IP4 24.124.37.3
> > > m=audio 0 RTP/AVP 0 1 3 99
> > > a=rtpmap:0 PCMU/8000
> > > a=rtpmap:1 1016/8000
> > > a=rtpmap:3 GSM/8000
> > > a=rtpmap:99 SX7300/8000
> > > m=video 0 RTP/AVP 31 34
> > > a=rtpmap:31 H261/90000
> > > a=rtpmap:34 H263/90000
> >
> > --
> > Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> > http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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Date: Fri, 3 Nov 2000 10:30:02 -0800

Hi :

Thanks for the response.

Another concern, section 5 states -

"Session expiration are mostly of interest to call stateful
proxy servers. However, a stateful proxy server MAY also follow
the rules described here"

Shouldn't it be mandated by the draft that, if a stateful proxy
(transaction stateful case) server is starting a Session Timer, 
it MUST insert a "Record-Route" header in the Invite, prior to 
proxying it.

Since, the procedure is symmetric, i.e. UAS can send a Re-Invite 
and become the master, to avoid such proxies being missed out and
later timing out, it MUST be on the path.

Also, is there any example when a transaction stateful only
proxy would like to start a session timer?

-Regards,
Manish Singh

> 
> Hi,
> 
> Sectoin 4, UAC behavior states :
> 
> "Note that it is possible that the calling UA is generating
> refreshes, and then it receives a re-INVITE. After following
> the rules for UAS described below, the calling UA now determines
> it is not supposed to generate refreshes..........."
> 
> Is their a specific reason for allowing the master switchover
> for generating the re-INVITES? After the initial negotation, if
> the UAC was responsible for generating re-INVITES, why would UAS
> in the middle of a session like to take control? 

The idea was just to make the whole process stateless, so that processing of
a re-INVITE is absolutely the same as processing the initial INVITE. As a
result of this choice, the roles will change over on re-INVITEs depending on
what level of support each side has. If each side both supports session
timer, the user sending the re-INVITE will become the one performing
refreshes if the re-invite completes successfully.

The stateless operation is needed, for example, to recover from crash-reboot
cycles.

I think this actually makes the implementation simpler, not more complex.

> 
> Can we add an Example flow for this scenario in section 8 of the
> draft. 

I will add text explaining the above, and also an example flow.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com


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From: "Shantala Manchenahally" <shantala@nortelnetworks.com>
To: "'Jonathan Lennox'" <lennox@cs.columbia.edu>, sip@lists.bell-labs.com
Subject: RE: [SIP] Pre-loaded Route headers
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Hi,

Just a few thoughts on the matter..

I agree that a Record is stripped before it gets to the Proxy. However if a
proxy see's a Route Header (specifying the next hop)in the initial request,
it can decide to add itself to the Record-Route. If every Proxy along the
path
obeys this rule then the pre-loaded route will be preserved for all
subsequent transactions. 

This wouldn't work for the cases where the pre-loaded Route doesn't specify
all the hops
between the UAC and the UAS. So a Proxy may not see a Route header at all,
even though the initial message did contain a Route header.

So I would think that a pre-loaded Route can effectively be applied for that
transaction alone.

Shantala.

-----Original Message-----
From: Jonathan Lennox [mailto:lennox@cs.columbia.edu]
Sent: Friday, November 03, 2000 12:47 PM
To: sip@lists.bell-labs.com
Subject: [SIP] Pre-loaded Route headers


I've been pondering details of the concept of "pre-loaded" Route headers --
Route headers that go in the initial transaction of a call, used to direct a
call to go through an outbound proxy server.

My question is, if you pre-load a Route header into an initial INVITE, do
those Route headers have to persist for the rest of the call?  Or do
subsequent transactions just build their Route headers based on the
Record-Routes that were inserted in that first transaction?

I think it would have to be the latter, since the destination UAS doesn't
see any of the pre-loaded Route headers.  The UAS is going to all its
routing based on those initial Record-Routes -- all the pre-loaded Routes
are stripped by the time the request gets to the UAS.

I was going to suggest that if you get a request with your own Route header
in it you be required to insert yourself as a Record-Route -- until I
remembered that you don't see Route headers for yourself.  (This causes a
lot of headaches, it seems.)

The problem is that this seems to be asymmetric with how Route headers
persist *after* the first transaction -- 2543bis says they remain even if
subsequent transactions don't put Record-Routes in for them.  This may be
the correct behavior, but it should probably be called out explicitly.

Thoughts?

-- 
Jonathan Lennox
lennox@cs.columbia.edu

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<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2652.35">
<TITLE>RE: [SIP] Pre-loaded Route headers</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Hi,</FONT>
</P>

<P><FONT SIZE=3D2>Just a few thoughts on the matter..</FONT>
</P>

<P><FONT SIZE=3D2>I agree that a Record is stripped before it gets to =
the Proxy. However if a proxy see's a Route Header (specifying the next =
hop)in the initial request, it can decide to add itself to the =
Record-Route. If every Proxy along the path</FONT></P>

<P><FONT SIZE=3D2>obeys this rule then the pre-loaded route will be =
preserved for all subsequent transactions. </FONT>
</P>

<P><FONT SIZE=3D2>This wouldn't work for the cases where the pre-loaded =
Route doesn't specify all the hops</FONT>
<BR><FONT SIZE=3D2>between the UAC and the UAS. So a Proxy may not see =
a Route header at all, even though the initial message did contain a =
Route header.</FONT></P>

<P><FONT SIZE=3D2>So I would think that a pre-loaded Route can =
effectively be applied for that transaction alone.</FONT>
</P>

<P><FONT SIZE=3D2>Shantala.</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Jonathan Lennox [<A =
HREF=3D"mailto:lennox@cs.columbia.edu">mailto:lennox@cs.columbia.edu</A>=
]</FONT>
<BR><FONT SIZE=3D2>Sent: Friday, November 03, 2000 12:47 PM</FONT>
<BR><FONT SIZE=3D2>To: sip@lists.bell-labs.com</FONT>
<BR><FONT SIZE=3D2>Subject: [SIP] Pre-loaded Route headers</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>I've been pondering details of the concept of =
&quot;pre-loaded&quot; Route headers --</FONT>
<BR><FONT SIZE=3D2>Route headers that go in the initial transaction of =
a call, used to direct a</FONT>
<BR><FONT SIZE=3D2>call to go through an outbound proxy server.</FONT>
</P>

<P><FONT SIZE=3D2>My question is, if you pre-load a Route header into =
an initial INVITE, do</FONT>
<BR><FONT SIZE=3D2>those Route headers have to persist for the rest of =
the call?&nbsp; Or do</FONT>
<BR><FONT SIZE=3D2>subsequent transactions just build their Route =
headers based on the</FONT>
<BR><FONT SIZE=3D2>Record-Routes that were inserted in that first =
transaction?</FONT>
</P>

<P><FONT SIZE=3D2>I think it would have to be the latter, since the =
destination UAS doesn't</FONT>
<BR><FONT SIZE=3D2>see any of the pre-loaded Route headers.&nbsp; The =
UAS is going to all its</FONT>
<BR><FONT SIZE=3D2>routing based on those initial Record-Routes -- all =
the pre-loaded Routes</FONT>
<BR><FONT SIZE=3D2>are stripped by the time the request gets to the =
UAS.</FONT>
</P>

<P><FONT SIZE=3D2>I was going to suggest that if you get a request with =
your own Route header</FONT>
<BR><FONT SIZE=3D2>in it you be required to insert yourself as a =
Record-Route -- until I</FONT>
<BR><FONT SIZE=3D2>remembered that you don't see Route headers for =
yourself.&nbsp; (This causes a</FONT>
<BR><FONT SIZE=3D2>lot of headaches, it seems.)</FONT>
</P>

<P><FONT SIZE=3D2>The problem is that this seems to be asymmetric with =
how Route headers</FONT>
<BR><FONT SIZE=3D2>persist *after* the first transaction -- 2543bis =
says they remain even if</FONT>
<BR><FONT SIZE=3D2>subsequent transactions don't put Record-Routes in =
for them.&nbsp; This may be</FONT>
<BR><FONT SIZE=3D2>the correct behavior, but it should probably be =
called out explicitly.</FONT>
</P>

<P><FONT SIZE=3D2>Thoughts?</FONT>
</P>

<P><FONT SIZE=3D2>-- </FONT>
<BR><FONT SIZE=3D2>Jonathan Lennox</FONT>
<BR><FONT SIZE=3D2>lennox@cs.columbia.edu</FONT>
</P>

<P><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>SIP mailing list</FONT>
<BR><FONT SIZE=3D2>SIP@lists.bell-labs.com</FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://lists.bell-labs.com/mailman/listinfo/sip" =
TARGET=3D"_blank">http://lists.bell-labs.com/mailman/listinfo/sip</A></F=
ONT>
</P>

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From sip-admin@lists.bell-labs.com  Fri Nov  3 16:04:03 2000
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Subject: [SIP] Re: Pre-loaded Route headers
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Date: Fri, 3 Nov 2000 16:01:41 -0500 (EST)

I believe there may be a problem with Pre-loaded Route headers and
stateless proxies, but it not a difficult one to solve.  But there
are side-affects that are not nice.

An initial INVITE is normally the one that accumulates the Record-Route
values; then they are stored at the UAS and converted into a Route header
for UAS->UAC transactions, and echoed back to the UAC in the 200-OK so
the UAC converts them into a Route header for UAC->UAS transactions.
A mid-call INVITE (or any other message) contains the Route header
value, but doesn't need to accumulate Record-Routes again.

But a stateless proxy can't really tell the difference between an
initial INVITE for a new session (one in which it should add itself
to the Record-Route) and a mid-call INVITE for an existing session (in
which it doesn't need to add itself).  So it has two choices: 1) always
add itself to a Record-Route, or 2) note there is a Route header and
assume therefore it is a mid-call message and do nothing.

Keying on the existence of a Route header to indicate a mid-call
message isn't explicitely disallowed by the spec (unless I missed
it somewhere), but it would work just fine - until a UAC tried to
use a pre-loaded Route header.  Then the proxy wouldn't get itself
into the Record-Route when it should.

So the solution is to have the proxy insert itself into the
Record-Route header *independent of whether it is an initial INVITE
or a mid-call INVITE*.  

Unfortunately there is a backward compatibility problem here - a proxy
that doesn't know to do this for a mid-call message will be dropped
from future messages.  If even one proxy along the path inserts a
Record-Route header on the mid-call message, the UAS and UAC will
trigger their Route re-calculation.  (Presumably if none of the proxies
did the Record-Route for a mid-call message, the UAC and UAS would
act like currently, and keep the Route from the initial INVITE).

Is this just a theoretical problem, or does it break real proxy
implementations?

Bill Marshall
wtm@research.att.com

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Subject: [SIP] Help with installation of sipd
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Hi,
   We are trying to install sipd (columbia release) on a linux box. But, before compiling sipd server, do we need to install the LDAP (Umich distribution)? If so then I guess we need root access for installation of LDAP??

   In the README.build of sipd there is an example for configuring  as follows:

  ./configure --with-db=/proj/irt-gc4/cinema/sun5 \
              --with-ldap=/proj/irt-gc4/cinema/sun5

 Now, we are not sure as to what should be in the sun5 directory? Should there be the sources for LDAP and berkely db? OR Is it the executables that needs to be present in this directory?

Any help and suggestions will be greatly appreciated.

Thanks
T.Singh

    

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Hello,
    according to section B. 6 in a new draft it is possible to put the
other party "on hold",
that is to ask it temporarily stop sending media, by sending re-INVITE
with the destina-
tion addresses (c= lines) set to 0.
What should we expect in 200 OK ? Should it be (upon acception of
re-INVITE) the
regular SDP with valid ports and destination addresses, since it stops
sending, but
still continue receiving ?

Thanx.

--
Vladislav Zubarev        mailto:vladis@vovida.com
Software Engineer
Vovida Networks, Inc.    http://www.vovida.com



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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Hello,
<br>&nbsp;&nbsp;&nbsp; according to section B. 6 in a new draft it is possible
to put the other party "on hold",
<br>that is to ask it temporarily stop sending media, by sending re-INVITE
with the destina-
<br>tion addresses (c= lines) set to 0.
<br>What should we expect in 200 OK ? Should it be (upon acception of re-INVITE)
the
<br>regular SDP with valid ports and destination addresses, since it stops
sending, but
<br>still continue receiving ?
<p>Thanx.
<pre>--&nbsp;
Vladislav Zubarev&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <A HREF="mailto:vladis@vovida.com">mailto:vladis@vovida.com</A>
Software Engineer&nbsp;&nbsp;&nbsp;
Vovida Networks, Inc.&nbsp;&nbsp;&nbsp; <A HREF="http://www.vovida.com">http://www.vovida.com</A></pre>
&nbsp;</html>

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Date: Fri, 3 Nov 2000 17:49:51 -0600

2543bis 4.2.1 INVITE:

  "The INVITE method indicates that the user or service is being invited
   to participate in a session. The message body MAY contain a
   description of the session to which the callee is being invited. For
   two-party calls, the caller indicates the type of media it is able to
   receive and possibly the media it is willing to send as well as their
   parameters such as network destination. A success response MUST
   indicate in its message body which media the callee wishes to receive
   and MAY indicate the media the callee is going to send."

How does a UA indicate (in case it wishes) the media it is going to send?

Uri

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From: "Singh, Manish" <manishs@trillium.com>
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Subject: RE: [SIP] Re: Pre-loaded Route headers
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Date: Fri, 3 Nov 2000 17:26:28 -0800

Hi :

If any proxy inserts a Record-Route header in mid-call
message, how will UAS and UAC re-calculate the Route
header?

A -INV-> P1 -INV(RR:P1)-> P2 -INV(RR:P1)-> B

In this case P1 decides to add Record-Route header
in the intial Invite, however P2 doesn't.

Now, if 'A' sends a re-Invite, the Route header mandates
only P1 to be in the path. However, if P1 proxies 
this re-Invite to another proxy P2' this time (depending
on SRV records), and P2' policy is to add Record-Route
in the mid-call message, when the re-Invite reaches
'B', how will it re-construct the correct Route header.

To examplify, pls consider these secnarios -

1. It is possible, that after P2, there is another
proxy P3, and P3 in the initial Invite had also 
added Record-Route, so the Route at 'B' was P3, P1.
With P2', in the Re-Invite case, it should then be 
P3, P2', P1.

A -> P1 -> P2 -> P3 -> B (Original Invite Path)

A -> P1 -> P2'-> P3 -> B (Re-Invite Path)

2. If P3 in the initial Invite was prior to P2, then
the initial Route at 'B' was still P3, P1. However,
after re-Invite it should be P2', P3, P1.

A -> P1 -> P3 -> P2 -> B (Original Invite Path)

A -> P1 -> P3 -> P2'-> B (Re-Invite Path)

So, how will UAS re-calculate the Route header?

Is there something I am missing?

Comments?

Thanks,
Manish Singh


-----Original Message-----
From: William Marshall [mailto:wtm@research.att.com]
Sent: Friday, November 03, 2000 1:02 PM
To: sip@lists.bell-labs.com
Subject: [SIP] Re: Pre-loaded Route headers


I believe there may be a problem with Pre-loaded Route headers and
stateless proxies, but it not a difficult one to solve.  But there
are side-affects that are not nice.

An initial INVITE is normally the one that accumulates the Record-Route
values; then they are stored at the UAS and converted into a Route header
for UAS->UAC transactions, and echoed back to the UAC in the 200-OK so
the UAC converts them into a Route header for UAC->UAS transactions.
A mid-call INVITE (or any other message) contains the Route header
value, but doesn't need to accumulate Record-Routes again.

But a stateless proxy can't really tell the difference between an
initial INVITE for a new session (one in which it should add itself
to the Record-Route) and a mid-call INVITE for an existing session (in
which it doesn't need to add itself).  So it has two choices: 1) always
add itself to a Record-Route, or 2) note there is a Route header and
assume therefore it is a mid-call message and do nothing.

Keying on the existence of a Route header to indicate a mid-call
message isn't explicitely disallowed by the spec (unless I missed
it somewhere), but it would work just fine - until a UAC tried to
use a pre-loaded Route header.  Then the proxy wouldn't get itself
into the Record-Route when it should.

So the solution is to have the proxy insert itself into the
Record-Route header *independent of whether it is an initial INVITE
or a mid-call INVITE*.  

Unfortunately there is a backward compatibility problem here - a proxy
that doesn't know to do this for a mid-call message will be dropped
from future messages.  If even one proxy along the path inserts a
Record-Route header on the mid-call message, the UAS and UAC will
trigger their Route re-calculation.  (Presumably if none of the proxies
did the Record-Route for a mid-call message, the UAC and UAS would
act like currently, and keep the Route from the initial INVITE).

Is this just a theoretical problem, or does it break real proxy
implementations?

Bill Marshall
wtm@research.att.com

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From sip-admin@lists.bell-labs.com  Sat Nov  4 04:43:01 2000
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Date: Sat, 4 Nov 2000 12:42:07 +0300

This is a multi-part message in MIME format.

------=_NextPart_000_000C_01C0465C.A478F560
Content-Type: text/plain;
	charset="koi8-r"
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Hi, all!
Could you help me in my SIPs study?
If proxy forwards a SIP request, it add Via header with its address. =
This ensures responses take the same path as the requests. OK!
If one proxy along the path is failure, what upstream proxy must do to =
route responses?  =20

Thank you!
Samorezov Vladimir.

------=_NextPart_000_000C_01C0465C.A478F560
Content-Type: text/html;
	charset="koi8-r"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Dkoi8-r" http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2614.3500" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#b8b8b8>
<DIV><FONT face=3D"Comic Sans MS" size=3D2>Hi, all!</FONT></DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2>Could you help me in my SIPs=20
study?</FONT></DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2>If proxy forwards a SIP =
request, it add=20
Via header with its address.&nbsp;This <FONT size=3D2>ensures responses =
take the=20
same path as the requests. OK!</FONT></FONT></DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2>If one proxy along the path =
is failure,=20
what&nbsp;upstream proxy must do to route responses?&nbsp;&nbsp; =
</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2>Thank you!</FONT></DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2>Samorezov=20
Vladimir.</FONT></DIV></BODY></HTML>

------=_NextPart_000_000C_01C0465C.A478F560--


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From: David Shrader <dshrader@master-consultant.com>
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Table 5 contains a note 2 regarding the Via field. It states that the UAS
removes first Via header field. This seems to be in direct contradiction to
the purpose of the Via field. When that proxy receives the response, if
there is no Via field in it, then 1) it can't know that it is meant for it
and 2) it wouldn't be able to identify the branch without the branch
parameter.

Am I missing something? What was meant by the note in Table 5?

----------------------------
David Shrader
Master Consultant, Inc.
dshrader@master-consultant.com
http://www.EyeForTheFuture.com


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From sip-admin@lists.bell-labs.com  Sat Nov  4 16:00:03 2000
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IANA wrote:
> 
> Dear Henning,
> 
> We apologize for the delay.  We have been in touch with the ADs and
> IANA will set up this new registry.
> 
> Would it be possible to get an initial file of values so that we can
> set this up.  It would also be helpful to have a small paragraph
> of text for the beginning of the file.  Will you be able to provide
> this?

The current list is at http://www.cs.columbia.edu/sip/headers.txt

> 
> In your IANA considerations section, you will need to include the procedure
> for deciding when and how a new one can be registered.

A suggestion is 

SIP Header field names are registered with IANA.  They do not require
working group or working group chair review, but {\SHOULD} be documented
in an RFC or Internet draft.  Headers {\SHOULDNOT} use the \header{X-}
prefix notation and {\MUSTNOT} duplicate the names of headers used by 
SMTP or HTTP unless the syntax is a compatible superset and the 
semantics are similar.  Some common and widely used header fields {\MAY}
be assigned one-letter compact forms (Section~\ref{sec:compact}).
Compact forms can only be assigned after SIP working group review.  In  
the absence of this working group, a designated expert.


> 
> Let me know if you have any questions with the above.

Thanks for your help.

> 
> Thank you,
> 
> Michelle Schipper
> IANA Administrator


-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Jonathan Rosenberg wrote:
 
[snip]

> Now, I'm going to open a huge can of worms here.
> 
> First, it seems I am in the minority of folks who think this approach
> (mixing the RR and From/Contact) is sensible. I'm willing to entertain the
> notion that preservation of a protocol invariant is less important than real
> protocol operational problems. Since the maintenance of this invariant is
> causing problems with spirals, perhaps it is not worth the pain, and we
> should return to the original notion of just using the list in the order you
> got it.

I don't like this. We should stick to the basic principal 
that Request URIs refer to the addressed entity. Breaking
that would just create confusion. Though I do have a slight 
problem in that we already weaken the premise by allowing 
overloading of the port value within the hostport.

> Now, to make things more complex, we have a slight problem that
> record-routing has troubles when the request URI in the incoming request is
> not a SIP URL. Thats because, as you may recall, the RR process in a proxy
> is to COPY the request URI into the RR, and then modify the maddr and port.
> Now, maddr and port are SIP URL parameters. There are no such things for the
> tel URL, for example. Thus, record routing a request with a tel URL is a bit
> problematic. You basically need to convert it to a SIP URL first, or do
> something else. Converting everything to SIP URLs may not always be possible
> down the road. One can make the argument, and I would agree with it, that
> the IP address and port of the proxy are not URI parameters, but
> record-route header parameters. We did the hack of putting them in the URI
> maddr and port since this is backwards compatible.
> 
> So, perhaps we should consider defining address and port record-route
> parameters. These would only be set when the request URI contained a non-SIP
> URL. Proxies would use these parameters if present for routing instead of
> the maddr and port in the URL.

There is a potential problem here and this looks like a 
reasonable solution. Pitty we can't extend it to all URLs.

Cheers,
Neil.
-- 
Ubiquity Software Corporation, UK        http://www.ubiquity.net

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Jonathan Rosenberg wrote:
> 
> > Am I missing the problem? The other RR suggestions are
> > certainly interesting ;)
> 
> The problem is that the unique via branch param allows you to
> *differentiate* between each request, but there is no easy way to know that
> a request corresponds to a specific state machine. Thats because (1) the
> proxy immediately before and/or after P1 may not actually record-route, so
> the branch param in the original INVITE could be totally different from the
> one in the re-INVITE or BYE, and (2) proxy P1 has no way to know what the
> branch param for requests in the REVERSE direction will look like after the
> original INVITE.

I agree there is this issue but is it really a problem for 
the spec? I saw this as just one of the additional challenges 
faced by a Proxy wishing to be call stateful. It isn't easy but 
I think it is already solvable without requiring further addition 
to the spec. If there is a neat solution that adds something more 
if everybody follows it then we should add it. If not we don't 
need more text.

Cheers,
Neil.
-- 
Ubiquity Software Corporation, UK        http://www.ubiquity.net

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From: "Hisham Khartabil" <hisham.khartabil@hotsip.com>
To: "Jonathan Lennox" <lennox@cs.columbia.edu>, <sip@lists.bell-labs.com>
Subject: RE: [SIP] Pre-loaded Route headers
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My questions are:

1. who would build this initial route header? It would have to be the user
2. how would this user know of the path?

Regards,
Hisham
Hotsip

-----Original Message-----
From: sip-admin@lists.bell-labs.com [mailto:sip-admin@lists.bell-labs.com]On
Behalf Of Jonathan Lennox
Sent: Friday, 3 November 2000 8:47 PM
To: sip@lists.bell-labs.com
Subject: [SIP] Pre-loaded Route headers

I've been pondering details of the concept of "pre-loaded" Route headers --
Route headers that go in the initial transaction of a call, used to direct a
call to go through an outbound proxy server.

My question is, if you pre-load a Route header into an initial INVITE, do
those Route headers have to persist for the rest of the call?  Or do
subsequent transactions just build their Route headers based on the
Record-Routes that were inserted in that first transaction?

I think it would have to be the latter, since the destination UAS doesn't
see any of the pre-loaded Route headers.  The UAS is going to all its
routing based on those initial Record-Routes -- all the pre-loaded Routes
are stripped by the time the request gets to the UAS.

I was going to suggest that if you get a request with your own Route header
in it you be required to insert yourself as a Record-Route -- until I
remembered that you don't see Route headers for yourself.  (This causes a
lot of headaches, it seems.)

The problem is that this seems to be asymmetric with how Route headers
persist *after* the first transaction -- 2543bis says they remain even if
subsequent transactions don't put Record-Routes in for them.  This may be
the correct behavior, but it should probably be called out explicitly.

Thoughts?

--
Jonathan Lennox
lennox@cs.columbia.edu

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See comments below.

/Bertil

Jonathan Rosenberg wrote:
> 
> 
> 
> > -----Original Message-----
> > From: Bertil Engelholm [mailto:Bertil.Engelholm@uab.ericsson.se]
> > Sent: Friday, November 03, 2000 10:54 AM
> > To: SIP@lists.bell-labs.com
> > Subject: [SIP] Identification problem
> >
> >
> > Hi,
> >
> > Sofar noone have responded to the problem below so this
> > third time I send it I try a new subject. I was hoping
> > that some SIP responsible people would come up with a
> > solution to the problem or atleast say if it is regarded
> > as a problem or not.
> 
> Well, back in the good old days, the SIP list generated just a few messages
> a day. I used to be able to respond almost immediately to every question.
> Now, as we approach the average of 30 messages per day, it is impossible for
> me, and many of the other SIP experts, to respond to every question on the
> list.
> 
> Anyway, to answer your question, I can think of a few additional
> possibiltiies:
> 
> 1. P1 use a different IP address (i.e., different maddr) as the request
> cycles through each time. This requires detection of the spiral and
> selection of a different address each time. This can be done statelessly,
> since the previous maddr inserted will appear somewhere in the Record-Route
> list when the request arrives again. This has the drawback of requiring a
> server to be configured with some number of IP addresses.

This is a little bit to much of a hack solution for my taste.
How many IP-addresses do you need ? How many times can you
spirale through a proxy ?

> 
> 2. The difference between the requests coming to P1 the first time and
> second time is not just the top Via, but also the contents of the Route
> header. What you can do is this: when the response to the initial INVITE
> comes, the response carries the complete RR trace. You should be able to
> infer from the request and the response what the Route header set will look
> like by the time the messages reach P1, and this will be different for both
> times the request arrives.

This might be a possibly solution but only if it's totally
forbidden for a proxy to jump off a route chain after some
requests. Since you should add new record-route for every
new request a proxy can technically remove himself from the
list in the middle of a call or is this forbidden ??

> 
> 3. As Sean has pointed out, the State header is another possibility. There
> is the unfortunate problem that this won't work if both UAs don't support
> the State header, so you pretty much need a non-State based solution anyway.

Isn't the State header a separate draft ? Can I require that all
devices have implemented this draft. I would prefer a solution
that is included in RFC2543.

> 
> Note that according to the spec now, the UAS does copy ONLY the maddr and
> port from the RR header; everything else comes from the Contact/From.
> 
> > The first solution is to remove the text where it says
> > that a UA should fill all Routes with values from
> > CONTACT or FROM. Instead the original addresses should
> > be kept. Personally I find it strange when you combine
> > addresses (from CONTACT/FROM) with a port number from
> > RECORD-ROUTE. This means that the total address now is
> > completely useless. It looks to me that you try to
> > avoid syntax changes in the protocol at all costs. I
> > don't understand why a change in semantics is regarded
> > as more backwards compatible than a change in the syntax ?
> 
> Because internal semantic processing does not require communications between
> devices. Thus, you can maintain interoperability. Its the "black box"
> paradigm; you have a hard time changing the interface between components in
> a system, but you can easily change the internal implementation and
> processing logic.
> 
> That aside, the motivation for doing this oddball combination was to try and
> preserve a "fundamental invariant" of SIP, which was that the request URI
> should always refer to an entity that the request is actually addressed to.
> If the UAS simply uses the Record-Routes as is for the Route header, this
> won't be true. The result is a certain amount of brittleness.
> 
> Now, I'm going to open a huge can of worms here.
> 
> First, it seems I am in the minority of folks who think this approach
> (mixing the RR and From/Contact) is sensible. I'm willing to entertain the
> notion that preservation of a protocol invariant is less important than real
> protocol operational problems. Since the maintenance of this invariant is
> causing problems with spirals, perhaps it is not worth the pain, and we
> should return to the original notion of just using the list in the order you
> got it.
> 
> Now, to make things more complex, we have a slight problem that
> record-routing has troubles when the request URI in the incoming request is
> not a SIP URL. Thats because, as you may recall, the RR process in a proxy
> is to COPY the request URI into the RR, and then modify the maddr and port.
> Now, maddr and port are SIP URL parameters. There are no such things for the
> tel URL, for example. Thus, record routing a request with a tel URL is a bit
> problematic. You basically need to convert it to a SIP URL first, or do
> something else. Converting everything to SIP URLs may not always be possible
> down the road. One can make the argument, and I would agree with it, that
> the IP address and port of the proxy are not URI parameters, but
> record-route header parameters. We did the hack of putting them in the URI
> maddr and port since this is backwards compatible.
> 
> So, perhaps we should consider defining address and port record-route
> parameters. These would only be set when the request URI contained a non-SIP
> URL. Proxies would use these parameters if present for routing instead of
> the maddr and port in the URL.
> 
> COmments? (I'm sure there are).
> 
> -Jonathan R.
> 
> >
> > I would have prefered to add a port number in the
> > maddr parameter because that's where I beleave it belongs.
> >
> > The second solution is to make it possible for a server
> > to add it's own parameter in RECORD-ROUTE and then get
> > it back in the REQUEST. Today a UAS only supposed to
> > copy maddr and port value from RECORD-ROUTE to ROUTE.
> > If all parameters was copied instead it would be possible
> > for a server to add anything it likes in RECORD-ROUTE
> > and get it back again. I beleave this kind of solution
> > have been discussed for states ?
> >
> > /Bertil
> >
> > Bertil Engelholm wrote:
> > >
> > > Hi again,
> > >
> > > I'm sorry to bring this up again but I just can't get things
> > > to work in the correct way. I have read through the old threads
> > > regarding this matter and still don't understand how it is
> > > suppose to work.
> > >
> > > It's this example again :
> > >
> > >   ---------                  --------                  --------
> > >   | a@c1  | -- INV u1@p1 --> | (s1) | -- INV u2@p2 --> |      |
> > >   ---------  (contact a@c1)  |      |                  |      |
> > >                              |  p1  |                  |  p2  |
> > >   ---------                  |      |                  |      |
> > >   | b@c2  | <- INV b@c2 ---- | (s2) | <- INV u3@p1 --- |      |
> > >   ---------                  --------                  --------
> > >
> > > If P1 is stateful it might allocate one "call" state machine for
> > > each path the "call" takes in P1 (s1 and s2 in the picture).
> > > These two state machines are identified by the REQUEST-URI
> > > that came in the two INVITEs since that's the only thing that
> > > differs the two from each other. (topmost VIA also differs,
> > > more about that later)
> > >
> > > P1 will now include the REQUEST-URI in the branch value it
> > > generates so that the responses can be directed to the right
> > > state machine when they arrive. And when the ACK is sent the
> > > original REQUEST-URI from ROUTE is used. So no problems here.
> > >
> > > The problem I have is when b@C2 sends a new request (eg. BYE).
> > > According to the specification b should/must/may (or whatever)
> > > overwrite the REQUEST-URI in the ROUTE headers using the
> > > REQUEST-URI from CONTACT or FROM.
> > >
> > > So how should P1 now be able to differ the two (BYE) requests
> > > from each other ? The Original REQUEST-URI was the only thing
> > > that P1 had to identify the two state machines.
> > >
> > > When reading the old threads regarding this matter I saw
> > > something about using topmost VIA in some way, but how ?
> > >
> > > By using the topmost VIA you can see that they are two different
> > > requests but how do you identify which state machine should receive
> > > the first request. (Someone might argue that it doesn't
> > matter but if
> > > P1 controls a firewall it DO matter. Maybe not for BYE but for a
> > > RE-INVITE it matters).
> > >
> > > --
> > > Bertil Engelholm
> > > AXE Research and Development        voice : +46 8 727 3499
> > > SIP Security                        Fax   : +46 8 647 8276
> > > S-126 25 Stockholm Sweden           E-mail:
> > > Bertil.Engelholm@uab.ericsson.se
> > >
> > > _______________________________________________
> > > SIP mailing list
> > > SIP@lists.bell-labs.com
> > > http://lists.bell-labs.com/mailman/listinfo/sip
> >
> > --
> > Bertil Engelholm
> > AXE Research and Development        voice : +46 8 727 3499
> > SIP Security                        Fax   : +46 8 647 8276
> > S-126 25 Stockholm Sweden           E-mail:
> > Bertil.Engelholm@uab.ericsson.se
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> >
> 
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Bertil Engelholm
AXE Research and Development        voice : +46 8 727 3499
SIP Security                        Fax   : +46 8 647 8276
S-126 25 Stockholm Sweden           E-mail:
Bertil.Engelholm@uab.ericsson.se

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Jonathan Rosenberg writes:

 > As Henning pointed out previously, allowing a UAC to insert a telephone
 > number into a request, which is then passed to the PSTN (via a gateway) as
 > the originating number, is a recipe for a security disaster. Nothing stops
 > me from inserting fake telephone numbers, and calling people pretending to
 > be someone else. You can expect a call from the FCC about that one.

why is that?  the proxy receives the invite message and authenticates
it.  then it checks from the user database if the telephone number in
the from field is among those that has been allocated to the user.  if
so, it accepts the call and passes it to the uas at the pstn gw. if not,
it rejects the invite.  the uas only talks to the proxy and thus knows
that it can use the number in the from field as calling party number.

 > So, in fact, this proposed used of the Also-From header is, in practice,
 > unusable without a massive security/PKI infrastructure which really doesn't
 > exist (you would need someone to certify that joe@foo.com owns the number
 > 555-1212. I don't even know who would be able to adequately certify this,
 > and I suspect no CA in their right mind would ever put this in a
 > certificate). 

if the user is my pstn customer, i have a record of the numbers that
have been allocated to it.
 > 
 > Perhaps I sound harsh, but I am trying really hard to push back on everyones
 > "lets add X to SIP" request. SIP is already an RFC, folks. We are obliged to
 > carefully scrutinize everything that gets added, and only put things in when
 > they are truly needed, broadly useful, and backwards compatible. This
 > Also-From thing has only one proposed application, which is interoperation
 > with the PSTN, and in this application is frought with serious security
 > issues. 

i too think that it is very bad idea to keep on adding new headers to
sip requests.  in some cases though, it would be nice if the proxy would
be allowed to rewrite the from address.

-- juha

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Date: Mon, 06 Nov 2000 15:24:41 +0100
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Jonathan Rosenberg wrote:
<snip>

> So, perhaps we should consider defining address and port record-route
> parameters. These would only be set when the request URI contained a non-SIP
> URL. Proxies would use these parameters if present for routing instead of
> the maddr and port in the URL.
>
> COmments? (I'm sure there are).
> -Jonathan R.

I like this approach except I would take it a step further and
do this for all URLs including SIP URLs. This allows maddr to go
back to its original intention for SIP URLs. Plus I hate special
cases; this seems (nominally) easier to implement.

/sean
--
Sean Olson <sean.olson@ericsson.com>
Ericsson Inc.


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From sip-admin@lists.bell-labs.com  Mon Nov  6 09:32:28 2000
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Neil Deason wrote:
> 
> Jonathan Rosenberg wrote:
> 
> [snip]
> 
> 
> > Now, to make things more complex, we have a slight problem that
> > record-routing has troubles when the request URI in the incoming request is
> > not a SIP URL. Thats because, as you may recall, the RR process in a proxy
> > is to COPY the request URI into the RR, and then modify the maddr and port.
> > Now, maddr and port are SIP URL parameters. There are no such things for the
> > tel URL, for example. Thus, record routing a request with a tel URL is a bit
> > problematic. You basically need to convert it to a SIP URL first, or do
> > something else. Converting everything to SIP URLs may not always be possible
> > down the road. One can make the argument, and I would agree with it, that
> > the IP address and port of the proxy are not URI parameters, but
> > record-route header parameters. We did the hack of putting them in the URI
> > maddr and port since this is backwards compatible.
> >
> > So, perhaps we should consider defining address and port record-route
> > parameters. These would only be set when the request URI contained a non-SIP
> > URL. Proxies would use these parameters if present for routing instead of
> > the maddr and port in the URL.
> 
> There is a potential problem here and this looks like a
> reasonable solution. Pitty we can't extend it to all URLs.

This is close to a solution I would like to have. I would
like to always have RR-parameters instead of URI-parameters.
The UA MUST always copy the complete RR-header(s) (including 
RR-parameters) into ROUTE. The UAS then take the contact/from
URI and copy it into ROUTE-URI. The only problem is to get the
RR-parameters back to the server that created them. Either
the servers have to copy the complete ROUTE header up as
request-URI (including the original RR-parameters) or the
ROUTE behaviour is changed so it behaves as VIA ie. a server
removes it's "own" ROUTE header instead of removing the next
servers ROUTE header. In this way you can add any parameters
you like in the RECORD-ROUTE and then get it back in ROUTE.

This is maybe to ask for too much but I beleave such a solution
would open up many new possibilities i.e. state transfer,
call state identification etc.  

/Bertil

-- 
Bertil Engelholm
AXE Research and Development        voice : +46 8 727 3499
SIP Security                        Fax   : +46 8 647 8276
S-126 25 Stockholm Sweden           E-mail:
Bertil.Engelholm@uab.ericsson.se

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From sip-admin@lists.bell-labs.com  Mon Nov  6 09:56:06 2000
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From: "Robert Sparks" <rsparks@dynamicsoft.com>
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Subject: RE: referred-by syntax (was RE: [SIP] (no subject))
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Date: Mon, 6 Nov 2000 08:53:20 -0600
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It's going to be a few more days before I can get the next
revision of the REFER draft out. Until then, here's what
list discussion has guided the Referred-By syntax into.

The motivations for change were

1) Allow URLs containing semicolons to be parsed correctly. This 
   introduced changing the referrer-url from SIPURL to name-addr |
   addr-spec. Allowing the referrenced URL to be separated is a
   little more difficult since it is not necessarily a SIP URL.
   This syntax attempts to resolve the problem by wrapping the URL
   with <> and requiring any <>s appearing in the URL to be escaped.

2) Allow the parameters to be completely unordered. This requires
   more text to augment the ABNF.

-------------------
     Referred-By      = ("Referred-By" | "b") ":"          referrer-url  
                                                  *(   ";" referenced-url 
                                                     | ";" ref-signature  )
     referrer-url     = ( name-addr | addr-spec )
     referenced-url   = "ref" "=" "<" URL ">" 
     ref-signature    = signature-scheme *( ";" sig-scheme-params ) 
     signature-scheme = "scheme" "=" token 
     sig-scheme-parms = token "=" ( token | quoted-string ) 

A Referred-By header MUST contain exactly one referenced-url. 
A Referred-By header MUST contain either zero or one ref-signature.
Any occurences of "<" or ">" within the URL value in referenced-url
must be escaped.

------------

RjS

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From: "Drage, Keith (Keith)" <drage@lucent.com>
To: "'james jack'" <sipjames@yahoo.com>,
        "SIP@lists.bell-labs.com" <sip@lists.bell-labs.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
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Date: Mon, 6 Nov 2000 15:39:17 -0000

Some minor tweaks to the original proposal, with the aim of aligning all the
text into a common format, and ensuring that we have a clearly specified set
of sender and receiver requirements, and not with any aim of technically
changing it..

> The header fields required, optional, and not applicable for each method
> are listed in Table 4 and Table 5. The table uses "o" to indicate
> optional, "m" mandatory and "-" for not applicable. 
> 
	"Optional" means that a UAC \MAY include the header in a request,
and a UAS \MAY ignore the header if present in the request. Similarly, an
optional response header implies that a UAS \MAY place the header in a
response, and the UAC \MAY ignore the header if present in a response it
receives. 

	"Mandatory" means the UAC \MUST include the header in a request, and
a UAS \MUST understand the header if present in a request it receives.
Similarly, a mandatory response header implies that a UAS \MUST place the
header in a response, and the UAC \MUST understand the header if present in
a response it receives. 

> "Not applicable" means the UAC \SHOULDNOT include the header in a request,
> and a UAS \SHOULD ignore the header if present in a request it receives.
> Similarly, a not applicable response header implies that a UAS \SHOULDNOT
> place the header in a response, and the UAC \SHOULD ignore the header if
> present in a response it receives. 
> 
> "m*" means the UAC \SHOULD include the header in a request, and a UAS
> \MUST process requests without that header field. Similarly, a mandatory
> response header implies that a UAS \SHOULD place the header in a response,
> and the UAC \MUST process responses without that header field. 
> 
Editorial note: Not sure about the phrasing of the receiver requirement in
the above paragraph, but there is definitely a receiver requirement needed.

> A "*" indicates that the header fields are needed only if message body is
> not
> empty. See sections 6.18, 6.19 and 8 for details.
> 
Keith

Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com

> ----------
> From: 	Jonathan Rosenberg[SMTP:jdrosen@dynamicsoft.com]
> Sent: 	31 October 2000 16:58
> To: 	'james jack'; SIP@lists.bell-labs.com
> Subject: 	RE: [SIP] discussion
> 
> These are ignored. Thats what optional means, although this probably needs
> to be clarified. The text in Section 6 currently reads:
> 
> The header fields required, optional and not applicable for each method
> are
> listed in Table 4 and Table 5.
> The table uses "o" to indicate optional, "m" mandatory and "-" for not
> applicable. "m*" indicates a header
> that SHOULD be sent, but servers need to be prepared to receive requests
> without that header field. A "*"
> indicates that the header fields are needed only if message body is not
> empty. See sections 6.18, 6.19 and 8
> for details.
> 
> 
> I would suggest:
> 
> The header fields required, optional, and not applicable for each method
> are
> listed in Table 4 and Table 5. The table uses "o" to indicate optional,
> "m"
> mandatory and "-" for not applicable. Optional means that a UAC \MAY
> include
> the header in a request, and a UAS \MAY ignore the header if present in
> the
> request. Similarly, an optional response header implies that a UAS \MAY
> place the header in a response, and the UAC \MAY ignore the header if
> present in a response it receives. A mandatory request header means the
> header \MUST be present in a request, and \MUST be understood by the UAS
> receiving the request. A mandatory response header means the header \MUST
> be
> present in the response, and the header \MUST be understood by the UAS
> processing the response. "Not applicable" for request headers means the
> header \SHOULDNOT be present in a request. If one is placed in a request,
> it
> \SHOULD be ignored by the UAS receiving the request. Similarly, a header
> "not applicable" for a response means the UAS \SHOULDNOT place the header
> in
> the response, and the UAC \SHOULD ignore the header in the response. "m*"
> indicates a header
> that SHOULD be sent, but servers need to be prepared to receive requests
> without that header field. A "*"
> indicates that the header fields are needed only if message body is not
> empty. See sections 6.18, 6.19 and 8
> for details.
> 
> 
> 
> Sound OK?
> 
> 
> 
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>  
> 
> > -----Original Message-----
> > From: james jack [mailto:sipjames@yahoo.com]
> > Sent: Tuesday, October 31, 2000 7:02 AM
> > To: SIP@lists.bell-labs.com
> > Subject: [SIP] discussion
> > 
> > 
> > Dear All:
> >       I have one question to ask, If an request
> > contains 
> > One header that is not applicable according to
> > RFC2543bis
> > 02, for example, in REGISTER request, there appear
> > Alert-info, Priority,
> > how to cope with this request, 400 response code or
> > discard this header
> > and continue to process? 
> >       Could anyone give me some advice or suggestion
> > for processing such stuation? 
> >      Best regards!
> >                                Yours:james
> >   
> > 
> > 
> > __________________________________________________
> > Do You Yahoo!?
> > Yahoo! Messenger - Talk while you surf!  It's FREE.
> > http://im.yahoo.com/
> > 
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> > 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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Subject: Re: [SIP] Help with installation of sipd
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On Friday, November 3 2000, "Tajindrapal Singh" wrote to "sip@lists.bell-labs.com" saying:

> Hi,
>    We are trying to install sipd (columbia release) on a linux box. But, before compiling sipd server, do we need to install the LDAP (Umich distribution)? If so then I guess we need root access for installation of LDAP??

No -- all you need is write access to a directory where you can install ldap
binaries.

>    In the README.build of sipd there is an example for configuring  as follows:
> 
>   ./configure --with-db=/proj/irt-gc4/cinema/sun5 \
>               --with-ldap=/proj/irt-gc4/cinema/sun5
> 
>  Now, we are not sure as to what should be in the sun5 directory? Should there be the sources for LDAP and berkely db? OR Is it the executables that needs to be present in this Any?

That should be the root of the installation directory for ldap and Berkeley
DB.   I.e., on our platform, the directory /proj/irt-gc4/cinema/sun5/lib
contains libdb.a, liblber.a, and libldap.a, and
/proj/irt-gc4/cinema/sun5/include contains db.h, lber.h, and ldap.h (and
other support files).

-- 
Jonathan Lennox
lennox@cs.columbia.edu

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Subject: [SIP] CFP: 2nd IP Telephony Workshop - Deadline 11/27/00
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                               Call for Papers

                         2nd IP Telephony Workshop

            April 2-3, 2001 - Columbia University, New York City
                 http://www.fokus.gmd.de/events/iptel2001/


Objectives
----------
Internet telephony is rapidly evolving from research to design
and deployment. The objectives of the IP Telephony Workshop are
to bring together researchers, developers, vendors and service
providers active in this area and stimulate discussion on
innovation, research, implementation, deployment experiences and
future directions.

Scope & Topics
--------------
Original technical articles related to IP telephony are solicited.
Only papers with significant technical content, not "white papers"
or tutorials, will be considered for publication:

     - Research papers (unique ideas, novel algorithms,
       architectures, measurements, theoretical and/or analytical
       contributions)
     - Surveys, state-of-the-art studies, technology comparisons
     - Implementation and deployment reports
     - Standardization reports

Particular areas of interest include, but are not limited to, the
following:

     - Integration with Internet services (e.g., web, instant
       messaging, games)
     - Added-value services (e.g., call centers, conferencing)
     - Mobility and 3rd generation wireless
     - Authentication, authorization, accounting, charging,
       settlement
     - QoS support
     - Security (e.g., privacy, authentication, certification
       authorities, firewall traversal)
     - Call signaling & processing
     - Feature creation
     - Supporting services (e.g., call routing, lookup services)
     - Audio & video encoding and transmission
     - Management and provisioning
     - Interworking with the PSTN
     - Design and deployment considerations (e.g., performance,
       scalability, reliability)

Important Dates
---------------
Full paper due                November 27th, 2000
Notification of acceptance    January 12th, 2001
Final version due             January 31st, 2001
Program published and         February 5th, 2001
registration opens
Workshop                      April 2nd-3rd, 2001

Submission Instructions
-----------------------
Authors are invited to submit full papers written in English
before November 27th, 2000. The submissions will be reviewed, and
accepted papers will be included in the program. Notifications of
acceptance will be sent out on January 12th, 2000. Deadline for
submission of camera-ready copies is January 31st, 2001. Authors
of accepted papers will need to sign a Copyright Transfer Form and
submit a Netbib entry.

Papers must be submitted electronically using the Web site at 
          http://www.cs.columbia.edu/iptel
Submissions must be in PDF or Postscript; any other documents 
cannot be accepted. Postscript papers must use only standard
PostScript fonts: Times Roman, Courier, Symbol, and Helvetica.  
Papers must be formatted according to the IEEE Transactions format
except for the font size, which MUST be 11pt. Templates are
available at the Web site
          http://www.fokus.gmd.de/events/iptel2001/cfp/
Because of the size limitation on the final manuscript, and to
ensure that the reviewed paper and the final version have a
similar size, papers with more than 11 pages cannot be reviewed.
Submissions must include: title, authors, affiliation, abstract,
list of keywords, and contact information. One of the authors of
each accepted paper must present the paper at iptel'2001.

Contact Address
---------------
Please, send all your inquiries regarding iptel2001 to
               iptel2001@egroups.com.

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Subject: [SIP] 200 OK Contact, firewalls, record-route
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Date: Mon, 6 Nov 2000 12:55:36 -0500 (EST)


I have a couple of questions regarding the interaction of the
contact header in the 200 OK, firewalls, and record-route
 
The spec says that a UA behind a firewall, provides the ip of the
firewall as its contact address, and if record-route headers exist,
this address is appended (by the UAC) to the route for subsequent
requests.

My question is this - what if the firewall proxy, due to some
independent feature logic, already added a record-route of itself
to the request ? Should the UAC that builds the route, verify that
the route is not already present before appending the contact address ?

Along those same lines, what should a proxy do that sees itself 
as the next route ? The spec says it should forward it unless that
will cause a loop. So, should it return a loop detected error ?,
pop that route, and forward it to the next route ?, or route
based upon the URI ?

thanks,

david



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Date: Mon, 06 Nov 2000 11:06:57 -0800


--------------C7AE93654F0CB74A72659D57
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Hello,
    according to section B. 6 in a new draft it is possible to put the
other party "on hold",
that is to ask it temporarily stop sending media, by sending re-INVITE
with the destina-
tion addresses (c= lines) set to 0.
What should we expect in 200 OK ? Should it be (upon acception of
re-INVITE) the
regular SDP with valid ports and destination addresses, since it stops
sending, but
still continue receiving ?

Thanx.


--
Vladislav Zubarev        mailto:vladis@vovida.com
Software Engineer
Vovida Networks, Inc.    http://www.vovida.com



--------------C7AE93654F0CB74A72659D57
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Hello,
<br>&nbsp;&nbsp;&nbsp; according to section B. 6 in a new draft it is possible
to put the other party "on hold",
<br>that is to ask it temporarily stop sending media, by sending re-INVITE
with the destina-
<br>tion addresses (c= lines) set to 0.
<br>What should we expect in 200 OK ? Should it be (upon acception of re-INVITE)
the
<br>regular SDP with valid ports and destination addresses, since it stops
sending, but
<br>still continue receiving ?
<p>Thanx.
<br>&nbsp;
<pre>--&nbsp;
Vladislav Zubarev&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <A HREF="mailto:vladis@vovida.com">mailto:vladis@vovida.com</A>
Software Engineer&nbsp;&nbsp;&nbsp;
Vovida Networks, Inc.&nbsp;&nbsp;&nbsp; <A HREF="http://www.vovida.com">http://www.vovida.com</A></pre>
&nbsp;</html>

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Date: Mon, 6 Nov 2000 14:13:07 -0500

As we approach the next SIP Bake-off, we are trying to improve the
body of tests called "Torture Tests".  These are SIP message sequences
that various parsers have had trouble with in the past.  New implementations
typically use these tests as the first level of interoperability testing.

We would like to have message sequences that you think are legal, but you
have seen a parser have trouble for one reason or another.  It doesn't
really matter what the message does, just that it be something a good
implementation should be able to cope with.

We are not really looking for you to get downright devious about it,
but we would like all implementations to not choke on legal messages.

If you have some, please email them to:
	Neil Deason [ndeason@ubiquity.net]
who has volunteer to collate them.

Thanks
  Brian

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Date: Fri, 3 Nov 2000 13:25:07 -0600

Sorry for the basic question.. but:

How can two SIP end points negotiate QoS, Bandwidth, vocoders ? I mean how it is done as part of the call setup? how it may be done during the call?
What headers are used for that (SIP headers?, SDP headers?)

Thanks

Uri



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Subject: RE: [SIP] Digest Authentication
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With respect to MD5 authentication, is the following the correct way to
compute the response tag of the authentication header for a 401
Unauthorized? I am assuming cnonce, noncecount and qop are not specified.

1. get the method name from CSeq of the response
2. get nonce, realm from the WWW-Authenticate header
3. Query uid and password from user

4. Following RFC 2617, setup the following:
 nonce      = WWW-Authenticate::nonce
 cnonce     = ""
 user       = query_user_for_uid()
 realm      = WWW-Authenticate::realm
 password   = query_user_for_password()
 algorithm  = "MD5"
 noncecount = ""
 method     = CSeq::method
 qop        = ""
 uri        = registrar

5. Calculate DigestCalcHA1(algorithm, user, realm, password, nonce,cnonce);
6. Calculate DigestCalcResponse(HA1, nonce,noncecount, cnonce, qop,
                                method, uri, HA2, response);

thanks,
Sunil Veluvali
Sylantro Systems

-----Original Message-----
From: Sunitha Kumar [mailto:skumar@vovida.com]
Sent: Friday, October 20, 2000 10:54 AM
To: Mark Wisdom
Cc: sip@lists.bell-labs.com
Subject: Re: [SIP] Digest Authentication


Mark: 
With respect to your question about mutual authentication, maybe PGP could
be considered as the way to go. 
Using PGP, the server can authenticate the client, and the client could
verify that this is the server. 
Using Digest for mutual authentication may require 2 independent msgs to go
from the server to client and client to server, 
each sending each other their generated nonces, and the other forming the
digest with that nonce. 
Just my 2c. 
sunitha 
  
  
  
Mark Wisdom wrote: 
I need some clarification on, and help with, digest authentication. 
rfc2617, section 3.2.1 states: 
> A nonce might, for example, be constructed as the base 64 
> encoding of 
> 
>     time-stamp H(time-stamp ":" ETag ":" private-key) 
Then in section 4.5 it states: 
> The server created "nonce" value is implementation dependent, 
> but if it contains a digest of the client IP, a time-stamp, 
> the resource ETag, and a private server key (as recommended 
> above) then a replay attack is not simple. 
rfc2543bis-02, section 14.3 states: 
> 4. The example procedure for choosing a nonce based on Etag 
>    does not work for SIP. 
It would be nice if rfc2543 provided an alternate example for 
creating a nonce. Can this be added to the rfc2543? 
Does anyone have any guidelines on creating a nonce for use 
with digest authentication? 
rfc2543bis-02, section 14.3 also states: 
> 5. The Authentication-Info and Proxy-Authentication-Info 
>    fields are not used in SIP. 
If I understand correctly, not allowing the Authentication-Info 
header eliminates support for mutual authenication (i.e. 
allowing the server to also authenticate itself to the client 
by proving to the client that it knows the client's secret). 
Does SIP allow for mutual authentication when using digest 
authentication? 
If not, can we modify rfc2543 so that mutual authentication is 
possible? 
Thanks, 
Mark Wisdom 
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If the held party wishes to still listen for something,
it can send the same SDP as it previously sent.

If the held party wishes the holder to quit
streaming, it can increment the SDP and
set the "c=" <connection address> to 0.0.0.0 .

----- Original Message -----
From: Vladislav Zubarev
To: sip@lists.bell-labs.com
Sent: Monday, November 06, 2000 2:06 PM
Subject: [SIP] putting media on hold


Hello,
    according to section B. 6 in a new draft it is possible to put the other
party "on hold",
that is to ask it temporarily stop sending media, by sending re-INVITE with
the destina-
tion addresses (c= lines) set to 0.
What should we expect in 200 OK ? Should it be (upon acception of re-INVITE)
the
regular SDP with valid ports and destination addresses, since it stops
sending, but
still continue receiving ?
Thanx.

--
Vladislav Zubarev        mailto:vladis@vovida.com
Software Engineer
Vovida Networks, Inc.    http://www.vovida.com



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What if the other party wishes still to receive media stream, is there any way
we can get
one-way stream, I mean the situation when one sends re-INVITE with RTP port set
to 0
for some particular m= line, however another party replies with SDP with valid
ports ?
Do we get one-way stream in this case ?
Thanx.

Jonathan Rosenberg wrote:

>
> -----Original Message-----
> From: Vladislav Zubarev [mailto:vladis@vovida.com]
> Sent: Tuesday, October 31, 2000 3:48 PM
> To: sip@lists.bell-labs.com
> Subject: [SIP] Deleting media streams
>
> >Hello,
> >    I have a question about deleting media stream procedure, described in a
> new SIP draft
> >(section B. 5).
> >
> >Let's say we opened two media streams, so now we send re-INVITE with SDP,
> where one of the
> >"m="
> >
> >lines contains RTP port set to 0 - we would like to delete this particular
> media stream.
> >
> >My question is what should we expect in 200 OK's SDP: should it contain the
> same "m=" line
> >with
> >
> >RTP port set to 0 or this "m=" line should be missing at all ?
>
> Same m line with port 0.
>
> There must always be the same number of m lines in the request as in the
> response.
>
> -Jonathan R.
>
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com

--
Vladislav Zubarev        mailto:vladis@vovida.com
Software Engineer
Vovida Networks, Inc.    http://www.vovida.com



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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
What if the other party wishes still to receive media stream, is there
any way we can get
<br>one-way stream, I&nbsp;mean the situation when one sends re-INVITE
with RTP&nbsp;port set to 0
<br>for some particular m= line, however another party replies with SDP
with valid ports ?
<br>Do we get one-way stream in this case ?
<br>Thanx.
<p>Jonathan Rosenberg wrote:
<blockquote TYPE=CITE>&nbsp;
<br>-----Original Message-----
<br>From: Vladislav Zubarev [<a href="mailto:vladis@vovida.com">mailto:vladis@vovida.com</a>]
<br>Sent: Tuesday, October 31, 2000 3:48 PM
<br>To: sip@lists.bell-labs.com
<br>Subject: [SIP] Deleting media streams
<p>>Hello,
<br>>&nbsp;&nbsp;&nbsp; I have a question about deleting media stream procedure,
described in a
<br>new SIP draft
<br>>(section B. 5).
<br>>
<br>>Let's say we opened two media streams, so now we send re-INVITE with
SDP,
<br>where one of the
<br>>"m="
<br>>
<br>>lines contains RTP port set to 0 - we would like to delete this particular
<br>media stream.
<br>>
<br>>My question is what should we expect in 200 OK's SDP: should it contain
the
<br>same "m=" line
<br>>with
<br>>
<br>>RTP port set to 0 or this "m=" line should be missing at all ?
<p>Same m line with port 0.
<p>There must always be the same number of m lines in the request as in
the
<br>response.
<p>-Jonathan R.
<p>---
<br>Jonathan D. Rosenberg&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
72 Eagle Rock Ave.
<br>Chief Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
First Floor
<br>dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
East Hanover, NJ 07936
<br>jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
FAX:&nbsp;&nbsp; (973) 952-5050
<br><a href="http://www.cs.columbia.edu/~jdrosen">http://www.cs.columbia.edu/~jdrosen</a>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
PHONE: (973) 952-5000
<br><a href="http://www.dynamicsoft.com">http://www.dynamicsoft.com</a></blockquote>

<pre>--&nbsp;
Vladislav Zubarev&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <A HREF="mailto:vladis@vovida.com">mailto:vladis@vovida.com</A>
Software Engineer&nbsp;&nbsp;&nbsp;
Vovida Networks, Inc.&nbsp;&nbsp;&nbsp; <A HREF="http://www.vovida.com">http://www.vovida.com</A></pre>
&nbsp;</html>

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Date: Mon, 6 Nov 2000 19:52:43 -0800


There's seems to be a lot of discussion lately about out-of band Route in
INVITE. Here's a another reason for correct processing of "out-of-band"
Route headers ("out-of-band" with respect to an INVITE initiated call-leg): 

When a UAC makes a OPTIONS request before performing a full INVITE, the UAS
may include a Contact in the 200 OK which be used as the direct destination
to send the initial INVITE. If a proxy needs to be on the path of the
initial INVITE (eg proxy is acting as ALG) then the call will fail if the
proxy and UAC doesn't take some additional action.

DOn't we need to add something such as the following to the spec?

1)A proxy SHOULD insert a Record-Route entry for itself on all OPTION's
requests, if it would have inserted itself on a corresponding INVITE.
2) UAC SHOULD generate a Route header (from the Record-Route in the OPTIONS
response) in the subsequent initial INVITE request if 
	- a Contact header in the OPTIONS 200 OK request is present and
used.
	- a Record-Route header is also present in the OPTIONS response.

[Actually a proxy should only need to insert itself into the Record-Route
header of the OPTIONS _response_ if the response contains a Contact (because
the reverse route will never be needed to an OPTIONS request); however this
just complicates behaviours and so I the above is a better recommendation
for the spec.]

Regards,

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 


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Date: Tue, 7 Nov 2000 01:37:53 -0500


  
-----Original Message-----
>From: Vladislav Zubarev [mailto:vladis@vovida.com]
>Sent: Monday, November 06, 2000 8:21 PM
>To: Jonathan Rosenberg; sip@lists.bell-labs.com
>Subject: Re: [SIP] Deleting media streams
>
>
>What if the other party wishes still to receive media stream, is there any
way we can get 
>one-way stream, I mean the situation when one sends re-INVITE with RTP port
set to 0 
>for some particular m= line, however another party replies with SDP with
valid ports ? 
>Do we get one-way stream in this case ? 

No. In the case of a stream deletion, the response should always have a port
of zero as well.

If a UA wants to change a stream to send only or recv-only, it can do that
by changing the sdp attribute for directions.

-Jonathan R.


-----Original Message----- 
From: Vladislav Zubarev [mailto:vladis@vovida.com] 
Sent: Tuesday, October 31, 2000 3:48 PM 
To: sip@lists.bell-labs.com 
Subject: [SIP] Deleting media streams 
>Hello, 
>    I have a question about deleting media stream procedure, described in a

new SIP draft 
>(section B. 5). 
> 
>Let's say we opened two media streams, so now we send re-INVITE with SDP, 
where one of the 
>"m=" 
> 
>lines contains RTP port set to 0 - we would like to delete this particular 
media stream. 
> 
>My question is what should we expect in 200 OK's SDP: should it contain the

same "m=" line 
>with 
> 
>RTP port set to 0 or this "m=" line should be missing at all ? 
Same m line with port 0. 
There must always be the same number of m lines in the request as in the 
response. 
-Jonathan R. 
--- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave. 
Chief Scientist                             First Floor 
dynamicsoft                                 East Hanover, NJ 07936 
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050 
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000 
http://www.dynamicsoft.com
-- 
Vladislav Zubarev        mailto:vladis@vovida.com
Software Engineer   
Vovida Networks, Inc.    http://www.vovida.com
  

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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Date: Tue, 7 Nov 2000 01:44:29 -0500



 

> -----Original Message-----
> From: David Daiker [mailto:ddaiker@cisco.com]
> Sent: Monday, November 06, 2000 12:56 PM
> To: sip@lists.bell-labs.com
> Subject: [SIP] 200 OK Contact, firewalls, record-route
> 
> 
> 
> I have a couple of questions regarding the interaction of the
> contact header in the 200 OK, firewalls, and record-route
>  
> The spec says that a UA behind a firewall, provides the ip of the
> firewall as its contact address, and if record-route headers exist,
> this address is appended (by the UAC) to the route for subsequent
> requests.

Be careful here. This treatment of Contact headers has about 1/2 sentence
mention in the spec. I guess it deserves more explanation.

The contact header sets the address where subsequent requests go. Thats it.
Now, a smart network can be set up whereby the UA inserts the address of the
firewall, and the firewall knows that this is happening, so it doesn't
record-route, and knows to translate this to something internal when
requests arrive.

In other words, this capability needs to be used in a coordinated fashion.
The proxy must know that the UA is doing something funky here. In general
usage, it should not make this assumption.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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Date: Tue, 7 Nov 2000 01:58:42 -0500



 

> -----Original Message-----
> From: Neil Deason [mailto:ndeason@ubiquity.net]
> Sent: Monday, November 06, 2000 5:14 AM
> To: Jonathan Rosenberg
> Cc: 'Bertil Engelholm'; SIP@lists.bell-labs.com
> Subject: Re: [SIP] Identification problem
> 
> 
> Jonathan Rosenberg wrote:
> > 
> > > Am I missing the problem? The other RR suggestions are
> > > certainly interesting ;)
> > 
> > The problem is that the unique via branch param allows you to
> > *differentiate* between each request, but there is no easy 
> way to know that
> > a request corresponds to a specific state machine. Thats 
> because (1) the
> > proxy immediately before and/or after P1 may not actually 
> record-route, so
> > the branch param in the original INVITE could be totally 
> different from the
> > one in the re-INVITE or BYE, and (2) proxy P1 has no way to 
> know what the
> > branch param for requests in the REVERSE direction will 
> look like after the
> > original INVITE.
> 
> I agree there is this issue but is it really a problem for 
> the spec? 

It depends. The problem here is that if the proxy has some state which is
different for each iteration of the request through the proxy, is there a
way to associate each request with the right piece of state? I think this is
a reasonable thing to expect to be possible, so the protocol should have
some way of supporting it.

Another potential solution is to slightly change the way Route headers are
constructed. Right now, the spec says:

----
If a UA finds a Record-Route header in a request received as a UAS, it
copies the Record-Route
maddr parameters and any port value, maintaining their ordering, to the
Route header field of future
requests issued as a UAC. Since the URIs contained in the Record-Route
header fields are not useful for
the reverse request path, the UA fills all other components of the Route
name-addr value with the name-addr
value found in the Contact or the From header field. The latter is used only
if there is no Contact
header field.
----

as I pointed out, this causes everything from the Contact/From to be copied
into Route, EXCEPT port and maddr. How about if we say that the Route
contains the user:password@host portion of the Contact/From, and everything
else is copied from the Record-Route? THis would include unknown URL
parameters.

The advantage here is that one could insert URL parameters into the
record-route, and get them back in the request URI. Not the prettiest way to
go. We could, as an alternative, simply mandate the State header in the bis
draft, which would provide a cleaner solution.

>> So, perhaps we should consider defining address and port record-route
>> parameters. These would only be set when the request URI contained a
non-SIP
>> URL. Proxies would use these parameters if present for routing instead of
>> the maddr and port in the URL.
>>
>> COmments? (I'm sure there are).
>> -Jonathan R.
>
>I like this approach except I would take it a step further and
>do this for all URLs including SIP URLs. This allows maddr to go
>back to its original intention for SIP URLs. Plus I hate special
>cases; this seems (nominally) easier to implement.

Conceptually that is clearly the better way. However, there is a backwards
compatibility problem. Older rfc2543 implementations will use the maddr to
forward (if they are working correctly, that is), but not an address
parameter of the Route header.

-Jonathan R.


---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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Date: Tue, 7 Nov 2000 02:07:52 -0500



 

> -----Original Message-----
> From: Fairlie-Cuninghame, Robert [mailto:rfairlie@nuera.com]
> Sent: Monday, November 06, 2000 10:53 PM
> To: sip@lists.bell-labs.com
> Subject: [SIP] OPTIONS and out-of-band Route.
> 
> 
> 
> There's seems to be a lot of discussion lately about out-of 
> band Route in
> INVITE. Here's a another reason for correct processing of 
> "out-of-band"
> Route headers ("out-of-band" with respect to an INVITE 
> initiated call-leg): 
> 
> When a UAC makes a OPTIONS request before performing a full 
> INVITE, the UAS
> may include a Contact in the 200 OK which be used as the 
> direct destination
> to send the initial INVITE. If a proxy needs to be on the path of the
> initial INVITE (eg proxy is acting as ALG) then the call will 
> fail if the
> proxy and UAC doesn't take some additional action.
> 
> DOn't we need to add something such as the following to the spec?

Well, this is a symptom of a more general problem, which is what is the
meaning of Record-Routing in requests in general, beyond INVITE. I would
propose the following:

If a proxy inserts a record-route into a request, that means it wishes to
receive all subsequent requests, in both directions, that contain the same
value of the To, From, and Call-ID. 

I like this, since then it works for MESSAGE and SUBSCRIBE, which have
similar requirements.

Now, to handle your OPTIONS case:

If a UAC initiates a call to a host as the result of an OPTIONS query, it
SHOULD use the same Call-ID as was used in the OPTIONS query. Note that as a
result of standard Record-Routing rules, this INVITE will contain any
routing information learned from the OPTIONS response.


This is now much more general purpose.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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Date: Tue, 7 Nov 2000 02:22:35 -0500



 

> -----Original Message-----
> From: Keith Robinson [mailto:Keith.Robinson@marconi.com]
> Sent: Thursday, October 26, 2000 12:01 PM
> To: Jonathan Rosenberg
> Cc: 'Christer Holmberg'; sip@lists.bell-labs.com; Igor Slepchin;
> sean.olson@ericsson.com
> Subject: RE: [SIP] Redirect server returning new From header
> 
> 
> 
> 
> With respect to the issue of delivering a calling line 
> identity from a SIP user
> into the PSTN, consider the following.
> 
> A SIP user wishes to be globally reachable; as this means he 
> has to be reachable
> from a POTS subscriber in the PSTN who can only dial digits 
> it follows that the
> SIP user must also be known by a dialled digit string. Therefore, the
> association of the SIP users URL and the dialled digit string 
> to reach this URL
> must be fixed so therefore verifiable; if this were not the 
> case then calls from
> the PSTN into SIP networks could not happen.

Well, such mappings (from a phone number to a SIP URL) are the subject of
enum. Who owns such mappings, and how they are managed, is currently the
subject of intense political and regulatory struggles. I'm hoping to stay
away from such things.

> 
> The UAC should not be allowed to supply  the CLI;  rather it 
> is supplied by the
> network to which he is connected and it is the responsibility 
> of the network to
> ensure that the person is who he says he is (verify FROM:) 
> and ensure that the
> correct CLI is delivered.

In this case, I would argue that what you want is to use the privacy spec:

http://search.ietf.org/internet-drafts/draft-dcsgroup-sip-privacy-02.txt

but always use a telephone number in there. All this network-level
authentication of names is very much a PSTN thing. I suspect the
Remote-Party-ID identification will be used most commonly in PSTN bypass or
PSTN emulation, in which case you only ever want a PSTN number anyway. For
IP to IP signaling, an end to end authentication mechanism is preferrable.

The next best thing is to update the privacy spec to allow multiple
addresses in there. This way, you don't touch the base spec, which is what I
really, really want to avoid.

Simon writes:
> > So, in fact, this proposed used of the Also-From header is, 
> in practice,
> > unusable without a massive security/PKI infrastructure 
> which really doesn't
> > exist (you would need someone to certify that joe@foo.com 
> owns the number
> > 555-1212. I don't even know who would be able to adequately 
> certify this,
> > and I suspect no CA in their right mind would ever put this in a
> > certificate).
> 
> This security infrastructure already exists in the current 
> telephone network -
> although it is implemented in hardware rather than software. 
> Phone companies have a
> network of trust with each other, and exchange ISUP messages 
> with each other only
> secure lines setup along these lines of trust. Where 
> signalling is exchanged with an
> end user (Q.931) checks are made to ensure the user is 
> authorised to send the numbers
> he sends. A similar trust network is required if we want to 
> have authenticated CLI in
> SIP. Currently SIP-T does require exactly the same (physical 
> or secure VPN) trust
> network as regular ISUP telephony. Indeed this is probably 
> one of the main
> constraining factors that limits whom SIP-T is available to.

Huh? SIP-T does not anywhere require the same trust network as regular ISUP
telephony. Where in the drafts does it say that?

Juha writes:
>  > As Henning pointed out previously, allowing a UAC to 
> insert a telephone
>  > number into a request, which is then passed to the PSTN 
> (via a gateway) as
>  > the originating number, is a recipe for a security 
> disaster. Nothing stops
>  > me from inserting fake telephone numbers, and calling 
> people pretending to
>  > be someone else. You can expect a call from the FCC about that one.
> 
> why is that?  the proxy receives the invite message and authenticates
> it.  then it checks from the user database if the telephone number in
> the from field is among those that has been allocated to the user.  if
> so, it accepts the call and passes it to the uas at the pstn 
> gw. if not,
> it rejects the invite.  the uas only talks to the proxy and thus knows
> that it can use the number in the from field as calling party number.

If the calling UA is known only by this number, you are correct. The privacy
specification addresses exactly this requirement. The problem is when I make
a call from my yahoo SIP account (sip:jdrosen@yahoo.com), and the yahoo
proxy needs to insert my phone number into the request. What I am saying is
that since yahoo does not own or adminster the PSTN numbering space, it has
no way to verifiably correlate my yahoo address with my phone number. The
same problem exists in the reverse; my phone company has no easy way to
reliably verify that a valid additional sip address for me is
jdrosen@yahoo.com. 

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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> 
> If a proxy inserts a record-route into a request, that means 
> it wishes to
> receive all subsequent requests, in both directions, that 
> contain the same
> value of the To, From, and Call-ID. 
> 
> I like this, since then it works for MESSAGE and SUBSCRIBE, which have
> similar requirements.

Agreed. Much more general. 
 
> Now, to handle your OPTIONS case:
> 
> If a UAC initiates a call to a host as the result of an 
> OPTIONS query, it
> SHOULD use the same Call-ID as was used in the OPTIONS query. 
> Note that as a
> result of standard Record-Routing rules, this INVITE will contain any
> routing information learned from the OPTIONS response.
> 
My only comment here is that Section 11 & 12 on the behavior of UAC's and
proxies is written from the point of view that a Call State Machine is
created by an "initial INVITE request". I believe it is this idea, that is,
importing "out-of-band" Route information from _before_ the Initial INVITE
Request, that is cause of the recent discussions/queries/confusion in this
area. In other words, in my mind the existing Record-Route rules description
doesn't quite stretch to unambiguously cover the pre-initial-invite
behaviour. I think a small clarification would be helpful.

Cheers,

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 

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Date: Tue, 7 Nov 2000 02:42:00 -0500



  
-----Original Message-----
>From: Samorezov Vladimir [mailto:samorezov@rts.loniis.ru]
>Sent: Saturday, November 04, 2000 4:42 AM
>To: sip@lists.bell-labs.com
>Subject: [SIP] failure of proxy
>
>
>Hi, all!
>Could you help me in my SIPs study?
>If proxy forwards a SIP request, it add Via header with its address. This
ensures responses >take the same path as the requests. OK!
>If one proxy along the path is failure, what upstream proxy must do to
route responses?   
>
>Thank you!
>Samorezov Vladimir.

This question is well timed. I've been having lots of private discussions on
just this subject, and I think we've got the solutions all worked out. There
are two cases here. One is routing of responses, and the other of subsequent
requests.

Regarding responses. The specific case is when proxy P1 forwards a request
to P2, and before the response comes, P1 fails. How does P2 forward the
response? This is related to what gets inserted into the Via header, and its
relation to the received param. Now, the current text states that if the
senders address is wrong, the proxy inserts the received param. However, if
the sent-by field in the Via contains a hostname that resolves to an SRV
record with many IP addresses, what is the meaning of "wrong"? The question
came up on a previous thread a long time ago, and the answer I gave was that
you would insert the received param if the source IP address in the request
didn't match any of the servers listed in DNS. I beleive now that this is
not the right solution. Rather, if the Via header contains a sent-by value
thats a domain name, the next hop should always insert a received param with
the source IP address.

Why is that? To deal with failures. P2 would sent the response to the
address in the received param. If it fails, P2 can instead take the hostname
in the sent-by field, and look that up in DNS to find an alternate server to
forward the response to. THis way, we can route responses around failures,
but under normal operation, ensure responses get routed back to the actual
server which forwarded the request. 

The second case is routing of subsequent requests. Lets say we send an
initial INVITE, and proxy P1 and P2 record-route. Now, midway through the
call, the proxy P2 fails. The caller sends a BYE. This hits P1. It sees a
Route header in the request (pointing to P2), pops it, and tries to send it
there. As this contains an maddr for P2, the request fails. In this case, I
would argue that P1 should then try the SRV procedure once more on the host
portion of the request URI, using the next server in DNS.

For things to work consistently, the SRV lookup process should be based on
the pseudo-random number generator I suggested in a previous thread; thats
the one based on a hash of the Call-ID, amongst other things. This gives us
consistent operation for SRV records in general.

Now, an interesting thing to note here. Lets say in this failure case, it
was the called party that hung up. The proxy downstream from P2 (say, P3),
pops its top Route, and tries to forward it to P2. But, that maddr (pointing
to P2), generates an ICMP error. So, it looks up the host portion of the
request URI in DNS and tries the SRV process from there. I will note that
this will only succeed if, and only if, *THE REQUEST URI ALWAYS REFERS TO
THE RESOUCE FOR WHOM THE REQUEST IS DESTINED*. Now, this relates to a
previous thread. The issue there was how a UAS would construct the Route
headers, and whether we should maintain this property that the R-URI always
refers to the entity for whom the request is targeted. I originally felt
this was critical, and was coming close to backing down. Now, it seems we
have another case to back up my original claim that maintaining this
property is important.

I'll make sure verbiage gets added to the spec on these failure handling
cases.

Thanks,
Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Date: Mon, 6 Nov 2000 23:47:14 -0800

> If the calling UA is known only by this number, you are 
> correct. The privacy
> specification addresses exactly this requirement. The problem 
> is when I make
> a call from my yahoo SIP account (sip:jdrosen@yahoo.com), and 
> the yahoo
> proxy needs to insert my phone number into the request. What 
> I am saying is
> that since yahoo does not own or adminster the PSTN numbering 
> space, it has
> no way to verifiably correlate my yahoo address with my phone 
> number. The
> same problem exists in the reverse; my phone company has no 
> easy way to
> reliably verify that a valid additional sip address for me is
> jdrosen@yahoo.com. 
Hi Jonathan,

Just because the SIP From address is jdrosen@freeemail.com, why is it not
possible for your phone company to identify you and hence determine the
correct CLI for the Remote-Party-Id?
[I am assuming there is no sharing of information between your email
provider and your phone company ..... but who knows what goes on these
days.]

What about "digital me" (www.digitalme.com) or verisign-like identification
certificates that allow multi-realm/universal identity authentication? What
about when these things become more prevalent? I ahven't looked into exactly
how these things work but should you be able to establish identity without a
direct security relationship between SIP endpoints, right? 

Cheers,

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 

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Date: Tue, 7 Nov 2000 02:47:26 -0500

I think this note is simply wrong. I don't even remember noticing it until
you pointed it out. Its in both bis and rfc2543. It should be stricken.

Henning - do you agree?

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: David Shrader [mailto:dshrader@master-consultant.com]
> Sent: Saturday, November 04, 2000 12:29 AM
> To: Henning Schulzrinne
> Cc: SIP List
> Subject: [SIP] rfc2543bis comment/question
> 
> 
> Table 5 contains a note 2 regarding the Via field. It states 
> that the UAS
> removes first Via header field. This seems to be in direct 
> contradiction to
> the purpose of the Via field. When that proxy receives the 
> response, if
> there is no Via field in it, then 1) it can't know that it is 
> meant for it
> and 2) it wouldn't be able to identify the branch without the branch
> parameter.
> 
> Am I missing something? What was meant by the note in Table 5?
> 
> ----------------------------
> David Shrader
> Master Consultant, Inc.
> dshrader@master-consultant.com
> http://www.EyeForTheFuture.com
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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Date: Tue, 7 Nov 2000 02:56:34 -0500

Comments below.

 

> -----Original Message-----
> From: William Marshall [mailto:wtm@research.att.com]
> Sent: Friday, November 03, 2000 4:02 PM
> To: sip@lists.bell-labs.com
> Subject: [SIP] Re: Pre-loaded Route headers
> 
> 
> I believe there may be a problem with Pre-loaded Route headers and
> stateless proxies, but it not a difficult one to solve.  But there
> are side-affects that are not nice.
> 
> An initial INVITE is normally the one that accumulates the 
> Record-Route
> values; then they are stored at the UAS and converted into a 
> Route header
> for UAS->UAC transactions, and echoed back to the UAC in the 200-OK so
> the UAC converts them into a Route header for UAC->UAS transactions.
> A mid-call INVITE (or any other message) contains the Route header
> value, but doesn't need to accumulate Record-Routes again.
> 
> But a stateless proxy can't really tell the difference between an
> initial INVITE for a new session (one in which it should add itself
> to the Record-Route) and a mid-call INVITE for an existing session (in
> which it doesn't need to add itself).  So it has two choices: 
> 1) always
> add itself to a Record-Route, or 2) note there is a Route header and
> assume therefore it is a mid-call message and do nothing.
> 
> Keying on the existence of a Route header to indicate a mid-call
> message isn't explicitely disallowed by the spec (unless I missed
> it somewhere), but it would work just fine - until a UAC tried to
> use a pre-loaded Route header.  Then the proxy wouldn't get itself
> into the Record-Route when it should.
> 
> So the solution is to have the proxy insert itself into the
> Record-Route header *independent of whether it is an initial INVITE
> or a mid-call INVITE*.  
> 
> Unfortunately there is a backward compatibility problem here - a proxy
> that doesn't know to do this for a mid-call message will be dropped
> from future messages.  If even one proxy along the path inserts a
> Record-Route header on the mid-call message, the UAS and UAC will
> trigger their Route re-calculation.  (Presumably if none of 
> the proxies
> did the Record-Route for a mid-call message, the UAC and UAS would
> act like currently, and keep the Route from the initial INVITE).

We had an extensive discussion on the list on just this topic. The result of
that discussion is already in the bis draft:

--
The Record-Route request and response header field is added to a request by
any proxy that insists on
being in the path of subsequent requests for the same call leg. A proxy
SHOULD add it to anyrequestfor
robustness, but a request route, once established, persists until the end of
the call leg, regardless of whether
the Record-Route header is present in subsequent requests.
--

So, what this is saying is that a proxy should always insert the
record-route, but a UA doesn't reload the record-routes if it has a set
already. This avoids the backwards compatibility issue. But, you ask, what
is the point of re-inserting them? THe answer is twofold. First, the
original idea is recovery from crash and restarts. The restarted UA wouldn't
have a record of the Record-Routes, and so when it gets a re-INVITE (via
session timer, perhaps), it will re-establish the route based on the
record-routes in the INVITE. Secondly, we have our recent discussion of
sending invites with loaded routes. If proxies follow the above
recommendation, they will always record-route, and the result is that the
initial forced route will actually get persisted, since the UAS has no
memory of a route when the initial request arrives.

So, what we need here is a little motivational text so people notice this
subtelty and implement it correctly, along with a reccomendation that
proxies SHOULD NOT consider a request with a Route header as indicative of a
mid-call or mid-session message.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Date: Tue, 7 Nov 2000 02:57:55 -0500

The text is generic, and not specific to SDP. SDP has no way to indicate
separate send and receive codecs for the same stream. 

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Baniel Uri-CUB001 [mailto:Uri.Baniel@motorola.com]
> Sent: Friday, November 03, 2000 6:50 PM
> To: SIP@lists.bell-labs.com
> Subject: [SIP] Media to send
> 
> 
> 2543bis 4.2.1 INVITE:
> 
>   "The INVITE method indicates that the user or service is 
> being invited
>    to participate in a session. The message body MAY contain a
>    description of the session to which the callee is being 
> invited. For
>    two-party calls, the caller indicates the type of media it 
> is able to
>    receive and possibly the media it is willing to send as 
> well as their
>    parameters such as network destination. A success response MUST
>    indicate in its message body which media the callee wishes 
> to receive
>    and MAY indicate the media the callee is going to send."
> 
> How does a UA indicate (in case it wishes) the media it is 
> going to send?
> 
> Uri
> 
> _______________________________________________
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> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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Date: Tue, 7 Nov 2000 03:01:37 -0500



 

> -----Original Message-----
> From: Baniel Uri-CUB001 [mailto:Uri.Baniel@motorola.com]
> Sent: Friday, November 03, 2000 2:25 PM
> To: sip@lists.bell-labs.com
> Subject: [SIP] SIP negotiation
> 
> 
> Sorry for the basic question.. but:
> 
> How can two SIP end points negotiate QoS, Bandwidth, vocoders 
> ? 

QoS, bandwidth, and other media stream reservations are handled via QoS
reservation protocols like RSVP. The relationship with SIP signaling is
loose. Worst case, there is a temporal correlation in the two. For more
info, check out the famed "manyfolks" draft:

http://search.ietf.org/internet-drafts/draft-manyfolks-sip-resource-01.txt


As for codecs, its part of the standard SDP usage defined in the appendix B
to RFC2543.


-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Date: Tue, 7 Nov 2000 03:05:24 -0500



 

> -----Original Message-----
> From: Agarwal, Aseem [mailto:aseem@trillium.com]
> Sent: Friday, November 03, 2000 1:20 PM
> To: 'SIP List'
> Cc: 'Jonathan Rosenberg'
> Subject: RE: [SIP] Sip- Message OPTION
> 
> 
> Hi Jonathan,
>   The draft (bis-2 section 4.2.3) says:
> "The response MAY contain a message body indicating the capabilities
> of the end system ..." 
> 
>   This gives an idea that OPTIONS should be relevant only for a UAS,
> since only a UAS can respond with the capabilities of the logged in
> users. The example given in section 16.8, also seems to suggest the
> same. 
>   However, theoretically, it is possible for a user to register
> his media capabilities as well in the REGISTER message with the
> Registrar. But I see this as a more dynamic thing which can 
> potentially change from a call to call. So, what is the usage of
> supporting OPTIONS at a Registrar server ? Kindly clarify.

That is not usage for a registrar. A SIP request addressed to a registrar
(sip:dynamicsoft.com) would not contain SDP in the response. A SIP request
addressed to an end user (sip:jdrosen@dynamicsoft.com) would possibly
contain SDP in the response. I will clarify in the spec.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Date: Tue, 7 Nov 2000 03:08:52 -0500




> -----Original Message-----
> From: Neil Deason [mailto:ndeason@ubiquity.net]
> Sent: Friday, November 03, 2000 8:34 AM
> To: Sip List
> Subject: [SIP] UAC Record Routing in draft-ietf-sip-call-flows-01.txt
> 
> 
> The way the UAC builds the Route on a final 200 response 
> in 3.1.2 of the call flows draft doesn't look quite 
> right ...
> 
> 3.1.2 Successful SIP to SIP through two proxies
> 
>    User A          Proxy 1          Proxy 2          User B
>      |                |                |                |
>      |   INVITE F1    |                |                |
>      |--------------->|                |                |
>      |     407 F2     |                |                |
>      |<---------------|                |                |
>      |     ACK F3     |                |                |
>      |--------------->|                |                |
>      |   INVITE F4    |                |                |
>      |--------------->|   INVITE F5    |                |
>      |    (100) F6    |--------------->|   INVITE F7    |
>      |<---------------|    (100) F8    |--------------->|
>      |                |<---------------|                |
>      |                |                |     180 F9     |
>      |                |    180 F10     |<---------------|
>      |     180 F11    |<---------------|                |
>      |<---------------|                |     200 F12    |
>      |                |    200 F13     |<---------------|
>      |     200 F14    |<---------------|                |
>      |<---------------|                |                |
>      |     ACK F15    |                |                |
>      |--------------->|    ACK F16     |                |
>      |                |--------------->|     ACK F17    |
>      |                |                |--------------->|
> 
> F14 200 OK Proxy 1 -> A
> 
>    SIP/2.0 200 OK
>    Via: SIP/2.0/UDP here.com:5060
>    Record-Route: <sip:UserB@there.com;maddr=ss2.wcom.com>,
>     <sip:UserB@there.com;maddr=ss1.wcom.com>
>    From: BigGuy <sip:UserA@here.com>
>    To: LittleGuy <sip:UserB@there.com>;tag=314159
>    Call-ID: 12345601@here.com
>    CSeq: 1 INVITE
>    Contact: LittleGuy <sip:UserB@there.com>
>    Content-Type: application/sdp
>    Content-Length: 134
> 
>    v=0
>    o=UserB 2890844527 2890844527 IN IP4 there.com
>    s=Session SDP
>    c=IN IP4 110.111.112.113
>    t=0 0
>    m=audio 3456 RTP/AVP 0
>    a=rtpmap:0 PCMU/8000
> 
> 
>    F15 ACK A -> Proxy 1
> 
>    ACK sip:UserB@there.com SIP/2.0
>    Via: SIP/2.0/UDP here.com:5060
>    Route: <sip:UserB@there.com;maddr=ss2.wcom.com>,
>     <sip:UserB@there.com;maddr=110.111.112.113>
>    From: BigGuy <sip:UserA@here.com>
>    To: LittleGuy <sip:UserB@there.com>;tag=314159
>    Call-ID: 12345601@here.com
>    CSeq: 1 ACK
>    Content-Length: 0
> 
> The UAC should build the Route header in the ACK by 
> copying the 200 response Record-Route header, 
> reversing the order and then adding the Contact as 
> the last entry. (Note the 1st entry in the Route is
> subsequently removed when the client actually sends F15).
> But in this case the last entry has become an IP address. 
> Shouldn't the UAC just copy the Contact and not try 
> and do any resolution of it?

Correct. This behavior is really bad, actually. It depends on being able to
perform a resolution of the Contact address outside of the realm where it is
supposed to be resolved. If there is a private DNS running, or some kind of
local information is placed in the hostname, this resolution will fail at
the called party.

Alan - can you please fix?

Thanks,
Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

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Subject: RE: [SIP] Errors handling on session description in ACK
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Date: Tue, 7 Nov 2000 03:19:09 -0500



 

> -----Original Message-----
> From: Piotr S. Kossowski [mailto:P.Kossowski@elka.pw.edu.pl]
> Sent: Wednesday, November 01, 2000 7:27 AM
> To: SIP discussion list
> Subject: [SIP] Errors handling on session description in ACK
> 
> 
> Hello,
> 
> When UAC send session description in INVITE request, UAS send 200 OK
> if it supports them or 606 Not Acceptable if not. Clear.
> (I assume ideal situation: no 1xx, etc.)
> 
> When UAC doesn't send session description in INVITE (what means
> that it's going to do that in ACK), UAS' 200 OK contains description.
> Next, UAC sees that it will not be able to support capabilities
> of UAS. Does it send session descr. as if it would send in INVITE,
> regardless of this failed handshake ? This scenario will fail,
> because there is no way of negotiation after UAC has sent ACK,
> since ACK doesn't generate response.

The ACK has an empty body, and the UAC sends a BYE right away.

> 
> 
> More generaly:
> 1. When UAC has an error and doesn't send session descr. in ACK (
>    INVITE did NOT contain session descr. as well), Should
>    UAS just ignore such ACK and treat it as *not received* ?

It should stop retransmitting the 200 OK. That aside, there are no media
sessions.

> 
> 2. Should garbage in body of type application/sdp generate
>    4xx response (which one ?) or 606 with appropriate code in 
>    Warning header (which one ?)

4xx, since the request is malformed. Which one? Plain old 400 is probably
fine.

> 
> 
> By the way,
> 
> I've compared bis-02 with draft-ietf-mmusic-sip-new-00.ps 
> (August '99).

Why? The only useful thing for comparison is rfc2543.

> 
> Section 10.5.1 Unreliable Transport Protocols (bis-02):
> 
> "(...)Response retransmissions cease when (...) the response has been
>  RETRANSMITTED seven times."
> 
> Section 10.5.1 (draft-ietf-mmusic-sip-new-00.ps):
> 
> "(...)Response retransmissions cease when (...) the response has been
>  TRANSMITTED seven times."

RFC2543 also says transmitted seven times.

I think it should still be transmitted, not retransmitted. 

Henning?

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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Date: Tue, 7 Nov 2000 03:29:22 -0500



 

> -----Original Message-----
> From: Jean-Francois Mule [mailto:jfmule@clarent.com]
> Sent: Tuesday, October 31, 2000 12:46 AM
> To: 'archow@hss.hns.com '; 'sip@lists.bell-labs.com '
> Subject: RE: [SIP] SIP proxies and CSeq numbers (was: SIP and T.38 fax
> cal ls - Internet Draft)
> 
> 
> Arjun:
> Thanks for your comments.  
> For the proxy, the 2 call legs constitute 2 distinct SIP 
> transactions so
> there is no particular reason to propagate the CSeq number 
> from one call leg
> to the other.  


No.

The call flow is a very standard re-INVITE type of thing. There is no reason
why this proxy should behave any differently from any other proxy. Proxies
do not modify CSeq. Period. There is no reason why this cannot be a standard
flow, and for purposes of interoperability, it should be. The flow you have
in the document is not compliant to the spec.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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Jonathan Rosenberg wrote:

> >
> >I like this approach except I would take it a step further and
> >do this for all URLs including SIP URLs. This allows maddr to go
> >back to its original intention for SIP URLs. Plus I hate special
> >cases; this seems (nominally) easier to implement.
>
> Conceptually that is clearly the better way. However, there is a backwards
> compatibility problem. Older rfc2543 implementations will use the maddr to
> forward (if they are working correctly, that is), but not an address
> parameter of the Route header.

This seems to be a backwards compatibility we can afford to break.
Will an older rfc2543 implementation really work with one based on bis
anyways? There seem to have been some substantial changes in the
mechanisms. At the very least, there is a syntactic change. (I had to
play this card because I believe implementations should handle this
change -- the addition of RR parameters)

/sean


>
> -Jonathan R.
>
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com


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Date: Tue, 7 Nov 2000 03:45:09 -0500



 

> -----Original Message-----
> From: Henning Schulzrinne [mailto:hgschul@attglobal.net]
> Sent: Sunday, October 29, 2000 8:30 AM
> To: Kevin Lingle
> Cc: sip@lists.bell-labs.com
> Subject: Re: [SIP] Some brief comments on the SIP MIB
> 
> > >
> > > - ULS should probably at least mention LDAP as an example.
> > 
> > ULS has undergone some changes.  We had a request to handle 
> prioritzed
> > backup addressess.  I also got some comments that having one generic
> > ULS address wasn't sufficient.  Systems may have multiple 
> resources they
> > use to determine where users are.
> > 
> > You can view the proposal on sip-mib@egroups.com archive
> > [http://www.egroups.com/message/sip-mib/40].  There has been no
> > feedback on the proposal yet (unfortunately that is typical).
> > I would appreciate if you could take a quick look.  Btw, the
> > current proposal does not actualy handle prioritized addresses
> > correctly (i need to consider that further).  What is your view
> > on having the ability to specify backup addresses ULS?
> 
> Probably needs discussion by those building proxies, as I can 
> primarily
> speak from our own experience. In our server, locations are 
> derived from
> several different sources
> 
> - LDAP
> - an SQL database for registrations
> - a set of user-configurable scripts that provide mappings 
> from SIP user
> ids to other user id's, e.g., resolution of email aliases.
> 
> While being able to change the LDAP URL via the management console is
> useful, I doubt that describing or changing the others is particularly
> helpful (and not likely to be feasible, given that just 
> connecting to a
> different SQL server just doesn't magically make the table definitions
> appear there).

I think the notion of generalzied, configurable ULS through SNMP is
fundamentally a hopeless cause. THe problem is that a location service is an
abstract concept. It can be the result of program execution, LDAP queries,
SQL queries, ENUM, finger, whois, local files, IVR interaction, etc. Your
assumption is that in all cases there is at least an IP address that
indicates where the service is. I don't think thats generally true. Certain
location services (local file, IVR interaction, program execution), don't
really have such a concept. 

So, there are two choices. Either completely drop this ULS thing, or
restrict it to a specific set of known location services which do have a
common set of configuration information. Access to an external database is
probably pretty broad but has a common set of information that you could
configure (IP address, username and password for access, port, transport,
database instance name)

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Subject: RE: [SIP] multiple responses ??
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Date: Tue, 7 Nov 2000 03:48:01 -0500

Hisham is correct on both counts.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Hisham Khartabil [mailto:hisham.khartabil@hotsip.com]
> Sent: Wednesday, October 25, 2000 2:10 AM
> To: Bertil Engelholm; SIP@lists.bell-labs.com
> Cc: Jonathan Rosenberg
> Subject: RE: [SIP] multiple responses ??
> 
> 
> 
> 
> > -----Original Message-----
> > From: sip-admin@lists.bell-labs.com
> > [mailto:sip-admin@lists.bell-labs.com]On Behalf Of Bertil Engelholm
> > Sent: 24. lokakuuta 2000 13:41
> > To: SIP@lists.bell-labs.com
> > Cc: Jonathan Rosenberg
> > Subject: Re: [SIP] multiple responses ??
> >
> >
> > Jonathan Rosenberg wrote:
> > >
> > >
> > >
> > > > -----Original Message-----
> > > > From: Bertil Engelholm [mailto:Bertil.Engelholm@uab.ericsson.se]
> > > > Sent: Monday, October 23, 2000 12:20 PM
> > > > To: sip@lists.bell-labs.com
> > > > Subject: [SIP] multiple responses ??
> > > >
> > > >
> > > > Hi,
> > > >
> > > > I have some questions regarding the possibility to forward
> > > > best vs. forward all responses.
> > >
> > > Forward all simply does not work.
> > >
> > > A proxy can forward one or more 200 responses (and 
> indeed, really must
> > > forward all of them), since those are ACKed end to end.
> > Non-200s are ACKed
> > > hop by hop. A proxy or client receiving multiple non-200
> > responses to the
> > > same request will have no way to distinguish them, and the
> > state machines
> > > currently defined would not work.
> >
> > OK, that simplifies things a bit.
> >
> > >
> > > >
> > > > First of all I wonder if this is not a function that should
> > > > be decided by the UA e.g. by setting a special header ?
> > >
> > > Caller preferences addresses some simple cases where a 
> UAC can request
> > > specific service from proxies. But, since 
> forward-all-responses doesn't
> > > work, having a caller prefs value for it doesn't seem that useful.
> >
> > Maybe I used the wrong term here. Forward all seems not allowed
> > as you explained above. But there are still two different 
> behaviours.
> > Either the proxy only forwards the best response or it forwards
> > all 200 responses. The problem I see is that a UA don't know if
> > it's going to receive one or several 200 responses. So it has
> > to be prepared to always receive several 200 responses. If you
> > MUST set a header to get all 200 responses instead of only the
> > best a UA can decide that it will not handle that case (by not
> > setting the header).
> 
> You must not set any header. You will get all 200 responses.  
> If you don't
> want to accept the second one, ACK it then send a BYE (then 
> wait for 200OK
> for BYE)
> 
> >
> >
> > >
> > > > It's the UA (the User) that knows if he like to get all
> > > > responses and choose himself or let the network choose the
> > > > best response. Why should each server take this decision ?
> > >
> > > We did it this way (forwarding only the best response) so 
> that it would
> > > scale and work in a reasonable fashion.
> > >
> > > >
> > > > Secondly I have a question regarding the proxy behaviour in
> > > > the following case.
> > > >
> > > >             ------      ------
> > > >             |    |----->| P2 |---->????
> > > > ------      |    |      ------
> > > > | Ca |----->| P1 |
> > > > ------      |    |      ------
> > > >             |    |----->| P3 |---->????
> > > >             ------      ------
> > > >
> > > > If Ca calls someone at P1 which forks the call to P2 and P3 how
> > > > should P1 behave in the following cases if it has the policy to
> > > > fork the best response.
> > > >
> > > > If P2 (or someone behind P2) have the policy to fork 
> all responses
> > > > it means that P1 can get several final responses.
> > >
> > > Only multiple 200.
> > >
> > > > Which of these
> > > > reponses should be used in the best choice by P1 ?
> > > > Only the first one that happens to arrive ?
> > > > Everyone that arrives before you get a response from P3 ?
> > >
> > > Beyond 200s, picking which is the best of the non-200 responses
> > is defined
> > > by the spec (600, then 300, then 400, then 500 in that order).
> > However, a
> > > proxy could choose an alternate definition of best amongst the
> > non-200 final
> > > responses it gets, since there are no interoperability issues
> > which choosing
> > > an alternate algorithm.
> > >
> > > >
> > > > What should be done with the rest of the responses that 
> might drop
> > > > in after you have sent your best response back ?
> > >
> > > There should only ever be 200s, which are forwarded upstream.
> > Anything else
> > > would be an error.
> > >
> > > >
> > > > You could actually have received an ACK back from Ca 
> when another
> > > > response drops in from P2. What to do in this case ?
> > >
> > > Won't happen under correct operating conditions.
> >
> > I'm not shure I understand why this can't happen. If P2
> > can send several 200 responses back to P1 I guess they can
> > come at any time (depending on when someone answers at the
> > other end). So lets say that the first 200 that is received
> > from P2 is choosen as the best result and sent back to Ca.
> > Ca now ACK this OK to P1 because Ca don't know it will
> > get more 200 responses later on. P1 sends the ACK to P2. Another
> > UA behind P2 somewhere now answers the same call which leads
> > to a second 200 response from P2 to P1. And this second
> > 200 response will belong to another CALL-LEG since the TO-tag
> > is different. Maybe I have misunderstood something completly
> > but this is how we have read the specification. If this
> > can't happen it would simplify things but I don't understand
> > why it can't happen.
> 
> This can happen.  You do exactly what i described above.
> 
> Regards,
> Hisham
> 
> >
> > >
> > > -Jonathan R.
> > > ---
> > > Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> > > Chief Scientist                             First Floor
> > > dynamicsoft                                 East Hanover, NJ 07936
> > > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > > http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> > > http://www.dynamicsoft.com
> > >
> > > _______________________________________________
> > > SIP mailing list
> > > SIP@lists.bell-labs.com
> > > http://lists.bell-labs.com/mailman/listinfo/sip
> >
> > --
> > Bertil Engelholm
> > AXE Research and Development        voice : +46 8 727 3499
> > SIP Security                        Fax   : +46 8 647 8276
> > S-126 25 Stockholm Sweden           E-mail:
> > Bertil.Engelholm@uab.ericsson.se
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 

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> Some minor tweaks to the original proposal, with the aim of
> aligning all the text into a common format, and ensuring that
> we have a clearly specified set of sender and receiver
> requirements, and not with any aim of technically changing it..
>
> > The header fields required, optional, and not applicable for each method
> > are listed in Table 4 and Table 5. The table uses "o" to indicate
> > optional, "m" mandatory and "-" for not applicable.
> >
> 	"Optional" means that a UAC \MAY include the header in a request,
> and a UAS \MAY ignore the header if present in the request. Similarly, an
> optional response header implies that a UAS \MAY place the header in a
> response, and the UAC \MAY ignore the header if present in a response it
> receives.

Immediately, Require springs to mind; surely a UAS MUST take notice
of Require?  What about Record-Route, or Accept?  While it seems
appropriate to identify UAC behaviour with Requests, and UAS
behaviour with Responses, I am not sure that it can be taken much
further except on a header-by-header basis.

> 	"Mandatory" means the UAC \MUST include the header in a request, and
> a UAS \MUST understand the header if present in a request it receives.
> Similarly, a mandatory response header implies that a UAS \MUST place the
> header in a response, and the UAC \MUST understand the header if
> present in
> a response it receives.

[...]


 - Jo.
--
  Jo Hornsby; IT Professional         --       mailto:jhornsby@ubiquity.net
 Ubiquity Software Corporation        --         http://www.ubiquity.net/
                                Hi, Chris! &:)


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> Agarwal, Aseem (aseem@trillium.com):
> 
> >     Pardon me if this has already been discussed earlier, but can
> > somebody verify the following behavior :
> 
>   You are almost correct.  I think your table is slightly misleading.
> 
> > 1. Unknown method
> >    Trans - Stateful proxy:     Forward if possible, else 405/501  
> >    Call - stateful proxy:      Forward if possible, else 405/501
> 
>   A proxy should proxy the request.  Only if the request is destined for
> the proxy (the Request-URI was 'sip:my.proxy.com') should the proxy
> return a 501.
> 
> >    Redirect server:            Return 405/501
> 
>   Again, if the request is destined for a user of the redirect server,
> the server should happily return a 300-class response to the unknown
> request.  It should only return a 501 if the request was destined for
> the redirect server (the Request-URI was 'sip:my.redirect.com') and the
> server didn't understand it.
[...]

Just a small nit: I would argue 405 for something like
'sip:my.proxy.com' (the proxy does understand the method, it
just doesn't like it for this Request-URI).

Cheers,


 - Jo.


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I wonder if it might be troublesome to start record-routing on any
messages exchanged prior to an intial INVITE as this will often be what
service logic triggers on. The service logic and management of the
triggers might become more complex and more brittle if one has to be
ready to start record-routing on a wider variety of conditions.

Anders Kristensen


Jonathan Rosenberg wrote:
> 
> 
> 
> > -----Original Message-----
> > From: Fairlie-Cuninghame, Robert [mailto:rfairlie@nuera.com]
> > Sent: Monday, November 06, 2000 10:53 PM
> > To: sip@lists.bell-labs.com
> > Subject: [SIP] OPTIONS and out-of-band Route.
> >
> >
> >
> > There's seems to be a lot of discussion lately about out-of
> > band Route in
> > INVITE. Here's a another reason for correct processing of
> > "out-of-band"
> > Route headers ("out-of-band" with respect to an INVITE
> > initiated call-leg):
> >
> > When a UAC makes a OPTIONS request before performing a full
> > INVITE, the UAS
> > may include a Contact in the 200 OK which be used as the
> > direct destination
> > to send the initial INVITE. If a proxy needs to be on the path of the
> > initial INVITE (eg proxy is acting as ALG) then the call will
> > fail if the
> > proxy and UAC doesn't take some additional action.
> >
> > DOn't we need to add something such as the following to the spec?
> 
> Well, this is a symptom of a more general problem, which is what is the
> meaning of Record-Routing in requests in general, beyond INVITE. I would
> propose the following:
> 
> If a proxy inserts a record-route into a request, that means it wishes to
> receive all subsequent requests, in both directions, that contain the same
> value of the To, From, and Call-ID.
> 
> I like this, since then it works for MESSAGE and SUBSCRIBE, which have
> similar requirements.
> 
> Now, to handle your OPTIONS case:
> 
> If a UAC initiates a call to a host as the result of an OPTIONS query, it
> SHOULD use the same Call-ID as was used in the OPTIONS query. Note that as a
> result of standard Record-Routing rules, this INVITE will contain any
> routing information learned from the OPTIONS response.

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> >
> > Section 10.5.1 Unreliable Transport Protocols (bis-02):
> >
> > "(...)Response retransmissions cease when (...) the response has been
> >  RETRANSMITTED seven times."
> >
> > Section 10.5.1 (draft-ietf-mmusic-sip-new-00.ps):
> >
> > "(...)Response retransmissions cease when (...) the response has been
> >  TRANSMITTED seven times."
> 
> RFC2543 also says transmitted seven times.
> 
> I think it should still be transmitted, not retransmitted.

Changed to "transmitted".

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Jonathan Rosenberg wrote:
> 
> I think this note is simply wrong. I don't even remember noticing it until
> you pointed it out. Its in both bis and rfc2543. It should be stricken.
> 
> Henning - do you agree?

Yes, it's gone now. It appeared in draft-ietf-mmusic-sip-09.txt, but
without explanation.

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Subject: Re: [SIP] OPTIONS and out-of-band Route.
From: David Shrader <dshrader@master-consultant.com>
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I agree. Since the Record-Route mechanism is predicated on the fact that the
proxy wants to remain in the call path because it is going to maintain state
on the call, it is extremely problematic if the record-route function occurs
on an OPTIONS. This would then, somehow, add pre-call-state prior to INVITE.
Suppose conditions change between the OPTIONS and the INVITE, such as
time-of-day routing decisions that cause the INVITE to routed a different
way. Are we then stuck because we had a pre-loaded Route header that
identifies a particular path?

I think we can generalize the record-route mechanism to operate on the
SUBSCRIBE/NOTIFY/MESSAGE type of requests, but I am concerned about
pre-loading the Route on an initial INVITE (or initial SUBSCRIBE).

David

----------------------------
David Shrader
Master Consultant, Inc.
dshrader@master-consultant.com
http://www.EyeForTheFuture.com

> From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
> Date: Mon, 6 Nov 2000 23:28:08 -0800
> To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, sip@lists.bell-labs.com
> Subject: RE: [SIP] OPTIONS and out-of-band Route.
> 
>> 
>> If a proxy inserts a record-route into a request, that means
>> it wishes to
>> receive all subsequent requests, in both directions, that
>> contain the same
>> value of the To, From, and Call-ID.
>> 
>> I like this, since then it works for MESSAGE and SUBSCRIBE, which have
>> similar requirements.
> 
> Agreed. Much more general.
> 
>> Now, to handle your OPTIONS case:
>> 
>> If a UAC initiates a call to a host as the result of an
>> OPTIONS query, it
>> SHOULD use the same Call-ID as was used in the OPTIONS query.
>> Note that as a
>> result of standard Record-Routing rules, this INVITE will contain any
>> routing information learned from the OPTIONS response.
>> 
> My only comment here is that Section 11 & 12 on the behavior of UAC's and
> proxies is written from the point of view that a Call State Machine is
> created by an "initial INVITE request". I believe it is this idea, that is,
> importing "out-of-band" Route information from _before_ the Initial INVITE
> Request, that is cause of the recent discussions/queries/confusion in this
> area. In other words, in my mind the existing Record-Route rules description
> doesn't quite stretch to unambiguously cover the pre-initial-invite
> behaviour. I think a small clarification would be helpful.
> 
> Cheers,
> 
> Robert.
> 
> -- My opinions are my own. I tried selling them once but everybody
> seems to already have one. --
> 
> _______________________________________________
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> SIP@lists.bell-labs.com
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> 


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Subject: RE: [SIP] Re: Pre-loaded Route headers
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Date: Tue, 7 Nov 2000 09:59:05 -0500 (EST)

There is informative text at the beginning of 6.34.1, and
also informative text in 6.34.2.  However, they contradict
each other.  There is no normative text for this situation
at all.

See, for example, lines 3-4 of 6.34.2:
  A UA builds the Route header field for subsequent requests
from the Record-Route header fields received in either a
response or a request.

I would take this to explicitely allow a mid-call change
of Record-Route/Route values.

What is needed is some normative text about the handling
of Record-Route in UAC/UAS/Proxies.  I believe the workable
combination is:

UAC:	MUST convert the Record-Route to a Route
	on the final response to the first request
	of a call leg, and use it for all future requests
	with the same call-leg identification.

	MUST convert the Record-Route from a provisional
	response into a Route for use solely in a
	PRACK and COMET.  
(This requirement is from 100rel, and was copied into manyfolks, but I 
don't understand how it might be different than first case above - unless 
it intends to allow mid-call requests with provisional responses to take 
a different route.)

	MUST NOT update Route value based on any Record-Route
	received in a subsequent request/response with the
	same call-leg identification.

UAS:	MUST convert the Record-Route to a Route on the
	first request with a new call-leg identification.

	Within a single transaction, MUST include the value of
	Record-Route received in the request in every
	response that requests a PRACK.
(Should this instead be to include the value of Record-Route received
in the first request with the same call-leg identification, to remove
the dependance on proxies doing the right thing?)

	MUST NOT update Route value based on any Record-Route
	received in a subsequent request/response with the
	same call-leg identification.

Proxy:	Either (1) MUST insert itself into the Record-Route
	header in every request for a particular call-leg
	identification, or (2) MUST NOT insert itself into
	any Record-Route header for a particular call-leg
	identification.
(Note this is a SHOULD in case #1 in 2543bis)

The proxy behavior is difficult for a stateless proxy, unless it makes the 
decision (to be in the Record-Route or not) independent of call-leg 
identification.


I believe the above requirements have the property that, if the proxies
do the right handling, the UAC/UAS can botch the handling and still be OK.
Further, if the UAC/UAS do the right handling, the proxies can be wrong
and everything will still work OK.

Bill Marshall
wtm@research.att.com


-----original message-----
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
To: "'William Marshall'" <wtm@research.att.com>, sip@lists.bell-labs.com
Subject: RE: [SIP] Re: Pre-loaded Route headers
Date: Tue, 7 Nov 2000 02:56:34 -0500

Comments below.

 

> -----Original Message-----
> From: William Marshall [mailto:wtm@research.att.com]
> Sent: Friday, November 03, 2000 4:02 PM
> To: sip@lists.bell-labs.com
> Subject: [SIP] Re: Pre-loaded Route headers
> 
> 
> I believe there may be a problem with Pre-loaded Route headers and
> stateless proxies, but it not a difficult one to solve.  But there
> are side-affects that are not nice.
> 
> An initial INVITE is normally the one that accumulates the 
> Record-Route
> values; then they are stored at the UAS and converted into a 
> Route header
> for UAS->UAC transactions, and echoed back to the UAC in the 200-OK so
> the UAC converts them into a Route header for UAC->UAS transactions.
> A mid-call INVITE (or any other message) contains the Route header
> value, but doesn't need to accumulate Record-Routes again.
> 
> But a stateless proxy can't really tell the difference between an
> initial INVITE for a new session (one in which it should add itself
> to the Record-Route) and a mid-call INVITE for an existing session (in
> which it doesn't need to add itself).  So it has two choices: 
> 1) always
> add itself to a Record-Route, or 2) note there is a Route header and
> assume therefore it is a mid-call message and do nothing.
> 
> Keying on the existence of a Route header to indicate a mid-call
> message isn't explicitely disallowed by the spec (unless I missed
> it somewhere), but it would work just fine - until a UAC tried to
> use a pre-loaded Route header.  Then the proxy wouldn't get itself
> into the Record-Route when it should.
> 
> So the solution is to have the proxy insert itself into the
> Record-Route header *independent of whether it is an initial INVITE
> or a mid-call INVITE*.  
> 
> Unfortunately there is a backward compatibility problem here - a proxy
> that doesn't know to do this for a mid-call message will be dropped
> from future messages.  If even one proxy along the path inserts a
> Record-Route header on the mid-call message, the UAS and UAC will
> trigger their Route re-calculation.  (Presumably if none of 
> the proxies
> did the Record-Route for a mid-call message, the UAC and UAS would
> act like currently, and keep the Route from the initial INVITE).

We had an extensive discussion on the list on just this topic. The result of
that discussion is already in the bis draft:

--
The Record-Route request and response header field is added to a request by
any proxy that insists on
being in the path of subsequent requests for the same call leg. A proxy
SHOULD add it to anyrequestfor
robustness, but a request route, once established, persists until the end of
the call leg, regardless of whether
the Record-Route header is present in subsequent requests.
--

So, what this is saying is that a proxy should always insert the
record-route, but a UA doesn't reload the record-routes if it has a set
already. This avoids the backwards compatibility issue. But, you ask, what
is the point of re-inserting them? THe answer is twofold. First, the
original idea is recovery from crash and restarts. The restarted UA wouldn't
have a record of the Record-Routes, and so when it gets a re-INVITE (via
session timer, perhaps), it will re-establish the route based on the
record-routes in the INVITE. Secondly, we have our recent discussion of
sending invites with loaded routes. If proxies follow the above
recommendation, they will always record-route, and the result is that the
initial forced route will actually get persisted, since the UAS has no
memory of a route when the initial request arrives.

So, what we need here is a little motivational text so people notice this
subtelty and implement it correctly, along with a reccomendation that
proxies SHOULD NOT consider a request with a Route header as indicative of a
mid-call or mid-session message.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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David Shrader wrote:
> 
> I agree. Since the Record-Route mechanism is predicated on the fact that the
> proxy wants to remain in the call path because it is going to maintain state
> on the call, it is extremely problematic if the record-route function occurs
> on an OPTIONS. This would then, somehow, add pre-call-state prior to INVITE.
> Suppose conditions change between the OPTIONS and the INVITE, such as
> time-of-day routing decisions that cause the INVITE to routed a different
> way. Are we then stuck because we had a pre-loaded Route header that
> identifies a particular path?
> 
> I think we can generalize the record-route mechanism to operate on the
> SUBSCRIBE/NOTIFY/MESSAGE type of requests, but I am concerned about
> pre-loading the Route on an initial INVITE (or initial SUBSCRIBE).

I don't understand this concern. The client presumably has a reason to
do the routing; nobody forces it to insert Route headers. The proxy
keeps no state of any kind, it just obeys the Route headers. Thus, I'm a
bit lost as to why this inconveniences the proxy in any way.


-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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> 2. In 4.2.5, it is noted that CANCEL requests cannot have Route headers.
>    Why not?  Are you assuming that CANCEL is only useful for an initial
>    request?  This breaks since we need to be able to CANCEL a pending
>    REFER or other extension method which doesn't complete quickly, for
>    example.

I doubt that the thinking here is CANCEL is only useful for an
initial request, more that there is no need to specify a Route
since proxies will always "know" which locations to cancel in
turn; CANCEL is not forwarded like other requests, thus a Route
is not necessary.


 - Jo.


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Subject: [SIP] More Record-Route Joy
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Hi,

I need debunking (or otherwise) &:)

The wording in bis02 in 6.34.2 reads:

    "If a UAC finds a Record-Route header in a final response,
     it copies it into Route header fields of all subsequent
     requests within the same call leg, reversing the order
     of fields, so that the first entry is the server closest
     to the UAC."

I think the fact that this only applies to the initial INVITE
is being thrashed out at the moment, but there is another
problem.

For the ACK, surely the Record-Route should be ignored unless
the response was 2xx, else things are going to go horribly
wrong.  This applies to the initial INVITE, as well as to
re-INVITEs; further, it should also apply to the
pre-initial-INVITEs (for want of a better term), "...where the
server can expect the client to retry for the same Call-Id,
as in 401 (Unauthorized) or 484 (Address Incomplete)."
[last para of 6.34.1].

In the non-2xx final response cases, the ACK should be sent
to the same place as the corresponding INVITE, with the same
Request-URI (as specified by 10.5.1).

Cheers,


 - Jo.
-- 
  Jo Hornsby; IT Professional         --       mailto:jhornsby@ubiquity.net
 Ubiquity Software Corporation        --         http://www.ubiquity.net/


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Hello!

(Im going to sound like a lawyer again. Yech! :-)). 

Here is the passage from the RFC that I am trying to interpret. In
section 1.4.3:

 If the client sent the request via unicast UDP, the response is sent
 to the address contained in the next Via header field (Section 6.46)
 of the response. 

I use the term client to refer to the originator of a message. My
questions / assumptions are below: 

1. The address in the first via header need not necessarily be the peer
address that the server reads from the udp socket when it gets the
message. Of course, it is in the client's best interest to use a
meaningful address.

2. If the server reads a request that is badly formatted except for the
via header, it can silently drop the message if it wants. In general,
the server is just being nice by returning error messages for udp
requests and is not obliged to do so.

Thank you in advance for your replies.

Regards,
Ranga.

-- 
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899. 
Tel: 301 975 3664 Fax: 301 590 0932

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Quoting Jonathan Rosenberg <jdrosen@dynamicsoft.com>:


> The advantage here is that one could insert URL parameters into the
> record-route, and get them back in the request URI. Not the prettiest way
> to
> go. We could, as an alternative, simply mandate the State header in the 
bis
> draft, which would provide a cleaner solution.
>
So, the State header will be a comma separated list (or multiple
state headers) of state ids
one per Route Header. However as one Route header is removed before sending
out the request, the corresponding State header is retained, only to
be removed by the subsequent proxy if it proceses the next route
header. In other words there should be n+1 State headers for
n Route headers in the request. This correlation process can
become tricky. Knowing whether the topmost State info is relevant
to the proxy can be difficult.

I would prefer to see this state info in the record route/route
headers.

Shiv

> >
> >I like this approach except I would take it a step further and
> >do this for all URLs including SIP URLs. This allows maddr to go
> >back to its original intention for SIP URLs. Plus I hate special
> >cases; this seems (nominally) easier to implement.
>
> Conceptually that is clearly the better way. However, there is a 
backwards
> compatibility problem. Older rfc2543 implementations will use the maddr 
to
> forward (if they are working correctly, that is), but not an address
> parameter of the Route header.
>
> -Jonathan R.
>
>
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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Quoting Jonathan Rosenberg :

> as I pointed out, this causes everything from the Contact/From to be 
copied
> into Route, EXCEPT port and maddr. How about if we say that the Route
> contains the user:password@host portion of the Contact/From, and 
everything
> else is copied from the Record-Route? THis would include unknown URL
> parameters.
>

Are we not going out of the way to retain the Request URI
to be the final destination. This has caused problems
is various contexts. Also an assumption is
being made in all cases is that the Request URI is routable.

What if a proxy receives a request with a URI that is not routable
by it. Consider an INVITE abc@pqr.stu.xyz.com. The proxy receiving
this request may not be able to route this because zone pqr may
not be resolvable by this proxy, however if this request is
sent to stu.xyz.com it may be. I would like to see request URIs
indicate intermediate hops. I think the Request URI should retain its 
flexible flavor.

In another thread:

>Now, an interesting thing to note here. Lets say in this failure case, it
>was the called party that hung up. The proxy downstream from P2 (say, P3),
>pops its top Route, and tries to forward it to P2. But, that maddr >
(pointing
>to P2), generates an ICMP error. So, it looks up the host portion of the
>request URI in DNS and tries the SRV process from there. I will note that
>this will only succeed if, and only if, *THE REQUEST URI ALWAYS REFERS TO
>THE RESOUCE FOR WHOM THE REQUEST IS DESTINED*. Now, this relates to a
>previous thread. The issue there was how a UAS would construct the Route
>headers, and whether we should maintain this property that the R-URI 
always
>refers to the entity for whom the request is targeted. I originally felt
>this was critical, and was coming close to backing down. Now, it seems we
>have another case to back up my original claim that maintaining this
>property is important.

I would like to find out why resort to the Request URI when you
could do better by sending the request to the next 'Route' header. Are
we going out of our way to preserve the Request URI? What if it is
not routable.

Shiv


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Subject: [SIP] Making draft-lennox-sip-reg-payload-01.txt a SIP WG item
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Date: Tue, 7 Nov 2000 17:19:20 -0500

At the last meeting, there was discussion of making 
draft-lennox-sip-reg-payload-01.txt a SIP WG item.
It was decided to take this discussion to the list.

If you have a problem with making this draft a WG
item, please say so now.  

Thanks
  Brian
  Joerg
  Dean


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Date: Tue, 7 Nov 2000 20:02:48 -0800

> 
> Another potential solution is to slightly change the way 
> Route headers are
> constructed. Right now, the spec says:
> 
> ----
> If a UA finds a Record-Route header in a request received as a UAS, it
> copies the Record-Route
> maddr parameters and any port value, maintaining their 
> ordering, to the
> Route header field of future
> requests issued as a UAC. Since the URIs contained in the Record-Route
> header fields are not useful for
> the reverse request path, the UA fills all other components 
> of the Route
> name-addr value with the name-addr
> value found in the Contact or the From header field. The 
> latter is used only
> if there is no Contact
> header field.
> ----
> 
> as I pointed out, this causes everything from the 
> Contact/From to be copied
> into Route, EXCEPT port and maddr. How about if we say that the Route
> contains the user:password@host portion of the Contact/From, 
> and everything
> else is copied from the Record-Route? THis would include unknown URL
> parameters.
> 
> The advantage here is that one could insert URL parameters into the
> record-route, and get them back in the request URI. Not the 
> prettiest way to
> go. 

I was thinking along the same lines - I think this would solve a couple of
problems. 

Firstly, I can't see it causing any backward-compatibility problems as any
"other parameters" in the Request-URI (ie implementation specific
Record-Route paramters) are simply going to be ignored by other proxies that
may see the Request-URI. That is, no changes required anywhere except in the
way the proxy builds its _own_ Record-Route entry - perfect. 

Secondly, this solves the stored Record-Route state problem (ie, the proxy
never sees the information stored in the Route entry because it is
overwritten ). This means that Record-Route-ing can be done statelessly -
probably important in OPTIONS and other non-INVITE Record-Routing that don't
have any BYE/Expire sequence to clean up the Route information.

We should however define a standard Record-Route "branch"-like options names
for implementation specific information - so we don't get clashes on
Request-URI option names. However, a proxy which understands the new Route
options should _only_ act upon the the route-state header _if_ it can
identify that the header was created by itself. 

In furtherance of the above and in solving the maddr:port problem for
non-sip URL's, then a proxy SHOULD also add a record-route address headers
(and MUST do so if the proxy adds a rroute state header ). How about .......

Record-Route: [Request-URI];rr-host=host:port;rr-state=balhblahblah

I've skipped a lot of details but I sure this is going to drawn enough
discussion. [Eg, how can a proxy can tell if a UAS will scrub the
record-route paaramters due to an older implementation?]

Cheers,

Robert.

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From: "Hisham Khartabil" <hisham.khartabil@hotsip.com>
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Subject: [SIP] SIP Extensions for Presence
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Date: Wed, 8 Nov 2000 10:55:29 +0200
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Hi all,

Apologies for the long email, but I have few questions regarding SIP
Extensions for Presence.

3. Definitions

"Presence User Agent (PUA): A Presence User Agent manipulates presence
information for a presentity. In SIP terms, this means that a PUA generates
REGISTER requests, conveying some kind of information about the presentity."

"Presence Agent (PA): A presence agent is a SIP user agent which is capable
of receiving SUBSCRIBE requests, responding to them, and generating
notifications of changes in presence state."

Q: what about the sip user agent that generates SUBSCRIBE request and
receiving notifications? Should we have a definition for that? Are we
assuming s/he is the SUBSCRIBER?

4.2 Message Flow
...
"In fact, it is possible that two different presence clients register for
the same presentity. In this case, the presence server, acting as a proxy,
may fork the SUBSCRIBE, which means it sends multiple copies of the request,
one to each presence client. Each client will presumably query its principal
about whether the subscription is acceptable. In the common case where a
principal has left their presence tool running at work, and then gone home
and started the tool there, the principal will accept the subscription at
home. This causes a success response to be generated, and forwarded to the
presence server (which is acting as a proxy), and then back to the
subscriber. The presence tool at home then assumes responsibility for that
subscription. The tool at work can continue to support subscriptions it
received previously."

Since I am not at work, my status will not change so no notifications will
be generated.  This does not give the subscriber and up to date notification
of my presence.

Q: how about if I don't want my subscriptions at work to continue and I want
them to be transferred home?  I know this is achieved over time since
SUBSCRIBE has Expires: header.  But this could take hours (or one hour).
There should be a way to transfer subscriptions.

...

"As an optimization, notifications do not need to be sent if the subscriber
is not actually online. This improves scalability."

Q: is this assuming that A is subscribed to B and vice versa? Otherwise, who
would a Presence Agent Know that the subscriber on offline?

...

"A SUBSCRIBE request MUST contain a Contact header. This indicates the
address(es) (as a SIP URL) to which the client would like to receive
notifications. This MUST be be one or more SIP addresses, containing the
fully qualified domain names for the host generating the subscription, or
the IP address of the host generating the subscription. Other addresses are
possible, supporting third party subscriptions. If it contains multiple
addresses, notifications will be sent to each address. If no Contact header
is present, no notifications will be sent."

Q: since a SUBSCRIBE request MUST contain a Contact header, why wouldn't you
respond with 4xx if no Contact is present?

General Question:
If a client sends a REGISTER or a SUBSCRIBE with Expires: 3600.  The client
then re-Registers or re-subscribe with Expires: 3600 again, but before the
first one has expired (say 600 secs left). Does the new timer get reset to
3600 secs or 4200 secs (3600 + whatever is left from the first request. 600
in our case).


Regards,
Hisham

0=====================================================================0
o Hisham Khartabil  -  Hotsip Finland                                 o
o mailto:hisham.khartabil@hotsip.com                                  o
o sip: Hisham.khartabil@hotsip.com                                    o
o mobile: +358 40 582 7484                                            o
o fax: +3589 856 642 10                                               o
0=====================================================================0






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shh@microappliances.com wrote:
> 
> Quoting Jonathan Rosenberg <jdrosen@dynamicsoft.com>:
> 
> > The advantage here is that one could insert URL parameters into the
> > record-route, and get them back in the request URI. Not the prettiest way
> > to
> > go. We could, as an alternative, simply mandate the State header in the
> bis
> > draft, which would provide a cleaner solution.
> >
> So, the State header will be a comma separated list (or multiple
> state headers) of state ids
> one per Route Header. However as one Route header is removed before sending
> out the request, the corresponding State header is retained, only to
> be removed by the subsequent proxy if it proceses the next route
> header. In other words there should be n+1 State headers for
> n Route headers in the request. This correlation process can
> become tricky. Knowing whether the topmost State info is relevant
> to the proxy can be difficult.

It doesn't have to be done that way. The State header could very 
easily include a value by which the proxy can identify itself. 
In fact the DCS work has already proposed something like this:

   State           = "State" ":" 1#(host ";" state-token 
                                *(";" state-token))
   state-token     =  token ["=" (*token | quoted-string)]

See:
http://search.ietf.org/internet-drafts/draft-ietf-sip-state-00.txt

A State header is much cleaner than a hack which overloads 
the meaning of SIP URL params in Record-Route. That type of 
approach looks to have greater potential for future deficiencies.

Cheers,
Neil.
-- 
Ubiquity Software Corporation, UK        http://www.ubiquity.net

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On Tue, 07 Nov 2000, M. Ranganathan wrote:
> Hello!
> 
> (Im going to sound like a lawyer again. Yech! :-)). 
> 
> Here is the passage from the RFC that I am trying to interpret. In
> section 1.4.3:
> 
>  If the client sent the request via unicast UDP, the response is sent
>  to the address contained in the next Via header field (Section 6.46)
>  of the response. 
> 
> I use the term client to refer to the originator of a message. My
> questions / assumptions are below: 
> 
> 1. The address in the first via header need not necessarily be the peer
> address that the server reads from the udp socket when it gets the
> message. Of course, it is in the client's best interest to use a
> meaningful address.

If a server receives a message and the originating host address does
not match the value in the top most via header, it adds a received
parameter which identifies the IP address of the client.  This received
parameter is used when routing messages back up stream.  Is that what
you're asking about?
    
> 2. If the server reads a request that is badly formatted except for the
> via header, it can silently drop the message if it wants. In general,
> the server is just being nice by returning error messages for udp
> requests and is not obliged to do so.

Is that a question or a statement?  I would expect an error response to
be sent if the message can be decoded enough to generate a valid
response.

> Thank you in advance for your replies.
> 
> Regards,
> Ranga.
> 
> -- 
> M.Ranganathan
> NIST Advanced Networking Technologies Group,
> 100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899. 
> Tel: 301 975 3664 Fax: 301 590 0932
> 
> _______________________________________________
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> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
-- 
Gethin Liddell
Ubiquity Software Corporation

http://www.ubiquity.net
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Jonathan Rosenberg wrote:

>
> For things to work consistently, the SRV lookup process should be based on
> the pseudo-random number generator I suggested in a previous thread; thats
> the one based on a hash of the Call-ID, amongst other things. This gives us
> consistent operation for SRV records in general.

As I replied in a previous thread (with a bad example) this hash based method
gives us nothing in general it only works if you locally cache DNS for the life
time of a transaction, for each transaction (hence truly stateless proxy servers
do not work ;^) ). The reason for this is simple  for the hash method to work
the result from the DNS query MUST be identical even in ordering of the RRs each
time. As an example of why this may not happen I'll admit my short comings, the
first time I implemented SRV lookups I was lazy I found the lowest priority RRs
and picked the first one. If I was writing a DNS server I may expect other
people to do this, and in a more thorough moment may decide to shuffle the
results my server returned for SRV records (this wouldn't provide the same level
of load balancing but could help).

Also it's possible the zone file might have changed.

If you are going to suggest this hashing proposal I'd also think about sorting
the results of SRV lookups, although zone file changes will still break it.

James Undery


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Sean Olson wrote:
>
> Jonathan Rosenberg wrote:
> 
> > >
> > >I like this approach except I would take it a step further and
> > >do this for all URLs including SIP URLs. This allows maddr to go
> > >back to its original intention for SIP URLs. Plus I hate special
> > >cases; this seems (nominally) easier to implement.
> >
> > Conceptually that is clearly the better way. However, there is 
> > a backwards compatibility problem. Older rfc2543 implementations
> > will use the maddr to forward (if they are working correctly,
> > that is), but not an address parameter of the Route header.
> 
> This seems to be a backwards compatibility we can afford to break.
> Will an older rfc2543 implementation really work with one based on bis
> anyways? There seem to have been some substantial changes in the
> mechanisms. At the very least, there is a syntactic change. (I had to
> play this card because I believe implementations should handle this
> change -- the addition of RR parameters)

Hmmm... this backwards compatibility thing is a pain, but how
about this:

2543 (Section 6.29) currently states:

    The Record-Route request and response header field is added to a
    request by any proxy that insists on being in the path of
    subsequent requests for the same call leg. It contains a globally
    reachable Request-URI that identifies the proxy server. Each proxy
    server adds its Request-URI to the beginning of the list.

[Note that the text says the proxy adds its Request-URI; it
doesn't say any more than that, so it doesn't seem unreasonable
to assume there is freedom for perhaps more exotic URI than
one might normally find.]

    The server copies the Record-Route header field unchanged into the
    response. (Record-Route is only relevant for 2xx responses.)

[I would highlight the "unchanged" in this sentence.]

    The calling user agent client copies the Record-Route header into a
    Route header field of subsequent requests within the same call
    leg, reversing the order of requests, so that the first entry is
    closest to the user agent client. If the response contained a
    Contact header field, the calling user agent adds its content as
    the last Route header. Unless this would cause a loop, any client
    MUST send any subsequent requests for this call leg to the first
    Request-URI in the Route request header field and remove that
    entry.

[And here we have the client sending to the Request-URI in popped
Route.  Again, it doesn't seem unreasonable to assume that the
Request-URI is copied verbatim; the only slight change might be
removal of any of (maddr, transport, ttl) parameters, since these
were not allowed in the Request-URI in 2543 (Section 2, just
before "Headers:".]

If the motivation behind retaining backwards compatibility is
purely a "we're going to be breaking the spirit of the RFCs/IETF",
then this may be a solution...

Instead of proxies copying the Request-URI, and mangling the
port and maddr, why not just get them to put _their_ SIP URL in,
and make the Request-URI a parameter of the SIP URL (suitably
escaped, of course).

So for instance, say I am an example.com proxy, listening on
port 6000.  I get the following:

    INVITE sip:alice@bobs-place.com SIP/2.0
    ...

I then forward this as (Alice gets about a bit):

    INVITE sip:alice@charlies-place.com SIP/2.0
    ...
    Record-Route: <sip:example.com:6000;uri=sip:alice@bobs-place.com>
    ...

Now if a 2543 client sends a subsequent request in the
caller -> callee direction, this proxy should see:

    INVITE sip:example.com:6000;uri=sip:alice@bobs-place.com SIP/2.0
    ...

which should be enough information. &:)

Now I know we have questioned whether the UA or proxy or whatever
will copy the entire Request-URI, but the spec certainly doesn't
say that only the user@host:port is copied, so I don't see this as
a valid argument.

If, OTOH, we are looking at backwards-compatibility from a "work
with existing SIP Implementations" standpoint, then I have to say
that so much of Record-Route wouldn't have worked anyway (the
spec says nothing about building Route:s in the callee -> caller
direction), is it really worth worrying?

With this approach, we can write into bis, say, that the called
UA, when forming requests, renames all "uri" parameters to
"original-uri"; and appends a new "uri" parameter with the value
of the Contact:/From:.

We now have the resilience that Jonathan was addressing (thread:
"failure of proxy), since each proxy along the Route can see
the final destination.  And also, if the request spiralled,
we know where on the Route we are (since we have retained the
original Request-URIs, too).

Now I know that this is a significant change, but as far as I can
tell, realistically, no proxy will as yet have been able to
Record-Route properly and/or remain call stateful, because we
are still finding bugs.  Thus it doesn't seem as if there can be
real objections to making such a changes, since hopefully they
are changes towards having a fully-working mechanism (okay, I
guess we think it works now, but it can't be more than 3 months
since we spotted that Record-Route: can break in the reverse
direction when a request has spiralled; how mature can
implementations possibly be?)

Thoughts?


 - Jo.


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Date: Wed, 8 Nov 2000 17:11:25 +0530



Hi,
Thanks for posting the proposed new grammar. I have a couple of doubts with
the new grammar. According to the grammar you have posted below
I can have any number of  referenced-urls and ref-signatures in any
combination. Further according to this grammar, we could have a
ref-signature without a
referenced url. Is this allowed?.. The draft-sip-cc-transfer-01  clearly
states that the refsignature "contains a signature over the concatenation
of referrer-url and referenced url". This implies that the referenced url
is mandatory ( as you have rightly pointed out in your mail). The new
grammar does not reflect this. Again with the new grammar i could leave out
the referenced-url and the refsignature alltogether. This is clearly not
allowed.  IMHO   the new grammar should be,

     Referred-By      = ("Referred-By" | "b") ":" referrer-url ";"
referenced-url [";" ref-signature ]

      referrer-url     = ( name-addr | addr-spec )

      referenced-url   = "ref" "=" "<" URL ">"

     ref-signature    = signature-scheme *( ";" sig-scheme-params )
      signature-scheme = "scheme" "=" token
      sig-scheme-parms = token "=" ( token | quoted-string )

The difference here is removing the " | " and the " * " operators  between
the referenced url and the ref-signature.


I have another doubt. In the draft it states that the Referred-By header
can be in any Request message. Why is this so?....It also states that the
refernced-url must contain the url in the refer-to header. Does this imply
that for a referred-by header to exist a refer-to header should also be
present, or does it mean in a REFER request you have both refer-to and
referred-by headers and when the Transferee does an INVITE to the target he
includes a referred-by header without a refer-to header ?...

What is the motivation for limiting the Refer-To header to just SIP-URL and
not nameaddr ? ....

In your next version of the draft could you inlcude some sample messages
with the REFER method in the call-flow diagrams?..thanks

regards,
Ashok


Ashok I Roy @ Hughes Software Systems




It's going to be a few more days before I can get the next
revision of the REFER draft out. Until then, here's what
list discussion has guided the Referred-By syntax into.

The motivations for change were

1) Allow URLs containing semicolons to be parsed correctly. This
   introduced changing the referrer-url from SIPURL to name-addr |
   addr-spec. Allowing the referrenced URL to be separated is a
   little more difficult since it is not necessarily a SIP URL.
   This syntax attempts to resolve the problem by wrapping the URL
   with <> and requiring any <>s appearing in the URL to be escaped.

2) Allow the parameters to be completely unordered. This requires
   more text to augment the ABNF.

-------------------
     Referred-By      = ("Referred-By" | "b") ":" referrer-url
                                 *(   ";" referenced-url | ";"
ref-signature  )
     referrer-url     = ( name-addr | addr-spec )
     referenced-url   = "ref" "=" "<" URL ">"
     ref-signature    = signature-scheme *( ";" sig-scheme-params )
     signature-scheme = "scheme" "=" token
     sig-scheme-parms = token "=" ( token | quoted-string )

A Referred-By header MUST contain exactly one referenced-url.
A Referred-By header MUST contain either zero or one ref-signature.
Any occurences of "<" or ">" within the URL value in referenced-url
must be escaped.

------------

RjS

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Date: Wed, 8 Nov 2000 11:27:28 -0000

I noted in Henning's list of headers for IANA that the Hide header is
deprecated.

In the bis-02 draft, there appears to be no text that indicates this, or
gives any details on what the term means as regards sending and receiving,
apart from some discussion in annex I. Is there some proposed text somewhere
that states SHOULD NOT for sending, and changes the "m" status in table 6
for proxies.

Keith
Keith Drage
Lucent Technologies
Tel: +44 1793 776249
Email: drage@lucent.com

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Some ramblings on the subject.
/sean

Jo Hornsby wrote:

Hmmm... this backwards compatibility thing is a pain, but how
about this:
<snip>

> If the motivation behind retaining backwards compatibility is
> purely a "we're going to be breaking the spirit of the RFCs/IETF",
> then this may be a solution...
>
> Instead of proxies copying the Request-URI, and mangling the
> port and maddr, why not just get them to put _their_ SIP URL in,
> and make the Request-URI a parameter of the SIP URL (suitably
> escaped, of course).

<snip>

> Now I know we have questioned whether the UA or proxy or whatever
> will copy the entire Request-URI, but the spec certainly doesn't
> say that only the user@host:port is copied, so I don't see this as
> a valid argument.

But this should be clarified. The UA/proxy MUST copy the entire URI
from the Route: header into the Request-URI without modification or
omission.

> If, OTOH, we are looking at backwards-compatibility from a "work
> with existing SIP Implementations" standpoint, then I have to say
> that so much of Record-Route wouldn't have worked anyway (the
> spec says nothing about building Route:s in the callee -> caller
> direction), is it really worth worrying?
>

I agree completely with this sentiment.

>
> With this approach, we can write into bis, say, that the called
> UA, when forming requests, renames all "uri" parameters to
> "original-uri"; and appends a new "uri" parameter with the value
> of the Contact:/From:.

I'm not sure I see a need for three URIs in this case. You
have in the Route: header:

1) The URI of the Proxy  (the main URI)
2) The original Request-URI of the request that set up the Route: (the
"original-uri" parameter)
3) The URI of the Contact:/From: of the request that set up the Route: (the
new "uri" parameter)

Do we need the full expressive power of a URI (#1) to know how to route to
the proxy?
Do we need both 2 & 3? (Did I misunderstand your proposal?)

> We now have the resilience that Jonathan was addressing (thread:
> "failure of proxy), since each proxy along the Route can see
> the final destination.

But how does this help? Which is more important: ensuring that
a message follows the static route of Route:  -or- ensuring that the message

gets to the final destination?

> And also, if the request spiralled,
> we know where on the Route we are (since we have retained the
> original Request-URIs, too).
>

This is an interesting idea.

>
> Now I know that this is a significant change, but as far as I can
> tell, realistically, no proxy will as yet have been able to
> Record-Route properly and/or remain call stateful, because we
> are still finding bugs.  Thus it doesn't seem as if there can be
> real objections to making such a changes, since hopefully they
> are changes towards having a fully-working mechanism (okay, I
> guess we think it works now, but it can't be more than 3 months
> since we spotted that Record-Route: can break in the reverse
> direction when a request has spiralled; how mature can
> implementations possibly be?)
>

Precis.

> Thoughts?
>  - Jo.


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Sean Olson wrote:
>
> Some ramblings on the subject.
> /sean
> 
> Jo Hornsby wrote:
> 
> Hmmm... this backwards compatibility thing is a pain, but how
> about this:
> <snip>
> 
> > If the motivation behind retaining backwards compatibility is
> > purely a "we're going to be breaking the spirit of the RFCs/IETF",
> > then this may be a solution...
> >
> > Instead of proxies copying the Request-URI, and mangling the
> > port and maddr, why not just get them to put _their_ SIP URL in,
> > and make the Request-URI a parameter of the SIP URL (suitably
> > escaped, of course).
> 
> <snip>
> 
> > Now I know we have questioned whether the UA or proxy or whatever
> > will copy the entire Request-URI, but the spec certainly doesn't
> > say that only the user@host:port is copied, so I don't see this as
> > a valid argument.
> 
> But this should be clarified. The UA/proxy MUST copy the entire URI
> from the Route: header into the Request-URI without modification or
> omission.

Oh very much so -- this should be clarified in the bis -- but
we'll have to deal with the fact that it's not nailed down in
the spec.

> > If, OTOH, we are looking at backwards-compatibility from a "work
> > with existing SIP Implementations" standpoint, then I have to say
> > that so much of Record-Route wouldn't have worked anyway (the
> > spec says nothing about building Route:s in the callee -> caller
> > direction), is it really worth worrying?
> >
> 
> I agree completely with this sentiment.
> 
> >
> > With this approach, we can write into bis, say, that the called
> > UA, when forming requests, renames all "uri" parameters to
> > "original-uri"; and appends a new "uri" parameter with the value
> > of the Contact:/From:.
> 
> I'm not sure I see a need for three URIs in this case. You
> have in the Route: header:
> 
> 1) The URI of the Proxy  (the main URI)
> 2) The original Request-URI of the request that set up the Route: (the
> "original-uri" parameter)
> 3) The URI of the Contact:/From: of the request that set up the 
> Route: (the
> new "uri" parameter)
> 
> Do we need the full expressive power of a URI (#1) to know how to
> route to the proxy?
> Do we need both 2 & 3? (Did I misunderstand your proposal?)

I think you may have misunderstood a subtle point that I'm
trying to achieve.  I think with this approach, we get the
behaviour that a heterogeneous (in the sense RFC 2543 and
state-of-the-art bis) network of SIP entities would work as
well as a network of purely 2543 SIP entities; that is,
Record-Route would work for requests in the caller -> callee
direction.  Additionally, we might even be lucky enough to
get Record-Route working in the callee -> caller direction,
although this would require that the caller was using a
state-of-the-art UA.

(This does assume that the entire URI is copied by 2543
entities, of course.)

The reason that I'm going to these lengths, I guess, is to
maintain just enough backwards compatibility.

> > We now have the resilience that Jonathan was addressing (thread:
> > "failure of proxy), since each proxy along the Route can see
> > the final destination.
> 
> But how does this help? Which is more important: ensuring that
> a message follows the static route of Route:  -or- ensuring that 
> the message gets to the final destination?

Probably depends on who I am.  If I'm the caller, I'd probably be
quite happy if the message got to its final destination and
"happened" to miss out all of my networks billing servers. &:)

But I understand what you're getting at (I think).  For all
intents and purposes, it's only the Route that matters; it
just gives me a warm fuzzy feeling to know that I'm also being
backed up by the Request-URI.

> > And also, if the request spiralled,
> > we know where on the Route we are (since we have retained the
> > original Request-URIs, too).
> >
> 
> This is an interesting idea.
> 
> >
> > Now I know that this is a significant change, but as far as I can
> > tell, realistically, no proxy will as yet have been able to
> > Record-Route properly and/or remain call stateful, because we
> > are still finding bugs.  Thus it doesn't seem as if there can be
> > real objections to making such a changes, since hopefully they
> > are changes towards having a fully-working mechanism (okay, I
> > guess we think it works now, but it can't be more than 3 months
> > since we spotted that Record-Route: can break in the reverse
> > direction when a request has spiralled; how mature can
> > implementations possibly be?)
> >
> 
> Precis.

Cheers,


 - Jo.


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"Drage, Keith (Keith)" wrote:
> 
> I noted in Henning's list of headers for IANA that the Hide header is
> deprecated.
> 
> In the bis-02 draft, there appears to be no text that indicates this, 
> or gives any details on what the term means as regards sending and 
> receiving, apart from some discussion in annex I. Is there some 
> proposed text somewhere that states SHOULD NOT for sending, and 
> changes the "m" status in table 6 for proxies.

Keith:

Digging into some of the (pertinent) emails I have saved, I found this
from Henning that may help:

> Date: Sat, 02 Sep 2000 22:21:39 -0400
> From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
> X-Mailer: Mozilla 4.7 [en]C-CCK-MCD BA45DSL  (WinNT; U)
> X-Accept-Language: en
> MIME-Version: 1.0
> To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
> Cc: sip@lists.bell-labs.com
> Subject: Re: [SIP] transaction identification
> References: <002e01c011c2$94a85000$4e34c3c1@ubiquity.co.uk> <39AC83A0.68A61CA@dy
> namicsoft.com> <39AE1096.95161017@uab.ericsson.se> <39AE38A0.526A55F8@ubiquity.n
> et> <39AF4ADB.B8751B55@dynamicsoft.com> <39AF51F7.CDBFE713@ericsson.fi> <39AF563
> A.8EEB995@dynamicsoft.com> <39AF7E14.FE7886BB@ubiquity.net> <39B06683.FE987438@d
> ynamicsoft.com>
> Content-Type: text/plain; charset=us-ascii
> Content-Transfer-Encoding: 7bit
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> X-UID: 0000000040
>  
>  
> > It might be worthwhile to take a step back and examine whether or not
> > this kind of fix (just needed for via hiding) can rather be addressed by
> > ditching via hiding and doing something else entirely. I don't think its
> > widely implemented, and there are definintely issues with the security
> > model (the next hop provides the feature for you).
> >
>  
> This is only one of the problems. With hindsight, my other concerns
> about Via hiding:
>  
> - complexity, particularly hidden "gotchas" that surface at various
> points (as in this instance);
>  
> - interference with loop detection and debugging;
>  
> - over-promise: unlike HTTP, where via-hiding makes some sense since all
> data is in one place, Via-hiding in SIP by itself does nothing to hide
> the caller. (I'm having a hard time thinking of applications where
> hiding the caller doesn't matter, but where hiding a proxy does.)
> There's all kind of other information leakage:
>  
>   - Contact
>   - Route/Record-Route
>   - SDP (various places, including o=, c=)
>   - possibly accidental leakage in User-Agent header and Call-ID,
> depending on how they're generated
>  
> Also, unless this is implemented everywhere, the feature is not likely
> to be very useful, without the sender having any recourse such as "don't
> route this request unless you can hide". I would guess that almost all
> existing proxies simply ignore the Hide header.
>  
> Thus, there is a danger of asking for via-hiding and then deluding
> oneself
> hat this indeed provides anonymity.
>  
> I think we should welcome opportunities to simplify the spec by removing
> marginally useful features.
>  
> For real anonymity, a back-to-back UA (that creates a "fresh" SDP entry
> and otherwise scrubs the request of identifying information) seems about
> the only viable approach.
>  
> Email has existed for many years without this feature, with "mixers" for
> anonymity.
>  
> If this is truly needed, I would argue that this should be an extension
> specified separately, with a Proxy-Require header.
>  
> Henning
>  
>  
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>  

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
Internet Software & eServices Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-413
Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184

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> Hi,
> Thanks for posting the proposed new grammar. I have a couple of
> doubts with
> the new grammar. According to the grammar you have posted below
> I can have any number of  referenced-urls and ref-signatures in any
> combination. Further according to this grammar, we could have a
> ref-signature without a
> referenced url. Is this allowed?..

Thats why there was normative text attached below the grammar.
I agree that the ABNF will allow what you describe. As far as I know,
we don't have a good way to express the requirements that the text
imposes in the grammar itself. Please reread the text below the grammar
and let me know if it answers your above questions sufficiently (I think
it does).

> The draft-sip-cc-transfer-01  clearly
> states that the refsignature "contains a signature over the concatenation
> of referrer-url and referenced url". This implies that the referenced url
> is mandatory ( as you have rightly pointed out in your mail). The new
> grammar does not reflect this. Again with the new grammar i could
> leave out
> the referenced-url and the refsignature alltogether. This is clearly not
> allowed.  IMHO   the new grammar should be,
>
>      Referred-By      = ("Referred-By" | "b") ":" referrer-url ";"
> referenced-url [";" ref-signature ]
>
>       referrer-url     = ( name-addr | addr-spec )
>
>       referenced-url   = "ref" "=" "<" URL ">"
>
>      ref-signature    = signature-scheme *( ";" sig-scheme-params )
>       signature-scheme = "scheme" "=" token
>       sig-scheme-parms = token "=" ( token | quoted-string )
>
> The difference here is removing the " | " and the " * " operators  between
> the referenced url and the ref-signature.

In an earlier thread, Jonathan brought up that this (which is similar to the
last version of the grammar) forced "ref=" to occur before any of the
signature
parameters - a restriction that doesn't exist for the other "sip-guidelines"
headers (there are order dependencies in things like www-authenticate, but
we
are trying not to replicate that characteristic).

>
>
> I have another doubt. In the draft it states that the Referred-By header
> can be in any Request message. Why is this so?....It also states that the
> refernced-url must contain the url in the refer-to header. Does this imply
> that for a referred-by header to exist a refer-to header should also be
> present, or does it mean in a REFER request you have both refer-to and
> referred-by headers and when the Transferee does an INVITE to the
> target he
> includes a referred-by header without a refer-to header ?...
>

The last scenario you described is correct. That answers why a Referred-By
header can appear in any request - if I send a REFER containing a
Referred-By
header to a sip URL with method FOO, the resulting FOO would have that
Referred-By header.

> What is the motivation for limiting the Refer-To header to just
> SIP-URL and
> not nameaddr ? ....

Umm... It doesn't. It says URL, not SIP-URL. You can, for example, REFER
to an http URL. (Remember that if you get a reference for a scheme you
don't understand or are not willing to process, you just return a 503 as
per section 3.5.

>
> In your next version of the draft could you inlcude some sample messages
> with the REFER method in the call-flow diagrams?..thanks

That's the plan (and its why the draft didn't come out last week)

>
> regards,
> Ashok
>
>
> Ashok I Roy @ Hughes Software Systems
>
>
>
>
> It's going to be a few more days before I can get the next
> revision of the REFER draft out. Until then, here's what
> list discussion has guided the Referred-By syntax into.
>
> The motivations for change were
>
> 1) Allow URLs containing semicolons to be parsed correctly. This
>    introduced changing the referrer-url from SIPURL to name-addr |
>    addr-spec. Allowing the referrenced URL to be separated is a
>    little more difficult since it is not necessarily a SIP URL.
>    This syntax attempts to resolve the problem by wrapping the URL
>    with <> and requiring any <>s appearing in the URL to be escaped.
>
> 2) Allow the parameters to be completely unordered. This requires
>    more text to augment the ABNF.
>
> -------------------
>      Referred-By      = ("Referred-By" | "b") ":" referrer-url
>                                  *(   ";" referenced-url | ";"
> ref-signature  )
>      referrer-url     = ( name-addr | addr-spec )
>      referenced-url   = "ref" "=" "<" URL ">"
>      ref-signature    = signature-scheme *( ";" sig-scheme-params )
>      signature-scheme = "scheme" "=" token
>      sig-scheme-parms = token "=" ( token | quoted-string )
>
> A Referred-By header MUST contain exactly one referenced-url.
> A Referred-By header MUST contain either zero or one ref-signature.
> Any occurences of "<" or ">" within the URL value in referenced-url
> must be escaped.
>
> ------------
>
> RjS
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>
>
>
>
>


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Gethin Liddell wrote:

>
> > 2. If the server reads a request that is badly formatted except for the
> > via header, it can silently drop the message if it wants. In general,
> > the server is just being nice by returning error messages for udp
> > requests and is not obliged to do so.
>
> Is that a question or a statement?  I would expect an error response to
> be sent if the message can be decoded enough to generate a valid
> response.

Is this a requirement  or a recommendation?  (i.e. are silent failures
disallowed by the spec. )

Thanks for the clarification on the other question.

Regards,
Ranga.


>

--
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
Tel: 301 975 3664 Fax: 301 590 0932



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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Gethin Liddell wrote:
<blockquote TYPE=CITE>&nbsp;
<br>> 2. If the server reads a request that is badly formatted except for
the
<br>> via header, it can silently drop the message if it wants. In general,
<br>> the server is just being nice by returning error messages for udp
<br>> requests and is not obliged to do so.
<p>Is that a question or a statement?&nbsp; I would expect an error response
to
<br>be sent if the message can be decoded enough to generate a valid
<br>response.</blockquote>

<p>Is this a requirement&nbsp; or a recommendation?&nbsp; (i.e. are silent
failures disallowed by the spec. )
<p>Thanks for the clarification on the other question.
<p>Regards,
<br>Ranga.
<br>&nbsp;
<blockquote TYPE=CITE><a href="http://lists.bell-labs.com/mailman/listinfo/sip"></a>&nbsp;</blockquote>

<pre>--&nbsp;
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.&nbsp;
Tel: 301 975 3664 Fax: 301 590 0932</pre>
&nbsp;</html>

--------------FCB466987B4BE51B576CF90A--


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Subject: [SIP] SIP registration draft - routing via home network
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Date: Wed, 8 Nov 2000 17:40:33 +0200

I have a question regarding draft-schulzrinne-sip-register-00. In the draft
it says:
--
5 Home Services while in Visited Network
   For some applications, the user would like to employ services of the
   home network while generating outbound requests in the visited
   network.  The visiting UA needs to detect that it is in a foreign
   network and insert a Route header pointing to its home proxy server.
--
I guess this would be another use for Route header within the initial INVITE
and would be "loose" source-routing as has been discussed in some other
threads. However, how does the UA learn about it's home proxy server? Static
configuration works well if there is a single proxy address always to be
used. But in some cases the home network may assign "serving" proxies
dynamically for registering users - even based on user's service control
needs. In this case 200 OK for REGISTER would need to carry information
about the proxy, right? Should this be done by using Record-Route in
REGISTER & 200 or by some other means? 

Regards,
	Markus


> -----Original Message-----
> From: EXT Henning Schulzrinne [mailto:schulzrinne@cs.columbia.edu]
> Sent: 06 October, 2000 18:03
> To: sip@lists.bell-labs.com
> Subject: [SIP] New draft on SIP registration
> 
> 
> I've written a quick draft on some of the issues related to how to do
> SIP registration when "visiting" networks. I would appreciate 
> feedback.
> The draft does not propose new SIP extensions, except for a possible
> Digest parameter to increase registration security.
> 
> The draft is
> http://www.ietf.org/internet-drafts/draft-schulzrinne-sip-regi
> ster-00.txt
> -- 
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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> 2. If the server reads a request that is badly formatted except for
> the via header, it can silently drop the message if it wants. In
> general, the server is just being nice by returning error messages
> for udp requests and is not obliged to do so. 

> Is that a question or a statement?  I would expect an error response
> to  be sent if the message can be decoded enough to generate a valid 
> response.

> Is this a requirement  or a recommendation?  (i.e. are silent
> failures disallowed by the spec. ) 

well the spec only says SHOULD when talking about the 400 response so
no its not mandatory

-- 
Gethin Liddell
Ubiquity Software Corporation

http://www.ubiquity.net
mailto:gethin@ubiquity.net

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From: Jean-Francois Mule <jfmule@clarent.com>
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Thank you Jim.
If there are no more comments, I will then post a quick update of the ID
reflecting the groups' comments:
	- addition of Record-route in call flow sample (see Cseq discussion with
Arju)
	- minor editorial comments received

Jean-Francois.
Clarent Corp.

> -----Original Message-----
> From: James_Renkel@3com.com [mailto:James_Renkel@3com.com]
> Sent: Wednesday, November 08, 2000 9:33 AM
> To: Jean-Francois Mule
> Cc: sip@lists.bell-labs.com
> Subject: RE: Request for information on using T.30/T.37/T.38 with
> SIP/SDP /RTP
>
>
>
>
> Jean-Francois,
>
> I and the other folk here at 3Com Carrier Networks that reviewed your
> draft
> (http://search.ietf.org/internet-drafts/draft-mule-sip-t38call
> flows-00.t
> xt)
> have found it very interesting and potentially useful, but we
> don't have
> any
> other more specific comments to make at this time. Perhaps as we get
> further
> into implementation we'll have questions or comments for you.
>
> Jim Renkel
> Director, Advanced Technology & System Engineering
> 3Com Carrier Networks Business
>

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Jonathan Rosenberg wrote:
> 
> 
> 
> > -----Original Message-----
> > From: Henning Schulzrinne [mailto:hgschul@attglobal.net]
> > Sent: Sunday, October 29, 2000 8:30 AM
> > To: Kevin Lingle
> > Cc: sip@lists.bell-labs.com
> > Subject: Re: [SIP] Some brief comments on the SIP MIB
> >
> > > >
> > > > - ULS should probably at least mention LDAP as an example.
> > >
> > > ULS has undergone some changes.  We had a request to handle
> > prioritzed
> > > backup addressess.  I also got some comments that having one generic
> > > ULS address wasn't sufficient.  Systems may have multiple
> > resources they
> > > use to determine where users are.
> > >
> > > You can view the proposal on sip-mib@egroups.com archive
> > > [http://www.egroups.com/message/sip-mib/40].  There has been no
> > > feedback on the proposal yet (unfortunately that is typical).
> > > I would appreciate if you could take a quick look.  Btw, the
> > > current proposal does not actualy handle prioritized addresses
> > > correctly (i need to consider that further).  What is your view
> > > on having the ability to specify backup addresses ULS?
> >
> > Probably needs discussion by those building proxies, as I can
> > primarily
> > speak from our own experience. In our server, locations are
> > derived from
> > several different sources
> >
> > - LDAP
> > - an SQL database for registrations
> > - a set of user-configurable scripts that provide mappings
> > from SIP user
> > ids to other user id's, e.g., resolution of email aliases.
> >
> > While being able to change the LDAP URL via the management console is
> > useful, I doubt that describing or changing the others is particularly
> > helpful (and not likely to be feasible, given that just
> > connecting to a
> > different SQL server just doesn't magically make the table definitions
> > appear there).
> 
> I think the notion of generalzied, configurable ULS through SNMP is
> fundamentally a hopeless cause. THe problem is that a location service is an
> abstract concept. It can be the result of program execution, LDAP queries,
> SQL queries, ENUM, finger, whois, local files, IVR interaction, etc. Your
> assumption is that in all cases there is at least an IP address that
> indicates where the service is. I don't think thats generally true. Certain
> location services (local file, IVR interaction, program execution), don't
> really have such a concept.

this is the general feeling i've been having about this.  it's good to hear
you say this.  that said, i'd still like to explore the issue a bit more.

> 
> So, there are two choices. Either completely drop this ULS thing, or
> restrict it to a specific set of known location services which do have a
> common set of configuration information. Access to an external database is
> probably pretty broad but has a common set of information that you could
> configure (IP address, username and password for access, port, transport,
> database instance name)

i can't drop ULS completely w/out replacing it with something.  primarily
because to this point, the ULS addr object has been used for a gateway to
specify the ip address of the sip proxy/redirect it uses.  

question: is calling a sip server an abuse or mis-use of the term "user 
location server"??  i was thinking that from a gateway's perspective, it 
looks to a sip server as one potential user location server (ie, doesn't know
where to go next to setup session to a terminating gateway)  now i'm 
wondering if that's confusing sip things with non-sip things like external
databases.  i've been using the term that way for a long time and noone 
has really objected.

so i will at least need sip server address info somewhere to replace ULS addr 
object if it goes away.  perhaps a specific sipServerAddr object added to SIP-UA-MIB??   
we were trying to also allow for backup sip server addresses... a request
we got a while back from someone on the sip-mib list.  do you think that is of value??

i'd like to try and identify a small set of known location services that we
think will be common enough that you'd expect to find them used by most sip 
entities.  if such a list can be identified then the question is how appropriate 
is that content for a standard sip mib?  if used by most sip entities, it would be 
convenience to provide the ability to address these resources.  the alternative
would be non-standard variety of ways to configure such things (other mibs). 
i could use some wisdom/guidance here, jonathan.

kevin
-- 
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
 Kevin R. Lingle       919.392.2029
 checkout: http://wwwin-eng.cisco.com/Eng/IOS/SNMP_WWW/mib-police.html
 Sometimes I think I understand everything, then I regain consciousness.
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=

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From: Alan Johnston <alan.johnston@wcom.com>
Subject: Re: [SIP] UAC Record Routing in draft-ietf-sip-call-flows-01.txt
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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I will fix this.  I thought RFC2543bis read that the maddr was required, but I
see upon re-reading that this is only true for proxies.

Alan.

Jonathan Rosenberg wrote:

> > -----Original Message-----
> > From: Neil Deason [mailto:ndeason@ubiquity.net]
> > Sent: Friday, November 03, 2000 8:34 AM
> > To: Sip List
> > Subject: [SIP] UAC Record Routing in draft-ietf-sip-call-flows-01.txt
> >
> >
> > The way the UAC builds the Route on a final 200 response
> > in 3.1.2 of the call flows draft doesn't look quite
> > right ...
> >
> > 3.1.2 Successful SIP to SIP through two proxies
> >
> >    User A          Proxy 1          Proxy 2          User B
> >      |                |                |                |
> >      |   INVITE F1    |                |                |
> >      |--------------->|                |                |
> >      |     407 F2     |                |                |
> >      |<---------------|                |                |
> >      |     ACK F3     |                |                |
> >      |--------------->|                |                |
> >      |   INVITE F4    |                |                |
> >      |--------------->|   INVITE F5    |                |
> >      |    (100) F6    |--------------->|   INVITE F7    |
> >      |<---------------|    (100) F8    |--------------->|
> >      |                |<---------------|                |
> >      |                |                |     180 F9     |
> >      |                |    180 F10     |<---------------|
> >      |     180 F11    |<---------------|                |
> >      |<---------------|                |     200 F12    |
> >      |                |    200 F13     |<---------------|
> >      |     200 F14    |<---------------|                |
> >      |<---------------|                |                |
> >      |     ACK F15    |                |                |
> >      |--------------->|    ACK F16     |                |
> >      |                |--------------->|     ACK F17    |
> >      |                |                |--------------->|
> >
> > F14 200 OK Proxy 1 -> A
> >
> >    SIP/2.0 200 OK
> >    Via: SIP/2.0/UDP here.com:5060
> >    Record-Route: <sip:UserB@there.com;maddr=ss2.wcom.com>,
> >     <sip:UserB@there.com;maddr=ss1.wcom.com>
> >    From: BigGuy <sip:UserA@here.com>
> >    To: LittleGuy <sip:UserB@there.com>;tag=314159
> >    Call-ID: 12345601@here.com
> >    CSeq: 1 INVITE
> >    Contact: LittleGuy <sip:UserB@there.com>
> >    Content-Type: application/sdp
> >    Content-Length: 134
> >
> >    v=0
> >    o=UserB 2890844527 2890844527 IN IP4 there.com
> >    s=Session SDP
> >    c=IN IP4 110.111.112.113
> >    t=0 0
> >    m=audio 3456 RTP/AVP 0
> >    a=rtpmap:0 PCMU/8000
> >
> >
> >    F15 ACK A -> Proxy 1
> >
> >    ACK sip:UserB@there.com SIP/2.0
> >    Via: SIP/2.0/UDP here.com:5060
> >    Route: <sip:UserB@there.com;maddr=ss2.wcom.com>,
> >     <sip:UserB@there.com;maddr=110.111.112.113>
> >    From: BigGuy <sip:UserA@here.com>
> >    To: LittleGuy <sip:UserB@there.com>;tag=314159
> >    Call-ID: 12345601@here.com
> >    CSeq: 1 ACK
> >    Content-Length: 0
> >
> > The UAC should build the Route header in the ACK by
> > copying the 200 response Record-Route header,
> > reversing the order and then adding the Contact as
> > the last entry. (Note the 1st entry in the Route is
> > subsequently removed when the client actually sends F15).
> > But in this case the last entry has become an IP address.
> > Shouldn't the UAC just copy the Contact and not try
> > and do any resolution of it?
>
> Correct. This behavior is really bad, actually. It depends on being able to
> perform a resolution of the Contact address outside of the realm where it is
> supposed to be resolved. If there is a private DNS running, or some kind of
> local information is placed in the hostname, this resolution will fail at
> the called party.
>
> Alan - can you please fix?
>
> Thanks,
> Jonathan R.
>
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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Subject: Re: [SIP] Some brief comments on the SIP MIB
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Kevin Lingle wrote:
> 

> 
> i can't drop ULS completely w/out replacing it with something.  primarily
> because to this point, the ULS addr object has been used for a gateway to
> specify the ip address of the sip proxy/redirect it uses.

That is not in any way related to a "location server". If you want to
specify the default outbound proxy (which this seems to be), that would
be useful, but it's a separate category from a location server.


-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Subject: [SIP] REGISTER with non-SIP URL
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Hi All,
    Can somebody please suggest what may be response for a REGISTER request
with a non-SIP URL, say a TEL URL?
    Should it be a 400 response or a 405 response?

Thanks 'N Regards,
Prasanna

Huawei Technologies
India

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http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2919.6307" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT color=3D#000080 face=3DVerdana size=3D2><SPAN =
class=3D725394504-07112000>Hi=20
All,</SPAN></FONT></DIV>
<DIV><FONT color=3D#000080 face=3DVerdana size=3D2><SPAN=20
class=3D725394504-07112000>&nbsp;&nbsp;&nbsp; Can somebody please =
suggest what may=20
be response for a REGISTER request with a non-SIP URL, say a TEL=20
URL?</SPAN></FONT></DIV>
<DIV><FONT color=3D#000080 face=3DVerdana size=3D2><SPAN=20
class=3D725394504-07112000>&nbsp;&nbsp;&nbsp; Should it be a 400 =
response or a 405=20
response?</SPAN></FONT></DIV>
<DIV><FONT color=3D#000080 face=3DVerdana size=3D2><SPAN=20
class=3D725394504-07112000></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000080 face=3DVerdana size=3D2><SPAN=20
class=3D725394504-07112000>Thanks 'N Regards,</SPAN></FONT></DIV>
<DIV><FONT color=3D#000080 face=3DVerdana size=3D2><SPAN=20
class=3D725394504-07112000>Prasanna</SPAN></FONT></DIV>
<DIV><FONT color=3D#000080 face=3DVerdana size=3D2><SPAN=20
class=3D725394504-07112000></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000080 face=3DVerdana size=3D2><SPAN=20
class=3D725394504-07112000>Huawei Technologies</SPAN></FONT></DIV>
<DIV><FONT color=3D#000080 face=3DVerdana size=3D2><SPAN=20
class=3D725394504-07112000>India</SPAN></FONT></DIV></BODY></HTML>

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--------------------------------------------------------------
HUAWEI (BANGALORE) BUSINESS OPERATION, BANGALORE, INDIA



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Date: Wed, 8 Nov 2000 11:56:43 -0000
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Hi,

This will be the third time that I have posted this message, without
receiving a satisfactory reply.
Please gimme some feedback!

Thanks
Phil

----------------------------------------------------------------------------
--

Hi,

Table 4 of bis02 states that a Contact header can occur in the following
response types:

1xx
2xx
3xx
485 (Ambiguous)

However, Table 5 states that Record-Route can occur in the following
response types:

2xx
401 (Unauthorized)
484 (Address Incomplete)

I don't see how a response can include a Record-Route, if it can't contain a
Contact.
I believe that 401 and 484 should be added to the list of response types for
a Contact header.

What do you think?

Cheers
Phil


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Date: Wed, 8 Nov 2000 11:33:19 -0600



Jean-Francois,

I and the other folk here at 3Com Carrier Networks that reviewed your draft
(http://search.ietf.org/internet-drafts/draft-mule-sip-t38callflows-00.txt)
have found it very interesting and potentially useful, but we don't have any
other more specific comments to make at this time. Perhaps as we get further
into implementation we'll have questions or comments for you.

Jim Renkel
Director, Advanced Technology & System Engineering
3Com Carrier Networks Business



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Date: Wed, 08 Nov 2000 15:28:35 -0500
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"Henning G. Schulzrinne" wrote:
> 
> Kevin Lingle wrote:
> >
> 
> >
> > i can't drop ULS completely w/out replacing it with something.  primarily
> > because to this point, the ULS addr object has been used for a gateway to
> > specify the ip address of the sip proxy/redirect it uses.
> 
> That is not in any way related to a "location server". If you want to
> specify the default outbound proxy (which this seems to be), that would
> be useful, but it's a separate category from a location server.

that was what i was afraid of :(  the mib needs to be corrected then.
funny, this uls addr object has been in the mib since -00 rev of the I-D
and the description has been very clear that a sip server was a possible 
use of the object.  

i am going to create a new object in the SIP-UA-MIB that is specifically
for configuring the address of the sip server a user agent use.  again,
is there any value in allowing for multiple server addresses to be 
configured (eg, for backup purposes)?  we had a request for that a while
back on sip-mib alias.

kevin
> 
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

-- 
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 Kevin R. Lingle       919.392.2029
 checkout: http://wwwin-eng.cisco.com/Eng/IOS/SNMP_WWW/mib-police.html
 Sometimes I think I understand everything, then I regain consciousness.
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Date: Wed, 8 Nov 2000 16:09:27 -0500 (EST)

Prasanna <itpl@in.huawei.com> wrote:
> Hi All,
>     Can somebody please suggest what may be response for a REGISTER request
> with a non-SIP URL, say a TEL URL?
>     Should it be a 400 response or a 405 response?

How about 200?

There are two URLs in the register request - the address-of-record
whose registration is to created or updated (the To: header), and
the place future non-REGISTER requests should be directed (the
Contact: header).

A tel: URL in the To: header is asking for a registration to be
created or updated with that address, which admittedly may end up
with less than global scope.

A tel: URL in the Contact: header is asking for future requests
to be directed to a PSTN gateway.

A tel: URL in both may be nearly useless, but there seems little
reason to reject it.  The sender obviously thought there was some
point in having such an entry at the registrar.

Bill Marshall
wtm@research.att.com

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how about this configuring sip server address in a user agent?
this table would be defined in the SIP-UA-MIB module and allow 
the old sipUserLocationServerAddress object in the SIP-COMMON-MIB
to no longer be used (incorrectly) for this purpose... and
actually be removed.

note: a table and the use of applIndex are inherent in the
      way all of the sip mib modules are defined 
      (see http://search.ietf.org/internet-drafts/draft-ietf-sip-mib-01.txt)


   -- 
   -- SIP Server Configuration 
   --     
   sipUACfgSipServerTable OBJECT-TYPE 
       SYNTAX      SEQUENCE OF SipUACfgSipServerEntry 
       MAX-ACCESS not-accessible 
       STATUS     current 
       DESCRIPTION 
            "This table contains SIP server configuration objects 
             applicable to each SIP user agent in this system.  The 
             instances of SIP entities are uniquely identified 
             by applIndex." 
       ::= { sipUACfgSipServer 1 } 
    
   sipUACfgSipServerEntry OBJECT-TYPE 
       SYNTAX     SipUACfgSipServerEntry 
       MAX-ACCESS not-accessible 
       STATUS     current 
       DESCRIPTION 
            "A row of server configuration." 
       INDEX { applIndex, sipUACfgSipServerIndex } 
       ::= { sipUACfgSipServerTable 1 } 
    
   SipUACfgSipServerEntry ::= 
       SEQUENCE { 
                sipUACfgSipServerAddrIndex   Unsigned32,
                sipUACfgSipServerAddrType    InetAddressType,
                sipUACfgSipServerAddr        InetAddress,
                sipUACfgSipServerStatus      RowStatus
       } 
   
   sipUACfgSipServerAddrIndex OBJECT-TYPE 
       SYNTAX     Unsigned32
       MAX-ACCESS not-accessible 
       STATUS     current 
       DESCRIPTION 
            "A unique identifier of a server address when
             multiple addresses are configured by the SIP entity.
             If one address isn't reachable, then another can
             be tried."
       ::= { sipUACfgSipServerEntry 1 } 

   sipUACfgSipServerAddrType OBJECT-TYPE 
       SYNTAX     InetAddressType
       MAX-ACCESS read-create
       STATUS     current 
       DESCRIPTION 
            "This object specifies the type of address contained
             in the associated instance of sipUACfgSipServerAddr."
       DEFVAL { ipv4 }
       REFERENCE "INET-ADDRESS-MIB (RFC 2851)"
       ::= { sipUACfgSipServerEntry 2 } 
    
   sipUACfgSipServerAddr OBJECT-TYPE 
       SYNTAX     InetAddress
       MAX-ACCESS read-create
       STATUS     current 
       DESCRIPTION 
            "This object specifies the address of a SIP server
             this user agent will use to proxy/redirect calls."
       ::= { sipUACfgSipServerEntry 3 } 

   sipUACfgSipServerAddrStatus OBJECT-TYPE 
       SYNTAX     RowStatus
       MAX-ACCESS read-create
       STATUS     current 
       DESCRIPTION 
            "This object is used to control rows in this table.
            
             'active'        : the row's information is completely
                               populated and that information is 
                               being used by the user agent.

             'notInService'  : the row's address is not being used
                               by the user agent, but will remain in the table.

             'notReady'      : key information is missing thus, preventing
                               the row from being made 'active' (eg, no
                               address specified).

             'createAndGo'   : only allowed if the manager also provides
                               a varbind for sipUACfgSipServerAddr object
                               in the same set operation.

             'createAndWait' : not applicable.

             'destroy'       : the row's address will no longer be used
                               by the user agent and the row will be
                               removed from the table."
       ::= { sipUACfgSipServerEntry 4 } 
"Henning G. Schulzrinne" wrote:
> 
> Kevin Lingle wrote:
> >
> 
> >
> > i can't drop ULS completely w/out replacing it with something.  primarily
> > because to this point, the ULS addr object has been used for a gateway to
> > specify the ip address of the sip proxy/redirect it uses.
> 
> That is not in any way related to a "location server". If you want to
> specify the default outbound proxy (which this seems to be), that would
> be useful, but it's a separate category from a location server.
> 
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

-- 
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
 Kevin R. Lingle       919.392.2029
 checkout: http://wwwin-eng.cisco.com/Eng/IOS/SNMP_WWW/mib-police.html
 Sometimes I think I understand everything, then I regain consciousness.
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=

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Date: Wed, 8 Nov 2000 15:19:57 -0600



Kevin,

I think there is a definite need for configuring multiple server addresses in a
SIP UAC. For example, our PSTN gateways can be configured for up to 8 SIP
proxies. Using REGISTERs, they determine which ones are alive, and then
distribute calls among the alive SIP proxies via one of a number of user
selectable algorithms.

I think this would be a very good addition to the SIP UA MIB.

Jim





Kevin Lingle <klingle@cisco.com> on 11/08/2000 02:28:35 PM

Sent by:  Kevin Lingle <klingle@cisco.com>


To:   "Henning G. Schulzrinne" <hgs@cs.columbia.edu>
cc:   Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@lists.bell-labs.com,
      sip-mib@egroups.com (James Renkel/MW/US/3Com)
Subject:  Re: [SIP] Some brief comments on the SIP MIB



"Henning G. Schulzrinne" wrote:
>
> Kevin Lingle wrote:
> >
>
> >
> > i can't drop ULS completely w/out replacing it with something.  primarily
> > because to this point, the ULS addr object has been used for a gateway to
> > specify the ip address of the sip proxy/redirect it uses.
>
> That is not in any way related to a "location server". If you want to
> specify the default outbound proxy (which this seems to be), that would
> be useful, but it's a separate category from a location server.

that was what i was afraid of :(  the mib needs to be corrected then.
funny, this uls addr object has been in the mib since -00 rev of the I-D
and the description has been very clear that a sip server was a possible
use of the object.

i am going to create a new object in the SIP-UA-MIB that is specifically
for configuring the address of the sip server a user agent use.  again,
is there any value in allowing for multiple server addresses to be
configured (eg, for backup purposes)?  we had a request for that a while
back on sip-mib alias.

kevin
>
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

--
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
 Kevin R. Lingle       919.392.2029
 checkout: http://wwwin-eng.cisco.com/Eng/IOS/SNMP_WWW/mib-police.html
 Sometimes I think I understand everything, then I regain consciousness.
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=

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Date: Wed, 8 Nov 2000 14:58:04 -0700


Hello, does anybody have stress test performance numbers of a SIP proxy,
UAC, and UAS? I am interested in a sustained call rate as well as a burst
call rate.  Thanks.

Jin-Gen Wang
Level 3 Communications

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From sip-admin@lists.bell-labs.com  Wed Nov  8 17:00:10 2000
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Subject: Re: [SIP] SIP registration draft - routing via home network
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I would like to avoid configuration as much as possible, as every
configuration option increases complexity. Thus, 'home' is unambiguously
defined by the user's From address. The server at that address can then
forward the request as needed. I would be curious why this is not
sufficient, as it is the same principle used for incoming calls.

We may be submitting a draft on general UA configuration soon; it might
be appropriate to include this there.

Also, with the discussion about Route headers, something like

Route: sip:callee@callee-domain.com ;at=home.com

would avoid problems on URL content (or maddr).


Markus.Isomaki@nokia.com wrote:
> 
> I have a question regarding draft-schulzrinne-sip-register-00. In the draft
> it says:
> --
> 5 Home Services while in Visited Network
>    For some applications, the user would like to employ services of the
>    home network while generating outbound requests in the visited
>    network.  The visiting UA needs to detect that it is in a foreign
>    network and insert a Route header pointing to its home proxy server.
> --
> I guess this would be another use for Route header within the initial INVITE
> and would be "loose" source-routing as has been discussed in some other
> threads. However, how does the UA learn about it's home proxy server? Static
> configuration works well if there is a single proxy address always to be
> used. But in some cases the home network may assign "serving" proxies
> dynamically for registering users - even based on user's service control
> needs. In this case 200 OK for REGISTER would need to carry information
> about the proxy, right? Should this be done by using Record-Route in
> REGISTER & 200 or by some other means?
> 
> Regards,
>         Markus

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Subject: Re: [SIP] Possible REFER problem?
To: etremblay@mediatrix.com (Eric Tremblay)
Cc: jdrosen@dynamicsoft.com ('Jonathan Rosenberg'),
        Cliff.Harris@nokia.com ('Cliff.Harris@nokia.com'),
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Eric Tremblay writes:
>I'm reviving and old thread ;)
>
>If you remember, the problem was when doing call transfer with consultation
>with the REFER request, how can we make sure that the INVITE request (from
>the Transferee to the Transfer Target) is going to reach the exact same
>User-Agent as the one that was consulted by the Transferor.  
>
>The solution was to use the header "Accept-Contact" and specify the contact
>of the transfer target.

<explanation of why this won't necessarily work snipped>

At the risk of sounding daft, I don't see why using the "Contact:"
header as the new "INVITE" URI is considered Evil in this case.

When explaining the difference between, say, the "From:" (or "To:")
field and the "Contact:" field, I generally think of "From:"/"To:" as
being the persistent URI for contacting the appropriate resource in 
an abstract sense (by resource, I mean "Adam Roach" or "helpdesk staffer" 
or "sales representative"), while the "Contact:" URI indicates "where to
get to me (as a particular instance of that resource) on this particular
device at this particular moment."

Obviously, the most useful application of this latter piece of information
is routing of subsequent messages for a particular call -- but I don't
see why it absolutely MUST be the only use for this information.

For the "Contact" information to be useful in any way, it needs to be
something that can be routed when used as a Request-URI; it wouldn't
work for "BYE" messages otherwise, right? 

So, if the transferee were passed this information to be used as 
the Request-URI, there should be no discernable problems with the 
INVITE (or other appropriate method) arriving at the partcular 
instance of the desired resource on a particular device.

Since no one else has raised this point, I'm going to assume that
I'm too blind to see the elephant in the room. So, could someone 
explain *why* using a Contact URI in this manner would be inappropriate
behavior?

--
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA

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From sip-admin@lists.bell-labs.com  Wed Nov  8 19:33:02 2000
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From: Jean-Francois Mule <jfmule@clarent.com>
Reply-To: Jean-Francois Mule <jfm@clarent.com>
To: drwalker@ss8networks.com, joon_maeng@vtel.com, klingle@cisco.com
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Subject: [SIP] Some comments on the SIP MIB - operState/AdminState (draft-ietf-sip-mib-01.txt)
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The comments here address mainly the application state variables: 2
variables in the current sip mib: sipServiceOperStatus and
sipServiceAdminStatus.

--- Usage status:
When the SIP entity is up, it would be nice to have a simple variable
showing its usage.  This way, as a sysadmin, I could query the app and if it
is up and idle, or up and busy.
Could we add a usage variable?   Something like:
   sipServiceUsageStatus OBJECT-TYPE
       SYNTAX     INTEGER

			idle(0),
			active(1),
			busy(2)
                  }
       MAX-ACCESS read-only
       STATUS     current
       DESCRIPTION
            "This object contains the current usage state of the
             SIP application.

             idle(0)         : The application is currently not in use.
             active(1)       : The application is in use, and has sufficient
spare operating capacity.
             busy(2)         : The application is in use but it has no spare
operating capacity."
       ::= { sipCommonCfgEntry x }


--- Regarding the number of state variables
The operational status variable contains numerous state information which
overlap.
For example, when the status is 'testing', the sysadmin does not know from
the MIB whether the service is up or down.  On large scale systems, we may
want to allow test procedures on some SIP functionality of the device while
calls are being served.  Unless we have a way to describe this properly, we
expose ourselves to implementation & operational issues.

The operability of the SIP service should be binary: enabled or disabled.
We could have another variable to reflect the procedural aspects (testing,
initializing, ...) or standby options.
Same applies to the AdminState:  I'd prefer keep a convention already
established where the admin state is either locked (sip application
administratively down), shutting down or unlocked.

Comments?
Without getting into too much detailed vars, I believe we could clarify the
admin/oper/usage states along with some procedural vars so that both vendors
and operators have a clear way of setting/querying the SIP apps behavior.

Cheers,
Jean-Francois Mule
Clarent Corp.

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From: Jean-Francois Mule <jfmule@clarent.com>
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Subject: [SIP] Some comments on the SIP MIB - configuration of security options (draft-ietf-sip-mib-01.txt)
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The current MIB allows to specify the authentication method that is used by
servers to authenticate request originators (sipProxyAuthMethod).   The
granularity is this config param means: a proxy will challenge all requests
or none.

There might be cases where a proxy requires authentication on some but not
all requests  (typically, authenticate REGISTER only then you are trusted
and can speak freely ;-), or authenticate REGISTER, INVITE, BYE -- for the
ACK, it is okay, etc.).

Shall we allow this level of granularity for the configuration of sip
proxies?

Jean-Francois Mule
Clarent Corp.


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To: Jean-Francois Mule <jfm@clarent.com>
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Subject: Re: [SIP] Some comments on the SIP MIB - configuration of security 
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Jean-Francois Mule wrote:
> 
> The current MIB allows to specify the authentication method that is used by
> servers to authenticate request originators (sipProxyAuthMethod).   The
> granularity is this config param means: a proxy will challenge all requests
> or none.
> 
> There might be cases where a proxy requires authentication on some but not
> all requests  (typically, authenticate REGISTER only then you are trusted
> and can speak freely ;-), or authenticate REGISTER, INVITE, BYE -- for the
> ACK, it is okay, etc.).
> 
> Shall we allow this level of granularity for the configuration of sip
> proxies?

Unfortunately, it's more complicated than that. Our server allows
configuration for each local user, so that (for example)
help@cs.columbia.edu requests, but does not require authentication for
non-REGISTER (but does for REGISTER), while calls to
chair@cs.columbia.edu requires authentication for all requests. I really
doubt you want to carry all of this in SNMP.

> 
> Jean-Francois Mule
> Clarent Corp.
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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From: "Sriram Parameswar" <sriramp@nortelnetworks.com>
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Subject: [SIP] Caller Preferences and MESSAGE method
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Hi,

I think there is definite value in adding the MESSAGE method to the
draft-ietf-sip-callerprefs-02. I see *all three* headers being of value to
the MESSAGE method i.e. Accept-Contact, Reject-Contact and
Request-Disposition.

Some examples:

1. The sender of the IM could require that his Message be delivered only to
a device with a class = "business".
   This could use the Accept-Contact header.
2. If you want to save your buddy valuable air-time charges you could send
an IM with the following header
   Reject-Contact: sip:mybuddy@downtown.com;mobility="mobile" 
3. You could use the Request-Disposition to ensure that the Contact list
returned for the Message is tried sequentially.

Thanks and Regards,

Sriram Parameswar
Nortel Networks

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	charset="iso-8859-1"
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<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2652.35">
<TITLE>Caller Preferences and MESSAGE method</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Hi,</FONT>
</P>

<P><FONT SIZE=3D2>I think there is definite value in adding the MESSAGE =
method to the draft-ietf-sip-callerprefs-02. I see *all three* headers =
being of value to the MESSAGE method i.e. Accept-Contact, =
Reject-Contact and Request-Disposition.</FONT></P>

<P><FONT SIZE=3D2>Some examples:</FONT>
</P>

<P><FONT SIZE=3D2>1. The sender of the IM could require that his =
Message be delivered only to a device with a class =3D =
&quot;business&quot;.</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; This could use the Accept-Contact =
header.</FONT>
<BR><FONT SIZE=3D2>2. If you want to save your buddy valuable air-time =
charges you could send an IM with the following header</FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp; Reject-Contact: =
sip:mybuddy@downtown.com;mobility=3D&quot;mobile&quot; </FONT>
<BR><FONT SIZE=3D2>3. You could use the Request-Disposition to ensure =
that the Contact list returned for the Message is tried =
sequentially.</FONT></P>

<P><FONT SIZE=3D2>Thanks and Regards,</FONT>
</P>

<P><FONT SIZE=3D2>Sriram Parameswar</FONT>
<BR><FONT SIZE=3D2>Nortel Networks</FONT>
</P>

</BODY>
</HTML>
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From sip-admin@lists.bell-labs.com  Thu Nov  9 00:02:05 2000
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From: Jean-Francois Mule <jfmule@clarent.com>
Reply-To: Jean-Francois Mule <jfm@clarent.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, archow@hss.hns.com,
        sip@lists.bell-labs.com
Subject: RE: [SIP] SIP proxies and CSeq numbers (was: SIP and T.38 fax cal ls - Internet Draft)
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If No, then No.
For my own records, can you quote the part of the current specification(s)
stating that proxies MUST NOT modify the Cseq?  I've screened rfc2543bis-02
again and I cannot find any statement that would flag the proposed call flow
as "not compliant" (the call flow in question illustrated a proxy modifying
Cseq on 2 call legs).  Any time the usage of Cseq is explained, it is in
reference to a particular call leg or SIP transaction.

It understand it is one thing to implement a profile of the spec that
follows the common sense to encourage interoperability(why would you ever
purposely increment Cseq in a proxy?), however, the spec should
unambiguously clarify this.  And this was the intent of this separate
thread.

Jean-Francois Mule
PS: I'll modify the call flow to reflect the same Cseq on both sides because
this is not the intent of the SIP-T.38 ID.  So, yes.

Jonathan Rosenberg wrote:
>
> No.
>
> The call flow is a very standard re-INVITE type of thing.
> There is no reason
> why this proxy should behave any differently from any other
> proxy. Proxies
> do not modify CSeq. Period. There is no reason why this
> cannot be a standard
> flow, and for purposes of interoperability, it should be. The
> flow you have
> in the document is not compliant to the spec.
>
> -Jonathan R.


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From sip-admin@lists.bell-labs.com  Thu Nov  9 01:41:04 2000
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Hi ,
Went through the distributable call state draft. It is a move in the
right
direction as the proxies should be holding as little state information
as
possible in order to scale up.

But it seems the idea of having a separate state header might not be
needed. The nesting of  state headers across proxies adds to
complexity and overhead. Existing UA implementations cannot
support state header.

An alternative will be to use the Record-route option already defined.
The proxy which adds a  record route header(means the proxy is
interested
in keeping track of the call.)can add the state info duly encrypted as a

parameter to record route header.

Now only the proxy which inserted the record route header needs to
interpret
the state parameters. Note that record route is valid per call leg
basis. Thus the
proxy can selectively choose call legs which it is interested in. .The
UA
implementations need not worry as they are supposed to insert the
Record-route
header unchanged into the responses.

As record - routing support is mandatory, this seems to be a better
solution that
needs very little changes. Also alleviates the need to live with one
more header!!!

This is what I feel from the first look...Or am I missing something?


Thanks & regards
Anuraj Ennai
SIP Designer
Wipro Technologies -Bangalore-India




Subject:
        [SIP] I-D ACTION:draft-ietf-sip-state-00.txt
   Date:
        Tue, 07 Nov 2000 06:05:14 -0500
   From:
        Internet-Drafts@ietf.org
     To:
        IETF-Announce: ;
    CC:
        sip@lists.bell-labs.com



A New Internet-Draft is available from the on-line Internet-Drafts
directories.
This draft is a work item of the Session Initiation Protocol Working
Group of the IETF.

        Title           : SIP Extensions for supporting Distributed Call
State
        Author(s)       : B. Marshall et al.
        Filename        : draft-ietf-sip-state-00.txt
        Pages           : 12
        Date            : 06-Nov-00

This document describes an extension to the Session Initiation
Protocol (SIP) that enables proxies to distribute call state to user
agents. The state information can be returned to the proxy when the
user agent requests a change in the characteristics of the active
call. By providing the ability to distribute state to the user
agents where it can be securely stored, proxy servers can remain
stateless for the duration of the call. This mechanism allows a
proxy server to provide services that depend on call state, while
still being stateless.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-state-00.txt

Internet-Drafts are also available by anonymous FTP. Login with the
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A list of Internet-Drafts directories can be found in
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Date: Thu, 9 Nov 2000 13:15:44 +0530



Robert , thanks for the clarifications.  I understood the text after the
grammar, just that i thought that the grammar should be sufficient to
reflect that.
Is there a problem with describing the grammar like this,
                Referred-By      = ("Referred-By" | "b") ":" referrer-url
";" (referred-by-params)
                referred-by-params   =  ";" referenced-url  [";"
ref-signature ]

                              | ";" ref-signature ";" referenced-url


I am not sure if this allowed by ABNF formal specs, but i don't see why it
shouldn't be.

>>>>>>



> What is the motivation for limiting the Refer-To header to just
> SIP-URL and
> not nameaddr ? ....

>>Umm... It doesn't. It says URL, not SIP-URL. You can, for example, REFER
>>to an http URL. (Remember that if you get a reference for a scheme you
>>don't understand or are not willing to process, you just return a 503 as
>>per section 3.5.

>>>>>>>>

Sorry about that....i wasn't too clear...i was asking you why the display
name was not allowed...

ex:  "ashok I roy " < airoy@hss.hns.com >

In the Refer-To header only the URL is allowed.

again why isn't that allowed in the referenced-url of the Referred-By
header ?
In the referenced-url it would be more appropriate to allow it, becuase the
new grammar mandates angle braces around
the URL.

Thanks.

regards,

-Ashok

-------------------------------------
Ashok I Roy @ Hughes Software Systems
-------------------------------------



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From: "Hisham Khartabil" <hisham.khartabil@hotsip.com>
To: <sip@lists.bell-labs.com>
Message-ID: <GEEMIMOPEJGBIEGHJBHDKENECAAA.hisham.khartabil@hotsip.com>
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Subject: [SIP] Contact header is a 200 response to SUBSCRIBE for IMPP
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Date: Thu, 9 Nov 2000 11:40:57 +0200
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Section 4.2 of the presence draft has the following examples:

SUBSCRIBE sip:jdrosen@dynamicsoft.com SIP/2.0
Via: SIP/2.0/UDP userpc.example.com
To: sip:jdrosen@dynamicsoft.com
From: sip:user@example.com
Contact: sip:user@userpc.example.com
Call-ID: knsd08alas9dy@3.4.5.6
CSeq: 1 SUBSCRIBE
Content-Length: 0



SIP/2.0 200 OK
Via: SIP/2.0/UDP presenceserver.dynamicsoft.com
Via: SIP/2.0/UDP userpc.example.com
To: sip:jdrosen@dynamicsoft.com
From: sip:user@example.com
Contact: sip:user@userpc.example.com
Record-Route: sip:jdrosen@example.com;maddr=presenceserver.dynamicsoft.com
Expires: 3600
Call-ID: knsd08alas9dy@3.4.5.6
CSeq: 1 SUBSCRIBE
Content-Type: application/xpidf+xml
Content-Length: 287

<?xml version="1.0"?>
<!DOCTYPE presence
 PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
 <presentity uri="sip:jdrosen@dynamicsoft.com;method="SUBSCRIBE">
  <atom id="779js0a98">
   <address uri="sip:jdrosen@dynamicsoft.com;method=INVITE">
    <status status="open"/>
   </address>
  </atom>
 </presentity>
</presence>


Then, it goes on to say:
"Notice that the response has mirrored many fields from the
request,including the To, From, Call-ID, CSeq, and Via headers. The PA has
also added a Contact header, providing an address where it can be reached.
The response also contains a Record-Route header. This header was presumably
inserted by the proxy (the address in the maddr header is that of a presence
server acting as a proxy), and is reflected in the 200 OK response. The
response also contains a presence document for the presentity, and a Contact
header. The Contact headers in a response to SUBSCRIBE list all the current
addresses which have been subscribed."

The example shows that the Contact header in the response does actually list
the current address which has been subscribed (user@userpc.example.com).
But I fail to see where the Contact header that the PA added providing an
address where it can be reached.

Regards,
Hisham Khartabil
Hotsip Finland




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From: "Hisham Khartabil" <hisham.khartabil@hotsip.com>
To: "Phil Hoffer" <phoffer@ubiquity.net>, <sip@lists.bell-labs.com>
Subject: RE: [SIP] Use of Record-Route and Contact Headers
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I think the way it works is that if a Contact header has been omitted in a
response and there is a record-route, you would use the URI in the To:
field.

Regards,
Hisham
Hotsip Finland

-----Original Message-----
From: sip-admin@lists.bell-labs.com [mailto:sip-admin@lists.bell-labs.com]On
Behalf Of Phil Hoffer
Sent: Wednesday, 8 November 2000 1:57 PM
To: sip@lists.bell-labs.com
Subject: [SIP] Use of Record-Route and Contact Headers

Hi,

This will be the third time that I have posted this message, without
receiving a satisfactory reply.
Please gimme some feedback!

Thanks
Phil

----------------------------------------------------------------------------
--

Hi,

Table 4 of bis02 states that a Contact header can occur in the following
response types:

1xx
2xx
3xx
485 (Ambiguous)

However, Table 5 states that Record-Route can occur in the following
response types:

2xx
401 (Unauthorized)
484 (Address Incomplete)

I don't see how a response can include a Record-Route, if it can't contain a
Contact.
I believe that 401 and 484 should be added to the list of response types for
a Contact header.

What do you think?

Cheers
Phil


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From sip-admin@lists.bell-labs.com  Thu Nov  9 04:55:02 2000
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Subject: Re: [SIP] SIP proxies and CSeq numbers (was: SIP and T.38 fax cal ls - 
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Jean-Francois Mule wrote:
> 
> If No, then No.
> For my own records, can you quote the part of the current specification(s)
> stating that proxies MUST NOT modify the Cseq?

It's a definitional thing, so to speak. If the CSeq is changed the
request doesn't belong to the same transaction and is simply a different
request. A proxy modifies requests in transit but by definition relayed
requests are equivalent to the original request in some sense, and
certainly must belong to the same transaction.

Anders

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From sip-admin@lists.bell-labs.com  Thu Nov  9 05:12:03 2000
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From: "Hisham Khartabil" <hisham.khartabil@hotsip.com>
To: "Jean-Francois Mule" <jfm@clarent.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, <archow@hss.hns.com>,
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Subject: RE: [SIP] SIP proxies and CSeq numbers (was: SIP and T.38 fax cal ls - Internet Draft)
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Section 12.1.3 of the bis states:

"The To, From, Call-ID, and Contact tags are copied exactly from the
original request. The proxy SHOULD change the Request-URI to indicate the
server where it intends to send the request."

So I guess it says nothing about the Cseq.  Frankly, I think it should state
all the headers that may not be copied and indicate that the rest MUST be
copied.

Regards,
Hisham Khartabil
Hotsip Finland


-----Original Message-----
From: sip-admin@lists.bell-labs.com [mailto:sip-admin@lists.bell-labs.com]On
Behalf Of Jean-Francois Mule
Sent: Thursday, 9 November 2000 7:01 AM
To: 'Jonathan Rosenberg'; archow@hss.hns.com; sip@lists.bell-labs.com
Subject: RE: [SIP] SIP proxies and CSeq numbers (was: SIP and T.38 fax cal
ls - Internet Draft)

If No, then No.
For my own records, can you quote the part of the current specification(s)
stating that proxies MUST NOT modify the Cseq?  I've screened rfc2543bis-02
again and I cannot find any statement that would flag the proposed call flow
as "not compliant" (the call flow in question illustrated a proxy modifying
Cseq on 2 call legs).  Any time the usage of Cseq is explained, it is in
reference to a particular call leg or SIP transaction.

It understand it is one thing to implement a profile of the spec that
follows the common sense to encourage interoperability(why would you ever
purposely increment Cseq in a proxy?), however, the spec should
unambiguously clarify this.  And this was the intent of this separate
thread.

Jean-Francois Mule
PS: I'll modify the call flow to reflect the same Cseq on both sides because
this is not the intent of the SIP-T.38 ID.  So, yes.

Jonathan Rosenberg wrote:
>
> No.
>
> The call flow is a very standard re-INVITE type of thing.
> There is no reason
> why this proxy should behave any differently from any other
> proxy. Proxies
> do not modify CSeq. Period. There is no reason why this
> cannot be a standard
> flow, and for purposes of interoperability, it should be. The
> flow you have
> in the document is not compliant to the spec.
>
> -Jonathan R.


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Suggested wording change that reflects the fact that proxies should copy
the whole request, not just the minimal header fields:

The proxy server {\MUST} copy all request header fields to the outgoing
request. It {\MAY} add other header fields.


Hisham Khartabil wrote:
> 
> Section 12.1.3 of the bis states:
> 
> "The To, From, Call-ID, and Contact tags are copied exactly from the
> original request. The proxy SHOULD change the Request-URI to indicate the
> server where it intends to send the request."
>

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Date: Thu, 09 Nov 2000 06:20:57 -0500

--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: MIME media types for ISUP and QSIG Objects
	Author(s)	: E. Zimmerer, J. Peterson, A. Vemuri, L. Ong, M. Watson, M. Zonoun
	Filename	: draft-ietf-sip-isup-mime-06.txt
	Pages		: 7
	Date		: 08-Nov-00
	
This document describes MIME types for application/ISUP and 
application/QSIG objects for use in SIP applications, according to 
the rules defined in RFC 2048 [1].  These types can be used to identify 
ISUP and QSIG objects within a SIP message such as INVITE or INFO, 
as might be implemented when using SIP between legacy systems.

A URL for this Internet-Draft is:
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	"FILE /internet-drafts/draft-ietf-sip-isup-mime-06.txt".
	
NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
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Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

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--OtherAccess
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Content-Type: text/plain
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ENCODING mime
FILE /internet-drafts/draft-ietf-sip-isup-mime-06.txt

--OtherAccess
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	name="draft-ietf-sip-isup-mime-06.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
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--OtherAccess--

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Hi,

Please note that in order to promote the development of new SIP
applications,
Startek Technologies provides a FREE SIP Testing Toolkit - The IPNess(tm)
SIP Messenger.

The SIP Messenger allows you to send, receive and monitor SIP Messages under
Windows-98 and Windows-NT.

The IPNess(tm) SIP Messenger can be downloaded from www.ipness.com .

Thanks,

Dov Ben-Asher.

Startek Technologies.



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From sip-admin@lists.bell-labs.com  Thu Nov  9 11:18:04 2000
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Hi,

My apologies for the very basic question, but I would really appreciate
if someone could drop a few lines to explain the benefit for a UA of
maintaining the same call id for all REGISTER  within a boot cycle.

Also, in the particular scenario where a UA is doing third party
registration for a few others, doesn't the call id become useless to
match requests and responses? (and the Cseq is the only way to match
them). Or should there be in this scenario a different Call ID for each
of the entities that are being registered by the third party??

thank you very much,

    Antonio.



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From sip-admin@lists.bell-labs.com  Thu Nov  9 11:26:41 2000
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Table 4 indicates that the Contact: header is mandatory "m" in the INVITE
request. Is this necessary for a re-INVITE message when the two endpoints
have already established communication.

I think this is similar to the Record-Route mechanism in which the route is
recorded once and applies to subsequent requests on the same call leg. For
that matter, if the Contact were to change. Do we change the
Record-Route/Route as well?

David

----------------------------
David Shrader
Master Consultant, Inc.
dshrader@master-consultant.com
http://www.EyeForTheFuture.com


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From sip-admin@lists.bell-labs.com  Thu Nov  9 11:59:02 2000
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Subject: Re: [SIP] Use of Record-Route and Contact Headers
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I must ask why you want to include a Contact: in either a
401 or a 484. In particular, I am very interested to know
what you would put in a Contact: that was returned in a 484?

(I agree in general that you should be able to use a Contact:
header anywhere you use a Record-Route:. I'm just curious
what applications you have in mind for these cases)

Regards
Sean Olson <sean.olson@ericsson.com>

Phil Hoffer wrote:

> Hi,
>
> This will be the third time that I have posted this message, without
> receiving a satisfactory reply.
> Please gimme some feedback!
>
> Thanks
> Phil
>
> ----------------------------------------------------------------------------
> --
>
> Hi,
>
> Table 4 of bis02 states that a Contact header can occur in the following
> response types:
>
> 1xx
> 2xx
> 3xx
> 485 (Ambiguous)
>
> However, Table 5 states that Record-Route can occur in the following
> response types:
>
> 2xx
> 401 (Unauthorized)
> 484 (Address Incomplete)
>
> I don't see how a response can include a Record-Route, if it can't contain a
> Contact.
> I believe that 401 and 484 should be added to the list of response types for
> a Contact header.
>
> What do you think?
>
> Cheers
> Phil
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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        joon_maeng@vtel.com, sip@lists.bell-labs.com, sip-mib@egroups.com
Subject: Re: [SIP] Some comments on the SIP MIB - configuration of security 
 options (draft-ietf-sip-mib-01.txt)
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this issue does look like it can be relatively complex and open to
a variety of solutions with different levels of granularity.  for the
standard sip mibs we should try and avoid complexity.  vendors should
be able to decide whether they want to express their more elaborate
auth configuration mechanisms via vendor specific mib extensions.
we should not do anything in the standard mib that would prevent, 
impede or conflict with vendors being able to do that.

i believe the sipProxyAuthMethod object's purpose to specify the 
method used and not get into the particulars of when that method
is applied.

that said, it does look like the value 'none' and it's current description 
go beyond that intended purpose. the description says 'none' means no 
authentication will be performed.  perhaps that is a point of confusion.   
maybe 'none' should  be changed to 'unknown' and the description altered 
to not imply anything about authentication being performed?  comments?

additionally, we recently got a request to alter the syntax of this
object so that multiple auth methods could be in play at once.  that
would be done by making the object's syntax BITS rather than INTEGER.
you could turn on bits for any combination of 'basic', 'digest', or
'pgp'.  the value of 0 would likely imply 'none'.   
so while we're on the subject are there any comments regarding this
proposed change (ie, to specify multiple auth methods)?   
i am currently planning on making this change, so if there are strong 
arguments against i'd like to hear them now rather than later.

kevin

"Henning G. Schulzrinne" wrote:
> 
> Jean-Francois Mule wrote:
> >
> > The current MIB allows to specify the authentication method that is used by
> > servers to authenticate request originators (sipProxyAuthMethod).   The
> > granularity is this config param means: a proxy will challenge all requests
> > or none.
> >
> > There might be cases where a proxy requires authentication on some but not
> > all requests  (typically, authenticate REGISTER only then you are trusted
> > and can speak freely ;-), or authenticate REGISTER, INVITE, BYE -- for the
> > ACK, it is okay, etc.).
> >
> > Shall we allow this level of granularity for the configuration of sip
> > proxies?
> 
> Unfortunately, it's more complicated than that. Our server allows
> configuration for each local user, so that (for example)
> help@cs.columbia.edu requests, but does not require authentication for
> non-REGISTER (but does for REGISTER), while calls to
> chair@cs.columbia.edu requires authentication for all requests. I really
> doubt you want to carry all of this in SNMP.
> 
> >
> > Jean-Francois Mule
> > Clarent Corp.
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

-- 
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
 Kevin R. Lingle       919.392.2029
 checkout: http://wwwin-eng.cisco.com/Eng/IOS/SNMP_WWW/mib-police.html
 Sometimes I think I understand everything, then I regain consciousness.
=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=

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> I must ask why you want to include a Contact: in either a
> 401 or a 484. In particular, I am very interested to know
> what you would put in a Contact: that was returned in a 484?

> (I agree in general that you should be able to use a Contact:
> header anywhere you use a Record-Route:. I'm just curious
> what applications you have in mind for these cases)
> 
> Regards
> Sean Olson <sean.olson@ericsson.com

Hi Sean,

Well if you consult Tables 5 in the bis draft you will see that it
states that it is possible to include a Record-Route in both a 401
and a 484 response.

The whole essense of Record-Route is to allow proxies to stay in
the signalling loop, which can be optimised by the use of Contacts.
So the set of responses which can contain a Record-Route, must be
a subset of the repsonses which can contain a Contact.

In both cases, I can see that the Contact would allow the caller
to recontact the callee directly, with credentials in the case of
a 401 response, or a gateway with extra digits in the case of
a 484 response. In both of these cases there is a very good chance
that the callee will retry with the extra information.

In both cases, a proxy on the original path might include a 
Record-Route in order to stay in the signalling loop.

Hence, I think that Contacts in 401 and 484 should be added
to table 5.

Just trying to clarify the spec for the community of SIP developers.
My reward might be in heaven.  :)

HTH
Cheers
Phil


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Jonathan Rosenberg writes:

 > If the calling UA is known only by this number, you are correct. The privacy
 > specification addresses exactly this requirement. The problem is when I make
 > a call from my yahoo SIP account (sip:jdrosen@yahoo.com), and the yahoo
 > proxy needs to insert my phone number into the request. What I am saying is
 > that since yahoo does not own or adminster the PSTN numbering space, it has
 > no way to verifiably correlate my yahoo address with my phone number. The
 > same problem exists in the reverse; my phone company has no easy way to
 > reliably verify that a valid additional sip address for me is
 > jdrosen@yahoo.com. 

i fully agree.  in the yahoo case, the sip/ss7 gateway could
authenticate the user and use its own pstn number as the calling number.

-- juha

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Subject: [SIP] bye and record route
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section 4.2.4 bye says:

If the INVITE request contained a Contact header, the callee SHOULD send
a BYE request to that address rather than the From address.

if the invite had record route header, shouldn't the bye request be send
to its maddr instead?

-- juha

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Date: Thu, 9 Nov 2000 13:59:24 -0500



After one last very minor tweak, I believe we are ready to have a
working-group last call on the ISUP-MIME draft, closing November 23.

The latest rev has been sent to the drafts editor and can also be downloaded
from:

http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-isup-mime-06.txt


And remember, vote early, vote often.

--
Dean Willis
SIP WG co-chair.

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Date: Thu, 9 Nov 2000 13:27:26 -0600

Which RFC defines Content-Disposition and its parameters?

-----Original Message-----
From: Dean Willis [mailto:dwillis@dynamicsoft.com]
Sent: Thursday, November 09, 2000 12:59 PM
To: IETF SIP (E-mail)
Subject: [SIP] WG Last Call on draft-ietf-isup-mime-06.txt




After one last very minor tweak, I believe we are ready to have a
working-group last call on the ISUP-MIME draft, closing November 23.

The latest rev has been sent to the drafts editor and can also be downloaded
from:

http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-isup-mime-06.txt


And remember, vote early, vote often.

--
Dean Willis
SIP WG co-chair.

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From sip-admin@lists.bell-labs.com  Thu Nov  9 14:35:02 2000
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There is a minor inconsistancy in the bis draft. Section 6.43 (To) indicates
that  "The UAS or redirect server copies the To header field into its
response, and MUST add a "tag" parameter."

However, Section 11.2 (Callee Issues Response) indicates that "The response
MUST copy the To, From, Call-ID, CSeq, and Via fields from the request.
Additionally, the responding UAS MUST add teh tag parmaeter to the To field
in the response if the request contained more than one Via header field."

I can see that the tag parameter is not really necessary if there is no
proxy, but the bis spec indicates otherwise in Section 6.43.

Which is the "correct" way to do it?

David

----------------------------
David Shrader
Master Consultant, Inc.
dshrader@master-consultant.com
http://www.EyeForTheFuture.com


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Juha Heinanen wrote:
> 
> section 4.2.4 bye says:
> 
> If the INVITE request contained a Contact header, the callee SHOULD 
> send a BYE request to that address rather than the From address.
> 
> if the invite had record route header, shouldn't the bye request be 
> send to its maddr instead?

Right.  When a UAC wants to send the request to it's next hop AND a
Route list has been constructed, it'll examine the topmost Route and
follow the rules listed in Section 1.4.2, "Locating a SIP Server" to
figure out the destination.  Step 1 in that section says, "If maddr
parameter exists, it becomes the /destination address/ used ..."

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
Internet Software & eServices Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-413
Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184

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Date: Thu, 9 Nov 2000 11:42:06 -0800

Hi,
  I did not find a "Changes since last draft.." section. Can you
please include this section or mention that no changes have been 
made.

Just a suggestion,
thanks,
aseem


-----Original Message-----
From: Dean Willis [mailto:dwillis@dynamicsoft.com]
Sent: Thursday, November 09, 2000 10:59 AM
To: IETF SIP (E-mail)
Subject: [SIP] WG Last Call on draft-ietf-isup-mime-06.txt




After one last very minor tweak, I believe we are ready to have a
working-group last call on the ISUP-MIME draft, closing November 23.

The latest rev has been sent to the drafts editor and can also be downloaded
from:

http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-isup-mime-06.txt


And remember, vote early, vote often.

--
Dean Willis
SIP WG co-chair.

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From sip-admin@lists.bell-labs.com  Thu Nov  9 15:54:03 2000
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Date: Thu, 09 Nov 2000 12:53:57 -0800

Hi,

The only change since draft 5 was to change the word "session" to "signal" 
in the "typical header" shown at the end of
section 3.1 on page 3.  This is an editorial correction, as draft 5 
introduced the disposition type "signal", but I neglected
to update the "typical header" accordingly.

Cheers,

L. Ong

  At 11:42 AM 11/9/2000 -0800, Agarwal, Aseem wrote:

>Hi,
>   I did not find a "Changes since last draft.." section. Can you
>please include this section or mention that no changes have been
>made.
>
>Just a suggestion,
>thanks,
>aseem
>
>-----Original Message-----
>From: Dean Willis 
>[<mailto:dwillis@dynamicsoft.com>mailto:dwillis@dynamicsoft.com]
>Sent: Thursday, November 09, 2000 10:59 AM
>To: IETF SIP (E-mail)
>Subject: [SIP] WG Last Call on draft-ietf-isup-mime-06.txt
>
>
>
>After one last very minor tweak, I believe we are ready to have a
>working-group last call on the ISUP-MIME draft, closing November 23.
>
>The latest rev has been sent to the drafts editor and can also be downloaded
>from:
>
><http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-isup-mime-06.txt>http 
>://www.softarmor.com/sipwg/drafts/draft-ietf-sip-isup-mime-06.txt
>
>And remember, vote early, vote often.
>
>--
>Dean Willis
>SIP WG co-chair.
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
><http://lists.bell-labs.com/mailman/listinfo/sip>http://lists.bell-labs.com 
>/mailman/listinfo/sip
>
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From sip-admin@lists.bell-labs.com  Thu Nov  9 17:23:01 2000
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From: David Shrader <dshrader@master-consultant.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: [SIP] Status of "serverfeatures" draft
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Date: Thu, 09 Nov 2000 17:14:10 -0500
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Jonathan,

What is the status of <draft-ietf-sip-serverfeatures-02.txt> which expired
September? I can see that most of its capabilities are in the bis draft.
However, I don't see anything related to the 421 code proposed in the draft.

That is, if a server requires a particular extension (such as 100rel) but
the client didn't include 100rel in a Supported header, how should it reject
the request?

In this particular example, the 100rel draft <draft-ietf-sip-100rel-02.txt>
indicates "If the request did not include either a Supported or Require
header indicating this feature, and the UAS wishes to send some provisional
responses reliably, the UAS \SHOULD reject the initial request and include a
Require header in the response, as per [4]" [4] is the "serverfeatures"
draft.

What code is used for this reject?

David
----------------------------
David Shrader
Master Consultant, Inc.
dshrader@master-consultant.com
http://www.EyeForTheFuture.com


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From sip-admin@lists.bell-labs.com  Thu Nov  9 17:26:14 2000
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Date: Thu, 9 Nov 2000 17:23:14 -0500 (EST)

Use of an added parameter on Record-Route was considered early
as a way to do the equivalent to State headers (like about 3-4
IETF meetings ago) and was rejected.

From our collective memory, there were several problems with 
overloading Route.

Record-Route headers actually FORCE the subsequent messages to
go to the proxy mentioned.  We wanted something that was disjoint
from routing, that would be usable if the message arrived there,
but didn't force all the future messages to be routed thee.

We also wanted the ability to insert multiple State headers in
a message, and to attach State headers to requests for transfers
(like when Refer-To is referring to an existing call that is to
be replaced).

There was also a good deal of confusion with using Route, that the
Route header for a proxy was removed before the request was sent
to it.  Therefore the state information had to be off by 
one entry in the Record-Route list, and that it was different
in the forward and reverse directions.

Bill Marshall (on behalf of all the DCS team who contributed to this response)
wtm@research.att.com

-----original message-----
From: "Anuraj Kunnummel Ennai" <anuraj.ennai@wipro.com>
To: sip@lists.bell-labs.com
Subject: Re:[SIP] I-D ACTION:draft-ietf-sip-state-00.txt
Date: Thu, 09 Nov 2000 12:21:55 +0530

Hi ,
Went through the distributable call state draft. It is a move in the right
direction as the proxies should be holding as little state information as
possible in order to scale up.

But it seems the idea of having a separate state header might not be
needed. The nesting of  state headers across proxies adds to
complexity and overhead. Existing UA implementations cannot
support state header.

An alternative will be to use the Record-route option already defined.
The proxy which adds a  record route header(means the proxy is interested
in keeping track of the call.)can add the state info duly encrypted as a
parameter to record route header.

Now only the proxy which inserted the record route header needs to interpret
the state parameters. Note that record route is valid per call leg basis. Thus the
proxy can selectively choose call legs which it is interested in. .The UA
implementations need not worry as they are supposed to insert the Record-route
header unchanged into the responses.

As record - routing support is mandatory, this seems to be a better solution that
needs very little changes. Also alleviates the need to live with one more header!!!

This is what I feel from the first look...Or am I missing something?


Thanks & regards
Anuraj Ennai
SIP Designer
Wipro Technologies -Bangalore-India


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From sip-admin@lists.bell-labs.com  Thu Nov  9 17:34:43 2000
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Date: Thu, 9 Nov 2000 16:10:58 -0600

It might be a basic question, sorry for that..

My understanding is that the "VIA" field serves two purposes:
1)Loop detection
2)Keeping proxies (which are part of the message forwarding path) in the response path.

My question is: Does that mean also stateless proxies will be in the response path? 
If YES, perhaps it worths to enable a proxy to add a Boolean tag, which says: "Hey I am adding myself to the VIA list just for the sake of loop prevention, but count me out for
the response thing"

Uri




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From sip-admin@lists.bell-labs.com  Thu Nov  9 19:05:05 2000
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Date: Thu, 09 Nov 2000 19:04:33 -0500
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David Shrader wrote:
> 
> There is a minor inconsistancy in the bis draft. Section 6.43 (To) indicates
> that  "The UAS or redirect server copies the To header field into its
> response, and MUST add a "tag" parameter."
> 
> However, Section 11.2 (Callee Issues Response) indicates that "The response
> MUST copy the To, From, Call-ID, CSeq, and Via fields from the request.
> Additionally, the responding UAS MUST add teh tag parmaeter to the To field
> in the response if the request contained more than one Via header field."
> 
> I can see that the tag parameter is not really necessary if there is no
> proxy, but the bis spec indicates otherwise in Section 6.43.
> 
> Which is the "correct" way to do it?

I've amended the text to remove the qualification. The more special
cases we can get rid of, the better.

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Subject: RE: [SIP] Caller Preferences and MESSAGE method
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Date: Fri, 10 Nov 2000 01:18:33 -0500

Caller preferences processing is really method independent. It would also be
useful for SUBSCRIBE. I'll be sure to update the doc to make this clearer.

-Jonathan R.


Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
  
-----Original Message-----
From: Sriram Parameswar [mailto:sriramp@nortelnetworks.com]
Sent: Wednesday, November 08, 2000 8:51 PM
To: 'sip@lists.bell-labs.com'
Cc: Jonathan Rosenberg (E-mail)
Subject: [SIP] Caller Preferences and MESSAGE method


Hi, 
I think there is definite value in adding the MESSAGE method to the
draft-ietf-sip-callerprefs-02. I see *all three* headers being of value to
the MESSAGE method i.e. Accept-Contact, Reject-Contact and
Request-Disposition.
Some examples: 
1. The sender of the IM could require that his Message be delivered only to
a device with a class = "business". 
   This could use the Accept-Contact header. 
2. If you want to save your buddy valuable air-time charges you could send
an IM with the following header 
   Reject-Contact: sip:mybuddy@downtown.com;mobility="mobile" 
3. You could use the Request-Disposition to ensure that the Contact list
returned for the Message is tried sequentially.
Thanks and Regards, 
Sriram Parameswar 
Nortel Networks 

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Date: Fri, 10 Nov 2000 01:28:20 -0500



 

> -----Original Message-----
> From: Baniel Uri-CUB001 [mailto:Uri.Baniel@motorola.com]
> Sent: Thursday, November 09, 2000 5:11 PM
> To: 'sip@lists.bell-labs.com'
> Subject: [SIP] VIA field
> 
> 
> It might be a basic question, sorry for that..
> 
> My understanding is that the "VIA" field serves two purposes:
> 1)Loop detection
> 2)Keeping proxies (which are part of the message forwarding 
> path) in the response path.
> 
> My question is: Does that mean also stateless proxies will be 
> in the response path? 

Yes.

> If YES, perhaps it worths to enable a proxy to add a Boolean 
> tag, which says: "Hey I am adding myself to the VIA list just 
> for the sake of loop prevention, but count me out for
> the response thing"

No.

First and foremost, its not backwards compatible. Remember, SIP is already
an RFC. It is not open season for "change SIP in all kinds of new ways". We
are working to (1) fix bugs in the  spec, (2) clarify things that are vague,
(3) describe cases not already covered in the spec. If something is clearly
defined and working correctly, I do not believe modifying it can be
justified.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Date: Fri, 10 Nov 2000 10:03:50 -0600
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I believe that's RFC1806.

--
Dean

Kevin Summers wrote:

> Which RFC defines Content-Disposition and its parameters?
>
> -----Original Message-----
> From: Dean Willis [mailto:dwillis@dynamicsoft.com]
> Sent: Thursday, November 09, 2000 12:59 PM
> To: IETF SIP (E-mail)
> Subject: [SIP] WG Last Call on draft-ietf-isup-mime-06.txt
>
> After one last very minor tweak, I believe we are ready to have a
> working-group last call on the ISUP-MIME draft, closing November 23.
>
> The latest rev has been sent to the drafts editor and can also be downloaded
> from:
>
> http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-isup-mime-06.txt
>
> And remember, vote early, vote often.
>
> --
> Dean Willis
> SIP WG co-chair.
>
> _______________________________________________
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>
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Date: Fri, 10 Nov 2000 10:24:16 -0600 (CST)
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I now have an I-D out telling how to describe an appropriate
ringback tone to be played to a caller for a SIP-initiated
voice call; it does so by including new headers in the
"180 Ringing" message. I currently plan to progress this as a 
non-workgroup informational draft, but am open to other
suggestions. Let me know if this topic is of interest to you. 
Comments solicited.

Abstract

     This document describes a mechanism by which an appropriate
     ringback tone may be played to the calling party when the called
     party's device is alerting. It is written specifcally to address
     the case where the Session Initiation Protocol (SIP) is used to
     initiate voice-over-IP calls. It also lists ringback
     characteristics for several countries.

http://search.ietf.org/internet-drafts/draft-roach-voip-ringtone-00.txt

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA

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Date: Fri, 10 Nov 2000 08:32:17 -0800

The bis draft references RFC1806 for the definition of the 
Content-Disposition header.

Cheers,

L. Ong

At 11:27 AM 11/9/2000 -0800, Kevin Summers wrote:

>Which RFC defines Content-Disposition and its parameters?
>
>-----Original Message-----
>From: Dean Willis 
>[<mailto:dwillis@dynamicsoft.com>mailto:dwillis@dynamicsoft.com]
>Sent: Thursday, November 09, 2000 12:59 PM
>To: IETF SIP (E-mail)
>Subject: [SIP] WG Last Call on draft-ietf-isup-mime-06.txt
>
>
>
>After one last very minor tweak, I believe we are ready to have a
>working-group last call on the ISUP-MIME draft, closing November 23.
>
>The latest rev has been sent to the drafts editor and can also be downloaded
>from:
>
><http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-isup-mime-06.txt>http 
>://www.softarmor.com/sipwg/drafts/draft-ietf-sip-isup-mime-06.txt
>
>And remember, vote early, vote often.
>
>--
>Dean Willis
>SIP WG co-chair.
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
><http://lists.bell-labs.com/mailman/listinfo/sip>http://lists.bell-labs.com 
>/mailman/listinfo/sip
>
>**********************************************************************
>This email and any files transmitted with it are confidential and
>intended solely for the use of the individual or entity to whom they
>are addressed. If you have received this email in error please notify
>the system manager.
>
>This footnote also confirms that this email message has been swept by
>MIMEsweeper for the presence of computer viruses.
>
>www.inip.com
>**********************************************************************
>
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>/mailman/listinfo/sip


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Date: Fri, 10 Nov 2000 10:44:25 -0500

It's in the base SIP spec.

> -----Original Message-----
> From: Kevin Summers [mailto:Kevin.Summers@ttimail.com]
> Sent: Thursday, November 09, 2000 2:27 PM
> To: IETF SIP (E-mail)
> Subject: RE: [SIP] WG Last Call on draft-ietf-isup-mime-06.txt
> 
> 
> Which RFC defines Content-Disposition and its parameters?
> 
> -----Original Message-----
> From: Dean Willis [mailto:dwillis@dynamicsoft.com]
> Sent: Thursday, November 09, 2000 12:59 PM
> To: IETF SIP (E-mail)
> Subject: [SIP] WG Last Call on draft-ietf-isup-mime-06.txt
> 
> 
> 
> 
> After one last very minor tweak, I believe we are ready to have a
> working-group last call on the ISUP-MIME draft, closing November 23.
> 
> The latest rev has been sent to the drafts editor and can 
> also be downloaded
> from:
> 
> http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-isup-mime-06.txt
> 
> 
> And remember, vote early, vote often.
> 
> --
> Dean Willis
> SIP WG co-chair.
> 
> _______________________________________________
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> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 
> 
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> 
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> 
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From: Bobby Sardana <bobby.sardana@mobilerain.com>
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Subject: Re: [SIP] WG Last Call on draft-ietf-isup-mime-06.txt
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Date: Fri, 10 Nov 2000 10:00:28 -0800

This is a multi-part message in MIME format.
--------------FC61928E5FE1992B8712FC98
Content-Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit

Greetings:

Communicating Presentation Information in Internet Messages:
       The Content-Disposition Header Field (RFC 2183)

Regards,

Bobby.Sardana@mobilerain.com

Kevin Summers wrote:

> Which RFC defines Content-Disposition and its parameters?
>
> -----Original Message-----
> From: Dean Willis [mailto:dwillis@dynamicsoft.com]
> Sent: Thursday, November 09, 2000 12:59 PM
> To: IETF SIP (E-mail)
> Subject: [SIP] WG Last Call on draft-ietf-isup-mime-06.txt
>
> After one last very minor tweak, I believe we are ready to have a
> working-group last call on the ISUP-MIME draft, closing November 23.
>
> The latest rev has been sent to the drafts editor and can also be downloaded
> from:
>
> http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-isup-mime-06.txt
>
> And remember, vote early, vote often.
>
> --
> Dean Willis
> SIP WG co-chair.
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>
> **********************************************************************
> This email and any files transmitted with it are confidential and
> intended solely for the use of the individual or entity to whom they
> are addressed. If you have received this email in error please notify
> the system manager.
>
> This footnote also confirms that this email message has been swept by
> MIMEsweeper for the presence of computer viruses.
>
> www.inip.com
> **********************************************************************
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

--------------FC61928E5FE1992B8712FC98
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Content-Disposition: attachment;
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begin:vcard 
n:Sardana;Bobby
x-mozilla-html:FALSE
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adr:;;;;;;
version:2.1
email;internet:bobby.sardana@mobilerain.com
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end:vcard

--------------FC61928E5FE1992B8712FC98--


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From sip-admin@lists.bell-labs.com  Fri Nov 10 14:44:22 2000
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From: Baniel Uri-CUB001 <Uri.Baniel@motorola.com>
To: "'sip@lists.bell-labs.com'" <sip@lists.bell-labs.com>
Subject: FW: [SIP] VIA field
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Date: Fri, 10 Nov 2000 13:43:28 -0600

Jonathan

No problem at all. It was just an idea to have in the back of the mind for future studies/work.
I know how important it is to retain backward compatibility and to handle the things that you have mentioned first,
only then to think of ideas like that good or bad as they may be.

Uri

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] 
Sent: Friday, November 10, 2000 12:28 AM
To: 'Baniel Uri-CUB001'; 'sip@lists.bell-labs.com'
Subject: RE: [SIP] VIA field




 

> -----Original Message-----
> From: Baniel Uri-CUB001 [mailto:Uri.Baniel@motorola.com]
> Sent: Thursday, November 09, 2000 5:11 PM
> To: 'sip@lists.bell-labs.com'
> Subject: [SIP] VIA field
> 
> 
> It might be a basic question, sorry for that..
> 
> My understanding is that the "VIA" field serves two purposes:
> 1)Loop detection
> 2)Keeping proxies (which are part of the message forwarding 
> path) in the response path.
> 
> My question is: Does that mean also stateless proxies will be 
> in the response path? 

Yes.

> If YES, perhaps it worths to enable a proxy to add a Boolean 
> tag, which says: "Hey I am adding myself to the VIA list just 
> for the sake of loop prevention, but count me out for
> the response thing"

No.

First and foremost, its not backwards compatible. Remember, SIP is already
an RFC. It is not open season for "change SIP in all kinds of new ways". We
are working to (1) fix bugs in the  spec, (2) clarify things that are vague,
(3) describe cases not already covered in the spec. If something is clearly
defined and working correctly, I do not believe modifying it can be
justified.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From sip-admin@lists.bell-labs.com  Fri Nov 10 16:54:03 2000
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From: "Culpepper, Bert" <bert.culpepper@intervoice-brite.com>
To: "Adam B. Roach" <Adam.Roach@Ericsson.com>, sip@lists.bell-labs.com
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Date: Fri, 10 Nov 2000 15:48:19 -0600

I think the ability to specify the ringback tone characteristics is useful
and expect some to find it desirable.  The (somewhat subtle) aspect of the
draft that I feel is also useful is that the device on the edge of the SIP
network (towards the caller) is responsible for providing ringback tone.  In
the PSTN, the switch at the edge of the network (towards the callee)
provides ringback.  In a PSTN-IP network where the call terminates in the IP
network, the device responsible for providing ringback is the ingress
gateway to the IP network IMO.  I've found that which device provides
ringback is unclear to some when discussing PSTN-IP interworking.

I realize the aspect I described is not the focus of your draft but the
topic does interest me :)

Regards,
Bert

-----Original Message-----
From: Adam B. Roach [mailto:Adam.Roach@Ericsson.com]
Sent: Friday, November 10, 2000 11:24 AM
To: sip@lists.bell-labs.com
Subject: [SIP] Ringback draft


I now have an I-D out telling how to describe an appropriate
ringback tone to be played to a caller for a SIP-initiated
voice call; it does so by including new headers in the
"180 Ringing" message. I currently plan to progress this as a 
non-workgroup informational draft, but am open to other
suggestions. Let me know if this topic is of interest to you. 
Comments solicited.

Abstract

     This document describes a mechanism by which an appropriate
     ringback tone may be played to the calling party when the called
     party's device is alerting. It is written specifcally to address
     the case where the Session Initiation Protocol (SIP) is used to
     initiate voice-over-IP calls. It also lists ringback
     characteristics for several countries.

http://search.ietf.org/internet-drafts/draft-roach-voip-ringtone-00.txt

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS
L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX
75081 USA

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From sip-admin@lists.bell-labs.com  Fri Nov 10 17:08:21 2000
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Date: Fri, 10 Nov 2000 10:02:29 -0500


I am tracking slot requests for SIPWG at:

http://www.softarmor.com/sipwg/meets/IETF49/requests.htm

If you make a request and don't see a timely response, please ping me.

BTW, I'll be hiding from the world during 17Nov-27Nov. At least I'll be
operating with reduced bandwidth during that interval.

--
Dean Willis
SIP WG cochair

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Date: Fri, 10 Nov 2000 11:00:52 -0500


My apologies. I believe only one word was changed between 5 and 6 -- changed
"session" to "signal" on page 3. But you are right, it should be mentioned
in "Changes".

--
Dean

-----Original Message-----
From: Agarwal, Aseem [mailto:aseem@trillium.com]
Sent: Thursday, November 09, 2000 1:42 PM
To: 'Dean Willis'; IETF SIP (E-mail)
Subject: RE: [SIP] WG Last Call on draft-ietf-isup-mime-06.txt


Hi,
  I did not find a "Changes since last draft.." section. Can you
please include this section or mention that no changes have been 
made.

Just a suggestion,
thanks,
aseem


-----Original Message-----
From: Dean Willis [mailto:dwillis@dynamicsoft.com]
Sent: Thursday, November 09, 2000 10:59 AM
To: IETF SIP (E-mail)
Subject: [SIP] WG Last Call on draft-ietf-isup-mime-06.txt




After one last very minor tweak, I believe we are ready to have a
working-group last call on the ISUP-MIME draft, closing November 23.

The latest rev has been sent to the drafts editor and can also be downloaded
from:

http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-isup-mime-06.txt


And remember, vote early, vote often.

--
Dean Willis
SIP WG co-chair.

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To: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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Subject: Re: [SIP] Ringback draft
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I would suggest making the location information somewhat more generic,
as this is useful for other applications, as pointed out in

http://www.ietf.org/internet-drafts/draft-schulzrinne-sip-911-01.txt

Also, there are related efforts in other areas that address the location
problem. It would seem appropriate to take these into consideration, in
the "tradition" of trying to leverage HTTP/web-related work going on, as
well as avoiding too many telephony-only extensions.

http://www.ietf.org/internet-drafts/draft-daviel-http-geo-header-02.txt

http://www.ietf.org/internet-drafts/draft-costanzo-dns-gl-03.txt

(The latter is obviously DNS, but the labeling may be applicable, as
shown in the 911 draft.)

Users can obviously omit details they'd rather not specify or don't
know. 

Also, there's an obvious relationship to RFC 2833 (RTP "tones") that
should be explored. Note also that you could use the Alert-Info for the
same purpose, by providing an external (cached) reference to whatever
national tones you'd like the caller to hear.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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From sip-admin@lists.bell-labs.com  Fri Nov 10 20:06:03 2000
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From: Jean-Francois Mule <jfmule@clarent.com>
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Date: Fri, 10 Nov 2000 17:02:25 -0800

OK - I agree it can get complex and it is certainly preferable to avoid this
in the MIB.  

The object sipProxyAuthMethod then refers to the capabilities:
- the ability of the SIP entity to perform Authentication, and, 
- in cases when authentication is performed, it specifies the method
(capabilities such as the supported algorithm(s) are in some subsequent
objects).

See comments below.
Jean-Francois Mule.
-------------------------
Kevin Lingle wrote:
> that said, it does look like the value 'none' and it's 
> current description 
> go beyond that intended purpose. the description says 'none' means no 
> authentication will be performed.  perhaps that is a point of 
> confusion.   
> maybe 'none' should  be changed to 'unknown' and the 
> description altered 
> to not imply anything about authentication being performed?  comments?

If we agree with the definition given above, I'd rather keep none with this
description:
   "This object specifies the authentication method that the SIP entity
supports 
    to authenticate request originators.
    none(1)            : no authentication is supported.
    basic(2)           : the basic authentication mechanism is supported.
    digest(3)          : the digest authentication mechanism is supported." 

> additionally, we recently got a request to alter the syntax of this
> object so that multiple auth methods could be in play at once.  that
> would be done by making the object's syntax BITS rather than INTEGER.
Yes, and this request confirms the intent of the var:  capabilities for the
supported authentication mechanims.

> you could turn on bits for any combination of 'basic', 'digest', or
> 'pgp'.  the value of 0 would likely imply 'none'.   
> so while we're on the subject are there any comments regarding this
> proposed change (ie, to specify multiple auth methods)?   
> i am currently planning on making this change, so if there are strong 
> arguments against i'd like to hear them now rather than later.
YES.
> 
> kevin

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From sip-admin@lists.bell-labs.com  Fri Nov 10 23:49:01 2000
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Subject: Re: [SIP] Ringback draft
From: David Shrader <dshrader@master-consultant.com>
To: SIP List <sip@lists.bell-labs.com>
Cc: "Adam B. Roach" <Adam.Roach@ericsson.com>
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Date: Fri, 10 Nov 2000 23:39:09 -0500
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Henning has a good point about leveraging the location work going on
elsewhere in the IETF. One question I then have is should the ringback tone
be based on my current location or rather on my home location so that the
caller hears what they expect to hear.

On the other hand, I always found the different tones in different countries
confusing. One advantage to not specifying them is that the originator has
his system customized for his location and he hears a consistent set of
tones regardless of where the callee is located.

I actually don't really like the idea of the originating side reconstructing
the ring pattern for the terminating side in such a specific manner. The use
of the Alert-Info: header provides an ideal solution to the problem.

If we are going to all this trouble for the callee to tell the calling party
what to hear, maybe its better to send a 183 and include the tones via RTP.
Maybe this is something the PSTN did right :-)

----------------------------
David Shrader
Master Consultant, Inc.
"Solution Oriented Engineering"
dshrader@master-consultant.com
http://www.EyeForTheFuture.com

> From: "Henning G. Schulzrinne" <hgs@cs.columbia.edu>
> Organization: Dept. of Computer Science, Columbia University
> Date: Fri, 10 Nov 2000 17:50:09 -0500
> To: "Adam B. Roach" <Adam.Roach@Ericsson.com>
> Cc: sip@lists.bell-labs.com
> Subject: Re: [SIP] Ringback draft
> 
> I would suggest making the location information somewhat more generic,
> as this is useful for other applications, as pointed out in
> 
> http://www.ietf.org/internet-drafts/draft-schulzrinne-sip-911-01.txt
> 
> Also, there are related efforts in other areas that address the location
> problem. It would seem appropriate to take these into consideration, in
> the "tradition" of trying to leverage HTTP/web-related work going on, as
> well as avoiding too many telephony-only extensions.
> 
> http://www.ietf.org/internet-drafts/draft-daviel-http-geo-header-02.txt
> 
> http://www.ietf.org/internet-drafts/draft-costanzo-dns-gl-03.txt
> 
> (The latter is obviously DNS, but the labeling may be applicable, as
> shown in the 911 draft.)
> 
> Users can obviously omit details they'd rather not specify or don't
> know. 
> 
> Also, there's an obvious relationship to RFC 2833 (RTP "tones") that
> should be explored. Note also that you could use the Alert-Info for the
> same purpose, by providing an external (cached) reference to whatever
> national tones you'd like the caller to hear.
> -- 
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 


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From sip-admin@lists.bell-labs.com  Sat Nov 11 01:00:03 2000
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From: "Cullen Jennings" <cjennings@vovida.com>
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One of the nice parts of sending the Ringback tone in the media channel is
that the media channel is established (at least in one direction) before the
callee picks up. This ensure that the first "Hello" of the person answering
the phone is not cut off. This early set up of RTP is probably even more
important when you consider that perhaps QoS and/or encryption are also
being set up for the RTP connection.

If the issue with ring back via RTP is that it wastes bandwidth, I imagine
some special "Ringback tone" style RTP packets could be defined.

Cullen Jennings


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Cullen Jennings wrote:
> 
> One of the nice parts of sending the Ringback tone in the media channel is
> that the media channel is established (at least in one direction) before the
> callee picks up. This ensure that the first "Hello" of the person answering
> the phone is not cut off. This early set up of RTP is probably even more
> important when you consider that perhaps QoS and/or encryption are also
> being set up for the RTP connection.
> 
> If the issue with ring back via RTP is that it wastes bandwidth, I imagine
> some special "Ringback tone" style RTP packets could be defined.

Well, RFC 2833 already allows to define any tone you want. You can even
do harmonies. All we need is a MIDI-to-RTP translator...

In general, I have to admit that I find the notion of having to
replicate the called country's tones a bit too much of the "it might be
a strange artifact of the phone system, but we have to replicate every
bit of it" syndrom. I can see the advantage that if I'm traveling, to
hear familiar tone sequences regardless of my location. However, that
should be done by configuring either my portable device or, temporarily,
the Internet pay phone to play familiar sounds. I can't say that I've
been confused or comforted by "foreign" ringback; the problem tends to
be more trying to understand the Swedish version of "this number has
been disconnected". More confusing than ringback are usually the various
busy and fast-busy tones.

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First off, I have to admit that I didn't explored RFC 2833.
But looking thru the "Ringback draft" thread of the SIP list, I thought
about payments, if we consider network with payment policy.
So my appologies for, perhaps, improper question: Who will be paying for 
transmission of RTP ringback tones if transaction is not succesfuly 
completed (e.g. rejected by callee). As i suppose, ringback RTP/UDP 
requires much more bandwith than SIP messages/UDP, and should not be
passed over ?...

Piotr


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References: <IPENKGOOAAAEMJNBAMNFMEIODAAA.cjennings@vovida.com> <3A0D5BC9.BA52328E@cs.columbia.edu> <3A0D765F.5006DA90@elka.pw.edu.pl>
Subject: Re: [SIP] Ringback draft
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SIP calls, where the endpoint is IP based will probably not be charged for
(the phone companies will slowly wake up to this fact), unless the call
involves a gateway to the PSTN, special service, or some QoS mechanism.

Simon Barber


----- Original Message -----
From: ""Piotr S. Kossowski"" <P.Kossowski@elka.pw.edu.pl>
Newsgroups: ietf.sip-wg
Sent: Saturday, November 11, 2000 8:39 AM
Subject: Re: [SIP] Ringback draft


>
> First off, I have to admit that I didn't explored RFC 2833.
> But looking thru the "Ringback draft" thread of the SIP list, I thought
> about payments, if we consider network with payment policy.
> So my appologies for, perhaps, improper question: Who will be paying for
> transmission of RTP ringback tones if transaction is not succesfuly
> completed (e.g. rejected by callee). As i suppose, ringback RTP/UDP
> requires much more bandwith than SIP messages/UDP, and should not be
> passed over ?...
>
> Piotr
>
>
> _______________________________________________
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> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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From sip-admin@lists.bell-labs.com  Sat Nov 11 18:38:00 2000
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From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [SIP] Ringback draft
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Payment policy that is user friendly would prevent charging for tones. IMHO,
charging for very short intervalls of time, such as the beginning of a
conversation, say for less than 30 seconds is more expensive to the service
provider than not charging at all for such short transients. Even if a
gateway is involved, charging for 10s of seconds does not seem to make
sense, given present ever dropping LD prices.

Henry

-----Original Message-----
From: sip-admin@lists.bell-labs.com
[mailto:sip-admin@lists.bell-labs.com]On Behalf Of Piotr S. Kossowski
Sent: Saturday, November 11, 2000 10:40 AM
To: SIP List
Subject: Re: [SIP] Ringback draft



First off, I have to admit that I didn't explored RFC 2833.
But looking thru the "Ringback draft" thread of the SIP list, I thought
about payments, if we consider network with payment policy.
So my appologies for, perhaps, improper question: Who will be paying for
transmission of RTP ringback tones if transaction is not succesfuly
completed (e.g. rejected by callee). As i suppose, ringback RTP/UDP
requires much more bandwith than SIP messages/UDP, and should not be
passed over ?...

Piotr


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From sip-admin@lists.bell-labs.com  Sun Nov 12 17:18:04 2000
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From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
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The draft on "Ringback tones in SIP-Based Telephony" shows impressive work
on preserving legacy country specific ring back tones. I wonder if we would
not be better of by adding a

SIP Ringback Tone
=================

With some optional fields where service providers could add innovative
greetings.

Henry

Henry Sinnreich
WorldCom
400 International Parkway,
Richardson, Texas, 75081



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Date: Sun, 12 Nov 2000 22:49:53 -0800 (PST)

Hi, Can my UA be registered with multiple Aliases?
(analogous to having a single mailbox for different
email addresses)
Zeeshan.


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Jo Hornsby wrote:
> 
> 
> If, OTOH, we are looking at backwards-compatibility from a "work
> with existing SIP Implementations" standpoint, then I have to say
> that so much of Record-Route wouldn't have worked anyway (the
> spec says nothing about building Route:s in the callee -> caller
> direction), is it really worth worrying?
> 
> With this approach, we can write into bis, say, that the called
> UA, when forming requests, renames all "uri" parameters to
> "original-uri"; and appends a new "uri" parameter with the value
> of the Contact:/From:.
> 
> We now have the resilience that Jonathan was addressing (thread:
> "failure of proxy), since each proxy along the Route can see
> the final destination.  And also, if the request spiralled,
> we know where on the Route we are (since we have retained the
> original Request-URIs, too).
> 
> Now I know that this is a significant change, but as far as I can
> tell, realistically, no proxy will as yet have been able to
> Record-Route properly and/or remain call stateful, because we
> are still finding bugs.  Thus it doesn't seem as if there can be
> real objections to making such a changes, since hopefully they
> are changes towards having a fully-working mechanism (okay, I
> guess we think it works now, but it can't be more than 3 months
> since we spotted that Record-Route: can break in the reverse
> direction when a request has spiralled; how mature can
> implementations possibly be?)
> 

I thought I were the only one having these thoughts. This is
the reason why I started to suggest simplifications to SIP some
months ago. It was quite obvious that the protocol was not
complete at that time (it's not even complete today). So
to argue that a specification that is not complete and still
contain major bugs have to be backwards compatible is very 
strange to me. But I guess I have to live with that ??

The kind of solution I would like to have to this problem is
a change of the ROUTE behaviour so it behaves like VIA i.e.
you remove your "own" entry from the list instead of removing
the next servers entry. In this way it would be possible to
add any RECORD-ROUTE parameter and get it back later in ROUTE 
(if the client copies the whole RECORD-ROUTE into ROUTE). This
would make the state draft obsolete and it would also solve
my problem with identifying the call state machine. Since
this change is a pure semantic change I see no reason why
it can't be done like this ;-). 

The only syntax change needed is to add a 'hostport' 
parameter to RECORD-ROUTE instead of maddr as Jonathan
pointed out. Since this is needed for TEL-URI etc. this
parameter has to be added but I would prefere to ALWAYS 
use this new parameter instead of sometimes using maddr.
 
/Bertil

-- 
Bertil Engelholm
AXE Research and Development        voice : +46 8 727 3499
SIP Security                        Fax   : +46 8 647 8276
S-126 25 Stockholm Sweden           E-mail:
Bertil.Engelholm@uab.ericsson.se

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Date: Mon, 13 Nov 2000 11:16:36 -0000
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> Hi, Can my UA be registered with multiple Aliases?
> (analogous to having a single mailbox for different
> email addresses)

Very much so (the only thing preventing this would be
a lack of Service Providers/Registrars willing to let
you do this).


 - Jo.


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From sip-admin@lists.bell-labs.com  Mon Nov 13 06:33:02 2000
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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: "'Henry Sinnreich'" <Henry.Sinnreich@wcom.com>
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Subject: RE: [SIP] SIP ringback tone
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Date: Mon, 13 Nov 2000 03:33:35 -0800

Henry,

Innovative ringback greetings definitely seem like a case for an early media
to me.

As an aside, many people still seem to erroroneously think that 183 Session
Progress should be used for early-media ring-back. 180 Ringing with SDP
should be used for ringback purposes, 183 with SDP should only be used for
call progress messages (ie, not alerting the user). I still think this needs
to be made clearer in the spec!

Cheers,

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 

> -----Original Message-----
> From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
> Sent: Monday, November 13, 2000 6:18 AM
> To: adam.roach@ericsson.com; IETF SIP (E-mail)
> Subject: [SIP] SIP ringback tone
> 
> 
> The draft on "Ringback tones in SIP-Based Telephony" shows 
> impressive work
> on preserving legacy country specific ring back tones. I 
> wonder if we would
> not be better of by adding a
> 
> SIP Ringback Tone
> =================
> 
> With some optional fields where service providers could add innovative
> greetings.
> 
> Henry
> 
> Henry Sinnreich
> WorldCom
> 400 International Parkway,
> Richardson, Texas, 75081
> 
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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From sip-admin@lists.bell-labs.com  Mon Nov 13 06:54:03 2000
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To: "Manish S. Jalan" <jalanmanish@rediffmail.com>
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Subject: [SIP] SIP provisioning
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Date: Mon, 13 Nov 2000 11:51:24 +0000
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Manish,

Thank you for your query. It is a very good question indeed. The
rational behind only having a get method for ListeningPoints is that the
scope JAIN SIP API does not include provisioning - the API assumes the
SIP stack has been set up previously by some other means. Currently this
implies the use of proprietary methods for setting/adding
ListeningPoints.

However there is a JMX-based JAIN OAM API (which currently handles SS7
stacks). The plan is to extend or provide a similar OAM API for the JAIN
IP protocols. Admittedly OAM is a lot less complicated for SIP stacks
than SS7 stacks, and there is a very strong case for just including the
addListeningPoint method to the JainSipStack interface, but this was
deemed inconsistent with other JAIN protocol API's.

Note, personally I would prefer leaving the OAM API in the SS7 domain,
and adding the necessary methods (only one in SIP!) to the Stack objects
in IP protocol APIs.

Regards,
Chris Harris



Subject: Seeking clarification on JainSipStack Interface
Date: 13 Nov 2000 04:00:04 -0000
From: "Manish S. Jalan" <jalanmanish@rediffmail.com>
To: "JAIN-SIP-interest@sun.com" <JAIN-SIP-interest@sun.com>

Interface JainSipStack defines methods to create providers, get
an array of Listening  points available to the stack, etc.
The default listening point may be a combination localhost and
standard sip port 5060. Now one may want to have additional
listening points(on some other ports). Hence not only should the
JainSipStack define a get method for the ListeningPoints list,
but should also have one to add to the list of ListeningPoints.
I may be wrong somewhere, as definitely there must have been some
rational behind, only defining a get method, and not a add
method. Kindly, clarify the rational. But if I am correct, then
please include addListeningPoint() method in the JainSipStack
interface.

Regards,
-Manish S. Jalan



_____________________________________________________
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Bertil Engelholm wrote:
>
> > If, OTOH, we are looking at backwards-compatibility from a "work
> > with existing SIP Implementations" standpoint, then I have to say
> > that so much of Record-Route wouldn't have worked anyway (the
> > spec says nothing about building Route:s in the callee -> caller
> > direction), is it really worth worrying?
> > 
> > With this approach, we can write into bis, say, that the called
> > UA, when forming requests, renames all "uri" parameters to
> > "original-uri"; and appends a new "uri" parameter with the value
> > of the Contact:/From:.
> > 
> > We now have the resilience that Jonathan was addressing (thread:
> > "failure of proxy), since each proxy along the Route can see
> > the final destination.  And also, if the request spiralled,
> > we know where on the Route we are (since we have retained the
> > original Request-URIs, too).
> > 
> > Now I know that this is a significant change, but as far as I can
> > tell, realistically, no proxy will as yet have been able to
> > Record-Route properly and/or remain call stateful, because we
> > are still finding bugs.  Thus it doesn't seem as if there can be
> > real objections to making such a changes, since hopefully they
> > are changes towards having a fully-working mechanism (okay, I
> > guess we think it works now, but it can't be more than 3 months
> > since we spotted that Record-Route: can break in the reverse
> > direction when a request has spiralled; how mature can
> > implementations possibly be?)
> > 
> 
> I thought I were the only one having these thoughts. This is
> the reason why I started to suggest simplifications to SIP some
> months ago. It was quite obvious that the protocol was not
> complete at that time (it's not even complete today). So
> to argue that a specification that is not complete and still
> contain major bugs have to be backwards compatible is very 
> strange to me. But I guess I have to live with that ??

I'm not sure that this is fair.  The protocol may fall slightly
short of being complete even today (although I might argue that
it is more our understanding that is uncomplete; the protocol
_may_ be spot on already), but where does the "major bugs" come
from?

> The kind of solution I would like to have to this problem is
> a change of the ROUTE behaviour so it behaves like VIA i.e.
> you remove your "own" entry from the list instead of removing
> the next servers entry. In this way it would be possible to
> add any RECORD-ROUTE parameter and get it back later in ROUTE 
> (if the client copies the whole RECORD-ROUTE into ROUTE). This
> would make the state draft obsolete and it would also solve
> my problem with identifying the call state machine. Since
> this change is a pure semantic change I see no reason why
> it can't be done like this ;-). 

I don't buy this at all.  And championing it on the basis that
it's purely a semantic change seems a little weak, at best.

In fact, I would argue that such a potentially subtle change
(older implementations would think they were doing the Right
Thing, while devices downstream would be suffering major
lossage) is dangerous.  I would consider it preferable to
do something like inventing Record-Route2:/Route2:, since
then at least the intent is clear all the way, 2543 or no.
(Please note that I'm not proposing we do this.)

> The only syntax change needed is to add a 'hostport' 
> parameter to RECORD-ROUTE instead of maddr as Jonathan
> pointed out. Since this is needed for TEL-URI etc. this
> parameter has to be added but I would prefere to ALWAYS 
> use this new parameter instead of sometimes using maddr.

Unfortunately, this is not backwards compatible, whereas
what I proposed -- if one is generous -- was completely
backwards compatible.  (Technically, I think it conflicts
with the bis drafts, but that is all?)

To be honest, I've been slightly disappointed with the
lack of response to my original suggestion; inflamatory or
otherwise.  I would have thought it was controversial
enough to merit a medium-sized fire extinguisher at the
very least. &:)

I think (please do correct if I wrong) that my solution
potentially addressed all the problems in and around
this thread; it even left the way open for state to be
stored in the Request-URIs (I would prefer to see the
State header used for this, but obviously there is then
an issue of what to do if a UA don't support it).

It'd be nice if we could try and get this nailed down
once and for all[0] before the Bake-Off, whichever hammer
we use.  That way, we can all offer up our bleeding-edge
I-knocked-it-up-during-the-flight-over-here implementations
there and then, and prove it works (or doesn't, as the
case may be).  I'm sure a "simple" scenario involving
a single spiral would suffice...

Cheers,


 - Jo.

[0] Until we realise the next bug, of course. &:)


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Subject: Re: [SIP] Multiple Aliases
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Date: Mon, 13 Nov 2000 08:48:02 -0500
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You could also register with multiple service providers.
----------------------------
David Shrader
Master Consultant, Inc.
dshrader@master-consultant.com
http://www.EyeForTheFuture.com

> From: "Jo Hornsby" <jhornsby@ubiquity.net>
> Date: Mon, 13 Nov 2000 11:16:36 -0000
> To: "zeeshan razzaque" <zrazzaque@yahoo.com>, <sip@lists.bell-labs.com>
> Subject: RE: [SIP] Multiple Aliases
> 
>> Hi, Can my UA be registered with multiple Aliases?
>> (analogous to having a single mailbox for different
>> email addresses)
> 
> Very much so (the only thing preventing this would be
> a lack of Service Providers/Registrars willing to let
> you do this).
> 
> 
> - Jo.
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 


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From sip-admin@lists.bell-labs.com  Mon Nov 13 09:04:05 2000
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Subject: Re: [SIP] SIP ringback tone
From: David Shrader <dshrader@master-consultant.com>
To: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
Cc: SIP List <sip@lists.bell-labs.com>
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Robert,

I don't agree that 183 with SDP should only be a call progress message. It
is a perfectly reasonable usage to support ringback. This is especially
necessary when interworking with the PSTN where ringback is delivered
inband. If you implement a system that works with the PSTN, I think we
shouldn't require it to support two modes of operation. It seems reasonable
that the use of 183 masks whether I'm going to the PSTN or another SIP
endpoint identified by a telephone number and allows the person or network
being called to provide whatever inband indication it chooses.

David

----------------------------
David Shrader
Master Consultant, Inc.
dshrader@master-consultant.com
http://www.EyeForTheFuture.com

> From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
> Date: Mon, 13 Nov 2000 03:33:35 -0800
> To: "'Henry Sinnreich'" <Henry.Sinnreich@wcom.com>
> Cc: "'sip@lists.bell-labs.com'" <sip@lists.bell-labs.com>
> Subject: RE: [SIP] SIP ringback tone
> 
> Henry,
> 
> Innovative ringback greetings definitely seem like a case for an early media
> to me.
> 
> As an aside, many people still seem to erroroneously think that 183 Session
> Progress should be used for early-media ring-back. 180 Ringing with SDP
> should be used for ringback purposes, 183 with SDP should only be used for
> call progress messages (ie, not alerting the user). I still think this needs
> to be made clearer in the spec!
> 
> Cheers,
> 
> Robert.
> 
> -- My opinions are my own. I tried selling them once but everybody
> seems to already have one. --
> 
>> -----Original Message-----
>> From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
>> Sent: Monday, November 13, 2000 6:18 AM
>> To: adam.roach@ericsson.com; IETF SIP (E-mail)
>> Subject: [SIP] SIP ringback tone
>> 
>> 
>> The draft on "Ringback tones in SIP-Based Telephony" shows
>> impressive work
>> on preserving legacy country specific ring back tones. I
>> wonder if we would
>> not be better of by adding a
>> 
>> SIP Ringback Tone
>> =================
>> 
>> With some optional fields where service providers could add innovative
>> greetings.
>> 
>> Henry
>> 
>> Henry Sinnreich
>> WorldCom
>> 400 International Parkway,
>> Richardson, Texas, 75081
>> 
>> 
>> 
>> _______________________________________________
>> SIP mailing list
>> SIP@lists.bell-labs.com
>> http://lists.bell-labs.com/mailman/listinfo/sip
>> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 


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Date: Mon, 13 Nov 2000 09:11:58 -0500



 

> -----Original Message-----
> From: James Undery [mailto:jundery@ubiquity.net]
> Sent: Wednesday, November 08, 2000 6:02 AM
> To: Jonathan Rosenberg
> Cc: sip@lists.bell-labs.com
> Subject: Re: [SIP] failure of proxy
> 
> 
> 
> 
> Jonathan Rosenberg wrote:
> 
> >
> > For things to work consistently, the SRV lookup process 
> should be based on
> > the pseudo-random number generator I suggested in a 
> previous thread; thats
> > the one based on a hash of the Call-ID, amongst other 
> things. This gives us
> > consistent operation for SRV records in general.
> 
> As I replied in a previous thread (with a bad example) this 
> hash based method
> gives us nothing in general it only works if you locally 
> cache DNS for the life
> time of a transaction, for each transaction (hence truly 
> stateless proxy servers
> do not work ;^) ). The reason for this is simple  for the 
> hash method to work
> the result from the DNS query MUST be identical even in 
> ordering of the RRs each
> time.

OK, so my proposal is that a UA or proxy that does a DNS query should order
entries of equal priority and preference using some sorting rule,
alphabetical or whatever. We really need for this to work somehow.

 As an example of why this may not happen I'll admit my 
> short comings, the
> first time I implemented SRV lookups I was lazy I found the 
> lowest priority RRs
> and picked the first one. If I was writing a DNS server I may 
> expect other
> people to do this, and in a more thorough moment may decide 
> to shuffle the
> results my server returned for SRV records (this wouldn't 
> provide the same level
> of load balancing but could help).
> 
> Also it's possible the zone file might have changed.

In that case, I think it is reasonable that requests may go to different
places.

-JOnathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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From sip-admin@lists.bell-labs.com  Mon Nov 13 09:18:05 2000
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
To: "'vkg@lucent.com'" <vkg@lucent.com>, Juha Heinanen <jh@lohi.eng.telia.fi>
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Date: Mon, 13 Nov 2000 09:15:47 -0500



 

> -----Original Message-----
> From: Vijay K. Gurbani [mailto:vkg@lucent.com]
> Sent: Thursday, November 09, 2000 2:25 PM
> To: Juha Heinanen
> Cc: sip@lists.bell-labs.com
> Subject: Re: [SIP] bye and record route
> 
> 
> Juha Heinanen wrote:
> > 
> > section 4.2.4 bye says:
> > 
> > If the INVITE request contained a Contact header, the callee SHOULD 
> > send a BYE request to that address rather than the From address.
> > 
> > if the invite had record route header, shouldn't the bye request be 
> > send to its maddr instead?
> 
> Right.  When a UAC wants to send the request to it's next hop AND a
> Route list has been constructed, it'll examine the topmost Route and
> follow the rules listed in Section 1.4.2, "Locating a SIP Server" to
> figure out the destination.  Step 1 in that section says, "If maddr
> parameter exists, it becomes the /destination address/ used ..."

The text in 4.2.4 as it reads now is incorrect. The BYE should be sent
using the regular routing mechanisms. THe text would seem to imply that you
would ignore record-routes for BYE, which is wrong, as Vijay has correctly
pointed out.

One of the cleanup items in my list is to consolidate all this text on next
hop computations into one place. Right now its all over the place and not
always consistent.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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Authors of Internet drafts:

Please do not include numeric citations, such as 

"SIP [1] is ..."

in abstracts. They produce dangling references in I-D announcements and
various other places. The I-D/RFC editor doesn't like those, either... 

Henning
(maintainer of the rfc.bib and i-d.bib BibTeX files and various lists of
SIP abstracts)
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Date: Mon, 13 Nov 2000 02:38:51 -0800 (PST)

Hi,

My name is Niranjan and am working on SIP Stack for
our Call Agent\MGC. I would like to know if there are
any HTTP Parser tools available for SIP ?

Please Clarify...

With Regards,
Niranjan Dahanakoti
Axes Technologies Pvt Ltd.



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: SIP Extensions for Caller Identity and Privacy
	Author(s)	: W. Marshall et al.
	Filename	: draft-ietf-sip-privacy-00.txt
	Pages		: 16
	Date		: 10-Nov-00
	
This document describes two extensions to the Session Initiation
Protocol (SIP) [4]. The extensions allow callers and callees to
maintain their privacy in an environment where one or more proxies
serve as intermediaries which can provide the identity of the
parties either directly or indirectly. The extensions allow the
parties to be identified either by name or by type, the latter of
which can be used to identify some group of callers and callees.

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: Integration of Resource Management and SIP
	Author(s)	: B. Marshall et al.
	Filename	: draft-ietf-sip-manyfolks-resource-00.txt
	Pages		: 25
	Date		: 10-Nov-00
	
This document discusses how network QoS and security establishment
can be made a precondition to sessions initiated by the Session
Initiation Protocol (SIP), and described by SDP. These preconditions
require that the participant reserve network resources (or establish
a secure media channel) before continuing with the session. We do
not define new QoS reservation or security mechanisms; these pre-
conditions simply require a participant to use existing resource
reservation and security mechanisms before beginning the session.
This results in a multi-phase call-setup mechanism, with the
resource management protocol interleaved between two phases of call
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available before the phone rings and the participants of the call
are 'invited' to participate.

A URL for this Internet-Draft is:
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Internet-Drafts are also available by anonymous FTP. Login with the username
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type "cd internet-drafts" and then
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or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


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From sip-admin@lists.bell-labs.com  Mon Nov 13 11:34:04 2000
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From: David Shrader <dshrader@master-consultant.com>
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Subject: [SIP] Registration, Routing,  and Record-Route
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Date: Mon, 13 Nov 2000 11:24:38 -0500
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I have a scenario that I am trying to solve using SIP. The pieces don't
quite seem to come together for me. Do you have any thoughts?

Given the following constraints (lets assume for this scenario):

1. A sip user agent must access a service provider network via a specific
proxy that is to be used for the life of the particular registration. (After
re-boot or network condition, a different proxy may be used to register.)

2. The service provider will route all calls to that user agent via the
specific proxy also for the life of the particular registration.

3. The SIP UA registers its current location with the network domain server
via the proxy.

4. The network domain server receiving an INVITE for the user, determines
the users current location and needs to now route all calls to the current
location via the proxy. Is there a SIP mechanism associated with the
registration that can indicate this?

I think that the REGISTER message in step 3 has a Request-URI of the network
domain server and is routed to the IP address of the proxy. The proxy would
then look at the Request-URI and forward the request on to the network
domain server. 

One thought I had was that a Record-Route added by the proxy would establish
a route that would be taken for all of subsequent registration requests on
that registration "call leg". This, however, would not necessarily apply to
subsequent INVITEs for new calls.

Anybody have any thoughts on this problem?

David

----------------------------
David Shrader
Master Consultant, Inc.
dshrader@master-consultant.com
http://www.EyeForTheFuture.com


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From sip-admin@lists.bell-labs.com  Mon Nov 13 12:04:14 2000
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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: "'sip@lists.bell-labs.com'" <sip@lists.bell-labs.com>
Subject: RE: [SIP] SIP ringback tone
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Date: Mon, 13 Nov 2000 09:04:42 -0800

The way the spec is written at the moment (see Section 7.1 - it changed
recently), a UAC should be able to accept any 1xx response (other than 100)
with SDP as indicating early media. eg 183, 182, 181 or 180. Obviously they
must have different meanings or there is no reason to have the different
response codes (COMET aside). Both the expired 183 draft (explicitly) and
the bis-02 draft (inferred) say that 183 indicates call progress events
_other_ than user alerting (eg "queued message", "please wait, verifying" or
"enter pin number" voice messages) --- yet the mechanism is the same for 180
and 183. [I just think it should be made clearer in the 183 description.]

I don't understand your point about requiring support for two mechanisms and
that it "masks whether I'm going to the PSTN or another SIP endpoint
identified by a telephone number". How so? And more importantly, why is this
necessary ?

In fact I would say the opposite is true. A SIP client that is not PSTN
centric and doesn't bother with early media playback should not have to know
that 183 is actually ringback on a PSTN SIP client. The client should simply
know that a received 180 indicates user alerting, 182 is queued and 183 is
some other call progress event.

Regards,

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 
> -----Original Message-----
> From: David Shrader [mailto:dshrader@master-consultant.com]
> Sent: Monday, November 13, 2000 9:54 PM
> To: Fairlie-Cuninghame, Robert
> Cc: SIP List
> Subject: Re: [SIP] SIP ringback tone
> 
> 
> Robert,
> 
> I don't agree that 183 with SDP should only be a call 
> progress message. It
> is a perfectly reasonable usage to support ringback. This is 
> especially
> necessary when interworking with the PSTN where ringback is delivered
> inband. If you implement a system that works with the PSTN, I think we
> shouldn't require it to support two modes of operation. It 
> seems reasonable
> that the use of 183 masks whether I'm going to the PSTN or another SIP
> endpoint identified by a telephone number and allows the 
> person or network
> being called to provide whatever inband indication it chooses.
> 
> David
> 
> ----------------------------
> David Shrader
> Master Consultant, Inc.
> dshrader@master-consultant.com
> http://www.EyeForTheFuture.com
> 
> > From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
> > Date: Mon, 13 Nov 2000 03:33:35 -0800
> > To: "'Henry Sinnreich'" <Henry.Sinnreich@wcom.com>
> > Cc: "'sip@lists.bell-labs.com'" <sip@lists.bell-labs.com>
> > Subject: RE: [SIP] SIP ringback tone
> > 
> > Henry,
> > 
> > Innovative ringback greetings definitely seem like a case 
> for an early media
> > to me.
> > 
> > As an aside, many people still seem to erroroneously think 
> that 183 Session
> > Progress should be used for early-media ring-back. 180 
> Ringing with SDP
> > should be used for ringback purposes, 183 with SDP should 
> only be used for
> > call progress messages (ie, not alerting the user). I still 
> think this needs
> > to be made clearer in the spec!
> > 
> > Cheers,
> > 
> > Robert.
> > 
> > -- My opinions are my own. I tried selling them once but everybody
> > seems to already have one. --
> > 
> >> -----Original Message-----
> >> From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
> >> Sent: Monday, November 13, 2000 6:18 AM
> >> To: adam.roach@ericsson.com; IETF SIP (E-mail)
> >> Subject: [SIP] SIP ringback tone
> >> 
> >> 
> >> The draft on "Ringback tones in SIP-Based Telephony" shows
> >> impressive work
> >> on preserving legacy country specific ring back tones. I
> >> wonder if we would
> >> not be better of by adding a
> >> 
> >> SIP Ringback Tone
> >> =================
> >> 
> >> With some optional fields where service providers could 
> add innovative
> >> greetings.
> >> 
> >> Henry
> >> 
> >> Henry Sinnreich
> >> WorldCom
> >> 400 International Parkway,
> >> Richardson, Texas, 75081
> >> 
> >> 
> >> 
> >> _______________________________________________
> >> SIP mailing list
> >> SIP@lists.bell-labs.com
> >> http://lists.bell-labs.com/mailman/listinfo/sip
> >> 
> > 
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> > 
> 

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From sip-admin@lists.bell-labs.com  Mon Nov 13 12:13:05 2000
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Subject: Re: [SIP] Ringback draft
To: hgs@cs.columbia.edu (Henning G. Schulzrinne)
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In-Reply-To: <3A0C7BA1.3B0BC3D8@cs.columbia.edu> from "Henning G. Schulzrinne" at Nov 10, 2000 05:50:09 PM
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Henning G. Schulzrinne writes:
>Note also that you could use the Alert-Info for the
>same purpose, by providing an external (cached) reference to whatever
>national tones you'd like the caller to hear.

I believe this to be the opposite of "Alert-Info."

My understanding is that "Alert-Info" indicates content to be rendered
to the callED party. My draft addresses content to be rendered to the
callING party.

Please correct me if I'm mistaken.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA

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From sip-admin@lists.bell-labs.com  Mon Nov 13 12:30:06 2000
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"Adam B. Roach" wrote:
> 
> Henning G. Schulzrinne writes:
> >Note also that you could use the Alert-Info for the
> >same purpose, by providing an external (cached) reference to whatever
> >national tones you'd like the caller to hear.
> 
> I believe this to be the opposite of "Alert-Info."
> 
> My understanding is that "Alert-Info" indicates content to be rendered
> to the callED party. My draft addresses content to be rendered to the
> callING party.
> 
> Please correct me if I'm mistaken.

You're right; as currently defined. What I meant and should have spelled
out is that it may make sense (if this functionality makes sense, which
I'm less than convinced...) to also define Alert-Info as a response
header, not just a request header.

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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From sip-admin@lists.bell-labs.com  Mon Nov 13 12:58:15 2000
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Subject: [SIP] new torture tests
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The following are 15 new proposed torture test messages 
based on the bis draft. None of them should cause good 
implementations to crash. Recommended responses to the 
illegal messages are given where needed.

I would greatly appreciate any additions/corrections to be
made quickly so we can try to get these folded into a new 
release of the call flows draft before the next bakeoff.

Cheers,
Neil.
-- 
Ubiquity Software Corporation, UK        http://www.ubiquity.net


1) Illegal enclosing of Request-URI  in "<>"
Server should respond with 400 + Appropriate Reason Phrase.

INVITE <sip:user@company.com> SIP/2.0
To: sip:user@company.com
From: sip:caller@university.edu
Call-ID: 1@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.180.130.133
Content-Type: application/sdp
Content-Length: 174

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC


2) Illegal LWS in elements of Request-URI
Server should respond with 400 + Appropriate Reason Phrase.

INVITE sip:user@company.com; transport=udp SIP/2.0
To: sip:user@company.com
From: sip:caller@university.edu
Call-ID: 2@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.180.130.133
Content-Type: application/sdp
Content-Length: 174

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC


3) Illegal >1 SP between elements of Request URI
Server should respond with 400 + Appropriate Reason Phrase.

INVITE sip:user@company.com  SIP/2.0
To: sip:user@company.com
From: sip:caller@university.edu
Call-ID: 3@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.180.130.133
Content-Type: application/sdp
Content-Length: 174

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC


4) Legal SIP URL in Request URI with escaped characters and
parameter.

INVITE sip%3Auser%40company.com;other-param=summit SIP/2.0
To: sip:user@company.com
From: sip:caller@university.edu
Call-ID: 4@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.180.130.133
Content-Type: application/sdp
Content-Length: 174

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC


5) Illegal use of escaped header in Request-URI
Proxies should strip off the escaped header before proxying on.

INVITE sip:user@company.com?Route=%3Csip:sip.example.com%3E
SIP/2.0
To: sip:user@company.com
From: sip:caller@university.edu
Call-ID: 5@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.180.130.133
Content-Type: application/sdp
Content-Length: 174

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC

6) Not understood URI in Req-URI but a SIP URL in To.
Server should respond 400 + Appropriate Reason Phrase

INVITE name:user SIP/2.0
To: sip:user@company.com
From: sip:caller@university.edu
Call-ID: 6@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.180.130.133
Content-Type: application/sdp
Content-Length: 174

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC


7) Legal no LWS between display name and <

OPTIONS sip:user@company.com SIP/2.0
To: sip:user@company.com
From: "caller"<sip:caller@example.com>
Call-ID: 1234abcd@10.0.0.1
CSeq: 1 OPTIONS
Via: SIP/2.0/UDP 135.180.130.133

8) Legal extra LWS between display name and <

OPTIONS sip:user@company.com SIP/2.0
To: sip:user@company.com
From: "caller"    <sip:caller@example.com>
Call-ID: 1234abcd@10.0.0.1
CSeq: 2 OPTIONS
Via: SIP/2.0/UDP 135.180.130.133

9) Illegal Non GMT time zone in SIP Date
A liberal server should correct this.

INVITE sip:user@company.com SIP/2.0
To: sip:user@company.com
From: sip:caller@university.edu
Call-ID: 7@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.180.130.133
Expires: Fri, 01 Jan 2010 16:00:00 EST
Content-Type: application/sdp
Content-Length: 174

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC

10) Legal message with passed Exipres date.
A Server should respond 408 Request Timeout.

INVITE sip:user@company.com SIP/2.0
To: sip:user@company.com
From: sip:caller@university.edu
Call-ID: 8@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.180.130.133
Expires: Thu, 01 Dec 1994 16:00:00 GMT
Content-Type: application/sdp
Content-Length: 174

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC


11) Legal message with zero max forwards
Proxies should return 483 Too Many Hops, a UAS responds 
as for OPTIONS.

INVITE sip:user@company.com SIP/2.0
To: sip:user@company.com
From: sip:caller@university.edu
Call-ID: 9@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.180.130.133
Max-Forwards: 0
Content-Type: application/sdp
Content-Length: 174

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC

12) Legal REGISTER with SIP URL containing escaped Route header
in Contact.

REGISTER sip:company.com SIP/2.0
To: sip:user@company.com
From: sip:user@company.com
Contact: sip:user@host.company.com
Call-ID: k345asrl3fdbv@10.0.0.1
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 135.180.130.133
Contact: sip:user@example.com?Route=%3Csip:sip.example.com%3E

13) Legal but long, includes Token for Call-ID.

INVITE sip:user@company.com SIP/2.0
To: "I have a user name of extreme proportion"
<sip:user@company.com:6000;other-param=1234567890somethingelselong1234567890>
From: sip:caller@university.edu
Call-ID:
kl24ahsd546folnyt2vbak9sad98u23naodiunzds09a3bqw0sdfbsk34poouymnae0043nsed09mfkvc74bd0cuwnms05dknw87hjpobd76f
CSeq: 1 INVITE
My-State: sldkjflzdsfaret0803adgaasd0afds0asdaasd
Via: SIP/2.0/UDP sip33.example.com
Via: SIP/2.0/UDP sip32.example.com
Via: SIP/2.0/UDP sip31.example.com
Via: SIP/2.0/UDP sip30.example.com
Via: SIP/2.0/UDP sip29.example.com
Via: SIP/2.0/UDP sip28.example.com
Via: SIP/2.0/UDP sip27.example.com
Via: SIP/2.0/UDP sip26.example.com
Via: SIP/2.0/UDP sip25.example.com
Via: SIP/2.0/UDP sip24.example.com
Via: SIP/2.0/UDP sip23.example.com
Via: SIP/2.0/UDP sip22.example.com
Via: SIP/2.0/UDP sip21.example.com
Via: SIP/2.0/UDP sip20.example.com
Via: SIP/2.0/UDP sip19.example.com
Via: SIP/2.0/UDP sip18.example.com
Via: SIP/2.0/UDP sip17.example.com
Via: SIP/2.0/UDP sip16.example.com
Via: SIP/2.0/UDP sip15.example.com
Via: SIP/2.0/UDP sip14.example.com
Via: SIP/2.0/UDP sip13.example.com
Via: SIP/2.0/UDP sip12.example.com
Via: SIP/2.0/UDP sip11.example.com
Via: SIP/2.0/UDP sip10.example.com
Via: SIP/2.0/UDP sip9.example.com
Via: SIP/2.0/UDP sip8.example.com
Via: SIP/2.0/UDP sip7.example.com
Via: SIP/2.0/UDP sip6.example.com
Via: SIP/2.0/UDP sip5.example.com
Via: SIP/2.0/UDP sip4.example.com
Via: SIP/2.0/UDP sip3.example.com
Via: SIP/2.0/UDP sip2.example.com
Via: SIP/2.0/UDP sip1.example.com
Via: SIP/2.0/UDP
host.example.com;received=135.180.130.133;branch=C1C3344E2710000000E299E568E7potato10potato0potato0
Content-Type: application/sdp

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC


14) Illegal, badly mangled message with mutliple To, 
From, Call-ID and CSeq headers.
Server should respond 400 + Appropriate Reason Phrase

OPTIONS sip:135.180.130.133 SIP/2.0
Via: SIP/2.0/UDP company.com:5604
From: sip:iuser@company.com
To: sip:user@135.180.130.133
Call-ID: 1804928587@company.com
CSeq: 1 OPTIONS
Expires: 0 0l@company.com
To: sip:user@135.180.130.133
Call-ID: 1804928587@company.com
CSeq: 1 OPTIONS
Contact: sip:host.company.com
Expires: 0xpires: 0sip:host.company.com
Expires: 0
Contact: sip:host.company.com

15) Legal message with an unusual amount of SDP attributes. 

INVITE sip:+1-650-555-2222@ss1.wcom.com;user=phone SIP/2.0
Via: SIP/2.0/UDP iftgw.there.com:5060
From: sip:+1-303-555-1111@ift.here.com;user=phone
To: sip:+1-650-555-2222@ss1.wcom.com;user=phone
Call-ID: 1717@ift.here.com
CSeq: 56 INVITE
Content-Type: application/sdp
Content-Length: 320

v=0
o=faxgw1 2890844527 2890844527 IN IP4 iftgw.there.com
s=Session SDP
c=IN IP4 iftmg.there.com
t=0 0
m=image 49172 udptl t38
a=T38FaxVersion:0
a=T38maxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:260
a=T38FaxUdpEC:t38UDPRedundancy

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From sip-admin@lists.bell-labs.com  Mon Nov 13 13:11:04 2000
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Date: Mon, 13 Nov 2000 11:33:21 -0500

If a PSTN originated call is redirected to a gateway, how would, for
example, a Q.931 redirected number or SS7 RDN/OCN be mapped into a SIP
INVITE message?  

I've been looking through various drafts to determine the appropriate
action but I can't seem to find the right one.  Could somone point me
in the right direction?

Thanks.

-- 
Peter Kazmier                                 http://www.kazmier.com
PGP Fingerprint   4FE7 8DA3 D0B5 9CAA 69DC  7243 1855 BC2E 4B43 5654

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Date: Mon, 13 Nov 2000 13:27:24 -0500

http://search.ietf.org/internet-drafts/draft-levy-sip-diversion-00.txt



At 11:33 AM 11/13/2000 -0500, Pete Kazmier wrote:
>If a PSTN originated call is redirected to a gateway, how would, for
>example, a Q.931 redirected number or SS7 RDN/OCN be mapped into a SIP
>INVITE message?
>
>I've been looking through various drafts to determine the appropriate
>action but I can't seem to find the right one.  Could somone point me
>in the right direction?
>
>Thanks.
>
>--
>Peter Kazmier                                 http://www.kazmier.com
>PGP Fingerprint   4FE7 8DA3 D0B5 9CAA 69DC  7243 1855 BC2E 4B43 5654
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip


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From: "Agarwal, Aseem" <aseem@trillium.com>
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Date: Mon, 13 Nov 2000 10:26:18 -0800

Hi,
  Is this document different from draft-manyfolks-sip-resource-01.txt ??
If not, should n't this document have a higher version number ?

Also, does this draft obsolete "Establishing QOS and Security Preconditions
for SDP Sessions", draft-ietf-mmusic-sdp-qos-00.txt ??

thanks,
aseem

-----Original Message-----
From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
Sent: Monday, November 13, 2000 3:41 AM
Cc: sip@lists.bell-labs.com
Subject: [SIP] I-D ACTION:draft-ietf-sip-manyfolks-resource-00.txt


A New Internet-Draft is available from the on-line Internet-Drafts
directories.
This draft is a work item of the Session Initiation Protocol Working Group
of the IETF.

	Title		: Integration of Resource Management and SIP
	Author(s)	: B. Marshall et al.
	Filename	: draft-ietf-sip-manyfolks-resource-00.txt
	Pages		: 25
	Date		: 10-Nov-00
	
This document discusses how network QoS and security establishment
can be made a precondition to sessions initiated by the Session
Initiation Protocol (SIP), and described by SDP. These preconditions
require that the participant reserve network resources (or establish
a secure media channel) before continuing with the session. We do
not define new QoS reservation or security mechanisms; these pre-
conditions simply require a participant to use existing resource
reservation and security mechanisms before beginning the session.
This results in a multi-phase call-setup mechanism, with the
resource management protocol interleaved between two phases of call
signaling. The objective of such a mechanism is to enable deployment
of robust IP Telephony services, by ensuring that resources are made
available before the phone rings and the participants of the call
are 'invited' to participate.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-manyfolks-resource-00.txt

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-sip-manyfolks-resource-00.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-ietf-sip-manyfolks-resource-00.txt".
	
NOTE:	The mail server at ietf.org can return the document in
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Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.


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From: "Robert Sparks" <rsparks@dynamicsoft.com>
To: "Adam B. Roach" <Adam.Roach@Ericsson.com>
Cc: <sip@lists.bell-labs.com>
Subject: RE: [SIP] Possible REFER problem?
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I had to rethink this whole conversation to respond,
and as part of the process captured the following notes.
Hopefully, they'll be a helpful summary:

The scenario that started this discussion consisted of
A calling B, A puts B on hold, A calls C, A then wants
to transfer B to C and try to ensure that the same B
gets the same C-endpoint that A was talking to.

The question was whether the Refer-To header in the REFER
message that A sends to B should contain the URL that A
used to reach C in the first place, or the URL that C
returned in the 200 OK to A's INVITE.

The Contact that C provided to A is only guaranteed to make
sense in the context of A's session with C. For example, the
hostname may be in a private namespace in a company's LAN.
A and B would both fail if they did a DNS lookup on it, but
the last proxy on the Route before that contact would have
no problems with it.

Even if C gave A a contact that A could resolve, there
is no guarantee that C will honor that contact outside the
call leg that A and C have set up.

Jonathan's suggested method is to place the contact that
C gave to A in the Accept-Contact header to increase the
probability (not ensure) that B reaches the same C-endpoint
that A did.

Given all that, and the likelyhood that for many (if not
most) instances, the contact provided in the Contact URL
_is_ routable, what problems exist with taking the following
approach?

  1) REFER B directly to the contact that C provided to A.
  2) If it fails, REFER B to C's well known URL with an Accept-Contact
     header containing the contact that C provided to A.

RjS


> -----Original Message-----
> From: sip-admin@lists.bell-labs.com
> [mailto:sip-admin@lists.bell-labs.com]On Behalf Of Adam B. Roach
> Sent: Wednesday, November 08, 2000 6:15 PM
> To: Eric Tremblay
> Cc: 'Jonathan Rosenberg'; 'Cliff.Harris@nokia.com'; Eric Tremblay;
> 'sip@lists.bell-labs.com'
> Subject: Re: [SIP] Possible REFER problem?
>
>
> Eric Tremblay writes:
> >I'm reviving and old thread ;)
> >
> >If you remember, the problem was when doing call transfer with
> consultation
> >with the REFER request, how can we make sure that the INVITE
> request (from
> >the Transferee to the Transfer Target) is going to reach the exact same
> >User-Agent as the one that was consulted by the Transferor.
> >
> >The solution was to use the header "Accept-Contact" and specify
> the contact
> >of the transfer target.
>
> <explanation of why this won't necessarily work snipped>
>
> At the risk of sounding daft, I don't see why using the "Contact:"
> header as the new "INVITE" URI is considered Evil in this case.
>
> When explaining the difference between, say, the "From:" (or "To:")
> field and the "Contact:" field, I generally think of "From:"/"To:" as
> being the persistent URI for contacting the appropriate resource in
> an abstract sense (by resource, I mean "Adam Roach" or "helpdesk staffer"
> or "sales representative"), while the "Contact:" URI indicates "where to
> get to me (as a particular instance of that resource) on this particular
> device at this particular moment."
>
> Obviously, the most useful application of this latter piece of information
> is routing of subsequent messages for a particular call -- but I don't
> see why it absolutely MUST be the only use for this information.
>
> For the "Contact" information to be useful in any way, it needs to be
> something that can be routed when used as a Request-URI; it wouldn't
> work for "BYE" messages otherwise, right?
>
> So, if the transferee were passed this information to be used as
> the Request-URI, there should be no discernable problems with the
> INVITE (or other appropriate method) arriving at the partcular
> instance of the desired resource on a particular device.
>
> Since no one else has raised this point, I'm going to assume that
> I'm too blind to see the elephant in the room. So, could someone
> explain *why* using a Contact URI in this manner would be inappropriate
> behavior?
>
> --
> Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E.
> Arapaho, MS L-04
> adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX
> 75081 USA
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>


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From: Billy Biggs <Billy_Biggs@3com.com>
To: Robert Sparks <rsparks@dynamicsoft.com>
Cc: "Adam B. Roach" <Adam.Roach@Ericsson.com>, sip@lists.bell-labs.com
Subject: Re: [SIP] Possible REFER problem?
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Robert Sparks (rsparks@dynamicsoft.com):

> [...]
> 
> The scenario that started this discussion consisted of A calling B, A
> puts B on hold, A calls C, A then wants to transfer B to C and try to
> ensure that the same B gets the same C-endpoint that A was talking to.
> 
> The question was whether the Refer-To header in the REFER message that
> A sends to B should contain the URL that A used to reach C in the
> first place, or the URL that C returned in the 200 OK to A's INVITE.

  C may have called A, in which case the well known URL is unknown.  You
can't use the From here either since it is outside of the scope.

> The Contact that C provided to A is only guaranteed to make
> sense in the context of A's session with C.  [...]

  I disagree with this.  The contact that C provided has to be available
at least for new calls within the local namespace, and if not, then
whatever translator in the middle (outbound proxy, transparent
masquerade host or whatever) has to know to check the Refer-To header
and mangle it also.

  Otherwise this gets too complicated.

> Jonathan's suggested method is to place the contact that C gave to A
> in the Accept-Contact header to increase the probability (not ensure)
> that B reaches the same C-endpoint that A did.

  And put that as a header parameter in the URL given in the Refer-To
header?  This seems like a big hack, especially since the well known URL
may not be known.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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From sip-admin@lists.bell-labs.com  Mon Nov 13 15:34:11 2000
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From: "Robert Sparks" <rsparks@dynamicsoft.com>
To: "Billy Biggs" <Billy_Biggs@3com.com>
Cc: "Adam B. Roach" <Adam.Roach@Ericsson.com>, <sip@lists.bell-labs.com>
Subject: RE: [SIP] Possible REFER problem?
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> > The scenario that started this discussion consisted of A calling B, A
> > puts B on hold, A calls C, A then wants to transfer B to C and try to
> > ensure that the same B gets the same C-endpoint that A was talking to.
> >
> > The question was whether the Refer-To header in the REFER message that
> > A sends to B should contain the URL that A used to reach C in the
> > first place, or the URL that C returned in the 200 OK to A's INVITE.
>
>   C may have called A, in which case the well known URL is unknown.  You
> can't use the From here either since it is outside of the scope.

Interesting thought. Really unusual preconditions though. "Let me transfer
you to this other person that just called me". In that case, the contact
they've provided is the best you've got unless you can get better
information
through another channel.

>
> > The Contact that C provided to A is only guaranteed to make
> > sense in the context of A's session with C.  [...]
>
>   I disagree with this.  The contact that C provided has to be available
> at least for new calls within the local namespace, and if not, then
> whatever translator in the middle (outbound proxy, transparent
> masquerade host or whatever) has to know to check the Refer-To header
> and mangle it also.
>
>   Otherwise this gets too complicated.

I'm not sure what/who you're disagreeing with? The contact that C provides
has only to be valid for subsequent requests in that call-leg. (I can't
imagine
anyone actually implementing it that way, but afaict, that's all the spec
requires).

>
> > Jonathan's suggested method is to place the contact that C gave to A
> > in the Accept-Contact header to increase the probability (not ensure)
> > that B reaches the same C-endpoint that A did.
>
>   And put that as a header parameter in the URL given in the Refer-To
> header?  This seems like a big hack, especially since the well known URL
> may not be known.
>

You didn't comment on the hybrid approach at the end of the message. I
think it addresses the case you are wanting to solve (as the primary
usage even).

RjS


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Robert Sparks (rsparks@dynamicsoft.com):

> > > The scenario that started this discussion consisted of A calling
> > > B, A puts B on hold, A calls C, A then wants to transfer B to C
> > > and try to ensure that the same B gets the same C-endpoint that A
> > > was talking to.
> > >
> > > The question was whether the Refer-To header in the REFER message
> > > that A sends to B should contain the URL that A used to reach C in
> > > the first place, or the URL that C returned in the 200 OK to A's
> > > INVITE.
> >
> >   C may have called A, in which case the well known URL is unknown.
> > You can't use the From here either since it is outside of the
> > scope.
> 
> Interesting thought. Really unusual preconditions though. "Let me
> transfer you to this other person that just called me". In that case,
> the contact they've provided is the best you've got unless you can get
> better information through another channel.

  It's not unusual when you consider the general problem: how to excuse
yourself from a 3-way call.

  I have two active call legs and I want to join the two remote
endpoints using REFER.  Everything else is just a special case of this
problem.

> > > The Contact that C provided to A is only guaranteed to make sense
> > > in the context of A's session with C.  [...]
> >
> >   I disagree with this.  The contact that C provided has to be
> > available at least for new calls within the local namespace, and
> > if not, then whatever translator in the middle (outbound proxy,
> > transparent masquerade host or whatever) has to know to check the
> > Refer-To header and mangle it also.
> >
> >   Otherwise this gets too complicated.
> 
> I'm not sure what/who you're disagreeing with? The contact that C
> provides has only to be valid for subsequent requests in that
> call-leg. (I can't imagine anyone actually implementing it that way,
> but afaict, that's all the spec requires).

  Since the call between A and C is pretty much guaranteed to be still
going on, this pedantic UA seems even more ridiculous.  For the sake of
interoperability the URI that you put in a Contact should be available
for requests from other UAs, and if you're a multi-user phone or
gateway, it should probably also have a username.

> > > Jonathan's suggested method is to place the contact that C gave to
> > > A in the Accept-Contact header to increase the probability (not
> > > ensure) that B reaches the same C-endpoint that A did.
> >
> >   And put that as a header parameter in the URL given in the
> > Refer-To header?  This seems like a big hack, especially since the
> > well known URL may not be known.
> 
> You didn't comment on the hybrid approach at the end of the message. I
> think it addresses the case you are wanting to solve (as the primary
> usage even).

  The hyrbid approach you suggested was:

  1) REFER B directly to the contact that C provided to A.
  2) If it fails, REFER B to C's well known URL with an Accept-Contact
     header containing the contact that C provided to A.

  I found another problem with part 2:  The proxy now has to do some
funky matching to try and correlate the Contact provided with those
known.  Pretty crazy, since in your scenario it's not the same as how it
registered, but I guess in some cases it is somehow similar (different
port, host in the same subnet, etc).

  Of course, this is all implementation specific- there are probably
other ways of convincing the transfer to work if it fails.  It would be
nice to eventually define some body type to indicate to the referrer why
the call failed (application/sip-status?).

  However I would NOT recommend attempting step 2 as it is error-prone
and only solves a very unlikely specific case.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Mon, 13 Nov 2000 16:28:50 -0500


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Hello!


Tbanks for posting these (it has a laxative effect on my parser :).   In particular, I like the fact that you
have posted expected return messages (it would be useful if the previous list also had such information).
I have a couple of questions on the following header where you have used escape sequences:

Can the COLON in the sip: of the URI be an escaped sequence?

Second, you have used %40 in place of the @ at the end of the user name. Is this legal?  If this is legal,
then   there is no way for the user to use an escape sequence for part of his or her name.

Regards

Ranga.

>
>
> 4) Legal SIP URL in Request URI with escaped characters and
> parameter.
>
> INVITE sip%3Auser%40company.com;other-param=summit SIP/2.0
> To: sip:user@company.com
> From: sip:caller@university.edu
> Call-ID: 4@10.0.0.1
> CSeq: 1 INVITE
> Via: SIP/2.0/UDP 135.180.130.133
> Content-Type: application/sdp
> Content-Length: 174
>
> v=0
> o=mhandley 29739 7272939 IN IP4 126.5.4.3
> s=SIP Call
> t=3149328700 0
> c=IN IP4 135.180.130.88
> m=audio 492170 RTP/AVP 0 12
> m=video 3227 RTP/AVP 31
> a=rtpmap:31 LPC
>
>

--
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
Tel: 301 975 3664 Fax: 301 590 0932



--------------EC43B8B7C90D7ACA4D51C223
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
&nbsp;
<br>Hello!
<br>&nbsp;
<p>Tbanks for posting these (it has a laxative effect on my parser :).&nbsp;&nbsp;
In particular, I&nbsp;like the fact that you have posted expected return
messages (it would be useful if the previous list also had such information).&nbsp;
I&nbsp;have a couple of questions on the following header where you have
used escape sequences:
<p>Can the COLON&nbsp;in the sip: of the URI&nbsp;be an escaped sequence?
<p>Second, you have used %40 in place of the @ at the end of the user name.
Is this legal?&nbsp; If this is legal, then&nbsp;&nbsp; there is no way
for the user to use an escape sequence for part of his or her name.
<p>Regards
<p>Ranga.
<p><br>
<BR>
<blockquote TYPE=CITE>&nbsp;
<p>4) Legal SIP URL in Request URI with escaped characters and
<br>parameter.
<p>INVITE sip%3Auser%40company.com;other-param=summit SIP/2.0
<br>To: sip:user@company.com
<br>From: sip:caller@university.edu
<br>Call-ID: 4@10.0.0.1
<br>CSeq: 1 INVITE
<br>Via: SIP/2.0/UDP 135.180.130.133
<br>Content-Type: application/sdp
<br>Content-Length: 174
<p>v=0
<br>o=mhandley 29739 7272939 IN IP4 126.5.4.3
<br>s=SIP Call
<br>t=3149328700 0
<br>c=IN IP4 135.180.130.88
<br>m=audio 492170 RTP/AVP 0 12
<br>m=video 3227 RTP/AVP 31
<br>a=rtpmap:31 LPC
<br>&nbsp;
<br><a href="http://lists.bell-labs.com/mailman/listinfo/sip"></a>&nbsp;</blockquote>

<pre>--&nbsp;
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.&nbsp;
Tel: 301 975 3664 Fax: 301 590 0932</pre>
&nbsp;</html>

--------------EC43B8B7C90D7ACA4D51C223--


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Date: Mon, 13 Nov 2000 14:01:07 -0800
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 Hello :

   I am wondering what mechanisms companies are planning to 
 deploy for SIP security.  I've been hearing PGP and TLS 
 mainly, but I didn't see any postings for extensions to 
 either protocols.  Please tell me if I've skipped over some
 draft. Thanks.

 - Tom Tang


Tom Tang
IP Unity
1575 McCandless Drive
Milpitas, CA 95035
(408) 957-0800 x154 (w)
(408) 957-0823 (f) 

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> > > > The Contact that C provided to A is only guaranteed to make sense
> > > > in the context of A's session with C.  [...]
> > >
> > >   I disagree with this.  The contact that C provided has to be
> > > available at least for new calls within the local namespace, and
> > > if not, then whatever translator in the middle (outbound proxy,
> > > transparent masquerade host or whatever) has to know to check the
> > > Refer-To header and mangle it also.
> > > > Jonathan's suggested method is to place the contact that C gave to
> > > > A in the Accept-Contact header to increase the probability (not
> > > > ensure) that B reaches the same C-endpoint that A did.
> > >
> > >   And put that as a header parameter in the URL given in the
> > > Refer-To header?  This seems like a big hack, especially since the
> > > well known URL may not be known.
> >
> > You didn't comment on the hybrid approach at the end of the message. I
> > think it addresses the case you are wanting to solve (as the primary
> > usage even).
>
>   The hyrbid approach you suggested was:
>
>   1) REFER B directly to the contact that C provided to A.
>   2) If it fails, REFER B to C's well known URL with an Accept-Contact
>      header containing the contact that C provided to A.
>
>   I found another problem with part 2:  The proxy now has to do some
> funky matching to try and correlate the Contact provided with those
> known.  Pretty crazy, since in your scenario it's not the same as how it
> registered, but I guess in some cases it is somehow similar (different
> port, host in the same subnet, etc).

Actually, in this scenario it would likely be _exactly_ what was registered.
This is one of the premises of caller-prefs, yes? The funky matching is
defined in that draft. If for some reason, it wasn't registered, or the
proxy involved didn't implement caller-prefs, the proxy's going to route
to the same places it would have without the hint.

>
>   However I would NOT recommend attempting step 2 as it is error-prone
> and only solves a very unlikely specific case.

We _must_ be talking past each other here. What's proposed at 2 is a
straightforward use of the caller-preferences mechanism. At the absolute
worst, 2 will result in exactly what you would have gotten if
the REFER to B contained only C's well known URL.


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Date: Mon, 13 Nov 2000 14:33:27 -0800

Hi,
   Has the following draft been updated ? Is YES, please point
   me to the latest version available.

   "The SIP PROPOSE Method", draft-onghe-mmusic-sip-propose-00.txt, Jun 1999

thanks,
aseem


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From: Billy Biggs <Billy_Biggs@3com.com>
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Subject: Re: [SIP] Possible REFER problem?
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Date: Mon, 13 Nov 2000 17:04:12 -0600

Robert Sparks (rsparks@dynamicsoft.com):

> >   The hyrbid approach you suggested was:
> >
> >   1) REFER B directly to the contact that C provided to A.
> >   2) If it fails, REFER B to C's well known URL with an
> >      Accept-Contact header containing the contact that C provided to
> >      A.
> >
> >   I found another problem with part 2:  The proxy now has to do some
> > funky matching to try and correlate the Contact provided with
> > those known.  Pretty crazy, since in your scenario it's not the
> > same as how it registered, but I guess in some cases it is somehow
> > similar (different port, host in the same subnet, etc).
> 
> Actually, in this scenario it would likely be _exactly_ what was
> registered.  [...] At the absolute worst, 2 will result in exactly
> what you would have gotten if the REFER to B contained only C's well
> known URL.

  I'm sorry, there were two discussed scenarios where this would be
needed, and I got confused:


  Case 1 (what you meant):

> [...] the hostname may be in a private namespace in a company's LAN.

  -> Refer-To: <sip:user-c@well-known.com?Accept-Contact=private.addr>

  The refer failed because the Contact was a private address, and
hopefully it was the same address used to register at well-known.com.
You are correct, this will help well-known.com reach the private
address.  Apologies.


  Case 2 (my confusion):

> [...] The contact that C provides has only to be valid for subsequent
> requests in that call-leg.

  -> Refer-To: <sip:user-c@well-known.com?Accept-Contact=addr:31337>

  Here we have a UA which fork()s off a new process per call which has
its own port.  Caller-preferences does nothing, but you are correct in
that it doesn't hurt.


  However, I think this just makes the algorithm over-complicated.
You're making an assumption about why the original REFER failed when
there are many other possible explinations:

 - The REFER target returned a 486 or other error response
 - The REFER target has a sever failure
 - Some authentication failure
 - ...

  It also only works if A initiated the call to C.

  Like I said, implementations are free to try harder, but I wouldn't
recommend it.

  Note that I plan on hacking my masquerade module code to also mangle
the Refer-To header.  That's the bane of the NAT implementor- to forever
be modifying the NAT code to handle the new protocol or protocol
extension.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Mon, 13 Nov 2000 16:59:49 -0800

Why/Where would you use a redirect server over a proxy server and vice
versa?

Thanks,
Dave

David J Schmidlin
Ext. 4869
JF3-2-D8




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From: Billy Biggs <Billy_Biggs@3com.com>
To: "Schmidlin, David" <david.schmidlin@intel.com>
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Date: Mon, 13 Nov 2000 19:37:54 -0600

Schmidlin, David (david.schmidlin@intel.com):

> Why/Where would you use a redirect server over a proxy server and vice
> versa?

  If you were a web server, when would you send a redirect and when
would you proxy?  It's the same answer.

  Redirect servers are cheap to code, for one.  But these are logical
roles not separate entities, so your question should be more specific.

  In general you proxy a message when you are providing some additional
feature rather than simply performing a lookup.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: RE: [SIP] new torture tests
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Date: Mon, 13 Nov 2000 21:48:48 -0500


Comments below.
 

> -----Original Message-----
> From: Neil Deason [mailto:ndeason@ubiquity.net]
> Sent: Monday, November 13, 2000 12:56 PM
> To: Sip List
> Subject: [SIP] new torture tests
> 
> 
> The following are 15 new proposed torture test messages 
> based on the bis draft. None of them should cause good 
> implementations to crash. Recommended responses to the 
> illegal messages are given where needed.

Well, some of your messages have only mild errors in them. Based on the
philosophy of being liberal with what you receive, I would not declare an
entity to have failed the test if it does not generate the specific error
code.

> 1) Illegal enclosing of Request-URI  in "<>"
> Server should respond with 400 + Appropriate Reason Phrase.
> 
> INVITE <sip:user@company.com> SIP/2.0
> To: sip:user@company.com
> From: sip:caller@university.edu
> Call-ID: 1@10.0.0.1
> CSeq: 1 INVITE
> Via: SIP/2.0/UDP 135.180.130.133
> Content-Type: application/sdp
> Content-Length: 174
> 
> v=0
> o=mhandley 29739 7272939 IN IP4 126.5.4.3
> s=SIP Call
> t=3149328700 0
> c=IN IP4 135.180.130.88
> m=audio 492170 RTP/AVP 0 12
> m=video 3227 RTP/AVP 31
> a=rtpmap:31 LPC
> 
> 
> 2) Illegal LWS in elements of Request-URI
> Server should respond with 400 + Appropriate Reason Phrase.
> 
> INVITE sip:user@company.com; transport=udp SIP/2.0
> To: sip:user@company.com
> From: sip:caller@university.edu
> Call-ID: 2@10.0.0.1
> CSeq: 1 INVITE
> Via: SIP/2.0/UDP 135.180.130.133
> Content-Type: application/sdp
> Content-Length: 174
> 
> v=0
> o=mhandley 29739 7272939 IN IP4 126.5.4.3
> s=SIP Call
> t=3149328700 0
> c=IN IP4 135.180.130.88
> m=audio 492170 RTP/AVP 0 12
> m=video 3227 RTP/AVP 31
> a=rtpmap:31 LPC

I suspect many implementations will accept the above message. 

> 
> 
> 3) Illegal >1 SP between elements of Request URI
> Server should respond with 400 + Appropriate Reason Phrase.
> 
> INVITE sip:user@company.com  SIP/2.0
> To: sip:user@company.com
> From: sip:caller@university.edu
> Call-ID: 3@10.0.0.1
> CSeq: 1 INVITE
> Via: SIP/2.0/UDP 135.180.130.133
> Content-Type: application/sdp
> Content-Length: 174
> 
> v=0
> o=mhandley 29739 7272939 IN IP4 126.5.4.3
> s=SIP Call
> t=3149328700 0
> c=IN IP4 135.180.130.88
> m=audio 492170 RTP/AVP 0 12
> m=video 3227 RTP/AVP 31
> a=rtpmap:31 LPC

Same here.

> 5) Illegal use of escaped header in Request-URI
> Proxies should strip off the escaped header before proxying on.
> 
> INVITE sip:user@company.com?Route=%3Csip:sip.example.com%3E
> SIP/2.0
> To: sip:user@company.com
> From: sip:caller@university.edu
> Call-ID: 5@10.0.0.1
> CSeq: 1 INVITE
> Via: SIP/2.0/UDP 135.180.130.133
> Content-Type: application/sdp
> Content-Length: 174
> 
> v=0
> o=mhandley 29739 7272939 IN IP4 126.5.4.3
> s=SIP Call
> t=3149328700 0
> c=IN IP4 135.180.130.88
> m=audio 492170 RTP/AVP 0 12
> m=video 3227 RTP/AVP 31
> a=rtpmap:31 LPC


I'm not sure I agree with the behavior here. If the proxy is not
company.com, it may simple forward the request to company.com without
stripping the Route header. I see nothing wrong with that.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From sip-admin@lists.bell-labs.com  Tue Nov 14 02:40:06 2000
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From: "Hisham Khartabil" <hisham.khartabil@hotsip.com>
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Subject: RE: [SIP] Gateway and Redirected Number Question
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Date: Tue, 14 Nov 2000 09:40:42 +0200
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Check this out

http://www.cs.columbia.edu/sip/drafts_pstn.html

there are a number of internet drafts there.

Regards,
Hisham Khartabil
Hotsip  Finland

-----Original Message-----
From: sip-admin@lists.bell-labs.com [mailto:sip-admin@lists.bell-labs.com]On
Behalf Of Pete Kazmier
Sent: Monday, 13 November 2000 6:33 PM
To: sip@lists.bell-labs.com
Subject: [SIP] Gateway and Redirected Number Question

If a PSTN originated call is redirected to a gateway, how would, for
example, a Q.931 redirected number or SS7 RDN/OCN be mapped into a SIP
INVITE message?

I've been looking through various drafts to determine the appropriate
action but I can't seem to find the right one.  Could somone point me
in the right direction?

Thanks.

--
Peter Kazmier                                 http://www.kazmier.com
PGP Fingerprint   4FE7 8DA3 D0B5 9CAA 69DC  7243 1855 BC2E 4B43 5654

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Henning:
> Since To and From are used to match call legs, they can't be =
replicated.
> (Besides minor problems like breaking every single existing SIP
> implementation...) Unlike for email, which can easily be copied, SIP
> call legs need a single destination. SMTP, after all, also does two
> transactions if you put in two different To fields.
>
>
>
> > Since this information only affects the display of call requests and =
reply
> > functionality I don't see the difficulty in implementing this. What
> problems can you
> > see? (I havn't though about the implications for comparing requests =
to
> see if they
> > are the same).
>
> That's one of the complications.
>
> SIP is not email, so doing a plain carry-over of headers is not likely
> to work well.
>
> Adding informational headers that carry no protocol implications is
> relatively harmless. For example, "Reply-To" might be a way to =
indicate
> addresses for possible return calls.

The Also-To: header would be purely informational - it would indicate to =
the UAS what
other UASs had been invited. The meaning of Reply-To: is different from =
From: - and
this difference could be just as useful for IMs and voice calls as it is =
for e-mail.
IMs in particular, due to their close similarity to e-mail, should =
retain the same
features users are used to with e-mail. My question is why should users =
lose this
functionality for either IMs or voice calls?

Simon Barber

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Hi,

This is something I've also been looking into.=20

One way would be to add a new header (e.g. Also-From). That header could
be inserted by a proxy that does know the request may end up in a
gateway. The UA may not know that, so he could put whatever string in
the user part of the From header (e.g. his e-mail address). If the proxy
then knows the UAs telephone number, for example by using some kind of
database lookup, which is needed by the gateway - for any possible
A-number procedure - the proxy can add that to the request.

Another way would be to just define a URI parameter for the From header,
but the bad thing about that is that it can not be inserted by a proxy
if it was not in the request from the UA (since the From header must not
be changed) by a proxy, or any other intermediate node.

Note, that this response is according to Clive's mail. I do NOT support
the idea of ALSO-TO headers, multiple FROM/TO headers and whatever was
proposed in the previous mails :)

Regards,

Christer Holmberg
Ericsson Finland

clive.dellard@bt.com wrote:
>=20
> I agree with what you say, however, if you consider a call originating =
from
> a SIP device (telephone or softphone) that ends up going through a =
gateway
> to the PSTN then it is a different story.
> Neither the user or the UAC can know where a call will end up, =
therefore the
> ability to be able to provide both SIP address and SIP Telephone =
number
> would be useful to providing a complete service as then an appropriate
> calling address is available wherever the call ends up. In the =
scenario I
> described the gateway could chose the Telephone number (ok I know =
there are
> verification issues but that's another story).
> Something like Simon's idea of an "ALSO-FROM" seems a good idea to me =
(this
> is similar to "two number delivery" in the ISDN CLIP service) older =
versions
> would just ignore the ALSO-FROM and it wouldn't affect the =
significance of
> the FROM and TO headers.
>=20
> Clive
>=20
> > -----Original Message-----
> > From: Jonathan Rosenberg [SMTP:jdrosen@dynamicsoft.com]
> > Sent: Wednesday, October 11, 2000 7:41 PM
> > To:   'clive.dellard@bt.com'; 'sip@lists.bell-labs.com'
> > Subject:      RE: [SIP] From header to ISUP CgPN mapping
> >
> >
> >
> >
> > > -----Original Message-----
> > > From: clive.dellard@bt.com [mailto:clive.dellard@bt.com]
> > > Sent: Wednesday, October 11, 2000 12:17 PM
> > > To: sip@lists.bell-labs.com
> > > Subject: [SIP] From header to ISUP CgPN mapping
> > >
> > >
> > > In the call flows document draft-ietf-sip-call-flows-01.txt,
> > > scenario 4.1.1,
> > > the message detail and text show User A using a SIP telephone
> > > number rather
> > > than a SIP address in the From header. The text at the
> > > beginning of section
> > > 4 also says "that User A still uses his/her SIP URL in the
> > > Contact header,
> > > since the call could be redirected back to the SIP network."
> > > This implies that User A has pre-knowledge that the call is
> > > going to route
> > > to the PSTN and has the ability to chose the address to be
> > > used in the From
> > > header.
> >
> > No.
> >
> > The From header is the logical identity of the user making the call. =
Since
> > the call is coming through a gateway, that user is some user on a =
PSTN
> > terminal somewhere. The identity in this case can either be a tel =
URL, or
> > a
> > SIP URL at the domain of the originating network.
> >
> > The Contact is used for signaling addressing. Its not a logical
> > identifier.
> > It answers the question "where should I send SIP messages to for the =
rest
> > of
> > this call". The SIP spec says this is supposed to be a SIP URL =
preferably
> > with a complete hostname, so that there are no ambiguities about =
where to
> > send requests.
> >
> > Neither of these have anything to do with knowing where the call is =
going.
> >
> > -Jonathan R.
> >
> > ---
> > Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> > Chief Scientist                             First Floor
> > dynamicsoft                                 East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> > http://www.dynamicsoft.com
>=20
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
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From sip-admin@lists.bell-labs.com  Tue Nov 14 04:10:06 2000
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Hi,

I have a question about the CSeq number value.


Consider the following case:

A UAC sends a request to a UAS using CSeq=1.

The UAC then sends another request to the same UAS using CSeq=2.
However, for some reason the UAS never gets the request. It may, for
example, be terminated by an intermediate proxy, sending a response to
it, or it may be cancelled before it reaches the UAS.

After that, the UAC sends a third request, using CSeq=3, to the UAS.

So, the UAS will receive two requests, the first and the third, but not
the second.

Chapter 6.20 in the RFCbis says that the CSeq number must be contiguous.
However, in this case the UAS will never receive the second request.
Should it "wait" for it (it will never arrive...) or what is the purpose
of the line in the RFCbis?

Regards,

Christer Holmberg
Ericsson Finland
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From sip-admin@lists.bell-labs.com  Tue Nov 14 04:12:30 2000
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Subject: Re: [SIP] Session-timer with INFO
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Hi,


>>Hi,
>>
>>First, this mail is NOT a proposal for a change. I would just like to
>>hear some opinions - or maybe it has already been discussed on the =
list?
>>
>>The INFO Method RFC says that for INFO requests without a message =
body,
>>a server supporting INFO MUST respond with response code 200. So, my
>>question is: why couldn't INFO be used for the session-timer
>>functionality? Is the ACK the (re-)INVITE provides really necessary =
for
>>the purpose?
>>
>>And, like the session-timer is defined now, the other party would not
>>even have to support the INFO method, because a 501 Not Implemented
>>would be sent, and the sender of the INFO would know the session is
>>"alive".
>>
>>Just some thoughts... Please let me know if there is something I =
haven't
>>taken into consideration.
>>
>=20
>I think the re-INVITE mechanism is far more better because the =
>information about what session is refreshed is carried in the SDP of =
the
>re-INVITE. The INFO approach has the significant disadvantage that =
>proxies that use the session timer might have to store a *lot* of state
>information about each session.

That is not true. The session is identified using the Call-ID.

In fact, another advantage of using INFO is that you don't have to send
a message body at all. Using re-INVITE, you must send a copy of the
original message, and the UAS must check if the SDP is a copy
(session-timer) or if it is a new SDP, when the session should be
modified. Again, we save resources. Comments?

Regards,

Christer Holmberg
Ericsson Finland



>=20
> Regards /Lars
>=20
> > Regards,
> >
> > Christer Holmberg
> > Ericsson Finland
>=20
> Lars Berggren       <mailto:lars.berggren@intertex.se>
> Intertex Data AB    tel: +46-8-6282828
> Sundbyberg, Sweden  fax: +46-8-6286414
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From sip-admin@lists.bell-labs.com  Tue Nov 14 04:15:35 2000
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Hi,

I found the archive, and the thread. The discussion was very much about
the fact that by using re-INVITEs the session can be "automatically"
re-established if the UAS has booted, and I guess this works fine as
long as we don't get any 18x responses, which may include ringtones,
messages and who knows what (we may even receive a 3xx Redirect message)
And, if the UAC uses INFO, and receive 481 Call leg/Transaction Does Not
Exist (e.g. if the UAS has booted), it can choose to send a re-INVITE,
and be prepared to receive 18x messages, to reestablish the session.

Also, I don't think the UAS will crash that often, compared to what we
would save by not having to send ACKs etc.

Regards,

Christer Holmberg
Ericsson Finland


Anders Kristensen wrote:
>=20
> We had this whole discussion about a year ago. Unfortunately the
> archives doesn't seem to go that far back, but it seems pointless to
> repeat that discussion. Is there an archive somewhere for the old =
list?
>=20
> Anders
>=20
> Christer Holmberg wrote:
> >
> > Hi,
> >
> > Thanks for the comments!
> >
> > >If the UA crashed and rebooted and forgot about the old session, it =
would
> > >respond with either a 481 call leg doesn't exist (you would know =
the call
> > >is down), or a 501 (you know the UA is up, but is the call?)
> > >
> > >thanks,
> > >-rohan
> >
> > I think that would be good, because then we could inform the user =
that
> > the call doesn't exist anymore.
> >
> > Let's assume the UAS crashes, reboots and the UAC sends the
> > session-timer re-INVITE. If the UAS has forgotten about the old =
session
> > it would consider the re-INVITE as a new INVITE, for a new session. =
It
> > would try to establish a new session - and we could suddenly start
> > receiving provisional 18x responses (with or without SDP bodies), =
and/or
> > a 200 response with a SDP body, to the session-timer re-INVITE. What =
do
> > we do then?
> >
> > Regards,
> >
> > Christer Holmberg
> > Ericsson Finland
> >
> > >
> > > At 10:55 AM 10/16/00 , Christer Holmberg wrote:
> > > >
> > > >Hi,
> > > >
> > > >First, this mail is NOT a proposal for a change. I would just =
like to
> > > >hear some opinions - or maybe it has already been discussed on =
the list?
> > > >
> > > >The INFO Method RFC says that for INFO requests without a message =
body,
> > > >a server supporting INFO MUST respond with response code 200. So, =
my
> > > >question is: why couldn't INFO be used for the session-timer
> > > >functionality? Is the ACK the (re-)INVITE provides really =
necessary for
> > > >the purpose?
> > > >
> > > >And, like the session-timer is defined now, the other party would =
not
> > > >even have to support the INFO method, because a 501 Not =
Implemented
> > > >would be sent, and the sender of the INFO would know the session =
is
> > > >"alive".
> > > >
> > > >Just some thoughts... Please let me know if there is something I =
haven't
> > > >taken into consideration.
> > > >
> > > >Regards,
> > > >
> > > >Christer Holmberg
> > > >Ericsson Finland
> > > >
>=20
> --
> Anders Kristensen
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From sip-admin@lists.bell-labs.com  Tue Nov 14 04:17:54 2000
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Hi,

>I actually agree with you (except I would have used a new method rather
>than INFO).

That is of course an option. I just thought that INFO, with no message
body, would work fine, since
the only thing the UAS has to do is to send a response (no communication
with the user/application is required).

>I just don't think it's very likely, given that the arguments are =
largely unchanged, that people will be >willing to modify the session =
timer mechanism at this late stage.

There are different ways of doing it. IF you would choose to change the
current specification, basically the only thing you would have to do is
to change the re-INVITEs to INFOs (or whatever method you choose to
use). The negotiation mechanism of the timer value, the extension
headers etc. would remain. Of course, a UA who wishes to send INFOs
would also have to support the method.

Another way would be to define a new mechanism - "session ping",
"session-timer light", or whatever you want to call it :)

>Anyway, maybe you should post the archive pointer and then we can at =
least have a fully informed discussion.

The link to the archieve is...=20

http://www.bell-labs.com/mailing-lists/sip/20000/subject.html

...and the subject is 'Session timer comments'.


Regards,

Christer Holmberg
Ericsson Finland



>=20
> Anders
>=20
> Christer Holmberg wrote:
> >
> > Hi,
> >
> > I found the archive, and the thread. The discussion was very much =
about
> > the fact that by using re-INVITEs the session can be "automatically"
> > re-established if the UAS has booted, and I guess this works fine as
> > long as we don't get any 18x responses, which may include ringtones,
> > messages and who knows what (we may even receive a 3xx Redirect =
message)
> > And, if the UAC uses INFO, and receive 481 Call leg/Transaction Does =
Not
> > Exist (e.g. if the UAS has booted), it can choose to send a =
re-INVITE,
> > and be prepared to receive 18x messages, to reestablish the session.
> >
> > Also, I don't think the UAS will crash that often, compared to what =
we
> > would save by not having to send ACKs etc.
> >
> > Regards,
> >
> > Christer Holmberg
> > Ericsson Finland
> >
> > Anders Kristensen wrote:
> > >
> > > We had this whole discussion about a year ago. Unfortunately the
> > > archives doesn't seem to go that far back, but it seems pointless =
to
> > > repeat that discussion. Is there an archive somewhere for the old =
list?
> > >
> > > Anders
> > >
> > > Christer Holmberg wrote:
> > > >
> > > > Hi,
> > > >
> > > > Thanks for the comments!
> > > >
> > > > >If the UA crashed and rebooted and forgot about the old =
session, it would
> > > > >respond with either a 481 call leg doesn't exist (you would =
know the call
> > > > >is down), or a 501 (you know the UA is up, but is the call?)
> > > > >
> > > > >thanks,
> > > > >-rohan
> > > >
> > > > I think that would be good, because then we could inform the =
user that
> > > > the call doesn't exist anymore.
> > > >
> > > > Let's assume the UAS crashes, reboots and the UAC sends the
> > > > session-timer re-INVITE. If the UAS has forgotten about the old =
session
> > > > it would consider the re-INVITE as a new INVITE, for a new =
session. It
> > > > would try to establish a new session - and we could suddenly =
start
> > > > receiving provisional 18x responses (with or without SDP =
bodies), and/or
> > > > a 200 response with a SDP body, to the session-timer re-INVITE. =
What do
> > > > we do then?
> > > >
> > > > Regards,
> > > >
> > > > Christer Holmberg
> > > > Ericsson Finland
> > > >
> > > > >
> > > > > At 10:55 AM 10/16/00 , Christer Holmberg wrote:
> > > > > >
> > > > > >Hi,
> > > > > >
> > > > > >First, this mail is NOT a proposal for a change. I would just =
like to
> > > > > >hear some opinions - or maybe it has already been discussed =
on the list?
> > > > > >
> > > > > >The INFO Method RFC says that for INFO requests without a =
message body,
> > > > > >a server supporting INFO MUST respond with response code 200. =
So, my
> > > > > >question is: why couldn't INFO be used for the session-timer
> > > > > >functionality? Is the ACK the (re-)INVITE provides really =
necessary for
> > > > > >the purpose?
> > > > > >
> > > > > >And, like the session-timer is defined now, the other party =
would not
> > > > > >even have to support the INFO method, because a 501 Not =
Implemented
> > > > > >would be sent, and the sender of the INFO would know the =
session is
> > > > > >"alive".
> > > > > >
> > > > > >Just some thoughts... Please let me know if there is =
something I haven't
> > > > > >taken into consideration.
> > > > > >
> > > > > >Regards,
> > > > > >
> > > > > >Christer Holmberg
> > > > > >Ericsson Finland
> > > > > >
> > >
> > > --
> > > Anders Kristensen
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Hi,

If I send you an INFO, or any other method that you don't support, you
can for example send a 501 Not Implemented response, and include an
Allow header with the methods you support. Of course, you can't stop the
UAC of sending the method again, if he chooses to...

You can not, however, tell an UAC not to send you re-INVITEs, because
then you would also forbid the UAC to send you INVITEs - and I don't
think you want to do that. Also, the specification says that a UA MUST
be able to receive INVITEs (which included re-INVITEs), so...

Regards,

Christer Holmberg
Ericsson Finland


hammer michael wrote:
>=20
> Is this the Internet version of "let them eat cake?"  Signal-spam is =
not
> more palatable than the normal variety.
>=20
> Is there a response code that can indicate the sentiment:  I don't =
support
> this method, don't try it again?  It might also be good to make sure =
that
> such undesirable sub-methods are indicated as "Unsupported."  If that
> sub-method is well-defined, then that header parameter should be =
applicable.
>=20
> Mike
>=20
> At 12:30 AM 10/17/2000 +0300, Christer Holmberg wrote:
>=20
> >Hi,
> >
> > >Good points. Here's one more problem you may want to consider.
> > >Whenever you invoke a ping-style method of deciding if a session is
> > >alive, you open up problems due to network jitter and bandwidth. =
For
> > >instance, you could easily have a packet delayed at an endpoint due =
to
> > >a large file transfer starting up, or some other bandwidth =
consuming
> > >event. In this case, your ping could be delayed long enough to =
cause
> > >the session to drop for no good reason.
> >
> >Well, this could be the case no matter which method (INVITE, INFO, or
> >whatever...) you use in the heartbeat request...
> >
> > >You're also creating the potential for artificial problems in =
wireless
> > >networks by using a ping-method to keep a session alive. During a
> > >handover there may be a period where packets are in effect "muted" =
for
> > >a short period of time. The mute period may not, by itself be long
> > >enough to trigger a ping-timer to pop and kill the session, but if =
the
> > >ping was near the edge of the time envelope to be accepted as
> > >"on-time" anyhow, once again, your session could very easily drop =
when
> > >the media was operating just fine.
> > >Also, keep in mind, the longer you set the ping timer for, the more
> > >network resources you will likely waste over time because of the
> > >additional latentcy in discovering a "dead" session.
> >
> >Just to clarify, this "ping-method" IS already defined for SIP
> >(draft-ietf-sip-session-timer-02.txt), and re-INVITEs are used to =
ping.
> >My idea was about replacing the re-INVITE with INFO, which mean we =
would
> >actually save resources since we don't have to send ACK every time.
> >
> > >This method has one other nasty side-effect. You'd be wasting =
network
> > >resources on an event that will likely not occur very often. After
> > >all, every INFO exchange except the very last one that invalidates =
the
> > >session could have very well not been sent without any negative =
impact
> > >on the session (otherwise they would have been the last one). So
> > >you've got proxies and UAs, and other transport resources being =
tied
> > >up processing packets that by and large are providing no value, and
> > >they're chewing up bandwidth budget that other services could be =
using
> > >instead.
> >
> >That is an interesting point, and it is valid also for a number of =
other
> >SIP extensions where extra messages are sent (e.g. PRACK), but it is
> >another discussion.
> >
> > >Please keep in mind that not everyone is looking at implementing =
SIP
> > >on a 10Mbit+ hardwired LAN (or has to scale up to tremendous =
numbers
> > >of users), pinging and triangle-routing can be the death-knell of
> > >these sorts of networks.
> >
> >Who said "bandwidth is free"? :)
> >
> > >Because of all of this, I think it would be a very bad idea for a =
UA
> > >to send an INFO to a client who repeatedly tells him "I don't
> > >understand INFO" to see if he's alive.
> >
> >The number of messages on the network will be the same no matter if =
we
> >send an INVITE which he understands, or an INFO he may not =
understand.
> >Also, when we receive the response for the INVITE we also have to =
send
> >back an ACK, which is not the case when INFO is used.
> >
> > >He may not understand INFO because that particular network doesn't
> > deem >the bandwidth required for this mechanism as being best used =
for
> > that >purpose. If someone tells you "don't send me this method" then
> > don't send >it to them.
> >
> >Again, this has nothing to do with the bandwith issue. Even if I send
> >re-INVITEs, which the other party understands, he can not tell me not =
to
> >send any more re-INVITEs.
> >
> > >Otherwise your implementation may be viewed as favorably as the =
record
> > >clubs that used to require you to send them a card every month =
telling
> > >them not to send you anything by some groups.
> >
> >Well, it's not up to me to decide when and if an operator will use =
the
> >session-timer, as defined in the draft, or not - I am just trying to
> >make it as good (and less resource consuming) if he choses to.
> >
> >Regards,
> >
> >Christer Holmberg
> >Ericsson Finland
> >
> >
> > >
> > > Brian Stucker
> > > Nortel Networks
> > >
> > > -----Original Message-----
> > > From: Culpepper, Bert [mailto:bert.culpepper@intervoice-brite.com]
> > > Sent: Monday, October 16, 2000 3:38 PM
> > > To: Christer Holmberg; sip@lists.bell-labs.com
> > > Cc: Rohan Mahy
> > > Subject: RE: [SIP] Session-timer with INFO
> > >
> > > Some things to think about.
> > >
> > > I believe INFO is _not_ to be used to modify the state of a =
session.
> > > I
> > > suppose a debate is possible about session-timer affecting session
> > > state - present vs. future and if INFO is only restricted to one.  =
But
> > > it's
> > > probably best to leave it alone.  Also, the INV-200-ACK exchange
> > > allows multiple parties to agree on the duration of the session.  =
If
> > > the
> > > recipient of the INFO request wants a smaller value (because its
> > > heartbeat interval is shorter) for the session timer it would have =
to
> > > send its own INFO request (unless you plan to keep the proposed
> > > Session-expires header and its use in a 200 OK).  Now four
> > > messages are needed.  In addition, there may be proxy issues when
> > > using INFO.  There's other behavior described in the draft too.
> > >
> > > > -----Original Message-----
> > > > From: Christer Holmberg =
[mailto:christer.holmberg@lmf.ericsson.se]
> > > > Sent: Monday, October 16, 2000 3:42 PM
> > > > To: sip@lists.bell-labs.com
> > > > Cc: Rohan Mahy
> > > > Subject: Re: [SIP] Session-timer with INFO
> > > >
> > > >
> > > >
> > > > Hi,
> > > >
> > > > Thanks for the comments!
> > > >
> > > > >If the UA crashed and rebooted and forgot about the old =
session, it
> > >
> > > > would
> > > > >respond with either a 481 call leg doesn't exist (you would =
know
> > > the
> > > > call
> > > > >is down), or a 501 (you know the UA is up, but is the call?)
> > > > >
> > > > >thanks,
> > > > >-rohan
> > > >
> > > > I think that would be good, because then we could inform the =
user
> > > that
> > > > the call doesn't exist anymore.
> > > >
> > > > Let's assume the UAS crashes, reboots and the UAC sends the
> > > > session-timer re-INVITE. If the UAS has forgotten about the old
> > > > session
> > > > it would consider the re-INVITE as a new INVITE, for a new =
session.
> > > It
> > > > would try to establish a new session - and we could suddenly =
start
> > > > receiving provisional 18x responses (with or without SDP =
bodies),
> > > > and/or
> > > > a 200 response with a SDP body, to the session-timer re-INVITE. =
What
> > >
> > > > do
> > > > we do then?
> > >
> > > Well, you could ignore the provisional responses (you know the =
other
> > > end
> > > is confused) and let the session get re-established or decide to
> > > terminate
> > > the session with CANCEL/BYE.
> > >
> > > Regards,
> > > Bert
> > >
> > > >
> > > > Regards,
> > > >
> > > > Christer Holmberg
> > > > Ericsson Finland
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > >
> > > > > At 10:55 AM 10/16/00 , Christer Holmberg wrote:
> > > > > >
> > > > > >Hi,
> > > > > >
> > > > > >First, this mail is NOT a proposal for a change. I would just
> > > like
> > > > to
> > > > > >hear some opinions - or maybe it has already been discussed =
on
> > > the
> > > > list?
> > > > > >
> > > > > >The INFO Method RFC says that for INFO requests without a =
message
> > >
> > > > body,
> > > > > >a server supporting INFO MUST respond with response code 200. =
So,
> > >
> > > > my
> > > > > >question is: why couldn't INFO be used for the session-timer
> > > > > >functionality? Is the ACK the (re-)INVITE provides really
> > > necessary
> > > > for
> > > > > >the purpose?
> > > > > >
> > > > > >And, like the session-timer is defined now, the other party =
would
> > >
> > > > not
> > > > > >even have to support the INFO method, because a 501 Not
> > > Implemented
> > > > > >would be sent, and the sender of the INFO would know the =
session
> > > is
> > > > > >"alive".
> > > > > >
> > > > > >Just some thoughts... Please let me know if there is =
something I
> > > > haven't
> > > > > >taken into consideration.
> > > > > >
> > > > > >Regards,
> > > > > >
> > > > > >Christer Holmberg
> > > > > >Ericsson Finland
> > > > > >
> > > >
> > >
> > > _______________________________________________
> > > SIP mailing list
> > > SIP@lists.bell-labs.com
> > > http://lists.bell-labs.com/mailman/listinfo/sip
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Subject: [SIP] SIP and T.38 fax calls - Internet Draft
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A new Internet-Draft has been submitted, dealing with real time fax calls
and SIP: call flows and BCP.

Comments and suggestions are very much appreciated.
Regards,
Jean-Francois Mule


-----Original Message-----
From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
Sent: Tuesday, October 17, 2000 3:47 AM
Subject: I-D ACTION:draft-mule-sip-t38callflows-00.txt


A New Internet-Draft is available from the on-line Internet-Drafts
directories.


	Title		: SIP T.38 Call Flow Examples And Best Current
Practice
	Author(s)	: J. Mule
	Filename	: draft-mule-sip-t38callflows-00.txt
	Pages		: 19
	Date		: 16-Oct-00

This document gives examples of SIP call flows for T.38 Internet fax
communications (SIP, the Session Initiation Protocol is defined in
RFC2543 [2] and T.38 is an ITU-T Recommendation [3]).  Elements in
these call flows include SIP User Agents, SIP Proxy Servers, and
Gateways to the PSTN (Public Switch Telephone Network).
This document introduces best current practices for T.38 fax calls:
a call starts with audio capabilities, and, upon fax tone detection,
T.38 fax capabilities are negotiated.  The T.38 fax call scenarios
include the detection of T.38 fax transmission by the receiving
side, or the emitting side, or both (in the latter case, a 'glare'
effect may appear).  The current version of this document only
covers the case when the fax tone is detected by the receiving side
(other cases were left for discussion and may be included in the
future).  Call flow diagrams and message details are shown.  A list
of IANA defined SDP attribute names for T.38 is summarized in
section 5.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-mule-sip-t38callflows-00.txt

Internet-Drafts are also available by anonymous FTP. Login with the
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Hi,

Using the Diversion header, you can inform an endpoint of which
Request-URIs have been used before the Request-URI of that specific
endpoint was used.

The case I am talking about, however, is when we can insert an "alias"
header for the From header, using a telephone number, if the From header
is, for example, an e-mail address. Using this new header, let us call
is Also-From, a proxy (or the calling endpoint itself) may also insert a
telephone number, which can then be used by PSTN Gateways for any kind
of A-number analysing. One could argue that the caller should always
include a telephone number in the From header, but the caller may not
know that the call will end up on the PSTN side, so...

I DO like the Diversion draft very much, though, but it doesn't solve
this specific problem :)

Regards,

Christer Holmberg
Ericsson Finland






Bryan Byerly wrote:
>=20
> Hi guys,
>=20
> What do you guys think about using the Diversion header
> (http://www.ietf.org/internet-drafts/draft-levy-sip-diversion-00.txt) =
instead of
> Also-From?
> It seems like a natural replacement to me.
>=20
> Thanks!
> Bryan
>=20
> Christer Holmberg wrote:
>=20
> > Hi,
> >
> > This is something I've also been looking into.
> >
> > One way would be to add a new header (e.g. Also-From). That header =
could
> > be inserted by a proxy that does know the request may end up in a
> > gateway. The UA may not know that, so he could put whatever string =
in
> > the user part of the From header (e.g. his e-mail address). If the =
proxy
> > then knows the UAs telephone number, for example by using some kind =
of
> > database lookup, which is needed by the gateway - for any possible
> > A-number procedure - the proxy can add that to the request.
> >
> > Another way would be to just define a URI parameter for the From =
header,
> > but the bad thing about that is that it can not be inserted by a =
proxy
> > if it was not in the request from the UA (since the From header must =
not
> > be changed) by a proxy, or any other intermediate node.
> >
> > Note, that this response is according to Clive's mail. I do NOT =
support
> > the idea of ALSO-TO headers, multiple FROM/TO headers and whatever =
was
> > proposed in the previous mails :)
> >
> > Regards,
> >
> > Christer Holmberg
> > Ericsson Finland
> >
> > clive.dellard@bt.com wrote:
> > >
> > > I agree with what you say, however, if you consider a call =
originating from
> > > a SIP device (telephone or softphone) that ends up going through a =
gateway
> > > to the PSTN then it is a different story.
> > > Neither the user or the UAC can know where a call will end up, =
therefore the
> > > ability to be able to provide both SIP address and SIP Telephone =
number
> > > would be useful to providing a complete service as then an =
appropriate
> > > calling address is available wherever the call ends up. In the =
scenario I
> > > described the gateway could chose the Telephone number (ok I know =
there are
> > > verification issues but that's another story).
> > > Something like Simon's idea of an "ALSO-FROM" seems a good idea to =
me (this
> > > is similar to "two number delivery" in the ISDN CLIP service) =
older versions
> > > would just ignore the ALSO-FROM and it wouldn't affect the =
significance of
> > > the FROM and TO headers.
> > >
> > > Clive
> > >
> > > > -----Original Message-----
> > > > From: Jonathan Rosenberg [SMTP:jdrosen@dynamicsoft.com]
> > > > Sent: Wednesday, October 11, 2000 7:41 PM
> > > > To:   'clive.dellard@bt.com'; 'sip@lists.bell-labs.com'
> > > > Subject:      RE: [SIP] From header to ISUP CgPN mapping
> > > >
> > > >
> > > >
> > > >
> > > > > -----Original Message-----
> > > > > From: clive.dellard@bt.com [mailto:clive.dellard@bt.com]
> > > > > Sent: Wednesday, October 11, 2000 12:17 PM
> > > > > To: sip@lists.bell-labs.com
> > > > > Subject: [SIP] From header to ISUP CgPN mapping
> > > > >
> > > > >
> > > > > In the call flows document draft-ietf-sip-call-flows-01.txt,
> > > > > scenario 4.1.1,
> > > > > the message detail and text show User A using a SIP telephone
> > > > > number rather
> > > > > than a SIP address in the From header. The text at the
> > > > > beginning of section
> > > > > 4 also says "that User A still uses his/her SIP URL in the
> > > > > Contact header,
> > > > > since the call could be redirected back to the SIP network."
> > > > > This implies that User A has pre-knowledge that the call is
> > > > > going to route
> > > > > to the PSTN and has the ability to chose the address to be
> > > > > used in the From
> > > > > header.
> > > >
> > > > No.
> > > >
> > > > The From header is the logical identity of the user making the =
call. Since
> > > > the call is coming through a gateway, that user is some user on =
a PSTN
> > > > terminal somewhere. The identity in this case can either be a =
tel URL, or
> > > > a
> > > > SIP URL at the domain of the originating network.
> > > >
> > > > The Contact is used for signaling addressing. Its not a logical
> > > > identifier.
> > > > It answers the question "where should I send SIP messages to for =
the rest
> > > > of
> > > > this call". The SIP spec says this is supposed to be a SIP URL =
preferably
> > > > with a complete hostname, so that there are no ambiguities about =
where to
> > > > send requests.
> > > >
> > > > Neither of these have anything to do with knowing where the call =
is going.
> > > >
> > > > -Jonathan R.
> > > >
> > > > ---
> > > > Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> > > > Chief Scientist                             First Floor
> > > > dynamicsoft                                 East Hanover, NJ =
07936
> > > > jdrosen@dynamicsoft.com                     FAX:   (973) =
952-5050
> > > > http://www.cs.columbia.edu/~jdrosen         PHONE: (973) =
952-5000
> > > > http://www.dynamicsoft.com
> > >
> > > _______________________________________________
> > > SIP mailing list
> > > SIP@lists.bell-labs.com
> > > http://lists.bell-labs.com/mailman/listinfo/sip
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Hi,

>>>Thus you are in support of not trying to implicitly use a method
>>>to do what a subordinate capability should be explicitly defined to =
do?
>>
>>I didn't say that. Of course it is nice if we can
>>get-two-things-by-doing-one, but in this case I just see some issues.
>>Again, first there is the thing that by using re-INVITEs we will
>>generate ACKs.
>>
>>Second, which I think is related to your mail, is when the UAS boots,
>>and the re-INVITE is seen as a new INVITE to establish a new session. =
We
>>may get 18x responses before the final response, and the final =
response
>>can be a 3xx, 4xx, 5xx or 6xx response, and in some cases it may be =
very
>>difficult for the UAC to behave in a good way. If we use INFO, and
>>receive 481, the UAS knows that the session doesn't exist anymore, and
>>can do whatever is needed, and then, if wanted, send a new INVITE to
>>re-establish the call. It gives the UAC more control, and the UAC may,
>>for some reason, even want to know if a session died.
>I don't believe this is the scenario that is addressed by the goal of =
>being able to re-establish the session.  Remember that the entity =
>managing the SIP session information and the entity participating in =
the >RTP session do not need to be the same machine.  So the scenario is =
that >the SIP UAS has lost track of the SIP state related of an RTP =
session, >but the RTP session is still active.  The re-INVITE would =
allow the SIP >UAS to re-establish SIP state for the session without a =
new call being >set up.  In this scenario, it would not send 18x, 3xx, =
4xx, 5xx or 6xx >responses.

Maybe not in this specific scenario, but it MAY happen depending of the
architecture and logic of the remote party. I would like the heartbeat
mechanism, and the logic to support it, to be as simple as possible,
i.e. you send a request and receive a response. Then, depending on the
response, the UAC can choose to do whatever he wishes to. In some
systems, for example, he may choose to use another UAS, or whatever, if
the session has died.
=20
>While it is true that there are other ways that this could have been =
>solved, it is pretty late in the game to make this kind of a change.  =
>Especially if the only reason for doing so is to avoid the ACK message.

Well, as has been discussed here, that is not the only reason, but I
still think it is a good reason.

>As a side note, we are in the process of creating a new version of the =
>draft that we hope will be used for working group last call.

One option, which has been proposed, was to define a new functionality,
"session ping", e.g. by using the INFO method extension.=20


Regards,

Christer Holmberg
Ericsson Finland


>=20
> Steve
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Hi,

>Thus you are in support of not trying to implicitly use a method to do =
>what a subordinate capability should be explicitly defined to do?

I didn't say that. Of course it is nice if we can
get-two-things-by-doing-one, but in this case I just see some issues.
Again, first there is the thing that by using re-INVITEs we will
generate ACKs.=20

Second, which I think is related to your mail, is when the UAS boots,
and the re-INVITE is seen as a new INVITE to establish a new session. We
may get 18x responses before the final response, and the final response
can be a 3xx, 4xx, 5xx or 6xx response, and in some cases it may be very
difficult for the UAC to behave in a good way. If we use INFO, and
receive 481, the UAS knows that the session doesn't exist anymore, and
can do whatever is needed, and then, if wanted, send a new INVITE to
re-establish the call. It gives the UAC more control, and the UAC may,
for some reason, even want to know if a session died.


Regards,

Christer Holmberg
Ericsson Finland

>=20
> Mike
>=20
> At 05:00 PM 10/17/2000 +0300, Christer Holmberg wrote:
>=20
> >Hi,
> >
> >If I send you an INFO, or any other method that you don't support, =
you
> >can for example send a 501 Not Implemented response, and include an
> >Allow header with the methods you support. Of course, you can't stop =
the
> >UAC of sending the method again, if he chooses to...
> >
> >You can not, however, tell an UAC not to send you re-INVITEs, because
> >then you would also forbid the UAC to send you INVITEs - and I don't
> >think you want to do that. Also, the specification says that a UA =
MUST
> >be able to receive INVITEs (which included re-INVITEs), so...
> >
> >Regards,
> >
> >Christer Holmberg
> >Ericsson Finland
> >
> >
> >hammer michael wrote:
> > >
> > > Is this the Internet version of "let them eat cake?"  Signal-spam =
is not
> > > more palatable than the normal variety.
> > >
> > > Is there a response code that can indicate the sentiment:  I don't =
support
> > > this method, don't try it again?  It might also be good to make =
sure that
> > > such undesirable sub-methods are indicated as "Unsupported."  If =
that
> > > sub-method is well-defined, then that header parameter should be
> > applicable.
> > >
> > > Mike
> > >
> > > At 12:30 AM 10/17/2000 +0300, Christer Holmberg wrote:
> > >
> > > >Hi,
> > > >
> > > > >Good points. Here's one more problem you may want to consider.
> > > > >Whenever you invoke a ping-style method of deciding if a =
session is
> > > > >alive, you open up problems due to network jitter and =
bandwidth. For
> > > > >instance, you could easily have a packet delayed at an endpoint =
due to
> > > > >a large file transfer starting up, or some other bandwidth =
consuming
> > > > >event. In this case, your ping could be delayed long enough to =
cause
> > > > >the session to drop for no good reason.
> > > >
> > > >Well, this could be the case no matter which method (INVITE, =
INFO, or
> > > >whatever...) you use in the heartbeat request...
> > > >
> > > > >You're also creating the potential for artificial problems in =
wireless
> > > > >networks by using a ping-method to keep a session alive. During =
a
> > > > >handover there may be a period where packets are in effect =
"muted" for
> > > > >a short period of time. The mute period may not, by itself be =
long
> > > > >enough to trigger a ping-timer to pop and kill the session, but =
if the
> > > > >ping was near the edge of the time envelope to be accepted as
> > > > >"on-time" anyhow, once again, your session could very easily =
drop when
> > > > >the media was operating just fine.
> > > > >Also, keep in mind, the longer you set the ping timer for, the =
more
> > > > >network resources you will likely waste over time because of =
the
> > > > >additional latentcy in discovering a "dead" session.
> > > >
> > > >Just to clarify, this "ping-method" IS already defined for SIP
> > > >(draft-ietf-sip-session-timer-02.txt), and re-INVITEs are used to =
ping.
> > > >My idea was about replacing the re-INVITE with INFO, which mean =
we would
> > > >actually save resources since we don't have to send ACK every =
time.
> > > >
> > > > >This method has one other nasty side-effect. You'd be wasting =
network
> > > > >resources on an event that will likely not occur very often. =
After
> > > > >all, every INFO exchange except the very last one that =
invalidates the
> > > > >session could have very well not been sent without any negative =
impact
> > > > >on the session (otherwise they would have been the last one). =
So
> > > > >you've got proxies and UAs, and other transport resources being =
tied
> > > > >up processing packets that by and large are providing no value, =
and
> > > > >they're chewing up bandwidth budget that other services could =
be using
> > > > >instead.
> > > >
> > > >That is an interesting point, and it is valid also for a number =
of other
> > > >SIP extensions where extra messages are sent (e.g. PRACK), but it =
is
> > > >another discussion.
> > > >
> > > > >Please keep in mind that not everyone is looking at =
implementing SIP
> > > > >on a 10Mbit+ hardwired LAN (or has to scale up to tremendous =
numbers
> > > > >of users), pinging and triangle-routing can be the death-knell =
of
> > > > >these sorts of networks.
> > > >
> > > >Who said "bandwidth is free"? :)
> > > >
> > > > >Because of all of this, I think it would be a very bad idea for =
a UA
> > > > >to send an INFO to a client who repeatedly tells him "I don't
> > > > >understand INFO" to see if he's alive.
> > > >
> > > >The number of messages on the network will be the same no matter =
if we
> > > >send an INVITE which he understands, or an INFO he may not =
understand.
> > > >Also, when we receive the response for the INVITE we also have to =
send
> > > >back an ACK, which is not the case when INFO is used.
> > > >
> > > > >He may not understand INFO because that particular network =
doesn't
> > > > deem >the bandwidth required for this mechanism as being best =
used for
> > > > that >purpose. If someone tells you "don't send me this method" =
then
> > > > don't send >it to them.
> > > >
> > > >Again, this has nothing to do with the bandwith issue. Even if I =
send
> > > >re-INVITEs, which the other party understands, he can not tell me =
not to
> > > >send any more re-INVITEs.
> > > >
> > > > >Otherwise your implementation may be viewed as favorably as the =
record
> > > > >clubs that used to require you to send them a card every month =
telling
> > > > >them not to send you anything by some groups.
> > > >
> > > >Well, it's not up to me to decide when and if an operator will =
use the
> > > >session-timer, as defined in the draft, or not - I am just trying =
to
> > > >make it as good (and less resource consuming) if he choses to.
> > > >
> > > >Regards,
> > > >
> > > >Christer Holmberg
> > > >Ericsson Finland
> > > >
> > > >
> > > > >
> > > > > Brian Stucker
> > > > > Nortel Networks
> > > > >
> > > > > -----Original Message-----
> > > > > From: Culpepper, Bert =
[mailto:bert.culpepper@intervoice-brite.com]
> > > > > Sent: Monday, October 16, 2000 3:38 PM
> > > > > To: Christer Holmberg; sip@lists.bell-labs.com
> > > > > Cc: Rohan Mahy
> > > > > Subject: RE: [SIP] Session-timer with INFO
> > > > >
> > > > > Some things to think about.
> > > > >
> > > > > I believe INFO is _not_ to be used to modify the state of a =
session.
> > > > > I
> > > > > suppose a debate is possible about session-timer affecting =
session
> > > > > state - present vs. future and if INFO is only restricted to =
one.  But
> > > > > it's
> > > > > probably best to leave it alone.  Also, the INV-200-ACK =
exchange
> > > > > allows multiple parties to agree on the duration of the =
session.  If
> > > > > the
> > > > > recipient of the INFO request wants a smaller value (because =
its
> > > > > heartbeat interval is shorter) for the session timer it would =
have to
> > > > > send its own INFO request (unless you plan to keep the =
proposed
> > > > > Session-expires header and its use in a 200 OK).  Now four
> > > > > messages are needed.  In addition, there may be proxy issues =
when
> > > > > using INFO.  There's other behavior described in the draft =
too.
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Christer Holmberg =
[mailto:christer.holmberg@lmf.ericsson.se]
> > > > > > Sent: Monday, October 16, 2000 3:42 PM
> > > > > > To: sip@lists.bell-labs.com
> > > > > > Cc: Rohan Mahy
> > > > > > Subject: Re: [SIP] Session-timer with INFO
> > > > > >
> > > > > >
> > > > > >
> > > > > > Hi,
> > > > > >
> > > > > > Thanks for the comments!
> > > > > >
> > > > > > >If the UA crashed and rebooted and forgot about the old =
session, it
> > > > >
> > > > > > would
> > > > > > >respond with either a 481 call leg doesn't exist (you would =
know
> > > > > the
> > > > > > call
> > > > > > >is down), or a 501 (you know the UA is up, but is the =
call?)
> > > > > > >
> > > > > > >thanks,
> > > > > > >-rohan
> > > > > >
> > > > > > I think that would be good, because then we could inform the =
user
> > > > > that
> > > > > > the call doesn't exist anymore.
> > > > > >
> > > > > > Let's assume the UAS crashes, reboots and the UAC sends the
> > > > > > session-timer re-INVITE. If the UAS has forgotten about the =
old
> > > > > > session
> > > > > > it would consider the re-INVITE as a new INVITE, for a new =
session.
> > > > > It
> > > > > > would try to establish a new session - and we could suddenly =
start
> > > > > > receiving provisional 18x responses (with or without SDP =
bodies),
> > > > > > and/or
> > > > > > a 200 response with a SDP body, to the session-timer =
re-INVITE. What
> > > > >
> > > > > > do
> > > > > > we do then?
> > > > >
> > > > > Well, you could ignore the provisional responses (you know the =
other
> > > > > end
> > > > > is confused) and let the session get re-established or decide =
to
> > > > > terminate
> > > > > the session with CANCEL/BYE.
> > > > >
> > > > > Regards,
> > > > > Bert
> > > > >
> > > > > >
> > > > > > Regards,
> > > > > >
> > > > > > Christer Holmberg
> > > > > > Ericsson Finland
> > > > > >
> > > > > >
> > > > > >
> > > > > >
> > > > > >
> > > > > >
> > > > > >
> > > > > > >
> > > > > > > At 10:55 AM 10/16/00 , Christer Holmberg wrote:
> > > > > > > >
> > > > > > > >Hi,
> > > > > > > >
> > > > > > > >First, this mail is NOT a proposal for a change. I would =
just
> > > > > like
> > > > > > to
> > > > > > > >hear some opinions - or maybe it has already been =
discussed on
> > > > > the
> > > > > > list?
> > > > > > > >
> > > > > > > >The INFO Method RFC says that for INFO requests without a =
message
> > > > >
> > > > > > body,
> > > > > > > >a server supporting INFO MUST respond with response code =
200. So,
> > > > >
> > > > > > my
> > > > > > > >question is: why couldn't INFO be used for the =
session-timer
> > > > > > > >functionality? Is the ACK the (re-)INVITE provides really
> > > > > necessary
> > > > > > for
> > > > > > > >the purpose?
> > > > > > > >
> > > > > > > >And, like the session-timer is defined now, the other =
party would
> > > > >
> > > > > > not
> > > > > > > >even have to support the INFO method, because a 501 Not
> > > > > Implemented
> > > > > > > >would be sent, and the sender of the INFO would know the =
session
> > > > > is
> > > > > > > >"alive".
> > > > > > > >
> > > > > > > >Just some thoughts... Please let me know if there is =
something I
> > > > > > haven't
> > > > > > > >taken into consideration.
> > > > > > > >
> > > > > > > >Regards,
> > > > > > > >
> > > > > > > >Christer Holmberg
> > > > > > > >Ericsson Finland
> > > > > > > >
> > > > > >
> > > > >
> > > > > _______________________________________________
> > > > > SIP mailing list
> > > > > SIP@lists.bell-labs.com
> > > > > http://lists.bell-labs.com/mailman/listinfo/sip
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Hi,


>>>A follow-up question:
>>>How does/does Also-From header differ from the Remote-Party-Id header
>>>(draft-dcsgroup-sip-privacy-02.txt)?
>>
>>Well, it could perhaps be possible to use that header, but the whole
>>concept of that draft is a little different than just adding an
>>telephone number alias to the caller, and by supporting it I guess you
>>would have to do more implementation than necessary for this specfic
>>need. Also, depending on what is put in Remote-Party-Id header, it is
>>not even sure it would solve the problem.
>
>I agree. Remote-Party-ID is supposed to be an address that the network =
>has "verified" to be the actual address of the originator. It is even =
>stripped if verification cannot take place. THis seems a bit at odds =
with >the desired function here.
>=20
>That said, I still do not understand this need for YAI (yet another
>identifier). This previous motivation involving the termination in the =
>PSTN or IP network is totally bogus. At this point, I am very, very, =
very
>reluctant to add new stuff without serious justification.

I agree, that this is simply an alias for the From header value, it is
maybe not such a good, and efficient, way to add a new header. Another
way would be to simply add a "telephone-number" URI parameter. In that
case, however, the UAC must insert it (since intermediate nodes are not
allowed to change to From header).

Another way would be for the UAS (the PSTN GW in this case) to send a
response, asking for the telephone number. A new INVITE, with the
telephone number in the From (in some cases the URI could even be
changed from a SIP-URL to a TEL-URL) header. And, if the
"telephone-number" URI parameter is used, the From URI would not have to
be changed in the new INVITE - the parameter would simply be added to
the old SIP-URL.


Regards,

Christer Holmberg
Ericsson Finland

=20
> -Jonathan R.
>=20
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
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Hi,

I think this issue is also related to another thread, 'Source
Routing/Route out of band'. If the reason for using re-INVITE is that
booted stateful proxies can re-establish the session, but they don't
support Routes in a new INVITE, I guess it will not work. Of course, an
INFO would not pass that proxy either, so it is a general problem, and
does not only affect the session-timer functionality.

Christer Holmberg
Ericsson Finland


> Brian Stucker wrote:
> 
> Your correct, I can't stop the UAC from sending the method again, but
> I'd like to plead to as many implementors as I can that it would
> definitely be preferable in any setting if you wouldn't. =)
> 
> Brian
> 
> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
> Sent: Tuesday, October 17, 2000 9:00 AM
> To: sip@lists.bell-labs.com
> Cc: hammer michael
> Subject: Re: [SIP] Session-timer with INFO
> 
> Hi,
> 
> If I send you an INFO, or any other method that you don't support, you
> 
> can for example send a 501 Not Implemented response, and include an
> Allow header with the methods you support. Of course, you can't stop
> the
> UAC of sending the method again, if he chooses to...
> 
> You can not, however, tell an UAC not to send you re-INVITEs, because
> then you would also forbid the UAC to send you INVITEs - and I don't
> think you want to do that. Also, the specification says that a UA MUST
> 
> be able to receive INVITEs (which included re-INVITEs), so...
> 
> Regards,
> 
> Christer Holmberg
> Ericsson Finland
> 
> hammer michael wrote:
> >
> > Is this the Internet version of "let them eat cake?"  Signal-spam is
> not
> > more palatable than the normal variety.
> >
> > Is there a response code that can indicate the sentiment:  I don't
> support
> > this method, don't try it again?  It might also be good to make sure
> that
> > such undesirable sub-methods are indicated as "Unsupported."  If
> that
> > sub-method is well-defined, then that header parameter should be
> applicable.
> >
> > Mike
> >
> > At 12:30 AM 10/17/2000 +0300, Christer Holmberg wrote:
> >
> > >Hi,
> > >
> > > >Good points. Here's one more problem you may want to consider.
> > > >Whenever you invoke a ping-style method of deciding if a session
> is
> > > >alive, you open up problems due to network jitter and bandwidth.
> For
> > > >instance, you could easily have a packet delayed at an endpoint
> due to
> > > >a large file transfer starting up, or some other bandwidth
> consuming
> > > >event. In this case, your ping could be delayed long enough to
> cause
> > > >the session to drop for no good reason.
> > >
> > >Well, this could be the case no matter which method (INVITE, INFO,
> or
> > >whatever...) you use in the heartbeat request...
> > >
> > > >You're also creating the potential for artificial problems in
> wireless
> > > >networks by using a ping-method to keep a session alive. During a
> 
> > > >handover there may be a period where packets are in effect
> "muted" for
> > > >a short period of time. The mute period may not, by itself be
> long
> > > >enough to trigger a ping-timer to pop and kill the session, but
> if the
> > > >ping was near the edge of the time envelope to be accepted as
> > > >"on-time" anyhow, once again, your session could very easily drop
> when
> > > >the media was operating just fine.
> > > >Also, keep in mind, the longer you set the ping timer for, the
> more
> > > >network resources you will likely waste over time because of the
> > > >additional latentcy in discovering a "dead" session.
> > >
> > >Just to clarify, this "ping-method" IS already defined for SIP
> > >(draft-ietf-sip-session-timer-02.txt), and re-INVITEs are used to
> ping.
> > >My idea was about replacing the re-INVITE with INFO, which mean we
> would
> > >actually save resources since we don't have to send ACK every time.
> 
> > >
> > > >This method has one other nasty side-effect. You'd be wasting
> network
> > > >resources on an event that will likely not occur very often.
> After
> > > >all, every INFO exchange except the very last one that
> invalidates the
> > > >session could have very well not been sent without any negative
> impact
> > > >on the session (otherwise they would have been the last one). So
> > > >you've got proxies and UAs, and other transport resources being
> tied
> > > >up processing packets that by and large are providing no value,
> and
> > > >they're chewing up bandwidth budget that other services could be
> using
> > > >instead.
> > >
> > >That is an interesting point, and it is valid also for a number of
> other
> > >SIP extensions where extra messages are sent (e.g. PRACK), but it
> is
> > >another discussion.
> > >
> > > >Please keep in mind that not everyone is looking at implementing
> SIP
> > > >on a 10Mbit+ hardwired LAN (or has to scale up to tremendous
> numbers
> > > >of users), pinging and triangle-routing can be the death-knell of
> 
> > > >these sorts of networks.
> > >
> > >Who said "bandwidth is free"? :)
> > >
> > > >Because of all of this, I think it would be a very bad idea for a
> UA
> > > >to send an INFO to a client who repeatedly tells him "I don't
> > > >understand INFO" to see if he's alive.
> > >
> > >The number of messages on the network will be the same no matter if
> we
> > >send an INVITE which he understands, or an INFO he may not
> understand.
> > >Also, when we receive the response for the INVITE we also have to
> send
> > >back an ACK, which is not the case when INFO is used.
> > >
> > > >He may not understand INFO because that particular network
> doesn't
> > > deem >the bandwidth required for this mechanism as being best used
> for
> > > that >purpose. If someone tells you "don't send me this method"
> then
> > > don't send >it to them.
> > >
> > >Again, this has nothing to do with the bandwith issue. Even if I
> send
> > >re-INVITEs, which the other party understands, he can not tell me
> not to
> > >send any more re-INVITEs.
> > >
> > > >Otherwise your implementation may be viewed as favorably as the
> record
> > > >clubs that used to require you to send them a card every month
> telling
> > > >them not to send you anything by some groups.
> > >
> > >Well, it's not up to me to decide when and if an operator will use
> the
> > >session-timer, as defined in the draft, or not - I am just trying
> to
> > >make it as good (and less resource consuming) if he choses to.
> > >
> > >Regards,
> > >
> > >Christer Holmberg
> > >Ericsson Finland
> > >
> > >
> > > >
> > > > Brian Stucker
> > > > Nortel Networks
> > > >
> > > > -----Original Message-----
> > > > From: Culpepper, Bert
> [mailto:bert.culpepper@intervoice-brite.com]
> > > > Sent: Monday, October 16, 2000 3:38 PM
> > > > To: Christer Holmberg; sip@lists.bell-labs.com
> > > > Cc: Rohan Mahy
> > > > Subject: RE: [SIP] Session-timer with INFO
> > > >
> > > > Some things to think about.
> > > >
> > > > I believe INFO is _not_ to be used to modify the state of a
> session.
> > > > I
> > > > suppose a debate is possible about session-timer affecting
> session
> > > > state - present vs. future and if INFO is only restricted to
> one.  But
> > > > it's
> > > > probably best to leave it alone.  Also, the INV-200-ACK exchange
> 
> > > > allows multiple parties to agree on the duration of the
> session.  If
> > > > the
> > > > recipient of the INFO request wants a smaller value (because its
> 
> > > > heartbeat interval is shorter) for the session timer it would
> have to
> > > > send its own INFO request (unless you plan to keep the proposed
> > > > Session-expires header and its use in a 200 OK).  Now four
> > > > messages are needed.  In addition, there may be proxy issues
> when
> > > > using INFO.  There's other behavior described in the draft too.
> > > >
> > > > > -----Original Message-----
> > > > > From: Christer Holmberg
> [mailto:christer.holmberg@lmf.ericsson.se]
> > > > > Sent: Monday, October 16, 2000 3:42 PM
> > > > > To: sip@lists.bell-labs.com
> > > > > Cc: Rohan Mahy
> > > > > Subject: Re: [SIP] Session-timer with INFO
> > > > >
> > > > >
> > > > >
> > > > > Hi,
> > > > >
> > > > > Thanks for the comments!
> > > > >
> > > > > >If the UA crashed and rebooted and forgot about the old
> session, it
> > > >
> > > > > would
> > > > > >respond with either a 481 call leg doesn't exist (you would
> know
> > > > the
> > > > > call
> > > > > >is down), or a 501 (you know the UA is up, but is the call?)
> > > > > >
> > > > > >thanks,
> > > > > >-rohan
> > > > >
> > > > > I think that would be good, because then we could inform the
> user
> > > > that
> > > > > the call doesn't exist anymore.
> > > > >
> > > > > Let's assume the UAS crashes, reboots and the UAC sends the
> > > > > session-timer re-INVITE. If the UAS has forgotten about the
> old
> > > > > session
> > > > > it would consider the re-INVITE as a new INVITE, for a new
> session.
> > > > It
> > > > > would try to establish a new session - and we could suddenly
> start
> > > > > receiving provisional 18x responses (with or without SDP
> bodies),
> > > > > and/or
> > > > > a 200 response with a SDP body, to the session-timer
> re-INVITE. What
> > > >
> > > > > do
> > > > > we do then?
> > > >
> > > > Well, you could ignore the provisional responses (you know the
> other
> > > > end
> > > > is confused) and let the session get re-established or decide to
> 
> > > > terminate
> > > > the session with CANCEL/BYE.
> > > >
> > > > Regards,
> > > > Bert
> > > >
> > > > >
> > > > > Regards,
> > > > >
> > > > > Christer Holmberg
> > > > > Ericsson Finland
> > > > >
> > > > >
> > > > >
> > > > >
> > > > >
> > > > >
> > > > >
> > > > > >
> > > > > > At 10:55 AM 10/16/00 , Christer Holmberg wrote:
> > > > > > >
> > > > > > >Hi,
> > > > > > >
> > > > > > >First, this mail is NOT a proposal for a change. I would
> just
> > > > like
> > > > > to
> > > > > > >hear some opinions - or maybe it has already been discussed
> on
> > > > the
> > > > > list?
> > > > > > >
> > > > > > >The INFO Method RFC says that for INFO requests without a
> message
> > > >
> > > > > body,
> > > > > > >a server supporting INFO MUST respond with response code
> 200. So,
> > > >
> > > > > my
> > > > > > >question is: why couldn't INFO be used for the
> session-timer
> > > > > > >functionality? Is the ACK the (re-)INVITE provides really
> > > > necessary
> > > > > for
> > > > > > >the purpose?
> > > > > > >
> > > > > > >And, like the session-timer is defined now, the other party
> would
> > > >
> > > > > not
> > > > > > >even have to support the INFO method, because a 501 Not
> > > > Implemented
> > > > > > >would be sent, and the sender of the INFO would know the
> session
> > > > is
> > > > > > >"alive".
> > > > > > >
> > > > > > >Just some thoughts... Please let me know if there is
> something I
> > > > > haven't
> > > > > > >taken into consideration.
> > > > > > >
> > > > > > >Regards,
> > > > > > >
> > > > > > >Christer Holmberg
> > > > > > >Ericsson Finland
> > > > > > >
> > > > >
> > > >
> > > > _______________________________________________
> > > > SIP mailing list
> > > > SIP@lists.bell-labs.com
> > > > http://lists.bell-labs.com/mailman/listinfo/sip
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Greetings Chris:

Just to add a pointer to this thread:

Maintaining trasaction state is an application level functionality and
*shouldn't* be implemented in the base stack. For applications like
state oriented proxy server, the JAIN SIP can provide enough hooks for
event delivery but the state has to be maintained by the proxy server.

Pleas keep up the good work. It is really appreciated.

Regards,

Bobby.Sardana@mobilerain.com

Chris Harris wrote:

> "M. Ranganathan" wrote:
>
>> Hi Chris,
>>
>> Thanks for replying so promptly (I am sure you are flooded with
>> email ).
>
> I get one or two :)
>
>>
>> As you pointed out, I suppose we cannot do away with making Messages
>>
>> into classes because of the tie-in with java events (unfortunately)
>> but
>> I still like the notion of being able to specify  interfaces where
>> ever
>> possible.
>
> I see where you're coming from alright, and I suppose changing header
> classes into interfaces, could fit in with Gethin's proposal nicely?
>
>>
>>  I dont quite follow  the motivation behind the following design
>> decision (excerpted from your reply):
>>
>> Maybe I've misunderstood what you're saying - but a JAIN SIP
>> implementation is only stateless when it comes to call state. The
>> JainSipProvider implementation must maintain transaction state - it
>> simply exposes a reference to a transaction via the transaction id
>> for
>> the convenience of the JainSipListener. The service does not have
>> total
>> control over transacton state - the stack implementation does.
>>
>> It appears that I have indded mis-understood the architecture.  Why
>> should the above be so? Can one not rely on the JainSipListener to
>> keep
>> transaction state also, thereby making the stack simply a way of
>> recognising messages and triggering  event handlers on the basis of
>> message type. This would (IMHO) be a cleaner model yet.    That
>> way,  we
>> can do away with transaction identifiers being passed back and forth
>> and
>> keep all of this information in the handler.  ( You may have to
>> extend
>> the architecture somewhat to deal with timeouts and failures so that
>> the
>> handler knows about transaction failure.) In any case, if this is
>> not a
>> viable idea, I would be grateful if you can explain why not (or
>> point me
>> to the portion of the spec that explains why not) so I can get a
>> better
>> understanding.
>
> It would be cleaner model, and personally I am in total agreement with
> you - but it means the application is forced to maintain transaction
> state, match up headers - which has been said places too much of a
> burden on applications. How about finding a way to let the application
> choose whether or not it wants to keep transaction state?
>
>>
>>
>> Regards,
>>
>> Ranga.
>>
>> Chris Harris wrote:
>>
>> > Hi Ranga,
>> >
>> > Response below...
>> >
>> > Chris
>> >
>> > "M. Ranganathan" wrote:
>> >
>> >> JAIN-SIP is a groovy idea. Hats off to the whole concept and many
>>
>> >> thanks to Chris and
>> >> the JAIN-SIP team for putting together a spec and more flower
>> power
>> >> to them :-) .  In
>> >> particular I like the notion of treating UAS,  Redirect and Proxy
>>
>> >> servers as merely
>> >> special cases of the general notion of Services and  the idea of
>> >> unifying these
>> >> notions with an event mechanism and  event handlers.  Second, I
>> like
>> >> the idea of
>> >> separation between a  stateless  event -oriented "stack" and a
>> bunch
>> >> of event
>> >> handlers that are triggered by the stack with all state
>> (including
>> >> transaction
>> >> state) being separated from the event stack via the JAIN-SIP
>> mapping
>> >> layer if you
>> >> will.  ( Correct me if I  wax poetic but have the wrong
>> >> understanding....)
>> >
>> >>
>> >>
>> >> I hope I  will not be viewed as being too critical,  but I would
>> >> like to go  one step
>> >> further than what Gethin is suggesting.   I am all for using
>> >> interfaces as much as
>> >> possible and leaving class hierarchy definitions to implemeters.
>> >> Defining class
>> >> hierarchies almost lays out an implementation and involves
>> >> considerable rework for
>> >> those who have a working java-based stack, but have not used the
>> >> same classes. OK it
>> >> is just a matter of labor to map things to the JAIN-SIP class
>> >> hierarchy but it  IS a
>> >> dis-incentive.
>> >>
>> >> 1. Lets use interfaces for all messages rather than actually
>> define
>> >> class
>> >> hierarchies  ( yes Chris has pointed out the historical precedent
>>
>> >> behind actually
>> >> defining class hierarchies for messages. IMHO,  there has to be a
>>
>> >> more compelling
>> >> reason than that.)
>> >
>> > I suppose all classes could be turned into interfaces, except for
>> the
>> > Message classes which must extend java.util.EventObject to fit
>> into
>> > the JAIN framework.
>> >
>> >>
>> >>
>> >> 2. The tie-in with the JAVA event mechanism has necessitated the
>> >> explicit definition
>> >> of class hierarchies in certain places.  Why not use callbacks
>> >> instead and do away
>> >> with this dependency as well?  (Basically the same thing as
>> events
>> >> but does not tie
>> >> into the JAVA event mechanism and its associated limitations.)
>> >
>> > Unfortunately the JAIN framework expilicitly relies on the Java
>> Event
>> > model, and I don't think callbacks would be accepted within the
>> JAIN
>> > community.
>> >
>> >>
>> >>
>> >> Nice work and keep it rolling!  Thanks.
>> >
>> > Couldn't do it without you guys - thank you!
>> >
>> >>
>> >>
>> >> Ranga.
>> >>
>> >> Gethin Liddell wrote:
>> >>
>> >> > All,
>> >> >
>> >> > This idea of abstraction of the JAIN API is certainly a valid
>> and
>> >> > interesting idea.  However, i'm a bit concerened that the
>> proposed
>> >>
>> >> > solution of many different packages, will only seek to
>> complicate
>> >> and
>> >> > maybe enlarge the API.
>> >> >
>> >> > I too see no reason why there is an EntityHeader etc... and i
>> also
>> >> do
>> >> > not see any requirement for an InviteMessage, AckMessage etc...
>>
>> >> >
>> >> > How about if the format of the API remains as is:
>> >> >
>> >> > jain.protocol.ip.sip
>> >> > jain.protocol.ip.sip.header
>> >> > jain.protocol.ip.sip.message
>> >> >
>> >> > except that the messages available in the message package
>> >> consisted of
>> >> > BasicRequest/Response, MinimalRequest/Response etc... where
>> each
>> >> > message is extened in turn (e.g. Minimal extends Basic,
>> Moderate
>> >> > extends Minimal etc...), of course as Chris points out it is
>> not
>> >> hard
>> >> > to see that a combination would be required that is not
>> provided.
>> >> So
>> >> > in the interest of flexibility, it could be possible for the
>> user
>> >> to
>> >> > plug-in to the message class what extra headers it requires.
>> The
>> >> only
>> >> > issue then is that the user will have to use a "public Header
>> >> > getHeader(String headerName)" and then cast the header to their
>>
>> >> desired
>> >> > header type. Either that or create their own Message class
>> >> > (extending the current ones) that simply does the casting
>> >> automatically
>> >> > for them.  Then after compiling, run an obfuscator on the code
>> and
>> >> it
>> >> > will automatically extract and only extract the Message,
>> Headers
>> >> and
>> >> > even methods that are used.
>> >> >
>> >> > This then will also cut down on the number of methods in the
>> >> Provider
>> >> > as there will be no need for the sendAck(AckMessage),
>> >> > sendInvite(InviteMessage) etc.. as they would be replaced by
>> the
>> >> > single method sendRequest(RequestMessage).
>> >> >
>> >> > I personally see three arguments for keeping the JAIN SIP stack
>> as
>> >> small
>> >> > as possible:
>> >> >
>> >> > 1) The first is for embedded applications where stack size has
>> to
>> >> be
>> >> > kept to a minimum as memory is at a premium.  So if just using
>> the
>> >>
>> >> > Basic message suffices then this will compile and run with a
>> small
>> >>
>> >> > number of classes present from the stack.  Don't forget,
>> >> obfuscation
>> >> > will help in extracting only the Messages, Headers and even
>> >> methods
>> >> > required from a jar file but it won't do everything.
>> >> >
>> >> > 2) The second is that users may be put off or confused by the
>> >> sheer
>> >> > size and complexity of the JAIN stack.  So if only using the
>> >> > BasicMessage gets them on the first rung of the ladder then it
>> is
>> >> a
>> >> > good starting point and allows them to build up to bigger and
>> >> better
>> >> > things.
>> >> >
>> >> > 3) We also have to consider who is going to implement the JAIN
>> SIP
>> >>
>> >> > stack. If it is indeed overly complicated then no-one or a
>> single
>> >> > vendor may end up implmenting the stack and thus its definition
>>
>> >> and
>> >> > all the hard work that has gone into it is useless.
>> >> >
>> >> > On Mon, 16 Oct 2000, Chris Harris wrote:
>> >> > > Itamar,
>> >> > >
>> >> > > The idea of levelling the API based on what messages, headers
>>
>> >> and response codes
>> >> > > are understood is certainly an interesting one - minimal,
>> basic,
>> >> redirection,
>> >> > > firewall-friendly, negotiation, authentication and "full".
>> >> > >
>> >> > > jain.protocol.ip.sip
>> >> > > jain.protocol.ip.sip.header
>> >> > > jain.protocol.ip.sip.message
>> >> > >
>> >> > > could then become...
>> >> > >
>> >> > > jain.protocol.ip.sip - listener, provider, stack, general
>> >> classes used at all
>> >> > > levels
>> >> > > jain.protocol.ip.sip.header - contains base Header class (and
>>
>> >> requestHeader,
>> >> > > ResponseHeader, EntityHeader, GeneralHeader)
>> >> > > jain.protocol.ip.sip.message - contains base Message class
>> >> > >
>> >> > > jain.protocol.ip.sip.minimal - general classes used only at
>> this
>> >> level and above
>> >> > >
>> >> > > jain.protocol.ip.sip.minimal.header - headers used only at
>> this
>> >> level and above
>> >> > > jain.protocol.ip.sip.minimal.message - messages only used at
>> >> this level and
>> >> > > above
>> >> > >
>> >> > > jain.protocol.ip.sip.basic - general classes used only at
>> this
>> >> level and above
>> >> > > jain.protocol.ip.sip.basic.header - headers used only at this
>>
>> >> level and above
>> >> > > jain.protocol.ip.sip.basic.message - messages used only at
>> this
>> >> level and above
>> >> > > ...
>> >> > > ....
>> >> > > jjain.protocol.ip.sip."full" - general classes only used at
>> this
>> >> level
>> >> > > jain.protocol.ip.sip."full".header - headers used only at
>> this
>> >> level
>> >> > > jain.protocol.ip.sip."full".message - messages used only at
>> this
>> >> level
>> >> > >
>> >> > > The API user could then pick the packages they need - say
>> they
>> >> initially only
>> >> > > want the minimal features, but then realise they need to use
>> >> their application
>> >> > > behind a firewall - then they can simply add the
>> >> firewall-friendly package. The
>> >> > > RFC says "In general, each capability listed builds on the
>> ones
>> >> above it" but it
>> >> > > is not hard to see that firewall-friendly and authentication
>> may
>> >> be desired
>> >> > > without redirection for example.
>> >> > >
>> >> > > Is this what you hand in mind Itamar?
>> >> >
>> >> > --
>> >> > Gethin Liddell
>> >> > Ubiquity Software Corporation
>> >> >
>> >> > http://www.ubiquity.net
>> >> > mailto:gethin@ubiquity.net
>> >> >
>> >> > _______________________________________________
>> >> > SIP mailing list
>> >> > SIP@lists.bell-labs.com
>> >> > http://lists.bell-labs.com/mailman/listinfo/sip
>> >>
>> >> --
>> >> M.Ranganathan
>> >> NIST Advanced Networking Technologies Group,
>> >> 100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
>> >> Tel: 301 975 3664 Fax: 301 590 0932
>> >>
>> >> Hf=E6j)b? =
b=B2=D4^>X=AC=B6=C6=DE-YZn=C7(s=1Bm=A7=FF=E5S=CBlm=E9e*=A6=ECr?=BF?=A8=A5=
?=A9=FF-+-Sw=E8=FE=C8=A9
>> >
>>
>> --
>> M.Ranganathan
>> NIST Advanced Networking Technologies Group,
>> 100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
>> Tel: 301 975 3664 Fax: 301 590 0932
>>
>> Hf=E6j)b? =
b=B2=D4^>X=AC=B6=C6=DE-YZn=C7(s=1Bm=A7=FF=E5S=CBlm=E9e*=A6=ECr?=BF?=A8=A5=
?=A9=FF-+-Sw=E8=FE=C8=A9
>

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From sip-admin@lists.bell-labs.com  Tue Nov 14 05:03:28 2000
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Hi,

>Do you actually mean the From: or do you mean the To:?

From.

>It seems a bit odd for a redirect server to tell the UAC what
>From: to use. Perhaps you could give an example of its usage.

Well, this is - once again - related to the PSTN/SIP GW issue, if the GW
needs a telephone number for some kind of A number analysis. When I make
a phone call from my SIP phone I may not know that the call may end up
on the PSTN side, so I can for example put my email address in the From
header. However, since the network architecture includes a GW to the
PSTN network, somewhere there could be a server which is able to do the
mapping between the email address user part and a telephone number user
part (if SIP URL is used).

Regards,

Christer Holmberg
Ericsson Finland


>=20
> Regards,
> Sean
>=20
> --
> Sean Olson <sean.olson@ericsson.com>
>=20
> Christer Holmberg wrote:
>=20
> > Another proposal on the issue, with another use of the "Also-From"
> > header.
> >
> > Currently, a redirect server add a new Request-URI (in the 3xx =
response
> > Contact header) to the UAC. An idea would be that it could also =
return a
> > new From header, for example with a telephone number. In this case, =
the
> > UAC, and intermediate proxies, would not need to insert any extra
> > headers or URI parameters in the request. This would be a new, =
optional,
> > feature in a redirect server.
> >
> > Comments?
> >
> > Christer Holmberg
> > Ericsson Finland
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Greetings:

Communicating Presentation Information in Internet Messages:
       The Content-Disposition Header Field (RFC 2183)

Regards,

Bobby.Sardana@mobilerain.com

Kevin Summers wrote:

> Which RFC defines Content-Disposition and its parameters?
>
> -----Original Message-----
> From: Dean Willis [mailto:dwillis@dynamicsoft.com]
> Sent: Thursday, November 09, 2000 12:59 PM
> To: IETF SIP (E-mail)
> Subject: [SIP] WG Last Call on draft-ietf-isup-mime-06.txt
>
> After one last very minor tweak, I believe we are ready to have a
> working-group last call on the ISUP-MIME draft, closing November 23.
>
> The latest rev has been sent to the drafts editor and can also be =
downloaded
> from:
>
> http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-isup-mime-06.txt
>
> And remember, vote early, vote often.
>
> --
> Dean Willis
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Another proposal on the issue, with another use of the "Also-From"
header.

Currently, a redirect server add a new Request-URI (in the 3xx response
Contact header) to the UAC. An idea would be that it could also return a
new From header, for example with a telephone number. In this case, the
UAC, and intermediate proxies, would not need to insert any extra
headers or URI parameters in the request. This would be a new, optional,
feature in a redirect server.

Comments?

Christer Holmberg
Ericsson Finland
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Chris Harris wrote:

>
>
> Bobby Sardana wrote:
>
>> Greetings Chris:
>>
>> Just to add a pointer to this thread:
>>
>> Maintaining trasaction state is an application level functionality
>> and *shouldn't* be implemented in the base stack. For applications
>> like state oriented proxy server, the JAIN SIP can provide enough
>> hooks for event delivery but the state has to be maintained by the
>> proxy server.
>
> Is simply sending up all received requests without any kind of
> transaction id really enough?

Yes. In the case of SIP there actually is no such thing as
TransactionID, compared to some other protocols like TCAP. The
TrasactionID in case of SIP is a composite collection of headers. The
headers, once parsed, are available to the application. The application
then should track what state it should maintain. It is equivalent to a
Call Model vs Encoding/Decoding of protocols like INAP.

> What are the hooks you refer to?

The hooks I refer to are the delegation event delivery mechanism via
JainSipListener.

> An application like a stateful proxy server will surely be able to
> handle this, but what about simple services? - shouldn't the API
> provide a convenient means to send responses, acks, cancel etc?

Here you are referring to the Call Model - which manages the
protocol/application state. All I am suggesting is that it should be
independent of the stack.

Regards,

Bobby.Sardana@mobilerain.com


>
>
>>
>>
>> Pleas keep up the good work. It is really appreciated.
>>
>> Regards,
>>
>> Bobby.Sardana@mobilerain.com
>>
>> Chris Harris wrote:
>>
>> > "M. Ranganathan" wrote:
>> >
>> >>  Hi Chris,
>> >>
>> >>  Thanks for replying so promptly (I am sure you are flooded with
>> >>  email ).
>> >
>> > I get one or two :)
>> >
>> >>
>> >>  As you pointed out, I suppose we cannot do away with making
>> >>  Messages
>> >>  into classes because of the tie-in with java events
>> >>  (unfortunately) but
>> >>  I still like the notion of being able to specify  interfaces
>> >>  where ever
>> >>  possible.
>> >
>> > I see where you're coming from alright, and I suppose changing
>> > header classes into interfaces, could fit in with Gethin's proposal
>> > nicely?
>> >
>> >>
>> >>   I dont quite follow  the motivation behind the following design
>> >>  decision (excerpted from your reply):
>> >>
>> >>  Maybe I've misunderstood what you're saying - but a JAIN SIP
>> >>  implementation is only stateless when it comes to call state. The
>> >>
>> >>  JainSipProvider implementation must maintain transaction state -
>> >>  it
>> >>  simply exposes a reference to a transaction via the transaction
>> >>  id for
>> >>  the convenience of the JainSipListener. The service does not have
>> >>  total
>> >>  control over transacton state - the stack implementation does.
>> >>
>> >>  It appears that I have indded mis-understood the architecture.
>> >>  Why
>> >>  should the above be so? Can one not rely on the JainSipListener
>> >>  to keep
>> >>  transaction state also, thereby making the stack simply a way of
>> >>  recognising messages and triggering  event handlers on the basis
>> >>  of
>> >>  message type. This would (IMHO) be a cleaner model yet.    That
>> >>  way,  we
>> >>  can do away with transaction identifiers being passed back and
>> >>  forth and
>> >>  keep all of this information in the handler.  ( You may have to
>> >>  extend
>> >>  the architecture somewhat to deal with timeouts and failures so
>> >>  that the
>> >>  handler knows about transaction failure.) In any case, if this is
>> >>  not a
>> >>  viable idea, I would be grateful if you can explain why not (or
>> >>  point me
>> >>  to the portion of the spec that explains why not) so I can get a
>> >>  better
>> >>  understanding.
>> >
>> > It would be cleaner model, and personally I am in total agreement
>> > with you - but it means the application is forced to maintain
>> > transaction state, match up headers - which has been said places
>> > too much of a burden on applications. How about finding a way to
>> > let the application choose whether or not it wants to keep
>> > transaction state?
>> >
>> >>
>> >>
>> >>  Regards,
>> >>
>> >>  Ranga.
>> >>
>> >>  Chris Harris wrote:
>> >>
>> >>  > Hi Ranga,
>> >>  >
>> >>  > Response below...
>> >>  >
>> >>  > Chris
>> >>  >
>> >>  > "M. Ranganathan" wrote:
>> >>  >
>> >>  >> JAIN-SIP is a groovy idea. Hats off to the whole concept and
>> >>  many
>> >>  >> thanks to Chris and
>> >>  >> the JAIN-SIP team for putting together a spec and more flower
>> >>  power
>> >>  >> to them :-) .  In
>> >>  >> particular I like the notion of treating UAS,  Redirect and
>> >>  Proxy
>> >>  >> servers as merely
>> >>  >> special cases of the general notion of Services and  the idea
>> >>  of
>> >>  >> unifying these
>> >>  >> notions with an event mechanism and  event handlers.  Second,
>> >>  I like
>> >>  >> the idea of
>> >>  >> separation between a  stateless  event -oriented "stack" and a
>> >>  bunch
>> >>  >> of event
>> >>  >> handlers that are triggered by the stack with all state
>> >>  (including
>> >>  >> transaction
>> >>  >> state) being separated from the event stack via the JAIN-SIP
>> >>  mapping
>> >>  >> layer if you
>> >>  >> will.  ( Correct me if I  wax poetic but have the wrong
>> >>  >> understanding....)
>> >>  >
>> >>  >>
>> >>  >>
>> >>  >> I hope I  will not be viewed as being too critical,  but I
>> >>  would
>> >>  >> like to go  one step
>> >>  >> further than what Gethin is suggesting.   I am all for using
>> >>  >> interfaces as much as
>> >>  >> possible and leaving class hierarchy definitions to
>> >>  implemeters.
>> >>  >> Defining class
>> >>  >> hierarchies almost lays out an implementation and involves
>> >>  >> considerable rework for
>> >>  >> those who have a working java-based stack, but have not used
>> >>  the
>> >>  >> same classes. OK it
>> >>  >> is just a matter of labor to map things to the JAIN-SIP class
>> >>  >> hierarchy but it  IS a
>> >>  >> dis-incentive.
>> >>  >>
>> >>  >> 1. Lets use interfaces for all messages rather than actually
>> >>  define
>> >>  >> class
>> >>  >> hierarchies  ( yes Chris has pointed out the historical
>> >>  precedent
>> >>  >> behind actually
>> >>  >> defining class hierarchies for messages. IMHO,  there has to
>> >>  be a
>> >>  >> more compelling
>> >>  >> reason than that.)
>> >>  >
>> >>  > I suppose all classes could be turned into interfaces, except
>> >>  for the
>> >>  > Message classes which must extend java.util.EventObject to fit
>> >>  into
>> >>  > the JAIN framework.
>> >>  >
>> >>  >>
>> >>  >>
>> >>  >> 2. The tie-in with the JAVA event mechanism has necessitated
>> >>  the
>> >>  >> explicit definition
>> >>  >> of class hierarchies in certain places.  Why not use callbacks
>> >>
>> >>  >> instead and do away
>> >>  >> with this dependency as well?  (Basically the same thing as
>> >>  events
>> >>  >> but does not tie
>> >>  >> into the JAVA event mechanism and its associated limitations.)
>> >>
>> >>  >
>> >>  > Unfortunately the JAIN framework expilicitly relies on the Java
>> >>  Event
>> >>  > model, and I don't think callbacks would be accepted within the
>> >>  JAIN
>> >>  > community.
>> >>  >
>> >>  >>
>> >>  >>
>> >>  >> Nice work and keep it rolling!  Thanks.
>> >>  >
>> >>  > Couldn't do it without you guys - thank you!
>> >>  >
>> >>  >>
>> >>  >>
>> >>  >> Ranga.
>> >>  >>
>> >>  >> Gethin Liddell wrote:
>> >>  >>
>> >>  >> > All,
>> >>  >> >
>> >>  >> > This idea of abstraction of the JAIN API is certainly a
>> >>  valid and
>> >>  >> > interesting idea.  However, i'm a bit concerened that the
>> >>  proposed
>> >>  >>
>> >>  >> > solution of many different packages, will only seek to
>> >>  complicate
>> >>  >> and
>> >>  >> > maybe enlarge the API.
>> >>  >> >
>> >>  >> > I too see no reason why there is an EntityHeader etc... and
>> >>  i also
>> >>  >> do
>> >>  >> > not see any requirement for an InviteMessage, AckMessage
>> >>  etc...
>> >>  >> >
>> >>  >> > How about if the format of the API remains as is:
>> >>  >> >
>> >>  >> > jain.protocol.ip.sip
>> >>  >> > jain.protocol.ip.sip.header
>> >>  >> > jain.protocol.ip.sip.message
>> >>  >> >
>> >>  >> > except that the messages available in the message package
>> >>  >> consisted of
>> >>  >> > BasicRequest/Response, MinimalRequest/Response etc... where
>> >>  each
>> >>  >> > message is extened in turn (e.g. Minimal extends Basic,
>> >>  Moderate
>> >>  >> > extends Minimal etc...), of course as Chris points out it is
>> >>  not
>> >>  >> hard
>> >>  >> > to see that a combination would be required that is not
>> >>  provided.
>> >>  >> So
>> >>  >> > in the interest of flexibility, it could be possible for the
>> >>  user
>> >>  >> to
>> >>  >> > plug-in to the message class what extra headers it
>> >>  requires.  The
>> >>  >> only
>> >>  >> > issue then is that the user will have to use a "public
>> >>  Header
>> >>  >> > getHeader(String headerName)" and then cast the header to
>> >>  their
>> >>  >> desired
>> >>  >> > header type. Either that or create their own Message class
>> >>  >> > (extending the current ones) that simply does the casting
>> >>  >> automatically
>> >>  >> > for them.  Then after compiling, run an obfuscator on the
>> >>  code and
>> >>  >> it
>> >>  >> > will automatically extract and only extract the Message,
>> >>  Headers
>> >>  >> and
>> >>  >> > even methods that are used.
>> >>  >> >
>> >>  >> > This then will also cut down on the number of methods in the
>> >>
>> >>  >> Provider
>> >>  >> > as there will be no need for the sendAck(AckMessage),
>> >>  >> > sendInvite(InviteMessage) etc.. as they would be replaced by
>> >>  the
>> >>  >> > single method sendRequest(RequestMessage).
>> >>  >> >
>> >>  >> > I personally see three arguments for keeping the JAIN SIP
>> >>  stack as
>> >>  >> small
>> >>  >> > as possible:
>> >>  >> >
>> >>  >> > 1) The first is for embedded applications where stack size
>> >>  has to
>> >>  >> be
>> >>  >> > kept to a minimum as memory is at a premium.  So if just
>> >>  using the
>> >>  >>
>> >>  >> > Basic message suffices then this will compile and run with a
>> >>  small
>> >>  >>
>> >>  >> > number of classes present from the stack.  Don't forget,
>> >>  >> obfuscation
>> >>  >> > will help in extracting only the Messages, Headers and even
>> >>  >> methods
>> >>  >> > required from a jar file but it won't do everything.
>> >>  >> >
>> >>  >> > 2) The second is that users may be put off or confused by
>> >>  the
>> >>  >> sheer
>> >>  >> > size and complexity of the JAIN stack.  So if only using the
>> >>
>> >>  >> > BasicMessage gets them on the first rung of the ladder then
>> >>  it is
>> >>  >> a
>> >>  >> > good starting point and allows them to build up to bigger
>> >>  and
>> >>  >> better
>> >>  >> > things.
>> >>  >> >
>> >>  >> > 3) We also have to consider who is going to implement the
>> >>  JAIN SIP
>> >>  >>
>> >>  >> > stack. If it is indeed overly complicated then no-one or a
>> >>  single
>> >>  >> > vendor may end up implmenting the stack and thus its
>> >>  definition
>> >>  >> and
>> >>  >> > all the hard work that has gone into it is useless.
>> >>  >> >
>> >>  >> > On Mon, 16 Oct 2000, Chris Harris wrote:
>> >>  >> > > Itamar,
>> >>  >> > >
>> >>  >> > > The idea of levelling the API based on what messages,
>> >>  headers
>> >>  >> and response codes
>> >>  >> > > are understood is certainly an interesting one - minimal,
>> >>  basic,
>> >>  >> redirection,
>> >>  >> > > firewall-friendly, negotiation, authentication and "full".
>> >>
>> >>  >> > >
>> >>  >> > > jain.protocol.ip.sip
>> >>  >> > > jain.protocol.ip.sip.header
>> >>  >> > > jain.protocol.ip.sip.message
>> >>  >> > >
>> >>  >> > > could then become...
>> >>  >> > >
>> >>  >> > > jain.protocol.ip.sip - listener, provider, stack, general
>> >>  >> classes used at all
>> >>  >> > > levels
>> >>  >> > > jain.protocol.ip.sip.header - contains base Header class
>> >>  (and
>> >>  >> requestHeader,
>> >>  >> > > ResponseHeader, EntityHeader, GeneralHeader)
>> >>  >> > > jain.protocol.ip.sip.message - contains base Message class
>> >>
>> >>  >> > >
>> >>  >> > > jain.protocol.ip.sip.minimal - general classes used only
>> >>  at this
>> >>  >> level and above
>> >>  >> > >
>> >>  >> > > jain.protocol.ip.sip.minimal.header - headers used only at
>> >>  this
>> >>  >> level and above
>> >>  >> > > jain.protocol.ip.sip.minimal.message - messages only used
>> >>  at
>> >>  >> this level and
>> >>  >> > > above
>> >>  >> > >
>> >>  >> > > jain.protocol.ip.sip.basic - general classes used only at
>> >>  this
>> >>  >> level and above
>> >>  >> > > jain.protocol.ip.sip.basic.header - headers used only at
>> >>  this
>> >>  >> level and above
>> >>  >> > > jain.protocol.ip.sip.basic.message - messages used only at
>> >>  this
>> >>  >> level and above
>> >>  >> > > ...
>> >>  >> > > ....
>> >>  >> > > jjain.protocol.ip.sip."full" - general classes only used
>> >>  at this
>> >>  >> level
>> >>  >> > > jain.protocol.ip.sip."full".header - headers used only at
>> >>  this
>> >>  >> level
>> >>  >> > > jain.protocol.ip.sip."full".message - messages used only
>> >>  at this
>> >>  >> level
>> >>  >> > >
>> >>  >> > > The API user could then pick the packages they need - say
>> >>  they
>> >>  >> initially only
>> >>  >> > > want the minimal features, but then realise they need to
>> >>  use
>> >>  >> their application
>> >>  >> > > behind a firewall - then they can simply add the
>> >>  >> firewall-friendly package. The
>> >>  >> > > RFC says "In general, each capability listed builds on the
>> >>  ones
>> >>  >> above it" but it
>> >>  >> > > is not hard to see that firewall-friendly and
>> >>  authentication may
>> >>  >> be desired
>> >>  >> > > without redirection for example.
>> >>  >> > >
>> >>  >> > > Is this what you hand in mind Itamar?
>> >>  >> >
>> >>  >> > --
>> >>  >> > Gethin Liddell
>> >>  >> > Ubiquity Software Corporation
>> >>  >> >
>> >>  >> > http://www.ubiquity.net
>> >>  >> > mailto:gethin@ubiquity.net
>> >>  >> >
>> >>  >> > _______________________________________________
>> >>  >> > SIP mailing list
>> >>  >> > SIP@lists.bell-labs.com
>> >>  >> > http://lists.bell-labs.com/mailman/listinfo/sip
>> >>  >>
>> >>  >> --
>> >>  >> M.Ranganathan
>> >>  >> NIST Advanced Networking Technologies Group,
>> >>  >> 100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
>> >>  >> Tel: 301 975 3664 Fax: 301 590 0932
>> >>  >>
>> >>  >> Hf=E6j)b? =
b=B2=D4^>X=AC=B6=C6=DE-YZn=C7(s=1Bm=A7=FF=E5S=CBlm=E9e*=A6=ECr?=BF?=A8=A5=
?=A9=FF-+-Sw=E8=FE=C8=A9
>> >>  >
>> >>
>> >>  --
>> >>  M.Ranganathan
>> >>  NIST Advanced Networking Technologies Group,
>> >>  100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
>> >>  Tel: 301 975 3664 Fax: 301 590 0932
>> >>
>> >>  Hf=E6j)b? =
b=B2=D4^>X=AC=B6=C6=DE-YZn=C7(s=1Bm=A7=FF=E5S=CBlm=E9e*=A6=ECr?=BF?=A8=A5=
?=A9=FF-+-Sw=E8=FE=C8=A9
>> >

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Hi,

>>Another proposal on the issue, with another use of the "Also-From"
>>header.
>>
>>Currently, a redirect server add a new Request-URI (in the 3xx =
response
>>Contact header) to the UAC. An idea would be that it could also return =
a
>>new From header, for example with a telephone number. In this case, =
the
>>UAC, and intermediate proxies, would not need to insert any extra
>>headers or URI parameters in the request. This would be a new, =
optional,
>>feature in a redirect server.
>=20
>A registrar is free to return SIP, tel:, mailto:, http: and any other
>kind of URLs in Contacts. Whatever it is you think you need the extra
>From for I think you'll find you can use Contact for.

I know, but the idea was to give the possibility not only to return a
new RequestURI - whatever format - in the Conact header, but also to
give a new From header, if for example the Redirect server knows that
the request will reach a PSTN/IP GW, and the current From header can for
example not be used for A number analysis.

I want to point out that this case is not about solving a bug in the
protocol itself, it has to do only with the interworking architecture
with PSTN.

Regards,

Christer Holmberg
Ericsson Finland
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Hi,

>>While it is true that there are other ways that this could have been
>>solved, it is pretty late in the game to make this kind of a change.  =
>>Especially if the only reason for doing so is to avoid the ACK =
message.
>Other reasons include the desire not to revive a
>dead session and to avoid the possibility of a new
>SDP in the 200 response.

True.


Regards,

Christer Holmberg
Ericsson Finland


>>As a side note, we are in the process of creating a new version of the
>>draft that we hope will be used for working group last call.
>=20
> Steve,
>=20
> Can a header or parameter be defined within
> the sip-session-timer to indicate that the sender
> is just trying to extend the Session-Timer and
> check session state, but the sender does NOT
> want the session revived if it is down?  The
> reverse of the boolean value could indicate a
> request to revive the session if it is down.
>=20
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
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From sip-admin@lists.bell-labs.com  Tue Nov 14 05:27:46 2000
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Hi,

>I see what you are trying to achieve here, but my concern is with the
>redirection rules.  When a UA gets a 3xx response, it issues a new =
>request. MUST that new request belong to the same call leg as the =
>original request?
>ie: same To: From: Call-id, but higher Cseq.  If so, your proposal =
breaks
>that rule.

It doesn't have to belong to the same call leg. It MAY do so, but you
MAY also change the To header (with the new value in the 3xx response
Contact header), and you MAY also change the Call-id value.

And, in any case, it doesn't affect anything else than my own
implementation, even if I use the same Call-id (which one would probably
do).=20
I mean - I don't even have to issue a new INVITE when I receive a 3xx,
so any possible intermediate SIP proxy etc. should not be affected.

Regards,

Christer Holmberg
Ericsson Finland




> -----Original Message-----
> From: sip-admin@lists.bell-labs.com =
[mailto:sip-admin@lists.bell-labs.com]On
> Behalf Of Christer Holmberg
> Sent: 24. lokakuuta 2000 14:02
> To: sip@lists.bell-labs.com
> Cc: Anders Kristensen
> Subject: Re: [SIP] Redirect server returning new From header
>=20
> Hi,
>=20
> >>Another proposal on the issue, with another use of the "Also-From"
> >>header.
> >>
> >>Currently, a redirect server add a new Request-URI (in the 3xx =
response
> >>Contact header) to the UAC. An idea would be that it could also =
return a
> >>new From header, for example with a telephone number. In this case, =
the
> >>UAC, and intermediate proxies, would not need to insert any extra
> >>headers or URI parameters in the request. This would be a new, =
optional,
> >>feature in a redirect server.
> >
> >A registrar is free to return SIP, tel:, mailto:, http: and any other
> >kind of URLs in Contacts. Whatever it is you think you need the extra
> >From for I think you'll find you can use Contact for.
>=20
> I know, but the idea was to give the possibility not only to return a
> new RequestURI - whatever format - in the Conact header, but also to
> give a new From header, if for example the Redirect server knows that
> the request will reach a PSTN/IP GW, and the current From header can =
for
> example not be used for A number analysis.
>=20
> I want to point out that this case is not about solving a bug in the
> protocol itself, it has to do only with the interworking architecture
> with PSTN.
>=20
> Regards,
>=20
> Christer Holmberg
> Ericsson Finland
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From sip-admin@lists.bell-labs.com  Tue Nov 14 05:29:26 2000
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From: =?ks_c_5601-1987?B?vK25rsjx?= <mhsuh@contela.com>
To: <mranga@nist.gov>
Cc: <sip@lists.bell-labs.com>
Subject: Re: [SIP] Content-Length values in RFC and call flows examples
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Greetings:

"M. Ranganathan" wrote:

> Here is a straightforward question regarding the examples in the RFC =
and
> sip call flows documents.
>
>  Are the  Content-Length values computed on the basis of CR-LF  (2
> characters) for  newline? (i.e. DOS vs UNIX convention)?

Yes.

Regards,

Bobby.Sardana@mobilerain.com

>
>
> Thank you  in advance for your replies.
>
> --
> M.Ranganathan
> NIST Advanced Networking Technologies Group,
> 100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
> Tel: 301 975 3664 Fax: 301 590 0932
>
> Hf=E6j)b? =
b=B2=D4^>X=AC=B6=C6=DE-YZn=C7(s=1Bm=A7=FF=E5S=CBlm=E9e*=A6=ECr?=BF?=A8=A5=
?=A9=FF-+-Sw=E8=FE=C8=A9

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From sip-admin@lists.bell-labs.com  Tue Nov 14 05:34:14 2000
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Hi,

Well, you may use the same call-ID value (the value should be unique for
the call, not for a single call leg), but you are still allowed to
change the To- and From headers.

And, chapter 7.3.3. says that the call-ID value can also be changed.

Regards,

Christer Holmberg
Ericsson Finland

Hisham Khartabil wrote:
>=20
> I can't find in the bis where it says that they belong to the same =
call leg,
> but in section 1.4.4 the last paragraph, it does say "The caller =
issues a
> new request,
> with the same call-ID but a higher CSeq, to the address returned by =
the
> first server"
>=20
> Hisham
>=20
> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
> Sent: 24. lokakuuta 2000 14:47
> To: sip@lists.bell-labs.com
> Cc: Hisham Khartabil
> Subject: Re: [SIP] Redirect server returning new From header
>=20
> Hi,
>=20
> >I see what you are trying to achieve here, but my concern is with the
> >redirection rules.  When a UA gets a 3xx response, it issues a new
> >request. MUST that new request belong to the same call leg as the =
>original
> request?
> >ie: same To: From: Call-id, but higher Cseq.  If so, your proposal =
breaks
> >that rule.
>=20
> It doesn't have to belong to the same call leg. It MAY do so, but you
> MAY also change the To header (with the new value in the 3xx response
> Contact header), and you MAY also change the Call-id value.
>=20
> And, in any case, it doesn't affect anything else than my own
> implementation, even if I use the same Call-id (which one would =
probably
> do).
> I mean - I don't even have to issue a new INVITE when I receive a 3xx,
> so any possible intermediate SIP proxy etc. should not be affected.
>=20
> Regards,
>=20
> Christer Holmberg
> Ericsson Finland
>=20
> > -----Original Message-----
> > From: sip-admin@lists.bell-labs.com
> [mailto:sip-admin@lists.bell-labs.com]On
> > Behalf Of Christer Holmberg
> > Sent: 24. lokakuuta 2000 14:02
> > To: sip@lists.bell-labs.com
> > Cc: Anders Kristensen
> > Subject: Re: [SIP] Redirect server returning new From header
> >
> > Hi,
> >
> > >>Another proposal on the issue, with another use of the "Also-From"
> > >>header.
> > >>
> > >>Currently, a redirect server add a new Request-URI (in the 3xx =
response
> > >>Contact header) to the UAC. An idea would be that it could also =
return a
> > >>new From header, for example with a telephone number. In this =
case, the
> > >>UAC, and intermediate proxies, would not need to insert any extra
> > >>headers or URI parameters in the request. This would be a new, =
optional,
> > >>feature in a redirect server.
> > >
> > >A registrar is free to return SIP, tel:, mailto:, http: and any =
other
> > >kind of URLs in Contacts. Whatever it is you think you need the =
extra
> > >From for I think you'll find you can use Contact for.
> >
> > I know, but the idea was to give the possibility not only to return =
a
> > new RequestURI - whatever format - in the Conact header, but also to
> > give a new From header, if for example the Redirect server knows =
that
> > the request will reach a PSTN/IP GW, and the current From header can =
for
> > example not be used for A number analysis.
> >
> > I want to point out that this case is not about solving a bug in the
> > protocol itself, it has to do only with the interworking =
architecture
> > with PSTN.
> >
> > Regards,
> >
> > Christer Holmberg
> > Ericsson Finland
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Hi,

>ok.  Thanks for clearing that up.
>=20
>A follow-up question:
>How does/does Also-From header differ from the Remote-Party-Id header
>(draft-dcsgroup-sip-privacy-02.txt)?

Well, it could perhaps be possible to use that header, but the whole
concept of that draft is a little different than just adding an
telephone number alias to the caller, and by supporting it I guess you
would have to do more implementation than necessary for this specfic
need. Also, depending on what is put in Remote-Party-Id header, it is
not even sure it would solve the problem.


Regards,

Christer Holmberg
Ericsson Finland

> Thanks for your help!
> Bryan
>=20
> Christer Holmberg wrote:
>=20
> > Hi,
> >
> > Using the Diversion header, you can inform an endpoint of which
> > Request-URIs have been used before the Request-URI of that specific
> > endpoint was used.
> >
> > The case I am talking about, however, is when we can insert an =
"alias"
> > header for the From header, using a telephone number, if the From =
header
> > is, for example, an e-mail address. Using this new header, let us =
call
> > is Also-From, a proxy (or the calling endpoint itself) may also =
insert a
> > telephone number, which can then be used by PSTN Gateways for any =
kind
> > of A-number analysing. One could argue that the caller should always
> > include a telephone number in the From header, but the caller may =
not
> > know that the call will end up on the PSTN side, so...
> >
> > I DO like the Diversion draft very much, though, but it doesn't =
solve
> > this specific problem :)
> >
> > Regards,
> >
> > Christer Holmberg
> > Ericsson Finland
> >
> > Bryan Byerly wrote:
> > >
> > > Hi guys,
> > >
> > > What do you guys think about using the Diversion header
> > > =
(http://www.ietf.org/internet-drafts/draft-levy-sip-diversion-00.txt) =
instead of
> > > Also-From?
> > > It seems like a natural replacement to me.
> > >
> > > Thanks!
> > > Bryan
> > >
> > > Christer Holmberg wrote:
> > >
> > > > Hi,
> > > >
> > > > This is something I've also been looking into.
> > > >
> > > > One way would be to add a new header (e.g. Also-From). That =
header could
> > > > be inserted by a proxy that does know the request may end up in =
a
> > > > gateway. The UA may not know that, so he could put whatever =
string in
> > > > the user part of the From header (e.g. his e-mail address). If =
the proxy
> > > > then knows the UAs telephone number, for example by using some =
kind of
> > > > database lookup, which is needed by the gateway - for any =
possible
> > > > A-number procedure - the proxy can add that to the request.
> > > >
> > > > Another way would be to just define a URI parameter for the From =
header,
> > > > but the bad thing about that is that it can not be inserted by a =
proxy
> > > > if it was not in the request from the UA (since the From header =
must not
> > > > be changed) by a proxy, or any other intermediate node.
> > > >
> > > > Note, that this response is according to Clive's mail. I do NOT =
support
> > > > the idea of ALSO-TO headers, multiple FROM/TO headers and =
whatever was
> > > > proposed in the previous mails :)
> > > >
> > > > Regards,
> > > >
> > > > Christer Holmberg
> > > > Ericsson Finland
> > > >
> > > > clive.dellard@bt.com wrote:
> > > > >
> > > > > I agree with what you say, however, if you consider a call =
originating from
> > > > > a SIP device (telephone or softphone) that ends up going =
through a gateway
> > > > > to the PSTN then it is a different story.
> > > > > Neither the user or the UAC can know where a call will end up, =
therefore the
> > > > > ability to be able to provide both SIP address and SIP =
Telephone number
> > > > > would be useful to providing a complete service as then an =
appropriate
> > > > > calling address is available wherever the call ends up. In the =
scenario I
> > > > > described the gateway could chose the Telephone number (ok I =
know there are
> > > > > verification issues but that's another story).
> > > > > Something like Simon's idea of an "ALSO-FROM" seems a good =
idea to me (this
> > > > > is similar to "two number delivery" in the ISDN CLIP service) =
older versions
> > > > > would just ignore the ALSO-FROM and it wouldn't affect the =
significance of
> > > > > the FROM and TO headers.
> > > > >
> > > > > Clive
> > > > >
> > > > > > -----Original Message-----
> > > > > > From: Jonathan Rosenberg [SMTP:jdrosen@dynamicsoft.com]
> > > > > > Sent: Wednesday, October 11, 2000 7:41 PM
> > > > > > To:   'clive.dellard@bt.com'; 'sip@lists.bell-labs.com'
> > > > > > Subject:      RE: [SIP] From header to ISUP CgPN mapping
> > > > > >
> > > > > >
> > > > > >
> > > > > >
> > > > > > > -----Original Message-----
> > > > > > > From: clive.dellard@bt.com [mailto:clive.dellard@bt.com]
> > > > > > > Sent: Wednesday, October 11, 2000 12:17 PM
> > > > > > > To: sip@lists.bell-labs.com
> > > > > > > Subject: [SIP] From header to ISUP CgPN mapping
> > > > > > >
> > > > > > >
> > > > > > > In the call flows document =
draft-ietf-sip-call-flows-01.txt,
> > > > > > > scenario 4.1.1,
> > > > > > > the message detail and text show User A using a SIP =
telephone
> > > > > > > number rather
> > > > > > > than a SIP address in the From header. The text at the
> > > > > > > beginning of section
> > > > > > > 4 also says "that User A still uses his/her SIP URL in the
> > > > > > > Contact header,
> > > > > > > since the call could be redirected back to the SIP =
network."
> > > > > > > This implies that User A has pre-knowledge that the call =
is
> > > > > > > going to route
> > > > > > > to the PSTN and has the ability to chose the address to be
> > > > > > > used in the From
> > > > > > > header.
> > > > > >
> > > > > > No.
> > > > > >
> > > > > > The From header is the logical identity of the user making =
the call. Since
> > > > > > the call is coming through a gateway, that user is some user =
on a PSTN
> > > > > > terminal somewhere. The identity in this case can either be =
a tel URL, or
> > > > > > a
> > > > > > SIP URL at the domain of the originating network.
> > > > > >
> > > > > > The Contact is used for signaling addressing. Its not a =
logical
> > > > > > identifier.
> > > > > > It answers the question "where should I send SIP messages to =
for the rest
> > > > > > of
> > > > > > this call". The SIP spec says this is supposed to be a SIP =
URL preferably
> > > > > > with a complete hostname, so that there are no ambiguities =
about where to
> > > > > > send requests.
> > > > > >
> > > > > > Neither of these have anything to do with knowing where the =
call is going.
> > > > > >
> > > > > > -Jonathan R.
> > > > > >
> > > > > > ---
> > > > > > Jonathan D. Rosenberg                       72 Eagle Rock =
Ave.
> > > > > > Chief Scientist                             First Floor
> > > > > > dynamicsoft                                 East Hanover, NJ =
07936
> > > > > > jdrosen@dynamicsoft.com                     FAX:   (973) =
952-5050
> > > > > > http://www.cs.columbia.edu/~jdrosen         PHONE: (973) =
952-5000
> > > > > > http://www.dynamicsoft.com
> > > > >
> > > > > _______________________________________________
> > > > > SIP mailing list
> > > > > SIP@lists.bell-labs.com
> > > > > http://lists.bell-labs.com/mailman/listinfo/sip
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Hi,

>>>I think the re-INVITE mechanism is far more better because the =
>information about what session is refreshed >>>is carried in the SDP of =
the re-INVITE. The INFO approach has the significant disadvantage that =
>proxies >>>that use the session timer might have to store a *lot* of =
state information about each session.
>>That is not true. The session is identified using the Call-ID.
>The call-id is the SIP-identifier of a session, yes. But, SIP
>initiates a multimedia session and if the proxy is doing stuff (like
>allocating some sort of network resources for a limited period of
>time) related to the multimedia session and is interested in knowing =
when
>that session ends it might add a session timer. Now, if all the
>information that set up the session gets repeated in the re-INVITE the
>proxy does not have to store call-id etc together with the multimedia
>session parameters to track which multimedia session the
>SIP-message refers to.

Well, that is true, but that is a very application specific issue. I
don't think it's a big problem for the proxy to store the necessary
session data (the endpoints have to store it too, to be able to resend
it in the re-INVITEs), together with the Call-Id (you need to store the
Call-ID in any case), compared to the fact that you save lots of
resources, both in the network and due to the fact that the logic of the
endpoints gets more simple.

Regards,

Christer Holmberg
Ericsson Finland


>=20
> Regards,
>=20
> /Lars
>=20
> >
> > In fact, another advantage of using INFO is that you don't have to =
send
> > a message body at all. Using re-INVITE, you must send a copy of the
> > original message, and the UAS must check if the SDP is a copy
> > (session-timer) or if it is a new SDP, when the session should be
> > modified. Again, we save resources. Comments?
> >
> > Regards,
> >
> > Christer Holmberg
> > Ericsson Finland
> >
> >
> >
> > >
> > > Regards /Lars
> > >
> > > > Regards,
> > > >
> > > > Christer Holmberg
> > > > Ericsson Finland
> > >
> > > Lars Berggren       <mailto:lars.berggren@intertex.se>
> > > Intertex Data AB    tel: +46-8-6282828
> > > Sundbyberg, Sweden  fax: +46-8-6286414
>=20
> Lars Berggren       <mailto:lars.berggren@intertex.se>
> Intertex Data AB    tel: +46-8-6282828
> Sundbyberg, Sweden  fax: +46-8-6286414
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Hi,

I would like to clarify one issue in the SIP Caller Preferences draft
(draft-ietf-sip-callerprefs-02.txt).

The case is when I talk to a SIP server acting as both proxy and
redirect. I want it to be able to send me 3xx responses ("redirect"),
and I want it to be able to act like a proxy ("proxy"). Also, if it
forks a request, and receives a 3xx response, I want it to be forwarded
back to me ("no-recurse").

The draft says (chapter 5.5 Request-Disposition) that the
recurese-feature is ignored if redirect has been requested, so I can not
use the following header:

Request-Disposition: proxy, redirect, no-recurse

Should I instead use two different headers in the message? I.e.:

Request-Disposition: proxy, no-recurse
Request-Disposition: redirect

(Note: I left the parallel- and fork-feature from the example on
purpose)


Regards,

Christer Holmberg
Ericsson Finland
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Hi,

>You can use URL headers to specify SIP message headers, e.g., the =
Contact >in the 302 may look like
>=20
>Contact: sip:chirster@ericsson?from=3Dtel:12345
>=20
>It's then up to the UAC to trust your redirect server and overwrite its
>default From header. See section 2 of 2543bis for details.

True, but in this case the redirect server (what I mean by that is
actually any SIP node, server or UA, with redirect functionality...)
will not know if the UAC is going to use the new header or not. For some
reason, the redirect server may behave in different ways (maybe, in some
case, it won't even allow the UAC to contact a certain host if the
feature is not supported...) depending on if it can give the UAC a new
>From header or not.

But, if we have a new header for this, the UAC can indicate in the
Supported header that it supports this feature. Then, the redirect
server can add the new "Also-From" header in the response, and indicate
in the Require header that the UAC must use it. Maybe this is possible
also in the case where you just add it to the Contact header, but in any
case I think you would have to define a new Supported/Require tag value
(?)

Regards,

Christer Holmberg
Ericsson Finland



>=20
> ---
> Igor Slepchin
>=20
> > -----Original Message-----
> > From: Christer Holmberg [mailto:christer.holmberg@lmf.ericsson.se]
> > Sent: Tuesday, October 24, 2000 5:39 AM
> > To: sip@lists.bell-labs.com
> > Cc: sean.olson@ericsson.com
> > Subject: Re: [SIP] Redirect server returning new From header
> >
> >
> >
> > Hi,
> >
> > >Do you actually mean the From: or do you mean the To:?
> >
> > From.
> >
> > >It seems a bit odd for a redirect server to tell the UAC what
> > >From: to use. Perhaps you could give an example of its usage.
> >
> > Well, this is - once again - related to the PSTN/SIP GW
> > issue, if the GW
> > needs a telephone number for some kind of A number analysis.
> > When I make
> > a phone call from my SIP phone I may not know that the call may end =
up
> > on the PSTN side, so I can for example put my email address
> > in the From
> > header. However, since the network architecture includes a GW to the
> > PSTN network, somewhere there could be a server which is able
> > to do the
> > mapping between the email address user part and a telephone
> > number user
> > part (if SIP URL is used).
> >
> > Regards,
> >
> > Christer Holmberg
> > Ericsson Finland
> >
> >
> > >
> > > Regards,
> > > Sean
> > >
> > > --
> > > Sean Olson <sean.olson@ericsson.com>
> > >
> > > Christer Holmberg wrote:
> > >
> > > > Another proposal on the issue, with another use of the =
"Also-From"
> > > > header.
> > > >
> > > > Currently, a redirect server add a new Request-URI (in
> > the 3xx response
> > > > Contact header) to the UAC. An idea would be that it
> > could also return a
> > > > new From header, for example with a telephone number. In
> > this case, the
> > > > UAC, and intermediate proxies, would not need to insert any =
extra
> > > > headers or URI parameters in the request. This would be a
> > new, optional,
> > > > feature in a redirect server.
> > > >
> > > > Comments?
> > > >
> > > > Christer Holmberg
> > > > Ericsson Finland
> >
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> I thought we had put this issue to bed.
>
> As Henning pointed out previously, allowing a UAC to insert a =
telephone
> number into a request, which is then passed to the PSTN (via a =
gateway) as
> the originating number, is a recipe for a security disaster. Nothing =
stops
> me from inserting fake telephone numbers, and calling people =
pretending to
> be someone else. You can expect a call from the FCC about that one.
>
> So, in fact, this proposed used of the Also-From header is, in =
practice,
> unusable without a massive security/PKI infrastructure which really =
doesn't
> exist (you would need someone to certify that joe@foo.com owns the =
number
> 555-1212. I don't even know who would be able to adequately certify =
this,
> and I suspect no CA in their right mind would ever put this in a
> certificate).

This security infrastructure already exists in the current telephone =
network -
although it is implemented in hardware rather than software. Phone =
companies have a
network of trust with each other, and exchange ISUP messages with each =
other only
secure lines setup along these lines of trust. Where signalling is =
exchanged with an
end user (Q.931) checks are made to ensure the user is authorised to =
send the numbers
he sends. A similar trust network is required if we want to have =
authenticated CLI in
SIP. Currently SIP-T does require exactly the same (physical or secure =
VPN) trust
network as regular ISUP telephony. Indeed this is probably one of the =
main
constraining factors that limits whom SIP-T is available to.

Your telephone company will be very well positioned to certify that you =
have the
right to present your telephone number. I would suggest that it will be =
your phone
company, or a third party acting on their behalf who will do so.

Authenticated CLI in SIP may play a very important part in keeping SIP =
spamming down.

Simon Barber

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Hi,

>First off, I'd like to agree with others here that this is a rehash of =
>whats already been discussed, and decided, in the past. If we revisit =
>every decision every few months as new people join the lists, we'll =
never >make progress.
>=20
>The use of INVITE along with the SDP has numerous benefits above just =
the
>INFO approach. It allows for crash recoveries, it allows for refreshes =
of
>session state in intermediate devices, such as proxies, and it follows =
>the soft-state model present in other similar protocols. Thus, it is =
more
>general purpose of a solution. Thats been our goal with SIP - to have =
>fairly general purpose components that could be used to build a variety =
>of different services.=20

I don't want to change the current draft, because it does provide a very
mechanism to keep sessions alive, and I agree that it would not take us
anywhere if we change things every time someone has a different opinion.

My point it is very "heavy", both on the network and the endpoint logic,
if the only thing I want is a session heartbeat on the signaling level.
Of course, I can personally chose to send INFOs wherever I want, but I
don't
think that is a good thing to do. And, that way no negotiation of the
timeout value, and of which party should send the messages, is possible,
why it would be much better to have a standard way on how to do it.


Regards,

Christer Holmberg
Ericsson Finland

>I'll agree INFO would eliminate a message, but compared to media, =
>signaling is still negligable. We've been through that one umpteen =
times, >and the horse is very dead.
>=20
>As for the question about how to make sure session timers don't keep =
>getting resent if the UAS doesn't support them, thats easy. It would =
>include a Require header in the request, listing "timer". If the UAS =
>doesn't support session timer, the request is rejected (no session is =
>setup either). Now, the UAC knows that the terminating UA doesn't =
support >the timer. So, it can either start the session without session =
timer at >all by sending a new INVITE with no Supported header and no =
>Session-Expires header, give up, or do session timer anyway if the =
>proxies want it (Supported header, no Session-Expires header).
>=20
>Brett also asks if the UAC can ask to not have the session revived if =
the
>UAS isn't active. The question really is, who gets to make that =
decision? >Is it realy the UAC who should ultimately decide whether a =
session >restart is appropriate? Since its really the UAS that needs to =
do it, it >seems like it should be its responsibility to decide.
>=20
> -Jonathan R.
>=20
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>=20
>=20
> > -----Original Message-----
> > From: Brett Tate [mailto:brett@broadsoft.com]
> > Sent: Tuesday, October 17, 2000 6:57 PM
> > To: sip@lists.bell-labs.com
> > Subject: Re: [SIP] Session-timer with INFO
> >
> >
> > > While it is true that there are other ways that this could have =
been
> > solved,
> > > it is pretty late in the game to make this kind of a
> > change.  Especially
> > if
> > > the only reason for doing so is to avoid the ACK message.
> >
> > Other reasons include the desire not to revive a
> > dead session and to avoid the possibility of a new
> > SDP in the 200 response.
> >
> > >
> > > As a side note, we are in the process of creating a new
> > version of the
> > draft
> > > that we hope will be used for working group last call.
> >
> > Steve,
> >
> > Can a header or parameter be defined within
> > the sip-session-timer to indicate that the sender
> > is just trying to extend the Session-Timer and
> > check session state, but the sender does NOT
> > want the session revived if it is down?  The
> > reverse of the boolean value could indicate a
> > request to revive the session if it is down.
> >
> >
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> >
>=20
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
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Date: Tue, 14 Nov 2000 17:12:07 +0530



Hi..
A possible additon to the torture test list

maddr = ipv6address

regards,
-ashok

-----------------
ashok i roy @hughes software systems



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"M. Ranganathan" wrote:
> 
> Tbanks for posting these (it has a laxative effect on my parser
> :).   In particular, I like the fact that you have posted
> expected return messages (it would be useful if the previous
> list also had such information). 

Please take note of Jonathan's advice about being liberal though.

> I have a couple of questions
> on the following header where you have used escape sequences:

There have been some considerable discussions on escaping SIP 
URLs in the past here. 

> Can the COLON in the sip: of the URI be an escaped sequence?
> 
> Second, you have used %40 in place of the @ at the end of the
> user name. Is this legal?  If this is legal, then   there is no
> way for the user to use an escape sequence for part of his or
> her name.

I think I have been over zealous here in my escaping.  
So I would suggest changing to the following start line:-

INVITE
sip:sip%3Auser%40example.com@company.com;other-param=summit
SIP/2.0

This re-raises a problem with section 2.1 of bis 02
an example SIP URL is a given as sip:juser@%65xample.com:5060 
I think this escaping within a host name is illegal.

Any corrections, better suggestions for this test most welcome.
Cheers,
Neil.
--
Ubiquity Software Corporation, UK        http://www.ubiquity.net

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Subject: Re: [SIP] new torture tests
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Date: Tue, 14 Nov 2000 11:40:57 +0000
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Jonathan Rosenberg wrote:
> 
> Comments below.
> 
> > The following are 15 new proposed torture test messages
> > based on the bis draft. None of them should cause good
> > implementations to crash. Recommended responses to the
> > illegal messages are given where needed.
> 
> Well, some of your messages have only mild errors in them. Based on the
> philosophy of being liberal with what you receive, I would not declare an
> entity to have failed the test if it does not generate the specific error
> code.

Good point, I will include text to this effect.

[snip]

> > 5) Illegal use of escaped header in Request-URI
> > Proxies should strip off the escaped header before proxying on.
> >
> > INVITE sip:user@company.com?Route=%3Csip:sip.example.com%3E
> > SIP/2.0
> > To: sip:user@company.com
> > From: sip:caller@university.edu
> > Call-ID: 5@10.0.0.1
> > CSeq: 1 INVITE
> > Via: SIP/2.0/UDP 135.180.130.133
> > Content-Type: application/sdp
> > Content-Length: 174
> >
> > v=0
> > o=mhandley 29739 7272939 IN IP4 126.5.4.3
> > s=SIP Call
> > t=3149328700 0
> > c=IN IP4 135.180.130.88
> > m=audio 492170 RTP/AVP 0 12
> > m=video 3227 RTP/AVP 31
> > a=rtpmap:31 LPC
> 
> I'm not sure I agree with the behavior here. If the proxy is not
> company.com, it may simple forward the request to company.com without
> stripping the Route header. I see nothing wrong with that.

Glad you think so too because this is what our proxy does. 
However, bis 02 says in 4.3 "A server which receives a 
SIP-URL with illegal elements removes them before further 
processing." So should this text be changed?

Cheers,
Neil
-- 
Ubiquity Software Corporation, UK        http://www.ubiquity.net

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From: "Keith Robinson" <Keith.Robinson@marconi.com>
To: Neil Deason <ndeason@ubiquity.net>
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Subject: Re: [SIP] new torture tests
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Date: Tue, 14 Nov 2000 12:11:38 +0000



I have the following comments/queries (based on bis-02)  on the proposed
additions for the torture test:

Message 2:

It is possible to parse this messge correctly even though it is strictly
illegal.

Message 3:

I don't think this message is illegal if reference is made to Appendix C which
states:

"All linear white space, including folding, has the same semantics as SP. A
recipient MAY replace any linear white space with a single SP before
interpreting the field value
or forwarding the message downstream"

Message 4:

Where does the spec imply that these characters can be replaced with their
escaped values in the context given ? In the case of the @ being replaced with
its escape sequence a conflict will occur if a client/proxy wants to use an
escaped @ in the user name, a situation described in the recent draft describing
registration in a visited domain.

Message 5:

Where does the spec imply that escaped headers cannot be used in the Reqest-URI
? The BNF for SIP URLs in section 2 gives the following rule:

headers = "?" header * ( "&" header )
header   = hname "=" hvalue
hname   = 1*(hnv-unreserved|unreserved|escaped)

which will allow escape seqences in SIP URL header fields.

Message 13:

The Call-ID: and last Via: headers span multiple lines but the LWS at the start
of the subsequent lines to indicate line folding (section 6.5 para. 1) is
missing.

Regards

K. Robinson















Neil Deason <ndeason@ubiquity.net> on 13/11/2000 17:56:04
                                                                                
                                                                                
                                                                                


                                                              
                                                              
                                                              
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 cc:      (bcc: Keith Robinson/MAIN/MC1)                      
                                                              
                                                              
                                                              
 Subject: [SIP] new torture tests                             
                                                              








The following are 15 new proposed torture test messages
based on the bis draft. None of them should cause good
implementations to crash. Recommended responses to the
illegal messages are given where needed.

I would greatly appreciate any additions/corrections to be
made quickly so we can try to get these folded into a new
release of the call flows draft before the next bakeoff.

Cheers,
Neil.
--
Ubiquity Software Corporation, UK        http://www.ubiquity.net


1) Illegal enclosing of Request-URI  in "<>"
Server should respond with 400 + Appropriate Reason Phrase.

INVITE <sip:user@company.com> SIP/2.0
To: sip:user@company.com
From: sip:caller@university.edu
Call-ID: 1@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.180.130.133
Content-Type: application/sdp
Content-Length: 174

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC


2) Illegal LWS in elements of Request-URI
Server should respond with 400 + Appropriate Reason Phrase.

INVITE sip:user@company.com; transport=udp SIP/2.0
To: sip:user@company.com
From: sip:caller@university.edu
Call-ID: 2@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.180.130.133
Content-Type: application/sdp
Content-Length: 174

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC


3) Illegal >1 SP between elements of Request URI
Server should respond with 400 + Appropriate Reason Phrase.

INVITE sip:user@company.com  SIP/2.0
To: sip:user@company.com
From: sip:caller@university.edu
Call-ID: 3@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.180.130.133
Content-Type: application/sdp
Content-Length: 174

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC


4) Legal SIP URL in Request URI with escaped characters and
parameter.

INVITE sip%3Auser%40company.com;other-param=summit SIP/2.0
To: sip:user@company.com
From: sip:caller@university.edu
Call-ID: 4@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.180.130.133
Content-Type: application/sdp
Content-Length: 174

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC


5) Illegal use of escaped header in Request-URI
Proxies should strip off the escaped header before proxying on.

INVITE sip:user@company.com?Route=%3Csip:sip.example.com%3E
SIP/2.0
To: sip:user@company.com
From: sip:caller@university.edu
Call-ID: 5@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.180.130.133
Content-Type: application/sdp
Content-Length: 174

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC

6) Not understood URI in Req-URI but a SIP URL in To.
Server should respond 400 + Appropriate Reason Phrase

INVITE name:user SIP/2.0
To: sip:user@company.com
From: sip:caller@university.edu
Call-ID: 6@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.180.130.133
Content-Type: application/sdp
Content-Length: 174

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC


7) Legal no LWS between display name and <

OPTIONS sip:user@company.com SIP/2.0
To: sip:user@company.com
From: "caller"<sip:caller@example.com>
Call-ID: 1234abcd@10.0.0.1
CSeq: 1 OPTIONS
Via: SIP/2.0/UDP 135.180.130.133

8) Legal extra LWS between display name and <

OPTIONS sip:user@company.com SIP/2.0
To: sip:user@company.com
From: "caller"    <sip:caller@example.com>
Call-ID: 1234abcd@10.0.0.1
CSeq: 2 OPTIONS
Via: SIP/2.0/UDP 135.180.130.133

9) Illegal Non GMT time zone in SIP Date
A liberal server should correct this.

INVITE sip:user@company.com SIP/2.0
To: sip:user@company.com
From: sip:caller@university.edu
Call-ID: 7@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.180.130.133
Expires: Fri, 01 Jan 2010 16:00:00 EST
Content-Type: application/sdp
Content-Length: 174

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC

10) Legal message with passed Exipres date.
A Server should respond 408 Request Timeout.

INVITE sip:user@company.com SIP/2.0
To: sip:user@company.com
From: sip:caller@university.edu
Call-ID: 8@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.180.130.133
Expires: Thu, 01 Dec 1994 16:00:00 GMT
Content-Type: application/sdp
Content-Length: 174

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC


11) Legal message with zero max forwards
Proxies should return 483 Too Many Hops, a UAS responds
as for OPTIONS.

INVITE sip:user@company.com SIP/2.0
To: sip:user@company.com
From: sip:caller@university.edu
Call-ID: 9@10.0.0.1
CSeq: 1 INVITE
Via: SIP/2.0/UDP 135.180.130.133
Max-Forwards: 0
Content-Type: application/sdp
Content-Length: 174

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC

12) Legal REGISTER with SIP URL containing escaped Route header
in Contact.

REGISTER sip:company.com SIP/2.0
To: sip:user@company.com
From: sip:user@company.com
Contact: sip:user@host.company.com
Call-ID: k345asrl3fdbv@10.0.0.1
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 135.180.130.133
Contact: sip:user@example.com?Route=%3Csip:sip.example.com%3E

13) Legal but long, includes Token for Call-ID.

INVITE sip:user@company.com SIP/2.0
To: "I have a user name of extreme proportion"
<sip:user@company.com:6000;other-param=1234567890somethingelselong1234567890>
From: sip:caller@university.edu
Call-ID:
kl24ahsd546folnyt2vbak9sad98u23naodiunzds09a3bqw0sdfbsk34poouymnae0043nsed09mfkvc74bd0cuwnms05dknw87hjpobd76f

CSeq: 1 INVITE
My-State: sldkjflzdsfaret0803adgaasd0afds0asdaasd
Via: SIP/2.0/UDP sip33.example.com
Via: SIP/2.0/UDP sip32.example.com
Via: SIP/2.0/UDP sip31.example.com
Via: SIP/2.0/UDP sip30.example.com
Via: SIP/2.0/UDP sip29.example.com
Via: SIP/2.0/UDP sip28.example.com
Via: SIP/2.0/UDP sip27.example.com
Via: SIP/2.0/UDP sip26.example.com
Via: SIP/2.0/UDP sip25.example.com
Via: SIP/2.0/UDP sip24.example.com
Via: SIP/2.0/UDP sip23.example.com
Via: SIP/2.0/UDP sip22.example.com
Via: SIP/2.0/UDP sip21.example.com
Via: SIP/2.0/UDP sip20.example.com
Via: SIP/2.0/UDP sip19.example.com
Via: SIP/2.0/UDP sip18.example.com
Via: SIP/2.0/UDP sip17.example.com
Via: SIP/2.0/UDP sip16.example.com
Via: SIP/2.0/UDP sip15.example.com
Via: SIP/2.0/UDP sip14.example.com
Via: SIP/2.0/UDP sip13.example.com
Via: SIP/2.0/UDP sip12.example.com
Via: SIP/2.0/UDP sip11.example.com
Via: SIP/2.0/UDP sip10.example.com
Via: SIP/2.0/UDP sip9.example.com
Via: SIP/2.0/UDP sip8.example.com
Via: SIP/2.0/UDP sip7.example.com
Via: SIP/2.0/UDP sip6.example.com
Via: SIP/2.0/UDP sip5.example.com
Via: SIP/2.0/UDP sip4.example.com
Via: SIP/2.0/UDP sip3.example.com
Via: SIP/2.0/UDP sip2.example.com
Via: SIP/2.0/UDP sip1.example.com
Via: SIP/2.0/UDP
host.example.com;received=135.180.130.133;branch=C1C3344E2710000000E299E568E7potato10potato0potato0

Content-Type: application/sdp

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s=SIP Call
t=3149328700 0
c=IN IP4 135.180.130.88
m=audio 492170 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC


14) Illegal, badly mangled message with mutliple To,
From, Call-ID and CSeq headers.
Server should respond 400 + Appropriate Reason Phrase

OPTIONS sip:135.180.130.133 SIP/2.0
Via: SIP/2.0/UDP company.com:5604
From: sip:iuser@company.com
To: sip:user@135.180.130.133
Call-ID: 1804928587@company.com
CSeq: 1 OPTIONS
Expires: 0 0l@company.com
To: sip:user@135.180.130.133
Call-ID: 1804928587@company.com
CSeq: 1 OPTIONS
Contact: sip:host.company.com
Expires: 0xpires: 0sip:host.company.com
Expires: 0
Contact: sip:host.company.com

15) Legal message with an unusual amount of SDP attributes.

INVITE sip:+1-650-555-2222@ss1.wcom.com;user=phone SIP/2.0
Via: SIP/2.0/UDP iftgw.there.com:5060
From: sip:+1-303-555-1111@ift.here.com;user=phone
To: sip:+1-650-555-2222@ss1.wcom.com;user=phone
Call-ID: 1717@ift.here.com
CSeq: 56 INVITE
Content-Type: application/sdp
Content-Length: 320

v=0
o=faxgw1 2890844527 2890844527 IN IP4 iftgw.there.com
s=Session SDP
c=IN IP4 iftmg.there.com
t=0 0
m=image 49172 udptl t38
a=T38FaxVersion:0
a=T38maxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:260
a=T38FaxUdpEC:t38UDPRedundancy

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Subject: Re: [SIP] new torture tests
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Date: Tue, 14 Nov 2000 13:09:40 +0000
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Keith Robinson wrote:
> 
> I have the following comments/queries (based on bis-02)  on the proposed
> additions for the torture test:

Thanks for the comments.

> Message 2:
> 
> It is possible to parse this messge correctly even though it is strictly
> illegal.

If you are clever enough you can parse anything ;) 
But is a fair point that following the "be liberal on what 
you accept .." is good and responding 400 to this would be 
harsh.

> Message 3:
> 
> I don't think this message is illegal if reference is made to Appendix C which
> states:
> 
> "All linear white space, including folding, has the same semantics as SP. A
> recipient MAY replace any linear white space with a single SP before
> interpreting the field value
> or forwarding the message downstream"

Hmm this is a MAY and indeed being liberal and accepting this 
is highly desirable. Going strictly from section 4.1 I read
this as being illegal. Authors, parser implementers care to 
comment on the matter?

> Message 4:
> 
> Where does the spec imply that these characters can be replaced with their
> escaped values in the context given ? In the case of the @ being replaced with
> its escape sequence a conflict will occur if a client/proxy wants to use an
> escaped @ in the user name, a situation described in the recent draft describing
> registration in a visited domain.

Yeah I got carried away on this one. I intend to change it to :

sip:sip%3Auser%40example.com@company.com;other-param=summit

> Message 5:
> 
> Where does the spec imply that escaped headers cannot be used in the Reqest-URI
> ? The BNF for SIP URLs in section 2 gives the following rule:
> 
> headers = "?" header * ( "&" header )
> header   = hname "=" hvalue
> hname   = 1*(hnv-unreserved|unreserved|escaped)
> 
> which will allow escape seqences in SIP URL header fields.

See Table 2 which indicates escaped headers are not allowed in 
the Req	URI.

> Message 13:
> 
> The Call-ID: and last Via: headers span multiple lines but the LWS at the start
> of the subsequent lines to indicate line folding (section 6.5 para. 1) is
> missing.

That is an unfortunate effect of my mail client's formatting.

Cheers,
Neil.
-- 
Ubiquity Software Corporation, UK        http://www.ubiquity.net

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There are two scenarios.

1. If A does not want to conference in C rather maintain two seperate calls
with B and C, then you're correct, A should generate a different call-Id for
C. 

2. If A wants to bring C into a conference call on a call going on with B,
then it shouldn't generate a new call-id for C. A can invite C with a Also
header telling B is also in a call, so C can also establish a signalling
relationship with B.

Hope this help...

>  -----Original Message-----
> From: 	sip-admin@lists.bell-labs.com
> [mailto:sip-admin@lists.bell-labs.com] 
> Sent:	Tuesday, November 14, 2000 8:46 AM
> To:	sip@lists.bell-labs.com
> Subject:	[SIP] Call-ID
> 
> 
> Dear all
> 
> 	I am just a beginner of SIP. I am a bit confuse about Call-ID. I am
> not sure that I clearly understand Call-ID or I miss understanding it.
> 	Is this example correct?
> 
> 	Suppose that A@A.COM would like to invite B@B.COM then it generates
> a INVITE request with a Call-ID as 111@A.COM.
> 	Then, if A@A.COM also want to invite C@C.COM then it generates
> another INVITE request with another Call-ID such as 222@A.COM.
> 
> 	Is it correct?
> 
> POP

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Subject: [SIP] query on multicast conference ...
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Hi,
     I am having one query regarding bahavior of the SIP user agent.
     Let terminal A,B and C are in a multicast conference.
     Terminal D wants to join the conference so it sends an INVITE
messege to
      terminal A with same session description.

             Can some one please tell me what could be the behavior of
terminal A,

    1. will it accept the INVITE and send Ack to terminal D.

        OR

    2. Reject the INVITE and  send RE - INVITE  to all terminals A, B
and D.

   Thanks in advance...

 rakesh jain


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Subject: Re: [SIP] new torture tests
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Date: Tue, 14 Nov 2000 15:18:50 +0100
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Neil Deason wrote:
> 
> Keith Robinson wrote:
> >
> > Message 2:
> >
> > It is possible to parse this messge correctly even though it is strictly
> > illegal.
> 
> If you are clever enough you can parse anything ;)
> But is a fair point that following the "be liberal on what
> you accept .." is good and responding 400 to this would be
> harsh.

I don't know about that. I believe it is the case that URLs can *never*
contain unencoded space characters. If this is true then the test seems
very reasonable.

--
Anders Kristensen

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From: Anders Kristensen <akristensen@dynamicsoft.com>
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Neil Deason wrote:
> 
> The following are 15 new proposed torture test messages
> based on the bis draft. None of them should cause good
> implementations to crash. Recommended responses to the
> illegal messages are given where needed.
> 
> I would greatly appreciate any additions/corrections to be
> made quickly so we can try to get these folded into a new
> release of the call flows draft before the next bakeoff.

I've included some below.

> 
> 12) Legal REGISTER with SIP URL containing escaped Route header
> in Contact.
> 
> REGISTER sip:company.com SIP/2.0
> To: sip:user@company.com
> From: sip:user@company.com
> Contact: sip:user@host.company.com
> Call-ID: k345asrl3fdbv@10.0.0.1
> CSeq: 1 REGISTER
> Via: SIP/2.0/UDP 135.180.130.133
> Contact: sip:user@example.com?Route=%3Csip:sip.example.com%3E

Actually, this request is illegal.

The bis-02 draft says about the Contact header (section 6.14):

   Even if the "display-name" is empty, the "name-addr" form
   MUST be used if the "addr-spec" contains a comma, semicolon,
   or question mark.

The SIP URI in the Contact header includes a '?' char, and so the
Contact value needs to be in addr-spec form:

Contact: <sip:user@example.com?Route=%3Csip:sip.example.com%3E>

Had the "extra stuff" been a parameter instead of a header it would be
parsed as a Contact parameter rather than a url-parameter. This would be
fine from the parsers point of view but is a common source of error
nonetheless.

In fact, this type of error is common enough to make it worth exercising
explicitly in the tests. I propose instead adding the following tests...


12a) Legal REGISTER with SIP URL containing escaped Route header
in Contact.

REGISTER sip:company.com SIP/2.0
To: sip:user@company.com
From: sip:user@company.com
Contact: sip:user@host.company.com
Call-ID: k345asrl3fdbv@10.0.0.1
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 135.180.130.133
Contact: <sip:user@example.com?Route=%3Csip:sip.example.com%3E>


12b) Illegal REGISTER with Contact containing header.
Server should respond with 400 + Appropriate Reason Phrase.

REGISTER sip:company.com SIP/2.0
To: sip:user@company.com
From: sip:user@company.com
Contact: sip:user@host.company.com
Call-ID: k345asrl3fdbv@10.0.0.1
CSeq: 1 REGISTER
Via: SIP/2.0/UDP 135.180.130.133
Contact: sip:user@example.com?Route=%3Csip:sip.example.com%3E

12c) Legal but probably doesn't mean what was intended. The 'user'
parameter should be interpreted as being a contact-param and
not a url-parameter.  The request should succeed but a subsequent
retrieval of the registration must NOT include "user=phone" as a
url-parameter.

REGISTER sip:bell-tel.com SIP/2.0
Via: SIP/2.0/UDP saturn.bell-tel.com
From: sip:watson@bell-tel.com
To: sip:watson@bell-tel.com
Call-ID: 70710@saturn.bell-tel.com
CSeq: 2 REGISTER
Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone

12d) Same as 12c but with this Contact header:

Contact: <sip:+1-972-555-2222@gw1.wcom.com>;user=phone


Other candidates...

16) Legal: display name in From/To/Contact contains multiple tokens and
is unqouted:

INVITE sip:t.watson@ieee.org SIP/2.0
Via:     SIP/2.0/UDP c.bell-tel.com
From:    A. Bell <sip:a.g.bell@bell-tel.com>
To:      T. Watson <sip:t.watson@ieee.org>
Call-ID: 31415@c.bell-tel.com
CSeq:    1 INVITE

17) Illegal: unquoted display name in From/To/Contact contains non-token
characters.
Server should respond with 400 + Appropriate Reason Phrase.

INVITE sip:t.watson@ieee.org SIP/2.0
Via:     SIP/2.0/UDP c.bell-tel.com
From:    Bell, Alexander <sip:a.g.bell@bell-tel.com>
To:      Watson, Thomas <sip:t.watson@ieee.org>
Call-ID: 31415@c.bell-tel.com
CSeq:    1 INVITE

18) Unknown (higher) protocol version in request line.
Server should respond with 400 + Appropriate Reason Phrase.

INVITE sip:t.watson@ieee.org SIP/7.0
Via:     SIP/2.0/UDP c.bell-tel.com
From:    A. Bell <sip:a.g.bell@bell-tel.com>
To:      T. Watson <sip:t.watson@ieee.org>
Call-ID: 31415@c.bell-tel.com
CSeq:    1 INVITE

19) Legal leading zero in protocol version numbers

INVITE sip:t.watson@ieee.org SIP/02.00
Via:     SIP/2.0/UDP c.bell-tel.com
From:    A. Bell <sip:a.g.bell@bell-tel.com>
To:      T. Watson <sip:t.watson@ieee.org>
Call-ID: 31415@c.bell-tel.com
CSeq:    1 INVITE


Cheers,
Anders

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Date: Tue, 14 Nov 2000 08:54:03 -0600

Mohammad Vakil (mvakil@sipcomm.com):

> 2. If A wants to bring C into a conference call on a call going on
> with B, then it shouldn't generate a new call-id for C. A can invite C
> with a Also header telling B is also in a call, so C can also
> establish a signalling relationship with B.

  Sorry, but the use of the Also header has been deprecated.  I doubt
you will find anyone who implements this or ever intends to.  It has
been deemed a "bad idea".

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Tue, 14 Nov 2000 10:46:57 -0500

I believe the difference is fundmentally one of control.

You use a proxy when you want to control processing of the call from the
point you receive it, and forward. A proxy can see the provisional and final
responses to the request. It can record route so that it sees whats going on
during the call. A redirect server, however, hands off control to the device
that sent it the INVITE. It will never see the final response to the
request, and not be contacted again for the remainder of the call. Its a
one-shot deal.

As such, redirect servers are really good for high volume, lookup style
transactions. They can almost be considered a form of database query, albeit
a SIP specific one. Proxies are generally needed for services and more
complex routing problems.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Billy Biggs [mailto:Billy_Biggs@3com.com]
> Sent: Monday, November 13, 2000 8:38 PM
> To: Schmidlin, David
> Cc: sip@lists.bell-labs.com
> Subject: Re: [SIP] Redirect VS Proxy Server
> 
> 
> Schmidlin, David (david.schmidlin@intel.com):
> 
> > Why/Where would you use a redirect server over a proxy 
> server and vice
> > versa?
> 
>   If you were a web server, when would you send a redirect and when
> would you proxy?  It's the same answer.
> 
>   Redirect servers are cheap to code, for one.  But these are logical
> roles not separate entities, so your question should be more specific.
> 
>   In general you proxy a message when you are providing some 
> additional
> feature rather than simply performing a lookup.
> 
> -- 
> Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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Date: Tue, 14 Nov 2000 16:14:27 +0000



I am not suggesting that it is legal to put unencoded spaces in the Request-URI;
I realise this is strictly illegal. The
point I am making is that if a parser knows it has got the Request-URI because
it is the text that lies between the Method and SIP-Version
elements it  can remove all unencoded spaces precisely because of the rule that
URIs cannot contain unencoded white spaces. If a
parser can do this without jeopardising the rest of the lexical analysis then it
would be harsh to determine that it is in error if it does not
return a 400 response but goes onto handle the request in a normal and correct
manner.

The type of parser that MUST return an error is one that is checking the output
of proxies/agents to make sure they do format messages strictly, i.e., checking
message generation as opposed to message parsing.









Anders Kristensen <akristensen@dynamicsoft.com> on 14/11/2000 14:18:50
                                                                                
                                                                                
                                                                                


                                                              
                                                              
                                                              
 To:      Neil Deason <ndeason@ubiquity.net>                  
                                                              
 cc:      Keith Robinson/MAIN/MC1@MCMAIN, Sip List            
          <sip@lists.bell-labs.com>                           
                                                              
                                                              
                                                              
 Subject: Re: [SIP] new torture tests                         
                                                              









Neil Deason wrote:
>
> Keith Robinson wrote:
> >
> > Message 2:
> >
> > It is possible to parse this messge correctly even though it is strictly
> > illegal.
>
> If you are clever enough you can parse anything ;)
> But is a fair point that following the "be liberal on what
> you accept .." is good and responding 400 to this would be
> harsh.

I don't know about that. I believe it is the case that URLs can *never*
contain unencoded space characters. If this is true then the test seems
very reasonable.

--
Anders Kristensen

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From: "Maddux, Michel" <Michel.Maddux@COMPAQ.com>
To: sip@lists.bell-labs.com
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Subject: [SIP] SIP - H.323 - INVITE - No media line.
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Date: Tue, 14 Nov 2000 15:00:29 -0600



In constructing a gateway connecting Sip to 323, the process of signalling a
call and then negotiating channels, for a Sip caller to 323 callee, is
relatively straightforward. Only one temporizing step is required, and that
is to not provide an RTCP address in opening channels from gateway (actually
Sip) to 323 callee.

The reverse case, of implementing 323 caller to Sip callee, is not so
straightforward. Signalling is tractable, but media channel negotiation is
murky. Two models occur:

1) When 323 calls, wait until it opens first media channel. Then issue the
INVITE, noting only a media line of that media type but with a phony port.
When Sip replies (200 OK presumed), complete the 323 media channel being
negotiated, giving it the Sip's media port. Then negotiate the reverse
channel with 323, saving its port, which is then transmitted to Sip via an
ACK with a message body correcting the 323 media port. If the 323 caller
then opens another media channel, repeat the above process via re-INVITE,
and continue to do so as long as 323 requests media channels.

2) Depend upon language from the 2543 spec, particularly section B.4 and the
ACK section, which state that an INVITE could be sent with NO media lines,
upon receipt of which the Sip callee would respond with a list of its
supported media (AND presumably providing a port for each of these). In this
scenario, an INVITE devoid of media would be issued immediately on receipt
of call from the 323 caller. The resulting 200 OK media list would then be
used to negotiate the media channels with the 323 caller, and would be
returned in final form in the message body of the ACK. [It is possible that
this model would devolve to the first model since it is not known when the
323 caller is through opening channels. Should a mid-call media change be
initiated by the 323 caller, the re-INVITE process would need be used,
anyway.]

We are interested in a recommendation as to how to do this. Are either of
the above models viable, or does a better model exist? What is the spec
status of section B.4?

_______________________________________________
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From: "Maddux, Michel" <Michel.Maddux@COMPAQ.com>
To: "'Billy Biggs'" <Billy_Biggs@3com.com>,
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Date: Tue, 14 Nov 2000 15:35:14 -0600

What is the best method then, if Also is deprecated? thanks. /m.


Michel A. Maddux
Senior IP Architect



-----Original Message-----
From: Billy Biggs [mailto:Billy_Biggs@3com.com]
Sent: Tuesday, November 14, 2000 7:54 AM
To: Mohammad Vakil
Cc: sip@lists.bell-labs.com
Subject: Re: [SIP] Call-ID


Mohammad Vakil (mvakil@sipcomm.com):

> 2. If A wants to bring C into a conference call on a call going on
> with B, then it shouldn't generate a new call-id for C. A can invite C
> with a Also header telling B is also in a call, so C can also
> establish a signalling relationship with B.

  Sorry, but the use of the Also header has been deprecated.  I doubt
you will find anyone who implements this or ever intends to.  It has
been deemed a "bad idea".

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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--------------76858BE3241BF9867A9908A9
Content-Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit

The annswers to most of your question in developing such a gateway you
can find in:

http://search.ietf.org/internet-drafts/draft-singh-sip-h323-01.txt

Have fun. :)

"Maddux, Michel" wrote:

> In constructing a gateway connecting Sip to 323, the process of signalling a
> call and then negotiating channels, for a Sip caller to 323 callee, is
> relatively straightforward. Only one temporizing step is required, and that
> is to not provide an RTCP address in opening channels from gateway (actually
> Sip) to 323 callee.
>
> The reverse case, of implementing 323 caller to Sip callee, is not so
> straightforward. Signalling is tractable, but media channel negotiation is
> murky. Two models occur:
>
> 1) When 323 calls, wait until it opens first media channel. Then issue the
> INVITE, noting only a media line of that media type but with a phony port.
> When Sip replies (200 OK presumed), complete the 323 media channel being
> negotiated, giving it the Sip's media port. Then negotiate the reverse
> channel with 323, saving its port, which is then transmitted to Sip via an
> ACK with a message body correcting the 323 media port. If the 323 caller
> then opens another media channel, repeat the above process via re-INVITE,
> and continue to do so as long as 323 requests media channels.
>
> 2) Depend upon language from the 2543 spec, particularly section B.4 and the
> ACK section, which state that an INVITE could be sent with NO media lines,
> upon receipt of which the Sip callee would respond with a list of its
> supported media (AND presumably providing a port for each of these). In this
> scenario, an INVITE devoid of media would be issued immediately on receipt
> of call from the 323 caller. The resulting 200 OK media list would then be
> used to negotiate the media channels with the 323 caller, and would be
> returned in final form in the message body of the ACK. [It is possible that
> this model would devolve to the first model since it is not known when the
> 323 caller is through opening channels. Should a mid-call media change be
> initiated by the 323 caller, the re-INVITE process would need be used,
> anyway.]
>
> We are interested in a recommendation as to how to do this. Are either of
> the above models viable, or does a better model exist? What is the spec
> status of section B.4?
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

--
Vladislav Zubarev        mailto:vladis@vovida.com
Software Engineer
Vovida Networks, Inc.    http://www.vovida.com



--------------76858BE3241BF9867A9908A9
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
The annswers to most of your question in developing such a gateway you
<br>can find in:
<p><A HREF="http://search.ietf.org/internet-drafts/draft-singh-sip-h323-01.txt">http://search.ietf.org/internet-drafts/draft-singh-sip-h323-01.txt</A>
<p>Have fun. :)
<p>"Maddux, Michel" wrote:
<blockquote TYPE=CITE>In constructing a gateway connecting Sip to 323,
the process of signalling a
<br>call and then negotiating channels, for a Sip caller to 323 callee,
is
<br>relatively straightforward. Only one temporizing step is required,
and that
<br>is to not provide an RTCP address in opening channels from gateway
(actually
<br>Sip) to 323 callee.
<p>The reverse case, of implementing 323 caller to Sip callee, is not so
<br>straightforward. Signalling is tractable, but media channel negotiation
is
<br>murky. Two models occur:
<p>1) When 323 calls, wait until it opens first media channel. Then issue
the
<br>INVITE, noting only a media line of that media type but with a phony
port.
<br>When Sip replies (200 OK presumed), complete the 323 media channel
being
<br>negotiated, giving it the Sip's media port. Then negotiate the reverse
<br>channel with 323, saving its port, which is then transmitted to Sip
via an
<br>ACK with a message body correcting the 323 media port. If the 323 caller
<br>then opens another media channel, repeat the above process via re-INVITE,
<br>and continue to do so as long as 323 requests media channels.
<p>2) Depend upon language from the 2543 spec, particularly section B.4
and the
<br>ACK section, which state that an INVITE could be sent with NO media
lines,
<br>upon receipt of which the Sip callee would respond with a list of its
<br>supported media (AND presumably providing a port for each of these).
In this
<br>scenario, an INVITE devoid of media would be issued immediately on
receipt
<br>of call from the 323 caller. The resulting 200 OK media list would
then be
<br>used to negotiate the media channels with the 323 caller, and would
be
<br>returned in final form in the message body of the ACK. [It is possible
that
<br>this model would devolve to the first model since it is not known when
the
<br>323 caller is through opening channels. Should a mid-call media change
be
<br>initiated by the 323 caller, the re-INVITE process would need be used,
<br>anyway.]
<p>We are interested in a recommendation as to how to do this. Are either
of
<br>the above models viable, or does a better model exist? What is the
spec
<br>status of section B.4?
<p>_______________________________________________
<br>SIP mailing list
<br>SIP@lists.bell-labs.com
<br><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></blockquote>

<pre>--&nbsp;
Vladislav Zubarev&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <A HREF="mailto:vladis@vovida.com">mailto:vladis@vovida.com</A>
Software Engineer&nbsp;&nbsp;&nbsp;
Vovida Networks, Inc.&nbsp;&nbsp;&nbsp; <A HREF="http://www.vovida.com">http://www.vovida.com</A></pre>
&nbsp;</html>

--------------76858BE3241BF9867A9908A9--


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Date: Tue, 14 Nov 2000 16:47:35 -0500



 

> -----Original Message-----
> From: Rakesh Jain [mailto:rakesh.jain@wipro.com]
> Sent: Saturday, October 14, 2000 10:20 AM
> To: sip@lists.bell-labs.com
> Subject: [SIP] query on multicast conference ...
> 
> 
> 
> Hi,
>      I am having one query regarding bahavior of the SIP user agent.
>      Let terminal A,B and C are in a multicast conference.
>      Terminal D wants to join the conference so it sends an INVITE
> messege to
>       terminal A with same session description.

Thats not what you would do for a large scale multicast conference.

There is a lot of confusion these days around conferences, largely because
there are several different models for how they operate. Some of these
models work with SIP today, and others require extensions.

I am preparing an I-D, due out the end of this week, that goes over the
various types of multiparty conferences you can have, how they work, how
they differ, etc. Keep posted.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Date: Tue, 14 Nov 2000 14:09:03 -0800

Hi SIP Experts,
	Here is a question that I have about a certain call flow that I am
trying to implement. Please look at the call flows and let me know what is
wrong in this picture.

     A			Proxy			B			C

     INVITE------------> 
				INVITE---------->
					<--------- 100 Trying
				      <--------- 180 Ringing
	<----------------- 180 Ringing	
				CANCEL ---------->
					<----------- 200 OK
					<----------- 487
				
				INVITE ----------------------------->
					<------------------------------ 100
Trying
					<------------------------------ 180
Ringing
	<----------------- 180 Ringing
					<------------------------------ 200
OK
	<----------------- 200 OK

	What I have noticed with the devices that I am testing this call
flow is that somehow they dont like the 180/200 OK that come in from C. Is
there a reason  for this ? I would love to know this.


Thanks a lot 


Sekhar Banerjee			  



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From: Billy Biggs <Billy_Biggs@3com.com>
To: "Maddux, Michel" <Michel.Maddux@COMPAQ.com>
Cc: Mohammad Vakil <mvakil@sipcomm.com>, sip@lists.bell-labs.com
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Date: Tue, 14 Nov 2000 16:12:46 -0600

Maddux, Michel (Michel.Maddux@COMPAQ.com):

> What is the best method then, if Also is deprecated? thanks. /m.

  The scenario asked about was a distributed conference:

> > 2. If A wants to bring C into a conference call on a call going on
> > with B, then it shouldn't generate a new call-id for C. A can invite
> > C with a Also header telling B is also in a call, so C can also
> > establish a signalling relationship with B.

  This form of conferencing, where each participant has a distinct
signalling relationship with each other, is difficult to manage.  It
seems that people are currently just implementing simpler forms of
conferences using MCUs, 3-Way calling, and handoff/transfer using REFER.

  Also was originally defined in the sip-cc-01 call control draft, which
is long expired.  It was not continued due to the complexity of the
draft, the many unresolved issues, and the dangerous overloading of
INVITE/BYE.

  Some further investigation was done by some folks at Dialogic in the
dmcs draft, but after discussion at the IETF, the 4th Bakeoff, and on
the mailing list, it was decided that further call control extensions be
defined more explicitly using new methods where applicable.  At least
that was my understanding.

  This philosophy is outlined in the Call Control Framework draft,
draft-campbell-sip-cc-framework-01.txt

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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From: Vijay Gurbani <vkg@lucent.com>
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Date: Tue, 14 Nov 2000 16:30:37 -0600
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Sekhar Banerjee wrote:
> 
> Hi SIP Experts,
>         Here is a question that I have about a certain call flow that I am
> trying to implement. Please look at the call flows and let me know what is
> wrong in this picture.
> 
>      A                  Proxy                   B                       C
> 
>      INVITE------------>
>                                 INVITE---------->
>                                       <--------- 100 Trying
>                                       <--------- 180 Ringing
>         <----------------- 180 Ringing
>                                 CANCEL ---------->
>                                         <----------- 200 OK
>                                         <----------- 487
> 
>                                 INVITE ----------------------------->
>                                         <------------------------------ 100
> Trying
>                                         <------------------------------ 180
> Ringing
>         <----------------- 180 Ringing
>                                         <------------------------------ 200
> OK
>         <----------------- 200 OK
> 
>         What I have noticed with the devices that I am testing this call
> flow is that somehow they dont like the 180/200 OK that come in from C. Is
> there a reason  for this ? I would love to know this.

There is no reason why A should reject a 180/200 OK from C; since it never
got a final response from B, a call leg has not yet been established at A.
On getting a 200 OK from C, a call leg should be established at A.

However, I do see a bug in your call flow; note your call flow:

A         Proxy      B              C
INV-------->
           INV------>
           <--------100
           <--------180
<---------180
           CANCEL----->
           <-----------200 OK
           <-----------487

           INV--------------------->

In your call flow, your proxy starts a second INVITE to C; it never sends an
ACK to B; it should.  Logically, B's INVITE is still pending and it will
continue re-transmitting 487 until ACK'ed or maximum response retransmission
limit has reached (which is 7 for an INVITE, if I am not mistaken). 

I also assume that the 200 OK from B->Proxy is for Cancel (you do not indicate 
this in your call flow).  If this is not a correct assumption and the 200 OK is 
for the INVITE, and since you want to CANCEL the transaction, you should ACK 
the 200 OK and then send a BYE.

- vijay
--
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Internet Software and eServices Group 
Lucent Technologies/Bell Labs Innovations 263 Shuman Blvd., Rm 1A-413
Naperville, Illinois 60566     Voice: +1 630 224 0216   Fax: +1 630 713 0184

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From sip-admin@lists.bell-labs.com  Tue Nov 14 18:03:03 2000
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From: Sekhar Banerjee <SBanerjee@lboard.com>
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Date: Tue, 14 Nov 2000 15:07:17 -0800

Hi Vijay,
	


Sekhar Banerjee wrote:
> 
> Hi SIP Experts,
>         Here is a question that I have about a certain call flow that I am
> trying to implement. Please look at the call flows and let me know what is
> wrong in this picture.
> 
>      A                  Proxy                   B                       C
> 
>      INVITE------------>
>                                 INVITE---------->
>                                       <--------- 100 Trying
>                                       <--------- 180 Ringing
>         <----------------- 180 Ringing
>                                 CANCEL ---------->
>                                         <----------- 200 OK
>                                         <----------- 487
> 
>                                 INVITE ----------------------------->
>                                         <------------------------------
100
> Trying
>                                         <------------------------------
180
> Ringing
>         <----------------- 180 Ringing
>                                         <------------------------------
200
> OK
>         <----------------- 200 OK
> 
>         What I have noticed with the devices that I am testing this call
> flow is that somehow they dont like the 180/200 OK that come in from C. Is
> there a reason  for this ? I would love to know this.
>Vijay wrote :
>There is no reason why A should reject a 180/200 OK from C; since it never
>got a final response from B, a call leg has not yet been established at A.
>On getting a 200 OK from C, a call leg should be established at A.


>However, I do see a bug in your call flow; note your call flow:

>A         Proxy      B              C
>INV-------->
>           INV------>
>           <--------100
>           <--------180
><---------180
>           CANCEL----->
>           <-----------200 OK
>           <-----------487
>           INV--------------------->

>In your call flow, your proxy starts a second INVITE to C; it never sends
an
>ACK to B; it should.  Logically, B's INVITE is still pending and it will
>continue re-transmitting 487 until ACK'ed or maximum response
retransmission
>limit has reached (which is 7 for an INVITE, if I am not mistaken). 
    Well .... that was something that I forgot to put in the diagram. I do
finish the INVITE transaction with the ACK.   

>I also assume that the 200 OK from B->Proxy is for Cancel (you do not
indicate 
>this in your call flow).  If this is not a correct assumption and the 200
OK is 
>for the INVITE, and since you want to CANCEL the transaction, you should
ACK 
>the 200 OK and then send a BYE.
   Yes ... the 200 OK is for the cancel.

   But my concern is with the tags that get attached with the responses
comming in from the end devices. Could that be the problem. I mean just
because the first 180 ringing had a different tag in the To field  than in
the second 180 ringing. Is that significant enough ? 



Thanks

Sekhar 

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Date: Tue, 14 Nov 2000 17:15:37 -0600

Sekhar Banerjee (SBanerjee@lboard.com):

>    But my concern is with the tags that get attached with the
> responses comming in from the end devices. Could that be the
> problem. I mean just because the first 180 ringing had a different
> tag in the To field  than in the second 180 ringing. Is that
> significant enough ? 

  The UA should have no problem seeing different tags on responses.

  If the UA can't handle it, it is broken.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Sekhar Banerjee wrote:
[...]
>    Yes ... the 200 OK is for the cancel.
> 
>    But my concern is with the tags that get attached with the responses
> comming in from the end devices. Could that be the problem. I mean just
> because the first 180 ringing had a different tag in the To field  than in
> the second 180 ringing. Is that significant enough ?

Not enough to make the UAC misbehave; the tags on the provisional responses
from different UASs will be different.  As I said earlier, the call leg has 
not yet been established until a final response arrives at the UAC.  The 200 
OK from C should be accepted by A.

- vijay
--
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Internet Software and eServices Group 
Lucent Technologies/Bell Labs Innovations 263 Shuman Blvd., Rm 1A-413
Naperville, Illinois 60566     Voice: +1 630 224 0216   Fax: +1 630 713 0184

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Date: Tue, 14 Nov 2000 20:06:42 -0500

> 1) When 323 calls, wait until it opens first media channel. 
> Then issue the
> INVITE, noting only a media line of that media type but with 
> a phony port.
> When Sip replies (200 OK presumed), complete the 323 media 
> channel being
> negotiated, giving it the Sip's media port. Then negotiate the reverse
> channel with 323, saving its port, which is then transmitted 
> to Sip via an
> ACK with a message body correcting the 323 media port. 

This is illegal in SIP (see section 4.2.1 in 2543bis). You should use
re-INVITE if you want to modify the session offered in INVITE request.

Thank you,
Igor Slepchin

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Again, call scenarios with different issues resolving (like this) can
be found in the following document:

http://search.ietf.org/internet-drafts/draft-singh-sip-h323-01.txt

Thanx.

Igor Slepchin wrote:

> > 1) When 323 calls, wait until it opens first media channel.
> > Then issue the
> > INVITE, noting only a media line of that media type but with
> > a phony port.
> > When Sip replies (200 OK presumed), complete the 323 media
> > channel being
> > negotiated, giving it the Sip's media port. Then negotiate the reverse
> > channel with 323, saving its port, which is then transmitted
> > to Sip via an
> > ACK with a message body correcting the 323 media port.
>
> This is illegal in SIP (see section 4.2.1 in 2543bis). You should use
> re-INVITE if you want to modify the session offered in INVITE request.
>
> Thank you,
> Igor Slepchin
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

--
Vladislav Zubarev        mailto:vladis@vovida.com
Software Engineer
Vovida Networks, Inc.    http://www.vovida.com



--------------E51163373267F2F75774436D
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Again, call scenarios with different issues resolving (like this) can
<br>be found in the following document:
<p><A HREF="http://search.ietf.org/internet-drafts/draft-singh-sip-h323-01.txt">http://search.ietf.org/internet-drafts/draft-singh-sip-h323-01.txt</A>
<p>Thanx.
<p>Igor Slepchin wrote:
<blockquote TYPE=CITE>> 1) When 323 calls, wait until it opens first media
channel.
<br>> Then issue the
<br>> INVITE, noting only a media line of that media type but with
<br>> a phony port.
<br>> When Sip replies (200 OK presumed), complete the 323 media
<br>> channel being
<br>> negotiated, giving it the Sip's media port. Then negotiate the reverse
<br>> channel with 323, saving its port, which is then transmitted
<br>> to Sip via an
<br>> ACK with a message body correcting the 323 media port.
<p>This is illegal in SIP (see section 4.2.1 in 2543bis). You should use
<br>re-INVITE if you want to modify the session offered in INVITE request.
<p>Thank you,
<br>Igor Slepchin
<p>_______________________________________________
<br>SIP mailing list
<br>SIP@lists.bell-labs.com
<br><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></blockquote>

<pre>--&nbsp;
Vladislav Zubarev&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <A HREF="mailto:vladis@vovida.com">mailto:vladis@vovida.com</A>
Software Engineer&nbsp;&nbsp;&nbsp;
Vovida Networks, Inc.&nbsp;&nbsp;&nbsp; <A HREF="http://www.vovida.com">http://www.vovida.com</A></pre>
&nbsp;</html>

--------------E51163373267F2F75774436D--


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Date: Tue, 14 Nov 2000 19:25:36 -0800

> 
> 12d) Same as 12c but with this Contact header:
> 
> Contact: <sip:+1-972-555-2222@gw1.wcom.com>;user=phone
> 
Can we add a torture test with a more taxing telephone subscriber
Request-URI. It is important that proxies and clients process these
correctly (even if they just ignore them or return a 404 Not Found).

INVITE
sip:+1-972-555-2222;phone-context=name%40domain;new=user?%22Route:%20X%40Y%3
bZ=W%22@gw1.wcom.com;user=phone SIP/2.0

Cheers,

Robert

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Date: Tue, 14 Nov 2000 20:30:13 -0800

> > 
> > 12d) Same as 12c but with this Contact header:
> > 
> > Contact: <sip:+1-972-555-2222@gw1.wcom.com>;user=phone
> > 
> Can we add a torture test with a more taxing telephone subscriber
> Request-URI. It is important that proxies and clients process these
> correctly (even if they just ignore them or return a 404 Not Found).
> 
> INVITE
> sip:+1-972-555-2222;phone-context=name%40domain;new=user?%22Ro
> ute:%20X%40Y%3
> bZ=W%22@gw1.wcom.com;user=phone SIP/2.0
> 
Actually, I think the colon after Route needs to be quoted as well.

Robert.
 

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Date: Wed, 15 Nov 2000 03:44:22 -0800 (PST)

Dear All:
   I want to consult one question, RFC2543 bis 02 as
for Digest Authentication referenced RFC2617, On
RFC2617,     where it is listed as the following:
  challenge        =  "Digest" digest-challenge
  digest-challenge  = 1#( realm | [ domain ] | nonce |
                 [ opaque ] |[ stale ] | [ algorithm ]
|              [ qop-options ] | [auth-param] )
      domain            = "domain" "=" <"> URI ( 1*SP
URI ) <">
      URI               =  Request-URI
      nonce             = "nonce" "=" nonce-value
      nonce-value       = quoted-string
      opaque            = "opaque" "=" quoted-string
      stale             = "stale" "=" ( "true" |
"false" )
      algorithm         = "algorithm" "=" ( "MD5" |
"MD5-sess" |
                           token )
      qop-options       = "qop" "=" <"> 1#qop-value
<">
      qop-value         = "auth" | "auth-int" | token

    I have no idea on MD5-sess, could anyone give me 
some information on MD5-sess, I will try to find the 
detailed description on MD5-see.
    Thanks in advance!
    Best regards!
                              Yours: Jim

   


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Subject: [SIP] Old Torture Tests
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Hello.

I think that "test7.txt" is wrong, the text says:

    This request message contains a new method, NEWMETHOD.
    A proxy should forward this using the same retransmission
    rules as INVITE. A UAS should reject it with an error,
    and list the available methods in the response.

My thinking is that the retransmission rules should be based on
the nominal retransmission rules; i.e., the ones in 10.4.1,
which describe behaviour for requests other than INVITE and ACK.

Should "test10.txt" now be deprecated, since multiplemessages
per datagram has been removed?  (This was done, wasn't it; I
didn't dream it? (:&)  Ditto "test18.txt"?

Cheers,


 - Jo.
-- 
  Jo Hornsby; IT Professional         --       mailto:jhornsby@ubiquity.net
 Ubiquity Software Corporation        --         http://www.ubiquity.net/


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Date: Tue, 14 Nov 2000 07:58:22 -0800 (PST)
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You need to be careful about what you mean by "end
to end".  Proxy-routed e-2-e traffic cannot use
TLS. Only an application layer protocol such as
PGP can do that, and you still probably want to
use hopwise crypto to protect the outer headers
from tampering and snooping.

If you're talking about hopwise crypto -- which
may lead to an end to end security association
with the two UA's -- TLS and IPsec are the usual
suspects. TLS doesn't work for UDP, so IPsec is
somewhat more general.

	      Mike

Tom Tang writes:
 > 
 >  Hello :
 > 
 >    I am wondering what mechanisms companies are planning to 
 >  deploy for SIP security.  I've been hearing PGP and TLS 
 >  mainly, but I didn't see any postings for extensions to 
 >  either protocols.  Please tell me if I've skipped over some
 >  draft. Thanks.
 > 
 >  - Tom Tang
 > 
 > 
 > Tom Tang
 > IP Unity
 > 1575 McCandless Drive
 > Milpitas, CA 95035
 > (408) 957-0800 x154 (w)
 > (408) 957-0823 (f) 
 > 
 > _______________________________________________
 > SIP mailing list
 > SIP@lists.bell-labs.com
 > http://lists.bell-labs.com/mailman/listinfo/sip
 > 

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Hello,

I am Al Costanzo and I wrote ID <draft-costanzo-dns-gl-03.txt>, it was
suggested to me to to bring it to the attention of this WG for
consideration.  If you are interested in the draft, I would be more than
willing with the working group to modify it to allow it to do what you need.

Regards,

Al Costanzo

---Exerpt of the email I received:

You may already be aware of it, your I-D <draft-costanzo-dns-gl-03.txt> on
Definition of the DNS GL Resource Record is of high interest to the IP
communications community. Especially, if you have any additional ideas how
to use this service for 911 calls from IP SIP phones. Suggest to bring it to
attention of the
IETF SIP WG list - sip@lists.bell-labs.com,
SIP Forum list - SIP Forum Discussion - discussion@sipforum.org

-----------------------------------------------------
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Date: Wed, 15 Nov 2000 09:15:45 -0500



 

> -----Original Message-----
> From: Jo Hornsby [mailto:jhornsby@ubiquity.net]
> Sent: Wednesday, November 15, 2000 8:11 AM
> To: sip@lists.bell-labs.com
> Subject: [SIP] Old Torture Tests
> 
> 
> Hello.
> 
> I think that "test7.txt" is wrong, the text says:
> 
>     This request message contains a new method, NEWMETHOD.
>     A proxy should forward this using the same retransmission
>     rules as INVITE. A UAS should reject it with an error,
>     and list the available methods in the response.
> 
> My thinking is that the retransmission rules should be based on
> the nominal retransmission rules; i.e., the ones in 10.4.1,
> which describe behaviour for requests other than INVITE and ACK.

Correct. This needs to be fixed.

> 
> Should "test10.txt" now be deprecated, since multiplemessages
> per datagram has been removed?  (This was done, wasn't it; I
> didn't dream it? (:&)  Ditto "test18.txt"?

Correct. In fact, the correct behavior now for test18 would be to reject
that request. Now that multiple messages per UDP datagram is deprecated, the
correct behavior for when the body is longer than the Content-Length is to
reject the request.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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Date: Wed, 15 Nov 2000 09:13:56 -0500



 

> -----Original Message-----
> From: Vijay Gurbani [mailto:vkg@lucent.com]
> Sent: Tuesday, November 14, 2000 6:57 PM
> To: Sekhar Banerjee
> Cc: Sip List
> Subject: Re: [SIP] What am I doing wrong ?
> 
> 
> Sekhar Banerjee wrote:
> [...]
> >    Yes ... the 200 OK is for the cancel.
> > 
> >    But my concern is with the tags that get attached with 
> the responses
> > comming in from the end devices. Could that be the problem. 
> I mean just
> > because the first 180 ringing had a different tag in the To 
> field  than in
> > the second 180 ringing. Is that significant enough ?
> 
> Not enough to make the UAC misbehave; the tags on the 
> provisional responses
> from different UASs will be different.  As I said earlier, 
> the call leg has 
> not yet been established until a final response arrives at 
> the UAC.  The 200 
> OK from C should be accepted by A.

Well, the UAC shouldn't misbehave, but that doesn't mean it isn't
misbehaving. This case (of getting different tags in the provisionals than
in the final) is probably not well tested in many implementations.

Its probably worth adding a torture test case for this.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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Jonathan Rosenberg wrote:
[...]
> Well, the UAC shouldn't misbehave, but that doesn't mean it isn't
> misbehaving. This case (of getting different tags in the provisionals than
> in the final) is probably not well tested in many implementations.
> 
> Its probably worth adding a torture test case for this.

Sure; I can write one up.  Who is maintaining the "Torture Document?"  I
can send it to him/her directly.

- vijay

- vijay
--
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Internet Software and eServices Group 
Lucent Technologies/Bell Labs Innovations 263 Shuman Blvd., Rm 1A-413
Naperville, Illinois 60566     Voice: +1 630 224 0216   Fax: +1 630 713 0184

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Vijay Gurbani wrote:
> 
> Jonathan Rosenberg wrote:
> [...]
> > Well, the UAC shouldn't misbehave, but that doesn't mean it isn't
> > misbehaving. This case (of getting different tags in the provisionals than
> > in the final) is probably not well tested in many implementations.
> >
> > Its probably worth adding a torture test case for this.
> 
> Sure; I can write one up.  Who is maintaining the "Torture Document?"  I
> can send it to him/her directly.
> 

I'm keeping the torture files. Once the discussion has died down, I'd
appreciate if somebody could send them to me so that I can add them to
the bakeoff web page.

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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We're moving computer equipment today, so it is possible that the SIP
web pages will be temporarily unavailable. There is no need to send me
email about this.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Vijay Gurbani wrote:
> 
> Jonathan Rosenberg wrote:
> [...]
> > Well, the UAC shouldn't misbehave, but that doesn't mean it isn't
> > misbehaving. This case (of getting different tags in the provisionals than
> > in the final) is probably not well tested in many implementations.
> >
> > Its probably worth adding a torture test case for this.
> 
> Sure; I can write one up.  Who is maintaining the "Torture Document?"  I
> can send it to him/her directly.

I am collecting together new torture tests for submission
to the call flows work and Henning's web site. They
need to be out in time for the next bakeoff.

Cheers,
Neil.
-- 
Ubiquity Software Corporation, UK        http://www.ubiquity.net

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All,

Now that the JAIN SIP API is using interfaces instead of classes for
messages and headers, the implementations of these interfaces will have
their own parsing mechanisms. Currently the API set methods throw
IllegalArgumentExceptions if the user attempts to use a null or
zero-length parameter. The question now is should an implementation
throw an IllegalArgument if it doesn't like the value, or should we
create an exception for anything implementation-specific.

An example of this is the AcceptLanguageHeader. The API allows any
String that is not null or zero-length be used as the language range.
Obviously an implementation should be stricter on the content of this
String.

The reason we are so tolerant of parameter values was that the headers
were originally defined as classes in the API, and we did not want to
include a full SIP parser in the API. However, now that interfaces are
used, an alternative would be to simply extend the definition of an
illegal argument - for example, the implementation should throw an
IllegalArgumentException if the language range does not conform to the
RFC. (Of course testing this would be another matter entirely!)

Another separate issue is - what should an implementation do if receives

something from the network that it cannot parse correctly? Should the
JAIN SIP application be made aware of it, or should the implementation
automatically send back the appropriate response to the network? This
obviously depends on whether the implementation is lazy-parsing or
parsing the entire message up front, and we have to accomodate both
scenarios (this is after all one of the main reasons we moved from
classes to interfaces)

Your opinions are of course welcomed.

Regards,
Chris Harris






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From: Alan Johnston <alan.johnston@wcom.com>
Subject: Re: [SIP] SIP ringback tone
To: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
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Robert,

In actual implementations, often a 183 will actually be sent during ringing, due
to the fact that often a gateway will be unable to determine definitively if
ringing or some other information is being sent in-band.  My concern is that
this reading will cause gateway vendors to assume that ringing is taking place,
send a 180 with SDP.  Then, the User Agent may also assume that ringing is
taking place, not play the early media stream to the user. (The bis spec
currently only says SHOULD.)  If the call then receives treatment, the user will
then have not idea why the call failed.

I agree that a User Agent should not have to know that it is talking to a
gateway to the PSTN.  What it does need to know is that media described in the
SDP of a 183 response MUST be played to the user, or the final outcome of a call
may not be known. (Since 183 is a new response code, there aren't any backwards
compatibility issues here, are there?)

Thanks,
Alan.

"Fairlie-Cuninghame, Robert" wrote:

> The way the spec is written at the moment (see Section 7.1 - it changed
> recently), a UAC should be able to accept any 1xx response (other than 100)
> with SDP as indicating early media. eg 183, 182, 181 or 180. Obviously they
> must have different meanings or there is no reason to have the different
> response codes (COMET aside). Both the expired 183 draft (explicitly) and
> the bis-02 draft (inferred) say that 183 indicates call progress events
> _other_ than user alerting (eg "queued message", "please wait, verifying" or
> "enter pin number" voice messages) --- yet the mechanism is the same for 180
> and 183. [I just think it should be made clearer in the 183 description.]
>
> I don't understand your point about requiring support for two mechanisms and
> that it "masks whether I'm going to the PSTN or another SIP endpoint
> identified by a telephone number". How so? And more importantly, why is this
> necessary ?
>
> In fact I would say the opposite is true. A SIP client that is not PSTN
> centric and doesn't bother with early media playback should not have to know
> that 183 is actually ringback on a PSTN SIP client. The client should simply
> know that a received 180 indicates user alerting, 182 is queued and 183 is
> some other call progress event.
>
> Regards,
>
> Robert.
>
> -- My opinions are my own. I tried selling them once but everybody
>         seems to already have one. --
> > -----Original Message-----
> > From: David Shrader [mailto:dshrader@master-consultant.com]
> > Sent: Monday, November 13, 2000 9:54 PM
> > To: Fairlie-Cuninghame, Robert
> > Cc: SIP List
> > Subject: Re: [SIP] SIP ringback tone
> >
> >
> > Robert,
> >
> > I don't agree that 183 with SDP should only be a call
> > progress message. It
> > is a perfectly reasonable usage to support ringback. This is
> > especially
> > necessary when interworking with the PSTN where ringback is delivered
> > inband. If you implement a system that works with the PSTN, I think we
> > shouldn't require it to support two modes of operation. It
> > seems reasonable
> > that the use of 183 masks whether I'm going to the PSTN or another SIP
> > endpoint identified by a telephone number and allows the
> > person or network
> > being called to provide whatever inband indication it chooses.
> >
> > David
> >
> > ----------------------------
> > David Shrader
> > Master Consultant, Inc.
> > dshrader@master-consultant.com
> > http://www.EyeForTheFuture.com
> >
> > > From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
> > > Date: Mon, 13 Nov 2000 03:33:35 -0800
> > > To: "'Henry Sinnreich'" <Henry.Sinnreich@wcom.com>
> > > Cc: "'sip@lists.bell-labs.com'" <sip@lists.bell-labs.com>
> > > Subject: RE: [SIP] SIP ringback tone
> > >
> > > Henry,
> > >
> > > Innovative ringback greetings definitely seem like a case
> > for an early media
> > > to me.
> > >
> > > As an aside, many people still seem to erroroneously think
> > that 183 Session
> > > Progress should be used for early-media ring-back. 180
> > Ringing with SDP
> > > should be used for ringback purposes, 183 with SDP should
> > only be used for
> > > call progress messages (ie, not alerting the user). I still
> > think this needs
> > > to be made clearer in the spec!
> > >
> > > Cheers,
> > >
> > > Robert.
> > >
> > > -- My opinions are my own. I tried selling them once but everybody
> > > seems to already have one. --
> > >
> > >> -----Original Message-----
> > >> From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
> > >> Sent: Monday, November 13, 2000 6:18 AM
> > >> To: adam.roach@ericsson.com; IETF SIP (E-mail)
> > >> Subject: [SIP] SIP ringback tone
> > >>
> > >>
> > >> The draft on "Ringback tones in SIP-Based Telephony" shows
> > >> impressive work
> > >> on preserving legacy country specific ring back tones. I
> > >> wonder if we would
> > >> not be better of by adding a
> > >>
> > >> SIP Ringback Tone
> > >> =================
> > >>
> > >> With some optional fields where service providers could
> > add innovative
> > >> greetings.
> > >>
> > >> Henry
> > >>
> > >> Henry Sinnreich
> > >> WorldCom
> > >> 400 International Parkway,
> > >> Richardson, Texas, 75081
> > >>
> > >>
> > >>
> > >> _______________________________________________
> > >> SIP mailing list
> > >> SIP@lists.bell-labs.com
> > >> http://lists.bell-labs.com/mailman/listinfo/sip
> > >>
> > >
> > > _______________________________________________
> > > SIP mailing list
> > > SIP@lists.bell-labs.com
> > > http://lists.bell-labs.com/mailman/listinfo/sip
> > >
> >
>
> _______________________________________________
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> SIP@lists.bell-labs.com
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Date: Wed, 15 Nov 2000 17:29:01 +0100

> -----Original Message-----
> From: Neil Deason [mailto:ndeason@ubiquity.net]
> Sent: 13 November 2000 18:56
> To: Sip List
> Subject: [SIP] new torture tests
> 
> The following are 15 new proposed torture test messages 
> based on the bis draft. None of them should cause good 
> implementations to crash. Recommended responses to the 
> illegal messages are given where needed.
> 
> I would greatly appreciate any additions/corrections to be
> made quickly so we can try to get these folded into a new 
> release of the call flows draft before the next bakeoff.
> 
> Cheers,
> Neil.

One test I realized is missing when I was at the last bake-off
is one that tests the following kind of To/From header:

	To: sip:pkj@axis.com ;tag=123456789@example.com

and makes sure the tag does not end up as a URL parameter.

I had not considered this legal before I got one like that 
sent to me (and studied the 2543 document for some time...)

//Peter

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Date: Wed, 15 Nov 2000 12:05:54 -0600

I think that Alan and Robert are "saying" the same thing.  My 
understanding is that a UAC, upon receipt of a 180 without 
 SDP, indicates to the user that the callee is being alerted.  
It can use whatever method it desires such as a pop-up 
message, playing an audible ringback tone, etc. to indicate 
this to the local user.  If the 180 included a SDP, the UAC can 
present the media stream or not.  However, I believe the UAC 
should cease its local indication (if any) upon receipt of a 183. 
The UAS sends a 183 (with SDP) to indicate that it will be 
sending early media (typically some call progress message).  
The local ringing indication should cease at the UAC so that 
the user can "hear" the early media (disposition of the call).

If the UAS wishes to provide ringback tone it can send a 180 
with SDP.  A 183 with SDP could be used too but why not use 
the Ringing response for ringback instead of a Session 
Progress response for ringback.  As Robert writes, a 180 
indicates a specific state.  The 180 with local alerting indication 
at the UAC minimizes network, UAC, and UAS resources 
during the setup phase and allows the UAC to locally indicate 
remote alerting however it likes.  If tone characteristics are 
present in a header of the 180 the UAC can "customize" its 
local alerting as Adam's Ringback I-D supports.

Regards,
Bert

-----Original Message-----
From: Alan Johnston [mailto:alan.johnston@wcom.com]
Sent: Wednesday, November 15, 2000 11:26 AM
To: Fairlie-Cuninghame, Robert
Cc: 'sip@lists.bell-labs.com'
Subject: Re: [SIP] SIP ringback tone


Robert,

In actual implementations, often a 183 will actually be sent during
ringing, due
to the fact that often a gateway will be unable to determine
definitively if
ringing or some other information is being sent in-band.  My concern
is that
this reading will cause gateway vendors to assume that ringing is
taking place,
send a 180 with SDP.  Then, the User Agent may also assume that
ringing is
taking place, not play the early media stream to the user. (The bis
spec
currently only says SHOULD.)  If the call then receives treatment, the
user will
then have not idea why the call failed.

I agree that a User Agent should not have to know that it is talking
to a
gateway to the PSTN.  What it does need to know is that media
described in the
SDP of a 183 response MUST be played to the user, or the final outcome
of a call
may not be known. (Since 183 is a new response code, there aren't any
backwards
compatibility issues here, are there?)

Thanks,
Alan.

"Fairlie-Cuninghame, Robert" wrote:

> The way the spec is written at the moment (see Section 7.1 - it
changed
> recently), a UAC should be able to accept any 1xx response (other
than 100)
> with SDP as indicating early media. eg 183, 182, 181 or 180.
Obviously they
> must have different meanings or there is no reason to have the
different
> response codes (COMET aside). Both the expired 183 draft
(explicitly) and
> the bis-02 draft (inferred) say that 183 indicates call progress
events
> _other_ than user alerting (eg "queued message", "please wait,
verifying" or
> "enter pin number" voice messages) --- yet the mechanism is the same
for 180
> and 183. [I just think it should be made clearer in the 183
description.]
>
> I don't understand your point about requiring support for two
mechanisms and
> that it "masks whether I'm going to the PSTN or another SIP endpoint
> identified by a telephone number". How so? And more importantly, why
is this
> necessary ?
>
> In fact I would say the opposite is true. A SIP client that is not
PSTN
> centric and doesn't bother with early media playback should not have
to know
> that 183 is actually ringback on a PSTN SIP client. The client
should simply
> know that a received 180 indicates user alerting, 182 is queued and
183 is
> some other call progress event.
>
> Regards,
>
> Robert.
>
> -- My opinions are my own. I tried selling them once but everybody
>         seems to already have one. --
> > -----Original Message-----
> > From: David Shrader [mailto:dshrader@master-consultant.com]
> > Sent: Monday, November 13, 2000 9:54 PM
> > To: Fairlie-Cuninghame, Robert
> > Cc: SIP List
> > Subject: Re: [SIP] SIP ringback tone
> >
> >
> > Robert,
> >
> > I don't agree that 183 with SDP should only be a call
> > progress message. It
> > is a perfectly reasonable usage to support ringback. This is
> > especially
> > necessary when interworking with the PSTN where ringback is
delivered
> > inband. If you implement a system that works with the PSTN, I
think we
> > shouldn't require it to support two modes of operation. It
> > seems reasonable
> > that the use of 183 masks whether I'm going to the PSTN or another
SIP
> > endpoint identified by a telephone number and allows the
> > person or network
> > being called to provide whatever inband indication it chooses.
> >
> > David
> >
> > ----------------------------
> > David Shrader
> > Master Consultant, Inc.
> > dshrader@master-consultant.com
> > http://www.EyeForTheFuture.com
> >
> > > From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
> > > Date: Mon, 13 Nov 2000 03:33:35 -0800
> > > To: "'Henry Sinnreich'" <Henry.Sinnreich@wcom.com>
> > > Cc: "'sip@lists.bell-labs.com'" <sip@lists.bell-labs.com>
> > > Subject: RE: [SIP] SIP ringback tone
> > >
> > > Henry,
> > >
> > > Innovative ringback greetings definitely seem like a case
> > for an early media
> > > to me.
> > >
> > > As an aside, many people still seem to erroroneously think
> > that 183 Session
> > > Progress should be used for early-media ring-back. 180
> > Ringing with SDP
> > > should be used for ringback purposes, 183 with SDP should
> > only be used for
> > > call progress messages (ie, not alerting the user). I still
> > think this needs
> > > to be made clearer in the spec!
> > >
> > > Cheers,
> > >
> > > Robert.
> > >
> > > -- My opinions are my own. I tried selling them once but
everybody
> > > seems to already have one. --
> > >
> > >> -----Original Message-----
> > >> From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
> > >> Sent: Monday, November 13, 2000 6:18 AM
> > >> To: adam.roach@ericsson.com; IETF SIP (E-mail)
> > >> Subject: [SIP] SIP ringback tone
> > >>
> > >>
> > >> The draft on "Ringback tones in SIP-Based Telephony" shows
> > >> impressive work
> > >> on preserving legacy country specific ring back tones. I
> > >> wonder if we would
> > >> not be better of by adding a
> > >>
> > >> SIP Ringback Tone
> > >> =================
> > >>
> > >> With some optional fields where service providers could
> > add innovative
> > >> greetings.
> > >>
> > >> Henry
> > >>
> > >> Henry Sinnreich
> > >> WorldCom
> > >> 400 International Parkway,
> > >> Richardson, Texas, 75081
> > >>
> > >>
> > >>
> > >> _______________________________________________
> > >> SIP mailing list
> > >> SIP@lists.bell-labs.com
> > >> http://lists.bell-labs.com/mailman/listinfo/sip
> > >>
> > >
> > > _______________________________________________
> > > SIP mailing list
> > > SIP@lists.bell-labs.com
> > > http://lists.bell-labs.com/mailman/listinfo/sip
> > >
> >
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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Subject: [SIP] Sip Grammars out there?
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Has anyone out there written a sip parser in yacc/bison?  I'm trying to find
a free grammar for it if anyone has.

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Not yacc but antlr (output language is java). Stay tuned... I will put one out
in a matter of 2 weeks  (I am documenting it and getting it approved for public
release).   It will be in th public domain (free).

Ranga.


David Frascone wrote:

> Has anyone out there written a sip parser in yacc/bison?  I'm trying to find
> a free grammar for it if anyone has.
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

--
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
Tel: 301 975 3664 Fax: 301 590 0932



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Not yacc but antlr (output language is java). Stay tuned... I&nbsp;will
put one out in a matter of 2 weeks&nbsp; (I&nbsp;am documenting it and
getting it approved for public release).&nbsp;&nbsp; It will be in th public
domain (free).
<p>Ranga.
<br>&nbsp;
<p>David Frascone wrote:
<blockquote TYPE=CITE>Has anyone out there written a sip parser in yacc/bison?&nbsp;
I'm trying to find
<br>a free grammar for it if anyone has.
<p>_______________________________________________
<br>SIP mailing list
<br>SIP@lists.bell-labs.com
<br><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></blockquote>

<pre>--&nbsp;
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.&nbsp;
Tel: 301 975 3664 Fax: 301 590 0932</pre>
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Chris Harris wrote:

> All,
>
> Now that the JAIN SIP API is using interfaces instead of classes for

Bravo!  Is it out for public review as yet?


>
> messages and headers, the implementations of these interfaces will have
> their own parsing mechanisms. Currently the API set methods throw
> IllegalArgumentExceptions if the user attempts to use a null or
> zero-length parameter. The question now is should an implementation
> throw an IllegalArgument if it doesn't like the value, or should we
> create an exception for anything implementation-specific.

I suggest we create an exception class for all parser specific exceptions
and have parsers subclass it to their taste (some may want to have finer
grained exceptions than others). How about a generic SIPParseException
class which is basically a "tagging" class from which implementations can
subclass specific exceptions. In general, for standard headers,  exceptions
on illegally formatted headers (see recommendation below) can be delayed or
thrown anytime (there is no request-reply model in SIP so the client UAS
will not be necessarily waiting for a reply although it could retry
endlessly....)



>
>
> An example of this is the AcceptLanguageHeader. The API allows any
> String that is not null or zero-length be used as the language range.
> Obviously an implementation should be stricter on the content of this
> String.

Yes but the parser need not be strict.

>
>
> The reason we are so tolerant of parameter values was that the headers
> were originally defined as classes in the API, and we did not want to
> include a full SIP parser in the API. However, now that interfaces are
> used, an alternative would be to simply extend the definition of an
> illegal argument - for example, the implementation should throw an
> IllegalArgumentException if the language range does not conform to the
> RFC. (Of course testing this would be another matter entirely!)
>
> Another separate issue is - what should an implementation do if receives
>
> something from the network that it cannot parse correctly? Should the
> JAIN SIP application be made aware of it, or should the implementation
> automatically send back the appropriate response to the network? This
> obviously depends on whether the implementation is lazy-parsing or
> parsing the entire message up front, and we have to accomodate both
> scenarios (this is after all one of the main reasons we moved from
> classes to interfaces)

By application I assume you are referring to the event handler portion of
the stack. Heres my take....

I think specific kinds of errors (for example unrecognized  method keywords
that identifies the header type)  should generate prompt notification as a
way of allowing implementers to build  extensions but that does not need to
apply to the entire message. Thus, what I am suggesting is that even a lazy
parser look at the first portion of the header and invoke a handler if the
header is not one of the pre-defined types supported by the parser.

However, this does not need to apply to all headers - for example if the
header is a standard header  (e.g. Contact header) but the rest of the
header is bad. Prompt SIP responses from the server under such
circumstances are not a requirement of the spec (as far as I know).  Hence,
the way I understand it (correct me if I am wrong)  it would be legal for
the server to do a number of things under this scenario including just
ignore the mangled header  (and drop it silently)  so this behavior is
best left unspecified in JAIN-SIP (gives implementations the ability to
choose to do whatever they want).

Thanks.


Regards,

Ranga.

>
>
> Your opinions are of course welcomed.
>
> Regards,
> Chris Harris
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

--
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
Tel: 301 975 3664 Fax: 301 590 0932



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&nbsp;
<br>Chris Harris wrote:
<blockquote TYPE=CITE>All,
<p>Now that the JAIN SIP API is using interfaces instead of classes for</blockquote>

<p><br>Bravo!&nbsp; Is it out for public review as yet?
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<br>messages and headers, the implementations of these interfaces will
have
<br>their own parsing mechanisms. Currently the API set methods throw
<br>IllegalArgumentExceptions if the user attempts to use a null or
<br>zero-length parameter. The question now is should an implementation
<br>throw an IllegalArgument if it doesn't like the value, or should we
<br>create an exception for anything implementation-specific.</blockquote>

<p>I&nbsp;suggest we create an exception class for all parser specific
exceptions and have parsers subclass it to their taste (some may want to
have finer grained exceptions than others). How about a generic SIPParseException
class which is basically a "tagging" class from which implementations can
subclass specific exceptions. In general, for standard headers,&nbsp; exceptions
on illegally formatted headers (see recommendation below)&nbsp;can be delayed
or thrown anytime (there is no request-reply model in SIP&nbsp;so the client
UAS will not be necessarily waiting for a reply although it could retry
endlessly....)
<br>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<p>An example of this is the AcceptLanguageHeader. The API allows any
<br>String that is not null or zero-length be used as the language range.
<br>Obviously an implementation should be stricter on the content of this
<br>String.</blockquote>

<p><br>Yes but the parser need not be strict.
<blockquote TYPE=CITE>&nbsp;
<p>The reason we are so tolerant of parameter values was that the headers
<br>were originally defined as classes in the API, and we did not want
to
<br>include a full SIP parser in the API. However, now that interfaces
are
<br>used, an alternative would be to simply extend the definition of an
<br>illegal argument - for example, the implementation should throw an
<br>IllegalArgumentException if the language range does not conform to
the
<br>RFC. (Of course testing this would be another matter entirely!)
<p>Another separate issue is - what should an implementation do if receives
<p>something from the network that it cannot parse correctly? Should the
<br>JAIN SIP application be made aware of it, or should the implementation
<br>automatically send back the appropriate response to the network? This
<br>obviously depends on whether the implementation is lazy-parsing or
<br>parsing the entire message up front, and we have to accomodate both
<br>scenarios (this is after all one of the main reasons we moved from
<br>classes to interfaces)</blockquote>

<p>By application I&nbsp;assume you are referring to the event handler
portion of the stack. Heres my take....
<p>I&nbsp;think specific kinds of errors (for example unrecognized&nbsp;
method keywords that identifies the header type)&nbsp; should generate
prompt notification as a way of allowing implementers to build&nbsp; extensions
but that does not need to apply to the entire message. Thus, what I am
suggesting is that even a lazy parser look at the first portion of the
header and invoke a handler if the header is not one of the pre-defined
types supported by the parser.
<p>However, this does not need to apply to all headers - for example if
the header is a standard header&nbsp; (e.g. Contact header)&nbsp;but the
rest of the header is bad. Prompt SIP&nbsp;responses from the server under
such circumstances are not a requirement of the spec (as far as I&nbsp;know).&nbsp;
Hence, the way I&nbsp;understand it (correct me if I&nbsp;am wrong)&nbsp;
it would be legal for the server to do a number of things under this scenario
including just ignore the mangled header&nbsp; (and drop it silently)&nbsp;
so this behavior is&nbsp; best left unspecified in JAIN-SIP (gives implementations
the ability to choose to do whatever they want).
<p>Thanks.
<br>&nbsp;
<p>Regards,
<p>Ranga.
<blockquote TYPE=CITE>&nbsp;
<p>Your opinions are of course welcomed.
<p>Regards,
<br>Chris Harris
<p>_______________________________________________
<br>SIP mailing list
<br>SIP@lists.bell-labs.com
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<pre>--&nbsp;
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.&nbsp;
Tel: 301 975 3664 Fax: 301 590 0932</pre>
&nbsp;
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From: Billy Biggs <Billy_Biggs@3com.com>
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Date: Wed, 15 Nov 2000 16:16:19 -0600


  I think Jo was on the right track with his suggestion of adding a
"uri" parameter, but I do not agree that it is worthwhile to add special
processing requirements to support a shaky failover solution.


  I believe that each proxy can decide for itself how much state
information to put in a Record-Route URI, and to provide a URI which
will route back to an appropriate destination for requests in each
direction:

  Method 1, transactions must go through this proxy to get through a
            firewall, no state required:
  --> R-R: <sip:firewall-proxy.com>


  Method 2, state kept with no failover:
  --> R-R: <sip:magic-cookie@proxy.ip.address>


  Method 3, state kept in shared database with friend proxies:
  --> R-R: <sip:magic-cookie@multi-dns-response-domain;maddr=specific-proxy>


  Method 4, need original request URI for internal state:
  --> R-R: <sip:sipproxy.com;orig-uri=sip:foo>


  Method 6, oldschool proxy, doesn't mess up algorithm, provides a
            failover only in one direction:
  --> R-R: <sip:orig-req-uri;maddr=myproxy>


  In each of these cases, the proxy chooses an appropriate URI based on
its needs for state or failure case.


  Jonathan brought up the "fundamental invariant":
 
> That aside, the motivation for doing this oddball combination was to
> try and preserve a "fundamental invariant" of SIP, which was that the
> request URI  should always refer to an entity that the request is
> actually addressed to.  [...]

  Personally, I think the request URI should reflect the next hop in the
Route and not the final destination, otherwise we will get problems with
transparent outbound proxies and clients that don't put the maddr in the
request URI.  Ugh.

  Also remember that in many Record-Route situations, the final
destination is not reachable except through the proxy.

  I don't think it is worthwhile to overcomplicate the protocol just to
provide a _possible_ (and maybe unlikely) failover solution.


  I like the algorithm for clients and proxies to be dead simple:

  1. Proxies prepend an address to the front of the R-R list.
  2. If a client receives a R-R in a request, it copies it to the Route
     directly and then appends the Contact.
  3. If a client receives a R-R in a final response, it reverse copies
     it to the Route and then appends the Contact.
  4. When a client sends a request, it pops the first route and then
     uses it as the Request-URI.

  This algorithm is currently followed by multiple existing
implementations.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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From: "Maddux, Michel" <Michel.Maddux@COMPAQ.com>
To: "'Vladislav Zubarev'" <vladis@vovida.com>,
        Igor Slepchin <ISlepchin@dynamicsoft.com>
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Subject: RE: [SIP] SIP - H.323 - INVITE - No media line.
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Date: Wed, 15 Nov 2000 16:41:03 -0600

RFC 2543 Section B.4 states:
 
<!--StartFragment-->
B.4 Delayed Media Streams

   In some cases, a caller may not know the set of media formats which
   it can support at the time it would like to issue an invitation. This
   is the case when the caller is actually a gateway to another protocol
   which performs media format negotiation after call setup. When this
   occurs, a caller MAY issue an INVITE with a session description that
   contains no media lines. The callee SHOULD interpret this to mean
   that the caller wishes to participate in a multimedia session
   described by the session description, but that the media streams are
   not yet known. The callee SHOULD return a session description
   indicating the streams and media formats it is willing to support,
   however. The caller MAY update the session description either in the
   ACK request or in a re-INVITE at a later time, once the streams are
   known.

The I-D - Singh/Schulzrinne states: "... [snip - to page 14...] The default
session description MUST be either empty or conatin media descriptio (m=)
lines indicating lthe minimal capabilities of any H.323 terminal handled by
the IWF."
 
 
The sip UAC that I'm using for testing either doesn't respond correctly
(e.g. with the appropriate media),
or the I-D is in conflict with the spec on this item, or perhaps I'm sending
an inappropriate value for the "empty"
string in the SDP m= line.  From my reading, I interpret the phrase "empty"
to mean the following line:
 
c=IN IP4 0.0.0.0
m=audio 0 RTP/AVP 0
 
Yet when I send this line, instead of getting back the expected line:
 
c=IN IP4 real.ip.address.here
m=audio 8000 RTP/AVP 8
 
I receive: m=audio 0 RTP/AVP 0
 
A blank media line is either:
 
m=
or no m= line at all.  In both case the UAC issues a rejection.  
 
so I'm still unable to ascertain how to setup the port.  Any comments?
thanks. /m.

-----Original Message-----
From: Vladislav Zubarev [mailto:vladis@vovida.com]
Sent: Tuesday, November 14, 2000 6:25 PM
To: Igor Slepchin
Cc: Maddux, Michel; sip@lists.bell-labs.com; Lawhorn, Tom; Byrne, Sam;
Hurlbert, John
Subject: Re: [SIP] SIP - H.323 - INVITE - No media line.


Again, call scenarios with different issues resolving (like this) can 
be found in the following document: 

http://search.ietf.org/internet-drafts/draft-singh-sip-h323-01.txt
<http://search.ietf.org/internet-drafts/draft-singh-sip-h323-01.txt>  


Thanx. 


Igor Slepchin wrote: 


> 1) When 323 calls, wait until it opens first media channel. 
> Then issue the 
> INVITE, noting only a media line of that media type but with 
> a phony port. 
> When Sip replies (200 OK presumed), complete the 323 media 
> channel being 
> negotiated, giving it the Sip's media port. Then negotiate the reverse 
> channel with 323, saving its port, which is then transmitted 
> to Sip via an 
> ACK with a message body correcting the 323 media port. 

This is illegal in SIP (see section 4.2.1 in 2543bis). You should use 
re-INVITE if you want to modify the session offered in INVITE request. 


Thank you, 
Igor Slepchin 


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-- 

Vladislav Zubarev         mailto:vladis@vovida.com
<mailto:vladis@vovida.com> 

Software Engineer   

Vovida Networks, Inc.     http://www.vovida.com <http://www.vovida.com> 
  


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Just curioius DFA version of the parser is faster than the NFA version
of the parser, which one could be faster? by how much?

"M. Ranganathan" wrote:

> Not yacc but antlr (output language is java). Stay tuned... I will put
> one out in a matter of 2 weeks  (I am documenting it and getting it
> approved for public release).   It will be in th public domain (free).
>
> Ranga.
>
>
> David Frascone wrote:
>
>> Has anyone out there written a sip parser in yacc/bison?  I'm trying
>> to find
>> a free grammar for it if anyone has.
>>
>> _______________________________________________
>> SIP mailing list
>> SIP@lists.bell-labs.com
>> http://lists.bell-labs.com/mailman/listinfo/sip
>
> --
> M.Ranganathan
> NIST Advanced Networking Technologies Group,
> 100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
> Tel: 301 975 3664 Fax: 301 590 0932
>
>

--

Jimmy zhang (jz@ipunity.com)

Software Engineer



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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<body text="#000000" bgcolor="#FFFFFF" link="#0000FF" vlink="#FF0000" alink="#000088">
Just curioius DFA version of the parser is faster than the NFA version
of the parser, which one could be faster? by how much?
<p>"M. Ranganathan" wrote:
<blockquote TYPE=CITE>Not yacc but antlr (output language is java). Stay
tuned... I will put one out in a matter of 2 weeks&nbsp; (I am documenting
it and getting it approved for public release).&nbsp;&nbsp; It will be
in th public domain (free).
<p>Ranga.
<br>&nbsp;
<p>David Frascone wrote:
<blockquote TYPE=CITE>Has anyone out there written a sip parser in yacc/bison?&nbsp;
I'm trying to find
<br>a free grammar for it if anyone has.
<p>_______________________________________________
<br>SIP mailing list
<br>SIP@lists.bell-labs.com
<br><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></blockquote>

<pre>--&nbsp;
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.&nbsp;
Tel: 301 975 3664 Fax: 301 590 0932</pre>
&nbsp;</blockquote>

<pre>--&nbsp;


Jimmy zhang (jz@ipunity.com)

Software Engineer</pre>
&nbsp;
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Ranga,

Thanks for the input! Comments below...

Regards,
Chris


M. Ranganathan wrote:

>  
> Chris Harris wrote:
> 
>> All,
>> 
>> Now that the JAIN SIP API is using interfaces instead of classes for
>> 
> 
> Bravo!  Is it out for public review as yet?
> 
It's not yet. I can email the current version to anyone who is 
interested. The first public draft will be available from Sun's site, 
for the next bake-off

> 
>  
> 
>>  
>> messages and headers, the implementations of these interfaces will have
>> their own parsing mechanisms. Currently the API set methods throw
>> IllegalArgumentExceptions if the user attempts to use a null or
>> zero-length parameter. The question now is should an implementation
>> throw an IllegalArgument if it doesn't like the value, or should we
>> create an exception for anything implementation-specific.
> 
> I suggest we create an exception class for all parser specific 
> exceptions and have parsers subclass it to their taste (some may want 
> to have finer grained exceptions than others). How about a generic 
> SIPParseException class which is basically a "tagging" class from 
> which implementations can subclass specific exceptions. In general, 
> for standard headers,  exceptions on illegally formatted headers (see 
> recommendation below) can be delayed or thrown anytime (there is no 
> request-reply model in SIP so the client UAS will not be necessarily 
> waiting for a reply although it could retry endlessly....)
> 
The implementations can certainly subclass such an exception - but the 
application will only know about the base exception from the API - so I 
think a SipParseException is sufficient. I assume you mean we keep the 
IllegalArgumentException for just the null argument case, and use the 
SipParseException for all other unparsable String arguments (including 
zero-length)?

We need to decide whether this should be a checked or runtime exception.

> 
>  
>  
> 
>>  
>> 
>> An example of this is the AcceptLanguageHeader. The API allows any
>> String that is not null or zero-length be used as the language range.
>> Obviously an implementation should be stricter on the content of this
>> String.
>> 
> 
> Yes but the parser need not be strict.
> 
OK - but chances are it will be stricter than the API :)

>>  
>> 
>> The reason we are so tolerant of parameter values was that the headers
>> were originally defined as classes in the API, and we did not want to
>> include a full SIP parser in the API. However, now that interfaces are
>> used, an alternative would be to simply extend the definition of an
>> illegal argument - for example, the implementation should throw an
>> IllegalArgumentException if the language range does not conform to the
>> RFC. (Of course testing this would be another matter entirely!)
>> 
>> Another separate issue is - what should an implementation do if receives
>> 
>> something from the network that it cannot parse correctly? Should the
>> JAIN SIP application be made aware of it, or should the implementation
>> automatically send back the appropriate response to the network? This
>> obviously depends on whether the implementation is lazy-parsing or
>> parsing the entire message up front, and we have to accomodate both
>> scenarios (this is after all one of the main reasons we moved from
>> classes to interfaces)
>> 
> By application I assume you are referring to the event handler portion 
> of the stack. Heres my take....
> 
> I think specific kinds of errors (for example unrecognized  method 
> keywords that identifies the header type)  should generate prompt 
> notification as a way of allowing implementers to build  extensions 
> but that does not need to apply to the entire message. Thus, what I am 
> suggesting is that even a lazy parser look at the first portion of the 
> header and invoke a handler if the header is not one of the 
> pre-defined types supported by the parser.
> 
Well the implementation should pass an unrecognised header to the JAIN 
SIP application (as a generic Header)- after all the JAIN SIP 
application might be aware of extensions that the underlying 
implementation knows nothing about.

> However, this does not need to apply to all headers - for example if 
> the header is a standard header  (e.g. Contact header) but the rest of 
> the header is bad.
> 
If lazy parsing is being used, this bad part will not be discovered 
until the application attempts to access it - so should the get methods 
throw the JainSipParseException too? (Again this may affect whether the 
exception should be checked or not)

> Prompt SIP responses from the server under such circumstances are not 
> a requirement of the spec (as far as I know).  Hence, the way 
> I understand it (correct me if I am wrong)  it would be legal for the 
> server to do a number of things under this scenario including just 
> ignore the mangled header  (and drop it silently)  so this behavior 
> is  best left unspecified in JAIN-SIP (gives implementations the 
> ability to choose to do whatever they want).
> 
Is dropping the broken header the usual way to handle this case? Do any 
SIP implementations keep the header and leave the header content as a 
string?

> Thanks.
>  
> 
> Regards,
> 
> Ranga.
> 
>>  
>> 
>> Your opinions are of course welcomed.
>> 
>> Regards,
>> Chris Harris
>> 
>> _______________________________________________
>> SIP mailing list
>> SIP@lists.bell-labs.com
>> http://lists.bell-labs.com/mailman/listinfo/sip
>> 
> -- 
> M.Ranganathan
> NIST Advanced Networking Technologies Group,
> 100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899. 
> Tel: 301 975 3664 Fax: 301 590 0932
>   



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"Empty" means SIP message does NOT contain SDP AT ALL, which I guess
means the same as it says in RFC2543 (no media lines).

The problem is the most SUAs do not support this "empty" INVITE right now.

The UA, which supports this feature for sure is CISCO IP PHONE 7960,
with the latest load. There could be some other implementations out there
as well.

Thanx.

"Maddux, Michel" wrote:

> RFC 2543 Section B.4 states:
>
> <!--StartFragment-->
> B.4 Delayed Media Streams
>
>    In some cases, a caller may not know the set of media formats which
>    it can support at the time it would like to issue an invitation. This
>    is the case when the caller is actually a gateway to another protocol
>    which performs media format negotiation after call setup. When this
>    occurs, a caller MAY issue an INVITE with a session description that
>    contains no media lines. The callee SHOULD interpret this to mean
>    that the caller wishes to participate in a multimedia session
>    described by the session description, but that the media streams are
>    not yet known. The callee SHOULD return a session description
>    indicating the streams and media formats it is willing to support,
>    however. The caller MAY update the session description either in the
>    ACK request or in a re-INVITE at a later time, once the streams are
>    known.
>
> The I-D - Singh/Schulzrinne states: "... [snip - to page 14...] The default
> session description MUST be either empty or conatin media descriptio (m=)
> lines indicating lthe minimal capabilities of any H.323 terminal handled by
> the IWF."
>
>
> The sip UAC that I'm using for testing either doesn't respond correctly
> (e.g. with the appropriate media),
> or the I-D is in conflict with the spec on this item, or perhaps I'm sending
> an inappropriate value for the "empty"
> string in the SDP m= line.  From my reading, I interpret the phrase "empty"
> to mean the following line:
>
> c=IN IP4 0.0.0.0
> m=audio 0 RTP/AVP 0
>
> Yet when I send this line, instead of getting back the expected line:
>
> c=IN IP4 real.ip.address.here
> m=audio 8000 RTP/AVP 8
>
> I receive: m=audio 0 RTP/AVP 0
>
> A blank media line is either:
>
> m=
> or no m= line at all.  In both case the UAC issues a rejection.
>
> so I'm still unable to ascertain how to setup the port.  Any comments?
> thanks. /m.
>
> -----Original Message-----
> From: Vladislav Zubarev [mailto:vladis@vovida.com]
> Sent: Tuesday, November 14, 2000 6:25 PM
> To: Igor Slepchin
> Cc: Maddux, Michel; sip@lists.bell-labs.com; Lawhorn, Tom; Byrne, Sam;
> Hurlbert, John
> Subject: Re: [SIP] SIP - H.323 - INVITE - No media line.
>
> Again, call scenarios with different issues resolving (like this) can
> be found in the following document:
>
> http://search.ietf.org/internet-drafts/draft-singh-sip-h323-01.txt
> <http://search.ietf.org/internet-drafts/draft-singh-sip-h323-01.txt>
>
> Thanx.
>
> Igor Slepchin wrote:
>
> > 1) When 323 calls, wait until it opens first media channel.
> > Then issue the
> > INVITE, noting only a media line of that media type but with
> > a phony port.
> > When Sip replies (200 OK presumed), complete the 323 media
> > channel being
> > negotiated, giving it the Sip's media port. Then negotiate the reverse
> > channel with 323, saving its port, which is then transmitted
> > to Sip via an
> > ACK with a message body correcting the 323 media port.
>
> This is illegal in SIP (see section 4.2.1 in 2543bis). You should use
> re-INVITE if you want to modify the session offered in INVITE request.
>
> Thank you,
> Igor Slepchin
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> <http://lists.bell-labs.com/mailman/listinfo/sip>
>
> --
>
> Vladislav Zubarev         mailto:vladis@vovida.com
> <mailto:vladis@vovida.com>
>
> Software Engineer
>
> Vovida Networks, Inc.     http://www.vovida.com <http://www.vovida.com>
>
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

--
Vladislav Zubarev        mailto:vladis@vovida.com
Software Engineer
Vovida Networks, Inc.    http://www.vovida.com



--------------B85EED76782735FE71D84619
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
"Empty"&nbsp;means SIP&nbsp;message does NOT contain SDP&nbsp;AT ALL, which
I&nbsp;guess
<br>means the same as it says in RFC2543 (no media lines).
<p>The problem is the most SUAs do not support this "empty" INVITE right
now.
<p>The UA, which supports this feature for sure is CISCO IP&nbsp;PHONE
7960,
<br>with the latest load. There could be some other implementations out
there
<br>as well.
<p>Thanx.
<p>"Maddux, Michel" wrote:
<blockquote TYPE=CITE>RFC 2543 Section B.4 states:
<p>&lt;!--StartFragment-->
<br>B.4 Delayed Media Streams
<p>&nbsp;&nbsp; In some cases, a caller may not know the set of media formats
which
<br>&nbsp;&nbsp; it can support at the time it would like to issue an invitation.
This
<br>&nbsp;&nbsp; is the case when the caller is actually a gateway to another
protocol
<br>&nbsp;&nbsp; which performs media format negotiation after call setup.
When this
<br>&nbsp;&nbsp; occurs, a caller MAY issue an INVITE with a session description
that
<br>&nbsp;&nbsp; contains no media lines. The callee SHOULD interpret this
to mean
<br>&nbsp;&nbsp; that the caller wishes to participate in a multimedia
session
<br>&nbsp;&nbsp; described by the session description, but that the media
streams are
<br>&nbsp;&nbsp; not yet known. The callee SHOULD return a session description
<br>&nbsp;&nbsp; indicating the streams and media formats it is willing
to support,
<br>&nbsp;&nbsp; however. The caller MAY update the session description
either in the
<br>&nbsp;&nbsp; ACK request or in a re-INVITE at a later time, once the
streams are
<br>&nbsp;&nbsp; known.
<p>The I-D - Singh/Schulzrinne states: "... [snip - to page 14...] The
default
<br>session description MUST be either empty or conatin media descriptio
(m=)
<br>lines indicating lthe minimal capabilities of any H.323 terminal handled
by
<br>the IWF."
<br>&nbsp;
<p>The sip UAC that I'm using for testing either doesn't respond correctly
<br>(e.g. with the appropriate media),
<br>or the I-D is in conflict with the spec on this item, or perhaps I'm
sending
<br>an inappropriate value for the "empty"
<br>string in the SDP m= line.&nbsp; From my reading, I interpret the phrase
"empty"
<br>to mean the following line:
<p>c=IN IP4 0.0.0.0
<br>m=audio 0 RTP/AVP 0
<p>Yet when I send this line, instead of getting back the expected line:
<p>c=IN IP4 real.ip.address.here
<br>m=audio 8000 RTP/AVP 8
<p>I receive: m=audio 0 RTP/AVP 0
<p>A blank media line is either:
<p>m=
<br>or no m= line at all.&nbsp; In both case the UAC issues a rejection.
<p>so I'm still unable to ascertain how to setup the port.&nbsp; Any comments?
<br>thanks. /m.
<p>-----Original Message-----
<br>From: Vladislav Zubarev [<a href="mailto:vladis@vovida.com">mailto:vladis@vovida.com</a>]
<br>Sent: Tuesday, November 14, 2000 6:25 PM
<br>To: Igor Slepchin
<br>Cc: Maddux, Michel; sip@lists.bell-labs.com; Lawhorn, Tom; Byrne, Sam;
<br>Hurlbert, John
<br>Subject: Re: [SIP] SIP - H.323 - INVITE - No media line.
<p>Again, call scenarios with different issues resolving (like this) can
<br>be found in the following document:
<p><a href="http://search.ietf.org/internet-drafts/draft-singh-sip-h323-01.txt">http://search.ietf.org/internet-drafts/draft-singh-sip-h323-01.txt</a>
<br>&lt;<a href="http://search.ietf.org/internet-drafts/draft-singh-sip-h323-01.txt">http://search.ietf.org/internet-drafts/draft-singh-sip-h323-01.txt</a>>
<p>Thanx.
<p>Igor Slepchin wrote:
<p>> 1) When 323 calls, wait until it opens first media channel.
<br>> Then issue the
<br>> INVITE, noting only a media line of that media type but with
<br>> a phony port.
<br>> When Sip replies (200 OK presumed), complete the 323 media
<br>> channel being
<br>> negotiated, giving it the Sip's media port. Then negotiate the reverse
<br>> channel with 323, saving its port, which is then transmitted
<br>> to Sip via an
<br>> ACK with a message body correcting the 323 media port.
<p>This is illegal in SIP (see section 4.2.1 in 2543bis). You should use
<br>re-INVITE if you want to modify the session offered in INVITE request.
<p>Thank you,
<br>Igor Slepchin
<p>_______________________________________________
<br>SIP mailing list
<br>SIP@lists.bell-labs.com
<br><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a>
<br>&lt;<a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a>>
<p>--
<p>Vladislav Zubarev&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a href="mailto:vladis@vovida.com">mailto:vladis@vovida.com</a>
<br>&lt;<a href="mailto:vladis@vovida.com">mailto:vladis@vovida.com</a>>
<p>Software Engineer
<p>Vovida Networks, Inc.&nbsp;&nbsp;&nbsp;&nbsp; <a href="http://www.vovida.com">http://www.vovida.com</a>
&lt;<a href="http://www.vovida.com">http://www.vovida.com</a>>
<br>&nbsp;
<p>_______________________________________________
<br>SIP mailing list
<br>SIP@lists.bell-labs.com
<br><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></blockquote>

<pre>--&nbsp;
Vladislav Zubarev&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <A HREF="mailto:vladis@vovida.com">mailto:vladis@vovida.com</A>
Software Engineer&nbsp;&nbsp;&nbsp;
Vovida Networks, Inc.&nbsp;&nbsp;&nbsp; <A HREF="http://www.vovida.com">http://www.vovida.com</A></pre>
&nbsp;</html>

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From sip-admin@lists.bell-labs.com  Wed Nov 15 20:55:04 2000
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Subject: RE: [SIP] SIP ringback tone
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Date: Wed, 15 Nov 2000 17:53:45 -0800

Right, so if a PSTN gateway knows that the callee is being alerted then it
should send a 180 with (or without) SDP. Otherwise, if the PSTN gateway only
knows that inband call progress tones are being provided by the PSTN network
(but not the callee alerting state) then I agree 183 is more appropriate. If
the network later confirms that the callee is being alerted, the gateway can
always send a subsequent 180 Ringing without SDP (or perhaps with a
duplicate SDP body for robustness).

Cheers,

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 

> -----Original Message-----
> From: Culpepper, Bert [mailto:bert.culpepper@intervoice-brite.com]
> Sent: Thursday, November 16, 2000 2:06 AM
> To: 'sip@lists.bell-labs.com'
> Cc: Alan Johnston; Fairlie-Cuninghame, Robert
> Subject: RE: [SIP] SIP ringback tone
> 
> 
> I think that Alan and Robert are "saying" the same thing.  My 
> understanding is that a UAC, upon receipt of a 180 without 
>  SDP, indicates to the user that the callee is being alerted.  
> It can use whatever method it desires such as a pop-up 
> message, playing an audible ringback tone, etc. to indicate 
> this to the local user.  If the 180 included a SDP, the UAC can 
> present the media stream or not.  However, I believe the UAC 
> should cease its local indication (if any) upon receipt of a 183. 
> The UAS sends a 183 (with SDP) to indicate that it will be 
> sending early media (typically some call progress message).  
> The local ringing indication should cease at the UAC so that 
> the user can "hear" the early media (disposition of the call).
> 
> If the UAS wishes to provide ringback tone it can send a 180 
> with SDP.  A 183 with SDP could be used too but why not use 
> the Ringing response for ringback instead of a Session 
> Progress response for ringback.  As Robert writes, a 180 
> indicates a specific state.  The 180 with local alerting indication 
> at the UAC minimizes network, UAC, and UAS resources 
> during the setup phase and allows the UAC to locally indicate 
> remote alerting however it likes.  If tone characteristics are 
> present in a header of the 180 the UAC can "customize" its 
> local alerting as Adam's Ringback I-D supports.
> 
> Regards,
> Bert
> 
> -----Original Message-----
> From: Alan Johnston [mailto:alan.johnston@wcom.com]
> Sent: Wednesday, November 15, 2000 11:26 AM
> To: Fairlie-Cuninghame, Robert
> Cc: 'sip@lists.bell-labs.com'
> Subject: Re: [SIP] SIP ringback tone
> 
> 
> Robert,
> 
> In actual implementations, often a 183 will actually be sent during
> ringing, due
> to the fact that often a gateway will be unable to determine
> definitively if
> ringing or some other information is being sent in-band.  My concern
> is that
> this reading will cause gateway vendors to assume that ringing is
> taking place,
> send a 180 with SDP.  Then, the User Agent may also assume that
> ringing is
> taking place, not play the early media stream to the user. (The bis
> spec
> currently only says SHOULD.)  If the call then receives treatment, the
> user will
> then have not idea why the call failed.
> 
> I agree that a User Agent should not have to know that it is talking
> to a
> gateway to the PSTN.  What it does need to know is that media
> described in the
> SDP of a 183 response MUST be played to the user, or the final outcome
> of a call
> may not be known. (Since 183 is a new response code, there aren't any
> backwards
> compatibility issues here, are there?)
> 
> Thanks,
> Alan.
> 
> "Fairlie-Cuninghame, Robert" wrote:
> 
> > The way the spec is written at the moment (see Section 7.1 - it
> changed
> > recently), a UAC should be able to accept any 1xx response (other
> than 100)
> > with SDP as indicating early media. eg 183, 182, 181 or 180.
> Obviously they
> > must have different meanings or there is no reason to have the
> different
> > response codes (COMET aside). Both the expired 183 draft
> (explicitly) and
> > the bis-02 draft (inferred) say that 183 indicates call progress
> events
> > _other_ than user alerting (eg "queued message", "please wait,
> verifying" or
> > "enter pin number" voice messages) --- yet the mechanism is the same
> for 180
> > and 183. [I just think it should be made clearer in the 183
> description.]
> >
> > I don't understand your point about requiring support for two
> mechanisms and
> > that it "masks whether I'm going to the PSTN or another SIP endpoint
> > identified by a telephone number". How so? And more importantly, why
> is this
> > necessary ?
> >
> > In fact I would say the opposite is true. A SIP client that is not
> PSTN
> > centric and doesn't bother with early media playback should not have
> to know
> > that 183 is actually ringback on a PSTN SIP client. The client
> should simply
> > know that a received 180 indicates user alerting, 182 is queued and
> 183 is
> > some other call progress event.
> >
> > Regards,
> >
> > Robert.
> >
> > -- My opinions are my own. I tried selling them once but everybody
> >         seems to already have one. --
> > > -----Original Message-----
> > > From: David Shrader [mailto:dshrader@master-consultant.com]
> > > Sent: Monday, November 13, 2000 9:54 PM
> > > To: Fairlie-Cuninghame, Robert
> > > Cc: SIP List
> > > Subject: Re: [SIP] SIP ringback tone
> > >
> > >
> > > Robert,
> > >
> > > I don't agree that 183 with SDP should only be a call
> > > progress message. It
> > > is a perfectly reasonable usage to support ringback. This is
> > > especially
> > > necessary when interworking with the PSTN where ringback is
> delivered
> > > inband. If you implement a system that works with the PSTN, I
> think we
> > > shouldn't require it to support two modes of operation. It
> > > seems reasonable
> > > that the use of 183 masks whether I'm going to the PSTN or another
> SIP
> > > endpoint identified by a telephone number and allows the
> > > person or network
> > > being called to provide whatever inband indication it chooses.
> > >
> > > David
> > >
> > > ----------------------------
> > > David Shrader
> > > Master Consultant, Inc.
> > > dshrader@master-consultant.com
> > > http://www.EyeForTheFuture.com
> > >
> > > > From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
> > > > Date: Mon, 13 Nov 2000 03:33:35 -0800
> > > > To: "'Henry Sinnreich'" <Henry.Sinnreich@wcom.com>
> > > > Cc: "'sip@lists.bell-labs.com'" <sip@lists.bell-labs.com>
> > > > Subject: RE: [SIP] SIP ringback tone
> > > >
> > > > Henry,
> > > >
> > > > Innovative ringback greetings definitely seem like a case
> > > for an early media
> > > > to me.
> > > >
> > > > As an aside, many people still seem to erroroneously think
> > > that 183 Session
> > > > Progress should be used for early-media ring-back. 180
> > > Ringing with SDP
> > > > should be used for ringback purposes, 183 with SDP should
> > > only be used for
> > > > call progress messages (ie, not alerting the user). I still
> > > think this needs
> > > > to be made clearer in the spec!
> > > >
> > > > Cheers,
> > > >
> > > > Robert.
> > > >
> > > > -- My opinions are my own. I tried selling them once but
> everybody
> > > > seems to already have one. --
> > > >
> > > >> -----Original Message-----
> > > >> From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
> > > >> Sent: Monday, November 13, 2000 6:18 AM
> > > >> To: adam.roach@ericsson.com; IETF SIP (E-mail)
> > > >> Subject: [SIP] SIP ringback tone
> > > >>
> > > >>
> > > >> The draft on "Ringback tones in SIP-Based Telephony" shows
> > > >> impressive work
> > > >> on preserving legacy country specific ring back tones. I
> > > >> wonder if we would
> > > >> not be better of by adding a
> > > >>
> > > >> SIP Ringback Tone
> > > >> =================
> > > >>
> > > >> With some optional fields where service providers could
> > > add innovative
> > > >> greetings.
> > > >>
> > > >> Henry
> > > >>
> > > >> Henry Sinnreich
> > > >> WorldCom
> > > >> 400 International Parkway,
> > > >> Richardson, Texas, 75081
> > > >>
> > > >>
> > > >>
> > > >> _______________________________________________
> > > >> SIP mailing list
> > > >> SIP@lists.bell-labs.com
> > > >> http://lists.bell-labs.com/mailman/listinfo/sip
> > > >>
> > > >
> > > > _______________________________________________
> > > > SIP mailing list
> > > > SIP@lists.bell-labs.com
> > > > http://lists.bell-labs.com/mailman/listinfo/sip
> > > >
> > >
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 
> _______________________________________________
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> 

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From sip-admin@lists.bell-labs.com  Wed Nov 15 21:01:04 2000
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Date: Wed, 15 Nov 2000 20:58:21 -0500

Minor comment on draft-byerly-sip-radius-00.txt: the draft seems to deviate
unnecessarily from the challenge and credentials syntax specified in
RFC2617, i.e., it uses semicolon to separate challenge/credentials
parameters instead of reusing comma-separation mandated by HTTP
Basic/Digest. I think consistency with HTTP should be maintained wherever
possible.

Thank you,
Igor Slepchin

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Date: Wed, 15 Nov 2000 20:49:59 -0800 (PST)

Dear All:
  I have one question to discuss, on RFC2543bis02 page
68, which says"The callee address provided in the
request was ambiguous. The response MAY contain a
listing of possible unambiguous addresses in Contact
headers.
   Revealing alternatives can infringe on privacy
concerns of the user or the organization. It MUST be
possible to configure a server to respond with status
404 (Not Found) or to suppress the listing of possible
choices if the request address was ambiguous."
  I can't understand how the UAS or server can judge
this address is ambiguous,if there is no such callee
after comparison, 404 response code can be sent out.
How can the server know under what situation 485 will 
be sent out.
   Could anyone give me some suggestion?
   Thanks in advance!
   Best regards!
                      Yours: Ford

If the server receives a request whose To address or
request-uri is ambiguous then the server may respond
with a 485 response.  The response usually contains a
list of un-ambiguous addresses that are semantically
similar to the ambiguous address.  However this
listing should be suppressed or displayed based on the
request address.  The pattern by which we can find if
an address is ambiguous or not available is not clear
to me.  Based on this method only we can either send a
485 or 404 response.

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From sip-admin@lists.bell-labs.com  Thu Nov 16 08:23:04 2000
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My comments below:

From: "Henry Sinnreich" <Henry.Sinnreich@wcom.com>
To: <AL@AKC.COM>
Cc: "Tom Redman" <tredman@cisco.com>; "Ikhlaq_Sidhu"
<Ikhlaq_Sidhu@3com.com>; "Jay Batson" <jbatson@pingtel.com>; "Jean Joseph
Brierre" <Jean.Joseph.Brierre@wcom.com>; "Ronald Bonica"
<Ronald.P.Bonica@wcom.com>
Sent: Sunday, November 12, 2000 3:40 AM
Subject: Definition of the DNS GL Resource Record


> Costanzo,
>
> You may already be aware of it, your I-D <draft-costanzo-dns-gl-03.txt> on
> Definition of the DNS GL Resource Record is of high interest to the IP
> communications community. Especially, if you have any additional ideas how
> to use this service for 911 calls from IP SIP phones. Suggest to bring it
to
> attention of the
> IETF SIP WG list - sip@lists.bell-labs.com,
> SIP Forum list - SIP Forum Discussion - discussion@sipforum.org
>
> Henry
>
> Henry Sinnreich
> WorldCom
> 400 International Parkway,
> Dept. 1489, 2A313
> Richardson, Texas, 75081
> Cell:   972-897-7723
> Office: 972-729-4271
>

 Hello,

 I am Al Costanzo, I submitted this email to the list but never saw it
posted so here it is again,,,

I wrote ID <draft-costanzo-dns-gl-03.txt>, it was  suggested to me (see
above) to to bring it to the attention of this WG for
 consideration, the draft has currently not moved forward, so there is time
to make modifications to it and
 re-submit it to the IESG.   would appreciate a comment back either way if
the WG is interested in using the draft,
I would be more than  willing to work with the working group to modify it to
allow it to do what you need for 911 calls from IP SIP phones, which is an
ideal usage for the GL RR.

Basically this draft allows physical locations to be stored as a RR in DNS,
the point being that this method of storing the loctions is much better
since everyone already uses DNS.

Please let me know what you think.

 Regards,

 Al Costanzo
>

>



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 I understand a REFER method is used to ask a User Agent to start
sending an INVITE to a party specified in the refer-to header.

What would be the consequence of cancelling the REFER transaction, would
it cancel the INVITE transaction which is in progress ?





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Date: Thu, 16 Nov 2000 06:29:03 -0500

--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: SIP Call Control
	Author(s)	: R. Sparks
	Filename	: draft-ietf-sip-cc-transfer-02.txt
	Pages		: 19
	Date		: 15-Nov-00
	
This document defines a SIP extension within the new Call Control 
Framework to provide Call Transfer capabilities.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-cc-transfer-02.txt

Internet-Drafts are also available by anonymous FTP. Login with the username
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type "cd internet-drafts" and then
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A list of Internet-Drafts directories can be found in
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or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


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Send a message to:
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In the body type:
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NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
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	server="mailserv@ietf.org"

Content-Type: text/plain
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ENCODING mime
FILE /internet-drafts/draft-ietf-sip-cc-transfer-02.txt

--OtherAccess
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	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

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--OtherAccess--

--NextPart--



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Date: Thu, 16 Nov 2000 10:30:47 -0500


--------------A3860FFF96F657C20940CF74
Content-Type: text/plain; charset=iso-8859-1
Content-Transfer-Encoding: base64

 This is  for the designers of SIP .    ( It is NOT a suggestion for
re-deisgn so dont pounce on me for asking this one) :

I am a bit puzzled by the way SIP messages were designed.   If
SIP allows for different kinds of headers and new  extension headers to
be added  (as evidenced by the plethora  of extension draft proposals),
does that subsume the need for message contents (and vice-versa) ?  For
example the sdp message contents can just be a MessageContents header
(albiet a long one).

Thanks

Ranga.

--
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
Tel: 301 975 3664 Fax: 301 590 0932



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<body text="#000000" bgcolor="#FFFFFF" link="#0000FF" vlink="#FF0000" alink="#000088">
&nbsp;This is&nbsp; for the designers of SIP .&nbsp;&nbsp;&nbsp; ( It is
NOT&nbsp;a suggestion for re-deisgn so dont pounce on me for asking this
one) :
<p>I&nbsp;am a bit puzzled by the way SIP&nbsp;messages were designed.&nbsp;&nbsp;
If SIP&nbsp;allows for different kinds of headers and new&nbsp; extension
headers to be added&nbsp; (as evidenced by the plethora&nbsp; of extension
draft proposals),&nbsp; does that subsume the need for message contents
(and vice-versa) ?&nbsp; For example the sdp message contents can just
be a MessageContents header (albiet a long one).
<p>Thanks
<p>Ranga.
<pre>--&nbsp;
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.&nbsp;
Tel: 301 975 3664 Fax: 301 590 0932</pre>
&nbsp;
</body>
</html>

--------------A3860FFF96F657C20940CF74--


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"M. Ranganathan" wrote:
> 
>  This is  for the designers of SIP .    ( It is NOT a suggestion for
> re-deisgn so dont pounce on me for asking this one) :
> 
> I am a bit puzzled by the way SIP messages were designed.   If
> SIP allows for different kinds of headers and new  extension headers
> to be added  (as evidenced by the plethora  of extension draft
> proposals),  does that subsume the need for message contents (and
> vice-versa) ?  For example the sdp message contents can just be a
> MessageContents header (albiet a long one).

The division roughly reflects a division between what describes sessions
as a whole (headers) and what describes sessions in their data stream
details (body). This makes it easier, for example, to upgrade to a new
session description format, without changing the software in proxies. In
practice, the idea is that proxies and redirect servers should keep
their hands and noses out of the message body, while being free to
inspect and, with restrictions, modify the headers. NATs, as usual,
break this model, but then NATs are warts on the Internet architecture,
so this is no surprise.

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Hello,
My colleagues, Arnoud van Wijk and Henry Sinnreich, and I have written the following draft: "A Proposed Set of Requirements for SIP Support for Deaf or Speech Impaired Customers".  The URL for this draft is:

http://search.ietf.org/internet-drafts/draft-gearhart-sip-deaf-req-00.txt

and would like to solicit comments from the SIP WG mailing list.  The abstract of our draft is as follows:

	SIP is attracting more and more attention as a valuable tool to
          enable and support voice and multimedia communications over the
          Internet.  However, a significant number of potential customers
          for SIP-based services are for various reasons unable to, or
          choose not to, use voice communication technologies. This document
          offers a proposed set of requirements and identifies some of the
          key issues that need resolution to allow full accessibility for
          this group to SIP-based Internet communications. This document is
          intended to continue the discussion begun in a previous draft (see
          Acknowledgements) and to lay a foundation for design and
          implementation discussions in future drafts.

Comments from deaf, hard-of-hearing, and/or speech-impaired individuals who have personal experience with frustrating communications are particularly welcome.  My colleague and co-author Arnoud van Wijk is deaf and I am a certified sign language interpreter, so between us we have attempted to represent both deaf and interpreter perspectives in our document.  However, we are aware that circumstances vary around the world for deaf, hard-of-hearing, and speech-impaired customers, so we are very interested in feedback from individuals who feel we may have missed something or need to clarify something.

Arnoud will make a brief presentation on this draft at the SIP WG meeting at IETF '49. 
Thank you!
____________________
Cathy Gearhart
cathy.gearhart@ericsson.com
(972) 583-5268


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<BODY>
<BR>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Gill Sans MT">Hello,</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Gill Sans MT">My =
colleagues, Arnoud van Wijk and Henry Sinnreich, and I have written the =
following draft: &quot;A Proposed Set of Requirements for SIP Support =
for Deaf or Speech Impaired Customers&quot;.&nbsp; The URL for this =
draft is:</FONT></P>

<P><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Gill Sans MT"><A =
HREF=3D"http://search.ietf.org/internet-drafts/draft-gearhart-sip-deaf-r=
eq-00.txt" =
TARGET=3D"_blank">http://search.ietf.org/internet-drafts/draft-gearhart-=
sip-deaf-req-00.txt</A></FONT></U>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Gill Sans MT">and would =
like to solicit comments from the SIP WG mailing list.&nbsp; The =
abstract of our draft is as follows:</FONT>
</P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Gill Sans MT">SIP is attracting more and more =
attention as a valuable tool to</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Gill Sans =
MT">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; enable and =
support voice and multimedia communications over the</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Gill Sans =
MT">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Internet.&nbsp; However, a significant number of potential =
customers</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Gill Sans =
MT">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; for =
SIP-based services are for various reasons unable to, or</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Gill Sans =
MT">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; choose not =
to, use voice communication technologies. This document</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Gill Sans =
MT">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; offers a =
proposed set of requirements and identifies some of the</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Gill Sans =
MT">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; key issues =
that need resolution to allow full accessibility for</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Gill Sans =
MT">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; this group =
to SIP-based Internet communications. This document is</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Gill Sans =
MT">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; intended to =
continue the discussion begun in a previous draft (see</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Gill Sans =
MT">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Acknowledgements) and to lay a foundation for design and</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Gill Sans =
MT">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
implementation discussions in future drafts.</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Gill Sans MT">Comments from =
deaf, hard-of-hearing, and/or speech-impaired individuals who have =
personal experience with frustrating communications are particularly =
welcome.&nbsp; My colleague and co-author Arnoud van Wijk is deaf and I =
am a certified sign language interpreter, so between us we have =
attempted to represent both deaf and interpreter perspectives in our =
document.&nbsp; However, we are aware that circumstances vary around =
the world for deaf, hard-of-hearing, and speech-impaired customers, so =
we are very interested in feedback from individuals who feel we may =
have missed something or need to clarify something.</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Gill Sans MT">Arnoud will =
make a brief presentation on this draft at the SIP WG meeting at IETF =
'49. </FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Gill Sans MT">Thank =
you!</FONT>
<BR><I><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Comic Sans =
MS">____________________</FONT></I>
<BR><I><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Comic Sans MS">Cathy =
Gearhart</FONT></I>
<BR><I><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Comic Sans =
MS">cathy.gearhart@ericsson.com</FONT></I>
<BR><I><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Comic Sans MS">(972) =
583-5268</FONT></I>
</P>

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Date: Thu, 16 Nov 2000 15:21:48 -0800

--=====================_79125457==_.ALT
Content-Type: text/plain; charset="us-ascii"

Hi,

I just submitted an ID for an extension to the REFER method that I discussed
with Robert Sparks at Pittsburgh.  Until it appears in the archive it is
available at:

        ftp://ftpeng.cisco.com/rmahy/draft-mahy-sip-189-00.txt


        A SIP Extension: Informational Responses to the REFER method

                Abstract  
This document defines a SIP extension to the REFER method of the SIP Call
Control Framework.  This extension provides for the ability to convey
information about the progress of the Referred request when that request is in
a provisional state for a long period of time (many seconds or minutes).

Comments welcome.

thanks,
-rohan



--=====================_79125457==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
Hi,<br>
<br>
I just submitted an ID for an extension to the REFER method that I
discussed with Robert Sparks at Pittsburgh.&nbsp; Until it appears in the
archive it is available at:<br>
<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><a href="ftp://ftpeng.cisco.com/rmahy/draft-mahy-sip-189-00.txt" eudora="autourl">ftp://ftpeng.cisco.com/rmahy/draft-mahy-sip-189-00.txt</a><br>
<br>
<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>A SIP
Extension: Informational Responses to the REFER method<br>
<br>
<x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab><x-tab>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</x-tab>Abstract
<dl>
<dd>This document defines a SIP extension to the REFER method of the SIP
Call Control Framework.&nbsp; This extension provides for the ability to
convey information about the progress of the Referred request when that
request is in a provisional state for a long period of time (many seconds
or minutes).<br>
<br>

</dl>Comments welcome.<br>
<br>
thanks,<br>
-rohan<br>
<br>
<br>
</html>

--=====================_79125457==_.ALT--


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Subject: [SIP] New ID for peer-to-peer 3pcc
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Date: Thu, 16 Nov 2000 16:46:58 -0800

Hi,

I just submitted an ID on a usage of REFER and presence for 3pcc which
doesn't require you to keep call state or stay in the signalling path.
Until it appears in the archive it can be found here:

	ftp://ftpeng.cisco.com/rmahy/draft-mahy-sip-peer-3pcc-00.txt

Using SIP for Peer-to-Peer Third-Party Call Control

Abstract

The 3rd party call control draft demonstrates a usage of SIP with some of
the enhancements of RFC2543bis.  This usage requires that a 3pcc controller
remain in the signaling path and maintain state for the duration of the
call.  While this is necessary in certain situations (ex: protocol
translation: H.323 to SIP, SIP to RTSP, HTTP to SIP), it is sub-optimal in
pure SIP environments.  This draft demonstrates a usage of the REFER method
and SIP for presence  to allow authorized peers to participate in call control.

Note that much of the functionality described in PHONECTL can be duplicated
using the proposed usage.

Also note that this is not an extension of SIP, merely a usage of SIP and
existing extensions.

Comments are welcome.
thanks,
-rohan

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Date: Fri, 17 Nov 2000 10:37:33 +0300

This is a multi-part message in MIME format.

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Hi, All !
I didn't find answer to this question in RFC.
How do proxy server search called UA? What are "routing tables" used? =
What algorithm?=20
What are expected in the future?

Thanks! Samorezov Vladimir!

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<DIV><FONT face=3D"Comic Sans MS" size=3D2>Hi, All !</FONT></DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2>I didn't find answer to this =
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<DIV><FONT face=3D"Comic Sans MS" size=3D2>How do proxy server search =
called UA?=20
What are "routing tables" used? What algorithm? </FONT></DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2>What&nbsp;are expected in the =

future?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2>Thanks! Samorezov=20
Vladimir!</FONT></DIV></BODY></HTML>

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Subject: Re: [SIP] ID extension of REFER
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Date: Fri, 17 Nov 2000 11:02:34 +0100
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Hello,

I am a little confused about the purpose of this I-D. The first question
that springs to mind is
why you don't use "183" for this purpose? The second question is whether
you
intend the 189 to lead to different timing than any other provisional
response?
(In other words, if the 189 does NOT cause different timing, than the
issue of a long
  period (> 37 seconds) between the INVITE and 200 OK is not really
solved. I realize
  this should be a very rare case.)

/sean

Rohan Mahy wrote:

>  Hi,
>
> I just submitted an ID for an extension to the REFER method that I
> discussed with Robert Sparks at Pittsburgh.  Until it appears in the
> archive it is available at:
>
> ftp://ftpeng.cisco.com/rmahy/draft-mahy-sip-189-00.txt
>
>
> A SIP Extension: Informational Responses to the REFER method
>
> Abstract
>
>      This document defines a SIP extension to the REFER method of the
>      SIP Call Control Framework.  This extension provides for the
>      ability to convey information about the progress of the Referred
>      request when that request is in a provisional state for a long
>      period of time (many seconds or minutes).
>
>
> Comments welcome.
>
> thanks,
> -rohan
>
>


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From sip-admin@lists.bell-labs.com  Fri Nov 17 07:08:07 2000
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        Sean Olson <sean.olson@uab.ericsson.se>,
        Jonathan Rosenberg <jdrosen@uab.ericsson.se>
Subject: Re: Record-Route (was Re: [SIP] Identification problem)
References: <3A07C09B.DAC8B5E5@ericsson.com> <001b01c04976$80fa26a0$4e34c3c1@ubiquity.co.uk> <8uvdf9$9f$1@lux2.datacom-lab.uab.ericsson.se>
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Date: Fri, 17 Nov 2000 13:07:38 +0100
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It's seems like we agree on removing the text from the
specification about overwriting the ROUTE-URI with the
URI from CONTACT/FROM. If this is done my problem to
identify the call state machine is solved. 

The other proposals you have concerns me a little bit.
Should we really use an URI for this kind of data. It
will probably work fine but it's to much of a hack in
my eyes. How about other types of URIs in ROUTE ?

As Jonathan pointed out some weeks ago there is no
maddr parameter in a TEL URL. So something needs to
be done to ROUTE to handle these kind of URLs. So
some parameter needs to be added to RECORD-ROUTE and
ROUTE (e.g. named 'raddr' for route address). 

Since this does not exists today it means that there
are no existing SIP implementations that can route on
a URI other than a SIP URL ? If this is true, how
backwards compatible do we need to be to fix this ?

So before a solution is decided this time may I suggest 
that it covers all known cases so we don't need to change
the RECORD-ROUTE/ROUTE behaviour again in the future.

/Bertil

Billy Biggs wrote:
> 
>   I think Jo was on the right track with his suggestion of adding a
> "uri" parameter, but I do not agree that it is worthwhile to add special
> processing requirements to support a shaky failover solution.
> 
>   I believe that each proxy can decide for itself how much state
> information to put in a Record-Route URI, and to provide a URI which
> will route back to an appropriate destination for requests in each
> direction:
> 
>   Method 1, transactions must go through this proxy to get through a
>             firewall, no state required:
>   --> R-R: <sip:firewall-proxy.com>
> 
>   Method 2, state kept with no failover:
>   --> R-R: <sip:magic-cookie@proxy.ip.address>
> 
>   Method 3, state kept in shared database with friend proxies:
>   --> R-R: <sip:magic-cookie@multi-dns-response-domain;maddr=specific-proxy>
> 
>   Method 4, need original request URI for internal state:
>   --> R-R: <sip:sipproxy.com;orig-uri=sip:foo>
> 
>   Method 6, oldschool proxy, doesn't mess up algorithm, provides a
>             failover only in one direction:
>   --> R-R: <sip:orig-req-uri;maddr=myproxy>
> 
>   In each of these cases, the proxy chooses an appropriate URI based on
> its needs for state or failure case.
> 
>   Jonathan brought up the "fundamental invariant":
> 
> > That aside, the motivation for doing this oddball combination was to
> > try and preserve a "fundamental invariant" of SIP, which was that the
> > request URI  should always refer to an entity that the request is
> > actually addressed to.  [...]
> 
>   Personally, I think the request URI should reflect the next hop in the
> Route and not the final destination, otherwise we will get problems with
> transparent outbound proxies and clients that don't put the maddr in the
> request URI.  Ugh.
> 
>   Also remember that in many Record-Route situations, the final
> destination is not reachable except through the proxy.
> 
>   I don't think it is worthwhile to overcomplicate the protocol just to
> provide a _possible_ (and maybe unlikely) failover solution.
> 
>   I like the algorithm for clients and proxies to be dead simple:
> 
>   1. Proxies prepend an address to the front of the R-R list.
>   2. If a client receives a R-R in a request, it copies it to the Route
>      directly and then appends the Contact.
>   3. If a client receives a R-R in a final response, it reverse copies
>      it to the Route and then appends the Contact.
>   4. When a client sends a request, it pops the first route and then
>      uses it as the Request-URI.
> 
>   This algorithm is currently followed by multiple existing
> implementations.
> 
> --
> Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Bertil Engelholm
AXE Research and Development        voice : +46 8 727 3499
SIP Security                        Fax   : +46 8 647 8276
S-126 25 Stockholm Sweden           E-mail:
Bertil.Engelholm@uab.ericsson.se

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Date: Fri, 17 Nov 2000 13:08:28 +0100
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Just one brief comment.
/sean

Chris Harris wrote:

> Ranga,
> Thanks for the input! Comments below...
> Regards,
> Chris
>
> M. Ranganathan wrote:
> >
> > Chris Harris wrote:
> >

<snip>

>
> > Prompt SIP responses from the server under such circumstances are not
> > a requirement of the spec (as far as I know).  Hence, the way
> > I understand it (correct me if I am wrong)  it would be legal for the
> > server to do a number of things under this scenario including just
> > ignore the mangled header  (and drop it silently)  so this behavior
> > is  best left unspecified in JAIN-SIP (gives implementations the
> > ability to choose to do whatever they want).
> >
> Is dropping the broken header the usual way to handle this case? Do any
> SIP implementations keep the header and leave the header content as a
> string?

To be really "future proof", you should return the header content as is,
even if it doesn't parse correctly (according to the parser's notion of
correctness). Dropping it seems very wrong.

> > Thanks.
> > Regards,
> > Ranga.


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Date: Fri, 17 Nov 2000 12:20:40 +0000
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Sean,

If dropping the malformed header is considered wrong, then one solution in
the JAIN context would be to use the generic Header class instead of the
particular subclass, say FromHeader. But this assumes that the header is
fully parsed before the object is created - if lazy-parsing is used the
FromHeader will be created based on the header name - and the fact that the
header content is malformed will not be discovered until the application
attempts to access the various parts of the header.

As an alternative, a JainSipParseException could be thrown by the get
methods, and if one is thrown, the application must then call the getValue()
method to get the string version of the malformed content?

Chris


Sean Olson wrote:

> Just one brief comment.
> /sean
>
> Chris Harris wrote:
>
> > Ranga,
> > Thanks for the input! Comments below...
> > Regards,
> > Chris
> >
> > M. Ranganathan wrote:
> > >
> > > Chris Harris wrote:
> > >
>
> <snip>
>
> >
> > > Prompt SIP responses from the server under such circumstances are not
> > > a requirement of the spec (as far as I know).  Hence, the way
> > > I understand it (correct me if I am wrong)  it would be legal for the
> > > server to do a number of things under this scenario including just
> > > ignore the mangled header  (and drop it silently)  so this behavior
> > > is  best left unspecified in JAIN-SIP (gives implementations the
> > > ability to choose to do whatever they want).
> > >
> > Is dropping the broken header the usual way to handle this case? Do any
> > SIP implementations keep the header and leave the header content as a
> > string?
>
> To be really "future proof", you should return the header content as is,
> even if it doesn't parse correctly (according to the parser's notion of
> correctness). Dropping it seems very wrong.
>
> > > Thanks.
> > > Regards,
> > > Ranga.
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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Date: Fri, 17 Nov 2000 17:55:11 +0530



Hi,
Is anyone using a SIP-loadgenerator, to test sipserver applications.Can
someone provide me info on this.
Regards,
N.Sudha



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From sip-admin@lists.bell-labs.com  Fri Nov 17 07:29:21 2000
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Subject: [SIP] JAIN SIP - some other issues
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Date: Fri, 17 Nov 2000 12:21:50 +0000
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A couple of other issues which need to be resolved are as follows:

The equals() methods are currently over-ridden e.g. AcceptHeader has a
method public boolean equals(AcceptHeader acceptHeader). Is this
over-riding desirable, or would the original signature public boolean
equals(object obj) be preferable?

The JainSipFactory has replaced the JainIPFactory due to the number of
SIP-specific create methods needed with interfaces. The JainSipFactory
defines a generic createJainSipObject(...) method that takes a String
(which is appended to the path name to give the fully qualified class
name of the object to be created). This method throws a
JainSipPeerUnavailableException if the class specified is not found.
This exception extends JainSipException, and is therefore a checked
exception.

The question is whether the other create methods in the factory e.g.
createAcceptHeader(...) should catch this exception (assume the JAIN SIP
implementation will contain an AcceptHeader implementation), or throw
it. I think it is reasonable to assume that an implementation will
provide all the classes needed (otherwise they will fail the TCK), but
the user may set the path name of the factory to a non-existent
implementation, say com.nosuchcompany, or the implementation they wish
to use may not be in the classpath. This would suggest that all create
methods in the factory should throw the JainSipPeerUnavailableException.

However, this will place a large burden on the application, since it
will have to catch this exception every time it tries to create a header
or message. My suggestion is to create a JainSipRuntimeException, and
have JainSipPeerUnavailableException extend it. Would this be considered
a reasonable solution?

Regards,
Chris Harris


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From sip-admin@lists.bell-labs.com  Fri Nov 17 08:22:16 2000
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From: "Keith Robinson" <Keith.Robinson@marconi.com>
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Date: Fri, 17 Nov 2000 12:41:10 +0000



I have a problem regarding the parsing of the contact line in 12C :

Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone

The accompanying text says that the 'user' parameter should be interpreted as
being a contact-param and not a url-parameter; I believe the opposite is true if
the BNF followed; here is why:

The first line of the contact rule is:

Contact = ("Contact" | "m") ":"
                    ("*"|(1#((name-addr|addr-spec)*((";"contact-params))))

having decoded the keyword Contact, the rule is now looking for name-addr or
addr-spec; as there is no display name or angled brackets then addr-spec is the
next rule to decode.

The rule for addr-spec is

addr-spec = SIP-URL|URI

which is matched perfectly by

sip:+1-972-555-222@gw1.wcom.com;user=phone

I do not see any reason why the decoding of the SIP-URL rule should stop before
the parameters if they satisfy the url-parameter spec  so therefore user=phone
would belong to url-parameters and not contact-parameters which is probably
intended.

Now to the particular problem I see. Consider the following line

Contact: sip:+1-972-555-2222@gw1.com.wcom;action=proxy

Using the above logic , paramter action=proxy would satisfy the other-param spec
in SIP-URL which is probably not what is intended.

I think the safest approach is to use the name-addr format in all cases and then
parameters which belong to url-parameters are placed inside the angled brackets
and parameters which belong to contact-params are placed otside the brackets,
thus

Contact: <sip:+1-972-555-2222@gw1.com.wcom;user=phone>;action=proxy

will always give desired behaviour.

The above also applies to other headers such as To and From.

Regards,
K. Robinson.




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From sip-admin@lists.bell-labs.com  Fri Nov 17 08:40:04 2000
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From: "Hisham Khartabil" <hisham.khartabil@hotsip.com>
To: "Keith Robinson" <Keith.Robinson@marconi.com>,
        "Anders Kristensen" <akristensen@dynamicsoft.com>
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Subject: RE: [SIP] new torture tests
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Date: Fri, 17 Nov 2000 15:39:58 +0200
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Section 6.5 in the Bis:

"
   The Contact, From and To header fields contain a URL. If the URL
   contains a comma, question mark or semicolon, the URL MUST be
   enclosed in angle brackets (< and >). Any URL parameters are
   contained within these brackets. If the URL is not enclosed in angle
   brackets, any semicolon-delimited parameters are header-parameters,
   not URL parameters.

"

I think this should answer your query.

Regards,
Hisham Khartabil
Hotsip Finland
www.hotsip.com


-----Original Message-----
From: sip-admin@lists.bell-labs.com [mailto:sip-admin@lists.bell-labs.com]On
Behalf Of Keith Robinson
Sent: Friday, 17 November 2000 2:41 PM
To: Anders Kristensen
Cc: Neil Deason; Sip List
Subject: Re: [SIP] new torture tests


I have a problem regarding the parsing of the contact line in 12C :

Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone

The accompanying text says that the 'user' parameter should be interpreted
as
being a contact-param and not a url-parameter; I believe the opposite is
true if
the BNF followed; here is why:

The first line of the contact rule is:

Contact = ("Contact" | "m") ":"
                    ("*"|(1#((name-addr|addr-spec)*((";"contact-params))))

having decoded the keyword Contact, the rule is now looking for name-addr or
addr-spec; as there is no display name or angled brackets then addr-spec is
the
next rule to decode.

The rule for addr-spec is

addr-spec = SIP-URL|URI

which is matched perfectly by

sip:+1-972-555-222@gw1.wcom.com;user=phone

I do not see any reason why the decoding of the SIP-URL rule should stop
before
the parameters if they satisfy the url-parameter spec  so therefore
user=phone
would belong to url-parameters and not contact-parameters which is probably
intended.

Now to the particular problem I see. Consider the following line

Contact: sip:+1-972-555-2222@gw1.com.wcom;action=proxy

Using the above logic , paramter action=proxy would satisfy the other-param
spec
in SIP-URL which is probably not what is intended.

I think the safest approach is to use the name-addr format in all cases and
then
parameters which belong to url-parameters are placed inside the angled
brackets
and parameters which belong to contact-params are placed otside the
brackets,
thus

Contact: <sip:+1-972-555-2222@gw1.com.wcom;user=phone>;action=proxy

will always give desired behaviour.

The above also applies to other headers such as To and From.

Regards,
K. Robinson.




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From sip-admin@lists.bell-labs.com  Fri Nov 17 08:42:04 2000
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Keith Robinson wrote:
> 
> I have a problem regarding the parsing of the contact line in 12C :
> 
> Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone
> 
> The accompanying text says that the 'user' parameter should be interpreted as
> being a contact-param and not a url-parameter; I believe the opposite is true if
> the BNF followed; here is why:

This goes back to that paragraph I was quoting... (bis-02 section 6.14):

   Even if the "display-name" is empty, the "name-addr" form
   MUST be used if the "addr-spec" contains a comma, semicolon,
   or question mark.

Since the name-addr form is NOT used it follows that the addr-spec part
does NOT contain a comma, semicolon, or question mark. Hence the
parameter cannot belong to the SIP URL and so must belong to the
Contact.  The correct parsing is not captured in the grammar which is
what makes this case tricky.

> 
> The first line of the contact rule is:
> 
> Contact = ("Contact" | "m") ":"
>                     ("*"|(1#((name-addr|addr-spec)*((";"contact-params))))
> 
> having decoded the keyword Contact, the rule is now looking for name-addr or
> addr-spec; as there is no display name or angled brackets then addr-spec is the
> next rule to decode.
> 
> The rule for addr-spec is
> 
> addr-spec = SIP-URL|URI
> 
> which is matched perfectly by
> 
> sip:+1-972-555-222@gw1.wcom.com;user=phone
> 
> I do not see any reason why the decoding of the SIP-URL rule should stop before
> the parameters if they satisfy the url-parameter spec  so therefore user=phone
> would belong to url-parameters and not contact-parameters which is probably
> intended.
> 
> Now to the particular problem I see. Consider the following line
> 
> Contact: sip:+1-972-555-2222@gw1.com.wcom;action=proxy
> 
> Using the above logic , paramter action=proxy would satisfy the other-param spec
> in SIP-URL which is probably not what is intended.

The above logic is wrong. This example is OK.

> 
> I think the safest approach is to use the name-addr format in all cases and then
> parameters which belong to url-parameters are placed inside the angled brackets
> and parameters which belong to contact-params are placed otside the brackets,
> thus

Absolutely. No discussion at all. The point of the torture tests,
though, is to test those rarely used constructs and try to stress
implementations. It's not a BCP. It would probably have been better if
the grammer hadn't allowed the addr-spec form to begin with.

--
Anders Kristensen

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From sip-admin@lists.bell-labs.com  Fri Nov 17 09:41:08 2000
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Reply-To: "Venkatesh Venkataramanan" <venkatesh.venkataramanan@wipro.com>
From: "Venkatesh Venkataramanan" <venkatesh.venkataramanan@wipro.com>
To: <sip@lists.bell-labs.com>
Subject:  [SIP] ID extension of REFER
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> Sorry I had more to add !! using 1xx class of responses gives more
> information to the transferor as to the state of the TT ... infact a lot
of
> applications (typically voice mail systems) usually support a feature
called
> "transfer to ringing", wherein, all they want to know is if the transfer
> target is Idle and would want to complete the transfer if the TT is idle.
If
> TT is busy, they end up abandoning transfers and putting the user back
into
> voice mail. Using the 1xx responses allows us to continue supporting such
> applications as well.
>
> Venkatesh
>
> ----- Original Message -----
> From: "Venkatesh Venkataramanan" <venkatesh.venkataramanan@wipro.com>
> To: "Sean Olson" <sean.olson@ericsson.com>
> Cc: <rohan@cisco.com>; "Robert Sparks" <rsparks@dynamicsoft.com>
> Sent: Thursday, November 16, 2000 4:02 PM
> Subject: Re: [SIP] ID extension of REFER
>
>
> > I agree. If REFER can be modified to accept any 1xx responses (the draft
> > currently talks about a 100 response/200/400-600 class responses), this
> new
> > ID probably is not required. One could reuse the 1xx progress
information
> > for REFER. The biggest advantage I see with this draft is the is the
> ability
> > of the transferor to get out of the transfer session at any time
(instead
> of
> > waiting for a final response). Model fits the traditional PSTN blind
> > transfers better.
> >
> > Venkatesh
> > ----- Original Message -----
> > From: "Sean Olson" <sean.olson@ericsson.com>
> > To: "Rohan Mahy" <rohan@cisco.com>
> > Cc: <sip@lists.bell-labs.com>
> > Sent: Friday, November 17, 2000 3:32 PM
> > Subject: Re: [SIP] ID extension of REFER
> >
> >
> > > Hello,
> > >
> > > I am a little confused about the purpose of this I-D. The first
question
> > > that springs to mind is
> > > why you don't use "183" for this purpose? The second question is
whether
> > > you
> > > intend the 189 to lead to different timing than any other provisional
> > > response?
> > > (In other words, if the 189 does NOT cause different timing, than the
> > > issue of a long
> > >   period (> 37 seconds) between the INVITE and 200 OK is not really
> > > solved. I realize
> > >   this should be a very rare case.)
> > >
> > > /sean
> > >
> > > Rohan Mahy wrote:
> > >
> > > >  Hi,
> > > >
> > > > I just submitted an ID for an extension to the REFER method that I
> > > > discussed with Robert Sparks at Pittsburgh.  Until it appears in the
> > > > archive it is available at:
> > > >
> > > > ftp://ftpeng.cisco.com/rmahy/draft-mahy-sip-189-00.txt
> > > >
> > > >
> > > > A SIP Extension: Informational Responses to the REFER method
> > > >
> > > > Abstract
> > > >
> > > >      This document defines a SIP extension to the REFER method of
the
> > > >      SIP Call Control Framework.  This extension provides for the
> > > >      ability to convey information about the progress of the
Referred
> > > >      request when that request is in a provisional state for a long
> > > >      period of time (many seconds or minutes).
> > > >
> > > >
> > > > Comments welcome.
> > > >
> > > > thanks,
> > > > -rohan
> > > >
> > > >
> > >
> > >
> > > _______________________________________________
> > > SIP mailing list
> > > SIP@lists.bell-labs.com
> > > http://lists.bell-labs.com/mailman/listinfo/sip
> > >
> > >
> >
>
>


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Date: Fri, 17 Nov 2000 07:59:19 -0500

In rfc 2543 bis-02, section 16.3, there is a example: 
Bell sends INVITE, sdp=coders 0, 3, 4, 5. 
Watson sends Trying..... 
Watson sends OK, sdp=coders 0 3. 
Bell sends ACK, no sdp. 

How do Bell and Watson determine what coder to use (0 or 3 are common - but
which one) since there is no coder negotiation in RTP.  

Thanks,
Duke


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From sip-admin@lists.bell-labs.com  Fri Nov 17 10:49:10 2000
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Date: Fri, 17 Nov 2000 09:47:42 -0600

Snyder, Duke (Duke.Snyder@inin.com):

> Bell sends INVITE, sdp=coders 0, 3, 4, 5. 
> Watson sends OK, sdp=coders 0 3. 
> 
> How do Bell and Watson determine what coder to use (0 or 3 are common
> - but which one) since there is no coder negotiation in RTP.  

  They each choose whichever they prefer.  Bell should be prepared to
receive any of 0, 3, 4 or 5 and Watson should be prepared to receive
either of 0 or 3.

  As well, they should note that the incoming codec can change at any
point in time, and that the incoming RTP can come from any IP address,
not just the one you are sending to.

  The media flow is not a two-way connection, it is two one-way
connections.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Fri, 17 Nov 2000 09:59:38 -0600

Bertil Engelholm (Bertil.Engelholm@uab.ericsson.se):

> The other proposals you have concerns me a little bit.  Should we
> really use an URI for this kind of data. It will probably work fine
> but it's to much of a hack in my eyes. How about other types of URIs
> in ROUTE ?

  It is not a hack:  You are sending the request to some state object on
a proxy so that it can update its state and then pass on the message.

> As Jonathan pointed out some weeks ago there is no maddr parameter in
> a TEL URL. So something needs to be done to ROUTE to handle these kind
> of URLs. So some parameter needs to be added to RECORD-ROUTE and ROUTE
> (e.g. named 'raddr' for route address). 

  You're suggesting that a proxy can put as its Record-Route:
  --> R-R: <http://my.proxy.com/rec-route-state>;raddr=proxy:5060

  Future requests are now sent like this:
  --> MYMETHOD http://my.proxy.com/rec-route-state SIP/2.0

  But IPly they are sent to the IP address proxy:5060

  This is dangerous because:

  1. The Request-URI is not routable by a transparent proxy in the
     middle.  If some firewall proxy snarfs the message, unless it
     records the real IP destination and port, it won't be able to route
     it to the appropriate destination.  This breaks the SIP model.

  2. This gets confusing, since in this case it is an http URI, but
     you're not sending it over http.

  I think that supporting this gives a small benefit at the cost of more
difficult implementation and non-compatibility with existing software.

  I don't believe it is worth it.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Subject: Re: Record-Route (was Re: [SIP] Identification problem)
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Billy Biggs wrote:

> Bertil Engelholm (Bertil.Engelholm@uab.ericsson.se):
>

<snip>

>
> > As Jonathan pointed out some weeks ago there is no maddr parameter in
> > a TEL URL. So something needs to be done to ROUTE to handle these kind
> > of URLs. So some parameter needs to be added to RECORD-ROUTE and ROUTE
> > (e.g. named 'raddr' for route address).
>
>   You're suggesting that a proxy can put as its Record-Route:
>   --> R-R: <http://my.proxy.com/rec-route-state>;raddr=proxy:5060
>
>   Future requests are now sent like this:
>   --> MYMETHOD http://my.proxy.com/rec-route-state SIP/2.0
>
>   But IPly they are sent to the IP address proxy:5060
>
>   This is dangerous because:
>
>   1. The Request-URI is not routable by a transparent proxy in the
>      middle.  If some firewall proxy snarfs the message, unless it
>      records the real IP destination and port, it won't be able to route
>      it to the appropriate destination.  This breaks the SIP model.
>
>   2. This gets confusing, since in this case it is an http URI, but
>      you're not sending it over http.
>
>   I think that supporting this gives a small benefit at the cost of more
> difficult implementation and non-compatibility with existing software.
>
>   I don't believe it is worth it.
>

I believe it is. The tel: URI is a good case for this.

There are three pieces of information that the Record-Route:
needs to convey:

1) What do you put in the Request-URI?
2) Where (IP address/port) do I send the message?
3) What is the context/state that the proxy associated with this RR?

The first item should be the addr-spec in the RR.
The second item should be separate and should be a parameter of the RR (not
of the addr-spec)
The third item can either be implied by the first item (provided it is
honored and not overriden
    by the Contact:) or it can be provided by a parameter of the RR.

Is this backwards compatible with RFC2543? No, because of the second item
which is
not a maddr parameter of the URI. This could be rectified for SIP URIs by
*both* adding
an maddr parameter to the SIP URI and adding a new raddr parameter to the
Record-Route:

Record-Route:
<sip:bob@money.com;maddr=proxy13.money.com>;raddr=proxy13.money.com

I don't see a clean way to provide backwards compatibility to RFC2543. This
is
mainly because the mechanisms in RFC2543 are broken in subtle ways.

my two cents
/sean
--
Sean Olson <sean.olson@ericsson.com>
Ericsson Inc.


>
> --
> Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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Subject: [SIP] New I-D on Application Server Components
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Date: Fri, 17 Nov 2000 12:41:23 -0500

Folks,

I've just submitted the following I-D to the archives:

http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-sip-app-component
s-00.txt

The draft describes an architecture for building large, complex applications
by breaking them into components. Examples of components are IVR servers,
conferencing servers, messaging servers, TTS servers, and so on. The draft
shows how SIP (along with HTTP in some cases) represents the ideal interface
between these components. The draft also discusses collection of DTMF in
this architecture, and how to unify it with speech recognition and web input
for service stimulus.

Thanks,
Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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To: Sean Olson <sean.olson@ericsson.com>
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [SIP] ID extension of REFER
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Date: Fri, 17 Nov 2000 10:28:23 -0800

At 02:02 AM 11/17/00 , Sean Olson wrote:
>Hello,
>
>I am a little confused about the purpose of this I-D. The first question
>that springs to mind is
>why you don't use "183" for this purpose? 

the semantics of 183 already means something different, and i did not think
I could overload those semantics cleanly.

>The second question is whether you
>intend the 189 to lead to different timing than any other provisional
>response?

A sends REFER to B
B sends INVITE to C

A would get 189s any time B got provisionals from C.  

If C just 200 OKs  B's INVITE, no 189s are sent.
If C sends 180 or 183 and answers right away, A still gets a 189.  If you
don't need the information you can ignore it.

thanks,
-rohan

>(In other words, if the 189 does NOT cause different timing, than the
>issue of a long
>  period (> 37 seconds) between the INVITE and 200 OK is not really
>solved. I realize
>  this should be a very rare case.)
>
>/sean
>
>Rohan Mahy wrote:
>
>>  Hi,
>>
>> I just submitted an ID for an extension to the REFER method that I
>> discussed with Robert Sparks at Pittsburgh.  Until it appears in the
>> archive it is available at:
>>
>> ftp://ftpeng.cisco.com/rmahy/draft-mahy-sip-189-00.txt
>>
>>
>> A SIP Extension: Informational Responses to the REFER method
>>
>> Abstract
>>
>>      This document defines a SIP extension to the REFER method of the
>>      SIP Call Control Framework.  This extension provides for the
>>      ability to convey information about the progress of the Referred
>>      request when that request is in a provisional state for a long
>>      period of time (many seconds or minutes).
>>
>>
>> Comments welcome.
>>
>> thanks,
>> -rohan
>>
>>
>


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Date: Fri, 17 Nov 2000 13:49:54 -0500


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I think even a lazy parser needs to look at just keyword that identifies the
header type (i.e. from: to: etc.)  immediately.  If no keyword matches, then
the header is unrecognized and the mechanism to deal with unrecognized headers
should be invoked promptly (thus allowing an implementation to extend SIP in a
well architected fashion). On the other hand, if there is a matching  header
identifier,  then there is no need to parse the rest of the header immediately.
It can be deferred until the application explicitly requests the
JAIN-SIP implementation to do so. Thus I think rather than handle this case
with just returning a header type like general header, you may wish to consider
an event mechanism that would invoke a handler for unsupported header (or throw
an exception if you prefer).

What I meant below by "dropping it" is not  the parser discarding the header
but  an application ( say  a proxy server)  discarding the header that  *is*
of one of the types defined in the RFC but does not parse correctly.  It is
actually somewhat tricky to insist that an application be required to respond
when a message contains a mangled header,   as the parser  must be able to
parse enough of the header to construct a meaningful response (and know whom to
respond to).  It appears that the thing to do under these circumstances would
be to make responses to mangled messages optional.  (Of course
"good" implementations will always try to respond promptly even under erroneous
conditions).

One clarification that Chris could help me with:  Say I buy a JAIN-SIP  proxy
server and I want to write an extension. Ideally, I would like to extend the
proxy server without touching its code. How could this be done (i.e. what
mechanisms have you built in)  with what you have designed ? (I am not claiming
it cant be done-- just that I don't understand how to do it. )


Regards,


Ranga.

Chris Harris wrote:

> Sean,
>
> If dropping the malformed header is considered wrong, then one solution in
> the JAIN context would be to use the generic Header class instead of the
> particular subclass, say FromHeader. But this assumes that the header is
> fully parsed before the object is created - if lazy-parsing is used the
> FromHeader will be created based on the header name - and the fact that the
> header content is malformed will not be discovered until the application
> attempts to access the various parts of the header.
>
> As an alternative, a JainSipParseException could be thrown by the get
> methods, and if one is thrown, the application must then call the getValue()
> method to get the string version of the malformed content?
>
> Chris
>
> Sean Olson wrote:
>
> > Just one brief comment.
> > /sean
> >
> > Chris Harris wrote:
> >
> > > Ranga,
> > > Thanks for the input! Comments below...
> > > Regards,
> > > Chris
> > >
> > > M. Ranganathan wrote:
> > > >
> > > > Chris Harris wrote:
> > > >
> >
> > <snip>
> >
> > >
> > > > Prompt SIP responses from the server under such circumstances are not
> > > > a requirement of the spec (as far as I know).  Hence, the way
> > > > I understand it (correct me if I am wrong)  it would be legal for the
> > > > server to do a number of things under this scenario including just
> > > > ignore the mangled header  (and drop it silently)  so this behavior
> > > > is  best left unspecified in JAIN-SIP (gives implementations the
> > > > ability to choose to do whatever they want).
> > > >
> > > Is dropping the broken header the usual way to handle this case? Do any
> > > SIP implementations keep the header and leave the header content as a
> > > string?
> >
> > To be really "future proof", you should return the header content as is,
> > even if it doesn't parse correctly (according to the parser's notion of
> > correctness). Dropping it seems very wrong.
> >
> > > > Thanks.
> > > > Regards,
> > > > Ranga.
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip

--
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
Tel: 301 975 3664 Fax: 301 590 0932



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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<body text="#000000" bgcolor="#FFFFFF" link="#0000FF" vlink="#FF0000" alink="#000088">
I&nbsp;think even a lazy parser needs to look at just keyword that identifies
the header type (i.e. from: to: etc.)&nbsp; immediately.&nbsp; If no keyword
matches, then the header is unrecognized and the mechanism to deal with
unrecognized headers should be invoked promptly (thus allowing an implementation
to extend SIP&nbsp;in a well architected fashion). On the other hand, if
there is a matching&nbsp; header identifier,&nbsp; then there is no need
to parse the rest of the header immediately. It can be deferred until the
application explicitly requests the JAIN-SIP&nbsp;implementation to do
so. Thus I&nbsp;think rather than handle this case with just returning
a header type like general header, you may wish to consider an event mechanism
that would invoke a handler for unsupported header (or throw an exception
if you prefer).
<p>What I&nbsp;meant below by "dropping it"&nbsp;is not&nbsp; the parser
discarding the header but&nbsp; an application ( say&nbsp; a proxy server)&nbsp;
discarding the header that&nbsp; *is*&nbsp; of one of the types defined
in the RFC but does not parse correctly.&nbsp; It is actually somewhat
tricky to insist that an application be required to respond when a message
contains a mangled header,&nbsp;&nbsp; as the parser&nbsp; must be able
to parse enough of the header to construct a meaningful response (and know
whom to respond to).&nbsp; It appears that the thing to do under these
circumstances would be to make responses to mangled messages optional.&nbsp;
(Of course "good"&nbsp;implementations will always try to respond promptly
even under erroneous conditions).
<p>One clarification that Chris could help me with:&nbsp; Say I&nbsp;buy
a JAIN-SIP&nbsp; proxy server and I&nbsp;want to write an extension. Ideally,
I would like to extend the proxy server without touching its code. How
could this be done (i.e. what mechanisms have you built in)&nbsp; with
what you have designed ? (I&nbsp;am not claiming it cant be done-- just
that I&nbsp;don't understand how to do it. )
<br>&nbsp;
<p>Regards,
<br>&nbsp;
<p>Ranga.
<p>Chris Harris wrote:
<blockquote TYPE=CITE>Sean,
<p>If dropping the malformed header is considered wrong, then one solution
in
<br>the JAIN context would be to use the generic Header class instead of
the
<br>particular subclass, say FromHeader. But this assumes that the header
is
<br>fully parsed before the object is created - if lazy-parsing is used
the
<br>FromHeader will be created based on the header name - and the fact
that the
<br>header content is malformed will not be discovered until the application
<br>attempts to access the various parts of the header.
<p>As an alternative, a JainSipParseException could be thrown by the get
<br>methods, and if one is thrown, the application must then call the getValue()
<br>method to get the string version of the malformed content?
<p>Chris
<p>Sean Olson wrote:
<p>> Just one brief comment.
<br>> /sean
<br>>
<br>> Chris Harris wrote:
<br>>
<br>> > Ranga,
<br>> > Thanks for the input! Comments below...
<br>> > Regards,
<br>> > Chris
<br>> >
<br>> > M. Ranganathan wrote:
<br>> > >
<br>> > > Chris Harris wrote:
<br>> > >
<br>>
<br>> &lt;snip>
<br>>
<br>> >
<br>> > > Prompt SIP responses from the server under such circumstances
are not
<br>> > > a requirement of the spec (as far as I know).&nbsp; Hence, the
way
<br>> > > I understand it (correct me if I am wrong)&nbsp; it would be
legal for the
<br>> > > server to do a number of things under this scenario including
just
<br>> > > ignore the mangled header&nbsp; (and drop it silently)&nbsp;
so this behavior
<br>> > > is&nbsp; best left unspecified in JAIN-SIP (gives implementations
the
<br>> > > ability to choose to do whatever they want).
<br>> > >
<br>> > Is dropping the broken header the usual way to handle this case?
Do any
<br>> > SIP implementations keep the header and leave the header content
as a
<br>> > string?
<br>>
<br>> To be really "future proof", you should return the header content
as is,
<br>> even if it doesn't parse correctly (according to the parser's notion
of
<br>> correctness). Dropping it seems very wrong.
<br>>
<br>> > > Thanks.
<br>> > > Regards,
<br>> > > Ranga.
<br>>
<br>> _______________________________________________
<br>> SIP mailing list
<br>> SIP@lists.bell-labs.com
<br>> <a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></blockquote>

<pre>--&nbsp;
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.&nbsp;
Tel: 301 975 3664 Fax: 301 590 0932</pre>
&nbsp;
</body>
</html>

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It's clear that interoperability is on the critical path to your success.
You can tap into dynamicsoft's experience and expertise by testing your SIP
systems with dynamicsoft's proxy & location servers.  To join the community
of users who already are benefitting from testing with us, go to
www.dynamicsoft.com and click "Send a request" in the "Spotlights" message
at the bottom of the page.  

---
Steven R. Donovan
Architect                           dynamicsoft                         
mailto:sdonovan@dynamicsoft.com     5100 Tennyson Parkway
sip:sdonovan@sip.dynamicsoft.com    Plano, Texas 75025
tel:+1-972-473-5469                 http://www.dynamicsoft.com

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Date: Fri, 17 Nov 2000 16:05:26 -0600

  Hi,

  We've written a brief draft describing the use of a new header,
Replaces, based slightly on the spirit of the old one from sip-cc-01.

  The purpose of this extension is to provide a method for allowing an
active call-leg to be replaced by a new incoming call-leg for the
purposes of call handoff using REFER.  It is intended to provide a
reasonable amount of backwards compatibility and also be easy to
understand.

  It is available online at:

  http://www.sip-happens.com/replaces/draft-biggs-sip-replaces-00.txt

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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From sip-admin@lists.bell-labs.com  Fri Nov 17 18:14:01 2000
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Subject: [SIP] SIP for Presence Bof Meeting at San Diego
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Date: Fri, 17 Nov 2000 18:11:18 -0500

Folks,

I'm pleased to announce that the SIP for presence and instant messaging work
is going to have a BoF session at the next IETF. The agenda has not been
updated yet to reflect it, but the meeting will replace one hour of the
current IMPP slot on Monday.

The proposed agenda is:

The SIP for Instant Messaging and Presence Leveraging Extensions
(SIMPLE) BoF session will investigate forming a working group to
standardize on SIP for presence as a transfer protocol supporting
within the Common Format for Presence and Instant Messaging (CPIM)
framework. 

Suggested reading:

SIP for presence:
---------------
http://search.ietf.org/internet-drafts/draft-rosenberg-impp-im-00.txt
http://search.ietf.org/internet-drafts/draft-rosenberg-impp-presence-00.txt
http://search.ietf.org/internet-drafts/draft-rosenberg-impp-qauth-00.txt


CPIM:
-----
http://www.ietf.org/rfc/rfc2779.txt
http://www.ietf.org/internet-drafts/draft-ietf-impp-cpim-00.txt

SIP Events:
----------
http://search.ietf.org/internet-drafts/draft-roach-sip-subscribe-notify-01.t
xt


Proposed agenda:

1. Update on status on IMPP and SIP for presence efforts
2. Quick overview of CPIM
3. Quick overview of SIP for IM and Presence
3. Relationship of SIP for presence and SIP events work
4. Charter bashing
    a. division of work between this group and sip group
    b. specific deliverables
5. Open issues in SIP for presence work


Looking forward to seeing everyone in San Diego,

Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

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Date: Fri, 17 Nov 2000 15:29:38 -0800
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Billy Biggs:
> Snyder, Duke (Duke.Snyder@inin.com):
>
> > Bell sends INVITE, sdp=coders 0, 3, 4, 5.
> > Watson sends OK, sdp=coders 0 3.
> >
> > How do Bell and Watson determine what coder to use (0 or 3 are common
> > - but which one) since there is no coder negotiation in RTP.
>
>   They each choose whichever they prefer.  Bell should be prepared to
> receive any of 0, 3, 4 or 5 and Watson should be prepared to receive
> either of 0 or 3.
>
>   As well, they should note that the incoming codec can change at any
> point in time, and that the incoming RTP can come from any IP address,
> not just the one you are sending to.
>
>   The media flow is not a two-way connection, it is two one-way
> connections.

They should also be able to receive multiple audio streams, with different
SSRC at a single RTP port. Most implementations out there today do not
conform to the spec and will not handle this. Could I do a quick survey of
implementations out there - what RTP functionality do you support (should
this be in the bake-off torture test?):

1) changing codecs on the fly
2) receiving packets that contain different amounts of audio
3) handling RTCP at all
4) receiving RTP from any address
5) multiple audio streams (how many?)
6) SSRC clash resolution
7) multicast

any others?

Our implementation will support all of the above, and 100s of audio streams.

Simon Barber


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From sip-admin@lists.bell-labs.com  Sat Nov 18 01:15:05 2000
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From: "farhan" <farhan@hotfoon.com>
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Subject: [SIP] a stupid question on 3way handshake
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Date: Sat, 18 Nov 2000 11:36:14 +0530
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why do we have a 3 way handshake for INVITE? is an INVITE-Response not
enough ? why?
i do know some of the issue involved with a 2 way handshake. but can someone
tell me about what are all the problems with just an INVITE-Response system
(without needing an ACK) .

- farhan


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Date: Fri, 17 Nov 2000 22:29:56 -0800
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This is to do with the rather confusing conecpt of the SIP transaction, and
reliability. In my opinion we need a document that clearly documents this,
since it is one of the most confusing aspects of the protocol - and it got
that way because the protocol has 'evolved'.

The INVITE retransmission will stop on receiving a provisional response, so
the ACK is needed to ensure the reliability (and stop retransmission of) the
final response.

(correct me if I'm wrong please!)

Simon.

----- Original Message -----
From: "farhan" <farhan@hotfoon.com>
To: <discussion@sipforum.org>; "Chris Harris" <charris@dynamicsoft.com>
Cc: <mranga@nist.gov>; <jainsip@sun.com>; <sip@lists.bell-labs.com>;
<discussion@sipforum.org>
Sent: Friday, November 17, 2000 10:06 PM
Subject: [SIP] a stupid question on 3way handshake


> why do we have a 3 way handshake for INVITE? is an INVITE-Response not
> enough ? why?
> i do know some of the issue involved with a 2 way handshake. but can
someone
> tell me about what are all the problems with just an INVITE-Response
system
> (without needing an ACK) .
>
> - farhan
>
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>


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From: Stuart Bargon <c9314310@studentmail.newcastle.edu.au>
Subject: RE: [SIP] a stupid question on 3way handshake
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I would suggest looking into the solutions developed to get around the two
armies problem. The classic computer science problem goes something like
this:

Two generals with two armies are besieging a city, each on the opposite side
of the city. They have very good clocks. They communicate by messengers that
may be killed on the way. The generals need to agree on a time when to
attack simultaneously the city. If they attack separately, they will lose.
If together they will win.

-----Original Message-----
From: sip-admin@lists.bell-labs.com
[mailto:sip-admin@lists.bell-labs.com]On Behalf Of farhan
Sent: Saturday, 18 November 2000 5:06 PM
To: discussion@sipforum.org; Chris Harris
Cc: mranga@nist.gov; jainsip@sun.com; sip@lists.bell-labs.com;
discussion@sipforum.org
Subject: [SIP] a stupid question on 3way handshake


why do we have a 3 way handshake for INVITE? is an INVITE-Response not
enough ? why?
i do know some of the issue involved with a 2 way handshake. but can someone
tell me about what are all the problems with just an INVITE-Response system
(without needing an ACK) .

- farhan


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SIP@lists.bell-labs.com
http://lists.bell-labs.com/mailman/listinfo/sip


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From: "farhan" <farhan@hotfoon.com>
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well simon i am glad that someone shares the confusion. is dr. rosenberg on
the list? i would love to hear from him on why the protocol got that way.


> This is to do with the rather confusing conecpt of the SIP transaction,
and
> reliability. In my opinion we need a document that clearly documents this,
> since it is one of the most confusing aspects of the protocol - and it got
> that way because the protocol has 'evolved'.
>
> The INVITE retransmission will stop on receiving a provisional response,
so
> the ACK is needed to ensure the reliability (and stop retransmission of)
the
> final response.
>
> (correct me if I'm wrong please!)
>
> Simon.



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Subject: Re: [SIP] a stupid question on 3way handshake
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farhan (farhan@hotfoon.com):

> why do we have a 3 way handshake for INVITE? is an INVITE-Response not
> enough ? why?
> i do know some of the issue involved with a 2 way handshake. but can
> someone tell me about what are all the problems with just an
> INVITE-Response system (without needing an ACK) .

  I see two reasons why an ACK system is useful:

1. Forking.  Remember that an initial request creates many call legs.
   Say that A sends a request to a proxy which forks in parallel to
   three locations:

       __                               __                 
    A  /\ ---------> Proxy -----------> /\   B    200 OK / T:u@p;tag=5
          SOMEMETHOD       \            __              
          T:u@p              ---------> /\   C    200 OK / T:u@p;tag=6
          F:A;tag=4          \          __              
                               -------> /\   D    200 OK / T:u@p;tag=7

   Say all three answer the request at the same time with a 200 OK.  We
   now have created three call legs: A-B (4-5), A-C (4-6), and A-D
   (4-7).

   Consider an unreliable transport (UDP).  How do we ensure that A
   receives all three of these responses?  The HTTP request-response
   model was not built to handle forking.

   One solution is to reliably transmit final responses.  Each of B, C,
   and D can retransmit their response until they get an acknowledgement
   that someone actually got it.  This is really a hack to the model,
   and you get some weird interactions:  For example, in SIP you must
   retransmit final responses to INVITEs even when used over TCP!

   SIP decided that this stronger form of reliability is only useful for
   INVITE requests.  It is critical for INVITE that all successful
   responses are received: otherwise we would get mysterious calls where
   one end thinks it's connected, but the other end never got the
   response.  So, INVITE is treated as a big special case where
   responses are retransmitted until they hear an ACK.

   Unfortunately, this means that requests such as MESSAGE cannot take
   full advantage of the forking capabilities of SIP, since you cannot
   reliably tell how many endpoints received a forked request.  Another
   problem method is SUBSCRIBE.

   Consequently, in order to build a deterministic system proxies should
   only parallel fork INVITEs, and wait for confirmation of a CANCEL
   before trying another branch of a non-INVITE sequential fork.

2. Optimization.  The assumption in SIP is that INVITE transactions take
   a long time, while everything else is pretty much immediate.  So, in
   order to optimize our flow, we retransmit the INVITE until we hear a
   provisional response, and then wait for the remote end to retransmit
   its final response.

   Unfortunately, this assumption is now false, and long-outstanding
   requests such as REFER must be continuously retransmitted to poll for
   the response when used over UDP.

  The main problem is trying to use the request/response model with the
forking concept.  If I were redesigning SIP, I would clean things up by
making every transaction a 3-way handshake for protection.  I guess you
would also want to fix the retransmissions over TCP thing too.  Ugh.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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>   I see two reasons why an ACK system is useful:
>
> 1. Forking.  Remember that an initial request creates many call legs.
>    Say that A sends a request to a proxy which forks in parallel to
>    three locations:

> 2. Optimization.  The assumption in SIP is that INVITE transactions take
>    a long time, while everything else is pretty much immediate.  So, in
>    order to optimize our flow, we retransmit the INVITE until we hear a
>    provisional response, and then wait for the remote end to retransmit
>    its final response.
>    Unfortunately, this assumption is now false, and long-outstanding
>    requests such as REFER must be continuously retransmitted to poll for
>    the response when used over UDP.

yes billy, forking is out of question without an ACKnowledgement. thanks for
pointing this out. however, as forking is not an essential behaviour,
couldn't it have moved away to
be invoked only under certain conditions like Requires:acknowledgment. ?


TCP/IP does a three way handshake too. it however has very different reasons
for the handshake. TCP/IP's three way handshake has been called 'necessary
and sufficient' by late richard stevens. tcp/ip handshake is not for a
transaction but for a stream creation. therefore, the starting sequences
have to be acknowlegded by both ends. There are actual two two way
handshakes in tcp/ip, called end point sends a syn and gets and
acknowledgement, then the called party sends syn and gets an
acknowledgement. The four way handshake is merged into a three way handshake
by riding the called party syn and ack into a single packet. However, that
is no parallel for SIP as I see it.

I am wondering if there are also issues of call parameter negotiation that
require (or get enhanced) by a three way handshake





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Date: Sat, 18 Nov 2000 07:06:43 -0600

farhan (farhan@hotfoon.com):

> yes billy, forking is out of question without an ACKnowledgement.
> thanks for pointing this out. however, as forking is not an essential
> behaviour, couldn't it have moved away to be invoked only under
> certain conditions like Requires:acknowledgment. ?

  Nope.  There is no way to tell if a request will fork, and for
security reasons the proxy server may not wish to disclose that the call
forked to multiple locations.

> [...] tcp/ip handshake is not for a transaction but for a stream
> creation. therefore, the starting sequences have to be acknowlegded by
> both ends. [...]

  Right, but in SIP you can setup a signalling connection (the parallel
to a TCP/IP stream) using any method, not just INVITE.  So, as you
noted, setup of a SIP connection definitely does not require a 3-way
handshake.  It's just that INVITE needs to be more robust and
responsive.

> I am wondering if there are also issues of call parameter negotiation
> that require (or get enhanced) by a three way handshake

  Not really.  A client can delay sending a session description by not
sending one in the INVITE and later sending it in the ACK, but that is
simply a convenient optimization.  In an INVITE transaction there are
only ever two session descriptions, one for each end.  For an API, the
ACK is only to stop retransmissions, there is no need to have a concept
of a 3-way handshake.

  One thing I forgot to mention: another reason why INVITE needs to have
this special mechanism is to ensure speedy call setup.  When the called
party picks up their phone the response is retransmitted to help setup
the call as quick as possible.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Subject: Re: [SIP] a stupid question on 3way handshake
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The speedy call setup is the main (and motivating) reason. The only
other alternative (with an unreliable transport) would have been to have
a separate, callee-initiated transaction that says "I accept the call".

All of this motivation, by the way, is contained in the spec in the
section on transport. The reason for not making other methods 3-way was
to make them simpler and to reduce message count, something that our
wireless friends are very worried about. The ACK does allow to delay the
session description, which is useful in some scenarios (some versions of
H.323 interoperation, for example), but that by itself is mostly a minor
advantage.

As usual, optimization yields complexity.

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Subject: Re: [SIP] a stupid question on 3way handshake
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>   I see two reasons why an ACK system is useful:
>
> 1. Forking.  Remember that an initial request creates many call legs.
>    Say that A sends a request to a proxy which forks in parallel to
>    three locations:

> 2. Optimization.  The assumption in SIP is that INVITE transactions take
>    a long time, while everything else is pretty much immediate.  So, in
>    order to optimize our flow, we retransmit the INVITE until we hear a
>    provisional response, and then wait for the remote end to retransmit
>    its final response.
>    Unfortunately, this assumption is now false, and long-outstanding
>    requests such as REFER must be continuously retransmitted to poll for
>    the response when used over UDP.

yes billy, forking is out of question without an ACKnowledgement. thanks for
pointing this out. however, as forking is not an essential behaviour,
couldn't it have moved away to
be invoked only under certain conditions like Requires:acknowledgment. ?


TCP/IP does a three way handshake too. it however has very different reasons
for the handshake. TCP/IP's three way handshake has been called 'necessary
and sufficient' by late richard stevens. tcp/ip handshake is not for a
transaction but for a stream creation. therefore, the starting sequences
have to be acknowlegded by both ends. There are actual two two way
handshakes in tcp/ip, called end point sends a syn and gets and
acknowledgement, then the called party sends syn and gets an
acknowledgement. The four way handshake is merged into a three way handshake
by riding the called party syn and ack into a single packet. However, that
is no parallel for SIP as I see it.

I am wondering if there are also issues of call parameter negotiation that
require (or get enhanced) by a three way handshake




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Subject: Re: [SIP] a stupid question on 3way handshake
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Hi,

A 2-way handshake would lead to the following situation:

A sends an INVITE to B and begin retransmitting it. After a while A
gives up and stops retransmitting the INVITE. A thinks that there is no
session established.

At the same time (when A is giving up), B sends a 200 OK back. Since B
observes that A has stopped retransmitting the INVITE the 200 OK in not
retransmitted, B thinks that the session is established...

A 3-way hand-shake helps synchronizing both ends during session
establishment.

Regards,

Gonzalo

Henning Schulzrinne wrote:
> 
> The speedy call setup is the main (and motivating) reason. The only
> other alternative (with an unreliable transport) would have been to have
> a separate, callee-initiated transaction that says "I accept the call".
> 
> All of this motivation, by the way, is contained in the spec in the
> section on transport. The reason for not making other methods 3-way was
> to make them simpler and to reduce message count, something that our
> wireless friends are very worried about. The ACK does allow to delay the
> session description, which is useful in some scenarios (some versions of
> H.323 interoperation, for example), but that by itself is mostly a minor
> advantage.
> 
> As usual, optimization yields complexity.
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo

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Subject: Re: [SIP] a stupid question on 3way handshake
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> A sends an INVITE to B and begin retransmitting it. After a while A
> gives up and stops retransmitting the INVITE. A thinks that there is no
> session established.
>
> At the same time (when A is giving up), B sends a 200 OK back. Since B
> observes that A has stopped retransmitting the INVITE the 200 OK in not
> retransmitted, B thinks that the session is established...

imagine the samething happening on the ACK. Party B keeps sending 200 OK
until it times out and just then Party A responds back with an ACK and
considers that the session is established. No, that is not a solution we
need an infinite way handshake in such a case.
- farhan


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Date: Sun, 19 Nov 2000 01:52:47 -0500

Sorry for taking so long to respond to many mails on the list the past few
weeks... I've been focusing on completing my dissertation...


 

> -----Original Message-----
> From: farhan [mailto:farhan@hotfoon.com]
> Sent: Saturday, November 18, 2000 7:24 AM
> To: Billy Biggs
> Cc: SIP List
> Subject: Re: [SIP] a stupid question on 3way handshake
> 
> 
> >   I see two reasons why an ACK system is useful:
> >
> > 1. Forking.  Remember that an initial request creates many 
> call legs.
> >    Say that A sends a request to a proxy which forks in parallel to
> >    three locations:
> 
> > 2. Optimization.  The assumption in SIP is that INVITE 
> transactions take
> >    a long time, while everything else is pretty much 
> immediate.  So, in
> >    order to optimize our flow, we retransmit the INVITE 
> until we hear a
> >    provisional response, and then wait for the remote end 
> to retransmit
> >    its final response.
> >    Unfortunately, this assumption is now false, and long-outstanding
> >    requests such as REFER must be continuously 
> retransmitted to poll for
> >    the response when used over UDP.
> 
> yes billy, forking is out of question without an 
> ACKnowledgement. thanks for
> pointing this out. 

I'm sorry, but this is not correct.

Forking is fine for non-invite requests. It will result in reception of a
single response (which is, IMHO, a very nice property), which is the best
amongst all those received. Forking of non-INVITEs is therefore useful for
requests where it isn't important to know how many places received the
request; simply knowing the best response is adequate. As an example,
consider a hypothetical new request called "EXIST" which checks whether
there is a UA in the network that meets the specifications (along the lines
of caller preferences) described in the request. This request can fork to
check numerous locations, and a 200 indicates a match. In this case, the
client doesn't care how many requests were ultimately delivered, or to whom.
Just the answer matters.

I would argue that SUBSCRIBE meets this criteria as well. The caller really
doesn't care who many servers are going to be handling the subscription. So
long as some server accepts it, and generates notifications for it, thats
all that matters.


Regarding the original question, my recollection of the main motivation of
special handling for INVITE was that we believe it was the only request for
whom a response may not arrive for a long time, as it required non-automated
processing to answer. Speedy delivery of acceptance was another benefit.

In hindsight, I would have removed this special treatment. One request, one
final response. No ack. Handling cases like this one (where the "response"
might not come for a while), is more readily handled using two sepate
transactions in each direction. So, we would have a new request, called
INITIATE (instead of INVITE). A 200 response is sent as soon as the request
is received by a UAS. When the call is answered, an ACCEPT message is sent
by the called party, and responded to immediately with a 200 OK upon
receipt.

Billy writes:
Optimization.  The assumption in SIP is that INVITE 
> transactions take
>    a long time, while everything else is pretty much 
> immediate.  So, in
>    order to optimize our flow, we retransmit the INVITE until 
> we hear a
>    provisional response, and then wait for the remote end to 
> retransmit
>    its final response.
> 
>    Unfortunately, this assumption is now false, and long-outstanding
>    requests such as REFER must be continuously retransmitted 
> to poll for
>    the response when used over UDP.

Actually, I have begun to think that refer treatment should follow the
two-transaction model above. There is a serious problem otherwise. The issue
is that the INVITE triggered by the REFER may take more than 32 seconds for
an answer. If this happens, the REFER transaction will timeout before the
INVITE it triggered completes.

So, my proposal would be that REFER generates an immediate 200 OK if the
recipient is willing to try to send the request. We can then have the
recipient of REFER send a request in the reverse direction (we can debate
whether this is NOTIFY or something else) once the triggered INVITE
completes successfully. 

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From: Billy Biggs <Billy_Biggs@3com.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: Re: [SIP] a stupid question on 3way handshake
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Date: Sun, 19 Nov 2000 02:06:14 -0600

Jonathan Rosenberg (jdrosen@dynamicsoft.com):

> [...] Forking of non-INVITEs is therefore useful for requests where it
> isn't important to know how many places received the request; simply
> knowing the best response is adequate. [...]
> 
> I would argue that SUBSCRIBE meets this criteria as well. The caller
> really doesn't care who many servers are going to be handling the
> subscription. So long as some server accepts it, and generates
> notifications for it, thats all that matters.

  It matters when trying to unsubscribe.  To be intelligence-
at-the-edges correct, an unsubscribe should be sent within the call-leg
created by the original SUBSCRIBE.  However, to be forking-safe, the
unsubscribe should be sent to the orignal address which was subscribed
to, without a tag.

  It is a minor complaint.  I know that to keep my implementation clean,
a SUBSCRIBE in my stack will create a single call leg, and any NOTIFYs
received outside of that call leg will be given 481s.  I don't buy the
"[...] subscribers SHOULD be prepared to receive notifications with
previously unknown tags in the "From" field [...]" argument.  Oh well.

> > The assumption in SIP is that INVITE transactions take a long time,
> > while everything else is pretty much immediate.  So, in order to
> > optimize our flow, we retransmit the INVITE until we hear a
> > provisional response, and then wait for the remote end to retransmit
> > its final response.
> > 
> > Unfortunately, this assumption is now false, and long-outstanding
> > requests such as REFER must be continuously retransmitted to poll
> > for the response when used over UDP.
> 
> Actually, I have begun to think that refer treatment should follow the
> two-transaction model above. There is a serious problem otherwise. The
> issue is that the INVITE triggered by the REFER may take more than 32
> seconds for an answer. If this happens, the REFER transaction will
> timeout before the INVITE it triggered completes.

  As well, there is no way to tell if the far end accepted the REFER and
is processing it, or if it will eventually return a 501.  Responding to
REFER immediately helps with error checking.

  We would also be less likely to have to worry about multiple
outstanding transactions on a call-leg.

  However, it is much more verbose, and a major change to the REFER
draft.  These were the two reasons I didn't suggest it earlier.  I might
submit a BYE/Also transfer draft if this proposal flies, or maybe
advocate louder it be added to bis.  Thoughts?

> So, my proposal would be that REFER generates an immediate 200 OK if
> the recipient is willing to try to send the request. We can then have
> the recipient of REFER send a request in the reverse direction (we can
> debate whether this is NOTIFY or something else) once the triggered
> INVITE completes successfully. 

  I'm all for it.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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        Henning Schulzrinne <hgs@cs.columbia.edu>,
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Subject: Re: [SIP] a stupid question on 3way handshake
References: <3A12B434.EE04C0E7@dynamicsoft.com> <3A130AB9.9A527F9E@nist.gov> <3A131F80.9060704@dynamicsoft.com> <3A151FBB.57D381CD@ericsson.com> <003201c05125$b222f890$1200a8c0@bigboy> <20001118052434.A32315@div8.net> <000a01c0515a$a2777c60$1200a8c0@bigboy> <20001118070643.A32698@div8.net> <3A16B47E.819A7F53@cs.columbia.edu> <3A16C519.8A75A321@lmf.ericsson.se> <001101c0519e$bfb867f0$1200a8c0@bigboy>
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Hi,

You were asking for the historical reason of having a 3-way handshake,
and that is what I answered.

Check page 6 of the first version of SIP and you will find the same
explanation as I gave.
http://www.cs.columbia.edu/sip/drafts/draft-ietf-mmusic-sip-00.ps

More comments inline:

farhan wrote:
> 
> > A sends an INVITE to B and begin retransmitting it. After a while A
> > gives up and stops retransmitting the INVITE. A thinks that there is no
> > session established.
> >
> > At the same time (when A is giving up), B sends a 200 OK back. Since B
> > observes that A has stopped retransmitting the INVITE the 200 OK in not
> > retransmitted, B thinks that the session is established...
> 
> imagine the samething happening on the ACK. Party B keeps sending 200 OK
> until it times out and just then Party A responds back with an ACK and
> considers that the session is established. No, that is not a solution we
> need an infinite way handshake in such a case.

Actually, if the network fails completely to deviler datagramas you
definitively have a problem. However, a 3-way handshake makes B
retransmit his 200 OK a number of times since the ACK has to been
received.

A big difference between the rest of the requests in SIP (that have a
2-way handshake) and INVITE (3-way handshake) is that the rest of the
requests are usually retransmitted by the UA until the transaction is
finished without the user stopping this process.

For example, if a user hangs up (he is using a SIP phone) his UA will
retransmit the BYE as many time as necessary until it gets a final
response.

However, the user can try to call someone and hang up before a final
response arrives. This stops the retransmission process of the INVITE.

But even if the 200 OK got lost, since the ACK will never arrive the
callee will not believe that there is an established session.

Best regards,

Gonzalo

> - farhan
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 31 18
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland

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Date: Sun, 19 Nov 2000 13:23:10 -0600
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Jonathan Rosenberg wrote:
[...]
> Forking is fine for non-invite requests. It will result in reception of a
> single response (which is, IMHO, a very nice property), which is the best
> amongst all those received. Forking of non-INVITEs is therefore useful for
> requests where it isn't important to know how many places received the
> request; simply knowing the best response is adequate. 
[...]

Except when multiple downstream UASes return a "best" response to a forked
non-INVITE request.  A while ago, we had a similar dialog regarding the
forking of a non-INVITE request like OPTIONS.  A forking proxy would send
the initial OPTIONS to several branches downstream; suppose 2 branches
returned a 200 OK.  Then the proxy would send the first 200 OK received up-
stream; absord the second 200 OK and send CANCEL on the remaining branches. 
Or the proxy could simply send multiple 200 OKs upstream and hope the UAC
can sort them out and pick the "best".

From a proxy's point of view, what exactly to do when multiple downstream 
UASes return a "best" response should be codified, IMHO.  Otherwise, we 
could run into scenarios where on certain requests the proxy ONLY sends one 
"best" response upstream and in other cases it must send all "best"
responses received upstream and leave the arbitration to the UAC.

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Internet Software and eServices Group 
Lucent Technologies/Bell Labs Innovations 263 Shuman Blvd., Rm 1A-413
Naperville, Illinois 60566     Voice: +1 630 224 0216   Fax: +1 630 713 0184

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From: Billy Biggs <Billy_Biggs@3com.com>
To: Vijay Gurbani <vkg@lucent.com>
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Subject: Re: [SIP] a stupid question on 3way handshake
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Vijay Gurbani (vkg@lucent.com):

> Jonathan Rosenberg wrote:
> [...]
> > Forking is fine for non-invite requests. It will result in reception
> > of a single response (which is, IMHO, a very nice property), which
> > is the best amongst all those received. Forking of non-INVITEs is
> > therefore useful for requests where it isn't important to know how
> > many places received the request; simply knowing the best response
> > is adequate. 
> [...]
> 
> Except when multiple downstream UASes return a "best" response to a
> forked non-INVITE request.  A while ago, we had a similar dialog
> regarding the forking of a non-INVITE request like OPTIONS.  A forking
> proxy would send the initial OPTIONS to several branches downstream;
> suppose 2 branches returned a 200 OK.  Then the proxy would send the
> first 200 OK received up- stream; absord the second 200 OK and send
> CANCEL on the remaining branches.  Or the proxy could simply send
> multiple 200 OKs upstream and hope the UAC can sort them out and pick
> the "best".

  The behavior is the same as for INVITE requests.  If you receive
multiple 200 OK responses, you MUST forward them up.  That has been the
consensus from everyone I've ever spoken with.

  However, since these responses are 'unreliable', dropping them is
simply a simulation of bad network conditions, but I wouldn't recommend
it.

  I was pretty sure this is spelled out in 2543, but I can't find an
appropriate quote.  Maybe it should be better spelled out.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Billy Biggs wrote:
[...]
> > Except when multiple downstream UASes return a "best" response to a
> > forked non-INVITE request.  A while ago, we had a similar dialog
> > regarding the forking of a non-INVITE request like OPTIONS.  A forking
> > proxy would send the initial OPTIONS to several branches downstream;
> > suppose 2 branches returned a 200 OK.  Then the proxy would send the
> > first 200 OK received up- stream; absord the second 200 OK and send
> > CANCEL on the remaining branches.  Or the proxy could simply send
> > multiple 200 OKs upstream and hope the UAC can sort them out and pick
> > the "best".
> 
>   The behavior is the same as for INVITE requests.  If you receive
> multiple 200 OK responses, you MUST forward them up.  That has been the
> consensus from everyone I've ever spoken with.

Hmmm, let's see...the RFC does indicate that a forking proxy MAY terminate
pending INVITE requests on other branches when it gets a 200 OK for at least 
one branch.  Following this logic, it proxies the 200 OK it got _first_ up-
stream and sends a CANCEL on all other branches.  Say it gets another 200
OK (INVITE) after it has send a CANCEL; the proxy can rest assured that the
CANCEL and the 200 OK (INVITE) "crossed each other on the wire."  Thus it
can locally ACK the 200 OK (INVITE) and then send a BYE on that branch.

So, what I was trying to get at is that a proxy may exhibit the behavior I 
describe above and only send the first 200 OK upstream; further 200 OK 
(INVITE) maybe processed locally.  

Certainly this is not be true for all requests.  It could very well be that 
forking proxies MUST send all 2xx responses to non-INVITE forked requests 
upstream as soon as they get it.  For INVITE requests, they may exhibit
the behavior described above.  We should simply state that in the RFC so 
there are no ambiguities.

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Internet Software and eServices Group 
Lucent Technologies/Bell Labs Innovations 263 Shuman Blvd., Rm 1A-413
Naperville, Illinois 60566     Voice: +1 630 224 0216   Fax: +1 630 713 0184

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Vijay Gurbani (vkg@lucent.com):

> >   The behavior is the same as for INVITE requests.  If you receive
> > multiple 200 OK responses, you MUST forward them up.  That has
> > been the consensus from everyone I've ever spoken with.
> 
> Hmmm, let's see...the RFC does indicate that a forking proxy MAY
> terminate pending INVITE requests on other branches when it gets a 200
> OK for at least one branch.  Following this logic, it proxies the 200
> OK it got _first_ up- stream and sends a CANCEL on all other branches.
> Say it gets another 200 OK (INVITE) after it has send a CANCEL; the
> proxy can rest assured that the CANCEL and the 200 OK (INVITE)
> "crossed each other on the wire."  Thus it can locally ACK the 200 OK
> (INVITE) and then send a BYE on that branch.

  Sorry, but that would be really really evil.

  From RFC2543, 12.4:

   2xx: The proxy MUST forward the response upstream towards the client,
        without sending an ACK downstream. After receiving a 2xx, the
        server MAY terminate all other pending requests by sending a
        CANCEL request and closing the TCP connection, if applicable.

  It's up to the client to BYE any additional calls -if it wants to-,
not the proxy.  It's perfectly acceptable to make one call and have two
seperate locations pick up!

  The text reads that the proxy MUST not ACK the response locally to
protect the UA.  That is for any 2xx responses received.  There might be
reasons why the proxy would choose not to cancel the other branches (if
there are no other branches?  if the other branch is a call logger?  who
knows), and I think that is why the MAY is there.

  This behavior is the same for non-INVITE requests.  I guess it could
be stated more explicitly though.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Sun, 19 Nov 2000 17:04:02 -0500



 

> -----Original Message-----
> From: Billy Biggs [mailto:Billy_Biggs@3com.com]
> Sent: Sunday, November 19, 2000 4:42 PM
> To: Vijay Gurbani
> Cc: Billy Biggs; Jonathan Rosenberg; SIP List
> Subject: Re: [SIP] a stupid question on 3way handshake
> 
> 
> Vijay Gurbani (vkg@lucent.com):
> 
> > >   The behavior is the same as for INVITE requests.  If you receive
> > > multiple 200 OK responses, you MUST forward them up.  That has
> > > been the consensus from everyone I've ever spoken with.
> > 
> > Hmmm, let's see...the RFC does indicate that a forking proxy MAY
> > terminate pending INVITE requests on other branches when it 
> gets a 200
> > OK for at least one branch.  Following this logic, it 
> proxies the 200
> > OK it got _first_ up- stream and sends a CANCEL on all 
> other branches.
> > Say it gets another 200 OK (INVITE) after it has send a CANCEL; the
> > proxy can rest assured that the CANCEL and the 200 OK (INVITE)
> > "crossed each other on the wire."  Thus it can locally ACK 
> the 200 OK
> > (INVITE) and then send a BYE on that branch.
> 
>   Sorry, but that would be really really evil.

Well, its not evil, but its not correct. Proxies don't initiate requests on
their own, excepting the "special" ones of BYE and ACK (and for ACK, its
only for non-200 responses. Proxies don't ACK 200 responses, only the UA
does).

>   The text reads that the proxy MUST not ACK the response locally to
> protect the UA.  That is for any 2xx responses received.  
> There might be
> reasons why the proxy would choose not to cancel the other 
> branches (if
> there are no other branches?  if the other branch is a call 
> logger?  who
> knows), and I think that is why the MAY is there.

Yes. In fact, some time back, one of the request-disposition parameters in
the caller preferences specification was something like "don't cancel". The
point is that this is good for requests where you want to reach everyone,
rather than just the first person to answer.

> 
>   This behavior is the same for non-INVITE requests.  I guess it could
> be stated more explicitly though.

I agree that forking behavior for non-INVITE is not well specified enough.
In fact, you previously write:

> > Except when multiple downstream UASes return a "best" response to a
> > forked non-INVITE request.  A while ago, we had a similar dialog
> > regarding the forking of a non-INVITE request like OPTIONS. 
>  A forking
> > proxy would send the initial OPTIONS to several branches downstream;
> > suppose 2 branches returned a 200 OK.  Then the proxy would send the
> > first 200 OK received up- stream; absord the second 200 OK and send
> > CANCEL on the remaining branches.  Or the proxy could simply send
> > multiple 200 OKs upstream and hope the UAC can sort them 
> out and pick
> > the "best".
> 
>   The behavior is the same as for INVITE requests.  If you receive
> multiple 200 OK responses, you MUST forward them up.  That 
> has been the
> consensus from everyone I've ever spoken with.

I'm sorry, that is not correct.

A proxy cannot forward multiple non-INVITE responses upstream.

A proxy cannot forward multiple non-INVITE responses upstream.

Just to be sure:

A proxy cannot forward multiple non-INVITE responses upstream.

I'll put a FAQ entry in there too.

It just doesn't work, plain and simple. The reliability mechanism as defined
just doesn't work for multiple responses, and there is no way to fix it
without pretty much adding in an ACK for non-INVITE requests. 

Billy also writes:
> > [...] Forking of non-INVITEs is therefore useful for 
> requests where it
> > isn't important to know how many places received the request; simply
> > knowing the best response is adequate. [...]
> > 
> > I would argue that SUBSCRIBE meets this criteria as well. The caller
> > really doesn't care who many servers are going to be handling the
> > subscription. So long as some server accepts it, and generates
> > notifications for it, thats all that matters.
> 
>   It matters when trying to unsubscribe.  To be intelligence-
> at-the-edges correct, an unsubscribe should be sent within 
> the call-leg
> created by the original SUBSCRIBE.  However, to be forking-safe, the
> unsubscribe should be sent to the orignal address which was subscribed
> to, without a tag.

Actually, I thought about this some more, and am not so sure that forking of
SUBCSRIBE is such a great idea. THe problem is not just unsubscribes, but
refreshes. One of the subscribe responses will come back, and it will likely
be record-routed. As a result, subsequent subscribes will go to just the one
server whose 200 response was forwarded upstream. Now, thats OK, but only if
the multiple servers are all *equal*, that is, each is providing
notifications for exactly the same set of devices for a particular
presentity. 

So, to modify my description somewhat, "forking of non-INVITE requests works
well when the client doesn't care from whom the response comes, and when the
services provided by each of the servers that receives the request is
identical."

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Date: Sun, 19 Nov 2000 16:28:03 -0600

Jonathan Rosenberg (jdrosen@dynamicsoft.com):

> [...] Proxies don't initiate requests on their own, excepting the
> "special" ones of BYE and ACK (and for ACK, its only for non-200
> responses. [...]

  When would a proxy ever send a BYE?

> A proxy cannot forward multiple non-INVITE responses upstream.

  I'm ok with this as long as it's clearly specified.

> Actually, I thought about this some more, and am not so sure that
> forking of SUBCSRIBE is such a great idea. THe problem is not just
> unsubscribes, but refreshes. One of the subscribe responses will come
> back, and it will likely be record-routed. As a result, subsequent
> subscribes will go to just the one server whose 200 response was
> forwarded upstream. Now, thats OK, but only if the multiple servers
> are all *equal*, that is, each is providing notifications for exactly
> the same set of devices for a particular presentity. 

  Similarily, if I get a NOTIFY and I don't recognize the tags, I can't
accept it and then unsubscribe since I don't have the route information
(nor do I have a Contact address, since NOTIFY can't have a Contact. ??)
The only sane thing to do is return a 481 and have that cancel the
subscription.

  It would be wrong to ask that proxies not fork SUBSCRIBEs, or to
suggest that SUBSCRIBE should always put in some 'don't-fork' caller
preferences parameter.  Forking of SUBSCRIBEs is useful.  For example,
subscribing to my voicemail status without using caller-preferences: the
SUBSCRIBE forks to all my phones and my voicemail, and hopefully only
the voicemail will accept it.

  Aside: Yet another reason you shouldn't always use the same local tag:
         You send a SUBSCRIBE and it comes back to yourself. ;-)
         Please take this error-prone suggestion out of bis.

  So, I guess we could suggest that when designing a SUBSCRIBE/NOTIFY
system, make sure that the URIs used don't fork to multiple subscription
handlers which handle the same event?

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Sun, 19 Nov 2000 16:28:48 -0800


Was there any consensus reached on Record-Route changes? It would be nice to
get this one nailed down.

Problems to be solved:
- handling of non-SIP Request-URI's properly.
- request transaction identification in proxies for reverse direction (I
forget the specifics)
- proxy doesn't see any inserted record-route headers in reverse direction
(eg loss of state or address info)
- detecting new behaviour in Record-Route response. 

I think the first three can be solved by the UAS only replacing the
user@host:port sections by the From/Contact in the reverse Route. The first
also requires a new Record-Route header to store the maddr. This is used in
preference to the maddr which might also be placed in the header for
backwards compat.

Detecting the new behaviour in the UAS? Any ideas? Having the UAS add a
Record-Route parameter or adding something to its COntact are both fairly
kludgey but would suffice. Proxies who support the new behavior but don't
detect this added parameter in the response know that they must store full
state for the call. 

Cheers,

Robert.

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From: Billy Biggs <Billy_Biggs@3com.com>
To: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
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Subject: Re: [SIP] Record-Route consensus
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Date: Sun, 19 Nov 2000 19:48:26 -0600

Fairlie-Cuninghame, Robert (rfairlie@nuera.com):

> Was there any consensus reached on Record-Route changes? It would be
> nice to get this one nailed down.

  In my opinion, the consensus is that going through all the addresses
in the Record-Route and changing them for the reverse direction is out.
Proxies should insert an address that indicates to them enough state to
perform their function, and the spec should probably give some examples.

  The only open issue is whether to allow non-SIP URIs in the Route.  In
order to support them, the suggestion is to define Record-Route
parameters to indicate the IP address, port, and transport.  I think
this is dangerous because it won't work with outbound proxies.

  Does anyone feel strongly otherwise?

  See my post:
    http://lists.bell-labs.com/pipermail/sip/2000q4/004005.html

  And Sean Olson's post on other URI schemes:
    http://lists.bell-labs.com/pipermail/sip/2000q4/004037.html

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Subject: [SIP] New I-D on multiparty conferencing models
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Date: Sun, 19 Nov 2000 22:19:19 -0500

Folks,

I submitted a draft last Friday on multiparty conferencing models. It shows
how to do multiparty conferencing six different ways with baseline SIP.
Until it appears, you can pick up a copy at:

http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-sip-conferencing-
models-00.txt

Thanks,
Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

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From sip-admin@lists.bell-labs.com  Mon Nov 20 01:12:05 2000
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Reply-To: "Venkatesh Venkataramanan" <venkatesh.venkataramanan@wipro.com>
From: "Venkatesh Venkataramanan" <venkatesh.venkataramanan@wipro.com>
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Subject: Re: [SIP] ID extension of REFER
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Wouldn't simply forwarding the provisionals received allow the initiator of
REFER know more about 'call progress' instead of translating them to a
single 189 ? Are there any specific reasons against not doing so? Please
elaborate.

Venkatesh
----- Original Message -----
From: "Rohan Mahy" <rohan@cisco.com>
To: "Sean Olson" <sean.olson@ericsson.com>
Cc: <sip@lists.bell-labs.com>
Sent: Friday, November 17, 2000 11:58 PM
Subject: Re: [SIP] ID extension of REFER


> At 02:02 AM 11/17/00 , Sean Olson wrote:
> >Hello,
> >
> >I am a little confused about the purpose of this I-D. The first question
> >that springs to mind is
> >why you don't use "183" for this purpose?
>
> the semantics of 183 already means something different, and i did not
think
> I could overload those semantics cleanly.
>
> >The second question is whether you
> >intend the 189 to lead to different timing than any other provisional
> >response?
>
> A sends REFER to B
> B sends INVITE to C
>
> A would get 189s any time B got provisionals from C.
>
> If C just 200 OKs  B's INVITE, no 189s are sent.
> If C sends 180 or 183 and answers right away, A still gets a 189.  If you
> don't need the information you can ignore it.
>
> thanks,
> -rohan
>
> >(In other words, if the 189 does NOT cause different timing, than the
> >issue of a long
> >  period (> 37 seconds) between the INVITE and 200 OK is not really
> >solved. I realize
> >  this should be a very rare case.)
> >
> >/sean
> >
> >Rohan Mahy wrote:
> >
> >>  Hi,
> >>
> >> I just submitted an ID for an extension to the REFER method that I
> >> discussed with Robert Sparks at Pittsburgh.  Until it appears in the
> >> archive it is available at:
> >>
> >> ftp://ftpeng.cisco.com/rmahy/draft-mahy-sip-189-00.txt
> >>
> >>
> >> A SIP Extension: Informational Responses to the REFER method
> >>
> >> Abstract
> >>
> >>      This document defines a SIP extension to the REFER method of the
> >>      SIP Call Control Framework.  This extension provides for the
> >>      ability to convey information about the progress of the Referred
> >>      request when that request is in a provisional state for a long
> >>      period of time (many seconds or minutes).
> >>
> >>
> >> Comments welcome.
> >>
> >> thanks,
> >> -rohan
> >>
> >>
> >
>
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>


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From sip-admin@lists.bell-labs.com  Mon Nov 20 01:20:07 2000
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Date: Mon, 20 Nov 2000 11:47:32 +0000
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Hi,
   In SIP-URL where spaces and line breaks are allowed.
   i have seen grammar in sip-rfc2543.what i feel is
   in between "sip:" and user,
                     user and password,
                     password and host.
                     host and port
                     port and url-parameter,
                     url parameters
                    url-parameter and header
                    headers.
                   Spaces are allowed.and where ever spaces are allowed
line breaks are also  allowed.
  i want to know whether  iam right or not.if iam wrong please tell me correct
 one.
                     thanks,
regards
Sreenivasa Reddy.D




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From sip-admin@lists.bell-labs.com  Mon Nov 20 01:49:04 2000
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Date: Mon, 20 Nov 2000 12:21:09 +0530



you are pretty much on track....some additions are
     between the user and the "@"
     between the "@" and the host
     between host and " : "
     between ":" and port
     between port and ";"
     between ";" and url
     and so on...you have left out separators. You can have spaces or line
folds between any 2 tokens , separators or otherwise.

-ashok
------------
ashok i roy @hughes software systems
-------------





sreenivasa reddy D <rsreenu@miel.mot.com> on 11/20/2000 05:17:32 PM

To:   sip <sip@lists.bell-labs.com>
cc:    (bcc: Ashok I Roy/HSSBLR)

Subject:  [SIP] spaces in SIP-URL




Hi,
   In SIP-URL where spaces and line breaks are allowed.
   i have seen grammar in sip-rfc2543.what i feel is
   in between "sip:" and user,
                     user and password,
                     password and host.
                     host and port
                     port and url-parameter,
                     url parameters
                    url-parameter and header
                    headers.
                   Spaces are allowed.and where ever spaces are allowed
line breaks are also  allowed.
  i want to know whether  iam right or not.if iam wrong please tell me
correct
 one.
                     thanks,
regards
Sreenivasa Reddy.D




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SIP@lists.bell-labs.com
http://lists.bell-labs.com/mailman/listinfo/sip





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Billy Biggs wrote:
>
> Fairlie-Cuninghame, Robert (rfairlie@nuera.com):
> 
> > Was there any consensus reached on Record-Route changes? It would be
> > nice to get this one nailed down.
> 
>   In my opinion, the consensus is that going through all the addresses
> in the Record-Route and changing them for the reverse direction is out.
> Proxies should insert an address that indicates to them enough state to
> perform their function, and the spec should probably give some examples.
> 
>   The only open issue is whether to allow non-SIP URIs in the Route.  In
> order to support them, the suggestion is to define Record-Route
> parameters to indicate the IP address, port, and transport.  I think
> this is dangerous because it won't work with outbound proxies.

I'd like to take this oppurtunity &:) to point out that the
solution I proposed -- just inserting the SIP URL of the proxy,
and embedding the Request-URI as a parameter thereof -- handles
non-SIP URIs.

I did go a step further, and suggest that called UAs did extra
mangling in the reverse direction, in that they insert the
Contact address of the caller UA as another parameter; however,
it doesn't so matter whether this is done or not, since it
would only come into play if the Route broke somehow.

So, for example, say I'm a proxy on the machine sip0.example.com,
listening on port 6000.  I receive a request:

    INVITE tel:999 SIP/2.0
    ...

I proxy on:

    INVITE sip:999@emergency.sip-gateways.com SIP/2.0
    ...
    Record-Route: <sip:sip0.example.com:6000;uri=tel:999>
    ...

Hopefully, it's now evident that in either direction, I am
going to be totally routeable (as long as sip0.example.com is
globally routeable), so as long as both caller and called UA
copy all parameters.

Unfortunately, we do lose the notion that the Request-URI
contains, somehow, the ultimate destination, but it would
be possible to perhaps have a BCP along the lines of
called UAs inserting another parameter indicating the
Contact/From (I guess it could literally be done as a
header parameter).  I did originally propose renaming
uri to original-uri, and making uri the Contact/From,
yada yada, but it does seem unecessarily painful to me
now.

Thoughts?


 - Jo.


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Jonathan Rosenberg wrote:
>
> Well, its not evil, but its not correct. Proxies don't initiate 
> requests on their own, excepting the "special" ones of BYE

Do you mean CANCEL?

> So, to modify my description somewhat, "forking of non-INVITE 
> requests works well when the client doesn't care from whom the
> response comes, and when the services provided by each of the
> servers that receives the request is identical."

I'm playing Devil's Advocate here, but isn't that better
achieved by SRV?

I like forking -- it's very powerful; however, I might be
inclined to suggest that we warn in bis that proxies
shouldn't be too keen to fork on non-INVITE requests, since
in a number of cases, it's probably not going to be so
helpful.

Cheers,


 - Jo.


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airoy@hss.hns.com wrote:
> 
> you are pretty much on track....some additions are
>      between the user and the "@"
>      between the "@" and the host
>      between host and " : "
>      between ":" and port
>      between port and ";"
>      between ";" and url
>      and so on...you have left out separators. You can have spaces or line
> folds between any 2 tokens , separators or otherwise.

Why do you think spaces here are legal? As far as I can tell rfc 2396,
"Uniform Resource Identifiers (URI): Generic Syntax" seems to clearly
spaces in URIs altogether. The comments on "implied *LWS" in appendix B
of rfc 2543 doesn't change this. 

Anders

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Jo Hornsby wrote:
> 
> I like forking -- it's very powerful; however, I might be
> inclined to suggest that we warn in bis that proxies
> shouldn't be too keen to fork on non-INVITE requests, since
> in a number of cases, it's probably not going to be so
> helpful.

Hmmm. If it's OK to register multiple SIP addresses at a registrar and
have a proxy fork INVITEs to those addresses it pretty much follows that
OPTIONS, SUBSCRIBE, and other requests must also be allowed to fork in
order to be useful.

Anders

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Billy Biggs wrote:
> 
> Fairlie-Cuninghame, Robert (rfairlie@nuera.com):
> 
> > Was there any consensus reached on Record-Route changes? It would be
> > nice to get this one nailed down.
> 
>   In my opinion, the consensus is that going through all the addresses
> in the Record-Route and changing them for the reverse direction is out.
> Proxies should insert an address that indicates to them enough state to
> perform their function, and the spec should probably give some examples.
> 
>   The only open issue is whether to allow non-SIP URIs in the Route.  In
> order to support them, the suggestion is to define Record-Route
> parameters to indicate the IP address, port, and transport.  I think
> this is dangerous because it won't work with outbound proxies.
> 

Hi,

I don't I understand why this will not work with outbound
proxies. If you have a firewall that can handle SIP that
firewall will need to work as a SIP proxy and thereby add
itself to the ROUTE chain. That 'proxy' also needs to be
addressable from outside the firewall how will you otherwise 
be able to call someone behind the firewall ?

/Bertil 

-- 
Bertil Engelholm
AXE Research and Development        voice : +46 8 727 3499
SIP Security                        Fax   : +46 8 647 8276
S-126 25 Stockholm Sweden           E-mail:
Bertil.Engelholm@uab.ericsson.se

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Anders Kristensen wrote:
> > 
> > I like forking -- it's very powerful; however, I might be
> > inclined to suggest that we warn in bis that proxies
> > shouldn't be too keen to fork on non-INVITE requests, since
> > in a number of cases, it's probably not going to be so
> > helpful.
> 
> Hmmm. If it's OK to register multiple SIP addresses at a registrar and
> have a proxy fork INVITEs to those addresses it pretty much follows that
> OPTIONS, SUBSCRIBE, and other requests must also be allowed to fork in
> order to be useful.

I think it's OPTIONS that bothers me the most.  You try to gather
some information on the UA you're about to call with an OPTIONS,
and it forks.  The result of the OPTIONS could be completely
useless, because it may well not be from the UA that you would
actually have ended up talking to if you'd done an INVITE instead.

I don't know -- maybe it's a silly point, but forking just seems to
slightly spoil OPTIONS.

Okay, so maybe Billy's suggestion was better: suggest sequential
forking as nominal behaviour for non-INVITE.  This doesn't help
at all with the OPTIONs case, but it's probably better for
things like BYE, for instance.


 - Jo.


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Subject: [SIP] Open Source?
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Date: Mon, 20 Nov 2000 10:01:29 GMT

Dear All,

Are there any open source SIP stacks for the Windows OS?

Regards,

Lewis.
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: SIP Telephony Service Examples
	Author(s)	: A. Johnston, R. Sparks, C. Cunningham, 
                          S. Donovan, K. Summers
	Filename	: draft-ietf-sip-service-examples-00.txt
	Pages		: 90
	Date		: 17-Nov-00
	
This document gives examples of SIP (Session Initiation Protocol)
telephony services.  This covers most features offered in so-called
Centrex offerings from local exchange carriers and PBX (Private
Branch Exchange) features.  Most of the services shown in this
document are implemented in the SIP User Agents, although some
require the assistance of a SIP Proxy.  Some require some extensions
to SIP including third party call control (3pcc) extensions such as
the REFER method. These features are not intented to be an exhaustive
set, but rather show implementations of common features likely to be
implemented on SIP IP Telephones in a business environment.

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From sip-admin@lists.bell-labs.com  Mon Nov 20 09:24:04 2000
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URLs, SIP or otherwise, cannot have unescaped spaces. Never have, never
will.

Anders Kristensen wrote:
> 
> airoy@hss.hns.com wrote:
> >
> > you are pretty much on track....some additions are
> >      between the user and the "@"
> >      between the "@" and the host
> >      between host and " : "
> >      between ":" and port
> >      between port and ";"
> >      between ";" and url
> >      and so on...you have left out separators. You can have spaces or line
> > folds between any 2 tokens , separators or otherwise.
> 
> Why do you think spaces here are legal? As far as I can tell rfc 2396,
> "Uniform Resource Identifiers (URI): Generic Syntax" seems to clearly
> spaces in URIs altogether. The comments on "implied *LWS" in appendix B
> of rfc 2543 doesn't change this.
> 
> Anders
> 

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Date: Mon, 20 Nov 2000 15:14:34 +0000



Henning Schulzrinne suggested that I submit the following questions about URL
comparison to the SIP list.

On page 21 of rfc2543bis02 , it says

(call this clause A)
"user, password, host, port and any url-parameter parameters of the URI must
match. If a component is omitted, it matches based on its default value."

So far, this is clear, but the paragraph goes on to say that

(call this clause B)
"Components not found in both URLs being compared are ignored"

Now consider these two URLs:

     sip:juser@example.com:6666
     sip:juser@example.com

Clause A means that these URLs compare as unequal, but clause B means that they
compare as equal.

Should clause B actually start with the word "other"?

(clause B')
"Other components not found in both URLs being compared are ignored"

(If so, is this actually the correct judgement? I would have thought that a URL
should not compare equal to a URL with an extra component. Why is it useful for
them to compare equal?)

And now another separate point: the text on page 21 goes on to say:

"Note that the two URLs example.com and example.com:5060, while considered
equal, may not lead to the same server, as the former causes a DNS SRV lookup,
while the latter only uses the A record".

However, this is not true --- on page 12 it says:

"3. The Request-URI is examined. If it contains an explicit port number other
than 5060, the next two steps are skipped".

The A record is only used if there is a port number other than 5060 ---- so
example.com and example.com:5060 both use the SRV record. (Of course this may
result in different machines actually being contacted, but that's a different
matter entirely, not to do with A records).

  Stuart Wray



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From sip-admin@lists.bell-labs.com  Mon Nov 20 10:18:04 2000
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Date: Mon, 20 Nov 2000 09:17:46 -0600

Bertil Engelholm (Bertil.Engelholm@uab.ericsson.se):

> Billy Biggs wrote:
> > 
> >   The only open issue is whether to allow non-SIP URIs in the Route.
> > In order to support them, the suggestion is to define Record-Route
> > parameters to indicate the IP address, port, and transport.  I
> > think this is dangerous because it won't work with outbound
> > proxies.
> 
> I don't I understand why this will not work with outbound proxies. If
> you have a firewall that can handle SIP that firewall will need to
> work as a SIP proxy and thereby add itself to the ROUTE chain. That
> 'proxy' also needs to be addressable from outside the firewall how
> will you otherwise be able to call someone behind the firewall ?

  Proxies don't need to add themselves to the Route.  So, say I buy a
SIP phone and I set its outbound proxy to be some SIP proxy on the
Internet which doesn't add itself to the route, and then some bozo puts
as the first Record-Route:

  R-R: <tel:54321>;sendto=12.43.52.23:5060;transport=udp

  What can I do?  The phone has to send the request to its outbound
proxy.  You get the same problem if you configure a SIP proxy to forward
all requests it gets to another proxy: you can't convey these parameters
in the request-URI.

  This shows why its too brittle to allow non-SIP URIs in the Route,
unless you force every proxy to know how to route based on tel: or http:
URIs... ;-)

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Mon, 20 Nov 2000 09:23:01 -0600

Jo Hornsby (jhornsby@ubiquity.net):

> >   The only open issue is whether to allow non-SIP URIs in the Route.
> > In order to support them, the suggestion is to define Record-Route
> > parameters to indicate the IP address, port, and transport.  I
> > think this is dangerous because it won't work with outbound
> > proxies.
> 
> I'd like to take this oppurtunity &:) to point out that the solution I
> proposed -- just inserting the SIP URL of the proxy, and embedding the
> Request-URI as a parameter thereof -- handles non-SIP URIs.

  Right, but this is a special case of not allowing SIP URIs.  If you're
a proxy that needs (for its state information or whatever) to know the
orignal request-URI it saw, then of course, throw it as a parameter.
But many (most? if not all?) proxies won't need this state to perform
their job.  Please see my post on this:

  http://lists.bell-labs.com/pipermail/sip/2000q4/004005.html

> I did go a step further, and suggest that called UAs did extra
> mangling in the reverse direction, [...]

  In my opinion, this increases the complexity for little gain.

  Basically, you're just proposing that we don't allow non-SIP URIs in
the Route, and I think this is completely reasonable.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Mon, 20 Nov 2000 09:25:42 -0600

Jo Hornsby (jhornsby@ubiquity.net):

> Okay, so maybe Billy's suggestion was better: suggest sequential
> forking as nominal behaviour for non-INVITE.  This doesn't help at all
> with the OPTIONs case, but it's probably better for things like BYE,
> for instance.

  That's funny, because it seems completely reasonable to me to parallel
fork a BYE.

  You would get a BYE at a proxy when the stupid UA at the other end
didn't put in a Contact.  You want to shut down the call quickly, so you
parallel fork to all registrations knowing that all but one will return
a 481.

  So much for my suggestion.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Billy Biggs wrote:
>
> Jo Hornsby (jhornsby@ubiquity.net):
> 
> > >   The only open issue is whether to allow non-SIP URIs in the Route.
> > > In order to support them, the suggestion is to define Record-Route
> > > parameters to indicate the IP address, port, and transport.  I
> > > think this is dangerous because it won't work with outbound
> > > proxies.
> > 
> > I'd like to take this oppurtunity &:) to point out that the solution I
> > proposed -- just inserting the SIP URL of the proxy, and embedding the
> > Request-URI as a parameter thereof -- handles non-SIP URIs.
> 
>   Right, but this is a special case of not allowing SIP URIs.  If you're
> a proxy that needs (for its state information or whatever) to know the
> orignal request-URI it saw, then of course, throw it as a parameter.
> But many (most? if not all?) proxies won't need this state to perform
> their job.  Please see my post on this:
> 
>   http://lists.bell-labs.com/pipermail/sip/2000q4/004005.html

Right.  So we're saying remove the bit where the bis specifies
that the Request-URI is copied into the Record-Route, and
instead note that the proxy is free to put any SIP URL in (of
course, it should be routable); and also remove the bit where
the bis specifies the Route Request-URI mangling by the called
UA.

> > I did go a step further, and suggest that called UAs did extra
> > mangling in the reverse direction, [...]
> 
>   In my opinion, this increases the complexity for little gain.

I certainly wouldn't argue this point.  It's just an option, if
someone feels they really need it.

>   Basically, you're just proposing that we don't allow non-SIP URIs in
> the Route, and I think this is completely reasonable.

Agreed.  After all, we already say that a proxy is not allowed
to mandate TCP (via the transport parameter) in Record-Route;
allowing non-SIP URIs, I would say, is orders of magnitude more
painful.

Cheers,


 - Jo.


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Date: Mon, 20 Nov 2000 09:55:59 -0600

Jo Hornsby (jhornsby@ubiquity.net):

> Right.  So we're saying remove the bit where the bis specifies that
> the Request-URI is copied into the Record-Route, and instead note that
> the proxy is free to put any SIP URL in (of course, it should be
> routable); and also remove the bit where the bis specifies the Route
> Request-URI mangling by the called UA.

  Totally.  It should put in some SIP URI that is routable in both
directions.

> >   Basically, you're just proposing that we don't allow non-SIP URIs
> > in the Route, and I think this is completely reasonable.
> 
> Agreed.  After all, we already say that a proxy is not allowed to
> mandate TCP (via the transport parameter) in Record-Route; allowing
> non-SIP URIs, I would say, is orders of magnitude more painful.

  Dude, you rock my world.

  Now, where's Sean Olson's argument? :)

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Mon, 20 Nov 2000 12:32:34 -0500

Absolutely correct. In fact, the only real difference between the chap based
authentication and digest should be the formatting of the response value.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Igor Slepchin [mailto:ISlepchin@dynamicsoft.com]
> Sent: Wednesday, November 15, 2000 8:58 PM
> To: sip@lists.bell-labs.com
> Subject: [SIP] draft-byerly-sip-radius-00.txt
> 
> 
> Minor comment on draft-byerly-sip-radius-00.txt: the draft 
> seems to deviate
> unnecessarily from the challenge and credentials syntax specified in
> RFC2617, i.e., it uses semicolon to separate challenge/credentials
> parameters instead of reusing comma-separation mandated by HTTP
> Basic/Digest. I think consistency with HTTP should be 
> maintained wherever
> possible.
> 
> Thank you,
> Igor Slepchin
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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Date: Mon, 20 Nov 2000 11:40:36 -0600

Hi
Is there any 'floor limitation' on the registration interval? I mean, let's say the registrar asks the client to register every 20 minutes.
Can the client decide to register every 1 minute? If YES, do we have any way to prevent malicious or badly configured client from overloading the network?
Thanks
Uri



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Date: Mon, 20 Nov 2000 09:38:33 -0800

I am open to this option.  Realize however, that REFER summarizes final
responses so summarizing provisionals is consistent with behavior that
exists today; and that simply forwarding other provisionals overloads their
semantics.

thanks,
-rohan

At 07:54 PM 11/18/00 , Venkatesh Venkataramanan wrote:
>Wouldn't simply forwarding the provisionals received allow the initiator of
>REFER know more about 'call progress' instead of translating them to a
>single 189 ? Are there any specific reasons against not doing so? Please
>elaborate.
>
>Venkatesh
>----- Original Message -----
>From: "Rohan Mahy" <rohan@cisco.com>
>To: "Sean Olson" <sean.olson@ericsson.com>
>Cc: <sip@lists.bell-labs.com>
>Sent: Friday, November 17, 2000 11:58 PM
>Subject: Re: [SIP] ID extension of REFER
>
>
>> At 02:02 AM 11/17/00 , Sean Olson wrote:
>> >Hello,
>> >
>> >I am a little confused about the purpose of this I-D. The first question
>> >that springs to mind is
>> >why you don't use "183" for this purpose?
>>
>> the semantics of 183 already means something different, and i did not
>think
>> I could overload those semantics cleanly.
>>
>> >The second question is whether you
>> >intend the 189 to lead to different timing than any other provisional
>> >response?
>>
>> A sends REFER to B
>> B sends INVITE to C
>>
>> A would get 189s any time B got provisionals from C.
>>
>> If C just 200 OKs  B's INVITE, no 189s are sent.
>> If C sends 180 or 183 and answers right away, A still gets a 189.  If you
>> don't need the information you can ignore it.
>>
>> thanks,
>> -rohan
>>
>> >(In other words, if the 189 does NOT cause different timing, than the
>> >issue of a long
>> >  period (> 37 seconds) between the INVITE and 200 OK is not really
>> >solved. I realize
>> >  this should be a very rare case.)
>> >
>> >/sean
>> >
>> >Rohan Mahy wrote:
>> >
>> >>  Hi,
>> >>
>> >> I just submitted an ID for an extension to the REFER method that I
>> >> discussed with Robert Sparks at Pittsburgh.  Until it appears in the
>> >> archive it is available at:
>> >>
>> >> ftp://ftpeng.cisco.com/rmahy/draft-mahy-sip-189-00.txt
>> >>
>> >>
>> >> A SIP Extension: Informational Responses to the REFER method
>> >>
>> >> Abstract
>> >>
>> >>      This document defines a SIP extension to the REFER method of the
>> >>      SIP Call Control Framework.  This extension provides for the
>> >>      ability to convey information about the progress of the Referred
>> >>      request when that request is in a provisional state for a long
>> >>      period of time (many seconds or minutes).
>> >>
>> >>
>> >> Comments welcome.
>> >>
>> >> thanks,
>> >> -rohan
>> >>
>> >>
>> >
>>
>>
>> _______________________________________________
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>> SIP@lists.bell-labs.com
>> http://lists.bell-labs.com/mailman/listinfo/sip
>>
>


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Date: Mon, 20 Nov 2000 12:53:02 -0500 (EST)

Aseem,

The new draft, draft-ietf-sip-manyfolks-resource-00, is an update
of previous draft-manyfolks-sip-resource-01.  Since its name changed
as a result of becomming a working group charter item, its version
number was reset to zero.

The previous contribution draft-ietf-mmusic-sdp-qos-00 expired
in December 1999; it has been superceeded by the current draft,
and its authors are some of the many folks.

Bill Marshall
wtm@research.att.com

-----original message-----
From: "Agarwal, Aseem" <aseem@trillium.com>
To: sip@lists.bell-labs.com
Subject: RE: [SIP] I-D ACTION:draft-ietf-sip-manyfolks-resource-00.txt
Date: Mon, 13 Nov 2000 10:26:18 -0800

Hi,
  Is this document different from draft-manyfolks-sip-resource-01.txt ??
If not, should n't this document have a higher version number ?

Also, does this draft obsolete "Establishing QOS and Security Preconditions
for SDP Sessions", draft-ietf-mmusic-sdp-qos-00.txt ??

thanks,
aseem


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Date: Mon, 20 Nov 2000 12:28:12 -0600

What does the first 'VIA' describe? (The one with the maddr)

Via: SIP/2.0/UDP 131.215.131.13;maddr=239.112.3.4;ttl=16;
Via: SIP/2.0/TCP 10.0.1.1;received=128.13.44.52

Uri

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Date: Mon, 20 Nov 2000 12:30:17 -0600


  I would like to thank both Jonathan and Henning for providing this
important first step in allowing us to discuss conferencing in SIP.
This helps both to clarify and focus future work.

  That said, I have some problems with the draft.  I will start with the
most critical.

> 7 Centralized Signaling, Distributed Media
> 
>    In this conferencing model, there is a centralized controller, as
>    in the dial-in and dial-out cases. However, the centralized server
>    handles signaling only. The media is still sent directly between
>    participants, using either multicast or multi-unicast.
>    Multi-unicast is when a user sends multiple packets (one for each
>    recipient, addressed to that recipient). This is referred to as a
>    "Decentralized Multipoint Conference" in H.323. Interestingly, this
>    conference model is possible baseline SIP.
> 
>    It works through third party call control [13]. The conference
>    server uses re-INVITEs to each participant when a new one joins.
>    The re- INVITEs add a media stream that gets sent to the new
>    participant (and similarly in the reverse direction).

  I don't think this works.

  Let's look at some example SDP:

     ...
     c=24.22.33.44
     m=audio 6646 RTP/AVP 0
     c=33.24.44.22
     m=audio 7348 RTP/AVP 1

  Wouldn't this mean that if you use codec 0 then send to 24.22.33.44,
and if you use codec 1 then send to 33.24.44.22?  What document implies
that I send using both codecs to each address?

  If you really want to do this, you're going to need a stronger
description protocol.  SDP just isn't good enough.

  UAs should not be expected to support this method as you describe.


> 5 Ad-hoc Centralized Conferences
> 
>    [...]
> 
>    It is also possible to transition from a end system mixed
>    conference (even one with a complex connection topology), to a
>    centralized conference server. One user takes responsibility for
>    initiating the transition. It proceeds as described above. However,
>    the REFER request is sent to all SIP peers adjacent to the user. In
>    addition, when a SIP UA receives a REFER, they must not only act on
>    it as described above, but also generate a REFER to any of their
>    adjacent SIP peers. In essence, the REFER message is propagated
>    along the connection graph, starting at the root (which is the user
>    who initiates the transition). The transition will work so long as
>    the graph has no cycles (which is needed anyway, as discussed
>    above), and so long as only one user attempts to initiate the
>    transition. If multiple users attempt to initiate the transition at
>    the same time, the conference will break into two disjoint ad-hoc
>    conferences, with membership depending on the temporal dynamics of
>    the REFER propagation.

  This implies that REFER propagation should be automatic, but no method
is given to signal this.

  It doesn't seem reasonable to expect UAs to propagate a REFER along
their conferences.  For example, say A is in a 3-way call performing
End System Mixing:

                       A
                 D    / \
                     B   C

  And say that B REFERs A to D.  Should A propagate this REFER to C?
No.  There is no way to know if D is a conference bridge or simply
another phone.

  I think you should take this paragraph out.  It is confusing and
unnecessary in 99% of cases.  If the user of A knows that D is a
conference bridge, A can initiate a REFER to C if it wants to.  This
action should not be automatic.


> 2.5 Discovering Participant Identities
> 
>    The identities of other participants in the conference is NOT known
>    through SIP. Rather, it is learned through RTP. UAs with degrees
>    greater than one are RTP mixers. As such, they take the RTCP SDES
>    of the streams they mix, and aggregrate them into the RTCP stream
>    sent out. Since RTCP messages are sent infrequently, there may be a
>    delay between when a user joins, and when their presence is known
>    to the other participants.

  You might want to mention in the security section that announcing
participants via RTCP is not always desirable.  For example, if the user
wants someone to listen in on the call without announcing their presence
to the remote agent.


> 9 Whats Missing - Full Mesh
> 
>    [...]
>
>    As a baseline model, we believe that each INVITE, 200 OK response,
>    and ACK simply contain a header called Members. This header is a
>    list of URLs, and for each URL, there is a parameter that indicates
>    whether they are in the conference right now, and when they joined,
>    or whether they were previously in the conference, and when they
>    left. A UA simply performs a re-INVITE as it receives new
>    information. A periodic re-INVITE (ala session timer [15] will also
>    be needed to heal partitions and deal with other conditions that
>    may arise).

  I don't think it is clean to try and put all that information into the
SIP header.  In my opinion, what is needed is first a generall call
description language, which can describe the state of the conference as
a whole.  This is useful both for managing a full mesh call, or watching
the status of a call through a conference bridge.

  See some old notes on conference control at:

  http://www.billybiggs.com/sip-conf/17-jun-00-conf-architecture-draft-1.txt

  Most of the document is useless, but I do talk about the need for a
description language which is seperate from the control of the
conference.  I will attempt to update these notes based on your draft.


  One last note about your draft:

> 2 End System Mixing
>
>    [...]
> 
>    Note that this model has the serious drawback that the conference
>    ends when the mixing UA leaves the call.

  You can mention our Replaces draft. :)

  http://www.sip-happens.com/replaces/

--
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Mon, 20 Nov 2000 13:53:32 -0500

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Section 6.46.1------------  2543bis.

A client that sends a request to a multicast address MUST add the 
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?maddr?
parameter to its Via header
field, and SHOULD add the ?ttl? parameter. (In that case, the maddr par=
ameter
SHOULD contain the desti-nation
multicast address, although under exceptional circumstances it MAY cont=
ain a
unicast address.) If a
server receives a request which contained an ?maddr? parameter in the t=
opmost
Via field, it SHOULD send
the response to the address listed in the ?maddr? parameter.


Sanjiv.





Baniel Uri-CUB001 <Uri.Baniel@motorola.com> on 11/20/2000 01:28:12 PM

To:   "'sip@lists.bell-labs.com'" <sip@lists.bell-labs.com>
cc:    (bcc: Sanjiv Deshpande/Telcordia)
Subject:  [SIP] VIA


=

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What does the first 'VIA' describe? (The one with the maddr)

Via: SIP/2.0/UDP 131.215.131.13;maddr=239.112.3.4;ttl=16;
Via: SIP/2.0/TCP 10.0.1.1;received=128.13.44.52

Uri

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Date: Mon, 20 Nov 2000 15:24:20 -0600

Hello

2543bis
section 1.4.3 SIP Transaction says:
..... If the request is sent via multicast UDP, the response is directed to the same multicast address and destination port.

My question is: assume a UAC sends an INVITE, which goes via P1 and P2. The P2 multicast it to the multicast group MG1, which P3 happens to be one of its members.
So to whom shall P3 return the response? is it to MG1? if that is the case, is P2 member of MG1? If yes why does the response go to the other group members ?

Thanks

Uri



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From: "Henning G. Schulzrinne" <hgs@cs.columbia.edu>
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Subject: Re: [SIP] New I-D on multiparty conferencing models
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Billy Biggs wrote:

> 
> > 7 Centralized Signaling, Distributed Media
> >

>   I don't think this works.
> 
>   Let's look at some example SDP:
> 
>      ...
>      c=24.22.33.44
>      m=audio 6646 RTP/AVP 0
>      c=33.24.44.22
>      m=audio 7348 RTP/AVP 1
> 
>   Wouldn't this mean that if you use codec 0 then send to 24.22.33.44,
> and if you use codec 1 then send to 33.24.44.22?  What document implies
> that I send using both codecs to each address?

SDP does. SDP does *not* say that you choose the media line you like.
All media lines are sent simultaneously; unfortunately, this is implied
(set of media) rather than stated. In practice, I doubt that any
existing application will do this, but it has been done, e.g., for two
simultaneous languages. The lack of distinction between 'and' and 'or'
is a major flaw in SDP.

> 
>   If you really want to do this, you're going to need a stronger
> description protocol.  SDP just isn't good enough.
> 
>   UAs should not be expected to support this method as you describe.

Indeed, I doubt that UAs will do this. They should support multiple
media lines with the same type, however, as this is useful beyond this
mixing model.


-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Date: Mon, 20 Nov 2000 17:32:38 -0600
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Hi:

    There are some inconsistencies for To Tag in the call flows.

    In 2.3 Consultation Hold:

   The To Tag "tag=314159"  in  F4 180 Ringing B -> Proxy 1
    is copied to the From header in the F10 Invite  request.

    But in 2.4 Unattended Transfer:

    The To Tag " tag=314159"  in F3 ACK B -> A
    is copied to the To header in the F4 Invite request.

    According to RFC2543bis, 2.3 Consultation Hold is correct.
    Please confirm.

    Thanks,

    Henry

      2.3 Consultation Hold
      ...

   /* User B places User A on hold. */

   F10 INVITE B -> Proxy 1

   INVITE sip:UserA@here.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   Route: <sip:UserA@here.com>
   From: LittleGuy <sip:UserB@there.com>;tag=314159
   To: BigGuy <sip:UserA@here.com>
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Contact: LittleGuy <sip:UserB@there.com>
   Content-Type: application/sdp
   Content-Length: ...

    The To Tag "tag=314159"  in  F4 180 Ringing B -> Proxy 1
    is copied to From header in the F10 Invite  request.

    2.4 Unattended Transfer

     /* Session is established between A and B.  User B then places User A
   on hold */

   F4 INVITE A -> B

   INVITE sip:UserB@there.com SIP/2.0
   Via: SIP/2.0/UDP there.com:5060
   From: BigGuy <sip:UserA@here.com>
   To: LittleGuy <sip:UserB@there.com>;tag=314159
   Call-ID: 12345601@here.com
   CSeq: 1 INVITE
   Contact: BigGuy <sip:UserA@here.com>
   Content-Type: application/sdp
   Content-Length: ...


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From: Billy Biggs <Billy_Biggs@3com.com>
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Date: Mon, 20 Nov 2000 17:56:08 -0600

Henning G. Schulzrinne (hgs@cs.columbia.edu):

> > > 7 Centralized Signaling, Distributed Media
> > >
> 
> >   I don't think this works.
> > 
> >   Let's look at some example SDP:
> > 
> >      ...
> >      c=24.22.33.44
> >      m=audio 6646 RTP/AVP 0
> >      c=33.24.44.22
> >      m=audio 7348 RTP/AVP 1
> > 
> >   Wouldn't this mean that if you use codec 0 then send to
> > 24.22.33.44, and if you use codec 1 then send to 33.24.44.22?
> > What document implies that I send using both codecs to each
> > address?
> 
> SDP does. SDP does *not* say that you choose the media line you like.
> All media lines are sent simultaneously; unfortunately, this is
> implied (set of media) rather than stated. In practice, I doubt that
> any existing application will do this, but it has been done, e.g., for
> two simultaneous languages. The lack of distinction between 'and' and
> 'or' is a major flaw in SDP.

  Ok, so if I see:
         m=audio 6646 RTP/AVP 0 1 2 3

  Then I get to choose, but if I see:
         m=audio 6646 RTP/AVP 0
         m=audio 6656 RTP/AVP 1

  Then I don't get to choose?  I'm supposed to send to both?

> >   If you really want to do this, you're going to need a stronger
> > description protocol.  SDP just isn't good enough.
> > 
> >   UAs should not be expected to support this method as you describe.
> 
> Indeed, I doubt that UAs will do this. They should support multiple
> media lines with the same type, however, as this is useful beyond this
> mixing model.

  Sorry, I don't follow.  If I see:
         c=22.22.22.22
         m=audio 6646 RTP/AVP 0
         c=33.33.33.33
         m=audio 6656 RTP/AVP 0

  Then it's asking me to send to both of them?

  I don't think that anyone will support this either.  I would say that
it is too late for SDP, and if you want to signal this, lets define a
stronger description protocol.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Billy Biggs wrote:
> 
>
>   Ok, so if I see:
>          m=audio 6646 RTP/AVP 0 1 2 3
> 
>   Then I get to choose, but if I see:
>          m=audio 6646 RTP/AVP 0
>          m=audio 6656 RTP/AVP 1
> 
>   Then I don't get to choose?  I'm supposed to send to both?

Yes. Each m line is independent and additive.


>   Sorry, I don't follow.  If I see:
>          c=22.22.22.22
>          m=audio 6646 RTP/AVP 0
>          c=33.33.33.33
>          m=audio 6656 RTP/AVP 0
> 
>   Then it's asking me to send to both of them?

Yes. I believe that sdr actually allows this, but haven't tested this.

> 
>   I don't think that anyone will support this either.  I would say that
> it is too late for SDP, and if you want to signal this, lets define a
> stronger description protocol.

That's a separate and necessary step.

> 
> --
> Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Henning G. Schulzrinne (hgs@cs.columbia.edu):

> >   [...] but if I see:
> >          m=audio 6646 RTP/AVP 0
> >          m=audio 6656 RTP/AVP 1
> >   Then I don't get to choose?  I'm supposed to send to both?
> 
> Yes. Each m line is independent and additive.

  This means that I could be asked to commit a huge denial of service
attack.  There are security implications here.

> >   Sorry, I don't follow.  If I see:
> >          c=22.22.22.22
> >          m=audio 6646 RTP/AVP 0
> >          c=33.33.33.33
> >          m=audio 6656 RTP/AVP 0
> > 
> >   Then it's asking me to send to both of them?
> 
> Yes. I believe that sdr actually allows this, but haven't tested this.

  Again there are security implications.  My client should be free to
reject this SDP as if the caller didn't specify a supported codec.
IIRC, this might be a 406 response and a shutdown of the call.

  The assertion in the draft that "Interestingly, this conference model
is possible baseline SIP" is misleading.  It implies that all SIP phones
are expected to support it.  I guess detecting the situation and
shutting down the call is acceptable?

> >   I don't think that anyone will support this either.  I would say
> > that it is too late for SDP, and if you want to signal this, lets
> > define a stronger description protocol.
> 
> That's a separate and necessary step.

  Well, then maybe the problem is that SDP is too powerful for use as a
base description protocol for SIP.  If supporting SDP means supporting:

  - Sending audio to multiple locations, and
  - Mixing audio from multiple sources

  ... then I think that will exceed the capacity of many embedded
phones.  It also might simply be too difficult for others to code.
Regardless, services which make use of these features must be aware that
they will not be able to interoperate with everybody.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Frankly, I don't see the difference in allowing media to be sent to one
address or two. There never was a guarantee that the media address is
the same as the signaling address (3pcc works based on that principle).
If you want denial of service, requesting a single video stream at high
rate is a far more effective mechanism than some piddling audio
streams... You don't need special SDP or end system capabilities for
that "feature".
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Date: Mon, 20 Nov 2000 19:14:06 -0600

Henning G. Schulzrinne (hgs@cs.columbia.edu):

> Frankly, I don't see the difference in allowing media to be sent to
> one address or two. There never was a guarantee that the media address
> is the same as the signaling address (3pcc works based on that
> principle).  If you want denial of service, requesting a single video
> stream at high rate is a far more effective mechanism than some
> piddling audio streams... You don't need special SDP or end system
> capabilities for that "feature".

  True, but you do need video.

  Regardless, lets just talk about the assumptions we're making about
the capability of the clients.

  I don't expect all clients to be able to support:

  - sending audio to multiple locations, and
  - mixing audio from multiple sources.

  I don't think your draft should either.  Is that unreasonable?

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
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Date: Mon, 20 Nov 2000 21:04:38 -0500


Just to make sure I'm clear, is this true:
Both SIP-URL and URI (as in the contact, to, from headers) can not contain
spaces.  



		Henning wrote:
		URLs, SIP or otherwise, cannot have unescaped spaces. Never
have, never
		will.

		Anders Kristensen wrote:
		> 
		> airoy@hss.hns.com wrote:
		> >
		> > you are pretty much on track....some additions are
		> >      between the user and the "@"
		> >      between the "@" and the host
		> >      between host and " : "
		> >      between ":" and port
		> >      between port and ";"
		> >      between ";" and url
		> >      and so on...you have left out separators. You can
have spaces or line
		> > folds between any 2 tokens , separators or otherwise.
		> 
		> Why do you think spaces here are legal? As far as I can
tell rfc 2396,
		> "Uniform Resource Identifiers (URI): Generic Syntax" seems
to clearly
		> spaces in URIs altogether. The comments on "implied *LWS"
in appendix B
		> of rfc 2543 doesn't change this.
		> 
		> Anders
		> 

		-- 
		Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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"Snyder, Duke" wrote:
> 
> Just to make sure I'm clear, is this true:
> Both SIP-URL and URI (as in the contact, to, from headers) can not contain
> spaces.
> 
>                 Henning wrote:
>                 URLs, SIP or otherwise, cannot have unescaped spaces. Never
> have, never
>                 will.

I'm sorry if my note was unclear, but the answer is that these URLs MUST
NOT contain unescaped spaces (%20 is obviously ok where it makes sense).

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Date: Mon, 20 Nov 2000 20:14:12 -0600

Henry Chen (hjlechen@cisco.com):

>  There are some inconsistencies for To Tag in the call flows.

  The example that Henry points out is of course a bug.

  Without having read the draft in depth, I would also like to request
to the authors that From tags be inserted everywhere.  From tags (on
initial requests) are important for call-leg matching in the presence of
forking proxies, which pretty much means everywhere.  I don't want to
encourage people not to use them.

  As well, it would be nice to see random tags per call.  You have UserB
always using the same tag.  As I've mentioned before on this list, this
is dangerous and a bad idea.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Billy Biggs wrote:

> 
>   As well, it would be nice to see random tags per call.  You have UserB
> always using the same tag.  As I've mentioned before on this list, this
> is dangerous and a bad idea.

Note that not everybody agrees with this. This is a trade-off between
features. Different people come to different conclusions as to what the
right trade-off is.

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Henning G. Schulzrinne (hgs@cs.columbia.edu):

> >   As well, it would be nice to see random tags per call.  You have
> > UserB always using the same tag.  As I've mentioned before on this
> > list, this is dangerous and a bad idea.
> 
> Note that not everybody agrees with this. This is a trade-off between
> features. Different people come to different conclusions as to what
> the right trade-off is.

  Sorry, I didn't realize this was a contended issue.

  Right now the text reads, "[...] but it is RECOMMENDED to maintain the
same tag across calls and instances of the UA to allow restarting of a
user agent."

  If that is to be kept, then please also mention that UAs which exhibit
this behavior be able to tell when a request loops back to themselves.
This can happen in many cases, and I've seen UAs crash because they
can't perform call-leg matching.

  Some cases:

  1. User makes a call through a proxy which forks back to themselves (I
     called Joe but Joe has his calls forwarded to me).

  2. User sends a SUBSCRIBE for their voicemail to sip:me@myproxy with
     an Accept-Contact: *;feature="voicemail", but the user doesn't have
     a voicemail server so the SUBSCRIBE comes back to the phone.

  3. User calls themselves through their own proxy to make sure their
     registration is active.

  I mean, sure, UAs can have the logic inside them to detect this case,
but in bis the suggestion is to always use the same tag without mention
of them.  You should also mention that soft-phones should avoid choosing
a tag based on the MAC address.

  I think the how-to-create-a-crash-proof-SIP-UA stuff should be in a
seperate document, not in the SIP RFC.  It's more complicated than just
choosing the same tag.  You have to realize that Routes won't be kept,
hope the far end sends a re-INVITE, etc...  IMHO, recommending that UAs
choose the same tag is asking for trouble.  Does noone else agree?

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Reply-To: "Venkatesh Venkataramanan" <venkatesh.venkataramanan@wipro.com>
From: "Venkatesh Venkataramanan" <venkatesh.venkataramanan@wipro.com>
To: "Rohan Mahy" <rohan@cisco.com>, "Sean Olson" <sean.olson@ericsson.com>
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References: <4.1.20001116151548.00cc2850@imop.cisco.com> <4.1.20001117102008.00a85c50@imop.cisco.com> <4.1.20001120093503.00cdc5c0@imop.cisco.com>
Subject: Re: [SIP] ID extension of REFER
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The main reason as to why I am for forwarding provisional responses is as
follows:

This allows the "Transferror" to decide when he/she wants to hang up, very
useful for certain applications, as I have mentioned in one of my previous
posts, which I repeat here for completeness...

Most transfers occur through a voice messaging system/IVR. Many of them
support a couple of nice features where in on a transfer request, they want
to know if the far end is busy, or if the far end is following a coverage
path(on no answer),  in which case they would want to abandon the transfer
request, and then voice a message saying, "The transferred extension is
busy/not available... To leave a message for <original_called_party> please
do so at the tone, to request another transfer press..." and so on.

On the other hand, there are other simpler applications would just want to
complete the transfer when they find the destination "ringing" and let the
caller follow the coverage path of the TT in case of no answer(this is just
to make sure that if the user mis-dialed and the voice messaging system did
a blind transfer, the user will not be stuck listening to an overflow tone
or would have to redial the entire number to reach another user).

I saw lack of ability to cleanly support such applications with the original
REFER based transfer solution. Your draft addresses this partly by sending a
provisional response of some sort, and builds an excellent mechanism for
supporting blind transfers(by dropping the necessity for the transferring UA
to wait until a final response for REFER is received), but I think allowing
provisional 1xx responses back to the transferring party would help
continuing support for such existing applications.

BTW, I thought  the only consolidation that REFER provided as regards final
response is that it consolidates a 3xx class response alone (Please correct
me if I am wrong, but as far as I know, any 400-600 class responses MAY be
issued in response to a REFER request. I don't know the rationale as to why
only 1xx and 3xx level responses were masked).

Venkatesh

----- Original Message -----
From: "Rohan Mahy" <rohan@cisco.com>
To: "Venkatesh Venkataramanan" <venkatesh.venkataramanan@wipro.com>; "Sean
Olson" <sean.olson@ericsson.com>
Cc: <sip@lists.bell-labs.com>
Sent: Monday, November 20, 2000 11:08 PM
Subject: Re: [SIP] ID extension of REFER


> I am open to this option.  Realize however, that REFER summarizes final
> responses so summarizing provisionals is consistent with behavior that
> exists today; and that simply forwarding other provisionals overloads
their
> semantics.
>
> thanks,
> -rohan
>
> At 07:54 PM 11/18/00 , Venkatesh Venkataramanan wrote:
> >Wouldn't simply forwarding the provisionals received allow the initiator
of
> >REFER know more about 'call progress' instead of translating them to a
> >single 189 ? Are there any specific reasons against not doing so? Please
> >elaborate.
> >
> >Venkatesh
> >----- Original Message -----
> >From: "Rohan Mahy" <rohan@cisco.com>
> >To: "Sean Olson" <sean.olson@ericsson.com>
> >Cc: <sip@lists.bell-labs.com>
> >Sent: Friday, November 17, 2000 11:58 PM
> >Subject: Re: [SIP] ID extension of REFER
> >
> >
> >> At 02:02 AM 11/17/00 , Sean Olson wrote:
> >> >Hello,
> >> >
> >> >I am a little confused about the purpose of this I-D. The first
question
> >> >that springs to mind is
> >> >why you don't use "183" for this purpose?
> >>
> >> the semantics of 183 already means something different, and i did not
> >think
> >> I could overload those semantics cleanly.
> >>
> >> >The second question is whether you
> >> >intend the 189 to lead to different timing than any other provisional
> >> >response?
> >>
> >> A sends REFER to B
> >> B sends INVITE to C
> >>
> >> A would get 189s any time B got provisionals from C.
> >>
> >> If C just 200 OKs  B's INVITE, no 189s are sent.
> >> If C sends 180 or 183 and answers right away, A still gets a 189.  If
you
> >> don't need the information you can ignore it.
> >>
> >> thanks,
> >> -rohan
> >>
> >> >(In other words, if the 189 does NOT cause different timing, than the
> >> >issue of a long
> >> >  period (> 37 seconds) between the INVITE and 200 OK is not really
> >> >solved. I realize
> >> >  this should be a very rare case.)
> >> >
> >> >/sean
> >> >
> >> >Rohan Mahy wrote:
> >> >
> >> >>  Hi,
> >> >>
> >> >> I just submitted an ID for an extension to the REFER method that I
> >> >> discussed with Robert Sparks at Pittsburgh.  Until it appears in the
> >> >> archive it is available at:
> >> >>
> >> >> ftp://ftpeng.cisco.com/rmahy/draft-mahy-sip-189-00.txt
> >> >>
> >> >>
> >> >> A SIP Extension: Informational Responses to the REFER method
> >> >>
> >> >> Abstract
> >> >>
> >> >>      This document defines a SIP extension to the REFER method of
the
> >> >>      SIP Call Control Framework.  This extension provides for the
> >> >>      ability to convey information about the progress of the
Referred
> >> >>      request when that request is in a provisional state for a long
> >> >>      period of time (many seconds or minutes).
> >> >>
> >> >>
> >> >> Comments welcome.
> >> >>
> >> >> thanks,
> >> >> -rohan
> >> >>
> >> >>
> >> >
> >>
> >>
> >> _______________________________________________
> >> SIP mailing list
> >> SIP@lists.bell-labs.com
> >> http://lists.bell-labs.com/mailman/listinfo/sip
> >>
> >
>
>
> _______________________________________________
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> SIP@lists.bell-labs.com
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>


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Subject: Re: [SIP] Record-Route consensus
From: David Shrader <dshrader@master-consultant.com>
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Where do we stand on which server removes the Route header? If I recall, the
current spec says that the preceding server remotes the route header for the
next destination. OTOH, I also saw posts suggesting that the server remove
its own Route header (like the Via header).

David

----------------------------
David Shrader
Master Consultant, Inc.
dshrader@master-consultant.com
http://www.EyeForTheFuture.com

> From: Billy Biggs <Billy_Biggs@3com.com>
> Date: Mon, 20 Nov 2000 09:55:59 -0600
> To: Jo Hornsby <jhornsby@ubiquity.net>
> Cc: Sean Olson <sean.olson@ericsson.com>, "Fairlie-Cuninghame, Robert"
> <rfairlie@nuera.com>, SIP List <sip@lists.bell-labs.com>
> Subject: Re: [SIP] Record-Route consensus
> 
> Jo Hornsby (jhornsby@ubiquity.net):
> 
>> Right.  So we're saying remove the bit where the bis specifies that
>> the Request-URI is copied into the Record-Route, and instead note that
>> the proxy is free to put any SIP URL in (of course, it should be
>> routable); and also remove the bit where the bis specifies the Route
>> Request-URI mangling by the called UA.
> 
> Totally.  It should put in some SIP URI that is routable in both
> directions.
> 
>>> Basically, you're just proposing that we don't allow non-SIP URIs
>>> in the Route, and I think this is completely reasonable.
>> 
>> Agreed.  After all, we already say that a proxy is not allowed to
>> mandate TCP (via the transport parameter) in Record-Route; allowing
>> non-SIP URIs, I would say, is orders of magnitude more painful.
> 
> Dude, you rock my world.
> 
> Now, where's Sean Olson's argument? :)
> 
> -- 
> Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 


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David Shrader (dshrader@master-consultant.com):

> Where do we stand on which server removes the Route header? If I
> recall, the current spec says that the preceding server remotes the
> route header for the next destination. OTOH, I also saw posts
> suggesting that the server remove its own Route header (like the Via
> header).

  I don't recall that thread, but I can't see how it would help
anything, and it would definitely break all existing implementations.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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We are looking to set up DNS SRV records for SIP servers at the 
upcoming bakeoff. Could anyone who has experience of doing this
on a Win 2000 Server please get in touch to liaise with the
hosts.

Many thanks,
Neil.
(on behalf of the SIP Bake-off Technical Program Committee)
-- 
Ubiquity Software Corporation, UK        http://www.ubiquity.net

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> -----Original Message-----
> From: Billy Biggs [mailto:Billy_Biggs@3com.com]
> Sent: Monday, November 20, 2000 10:26 AM
> To: Jo Hornsby
> Cc: SIP List
> Subject: Re: [SIP] a stupid question on 3way handshake
> 
> 
> Jo Hornsby (jhornsby@ubiquity.net):
> 
> > Okay, so maybe Billy's suggestion was better: suggest sequential
> > forking as nominal behaviour for non-INVITE.  This doesn't 
> help at all
> > with the OPTIONs case, but it's probably better for things like BYE,
> > for instance.
> 
>   That's funny, because it seems completely reasonable to me 
> to parallel
> fork a BYE.
> 
>   You would get a BYE at a proxy when the stupid UA at the other end
> didn't put in a Contact.  You want to shut down the call 
> quickly, so you
> parallel fork to all registrations knowing that all but one 
> will return
> a 481.

Excellent point. Good case for forking a non-INVITE in parallel.

Jo writes:
> I think it's OPTIONS that bothers me the most.  You try to gather
> some information on the UA you're about to call with an OPTIONS,
> and it forks.  The result of the OPTIONS could be completely
> useless, because it may well not be from the UA that you would
> actually have ended up talking to if you'd done an INVITE instead.
> 
> I don't know -- maybe it's a silly point, but forking just seems to
> slightly spoil OPTIONS.

Using OPTIONS to gather information on the UA you're about to call (beyond
that there is a UA that you can talk to) has other problems. Service logic
in a proxy may anyway dictate that the OPTIONS goes one way, and then a
followup INVITE goes a different way (some kind of randomized load
balancing). Furthermore, what would you realistically do with all the
OPTIONS responses even if you could get them all?

If you want to do something specific, like find out if there is a UA that
supports video, you are far better off sending an OPTIONS with a caller
preferences header. This way, if you get back any sucess responses, you know
there is a UA that meets your requirements.

Jo also writes:
> Jonathan Rosenberg wrote:
> >
> > Well, its not evil, but its not correct. Proxies don't initiate 
> > requests on their own, excepting the "special" ones of BYE
> 
> Do you mean CANCEL?

Yes, sorry.

-Jonathan R.


---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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Please, lets use the sip list for technical discussions. I've removed the
sip forum list from the to line.

To answer your question:

 

> -----Original Message-----
> From: "ferreiro_j"@tsm.es [mailto:"ferreiro_j"@tsm.es]
> Sent: Monday, November 20, 2000 10:59 AM
> To: simon@firetalk.com; farhan
> Cc: discussion@sipforum.org
> Subject: Re: [SIPFORUM] Re: [SIP] a stupid question on 3way handshake
> 
> 
> 
> 
> 11/20/2000 04:58 PM
> (subscript: Javier Ferreiro Garcia@TSM)
> 
> 
> >The INVITE retransmission will stop on receiving a 
> provisional response, so
> >the ACK is needed to ensure the reliability (and stop 
> retransmission of) the
> >final response.
> 
> I´M NOT A DEVELOPER, BUT I UNDERSTAND THE ACK METHOD AS THE 
> BASIS FOR THE FORK
> FUNCTIONALITY.
> WHEN FORKING (i.e. SENDING MULTIPLE INVITE REQUESTS), THE 
> SESSION SHOULD NOT BE
> AUTOMATICALLY ESTABLISHED WHEN THE CALLER RECEIVES THE INVITE 
> RESPONSE FROM THE
> CALLEE, BUT RATHER WHEN THE CALLER ITSELF AGREES TO ESTABLISH 
> THE SESSION WITH
> ANY PARTICULAR CALLEE FROM THE ONES THAT SENT A POSITIVE 
> INVITE RESPONSE.
> IT ALLOWS IN THIS CASE THE CALLER TO SELECT ONE OF THE 
> PROPOSED LEGS, SENDING
> THE ACK TO THE SELECTED ONE(s) AND CANCELLING THE REST OF THE PENDING
> INVITATIONS.

You wouldn't use CANCEL to terminate the other ones that have responded
positively. For those, you would send BYE. CANCEL is only useful if the
other end hasn't answered.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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Date: Tue, 21 Nov 2000 04:44:35 -0500



 

> -----Original Message-----
> From: Baniel Uri-CUB001 [mailto:Uri.Baniel@motorola.com]
> Sent: Monday, November 20, 2000 12:41 PM
> To: 'sip@lists.bell-labs.com'
> Subject: [SIP] SIP Registration minimal interval?
> 
> 
> Hi
> Is there any 'floor limitation' on the registration interval? 
> I mean, let's say the registrar asks the client to register 
> every 20 minutes.
> Can the client decide to register every 1 minute? If YES, do 
> we have any way to prevent malicious or badly configured 
> client from overloading the network?

Yes, you would call the protocol police. I'll look the number up and send it
to you :)

Seriously, there is nothing a protocol can do to stop users from sending
messages. Thats not a SIP thing, its an IP thing.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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Date: Tue, 21 Nov 2000 01:57:35 -0800

DO that would _really_ confuse existing Record-Route proxies.

We have a few options:
1) Scrap current mechanism completely and come up with something that works
properly.
This should have the benefits that new proxies will be ensured of the
correct behaviour but it is not backwardly-compatiable with the old proxies
(preferably violently so).

Or 
2) Cobble something together from old record-route system so that:
a) old Record-Route proxies can still exist in the path (with known
problems), but
b) newer proxies can detect that the UAS will construct the reverse Route
correctly and
c) that older proxies will forward the Route entries correctly (assuming
they implemented the spec correctly).

Or lastly,
3) Cobble something together from old record route system so that:
A proxy can't tell if the UAS or other older proxies are going to screw
something up by using the old record-route behaviour.
This is the worst possible solution.

I think the current suggestions only fall into the last category which gives
you the worst of both worlds. Think about it, if a proxy wants to be
Record-Route'd there is a usually a reason for it. So it is next to useless
if the proxy can only be "semi-sure" that it will correctly receive reverse
requests. 

IMO opinion we need to come up with a RR proposal that is either
pro-actively _not_ backwardly compatiable or that the proxy can at least
detect that an older Record-Route scheme is being used by the UAS and/or
other proxies.

I have yet to hear suggestions falling into the second category..... so
here's mine: <grin>

- If the UAS receiving the Record-Routes was to put a special contact
parameter into it's Contact then this would at least solve the
"non-detection problem" of downstrea, proxies. 
- The UAS only changes the user@host:port part when building the reverse
Route header (from From or COntact). This leaves all R-R parameter. This
means that non-SIP Request-URI's should be work (and is backwards compat).
- I beleive an older proxy would/should still pass the Record Route
parameters onto the next hop (in the Request-URI) as long as you don't use
angle brackets in the R-R header.  That is, the record route parameters are
actually treated as "other" URL parameters in the Request-URI (which is
legal).

This I think should fall into the second (most preferable) category.

Comments?

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 

> -----Original Message-----
> From: David Shrader [mailto:dshrader@master-consultant.com]
> Sent: Tuesday, November 21, 2000 12:59 PM
> To: SIP List
> Subject: Re: [SIP] Record-Route consensus
> 
> 
> Where do we stand on which server removes the Route header? 
> If I recall, the
> current spec says that the preceding server remotes the route 
> header for the
> next destination. OTOH, I also saw posts suggesting that the 
> server remove
> its own Route header (like the Via header).
> 
> David
> 
> ----------------------------
> David Shrader
> Master Consultant, Inc.
> dshrader@master-consultant.com
> http://www.EyeForTheFuture.com
> 
> > From: Billy Biggs <Billy_Biggs@3com.com>
> > Date: Mon, 20 Nov 2000 09:55:59 -0600
> > To: Jo Hornsby <jhornsby@ubiquity.net>
> > Cc: Sean Olson <sean.olson@ericsson.com>, 
> "Fairlie-Cuninghame, Robert"
> > <rfairlie@nuera.com>, SIP List <sip@lists.bell-labs.com>
> > Subject: Re: [SIP] Record-Route consensus
> > 
> > Jo Hornsby (jhornsby@ubiquity.net):
> > 
> >> Right.  So we're saying remove the bit where the bis specifies that
> >> the Request-URI is copied into the Record-Route, and 
> instead note that
> >> the proxy is free to put any SIP URL in (of course, it should be
> >> routable); and also remove the bit where the bis specifies 
> the Route
> >> Request-URI mangling by the called UA.
> > 
> > Totally.  It should put in some SIP URI that is routable in both
> > directions.
> > 
> >>> Basically, you're just proposing that we don't allow non-SIP URIs
> >>> in the Route, and I think this is completely reasonable.
> >> 
> >> Agreed.  After all, we already say that a proxy is not allowed to
> >> mandate TCP (via the transport parameter) in Record-Route; allowing
> >> non-SIP URIs, I would say, is orders of magnitude more painful.
> > 
> > Dude, you rock my world.
> > 
> > Now, where's Sean Olson's argument? :)
> > 
> > -- 
> > Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> > http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com
> > 
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> > 
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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Date: Tue, 21 Nov 2000 05:10:31 -0500



 

> -----Original Message-----
> From: Billy Biggs [mailto:Billy_Biggs@3com.com]
> Sent: Monday, November 20, 2000 7:54 PM
> To: Henning G. Schulzrinne
> Cc: Jonathan Rosenberg; SIP List
> Subject: Re: [SIP] New I-D on multiparty conferencing models
> 
> 
> Henning G. Schulzrinne (hgs@cs.columbia.edu):
> 
> > >   [...] but if I see:
> > >          m=audio 6646 RTP/AVP 0
> > >          m=audio 6656 RTP/AVP 1
> > >   Then I don't get to choose?  I'm supposed to send to both?
> > 
> > Yes. Each m line is independent and additive.
> 
>   This means that I could be asked to commit a huge denial of service
> attack.  There are security implications here.

I don't see this as fundmentally different. I could specify a multicast
address that has gazillions of receives, and do far more damage than
enumerating a list of unicast addresses.  Or, I could send you lots of
INVITEs with different SDPs. You are always at liberty to implement policy
to reject a call when you want. The point here is that there are legitimate,
and indeed, useful cases where multiple m lines of the same type are
present. This is one. If you read draft-rosenberg-sip-app-components, you
will see other important uses that center around dtmf.

I believe the use of multiple m lines is well within the spirit of SDP.
Indeed, in the original model of a multicast, multimedia conference, this is
quite reasonable. Consider a session thats a shuttle launch. I have one
audio stream which is the launch itself, and a separate one which is the
audio commentary from the reporter. I also have a video stream of it.
Perfectly reasonable. 

Now, whether this is supported is a separate quesiton. Part of the
motivation of this draft, and the 3pcc draft, is to show that these
seemingly subtle things, if you do them, can open a wealth of services and
features. 3pcc requires, for example, uses to accept re-INVITEs with
modified connection addresses. SIP doesn't say anything specific about this.
Likely people don't do it. The draft helps show why you should. In fact, at
the last bakeoff, we had people who added support for it in order to enable
3pcc apps. 

>   The assertion in the draft that "Interestingly, this 
> conference model
> is possible baseline SIP" is misleading.  It implies that all 
> SIP phones
> are expected to support it. 

Right now, I expect no one supports it. Down the road, I would expect that
many phones support it, and those that don't, reject the call with some
reasonable response. 

 I guess detecting the situation and
> shutting down the call is acceptable?
> 
> > >   I don't think that anyone will support this either.  I would say
> > > that it is too late for SDP, and if you want to signal this, lets
> > > define a stronger description protocol.
> > 
> > That's a separate and necessary step.
> 
>   Well, then maybe the problem is that SDP is too powerful 
> for use as a
> base description protocol for SIP.  If supporting SDP means 
> supporting:
> 
>   - Sending audio to multiple locations, and
>   - Mixing audio from multiple sources
> 
>   ... then I think that will exceed the capacity of many embedded
> phones. 

Mixing, perhaps. I'll note that I know of at least one embedded phone that
does this, though. As for sending to multiple locations, thats trivial when
the codec is the same. In fact, the RTP library we wrote, you can specify
multiple unicast destinations when you set up the session, and it sends to
all of them for you.

> It also might simply be too difficult for others to code.

I think you are overreacting a bit to the complexity involved for sending
the same audio stream to multiple destinations.

Billy initially wrote:
> >    It is also possible to transition from a end system mixed
> >    conference (even one with a complex connection topology), to a
> >    centralized conference server. One user takes responsibility for
> >    initiating the transition. It proceeds as described 
> above. However,
> >    the REFER request is sent to all SIP peers adjacent to 
> the user. In
> >    addition, when a SIP UA receives a REFER, they must not 
> only act on
> >    it as described above, but also generate a REFER to any of their
> >    adjacent SIP peers. In essence, the REFER message is propagated
> >    along the connection graph, starting at the root (which 
> is the user
> >    who initiates the transition). The transition will work 
> so long as
> >    the graph has no cycles (which is needed anyway, as discussed
> >    above), and so long as only one user attempts to initiate the
> >    transition. If multiple users attempt to initiate the 
> transition at
> >    the same time, the conference will break into two disjoint ad-hoc
> >    conferences, with membership depending on the temporal 
> dynamics of
> >    the REFER propagation.
> 
>   This implies that REFER propagation should be automatic, 
> but no method
> is given to signal this.

No signaling means is needed. The point here is that if a UA is doing this
kind of mixing, and it receives a REFER, it has to propagate it for the
conference to continue to function. Its a processing requirement that goes
along with support for the model in general. Not all UAs that support REFER
would do this; only ones actively performing this mixing role in a
conference.

>   It doesn't seem reasonable to expect UAs to propagate a REFER along
> their conferences.  For example, say A is in a 3-way call performing
> End System Mixing:
> 
>                        A
>                  D    / \
>                      B   C
> 
>   And say that B REFERs A to D.  Should A propagate this REFER to C?
> No.  There is no way to know if D is a conference bridge or simply
> another phone.

Hmm. This is a good point. I could wave my hands here and say "well, B
should know better than to transfer a multiparty call to a non-bridge
entity". But, thats somewhat unsatisfying. There may be a need to signal
whether REFER propagation is needed. I need to think about it.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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Date: Tue, 21 Nov 2000 05:27:00 -0500



 

> -----Original Message-----
> From: Stuart Wray [mailto:Stuart.Wray@marconi.com]
> Sent: Monday, November 20, 2000 10:15 AM
> To: sip@lists.bell-labs.com
> Subject: [SIP] URL comparison
> 
> 
> 
> 
> Henning Schulzrinne suggested that I submit the following 
> questions about URL
> comparison to the SIP list.
> 
> On page 21 of rfc2543bis02 , it says
> 
> (call this clause A)
> "user, password, host, port and any url-parameter parameters 
> of the URI must
> match. If a component is omitted, it matches based on its 
> default value."
> 
> So far, this is clear, but the paragraph goes on to say that
> 
> (call this clause B)
> "Components not found in both URLs being compared are ignored"
> 
> Now consider these two URLs:
> 
>      sip:juser@example.com:6666
>      sip:juser@example.com
> 
> Clause A means that these URLs compare as unequal, but clause 
> B means that they
> compare as equal.
> 
> Should clause B actually start with the word "other"?
> 
> (clause B')
> "Other components not found in both URLs being compared are ignored"
> 
> (If so, is this actually the correct judgement? I would have 
> thought that a URL
> should not compare equal to a URL with an extra component. 
> Why is it useful for
> them to compare equal?)

Hmm. Good question. I honestly can't recall why we said this. I would tend
to agree that:

sip:user@host:5050;foo=bar

and

sip:user@host:5050

are not the same.

Unless someone can come up with a good reason why this should stay, I would
argue that we change this to say "If one of the URLs has a component that is
not present (and has no default value) in the other URL, the URLs are not
equal".


> 
> And now another separate point: the text on page 21 goes on to say:
> 
> "Note that the two URLs example.com and example.com:5060, 
> while considered
> equal, may not lead to the same server, as the former causes 
> a DNS SRV lookup,
> while the latter only uses the A record".
> 
> However, this is not true --- on page 12 it says:
> 
> "3. The Request-URI is examined. If it contains an explicit 
> port number other
> than 5060, the next two steps are skipped".
> 
> The A record is only used if there is a port number other 
> than 5060 ---- so
> example.com and example.com:5060 both use the SRV record. (Of 
> course this may
> result in different machines actually being contacted, but 
> that's a different
> matter entirely, not to do with A records).

This is an error, I believe, resulting in the evolution of this text. We
modified the SRV procedure so that example.com and example.com:5060 were,
indeed, pointing to the same server, but didn't update the text on top which
pointed out the problem.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Robert Fairlie-Cuninghame wrote:
>
> We have a few options:
> 1) Scrap current mechanism completely and come up with something
> that works properly.  This should have the benefits that new
> proxies will be ensured of the correct behaviour but it is not
> backwardly-compatiable with the old proxies (preferably violently
> so).
>
> Or
> 2) Cobble something together from old record-route system so that:
> a) old Record-Route proxies can still exist in the path (with known
> problems), but
> b) newer proxies can detect that the UAS will construct the reverse
> Route correctly and
> c) that older proxies will forward the Route entries correctly
> (assuming they implemented the spec correctly).
>
> Or lastly,
> 3) Cobble something together from old record route system so that:
> A proxy can't tell if the UAS or other older proxies are going to
> screw something up by using the old record-route behaviour.
> This is the worst possible solution.

I'm not sure how you come to this conclusion.

If I requote 2543, section 6.29:

   The server copies the Record-Route header field unchanged into the
   response. (Record-Route is only relevant for 2xx responses.)

   The calling user agent client copies the Record-Route header into a
   Route header field of subsequent requests within the same call leg,
   reversing the order of requests, so that the first entry is closest
   to the user agent client. If the response contained a Contact header
   field, the calling user agent adds its content as the last Route
   header. Unless this would cause a loop, any client MUST send any
   subsequent requests for this call leg to the first Request-URI in the
   Route request header field and remove that entry.

If I were feeling particularly uncharitable, then I would say a
(calling) 2543-compliant UA that did not copy _all_ of the URI
parameters was seriously broken -- it had no business assuming
that it was allowed to mangle the Record-Route values in any
way.

Further, looking to section 2 of the same:

   The transport, maddr, and ttl parameters MUST NOT be used in the From
   and To header fields and the Request-URI; they are ignored if
   present.

   Headers: Headers of the SIP request can be defined with the "?"
        mechanism within a SIP URL. The special hname "body" indicates
        that the associated hvalue is the message-body of the SIP INVITE
        request. Headers MUST NOT be used in the From and To header
        fields and the Request-URI; they are ignored if present.  hname
        and hvalue are encodings of a SIP header name and value,
        respectively. All URL reserved characters in the header names
        and values MUST be escaped.

   Method: The method of the SIP request can be specified with the
        method parameter.  This parameter MUST NOT be used in the From
        and To header fields and the Request-URI; they are ignored if
        present.

Therefore, a 2543-compliant client (be it proxy or UA) again would
have no business removing any parameters that it did not recognise,
besides the ones listed above.  It's a bit of a shame that maddr
was included, but we don't actually need that parameter with this
proposal, anyway. &:)

> I think the current suggestions only fall into the last category
> which gives you the worst of both worlds. Think about it, if a
> proxy wants to be Record-Route'd there is a usually a reason
> for it. So it is next to useless if the proxy can only be
> "semi-sure" that it will correctly receive reverse requests.

This is true -- I find it very unlikely that 2543-compliant called
UAs would ever do The Right Thing when sending requests; however,
in the end, even ensuring that one is a pure 3000 (imaginary new
RFC) network, isn't enough in itself.

That is because UAs can still misbehave -- maliciously, or
otherwise.  Record-Route is a mechanism that allows proxies to
ask that they are kept in the signalling path; however, they
can't insist such that the signalling path will be torn down
if they are not.

If a proxy is to know that it's going to stay in the signalling
path, then it will be achieved with trust relationships, and
various networking magic.  Most often, this will probably
consist of talking to vendors/service-providers that offer
guarantees; and ultimately, I think that this is the best that
one can do.

> IMO opinion we need to come up with a RR proposal that is either
> pro-actively _not_ backwardly compatiable or that the proxy can at least
> detect that an older Record-Route scheme is being used by the UAS and/or
> other proxies.
>
> I have yet to hear suggestions falling into the second category..... so
> here's mine: <grin>
>
> - If the UAS receiving the Record-Routes was to put a special contact
> parameter into it's Contact then this would at least solve the
> "non-detection problem" of downstrea, proxies.
> - The UAS only changes the user@host:port part when building the reverse
> Route header (from From or COntact). This leaves all R-R parameter. This
> means that non-SIP Request-URI's should be work (and is backwards compat).
> - I beleive an older proxy would/should still pass the Record Route
> parameters onto the next hop (in the Request-URI) as long as you don't use
> angle brackets in the R-R header.  That is, the record route
> parameters are
> actually treated as "other" URL parameters in the Request-URI (which is
> legal).
>
> This I think should fall into the second (most preferable) category.
>
> Comments?

This will work -- of course.  It's just that I see no real need
to make the differentiation (old vs. new), when on paper (ignoring
bis (:&), we already have interoperation[0].

Cheers,


 - Jo.

[0] With new UAs that don't mangle the Route Request-URIs, of course.


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Erik,

The implication of what you say is that all set methods in the header
interfaces which take String arguments must throw a checked
JainSipParseException. I think this is reasonable given that different
implementations will be more tolerant than others. However, in order to allow
lazy parsing by an implementation, all get methods in the header interfaces
(regardless of return type) must throw a checked parse exception. This is a
lot of checked exceptions - and will make life more awkward for the
application developer. It would certainly be much less effort to not allow
lazy parsing - there would be no need to throw a parse exception for all get
methods - greatly reducing the number of checked exceptions. Unfortunately it
has been said that JAIN SIP support for lazy parsing is a must. If it is then
I guess there is no other choice than to include the exceptions?

Regards,
Chris


Erik Nissen wrote:

> Chris Harris wrote:
>
> >
> > We need to decide whether this should be a checked or runtime exception.
>
> The rule of thumb is to use:
> * checked exceptions when the program should handle situations which
> can/will occur.
> * runtime exceptions when there are programming error (God forbid :-)
>
> So:
> Parse errors should throw checked exceptions.
> Becaurse parse errors are outside the control of the programmer
> and he/she should be forced to handle such conditions in the code.
>
> Null/wrong arguments should throw runtime exceptions.
>
> If it is possible, reduce the number of methods throwing (checked)
> exceptions
> to a minimum.
>
> Best regards,
> /Erik Nissen,
> Ericsson Utveckling AB


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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: "'Jo Hornsby'" <jhornsby@ubiquity.net>
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Date: Tue, 21 Nov 2000 03:27:52 -0800

Jo, you're right, I was referring the to "new" old Record-Route scheme
presented in the bis draft not the "old" old RFC2543 R-R scheme. [ I also
admit to a fading memory of the old RFC specifics :) ] Its the Route
mangling in the _UAS_ (not the proxies) that is causing us the grief now
(ala the bis). The route mangling was to maintain the protocol invariant of
the Request-URI (not very well) which even JR agrees might be causing more
pain than its worth.

One could say "why bother worrying about backwards compat - nobody
implemented the Route mangling anyway". Is this true? Or have a sufficient
number of UAS's implemented this to make it worthwhile to put some thought
into detecting if Route mangling will not occur ?

If it is not worthwhile worrying about bis implementations, then the fix is
simple, right? Each proxy simply puts its own globally routeable address
into the Record-Route URL and then go back to the simpler RFC2543 behaviour.

Otherwise, well we also have to have the UAS do something indicative that
the "new old old" behaviour is implemented. 

I think I can guess what the consensus will be, "Don't worry, be happy." I
think I would tend to agree. :) 

[As an aside, IMO, the argument that a mutual trust relationship is required
to ensure a proxy stays in the signalling path, doesn't apply to justifying
why something can't be _expected_ to succeed between a set of co-operating
(ie, non-intentionally-malicous) UA and proxies].   

Regards,

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 
  
> -----Original Message-----
> From: Jo Hornsby [mailto:jhornsby@ubiquity.net]
> Sent: Tuesday, November 21, 2000 6:31 PM
> To: Fairlie-Cuninghame, Robert; 'David Shrader'; SIP List
> Subject: RE: [SIP] Record-Route consensus
> 
> 
> Robert Fairlie-Cuninghame wrote:
> >
> > We have a few options:
> > 1) Scrap current mechanism completely and come up with something
> > that works properly.  This should have the benefits that new
> > proxies will be ensured of the correct behaviour but it is not
> > backwardly-compatiable with the old proxies (preferably violently
> > so).
> >
> > Or
> > 2) Cobble something together from old record-route system so that:
> > a) old Record-Route proxies can still exist in the path (with known
> > problems), but
> > b) newer proxies can detect that the UAS will construct the reverse
> > Route correctly and
> > c) that older proxies will forward the Route entries correctly
> > (assuming they implemented the spec correctly).
> >
> > Or lastly,
> > 3) Cobble something together from old record route system so that:
> > A proxy can't tell if the UAS or other older proxies are going to
> > screw something up by using the old record-route behaviour.
> > This is the worst possible solution.
> 
> I'm not sure how you come to this conclusion.
> 
> If I requote 2543, section 6.29:
> 
>    The server copies the Record-Route header field unchanged into the
>    response. (Record-Route is only relevant for 2xx responses.)
> 
>    The calling user agent client copies the Record-Route header into a
>    Route header field of subsequent requests within the same call leg,
>    reversing the order of requests, so that the first entry is closest
>    to the user agent client. If the response contained a 
> Contact header
>    field, the calling user agent adds its content as the last Route
>    header. Unless this would cause a loop, any client MUST send any
>    subsequent requests for this call leg to the first 
> Request-URI in the
>    Route request header field and remove that entry.
> 
> If I were feeling particularly uncharitable, then I would say a
> (calling) 2543-compliant UA that did not copy _all_ of the URI
> parameters was seriously broken -- it had no business assuming
> that it was allowed to mangle the Record-Route values in any
> way.
> 
> Further, looking to section 2 of the same:
> 
>    The transport, maddr, and ttl parameters MUST NOT be used 
> in the From
>    and To header fields and the Request-URI; they are ignored if
>    present.
> 
>    Headers: Headers of the SIP request can be defined with the "?"
>         mechanism within a SIP URL. The special hname "body" indicates
>         that the associated hvalue is the message-body of the 
> SIP INVITE
>         request. Headers MUST NOT be used in the From and To header
>         fields and the Request-URI; they are ignored if 
> present.  hname
>         and hvalue are encodings of a SIP header name and value,
>         respectively. All URL reserved characters in the header names
>         and values MUST be escaped.
> 
>    Method: The method of the SIP request can be specified with the
>         method parameter.  This parameter MUST NOT be used in the From
>         and To header fields and the Request-URI; they are ignored if
>         present.
> 
> Therefore, a 2543-compliant client (be it proxy or UA) again would
> have no business removing any parameters that it did not recognise,
> besides the ones listed above.  It's a bit of a shame that maddr
> was included, but we don't actually need that parameter with this
> proposal, anyway. &:)
> 
> > I think the current suggestions only fall into the last category
> > which gives you the worst of both worlds. Think about it, if a
> > proxy wants to be Record-Route'd there is a usually a reason
> > for it. So it is next to useless if the proxy can only be
> > "semi-sure" that it will correctly receive reverse requests.
> 
> This is true -- I find it very unlikely that 2543-compliant called
> UAs would ever do The Right Thing when sending requests; however,
> in the end, even ensuring that one is a pure 3000 (imaginary new
> RFC) network, isn't enough in itself.
> 
> That is because UAs can still misbehave -- maliciously, or
> otherwise.  Record-Route is a mechanism that allows proxies to
> ask that they are kept in the signalling path; however, they
> can't insist such that the signalling path will be torn down
> if they are not.
> 
> If a proxy is to know that it's going to stay in the signalling
> path, then it will be achieved with trust relationships, and
> various networking magic.  Most often, this will probably
> consist of talking to vendors/service-providers that offer
> guarantees; and ultimately, I think that this is the best that
> one can do.
> 
> > IMO opinion we need to come up with a RR proposal that is either
> > pro-actively _not_ backwardly compatiable or that the proxy 
> can at least
> > detect that an older Record-Route scheme is being used by 
> the UAS and/or
> > other proxies.
> >
> > I have yet to hear suggestions falling into the second 
> category..... so
> > here's mine: <grin>
> >
> > - If the UAS receiving the Record-Routes was to put a 
> special contact
> > parameter into it's Contact then this would at least solve the
> > "non-detection problem" of downstrea, proxies.
> > - The UAS only changes the user@host:port part when 
> building the reverse
> > Route header (from From or COntact). This leaves all R-R 
> parameter. This
> > means that non-SIP Request-URI's should be work (and is 
> backwards compat).
> > - I beleive an older proxy would/should still pass the Record Route
> > parameters onto the next hop (in the Request-URI) as long 
> as you don't use
> > angle brackets in the R-R header.  That is, the record route
> > parameters are
> > actually treated as "other" URL parameters in the 
> Request-URI (which is
> > legal).
> >
> > This I think should fall into the second (most preferable) category.
> >
> > Comments?
> 
> This will work -- of course.  It's just that I see no real need
> to make the differentiation (old vs. new), when on paper (ignoring
> bis (:&), we already have interoperation[0].
> 
> Cheers,
> 
> 
>  - Jo.
> 
> [0] With new UAs that don't mangle the Route Request-URIs, of course.
> 

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To: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
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Subject: RE: [SIP] Record-Route consensus
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Robert Fairlie-Cuninghame wrote:
>
> Jo, you're right, I was referring the to "new" old Record-Route scheme
> presented in the bis draft not the "old" old RFC2543 R-R scheme. [ I also
> admit to a fading memory of the old RFC specifics :) ]

Oh, right! &:)

> Its the Route
> mangling in the _UAS_ (not the proxies) that is causing us the grief now
> (ala the bis). The route mangling was to maintain the protocol
> invariant of
> the Request-URI (not very well) which even JR agrees might be causing more
> pain than its worth.
>
> One could say "why bother worrying about backwards compat - nobody
> implemented the Route mangling anyway". Is this true? Or have a sufficient
> number of UAS's implemented this to make it worthwhile to put some thought
> into detecting if Route mangling will not occur ?
>
> If it is not worthwhile worrying about bis implementations, then
> the fix is
> simple, right? Each proxy simply puts its own globally routeable address
> into the Record-Route URL and then go back to the simpler RFC2543
> behaviour.

> Otherwise, well we also have to have the UAS do something indicative that
> the "new old old" behaviour is implemented.

Very much so.  I was basically working on the premise of "screw
the current bis". &:)

As you've noted, I don't think we can have one solution which will
work across the board.

> [As an aside, IMO, the argument that a mutual trust relationship
> is required to ensure a proxy stays in the signalling path, doesn't
> apply to justifying why something can't be _expected_ to succeed
> between a set of co-operating (ie, non-intentionally-malicous) UA
> and proxies].

True.  True.

However, I think it's likely in practise that UAs will be
"required" to conform to RFC N (where N is the next SIP RFC,
or whatever), if they are going to be used in certain networks;
this is what I was alluding (albeit poorly) to.

Cheers,


 - Jo.


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Billy Biggs wrote:
>   If that is to be kept, then please also mention that UAs which exhibit
> this behavior be able to tell when a request loops back to themselves.
> This can happen in many cases, and I've seen UAs crash because they
> can't perform call-leg matching.
> 
>   Some cases:
> 
>   1. User makes a call through a proxy which forks back to themselves (I
>      called Joe but Joe has his calls forwarded to me).
> 
>   2. User sends a SUBSCRIBE for their voicemail to sip:me@myproxy with
>      an Accept-Contact: *;feature="voicemail", but the user doesn't have
>      a voicemail server so the SUBSCRIBE comes back to the phone.

Am I right in thinking that both of these cases are reasonably
easily solved if the UA always inserts a From tag, too; or is
the problem more subtle than that?

Cheers,


 - Jo.


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Date: Tue, 21 Nov 2000 14:53:20 +0700 (GMT)


Dear all

	Is there any SDP API public for common usage?

Mr. Voravit H.


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From: "Hisham Khartabil" <hisham.khartabil@hotsip.com>
To: <sip@lists.bell-labs.com>
Subject: RE: [SIP] Contact header is a 200 response to SUBSCRIBE for IMPP
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Hi,
I'm posting the following again since there was no reply the first time
around

Regards,
Hisham Khartabil
Hotsip Finland
www.hotsip.com

-----Original Message-----
From: sip-admin@lists.bell-labs.com [mailto:sip-admin@lists.bell-labs.com]On
Behalf Of Hisham Khartabil
Sent: Thursday, 9 November 2000 11:41 AM
To: sip@lists.bell-labs.com
Subject: [SIP] Contact header is a 200 response to SUBSCRIBE for IMPP

Section 4.2 of the presence draft has the following examples:

SUBSCRIBE sip:jdrosen@dynamicsoft.com SIP/2.0
Via: SIP/2.0/UDP userpc.example.com
To: sip:jdrosen@dynamicsoft.com
From: sip:user@example.com
Contact: sip:user@userpc.example.com
Call-ID: knsd08alas9dy@3.4.5.6
CSeq: 1 SUBSCRIBE
Content-Length: 0



SIP/2.0 200 OK
Via: SIP/2.0/UDP presenceserver.dynamicsoft.com
Via: SIP/2.0/UDP userpc.example.com
To: sip:jdrosen@dynamicsoft.com
From: sip:user@example.com
Contact: sip:user@userpc.example.com
Record-Route: sip:jdrosen@example.com;maddr=presenceserver.dynamicsoft.com
Expires: 3600
Call-ID: knsd08alas9dy@3.4.5.6
CSeq: 1 SUBSCRIBE
Content-Type: application/xpidf+xml
Content-Length: 287

<?xml version="1.0"?>
<!DOCTYPE presence
 PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
<presence>
 <presentity uri="sip:jdrosen@dynamicsoft.com;method="SUBSCRIBE">
  <atom id="779js0a98">
   <address uri="sip:jdrosen@dynamicsoft.com;method=INVITE">
    <status status="open"/>
   </address>
  </atom>
 </presentity>
</presence>


Then, it goes on to say:
"Notice that the response has mirrored many fields from the
request,including the To, From, Call-ID, CSeq, and Via headers. The PA has
also added a Contact header, providing an address where it can be reached.
The response also contains a Record-Route header. This header was presumably
inserted by the proxy (the address in the maddr header is that of a presence
server acting as a proxy), and is reflected in the 200 OK response. The
response also contains a presence document for the presentity, and a Contact
header. The Contact headers in a response to SUBSCRIBE list all the current
addresses which have been subscribed."

The example shows that the Contact header in the response does actually list
the current address which has been subscribed (user@userpc.example.com).
But I fail to see where the Contact header that the PA added providing an
address where it can be reached.

Regards,
Hisham Khartabil
Hotsip Finland




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From: "Hisham Khartabil" <hisham.khartabil@hotsip.com>
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Take 2:

Hi all,

Apologies for the long email, but I have few questions regarding SIP
Extensions for Presence.

3. Definitions

"Presence User Agent (PUA): A Presence User Agent manipulates presence
information for a presentity. In SIP terms, this means that a PUA generates
REGISTER requests, conveying some kind of information about the presentity."

"Presence Agent (PA): A presence agent is a SIP user agent which is capable
of receiving SUBSCRIBE requests, responding to them, and generating
notifications of changes in presence state."

Q: what about the sip user agent that generates SUBSCRIBE request and
receiving notifications? Should we have a definition for that? Are we
assuming s/he is the SUBSCRIBER?

4.2 Message Flow
...
"In fact, it is possible that two different presence clients register for
the same presentity. In this case, the presence server, acting as a proxy,
may fork the SUBSCRIBE, which means it sends multiple copies of the request,
one to each presence client. Each client will presumably query its principal
about whether the subscription is acceptable. In the common case where a
principal has left their presence tool running at work, and then gone home
and started the tool there, the principal will accept the subscription at
home. This causes a success response to be generated, and forwarded to the
presence server (which is acting as a proxy), and then back to the
subscriber. The presence tool at home then assumes responsibility for that
subscription. The tool at work can continue to support subscriptions it
received previously."

Since I am not at work, my status will not change so no notifications will
be generated.  This does not give the subscriber and up to date notification
of my presence.

Q: how about if I don't want my subscriptions at work to continue and I want
them to be transferred home?  I know this is achieved over time since
SUBSCRIBE has Expires: header.  But this could take hours (or one hour).
There should be a way to transfer subscriptions.

...

"As an optimization, notifications do not need to be sent if the subscriber
is not actually online. This improves scalability."

Q: is this assuming that A is subscribed to B and vice versa? Otherwise, who
would a Presence Agent Know that the subscriber on offline?

...

"A SUBSCRIBE request MUST contain a Contact header. This indicates the
address(es) (as a SIP URL) to which the client would like to receive
notifications. This MUST be be one or more SIP addresses, containing the
fully qualified domain names for the host generating the subscription, or
the IP address of the host generating the subscription. Other addresses are
possible, supporting third party subscriptions. If it contains multiple
addresses, notifications will be sent to each address. If no Contact header
is present, no notifications will be sent."

Q: since a SUBSCRIBE request MUST contain a Contact header, why wouldn't you
respond with 4xx if no Contact is present?



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From: "Hisham Khartabil" <hisham.khartabil@hotsip.com>
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Hi,

The REGISTER message in example 13.2 of the SIP for Presence draft has the
method in the Contact header as method=MESSAGE.  It should say SUBSCRIBE.

Regards,
Hisham


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From: Billy Biggs <Billy_Biggs@3com.com>
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Jo Hornsby (jhornsby@ubiquity.net):

> Billy Biggs wrote:
> >   If that is to be kept, then please also mention that UAs which
> > exhibit this behavior be able to tell when a request loops back to
> > themselves.  This can happen in many cases, and I've seen UAs
> > crash because they can't perform call-leg matching.
> > 
> >   Some cases:
> > 
> >   1. User makes a call through a proxy which forks back to
> >      themselves (I called Joe but Joe has his calls forwarded to
> >      me).
> > 
> >   2. User sends a SUBSCRIBE for their voicemail to sip:me@myproxy
> >      with an Accept-Contact: *;feature="voicemail", but the user
> >      doesn't have a voicemail server so the SUBSCRIBE comes back to
> >      the phone.
> 
> Am I right in thinking that both of these cases are reasonably easily
> solved if the UA always inserts a From tag, too; or is the problem
> more subtle than that?

  Their suggestion was that the UA always use the same local tag so when
you're rebooted and see a re-INVITE you know not to return a 481 cause
it's actually for you.  It's in bis-02.  For that to work, the UA will
always put in a from tag on outgoing requests, but it will always be the
same.  As well, the UA will always answer with the same to tag on
outgoing responses.

  So, UA sends a request, it loops back, they end up putting the same
tag as the To and as the From...  You see the problem?  Call-leg
endpoint matching is based on:

  call-id,local-uri,local-tag,remote-uri,remote-tag

  And you'll end up with the same key for both.  The code gets cleaner
if you always pick random tags.  Unfortunately, call-leg-end matching is
very difficult.


  As an aside, things would be alot more robust if tags were mandatory.
Ideally, I want to match on:

  call-id,local-tag,remote-tag

  (or for an unconnected call-leg end):

  call-id,local-tag

  Much nicer than this total hack of trying to compare the URIs,
especially when they aren't SIP URIs.  Oh well.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Jonathan Rosenberg wrote:
> 
>
> 
> Yes, you would call the protocol police. I'll look the number up and send it
> to you :)
> 
> Seriously, there is nothing a protocol can do to stop users from sending
> messages. Thats not a SIP thing, its an IP thing.
> 

At least in the United States, interfering with the operation of a
computer system can be a federal offense, so you can indeed call the FBI
as the "protocol police". Yahoo and others did when they were on the
receiving end of various DDOS attacks.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Jonathan Rosenberg wrote:

> Hmm. Good question. I honestly can't recall why we said this. I would tend
> to agree that:
>
> sip:user@host:5050;foo=bar
>
> and
>
> sip:user@host:5050
>
> are not the same.
>
> Unless someone can come up with a good reason why this should stay, I would
> argue that we change this to say "If one of the URLs has a component that is
> not present (and has no default value) in the other URL, the URLs are not
> equal".

This depends on what semantics you apply to that additional component. I
would say in this case that these two URLs are equal when considered
in the context of a To: or a From:, but unequal in the context of a Request-URI.

If you want to make these two cases consistent, I would lean towards considering

these two URLs equal in both cases. Otherwise, we should just do a case
insensitive
string comparison, port included.

/sean



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Date: Tue, 21 Nov 2000 10:57:33 -0500

Hi -
 I have a question on registration -
Let's say I have client (A) who is configured to register with (B),  where B
is a proxy server. When A attempts to register with B, B decides to redirect
the request to another proxy server C. (If I understand the spec correctly,
I believe that I could accomplish this by having B respond to the
registration with a 301 Moved Permanently, including a Contact header for C
in the response.)

The question is this: Does this redirection only impact future registration
requests, or does it impact all future requests - such as INVITE?

What I am trying to accomplish is to have A direct all of its future
requests (not just registration) to the new proxy server (C). Will the
redirect that I mention above accomplish this?

Thanks,
Bill

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Sean Olson wrote:
> 
> Jonathan Rosenberg wrote:
> 
> > Hmm. Good question. I honestly can't recall why we said this. I would tend
> > to agree that:
> >
> > sip:user@host:5050;foo=bar
> >
> > and
> >
> > sip:user@host:5050
> >
> > are not the same.
> >
> > Unless someone can come up with a good reason why this should stay, I would
> > argue that we change this to say "If one of the URLs has a component that is
> > not present (and has no default value) in the other URL, the URLs are not
> > equal".

The problem is that with new parameters, older applications won't know
whether the parameter has a default value and what that might be. Thus,
you can either

- compare well-known parameters (RFC 2543) according to this rule and
new parameters according to "ignore" or "unequal if one has it, the
other doesn't" rule. This seems less than satisfactory in terms of
robustness and simplicity.

- Declare missing parameters as unequal to any existing parameter. That
doesn't work, due to default values.

- Ignore ;-parameters that are only present in one of the URLs being
compared. This seems to work for most practical cases. Any examples
where this would break things?

> 
> This depends on what semantics you apply to that additional component. I
> would say in this case that these two URLs are equal when considered
> in the context of a To: or a From:, but unequal in the context of a Request-URI.

Having two different comparison routines seems like a recipe for subtle
interoperation problems.

> 
> If you want to make these two cases consistent, I would lean towards considering
> 
> these two URLs equal in both cases. 

That would be my preference. This is in the spirit of "you can add
parameters/headers, but don't expect them to matter unless Require'd".

> Otherwise, we should just do a case
> insensitive
> string comparison, port included.
> 
> /sean
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Subject: Re: [SIP] From tags (was: I-D 
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Billy Biggs wrote:
> 
> Jo Hornsby (jhornsby@ubiquity.net):


> 
>   So, UA sends a request, it loops back, they end up putting the same
> tag as the To and as the From...  You see the problem?  Call-leg
> endpoint matching is based on:
> 
>   call-id,local-uri,local-tag,remote-uri,remote-tag
> 
>   And you'll end up with the same key for both.  The code gets cleaner
> if you always pick random tags.  Unfortunately, call-leg-end matching is
> very difficult.

I'm not sure I follow this logic. If A calls B and B proxies back to A,
we get A sending

INVITE B
From: A
To: B

The proxy forwards it to A again:

INVITE A
From: A
To: B

A can easily tell that it was the origin of the request ("From: A"), so
I'm not sure how tags get involved here. In this example, the request
doesn't even have To tags.

> 
>   As an aside, things would be alot more robust if tags were mandatory.
> Ideally, I want to match on:
> 
>   call-id,local-tag,remote-tag
> 
>   (or for an unconnected call-leg end):
> 
>   call-id,local-tag
> 
>   Much nicer than this total hack of trying to compare the URIs,
> especially when they aren't SIP URIs.  Oh well.

I doubt that anything but tel: URLs will be useful in this context, so
I'm not sure how big a problem this is. I assume by "local-tag" you mean
the From tag?
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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From: Billy Biggs <Billy_Biggs@3com.com>
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Date: Tue, 21 Nov 2000 11:26:00 -0600

Henning G. Schulzrinne (hgs@cs.columbia.edu):

> >   So, UA sends a request, it loops back, they end up putting the
> > same tag as the To and as the From...  You see the problem?
> > Call-leg endpoint matching is based on:
> > 
> >   call-id,local-uri,local-tag,remote-uri,remote-tag
> > 
> >   And you'll end up with the same key for both.  The code gets
> > cleaner if you always pick random tags.  Unfortunately,
> > call-leg-end matching is very difficult.
> 
> I'm not sure I follow this logic. If A calls B and B proxies back to
> A, we get A sending
> 
> INVITE B
> From: A
> To: B
> 
> The proxy forwards it to A again:
> 
> INVITE A
> From: A
> To: B
> 
> A can easily tell that it was the origin of the request ("From: A"),
> so I'm not sure how tags get involved here. In this example, the
> request doesn't even have To tags.

  What if A and B are the same URI?  How do I tell them apart?  The user
answers the call, I send a 200 OK to the INVITE and then ACK it.  Now
one of the lines hangs up and I send a BYE: how can I tell which
call-leg end the BYE was meant for when it comes back?  Was I even able
to match that ACK properly?  Do I have to special-case calls to myself?
That would be really annoying, especially if I'm a gateway.

  Also note that if you have two UAs which always use the same tags, and
there are two calls up between them, you're relying on the call-id for
each call to be unique in order to tell them apart.  I'm not convinced
this will always be the case, as there have been hints that the call-id
can be reused to infer special behavior (see my Replaces draft for a
cleaner solution).

> [...] I assume by "local-tag" you mean the From tag?

  Well, when I send a request I put the localuri/localtag as the from
and the remoteuri/remotetag as the to.  When I receive an initial
request, I populate the localuri from the to and add a localtag, and the
remoteuri/remotetag is taken from the from.

  I don't like calling them 'To tags' or 'From tags' because it gets
confusing.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Billy Biggs wrote:
> 

> > I'm not sure I follow this logic. If A calls B and B proxies back to
> > A, we get A sending
> >
> > INVITE B
> > From: A
> > To: B
> >
> > The proxy forwards it to A again:
> >
> > INVITE A
> > From: A
> > To: B
> >
> > A can easily tell that it was the origin of the request ("From: A"),
> > so I'm not sure how tags get involved here. In this example, the
> > request doesn't even have To tags.
> 
>   What if A and B are the same URI?  How do I tell them apart? 

Are you saying that A wants to talk to himself? How else would they be
the same under normal circumstances? A gateway may want to connect two
ports on the same box through the network, but they would have different
phone numbers, i.e., different From/To/Request-URIs.

I also don't see that if you do normal loop detection, you couldn't
handle calls to yourself. Responses and ACKs go to the Via or Contact,
respectively.

Again, given that in this scenario, tags aren't even involved, I'm at a
loss as to why they can matter.


>  The user
> answers the call, I send a 200 OK to the INVITE and then ACK it.  Now
> one of the lines hangs up and I send a BYE: how can I tell which
> call-leg end the BYE was meant for when it comes back?  Was I even able
> to match that ACK properly?  Do I have to special-case calls to myself?

Calls to myself are probably a case for the psychologist or a support
group :-)

> That would be really annoying, especially if I'm a gateway.
> 
>   Also note that if you have two UAs which always use the same tags, and
> there are two calls up between them, you're relying on the call-id for
> each call to be unique in order to tell them apart.  I'm not convinced
> this will always be the case, as there have been hints that the call-id
> can be reused to infer special behavior (see my Replaces draft for a
> cleaner solution).

If anything, we're probably moving away from overloading the call-id,
primarily because you can't control it when somebody "calls in".

> 
> > [...] I assume by "local-tag" you mean the From tag?
> 
>   Well, when I send a request I put the localuri/localtag as the from
> and the remoteuri/remotetag as the to.  When I receive an initial
> request, I populate the localuri from the to and add a localtag, and the
> remoteuri/remotetag is taken from the from.
> 
>   I don't like calling them 'To tags' or 'From tags' because it gets
> confusing.

I find the designation local/remote tags confusing...

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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From: Billy Biggs <Billy_Biggs@3com.com>
To: "Moloney, Bill" <BMoloney@sonusnet.com>
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Subject: Re: [SIP] Registration Question
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Moloney, Bill (BMoloney@sonusnet.com):

> I have a question on registration -
> [...]
> I believe that I could accomplish this by having B respond to the
> registration with a 301 Moved Permanently, including a Contact header
> for C in the response.)
> 
> The question is this: Does this redirection only impact future
> registration requests, or does it impact all future requests - such as
> INVITE?

  The redirection only impacts the specific registration transaction and
(optionally) future registration transactions.

  Requests such as INVITE go to their destination directly, and have
nothing to do with the registration.

> What I am trying to accomplish is to have A direct all of its future
> requests (not just registration) to the new proxy server (C). Will the
> redirect that I mention above accomplish this?

  This is called an outbound proxy, where even though the message is
directed to, say, 'sip:5551212@sip.billybiggs.com', you send the message
to 'my.local.proxy.com' which then forwards it on.

  The outbound proxy is configured directly on the SIP end device, or it
can receive it from DHCP.  There is no way to signal using REGISTER or
whatever that the proxy should be used as an outbound proxy, nor would
it make sense to provide such signalling.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Tue, 21 Nov 2000 12:02:41 -0600

Henning G. Schulzrinne (hgs@cs.columbia.edu):

> >   What if A and B are the same URI?  How do I tell them apart? 
> 
> Are you saying that A wants to talk to himself? How else would they be
> the same under normal circumstances? A gateway may want to connect two
> ports on the same box through the network, but they would have
> different phone numbers, i.e., different From/To/Request-URIs.

  Yes, A wants to talk to himself.  This is a perfectly reasonable
diagnostic to see if my registration is active.  We do it routinely
here.  I don't see why it is acceptable to have it fail.

  I mean, you can telnet to yourself, right?

  What happens when you SUBSCRIBE to your own presence information so
you can watch how you appear online?  Those NOTIFYs have to map
correctly somewhere.

> I also don't see that if you do normal loop detection, you couldn't
> handle calls to yourself. Responses and ACKs go to the Via or Contact,
> respectively.

  Well, UAs don't forward requests and therefore don't do loop
detection.  Sure you might properly handle responses to the single
request, but the rest of the requests in that call will fail.

> If anything, we're probably moving away from overloading the call-id,
> primarily because you can't control it when somebody "calls in".

  Great.  I'd also love to hear what you think of Replaces.
  http://www.sip-happens.com/replaces/

> > > [...] I assume by "local-tag" you mean the From tag?
> > 
> >   Well, when I send a request I put the localuri/localtag as the
> > from and the remoteuri/remotetag as the to.  When I receive an
> > initial request, I populate the localuri from the to and add a
> > localtag, and the remoteuri/remotetag is taken from the from.
> > 
> >   I don't like calling them 'To tags' or 'From tags' because it gets
> > confusing.
> 
> I find the designation local/remote tags confusing...

  From an implementation point of view it isn't.  When you receive a SIP
message, you need to first match it to see if its related to any
currently active call leg ends.  At this point we don't care if its a
request or response, so To/From becomes ambiguous.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Tue, 21 Nov 2000 13:20:08 -0500

>  The outbound proxy is configured directly on the SIP end device, or it
>can receive it from DHCP.  There is no way to signal using REGISTER or
>whatever that the proxy should be used as an outbound proxy, nor would
>it make sense to provide such signalling.

I think it does make sense in certain instances. For example, the outbound
proxy server is getting overloaded, is going down for scheduled
maintainence, or an additional server is being added. Rather than needing to
go to each SIP end device, the proxy server would be able to redirect the
end device.

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Date: Tue, 21 Nov 2000 12:28:28 -0600

Moloney, Bill (BMoloney@sonusnet.com):

> >  The outbound proxy is configured directly on the SIP end device, or
> >  it can receive it from DHCP.  There is no way to signal using
> >  REGISTER or whatever that the proxy should be used as an outbound
> >  proxy, nor would it make sense to provide such signalling.
> 
> I think it does make sense in certain instances. For example, the
> outbound proxy server is getting overloaded, is going down for
> scheduled maintainence, or an additional server is being added. Rather
> than needing to go to each SIP end device, the proxy server would be
> able to redirect the end device.

  Your outbound proxy should be a DNS name, not an IP address.  So, just
change the DNS entry.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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The proposed new Torture Test messages have been added to the 
main SIP web site:

http://www.cs.columbia.edu/~hgs/sip/bakeoff/testmsg.html

The new ones are test21.txt onwards. They should also appear in 
the new verion of the SIP Message Flows draft shortly.

SIP bakeoff entrants are strongly encouraged to run these tests. 
In particular entrants to advanced scenario testing at bakeoffs 
will be expected to have passed all torture tests. 

Comments, corrections etc welcome.
Neil.
-- 
Ubiquity Software Corporation, UK        http://www.ubiquity.net

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Jonathan Rosenberg wrote:
> 

> > The A record is only used if there is a port number other
> > than 5060 ---- so
> > example.com and example.com:5060 both use the SRV record. (Of
> > course this may
> > result in different machines actually being contacted, but
> > that's a different
> > matter entirely, not to do with A records).
> 
> This is an error, I believe, resulting in the evolution of this text. We
> modified the SRV procedure so that example.com and example.com:5060 were,
> indeed, pointing to the same server, but didn't update the text on top which
> pointed out the problem.

This should already be gone from the -02 draft draft.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Date: Tue, 21 Nov 2000 20:08:49 +0100

In test30.txt if the Expires value is in absolute formate,
then a Date header should be supplied too. Otherwise a UA
that has no real-time clock will not be able to tell whether
this is in the future or past. Actually I think the Expires
value should be compared with the Date header if available
since the UA on the other end may have an incorrect notion
of time...

Also, I am still lacking a test for this kind of To/From header:

	To: sip:pkj@axis.com ;tag=123456789@example.com

and make sure the tag does not end up as a URL parameter...

//Peter

> -----Original Message-----
> From: Neil Deason [mailto:ndeason@ubiquity.net]
> Sent: 21 November 2000 19:30
> To: Sip List
> Subject: [SIP] More Torture Test Messages Available
> 
> The proposed new Torture Test messages have been added to the 
> main SIP web site:
> 
> http://www.cs.columbia.edu/~hgs/sip/bakeoff/testmsg.html
> 
> The new ones are test21.txt onwards. They should also appear in 
> the new verion of the SIP Message Flows draft shortly.
> 
> SIP bakeoff entrants are strongly encouraged to run these tests. 
> In particular entrants to advanced scenario testing at bakeoffs 
> will be expected to have passed all torture tests. 
> 
> Comments, corrections etc welcome.
> Neil.
> -- 
> Ubiquity Software Corporation, UK        http://www.ubiquity.net

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From sip-admin@lists.bell-labs.com  Tue Nov 21 15:47:14 2000
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Date: Tue, 21 Nov 2000 10:37:43 -0800
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Hi,
There are some problems in the compilation of the vovida sipstack and
the ua on RedHat 7.0 . Here are a few changes required for the fix

1. If you get a problem with the forward declaration of the class remove

it and include the header file
2. If there is a problem with compilation due to the use of vector then
include <deque>
    This can occur in a few files like:
 -  SipAccept
 -  SipAcceptEncoding
 -  SipAcceptLanguage
     to name a few

If there are any other issues in compilation please let me know.

Thanks
Amit Choksi


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1 Deprecate Early Media

  Listed below are some severe problems with early media.  These
  problems can be solved by connecting (200 OK) before sending media.
  This creates a billing problem for PSTN gateways which can be solved
  by either:

  1. Separating the billing system from the SIP signalling.  Everyone
     sends detailed CDRs via non-SIP means.

  2. Changing the billing contract to charge for ringback and error
     sessions, just like AT&T cell phones do today.

  The issues listed below are causing us immediate problems.  I focus on
  PSTN gateways, but the arguments apply to other uses of early media.

  While this has come up before on the list and been ignored, I feel
  that early media should be deprecated quickly to avoid a larger mess.

  Comments appreciated.


2 History of Early Media

  The early media concept originated with the need for PSTN-style inband
  altering.  The primary reason for not initiating a full SIP session was
  for billing reasons.  As Steve Donovan wrote, "[we] Need [the] ability
  to have media session setup prior to network-to-network charging." [1]

  See [2] for more historical context.

  Apart from that, early media is convenient for gateways as it makes
  the mapping from Q.931 to SIP more direct.


3 Current Major Problems

  3.1 Services Which Use Early Two-Way Audio/Signalling
  3.2 Gateways Which Can't Determine Call Failure
  3.3 Handling by Forking Proxies which Fork the Call in Parallel
  3.4 Handling by Forking Proxies who Sequentially Fork the Call
  3.5 Inability to Provide User Recourse
  3.6 Difficult Negotiation of Media


3.1 Services Which Use Early Two-Way Audio/Signalling

  Some PSTN services, usually 800-number services, try and avoid being
  charged extra by deferring acceptance of a call as long as they can,
  relying on still receiving DTMF digits over the half-connected call.

  This was explained by Cliff Harris [3] as follows:

  "I believe that Q.931 does allow cut-through audio to be set up during
   the "overlapped dialing" phase of the call. In this scenario, while
   the caller is still issuing INFO messages containing the dialed
   digits, the callee sends a PROGRESS message indicating the presence
   of cut-through audio. The caller sets up an audio path in the
   backwards direction only.  Note that in this situation, the DTMF
   digits are being sent in the signalling path, not in the media path."

  Examples: 1-800-CALL-ATT, Verizon Wireless, American Airlines/United
            Reservations, and many other 1-800 IVR systems.

  As far as we can tell, there is no way for the gateway to identify
  this problem.


3.2 Gateways Which Can't Determine Call Failure

  Based on my experience, some PSTN gateways seem to have problems
  determining when the call has failed.  Henning mentioned this on the
  list [4]:

  "This goes back to an earlier, long-ago discussion whether it's better
   to use 4xx/5xx for announcements, but if I remember right, the
   conclusion was that this wasn't always feasible since gateways
   wouldn't be able to distinguish failure announcements from ring tone.
   (I thought the announcement tri-tone was for stuff like that...)"

  In this sort of case, UAs which delay sending SDP until the ACK will
  think the call is still ringing when really a busy signal is being
  played.


3.3 Handling by Forking Proxies which Fork the Call in Parallel

  The most obvious problem is when a call forks in parallel to two user
  agents, each which send early media.  As Henning mentioned [4]:

  "If "early media" is an announcement like "the number you have dialed
   has been changed. The new number is ...", mixing with ring tone from
   another destination would be very confusing."

  The UA can't cancel the branches, and the proxy can't be expected to
  cancel branches which return SDP.


3.4 Handling by Forking Proxies which Sequentially Fork the Call

  The problem here is when gateways can't tell that they're sending an
  early busy signal.  The user dials a number, hears a busy signal, and
  hangs up before the proxy has a chance to time out and switch to the
  next contact.

  Even worse, you call through a proxy which first forks to one of these
  interactive IVRs, asking you for your PIN number.  As you're typing it
  in, the proxy cancels the branch and moves on to the next contact.


3.5 Inability to Provide User Recourse

  Without using reliable provisional responses, there is no method to
  allow a user agent to place early media on hold, redirect multiple
  incoming sessions, or otherwise change its session description short
  of shutting down the call request.


3.6 Difficult Negotiation of Media

  A complete system which relies on early media must support reliable
  provisional responses in order for RTCP to work properly.  Other
  description protocols may have more difficult problems.


4 References

  [1] http://www.ietf.org/proceedings/99jul/slides/mmusic-sip183-99jul/

  [2] http://www.cs.columbia.edu/sip/drafts/draft-ietf-sip-183-00.txt

  [3] http://lists.bell-labs.com/pipermail/sip/2000q3/002352.html

  [4] http://lists.bell-labs.com/pipermail/sip/2000q3/002297.html


-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Hi,
Please ignore my previous mail about the compilation on redhat 7, I am
sorry for that.


Thanks
Amit Choksi




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Billy Biggs wrote:
> 
> 1 Deprecate Early Media
> 
>   Listed below are some severe problems with early media.  These
>   problems can be solved by connecting (200 OK) before sending media.
>   This creates a billing problem for PSTN gateways which can be solved
>   by either:
> 
>   1. Separating the billing system from the SIP signalling.  Everyone
>      sends detailed CDRs via non-SIP means.
> 
>   2. Changing the billing contract to charge for ringback and error
>      sessions, just like AT&T cell phones do today.
> 
>   The issues listed below are causing us immediate problems.  I focus on
>   PSTN gateways, but the arguments apply to other uses of early media.

The sooner, the better... However, it should be noted that "early media"
has an additional advantage in that it does allow the caller to figure
out automatically when to start sending audio. If you had an automatic
caller (as is common for various telemarketers or when Al Gore calls you
at home before elections), it would not work. Not that I would consider
this a major problem, as this appears only when calling IVR systems
which presumably don't buy aluminum siding and are too dumb to even
create a pregnant chad.

For those cases, where the gateway can determine when the call has been
connected, sending a re-INVITE to update the sendonly/recvonly
designation would seem to be preferable.

For billing, I'm not sure that early media is particularly useful except
possibly in the case of bridging ISUP islands. The direction indication
in SDP would help here, too. For gateways, the gateway presumably knows
how long the call into the PSTN was and can thus bill accordingly. I
assume gateways are smart enough to recognize the three-tone sequence
even if they are not ISUP connected. For IP billing, billing based on
resource reservation is the only long-term feasible approach in any
event.

A related question is whether any SIP implementations actually handle
early media. It seems harmless to have in pure IP systems, if it gets
never exercised by gateways.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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How do you change codec in a SIP call without disconnecting the established
call?

Mo Zonoun

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<P><FONT SIZE=2>How do you change codec in a SIP call without disconnecting the established call?</FONT>
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(re)Invite with a new session description
=20
Pascal Dor=E9

-----Original Message-----
From: Mo Zonoun [mailto:mzonoun@nortelnetworks.com]
Sent: Tuesday, November 21, 2000 1:47 PM
To: IETF SIP (E-mail)
Subject: [SIP] Changing Codec



How do you change codec in a SIP call without disconnecting the =
established
call?=20

Mo Zonoun=20


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class=3D732580222-21112000></SPAN></FONT>&nbsp;</DIV>
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Dor=E9</SPAN></FONT></DIV>
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Subject: [SIP] I-Ds
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Date: Tue, 21 Nov 2000 14:19:58 -0800

Two new I-Ds have been submitted with relevance
to the SIP, AAAarch, and RAP working groups.


	Title		: QoS and AAA Usage with SIP Based IP Communications
	Author(s)	: G. Gross, H. Sinnreich, D. Rawlins, S. Thomas
	Filename	: draft-gross-sipaq-00.txt
	Pages		: 23
	Date		: 20-Nov-00
	
This document specifies the architecture, protocols and messages for 
SIP based IP communications between Internet domains in support of 
QoS and AAA. Detailed message flows and message contents are 
discussed and specified. AAA requirements including inter-domain and 
user authorization in mobile and non-mobile environments are 
addressed. A solution is proposed where session setup and teardown 
are linked with QoS and AAA signaling using AAA policy servers and 
clearinghouses.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-gross-sipaq-00.txt


	Title		: COPS Usage for SIP
	Author(s)	: G. Gross, D. Rawlins, H. Sinnreich, S. Thomas
	Filename	: draft-gross-cops-sip-00.txt
	Pages		: 13
	Date		: 20-Nov-00
	
The development of QoS enhanced IP communication requires an 
infrastructure that supports AAA services. General accounting and 
settlement on the intra-domain level and inter-domain level are 
necessary. As part of the solution to these requirements, this 
document describes a new COPS outsourcing extension for SIP . A 
description of COPS objects that have special meaning in the SIP 
environment is provided. The usage of the COPS extension for SIP is 
given for clients and servers.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-gross-cops-sip-00.txt




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From: Billy Biggs <Billy_Biggs@3com.com>
To: "Henning G. Schulzrinne" <hgs@cs.columbia.edu>
Cc: SIP List <sip@lists.bell-labs.com>
Subject: Re: [SIP] Deprecate Early Media
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Date: Tue, 21 Nov 2000 16:36:21 -0600


  Henning,

  Thanks for your support. :)

> A related question is whether any SIP implementations actually handle
> early media.

  You missed my urgency.  For the past 11 months I have been unable to
use 1-800 IVRs from the only phone on my desk, and get to see my phone
say 'Ringing' when I'm clearly hearing a busy signal.

  Quite frankly, I'm fed up.

  Yes, phones already receive it, and gateways already send it.

> [...] It seems harmless to have in pure IP systems, if it gets never
> exercised by gateways.

  I'm not sure what you mean.  Without any gateways, you still have all
the forking problems, the need for reliable provisional responses, ...
It's still a big mess.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Tue, 21 Nov 2000 17:46:55 -0500

Typo in test case 38:

Contact: <sip:+1-972-555-2222@gw1.wcom.com>;user=phone

Here, user=phone is actually a header parameter, opposite to what the
description says. You probably want the closing angle bracket after the
parameter.

Test case 41:

I'd argue that this message should be rejected with a 400. Section 4.1 lists
SIP version as a literal and doesn't say anything about ignoring leading
zeros. As an example of why rejecting it is a good idea, consider version
numbers like 2.1 and 2.01 - they are unlikely to refer to the same version.

Interestingly, Appendix C mentions that literals are case-insensitive. I
think it is reasonable to explicitly state that the SIP-Version is
case-sensitive. This would make Via definition consistent: it currently
defines protocol-name as either literal "SIP" (case-insensitive) or token
(case-sensitive).

Thank you,
Igor Slepchin


> -----Original Message-----
> From: Neil Deason [mailto:ndeason@ubiquity.net]
> Sent: Tuesday, November 21, 2000 1:30 PM
> To: Sip List
> Subject: [SIP] More Torture Test Messages Available
> 
> 
> The proposed new Torture Test messages have been added to the 
> main SIP web site:
> 
> http://www.cs.columbia.edu/~hgs/sip/bakeoff/testmsg.html
> 
> The new ones are test21.txt onwards. They should also appear in 
> the new verion of the SIP Message Flows draft shortly.
> 
> SIP bakeoff entrants are strongly encouraged to run these tests. 
> In particular entrants to advanced scenario testing at bakeoffs 
> will be expected to have passed all torture tests. 
> 
> Comments, corrections etc welcome.
> Neil.
> -- 
> Ubiquity Software Corporation, UK        http://www.ubiquity.net
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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Subject: [SIP] New I-D: draft-hamer-sip-session-auth-00.txt
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Date: Tue, 21 Nov 2000 10:37:09 -0500

This message is in MIME format. Since your mail reader does not understand
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Folks,

My colleague and I have submitted the following draft to the SIP WG and
would appreciate any comments
or questions prior to the next meeting in San Diego.

Cheers,
Louis-Nicolas Hamer


-----Original Message-----
From:	Internet-Drafts@ietf.org [SMTP:Internet-Drafts@ietf.org]
Sent:	Monday, November 13, 2000 6:40 AM
To:	IETF-Announce
Subject:	I-D ACTION:draft-hamer-sip-session-auth-00.txt

A New Internet-Draft is available from the on-line Internet-Drafts
directories.


	Title		: Session setup with media authorization
	Author(s)	: L. Hamer, B. Gage
	Filename	: draft-hamer-sip-session-auth-00.txt
	Pages		: 23
	Date		: 10-Nov-00
	
Current proposals dealing with authorization of media streams for 
multimedia services like IP telephony and video assume a pre- 
established relationship between elements of the network (e.g. 
session managers, edge routers, policy servers and end hosts). In 
some environments, however, such pre-established relationships may 
not exist either due to the complexity of creating these 
associations a priori (e.g. in a network with many elements), or due 
to the different business entities involved (e.g. service provider 
and access provider), or due to the dynamic nature of these 
associations (e.g. in a mobile environment). 
In this document, we describe scenarios where there is no pre-
established relationship between entities and describe mechanisms 
for exchanging information between network elements in order to 
authorize the use of resources for a service and to co-ordinate 
actions between the session and bearer control planes.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-hamer-sip-session-auth-00.txt

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-hamer-sip-session-auth-00.txt".

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or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
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charset=3Diso-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2652.35">
<TITLE>New I-D: draft-hamer-sip-session-auth-00.txt</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D1 FACE=3D"Arial">Folks,</FONT>
</P>

<P><FONT SIZE=3D1 FACE=3D"Arial">My colleague and I have submitted the =
following draft to the SIP WG and would appreciate any comments</FONT>
<BR><FONT SIZE=3D1 FACE=3D"Arial">or questions prior to the next =
meeting in San Diego.</FONT>
</P>

<P><FONT SIZE=3D1 FACE=3D"Arial">Cheers,</FONT>
<BR><FONT SIZE=3D1 FACE=3D"Arial">Louis-Nicolas Hamer</FONT>
</P>
<BR>

<P><FONT SIZE=3D1 FACE=3D"Arial">-----Original Message-----</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">From:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Internet-Drafts@ietf.org =
[SMTP:Internet-Drafts@ietf.org]</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Monday, November 13, 2000 6:40 AM</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">IETF-Announce</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT>=
</B> <FONT SIZE=3D1 FACE=3D"Arial">I-D =
ACTION:draft-hamer-sip-session-auth-00.txt</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">A New Internet-Draft is available from =
the on-line Internet-Drafts directories.</FONT>
</P>
<BR>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">Title&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : Session setup with media =
authorization</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : L. =
Hamer, B. Gage</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
draft-hamer-sip-session-auth-00.txt</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">Pages&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 23</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">Date&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 10-Nov-00</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR><FONT SIZE=3D2 FACE=3D"Arial">Current proposals dealing with =
authorization of media streams for </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">multimedia services like IP telephony =
and video assume a pre- </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">established relationship between =
elements of the network (e.g. </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">session managers, edge routers, =
policy servers and end hosts). In </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">some environments, however, such =
pre-established relationships may </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">not exist either due to the =
complexity of creating these </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">associations a priori (e.g. in a =
network with many elements), or due </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">to the different business entities =
involved (e.g. service provider </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">and access provider), or due to the =
dynamic nature of these </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">associations (e.g. in a mobile =
environment). </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">In this document, we describe =
scenarios where there is no pre-</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">established relationship between =
entities and describe mechanisms </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">for exchanging information between =
network elements in order to </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">authorize the use of resources for a =
service and to co-ordinate </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">actions between the session and =
bearer control planes.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">A URL for this Internet-Draft =
is:</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-hamer-sip-session-auth=
-00.txt" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-hamer-sip-se=
ssion-auth-00.txt</A></FONT></U>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Internet-Drafts are also available by =
anonymous FTP. Login with the username</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&quot;anonymous&quot; and a password =
of your e-mail address. After logging in,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">type &quot;cd internet-drafts&quot; =
and then</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">&quot;get =
draft-hamer-sip-session-auth-00.txt&quot;.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">A list of Internet-Drafts directories =
can be found in</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://www.ietf.org/shadow.html" =
TARGET=3D"_blank">http://www.ietf.org/shadow.html</A></FONT></U><FONT =
SIZE=3D2 FACE=3D"Arial"> </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">or</FONT><U> <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt" =
TARGET=3D"_blank">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</A></FONT></=
U>
</P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Arial">Internet-Drafts can also be obtained =
by e-mail.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Send a message to:</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">mailserv@ietf.org.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">In the body type:</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">&quot;FILE =
/internet-drafts/draft-hamer-sip-session-auth-00.txt&quot;.</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<BR><FONT SIZE=3D2 FACE=3D"Arial">NOTE:&nbsp;&nbsp; The mail server at =
ietf.org can return the document in</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">MIME-encoded form by using the &quot;mpack&quot; =
utility.&nbsp; To use this</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">feature, insert the command &quot;ENCODING mime&quot; =
before the &quot;FILE&quot;</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">command.&nbsp; To decode the response(s), you will need =
&quot;munpack&quot; or</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">a MIME-compliant mail reader.&nbsp; Different =
MIME-compliant mail readers</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">exhibit different behavior, especially when dealing =
with</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">&quot;multipart&quot; MIME messages (i.e. documents =
which have been split</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">up into multiple messages), so check your local =
documentation on</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2 =
FACE=3D"Arial">how to manipulate these messages.</FONT>
</P>
<BR>

</BODY>
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From: Billy Biggs <Billy_Biggs@3com.com>
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Subject: Re: [SIP] Changing Codec
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Pascal Doré (pdore@mediatrix.com):

> > How do you change codec in a SIP call without disconnecting the
> > established call? 
>
> (re)Invite with a new session description

  Actually, it depends on if you are talking about send codec or receive
codec.

  If you want to change the codec you are sending, then as long as the
other end indicated support, you can change any time you like.  For
example, if I send you SDP that says:

   m=audio 8552 RTP/AVP 0 1 2

  Then you can freely switch between sending audio using codec 0, 1 or
2.  If you're feeling cruel, you can use a different codec for every
packet.


  If you want to change which codec you are receiving from the far end,
the best you can do is offer a different set of choices using a
re-INVITE.  So, first maybe you send:

   m=audio 8552 RTP/AVP 0 1 2

  And the far end is sending you codec 1, but you only want to receive
codec 0, then you can always re-INVITE with:

   m=audio 8552 RTP/AVP 0

  And force it to use either codec 0 or nothing.  Be prepared though
that if the remote end can't send codec 0, it might tear down the call
(or at best audio won't flow).

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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From sip-admin@lists.bell-labs.com  Tue Nov 21 21:50:24 2000
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Lars Berggren wrote:
> 

> 
> I agree with Billy and would suggest that tags should be unique to each
> instance of a user agent and maybe even unique to a call.

They are already unique to each instance of a user agent, as described.
If you have two SIP UAs running on the same host (on different ports,
presumably), they will have different tags. The same "brand" of UA on
different hosts will have different UAs. If it were useful, having
different tags for From and To from the same UA may even be useful,
although it would be nice to have a scenario for that.

I have yet to understand why re-using the same tag across restarts hurts
anything. After all, you're not worried about confusing messages when a
SIP UA is revived.

Maybe we should look at the self-call in a bit more detail (I'm assuming
From tags here):

INVITE self
To: self
From: self;tag=T
Via: self

This will be "filed" under to=self/*, from=self/T. Any incoming request
will have their From/To reversed and mapped. As long as the two tags

The response will be

200 OK
To: self;tag=T
From: self;tag=T

This will map to the same entry. A BYE will then use

BYE self
To: self;tag=T
From: self;tag=T

The major problem occurs probably not so much because of mapping, but
rather since this is treated as a re-INVITE while the first one is still
pending.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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From sip-admin@lists.bell-labs.com  Tue Nov 21 22:08:02 2000
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Igor Slepchin wrote:
> 
> Typo in test case 38:
> 
> Contact: <sip:+1-972-555-2222@gw1.wcom.com>;user=phone
> 
> Here, user=phone is actually a header parameter, opposite to what the
> description says. You probably want the closing angle bracket after the
> parameter.

Fixed.

> 
> Test case 41:
> 
> I'd argue that this message should be rejected with a 400. Section 4.1 lists
> SIP version as a literal and doesn't say anything about ignoring leading
> zeros. As an example of why rejecting it is a good idea, consider version
> numbers like 2.1 and 2.01 - they are unlikely to refer to the same version.

Version numbers are not real numbers. Typically, you have
2.1,2.2,...,2.9,2.10,2.11, which isn't exactly numerical progression.
Well, the zero issue isn't quite as clear-cut, as RFC 2616 says:

   The version of an HTTP message is indicated by an HTTP-Version field
   in the first line of the message.

       HTTP-Version   = "HTTP" "/" 1*DIGIT "." 1*DIGIT

   Note that the major and minor numbers MUST be treated as separate
   integers and that each MAY be incremented higher than a single digit.
   Thus, HTTP/2.4 is a lower version than HTTP/2.13, which in turn is
   lower than HTTP/12.3. Leading zeros MUST be ignored by recipients and
   MUST NOT be sent.

I also just tried 

GET / http/1.1
Host: www.cs.columbia.edu

(which is a standard Apache server)

and this works fine.



> 
> Interestingly, Appendix C mentions that literals are case-insensitive. I
> think it is reasonable to explicitly state that the SIP-Version is
> case-sensitive. This would make Via definition consistent: it currently
> defines protocol-name as either literal "SIP" (case-insensitive) or token
> (case-sensitive).

Wait, if protocol is defined as, say

protocol-token = "FOO" | "BAR"

it would be CI. This is consistent with the use in RFC 2616, where all
tokens are defined as CI (such as character-set, content-coding,
transfer-coding, type, subtype, etc.


-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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From: Billy Biggs <Billy_Biggs@3com.com>
To: "Henning G. Schulzrinne" <hgs@cs.columbia.edu>
Cc: Lars Berggren <lars@intertex.se>, Jo Hornsby <jhornsby@ubiquity.net>,
        Alan Johnston <alan.johnston@wcom.com>,
        Henry Chen <hjlechen@cisco.com>, SIP List <sip@lists.bell-labs.com>
Subject: Re: [SIP] From tags (was: I-D ACTION:draft-ietf-sip-service-examples-00.txt)
Message-ID: <20001121214009.A12322@div8.net>
References: <Pine.LNX.4.21.0011220050390.4670-100000@felix.intertex.se> <3A1B33DB.3D30D366@cs.columbia.edu>
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Date: Tue, 21 Nov 2000 21:40:09 -0600

Henning G. Schulzrinne (hgs@cs.columbia.edu):

> I have yet to understand why re-using the same tag across restarts
> hurts anything. After all, you're not worried about confusing messages
> when a SIP UA is revived.
> 
> Maybe we should look at the self-call in a bit more detail
> [...]
> The major problem occurs probably not so much because of mapping, but
> rather since this is treated as a re-INVITE while the first one is
> still pending.

  Sorry I don't quite understand what you mean.  It looks like your
email got a bit truncated.

  To help, I'm going to describe a scenario where a UA calls itself, and
that UA always uses a local tag (To or From) of 7.  Hopefully this will
help to clarify the problem.


   I decide to call myself.  I create a call-leg struct to represent it.
Let's call it "Line 1".  I send out the initial INVITE:

  [From Line 1]
  INVITE sip:me@proxy
  Via: SIP/2.0/UDP my.ip
  To: <sip:me@proxy>
  From: <sip:me@proxy>;tag=7
  Call-ID: 5

  This eventually loops back to me.  Now I see coming in:

  INVITE sip:me@proxy
  Via: SIP/2.0/UDP some.ip
  To: <sip:me@proxy>
  From: <sip:me@proxy>;tag=7
  Call-ID: 5

  I check and see there is no To-tag, so this is a new incoming call.  I
create a call struct to represent it, called "Line 2".  Line 2 now
begins ringing.  I respond with a 180 and the user switches lines and
answers the incoming call:

  [From Line 2]
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP some.ip
  To: <sip:me@proxy>;tag=7
  From: <sip:me@proxy>;tag=7
  Call-ID: 5

  When I receive this response, I find the related pending transaction.
Arguably, I could first find which call-leg it is destined to (and
immediately have a problem), but say that I do transaction mapping first
so we don't break yet.  I match the OK and send an ACK:

  [From Line 1]
  ACK sip:me@contactaddr
  Via: SIP/2.0/UDP my.ip
  To: <sip:me@proxy>;tag=7
  From: <sip:me@proxy>;tag=7
  Call-ID: 5

  At which point the other pending transaction also correctly matches
this ACK to it.

  Now, on Line 2, I decide to hang up.  I've had enough of talking to
myself.  So, I send a BYE:

  [From Line 2]
  BYE sip:me@contact.addr
  Via: SIP/2.0/UDP my.ip
  To: <sip:me@proxy>;tag=7
  From: <sip:me@proxy>;tag=7
  Call-ID: 5

  I now see this incoming BYE:

  BYE sip:me@contact.addr
  Via: SIP/2.0/UDP some.ip
  To: <sip:me@proxy>;tag=7
  From: <sip:me@proxy>;tag=7
  Call-ID: 5

  Which one of the calls is it for?  Is it for the line which originated
the call, or the line which received the call?

  Using unique tags per call avoids any confusion even in this case, and
also protects the UA against clients which always use the same tag and
happen to make two calls with the same Call-ID.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Tue, 21 Nov 2000 21:26:57 -0800

The colon in the user part of Test 36 (the long telephone subscriber) needs
to be quoted. Perhaps you could also put a similar long telephone-subscriber
URL in the Contact - this would test whether the URL was correctly
reproduced if the call is accepted.

Cheers,

Robert.

> -----Original Message-----
> From: Neil Deason [mailto:ndeason@ubiquity.net]
> Sent: Wednesday, November 22, 2000 2:30 AM
> To: Sip List
> Subject: [SIP] More Torture Test Messages Available
> 
> 
> The proposed new Torture Test messages have been added to the 
> main SIP web site:
> 
> http://www.cs.columbia.edu/~hgs/sip/bakeoff/testmsg.html
> 
> The new ones are test21.txt onwards. They should also appear in 
> the new verion of the SIP Message Flows draft shortly.
> 
> SIP bakeoff entrants are strongly encouraged to run these tests. 
> In particular entrants to advanced scenario testing at bakeoffs 
> will be expected to have passed all torture tests. 
> 
> Comments, corrections etc welcome.
> Neil.
> -- 
> Ubiquity Software Corporation, UK        http://www.ubiquity.net
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
To: "'Hisham Khartabil'" <hisham.khartabil@hotsip.com>,
        sip@lists.bell-labs.com
Subject: RE: [SIP] Contact header is a 200 response to SUBSCRIBE for IMPP
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Date: Wed, 22 Nov 2000 03:52:55 -0500



 

> -----Original Message-----
> From: Hisham Khartabil [mailto:hisham.khartabil@hotsip.com]
> Sent: Tuesday, November 21, 2000 8:44 AM
> To: sip@lists.bell-labs.com
> Subject: RE: [SIP] Contact header is a 200 response to SUBSCRIBE for
> IMPP
> 
> 
> Hi,
> I'm posting the following again since there was no reply the 
> first time
> around
> 
> Regards,
> Hisham Khartabil
> Hotsip Finland
> www.hotsip.com
> 
> -----Original Message-----
> From: sip-admin@lists.bell-labs.com 
> [mailto:sip-admin@lists.bell-labs.com]On
> Behalf Of Hisham Khartabil
> Sent: Thursday, 9 November 2000 11:41 AM
> To: sip@lists.bell-labs.com
> Subject: [SIP] Contact header is a 200 response to SUBSCRIBE for IMPP
> 
> Section 4.2 of the presence draft has the following examples:
> 
> SUBSCRIBE sip:jdrosen@dynamicsoft.com SIP/2.0
> Via: SIP/2.0/UDP userpc.example.com
> To: sip:jdrosen@dynamicsoft.com
> From: sip:user@example.com
> Contact: sip:user@userpc.example.com
> Call-ID: knsd08alas9dy@3.4.5.6
> CSeq: 1 SUBSCRIBE
> Content-Length: 0
> 
> 
> 
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP presenceserver.dynamicsoft.com
> Via: SIP/2.0/UDP userpc.example.com
> To: sip:jdrosen@dynamicsoft.com
> From: sip:user@example.com
> Contact: sip:user@userpc.example.com
> Record-Route: 
> sip:jdrosen@example.com;maddr=presenceserver.dynamicsoft.com
> Expires: 3600
> Call-ID: knsd08alas9dy@3.4.5.6
> CSeq: 1 SUBSCRIBE
> Content-Type: application/xpidf+xml
> Content-Length: 287
> 
> <?xml version="1.0"?>
> <!DOCTYPE presence
>  PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
> <presence>
>  <presentity uri="sip:jdrosen@dynamicsoft.com;method="SUBSCRIBE">
>   <atom id="779js0a98">
>    <address uri="sip:jdrosen@dynamicsoft.com;method=INVITE">
>     <status status="open"/>
>    </address>
>   </atom>
>  </presentity>
> </presence>
> 
> 
> Then, it goes on to say:
> "Notice that the response has mirrored many fields from the
> request,including the To, From, Call-ID, CSeq, and Via 
> headers. The PA has
> also added a Contact header, providing an address where it 
> can be reached.
> The response also contains a Record-Route header. This header 
> was presumably
> inserted by the proxy (the address in the maddr header is 
> that of a presence
> server acting as a proxy), and is reflected in the 200 OK 
> response. The
> response also contains a presence document for the 
> presentity, and a Contact
> header. The Contact headers in a response to SUBSCRIBE list 
> all the current
> addresses which have been subscribed."
> 
> The example shows that the Contact header in the response 
> does actually list
> the current address which has been subscribed 
> (user@userpc.example.com).
> But I fail to see where the Contact header that the PA added 
> providing an
> address where it can be reached.

Err, we have inconsistent text here on what the meaning of Contact header is
in the 200 response. The problems relate to the dual usage of SUBSCRIBE to
have INVITE-like record-routing semantics, and REGISTER like informational
semantics. 

There are two choices:

1. We go all the way to INVITE like semantics, so that the Contact in the
response is the PA address. The Contact address in the SUBSCRIBE itself is
therefore not mirrored.
2. We maintain a hybrid, and instead of having the PA insert a Contact into
the response, the PA itself "record-routes", even though its not a proxy. 

I'd prefer (1) at this point, especially now that we have dropped the
multiple Contacts in SUBSCRIBE.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From sip-admin@lists.bell-labs.com  Wed Nov 22 04:27:03 2000
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From: "Hisham Khartabil" <hisham.khartabil@hotsip.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>, <sip@lists.bell-labs.com>
Subject: RE: [SIP] Contact header is a 200 response to SUBSCRIBE for IMPP
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Date: Wed, 22 Nov 2000 11:28:05 +0200
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I wasn't aware the "multiple contacts" has been dropped from SUBSCRIBE. Why
was the dropped?
Even then, why wouldn't the 200 responses contain the contacts of the
individual subscriptions (registration like)?

I agree with option 1 though. Knowing where to directly contact the PA is
much more useful than getting a response with contacts you subscribed to
(they should be stored somewhere on the client anyway)

Hisham

-----Original Message-----
From: sip-admin@lists.bell-labs.com [mailto:sip-admin@lists.bell-labs.com]On
Behalf Of Jonathan Rosenberg
Sent: Wednesday, 22 November 2000 10:53 AM
To: 'Hisham Khartabil'; sip@lists.bell-labs.com
Subject: RE: [SIP] Contact header is a 200 response to SUBSCRIBE for IMPP




> -----Original Message-----
> From: Hisham Khartabil [mailto:hisham.khartabil@hotsip.com]
> Sent: Tuesday, November 21, 2000 8:44 AM
> To: sip@lists.bell-labs.com
> Subject: RE: [SIP] Contact header is a 200 response to SUBSCRIBE for
> IMPP
>
>
> Hi,
> I'm posting the following again since there was no reply the
> first time
> around
>
> Regards,
> Hisham Khartabil
> Hotsip Finland
> www.hotsip.com
>
> -----Original Message-----
> From: sip-admin@lists.bell-labs.com
> [mailto:sip-admin@lists.bell-labs.com]On
> Behalf Of Hisham Khartabil
> Sent: Thursday, 9 November 2000 11:41 AM
> To: sip@lists.bell-labs.com
> Subject: [SIP] Contact header is a 200 response to SUBSCRIBE for IMPP
>
> Section 4.2 of the presence draft has the following examples:
>
> SUBSCRIBE sip:jdrosen@dynamicsoft.com SIP/2.0
> Via: SIP/2.0/UDP userpc.example.com
> To: sip:jdrosen@dynamicsoft.com
> From: sip:user@example.com
> Contact: sip:user@userpc.example.com
> Call-ID: knsd08alas9dy@3.4.5.6
> CSeq: 1 SUBSCRIBE
> Content-Length: 0
>
>
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP presenceserver.dynamicsoft.com
> Via: SIP/2.0/UDP userpc.example.com
> To: sip:jdrosen@dynamicsoft.com
> From: sip:user@example.com
> Contact: sip:user@userpc.example.com
> Record-Route:
> sip:jdrosen@example.com;maddr=presenceserver.dynamicsoft.com
> Expires: 3600
> Call-ID: knsd08alas9dy@3.4.5.6
> CSeq: 1 SUBSCRIBE
> Content-Type: application/xpidf+xml
> Content-Length: 287
>
> <?xml version="1.0"?>
> <!DOCTYPE presence
>  PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd">
> <presence>
>  <presentity uri="sip:jdrosen@dynamicsoft.com;method="SUBSCRIBE">
>   <atom id="779js0a98">
>    <address uri="sip:jdrosen@dynamicsoft.com;method=INVITE">
>     <status status="open"/>
>    </address>
>   </atom>
>  </presentity>
> </presence>
>
>
> Then, it goes on to say:
> "Notice that the response has mirrored many fields from the
> request,including the To, From, Call-ID, CSeq, and Via
> headers. The PA has
> also added a Contact header, providing an address where it
> can be reached.
> The response also contains a Record-Route header. This header
> was presumably
> inserted by the proxy (the address in the maddr header is
> that of a presence
> server acting as a proxy), and is reflected in the 200 OK
> response. The
> response also contains a presence document for the
> presentity, and a Contact
> header. The Contact headers in a response to SUBSCRIBE list
> all the current
> addresses which have been subscribed."
>
> The example shows that the Contact header in the response
> does actually list
> the current address which has been subscribed
> (user@userpc.example.com).
> But I fail to see where the Contact header that the PA added
> providing an
> address where it can be reached.

Err, we have inconsistent text here on what the meaning of Contact header is
in the 200 response. The problems relate to the dual usage of SUBSCRIBE to
have INVITE-like record-routing semantics, and REGISTER like informational
semantics.

There are two choices:

1. We go all the way to INVITE like semantics, so that the Contact in the
response is the PA address. The Contact address in the SUBSCRIBE itself is
therefore not mirrored.
2. We maintain a hybrid, and instead of having the PA insert a Contact into
the response, the PA itself "record-routes", even though its not a proxy.

I'd prefer (1) at this point, especially now that we have dropped the
multiple Contacts in SUBSCRIBE.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Date: Wed, 22 Nov 2000 09:03:48 +0000



Henning Schulzrinne wrote:

> The problem is that with new parameters, older applications won't know
> whether the parameter has a default value and what that might be.

I don't think this problem has a neat solution. I think what is really
wanted is that URLs which are semantically equivalent should
compare as equal. But if a new parameter changes the semantics
how can older applications ever make the right judgement?

> Thus,
> you can either
>
> - compare well-known parameters (RFC 2543) according to this rule and
> new parameters according to "ignore" or "unequal if one has it, the
> other doesn't" rule. This seems less than satisfactory in terms of
> robustness and simplicity.
>
> - Declare missing parameters as unequal to any existing parameter. That
> doesn't work, due to default values.
>
> - Ignore ;-parameters that are only present in one of the URLs being
> compared. This seems to work for most practical cases. Any examples
> where this would break things?

In this last case do you really mean to ignore the port number if it is not
present in both URLs?

  Stuart Wray



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Date: Wed, 22 Nov 2000 04:29:33 -0500



 

> -----Original Message-----
> From: Hisham Khartabil [mailto:hisham.khartabil@hotsip.com]
> Sent: Wednesday, November 22, 2000 4:28 AM
> To: Jonathan Rosenberg; sip@lists.bell-labs.com
> Subject: RE: [SIP] Contact header is a 200 response to SUBSCRIBE for
> IMPP
> 
> 
> I wasn't aware the "multiple contacts" has been dropped from 
> SUBSCRIBE.

That thread may have been on sip-events. 

> Why
> was the dropped?

It doesn't work with record-routing.

> Even then, why wouldn't the 200 responses contain the contacts of the
> individual subscriptions (registration like)?

Because SUBSCRIBE is more like INVITE than REGISTER.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
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Date: Wed, 22 Nov 2000 04:40:23 -0500

Contact should be sent in every INVITE. Due to crash and restart, what is a
re-INVITE for one user is a new INVITE for another.

If the Contact is different, yes, it should be updated in the Route set.
This is needed for certain proposed mobility applications, and I suspect has
some other very useful properties ala third party call control.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: David Shrader [mailto:dshrader@master-consultant.com]
> Sent: Thursday, November 09, 2000 11:17 AM
> To: SIP List
> Subject: [SIP] Contact: header in re-INVITE
> 
> 
> Table 4 indicates that the Contact: header is mandatory "m" 
> in the INVITE
> request. Is this necessary for a re-INVITE message when the 
> two endpoints
> have already established communication.
> 
> I think this is similar to the Record-Route mechanism in 
> which the route is
> recorded once and applies to subsequent requests on the same 
> call leg. For
> that matter, if the Contact were to change. Do we change the
> Record-Route/Route as well?
> 
> David
> 
> ----------------------------
> David Shrader
> Master Consultant, Inc.
> dshrader@master-consultant.com
> http://www.EyeForTheFuture.com
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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Date: Wed, 22 Nov 2000 05:20:35 -0500

Finally getting around to this.... sorry for the delay:

 

> -----Original Message-----
> From: Robin Escolme [mailto:Robin.Escolme@marconi.com]
> Sent: Thursday, November 02, 2000 10:45 AM
> To: Jonathan Rosenberg
> Cc: sip@lists.bell-labs.com
> Subject: Re: [SIP] I-D ACTION:draft-ietf-sip-session-timer-03.txt
> 
> 
> 
> 
> Hi,
> 
> Some comments and questions :
> 
>    Section 5.2, CASE I - UAC supports session timers. The 
> text states "In this
>    case the proxy remembers that it inserted a 
> Session-Expires header into the
>    request and that the UAC supported sesion timer". The 
> proxy may not have
>    inserted a Session-Expires header - it would only have 
> done so if the there
>    wasn't one in the request from the UAC.

I rewrote the text for CASE I. There are actually two sub-cases. One is when
the proxy is the one closest to the UAS (in which case it inserts
Session-TImer and Require into the response), and the other case is for all
other proxies. For those proxies, this case is indistinguishable from CASE
IV.



>    CASE II - UAC Doesn't support, UAS doesn't support. There 
> is, presumably,
>    nothing to stop the proxy, if its policy dictates, from 
> running a local
>    session timer, the expiration of which would result in it 
> de-allocating any
>    resources - and in particular, closing any pin-holes 
> opened across a
>    firewall, with no indication at all to the end points. 
> Whilst this isn't very
>    nice for the end points, since their RTP packets would 
> suddenly cease, the
>    proxy is trying to protect it's network resources from 
> unreliable end-points.
>    The session timer used be longer that normal, given there 
> is no opportunity
>    to refresh. Alternatively, cannot the proxy issue a 
> re-INVITE to each UA,
>    using the same SDP used orginally?

No, it cannot. Proxies do not initiate requests except for ACK and CANCEL. 

If the proxy wants to run a local timer on its own, well, thats its own
choice. There are no protocol implications of that, and it has nothing to do
with this session timer specification. Thus, I see no need to discuss it
here.

>    Section 6, Para 2 states "If the the UAS places a 
> Session-Expires header into
>    the response, and the request contained a Supported header 
> with the value
>    "timer",.......". This implies that the UAS may place a 
> Session-Espires
>    header into the response when the request did not contain 
> a Supported header
>    with the value "timer". However,  the table further down 
> says that if the
>    request contained neither a Supported header with the 
> value "timer", nor a
>    Session-Expires header, there there will be no timers. 
> Surely, if the UAS
>    supports session timers it can place a Session-Expires 
> header in the
>    response, so that it indicates to proxies that it is 
> running a session timer.
>    If this is the case, then the behavoir in the foirst row 
> of this table should
>    indicate that the UAS MAY add a session-Expires header to 
> the response. This
>    then equates to CASE III in section 5.2.

Yes, you are correct. I have updated the table.


>    Section 6, last para suggests that a 200 OK in response to 
> a re-INVITE may
>    not contain a Session-Expire header. How can this be, 
> unless a proxy or the
>    UAC has removed it, which I thought was not allowed?

No; the case is when the proxies change their mind and decide that they no
longer want session timer. In this case, the re-invite had a supported
header, but no session-expires. No proxies were interested, so none inserted
session-expires. UAS doesn't support, so the response has no
session-expires, no Require. I added a sentence explaining when this
happens.

>    Section 8.5 - call flow for re-INVITE. The Calling UA send 
> the INVITE to the
>    SPS with a Session-Expires header. The note against the 
> next message, INVITE
>    from SPS to Called UA says the SPS adds the S-E header, 
> but it was already
>    there.

Copy-paste error. Fixed.

>    General - ther terms UAC and UAS I have taken to be 
> synonymous with Calling
>    UAC and Called UAC, respectively. The document makes use 
> of both terms at
>    varying times. Are these terms interchangeable or is there 
> some implied
>    difference.

THe roles of UAC and UAS are on a transaction by transaction basis. The
initiator of a request is the UAS. Recipient is UAS. Caller and called party
reflect the initial invite request. I believe the terms are used consistent
with this definition. This definition is the one present in rfc2543.


>    Section 9 of RFC2543bis02 lists abbreviated forms for 
> common header fields.
>    The session timer feature is valuable and may be insisted 
> upon by many
>    networks, bringing the Session Expires header into common 
> use. You should
>    consider an abbreviated form.

Good idea. I've taken 'x', for eXpires.


Thanks for your comments,

Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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Date: Wed, 22 Nov 2000 05:26:58 -0500



 

> -----Original Message-----
> From: Singh, Manish [mailto:manishs@trillium.com]
> Sent: Friday, November 03, 2000 1:30 PM
> To: 'Jonathan Rosenberg'; Singh, Manish; 'Robin Escolme'
> Cc: sip@lists.bell-labs.com
> Subject: RE: [SIP] I-D ACTION:draft-ietf-sip-session-timer-03.txt
> 
> 
> Hi :
> 
> Thanks for the response.
> 
> Another concern, section 5 states -
> 
> "Session expiration are mostly of interest to call stateful
> proxy servers. However, a stateful proxy server MAY also follow
> the rules described here"
> 
> Shouldn't it be mandated by the draft that, if a stateful proxy
> (transaction stateful case) server is starting a Session Timer, 
> it MUST insert a "Record-Route" header in the Invite, prior to 
> proxying it.

No. Its not useful to the proxy to ask for a session timer and then not
record-route, but nothing breaks if you don't. I'll mention that its silly
not to.

> 
> Since, the procedure is symmetric, i.e. UAS can send a Re-Invite 
> and become the master, to avoid such proxies being missed out and
> later timing out, it MUST be on the path.
> 
> Also, is there any example when a transaction stateful only
> proxy would like to start a session timer?

Protocol processing has nothing to do with whether you are transaction
stateful or call stateful. This fact is hidden from anyone outside of the
proxy. Any of the flows, as a result, are the same for both cases.

Thanks,
Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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I=B4m looking for SIP Tutorial
Does anyone have a SIP briefly tutorial, or sugestion?

Thanks
Edson


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From sip-admin@lists.bell-labs.com  Wed Nov 22 08:20:54 2000
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In (expired) ID draft-sparks-sip-multiproxy-auth-00:

"A UAC should be prepared to terminate the deadlock situation caused by a
proxy in the chain that expires a challenge after its first successful
response."
Can somebody give an example of call flow resulting in such a deadlock? I
tried many scenarios and I can't figure out what is meant.
Thanks.

Alexandre Charest
Software Designer
Mediatrix Telecom Inc. (www.mediatrix.com)
tél:(819) 829-8749 #275
fax: (819) 829-5100


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Peter Kjellerstedt wrote:
> 
> In test30.txt if the Expires value is in absolute formate,
> then a Date header should be supplied too. Otherwise a UA
> that has no real-time clock will not be able to tell whether
> this is in the future or past. Actually I think the Expires
> value should be compared with the Date header if available
> since the UA on the other end may have an incorrect notion
> of time...

But this isn't a MUST so the test message is legal, bear in
mind these are torture tests not examples of good practice.

> Also, I am still lacking a test for this kind of To/From header:
> 
>         To: sip:pkj@axis.com ;tag=123456789@example.com
> 
> and make sure the tag does not end up as a URL parameter...

Sorry I forgot to add that one but tests 37 and 38 are pretty 
similar to this although admittedly they are for a registrar 
not UAS.

Cheers,
Neil.
-- 
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Stuart Wray wrote:
> 
> Henning Schulzrinne wrote:
> 
> > The problem is that with new parameters, older applications won't know
> > whether the parameter has a default value and what that might be.
> 
> I don't think this problem has a neat solution. I think what is really
> wanted is that URLs which are semantically equivalent should
> compare as equal. But if a new parameter changes the semantics
> how can older applications ever make the right judgement?
> 

There is a difference between changing the semantics and changing the
comparison rules. New parameters can well change the semantics without
affecting the To/From call leg comparison which is at issue here.
Semantics changes that are mandatory to implement obviously require a
Require header.

> In this last case do you really mean to ignore the port number if it is not
> present in both URLs?

I worded my statement carefully to say ;-parameter. The port number does
not fall into this category.

> 
>   Stuart Wray

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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edson.fassoni@tropiconet.com.br wrote:
> 
> I´m looking for SIP Tutorial
> Does anyone have a SIP briefly tutorial, or sugestion?

See http://www.cs.columbia.edu/sip for numerous articles and slides.
There are also various conferences that offer SIP tutorials.

> 
> Thanks
> Edson

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From sip-admin@lists.bell-labs.com  Wed Nov 22 09:23:10 2000
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"Fairlie-Cuninghame, Robert" wrote:
> 
> The colon in the user part of Test 36 (the long telephone subscriber) needs
> to be quoted. Perhaps you could also put a similar long telephone-subscriber
> URL in the Contact - this would test whether the URL was correctly
> reproduced if the call is accepted.

Actually looking more closely at the URL in the Request URI of 
test36.txt ...

sip:+1-972-555-2222;phone-context=name%40domain;new=user?%22Route:%20X%40Y%3bZ=W%22@gw1.wcom.com;user=phone

I think the ':' might be OK as it becomes part of a hname in 
an escaped header as it follows a '?', i.e. from parsing it 
into:
	%22Route:%20X%40Y%3bZ = W%22@gw1.wcom.com;user=phone
        	       header = hvalue
However this would then mean the following  ';' and '=' in the 
hvalue need to be escaped.

Also we would back into the problem of escaped headers not being
allowed in Request-URIs. I am a bit uncertain about this now.
The spec indicates that they are not allowed (Table 2) and 
should be removed by a proxy before futher processing (4.3).
However there seems to be a feeling that this may not be
desirable
- should there be a spec change here?

Cheers,
Neil.
-- 
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From: "Hisham Khartabil" <hisham.khartabil@hotsip.com>
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Hi,

section 6.46.3 in rfc2543bis says:

Via header fields in responses are processed by a proxy or UAC according to
the following rules:
1. The first Via header field should indicate the proxy or client processing
this response. If it does not, discard the message. Otherwise, remove this
Via field.
2. If there is no second Via header field, this response is destined for
this client. Otherwise, use this Via field as the destination, as described
in Section 6.46.5.

is that implying that a UAC has to forward the request, even if it is an end
user?

Regards,
Hisham Khartabil
Hotsip Finland
www.hotsip.com


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From: Peter Kjellerstedt <peter.kjellerstedt@axis.com>
To: "'Neil Deason'" <ndeason@ubiquity.net>
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Subject: RE: [SIP] More Torture Test Messages Available
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Date: Wed, 22 Nov 2000 15:32:52 +0100

> -----Original Message-----
> From: Neil Deason [mailto:ndeason@ubiquity.net]
> Sent: 22 November 2000 14:51
> To: Peter Kjellerstedt
> Cc: 'sip@lists.bell-labs.com'
> Subject: Re: [SIP] More Torture Test Messages Available
> 
> Peter Kjellerstedt wrote:
> > 
> > In test30.txt if the Expires value is in absolute format,
> > then a Date header should be supplied too. Otherwise a UA
> > that has no real-time clock will not be able to tell whether
> > this is in the future or past. Actually I think the Expires
> > value should be compared with the Date header if available
> > since the UA on the other end may have an incorrect notion
> > of time...
> 
> But this isn't a MUST so the test message is legal, bear in
> mind these are torture tests not examples of good practice.

No it is not a MUST, but you cannot expect me to respond with
a 408 if you do not provide me with your time (as I have no RTC...)
The description of the test should at least state what will
happen for servers without knowledge of time. You could also have
another identical test, but with a Date header. And you could
actually have a third test with a Date and an Expires header
that both are in the past, but where the Expires header is
in the future compared to the supplied Date header...

> > Also, I am still lacking a test for this kind of To/From header:
> > 
> >         To: sip:pkj@axis.com ;tag=123456789@example.com
> > 
> > and make sure the tag does not end up as a URL parameter...
> 
> Sorry I forgot to add that one but tests 37 and 38 are pretty 
> similar to this although admittedly they are for a registrar 
> not UAS.

True, test 37 tests it. I did not notice it as it was a REGISTER
request, and I do not really handle REGISTER requests :) It would
probably be a good idea to have an INVITE test for this too.

> Cheers,
> Neil.

//Peter

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Subject: RE: [SIP] Via processing rules
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> section 6.46.3 in rfc2543bis says:
> 
> Via header fields in responses are processed by a proxy or UAC 
> according to the following rules:
> 1. The first Via header field should indicate the proxy or client 
> processing this response. If it does not, discard the message.
> Otherwise, remove this Via field.
> 2. If there is no second Via header field, this response is destined
> for this client. Otherwise, use this Via field as the destination,
> as described in Section 6.46.5.
> 
> is that implying that a UAC has to forward the request, even if 
> it is an end user?

(I think you mean response?)

I don't see how, unless a UAC received a response that had more
than one Via.  In such a case, either the UAC wasn't behaving
as a UAC (i.e., it was a proxy) when it sent the corresponding
request; or something downstream has gone ka ka.


 - Jo.


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From sip-admin@lists.bell-labs.com  Wed Nov 22 09:42:03 2000
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From: Lars Berggren <lars@intertex.se>
To: Billy Biggs <Billy_Biggs@3com.com>
Cc: "Henning G. Schulzrinne" <hgs@cs.columbia.edu>,
        Jo Hornsby <jhornsby@ubiquity.net>,
        Alan Johnston <alan.johnston@wcom.com>,
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Subject: Re: [SIP] From tags (was: I-D ACTION:draft-ietf-sip-service-examples-00.txt)
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Date: Wed, 22 Nov 2000 01:48:01 +0100 (CET)

On Tue, 21 Nov 2000, Billy Biggs wrote:

> Henning G. Schulzrinne (hgs@cs.columbia.edu):
> 
> > >   So, UA sends a request, it loops back, they end up putting the
> > > same tag as the To and as the From...  You see the problem?
> > > Call-leg endpoint matching is based on:
> > > 
> > >   call-id,local-uri,local-tag,remote-uri,remote-tag
> > > 
> > >   And you'll end up with the same key for both.  The code gets
> > > cleaner if you always pick random tags.  Unfortunately,
> > > call-leg-end matching is very difficult.
> > 
> > I'm not sure I follow this logic. If A calls B and B proxies back to
> > A, we get A sending
> > 
> > INVITE B
> > From: A
> > To: B
> > 
> > The proxy forwards it to A again:
> > 
> > INVITE A
> > From: A
> > To: B
> > 
> > A can easily tell that it was the origin of the request ("From: A"),
> > so I'm not sure how tags get involved here. In this example, the
> > request doesn't even have To tags.
> 
>   What if A and B are the same URI?  How do I tell them apart?  The user
> answers the call, I send a 200 OK to the INVITE and then ACK it.  Now
> one of the lines hangs up and I send a BYE: how can I tell which
> call-leg end the BYE was meant for when it comes back?  Was I even able
> to match that ACK properly?  Do I have to special-case calls to myself?
> That would be really annoying, especially if I'm a gateway.
> 

At a first look I think this is solved by the following phrase in
section 6.25 in rfc2543:

   The "tag" MAY appear in the From field of a request. It MUST be present
   when it is possible that two instances of a user sharing a SIP address
   can make call invitations with the same Call-ID.

This seems to imply that the tags should be different to be useful.

However, as Billy mentioned, a piece of new text is added which says:

   A single user agent maintains the same tag at least within the same
   Call-ID, but it is RECOMMENDED to maintain the same tag across calls
   and instances of the UA application to allow restarting of a user
   agent.

Since tags primarily are used for differentiating of multiple instances of
a SIP address, I think having the same tag across instances of the UA
might break the protocol more than it helps. And, why does the UA
crash anyway? Well, maybe because identical tags across UAs creates shaky
scenarios as the one described earlier in this thread. And if the tags are
unique to each UA, the UA may be less likely to have to reboot in the
first place.

I agree with Billy and would suggest that tags should be unique to each
instance of a user agent and maybe even unique to a call.

/Lars

Lars Berggren       <mailto:lars.berggren@intertex.se>
Intertex Data AB    tel: +46-8-6282828
Sundbyberg, Sweden  fax: +46-8-6286414


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To: "Billy Biggs" <Billy_Biggs@3com.com>,
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Subject: Re: [SIP] Changing Codec
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Date: Wed, 22 Nov 2000 09:37:33 +0530
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But if this (re)INVITE fails, shouldn't the original media agreed upon
prevail ? That's how I understand (re)INVITES(unlike what you mention about
the remote UA tearing down the call). Please clarify.
Venkatesh
----- Original Message -----
From: "Billy Biggs" <Billy_Biggs@3com.com>
To: "Pascal Doré" <pdore@mediatrix.com>
Cc: "Mo Zonoun" <mzonoun@nortelnetworks.com>; "SIP List"
<sip@lists.bell-labs.com>
Sent: Wednesday, November 22, 2000 6:13 AM
Subject: Re: [SIP] Changing Codec


Pascal Doré (pdore@mediatrix.com):

> > How do you change codec in a SIP call without disconnecting the
> > established call?
>
> (re)Invite with a new session description

  Actually, it depends on if you are talking about send codec or receive
codec.

  If you want to change the codec you are sending, then as long as the
other end indicated support, you can change any time you like.  For
example, if I send you SDP that says:

   m=audio 8552 RTP/AVP 0 1 2

  Then you can freely switch between sending audio using codec 0, 1 or
2.  If you're feeling cruel, you can use a different codec for every
packet.


  If you want to change which codec you are receiving from the far end,
the best you can do is offer a different set of choices using a
re-INVITE.  So, first maybe you send:

   m=audio 8552 RTP/AVP 0 1 2

  And the far end is sending you codec 1, but you only want to receive
codec 0, then you can always re-INVITE with:

   m=audio 8552 RTP/AVP 0

  And force it to use either codec 0 or nothing.  Be prepared though
that if the remote end can't send codec 0, it might tear down the call
(or at best audio won't flow).

--
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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 But if this (re)INVITE fails, shouldn't the original media agreed upon
 prevail ? That's how I understand (re)INVITES(unlike what you mention about
 the remote UA tearing down the call). Please clarify.
 Venkatesh
> ----- Original Message -----
> From: "Billy Biggs" <Billy_Biggs@3com.com>
> To: "Pascal Doré" <pdore@mediatrix.com>
> Cc: "Mo Zonoun" <mzonoun@nortelnetworks.com>; "SIP List"
> <sip@lists.bell-labs.com>
> Sent: Wednesday, November 22, 2000 6:13 AM
> Subject: Re: [SIP] Changing Codec
>
>
> Pascal Doré (pdore@mediatrix.com):
>
> > > How do you change codec in a SIP call without disconnecting the
> > > established call?
> >
> > (re)Invite with a new session description
>
>   Actually, it depends on if you are talking about send codec or receive
> codec.
>
>   If you want to change the codec you are sending, then as long as the
> other end indicated support, you can change any time you like.  For
> example, if I send you SDP that says:
>
>    m=audio 8552 RTP/AVP 0 1 2
>
>   Then you can freely switch between sending audio using codec 0, 1 or
> 2.  If you're feeling cruel, you can use a different codec for every
> packet.
>
>
>   If you want to change which codec you are receiving from the far end,
> the best you can do is offer a different set of choices using a
> re-INVITE.  So, first maybe you send:
>
>    m=audio 8552 RTP/AVP 0 1 2
>
>   And the far end is sending you codec 1, but you only want to receive
> codec 0, then you can always re-INVITE with:
>
>    m=audio 8552 RTP/AVP 0
>
>   And force it to use either codec 0 or nothing.  Be prepared though
> that if the remote end can't send codec 0, it might tear down the call
> (or at best audio won't flow).
>
> --
> Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>
>
>


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From: "Hisham Khartabil" <hisham.khartabil@hotsip.com>
To: "Jo Hornsby" <jhornsby@ubiquity.net>, <sip@lists.bell-labs.com>
Subject: RE: [SIP] Via processing rules
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Date: Wed, 22 Nov 2000 16:43:10 +0200
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Sorry, I did mean response.

I think the text should explicitly say that in this situation, something
downstream has gone ka ka (well, not exactly that:-) )  and how to handle
this message.  Discard it, perhaps.

Regards,
Hisham

-----Original Message-----
From: Jo Hornsby [mailto:jhornsby@ubiquity.net]
Sent: Wednesday, 22 November 2000 4:32 PM
To: Hisham Khartabil; sip@lists.bell-labs.com
Subject: RE: [SIP] Via processing rules

> section 6.46.3 in rfc2543bis says:
>
> Via header fields in responses are processed by a proxy or UAC
> according to the following rules:
> 1. The first Via header field should indicate the proxy or client
> processing this response. If it does not, discard the message.
> Otherwise, remove this Via field.
> 2. If there is no second Via header field, this response is destined
> for this client. Otherwise, use this Via field as the destination,
> as described in Section 6.46.5.
>
> is that implying that a UAC has to forward the request, even if
> it is an end user?

(I think you mean response?)

I don't see how, unless a UAC received a response that had more
than one Via.  In such a case, either the UAC wasn't behaving
as a UAC (i.e., it was a proxy) when it sent the corresponding
request; or something downstream has gone ka ka.


 - Jo.


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Date: Wed, 22 Nov 2000 06:12:31 -0500

--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: SIP Traversal through Residential and Enterprise NATs 
                          and Firewalls
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-rosenberg-sip-entfw-00.txt
	Pages		: 15
	Date		: 21-Nov-00
	
In this draft, we discuss how SIP can traverse enterprise and
residential firewalls and NATs. This environment is challenging
because we assume here that the end user has little or no control
over the firewall or NAT, and that the firewall or NAT is completely
ignorant of SIP

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From sip-admin@lists.bell-labs.com  Wed Nov 22 10:11:32 2000
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Date: Wed, 22 Nov 2000 16:57:45 +0200



> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Wed, November 22, 2000 11:40 AM
> To: 'David Shrader'; SIP List
> Subject: RE: [SIP] Contact: header in re-INVITE
> 
> 
> Contact should be sent in every INVITE. Due to crash and 
> restart, what is a
> re-INVITE for one user is a new INVITE for another.

Shouldn't the INVITE be rejected with a 481 in this case because it arrives
with an unknown To tag?

  Itamar
> 
> If the Contact is different, yes, it should be updated in the 
> Route set.
> This is needed for certain proposed mobility applications, 
> and I suspect has
> some other very useful properties ala third party call control.
> 
> -Jonathan R.
> 
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>  
> 
> > -----Original Message-----
> > From: David Shrader [mailto:dshrader@master-consultant.com]
> > Sent: Thursday, November 09, 2000 11:17 AM
> > To: SIP List
> > Subject: [SIP] Contact: header in re-INVITE
> > 
> > 
> > Table 4 indicates that the Contact: header is mandatory "m" 
> > in the INVITE
> > request. Is this necessary for a re-INVITE message when the 
> > two endpoints
> > have already established communication.
> > 
> > I think this is similar to the Record-Route mechanism in 
> > which the route is
> > recorded once and applies to subsequent requests on the same 
> > call leg. For
> > that matter, if the Contact were to change. Do we change the
> > Record-Route/Route as well?
> > 
> > David
> > 
> > ----------------------------
> > David Shrader
> > Master Consultant, Inc.
> > dshrader@master-consultant.com
> > http://www.EyeForTheFuture.com
> > 
> > 
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> > 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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From sip-admin@lists.bell-labs.com  Wed Nov 22 10:14:37 2000
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Two other good sites for info on SIP.
http://www.sipforum.org/ and http://www.sipcenter.com/

Gordon Random
Director of Network Services
Medetalk, Inc.
grandom@medetalk.com

-----Original Message-----
From: sip-admin@lists.bell-labs.com
[mailto:sip-admin@lists.bell-labs.com]On Behalf Of
edson.fassoni@tropiconet.com.br
Sent: Wednesday, November 22, 2000 6:29 AM
To: sip@lists.bell-labs.com
Subject: [SIP] SIP Tutorial


I´m looking for SIP Tutorial
Does anyone have a SIP briefly tutorial, or sugestion?

Thanks
Edson



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From sip-admin@lists.bell-labs.com  Wed Nov 22 10:18:04 2000
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Hisham Khartabil wrote:
> 
> Sorry, I did mean response.
> 
> I think the text should explicitly say that in this situation, something
> downstream has gone ka ka (well, not exactly that:-) )  and how to handle
> this message.  Discard it, perhaps.

Sorry, but unless we want a 400-page spec, the spec won't be able to
spell out every possible error situation that anybody can dream up,
particularly since error handling is up to the implementation. Some will
be very lenient (or just sloppy), others will be flagging the slightest
violation and send along bake-off invitations.


-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Date: Wed, 22 Nov 2000 09:32:53 -0600

Venkatesh Venkataramanan (venkatesh.venkataramanan@wipro.com):


> >   And the far end is sending you codec 1, but you only want to receive
> > codec 0, then you can always re-INVITE with:
> >
> >    m=audio 8552 RTP/AVP 0
> >
> >   And force it to use either codec 0 or nothing.  Be prepared though
> > that if the remote end can't send codec 0, it might tear down the call
> > (or at best audio won't flow).

>  But if this (re)INVITE fails, shouldn't the original media agreed
>  upon prevail ? That's how I understand (re)INVITES(unlike what you
>  mention about the remote UA tearing down the call). Please clarify.

  I think this was the intention of some folks on this list- that a
rejected re-INVITE means the old session stays up.

  However, I would be more inclined to accept the re-INVITE and shut
down the call.  The SDP was not malformed, and the remote side has
clearly changed its session and is now listening for a different codec.

  I don't have any implementation experience with multiple codecs
though (not with kphone).

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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From sip-admin@lists.bell-labs.com  Wed Nov 22 10:51:20 2000
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Actually, I think it is better for the side that issued the re-INVITE to
make the decision about whether to return to the original media or tear down
the call.  If the decision is to return to the original media then no more
SIP messages are required since the other side already rejected the new
media.  If the decision is to terminate the call because the original media
can no longer be supported then a BYE can be issued.

-Maroula Bratakos

-----Original Message-----
From: Billy Biggs [mailto:Billy_Biggs@3com.com]
Sent: Wednesday, November 22, 2000 7:33 AM
To: Venkatesh Venkataramanan
Cc: sip@lists.bell-labs.com; Pascal Doré; Mo Zonoun
Subject: Re: [SIP] Changing Codec


Venkatesh Venkataramanan (venkatesh.venkataramanan@wipro.com):


> >   And the far end is sending you codec 1, but you only want to receive
> > codec 0, then you can always re-INVITE with:
> >
> >    m=audio 8552 RTP/AVP 0
> >
> >   And force it to use either codec 0 or nothing.  Be prepared though
> > that if the remote end can't send codec 0, it might tear down the call
> > (or at best audio won't flow).

>  But if this (re)INVITE fails, shouldn't the original media agreed
>  upon prevail ? That's how I understand (re)INVITES(unlike what you
>  mention about the remote UA tearing down the call). Please clarify.

  I think this was the intention of some folks on this list- that a
rejected re-INVITE means the old session stays up.

  However, I would be more inclined to accept the re-INVITE and shut
down the call.  The SDP was not malformed, and the remote side has
clearly changed its session and is now listening for a different codec.

  I don't have any implementation experience with multiple codecs
though (not with kphone).

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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From sip-admin@lists.bell-labs.com  Wed Nov 22 10:56:19 2000
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Subject: [SIP] INVITE/200 OK/BYE or INVITE/200 OK/ACK/BYE?
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Hi:

Assume a UAC sent an INVITE request which reached, through a forking proxy, 
to 2 UASes.  Both of those UASes send a 200 OK upstream.  The UAC now gets 2
200 OKs and wants to talk to one and reject the other.
 
Based on some logic, the UAC decides to establish a session with UAS 1, so 
it sends it a ACK to UAS 1.
 
After sending an ACK to UAS 1, the UAC wants to reject UAS 2.  Which is the 
appropriate message flow:
 
   UAC         UAS 2
   ACK--------->
   BYE--------->
   <---------- 200 OK (BYE)
 
OR:
 
   UAC         UAS 2
   BYE--------->
   <---------- 200 OK (BYE)
 
I believe the first flow is correct (bis, Section 4.2.4, "BYE": 
   "A BYE request from either called or calling party terminates any pending
   INVITE at a UA, but the INVITE request MUST be completed with a final 
   response and ACK")
 
However, reading some of the postings I get the impression that jives with 
the second flow.
 
So, which is it?  

Could the bis draft include something in Section 16.5 "Forking Proxy" call 
flows; especially where Bell gets two 200 responses from X and Y.  The text 
now says: "He can either send an ACK to both if human intelligence is needed 
to determine who he wants to talk to or he can automatically reject one of 
the two calls."  
 
It would be good if the text showed how to reject the other call: by sending 
an ACK immediately followed by a BYE (if first flow is correct), or by 
sending a BYE (if second flow is correct).
 
Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Internet Software and eServices Group 
Lucent Technologies/Bell Labs Innovations 263 Shuman Blvd., Rm 1A-413
Naperville, Illinois 60566     Voice: +1 630 224 0216   Fax: +1 630 713 0184

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From sip-admin@lists.bell-labs.com  Wed Nov 22 11:04:48 2000
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Date: Wed, 22 Nov 2000 09:52:53 -0600

Why don't you have two fixed and permanent tags: a "from" tag and a "to"
tag? Won't that solve the problems with calling yourself?

I think you're taking the text in rfc-bis too literally: it is really
recommending that the UA be able to recognize its own tags across reboots.
Having two tags instead of one doesn't interfere with the intention.

> -----Original Message-----
> From: EXT Billy Biggs [mailto:Billy_Biggs@3com.com]
> Sent: Tuesday, November 21, 2000 10:40 PM
> To: Henning G. Schulzrinne
> Cc: Lars Berggren; Jo Hornsby; Alan Johnston; Henry Chen; SIP List
> Subject: Re: [SIP] From tags (was: I-D
> ACTION:draft-ietf-sip-service-examples-00.txt)
> 
> 
> Henning G. Schulzrinne (hgs@cs.columbia.edu):
> 
> > I have yet to understand why re-using the same tag across restarts
> > hurts anything. After all, you're not worried about 
> confusing messages
> > when a SIP UA is revived.
> > 
> > Maybe we should look at the self-call in a bit more detail
> > [...]
> > The major problem occurs probably not so much because of 
> mapping, but
> > rather since this is treated as a re-INVITE while the first one is
> > still pending.
> 
>   Sorry I don't quite understand what you mean.  It looks like your
> email got a bit truncated.
> 
>   To help, I'm going to describe a scenario where a UA calls 
> itself, and
> that UA always uses a local tag (To or From) of 7.  Hopefully 
> this will
> help to clarify the problem.
> 
> 
>    I decide to call myself.  I create a call-leg struct to 
> represent it.
> Let's call it "Line 1".  I send out the initial INVITE:
> 
>   [From Line 1]
>   INVITE sip:me@proxy
>   Via: SIP/2.0/UDP my.ip
>   To: <sip:me@proxy>
>   From: <sip:me@proxy>;tag=7
>   Call-ID: 5
> 
>   This eventually loops back to me.  Now I see coming in:
> 
>   INVITE sip:me@proxy
>   Via: SIP/2.0/UDP some.ip
>   To: <sip:me@proxy>
>   From: <sip:me@proxy>;tag=7
>   Call-ID: 5
> 
>   I check and see there is no To-tag, so this is a new 
> incoming call.  I
> create a call struct to represent it, called "Line 2".  Line 2 now
> begins ringing.  I respond with a 180 and the user switches lines and
> answers the incoming call:
> 
>   [From Line 2]
>   SIP/2.0 200 OK
>   Via: SIP/2.0/UDP some.ip
>   To: <sip:me@proxy>;tag=7
>   From: <sip:me@proxy>;tag=7
>   Call-ID: 5
> 
>   When I receive this response, I find the related pending 
> transaction.
> Arguably, I could first find which call-leg it is destined to (and
> immediately have a problem), but say that I do transaction 
> mapping first
> so we don't break yet.  I match the OK and send an ACK:
> 
>   [From Line 1]
>   ACK sip:me@contactaddr
>   Via: SIP/2.0/UDP my.ip
>   To: <sip:me@proxy>;tag=7
>   From: <sip:me@proxy>;tag=7
>   Call-ID: 5
> 
>   At which point the other pending transaction also correctly matches
> this ACK to it.
> 
>   Now, on Line 2, I decide to hang up.  I've had enough of talking to
> myself.  So, I send a BYE:
> 
>   [From Line 2]
>   BYE sip:me@contact.addr
>   Via: SIP/2.0/UDP my.ip
>   To: <sip:me@proxy>;tag=7
>   From: <sip:me@proxy>;tag=7
>   Call-ID: 5
> 
>   I now see this incoming BYE:
> 
>   BYE sip:me@contact.addr
>   Via: SIP/2.0/UDP some.ip
>   To: <sip:me@proxy>;tag=7
>   From: <sip:me@proxy>;tag=7
>   Call-ID: 5
> 
>   Which one of the calls is it for?  Is it for the line which 
> originated
> the call, or the line which received the call?
> 
>   Using unique tags per call avoids any confusion even in 
> this case, and
> also protects the UA against clients which always use the same tag and
> happen to make two calls with the same Call-ID.
> 
> -- 
> Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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Subject: Re: [SIP] INVITE/200 OK/BYE or INVITE/200 OK/ACK/BYE?
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Date: Wed, 22 Nov 2000 10:03:25 -0600

Vijay Gurbani (vkg@lucent.com):

> After sending an ACK to UAS 1, the UAC wants to reject UAS 2.  Which
> is the appropriate message flow:
>  
>    UAC         UAS 2
>    ACK--------->
>    BYE--------->
>    <---------- 200 OK (BYE)
>  
> OR:
>  
>    UAC         UAS 2
>    BYE--------->
>    <---------- 200 OK (BYE)

  The second flow is incorrect, you must ACK the OK.  This is a known
area of confusion in the bis draft and it has already been mentioned on
this list that it will be clarified.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Wed, 22 Nov 2000 11:09:47 -0500



 

> -----Original Message-----
> From: Vijay Gurbani [mailto:vkg@lucent.com]
> Sent: Wednesday, November 22, 2000 10:37 AM
> To: SIP LIST
> Subject: [SIP] INVITE/200 OK/BYE or INVITE/200 OK/ACK/BYE?
> 
> 
> Hi:
> 
> Assume a UAC sent an INVITE request which reached, through a 
> forking proxy, 
> to 2 UASes.  Both of those UASes send a 200 OK upstream.  The 
> UAC now gets 2
> 200 OKs and wants to talk to one and reject the other.
>  
> Based on some logic, the UAC decides to establish a session 
> with UAS 1, so 
> it sends it a ACK to UAS 1.
>  
> After sending an ACK to UAS 1, the UAC wants to reject UAS 2. 
>  Which is the 
> appropriate message flow:
>  
>    UAC         UAS 2
>    ACK--------->
>    BYE--------->
>    <---------- 200 OK (BYE)
>  
> OR:
>  
>    UAC         UAS 2
>    BYE--------->
>    <---------- 200 OK (BYE)
>  
> I believe the first flow is correct (bis, Section 4.2.4, "BYE": 
>    "A BYE request from either called or calling party 
> terminates any pending
>    INVITE at a UA, but the INVITE request MUST be completed 
> with a final 
>    response and ACK")

Correct. rfc2543 allowed you to skip the ACK. Bis mandates it. The reason is
funamental theorem number 13 of SIP, "all transactions complete independent
of any other transactions". This is needed in order to implement a
reasonable UA that separates low level transaction processing from higher
level call manipulations. Its also needed for sanity in proxies which are
transactional.

>  
> However, reading some of the postings I get the impression 
> that jives with 
> the second flow.
>  
> So, which is it?  
> 
> Could the bis draft include something in Section 16.5 
> "Forking Proxy" call 
> flows; especially where Bell gets two 200 responses from X 
> and Y.  The text 
> now says: "He can either send an ACK to both if human 
> intelligence is needed 
> to determine who he wants to talk to or he can automatically 
> reject one of 
> the two calls."  
>  
> It would be good if the text showed how to reject the other 
> call: by sending 
> an ACK immediately followed by a BYE (if first flow is 
> correct), or by 
> sending a BYE (if second flow is correct).

Thanks; we'll update.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From: Billy Biggs <Billy_Biggs@3com.com>
To: Cliff.Harris@nokia.com
Cc: SIP List <sip@lists.bell-labs.com>
Subject: Crash-Proof UAs (was Re: [SIP] From tags)
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Date: Wed, 22 Nov 2000 10:34:07 -0600

Cliff.Harris@nokia.com (Cliff.Harris@nokia.com):

> Why don't you have two fixed and permanent tags: a "from" tag and a
> "to" tag?  Won't that solve the problems with calling yourself?
> 
> I think you're taking the text in rfc-bis too literally: it is really
> recommending that the UA be able to recognize its own tags across
> reboots.  Having two tags instead of one doesn't interfere with the
> intention.

  Building a crash-proof UA is not as simple as bis makes it out to be.
Rather than putting little hints throughout bis, there should be a
separate document describing methods and their problems.

  My point was that the recommendation will lead to broken behavior.


  To start this seperate document, here is a list of requirements for
Crash-proofness without saving call-state for all active calls:

  1. The ability to identify a tag as having originated from the
     pre-crash UA which this new UA is subsuming.  A better idea is to
     have the tag in two parts, one which is a hash of something unique
     to the UA, the other something random.  These tags MUST be unique
     to the instance.

  2. The ability to choose increasing cseq numbers across reboots and
     across calls.  Just trying to work through what's needed here is
     very difficult.  If you want to keep cseqs monotonotically
     increasing in each active call, then your cseq must be based on a
     hash of the call-leg information!  Ouch!

  3. Some security mechanism such that attacking UAs can't suddenly
     appear on your phone as if you were already in a call without the
     phone ringing!

  4. Some scheme whereby the remote end knows to send a re-INVITE to
     kickstart the call.  I have doubts about the following scenario:

     Say that Bob is on the phone to Jill:
     Bob: "blah blah... Jill?  Jill are you there?"

     Bob suddenly realizes that Jill isn't there!  Does Bob:

     a) Hang up the call and call Jill back
  or b) Put Jill on hold, and then take Jill off hold, with hopes the
        phone at the other end has just crashed.

     I find (b) to be far less likely.

  Because of the complexity, this should not be discussed in bis.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Wed, 22 Nov 2000 11:35:44 -0500





Q:
INVITE Sent by UAC got forked and reached two UAS1 and UAS2 ,  the UAC decided
to talk to UAS1. Is it possible that UAS2 (Assuming it is a client also) can
make a call to someone else.


What I wanted to know is, A user can be contacted at multiple location if one of
the location answers a CALL is it possible for somebody else on behalf of that
user make CALLS or receive CALLS....


Thanks
Sanjiv.



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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
To: "'Sanjiv Deshpande'" <sdeshpan@telcordia.com>, sip@lists.bell-labs.com
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Date: Wed, 22 Nov 2000 11:59:06 -0500

SIP is just a protocol. Thats it. Its a tool. If you want device A to call
device B, it tells you how to do that. Like all tools, you can use them in
many different ways. When you buy a hammer in Home Depot, does it tell you
whether to use your left hand or right? How many fingers to grip with it?
The size of the nails to use with it? Of course not. 

However, for some reason that I cannot fathom, many people seem to believe
the opposite is true for protocols; that is, protocols need to insist on how
syntax and semantics AND how they are applied to solve real world problems.
This approach just results in systems that are useful to no one. Would you
buy a hammer that said "for nails of length 1 inch only?" I doubt it.

So, to be specific, SIP doesn't tell you how to do any of the following:

1. the color of the phones that use SIP
2. the GUI for SIP calls
3. how you build a network of proxy servers
4. whether a phone can make calls while its in a call
5. qualifications for human beings to make use of SIP softphones (i.e., IQ
levels, height, or what have you).
6. how many simultaneous calls the device can be in

Thanks,
Jonathan R.



---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Sanjiv Deshpande [mailto:sdeshpan@telcordia.com]
> Sent: Wednesday, November 22, 2000 11:36 AM
> To: sip@lists.bell-labs.com
> Subject: [SIP] multiple location
> 
> 
> 
> 
> 
> 
> Q:
> INVITE Sent by UAC got forked and reached two UAS1 and UAS2 , 
>  the UAC decided
> to talk to UAS1. Is it possible that UAS2 (Assuming it is a 
> client also) can
> make a call to someone else.
> 
> 
> What I wanted to know is, A user can be contacted at multiple 
> location if one of
> the location answers a CALL is it possible for somebody else 
> on behalf of that
> user make CALLS or receive CALLS....
> 
> 
> Thanks
> Sanjiv.
> 
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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Subject: [SIP] draft-rosenberg-sip-app-components-00.txt
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Hi,

In the draft
http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-sip-app-components-00.txt

you propose to invoke text to speech services using an INVITE with two
"m" sections. One sendonly for the text and the other recvonly for the
speech (audio).

I see a TSS server as a particular case of a transcoder, that receives
media in one format (e.g. GSM) and converts it to another format (e.g.
PCM).

And I see a transcoder as a particular case of a conference until (MCU).
It just takes media from one participant and sends it to the other
(adjusting the format).

Since conference units are usually INVITEd using one INVITE per
conference member, I think transcoder services should be invoked in the
same way (with two INVITEs) as it is described in section 3 of:
http://search.ietf.org/internet-drafts/draft-camarillo-3pcc-qos-00.txt

Thus, I would invoke TSS services in the same way. I would send an
INVITE with an "m"section consisting of audio and a second INVITE with
an "m" section consisting of text. The TSS server would related both
INVITEs using the Request-URI (as a conference unit does).

I believe this would be a more general way of invoking such a service.

Opinions?

A minor typo: in section 9 Peter Mataga seems to share Jonathan's email
address.

Regards,

Gonzalo
-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 31 18
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland

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Subject: [SIP] New Section 6.34 (was: Record-Route consensus)
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Date: Wed, 22 Nov 2000 17:45:24 -0000
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Okay, here's a proposal for a new Section 6.34.  Hopefully, this
is as backwardly-compatible (to 2543) as we could possibly be
(that is, it should work with a 2543 proxy, and it even has a
good chance of working with a 2543 UA, if it was the one calling).

Also, this should solve using Record-Route when a proxy receives
a request with a non-SIP Request-URI.  The changes are basically
now that a proxy is free to put just about any URI into a
Record-Route value, as long as it's a UDP SIP URL; and the called
UA no-longer does the nasty mangling when copying into the Route.

Anyway, without further ado...



6.34 Record-Route


6.34.1 Operation

 The Record-Route request and response header field is added to a
request by any proxy that insists on being in the path of subsequent
requests for the same call leg.  A proxy SHOULD add it to any request
for robustness, but a request route, once established, persists until
the end of the call leg, regardless of whether the Record-Route header
is present in subsequent requests.

    Some proxies, such as those controlling firewalls or in an
    automatic call distribution (ACD) system, need to maintain call
    state and thus need to receive any BYE, re-INVITE and ACK packets
    for the call.

 Each Record-Route header field contains a globally reachable SIP URL
that identifies a proxy server.  Any proxy that wishes to remain in
the signalling path of a call leg adds its own SIP URL to the
beginning of the list of these fields.  This SIP URL MUST represent an
address that is reachable via UDP.  Requests between both user agents
involved in the call leg, in either direction, traverse this route.

    Inclusion of, say, a TCP transport parameter may prevent a UA that
    supports only UDP but reached the proxy inserting the Record-Route
    via another proxy from reaching this proxy.

 The UAS copies the Record-Route request header field unchanged into
the response.  (Record-Route is only relevant for 2xx responses and
responses where the server can expect the client to retry for the same
Call-ID, as in 401 (Unauthorized) or 484 (Address Incomplete).)

 *** Should we remove this bit about 401 and 484?  It makes things
     more complicated, and also what if a proxy has collected a load
     of WWW-Authenticate and Proxy-Authenticate headers into a
     401 (Section 12.4 of bis2) -- the Record-Route could be arbitrary.


6.34.2 Construction of Route header

 If a UAC finds a list of Record-Route header fields in a final
response to the initial request in a call leg, it copies this list
into Route header fields of all subsequent requests within the same
call leg, reversing the order of fields, so that the first entry is
the server closest to the UAC.

 *** I haven't said initial INVITE here, since wasn't there some talk about
     abstracting it for, say SUBSCRIBE/NOTIFY purposes?

 If a UAS finds a list of Record-Route header fields in the initial
request in a call leg, it copies this list, maintaining its ordering,
into the Route header fields of all subsequent requests within the
same call leg.

 In either case, the UA MUST copy all SIP URLs verbatim.

    This allows a proxy to add a SIP URL parameter indicating a
    request's original Request-URI, for instance.

 If the request/response featured a Contact header field, the UA Contact
header value is appended to the list of Route header fields.  If this
Contact header value is found to have changed in a subsequent request
or response, then the new value MUST then be used.  Note that the same
is not true of changes to list of Record-Route header values.

 *** If no Contact, should add the From?


6.34.3 Request Destination

If a client finds a list of Route header fields in a request, it
SHOULD send the request to the first SIP URL in the list.  A client
MAY forward the request to a designated proxy instead, for example, if
it lacks DNS resolution capability.  If a client uses the first Route
entry to route the request, it removes it.


6.34.4 Syntax

The Record-Route header field has the following syntax:

    Record-Route = "Record-Route" ":" 1#(name-addr *(";" rr-param))
        rr-param = generic-param


6.34.5 Example

Example for a request where the proxy servers at vanity-domain.com,
ieee.org, and sunnydale-high.edu, in that order, insist on being part
of subsequent request paths:

    Record-Route: <sip:buffy@sunnydale-high.edu>
      <sip:ieee.org;uri=sip:jo@ieee.org>
      <sip:sip.vanity-domain.com:6000;uri=tel:+1-972-555-2222>

 *** I took the liberty of making the example Record-Route slightly
     more exotic. &;)


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All,

I just wanted to inform you that an updated version of the AAA
Requirements for IP Telephony/Multimedia  I-D is now available for
preview on
http://www.diameter.org/draft-ietf-aaa-solutions-01.txt
The I-D was submitted to the secretariat, but it will take a few days
for it to be available on the archives.

Have a great Thanksgiving!

/Tony Johansson




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From: Igor Slepchin <ISlepchin@dynamicsoft.com>
To: "'Henning G. Schulzrinne'" <hgs@cs.columbia.edu>
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Date: Wed, 22 Nov 2000 13:22:05 -0500

> -----Original Message-----
> From: Henning G. Schulzrinne [mailto:hgs@cs.columbia.edu]
>
> > Test case 41:
> > 
> > I'd argue that this message should be rejected with a 400. 
>
>    The version of an HTTP message is indicated by an 
> HTTP-Version field
>    in the first line of the message.
> 
>        HTTP-Version   = "HTTP" "/" 1*DIGIT "." 1*DIGIT
> 
>    Note that the major and minor numbers MUST be treated as separate
>    integers and that each MAY be incremented higher than a 
> single digit.

Sounds fine except that I don't think this is ever referred to in 2543bis.
In fact, the grammar for SIP-Version explicitly disables these tricks:

SIP-Version="SIP/2.0"

Note that SIP-Version is a literal. I'm not sure what the value of allowing
"SIP/02.00" is, but if that's the intention, it should probably be mentioned
in the spec. BTW, the change would not be backwards compatible; if it's
made, it should probably be mentioned that leading zeros should not be
generated.

> Wait, if protocol is defined as, say
> 
> protocol-token = "FOO" | "BAR"
> 
> it would be CI. This is consistent with the use in RFC 2616, where all
> tokens are defined as CI (such as character-set, content-coding,
> transfer-coding, type, subtype, etc.

Absolutely. I was just being picky about the grammar of Via headers: it
allows protocol name to be something other than SIP, namely, it allows it to
be a token. token is case-sensitive, which implies that any protocol name
but SIP would be case-sensitive there. Maybe it's easier to just remove
token from protocol-name production in Via grammar (after all, what's the
meaning of sending a SIP message that is not SIP?)

---
Igor Slepchin

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Date: Wed, 22 Nov 2000 14:05:33 -0500

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I wanted to draw attention to the following draft which we have submitted.
It tackles, and provides what we believe to be at least a workable solution,
to the problem of SIP traversal through enterprise firewalls and NATs when
there is no ALG at all associated with, or inside of, the firewall. 

Thanks,
Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

-----Original Message-----
From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
Sent: Wednesday, November 22, 2000 6:13 AM
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Cc: sip@lists.bell-labs.com
Subject: I-D ACTION:draft-rosenberg-sip-entfw-00.txt


A New Internet-Draft is available from the on-line Internet-Drafts
directories.


	Title		: SIP Traversal through Residential and Enterprise
NATs 
                          and Firewalls
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-rosenberg-sip-entfw-00.txt
	Pages		: 15
	Date		: 21-Nov-00
	
In this draft, we discuss how SIP can traverse enterprise and
residential firewalls and NATs. This environment is challenging
because we assume here that the end user has little or no control
over the firewall or NAT, and that the firewall or NAT is completely
ignorant of SIP

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-rosenberg-sip-entfw-00.txt

Internet-Drafts are also available by anonymous FTP. Login with the username
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        "'Henning G. Schulzrinne'" <hgs@cs.columbia.edu>
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> > Wait, if protocol is defined as, say
> > 
> > protocol-token = "FOO" | "BAR"
> > 
> > it would be CI. This is consistent with the use in RFC 
> 2616, where all
> > tokens are defined as CI (such as character-set, content-coding,
> > transfer-coding, type, subtype, etc.
> 
> Absolutely. I was just being picky about the grammar of Via 
> headers: it <...>

Arrgh, I guess I missed your point that tokens are CI by default. The reason
I missed it is that case-insensitivity of literals is mentioned in Appendix
C whereas that of tokens is described in section 6.5.

Thank you,
Igor Slepchin

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hi recently downloaded the vovida software for sip. Having difficulties
in compiling the UA part on solaris 4.2 and above. Has anyone done that?
Any info on UA structures/classes hiearchy and complilation will be a
added plus! 
Thanks
Ramana

___________________________________________________________________
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Subject: RE: [SIP] vovida sip-ua
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You can sign up for email list at vovida. All these kinds of problems
specific to Vovida's stack are discussed there.


-----Original Message-----
From: sip-admin@lists.bell-labs.com
[mailto:sip-admin@lists.bell-labs.com]On Behalf Of Ramana Bhamidipati
Sent: Wednesday, November 22, 2000 3:04 PM
To: sip@lists.bell-labs.com
Subject: [SIP] vovida sip-ua


hi recently downloaded the vovida software for sip. Having difficulties
in compiling the UA part on solaris 4.2 and above. Has anyone done that?
Any info on UA structures/classes hiearchy and complilation will be a
added plus!
Thanks
Ramana

___________________________________________________________________
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all in one place - sign up today at http://www.zdnetonebox.com


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Subject: [SIP] New ISUP mapping draft
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A new ISUP to SIP mapping draft had been completed. Due to the
overwhelming number of contributions pouring in prior to the
November 24th deadline, the I-D editor is currently running
somewhat slower than usual. Until it becomes available from
the IETF website, you may retrieve a copy from:

http://ia.yimg.com/a/5e379c0a/h/48c55585/draft-ietf-sip-isup-00.txt

Please post all comments and replies to the sip-t@egroups.com
mailing list. Thanks.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA

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From sip-admin@lists.bell-labs.com  Wed Nov 22 17:03:07 2000
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
To: "'Bratakos, Maroula'" <mbratakos@nuera.com>,
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        Venkatesh Venkataramanan <venkatesh.venkataramanan@wipro.com>
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Subject: RE: [SIP] Changing Codec
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Either way is fine. Its a matter of local choice. I don't think the spec
needs to say anything except that a rejection of a reinvite returns to the
previous session state.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Bratakos, Maroula [mailto:mbratakos@nuera.com]
> Sent: Wednesday, November 22, 2000 10:51 AM
> To: 'Billy Biggs'; Venkatesh Venkataramanan
> Cc: sip@lists.bell-labs.com; Pascal Doré; Mo Zonoun
> Subject: RE: [SIP] Changing Codec
> 
> 
> Actually, I think it is better for the side that issued the 
> re-INVITE to
> make the decision about whether to return to the original 
> media or tear down
> the call.  If the decision is to return to the original media 
> then no more
> SIP messages are required since the other side already 
> rejected the new
> media.  If the decision is to terminate the call because the 
> original media
> can no longer be supported then a BYE can be issued.
> 
> -Maroula Bratakos
> 
> -----Original Message-----
> From: Billy Biggs [mailto:Billy_Biggs@3com.com]
> Sent: Wednesday, November 22, 2000 7:33 AM
> To: Venkatesh Venkataramanan
> Cc: sip@lists.bell-labs.com; Pascal Doré; Mo Zonoun
> Subject: Re: [SIP] Changing Codec
> 
> 
> Venkatesh Venkataramanan (venkatesh.venkataramanan@wipro.com):
> 
> 
> > >   And the far end is sending you codec 1, but you only 
> want to receive
> > > codec 0, then you can always re-INVITE with:
> > >
> > >    m=audio 8552 RTP/AVP 0
> > >
> > >   And force it to use either codec 0 or nothing.  Be 
> prepared though
> > > that if the remote end can't send codec 0, it might tear 
> down the call
> > > (or at best audio won't flow).
> 
> >  But if this (re)INVITE fails, shouldn't the original media agreed
> >  upon prevail ? That's how I understand (re)INVITES(unlike what you
> >  mention about the remote UA tearing down the call). Please clarify.
> 
>   I think this was the intention of some folks on this list- that a
> rejected re-INVITE means the old session stays up.
> 
>   However, I would be more inclined to accept the re-INVITE and shut
> down the call.  The SDP was not malformed, and the remote side has
> clearly changed its session and is now listening for a 
> different codec.
> 
>   I don't have any implementation experience with multiple codecs
> though (not with kphone).
> 
> -- 
> Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 
> _______________________________________________
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> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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From sip-admin@lists.bell-labs.com  Wed Nov 22 17:06:56 2000
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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        Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
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Subject: RE: [SIP] Contact: header in re-INVITE
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Date: Wed, 22 Nov 2000 17:01:13 -0500

That depends on whether the UAS is using To tags that are persistent across
reboots. There is a separate thread running now on this subject.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Itamar Gilad [mailto:ItamarG@tlv.radvision.com]
> Sent: Wednesday, November 22, 2000 9:58 AM
> To: 'Jonathan Rosenberg'; 'David Shrader'; SIP List
> Subject: RE: [SIP] Contact: header in re-INVITE
> 
> 
> 
> 
> > -----Original Message-----
> > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > Sent: Wed, November 22, 2000 11:40 AM
> > To: 'David Shrader'; SIP List
> > Subject: RE: [SIP] Contact: header in re-INVITE
> > 
> > 
> > Contact should be sent in every INVITE. Due to crash and 
> > restart, what is a
> > re-INVITE for one user is a new INVITE for another.
> 
> Shouldn't the INVITE be rejected with a 481 in this case 
> because it arrives
> with an unknown To tag?
> 
>   Itamar
> > 
> > If the Contact is different, yes, it should be updated in the 
> > Route set.
> > This is needed for certain proposed mobility applications, 
> > and I suspect has
> > some other very useful properties ala third party call control.
> > 
> > -Jonathan R.
> > 
> > ---
> > Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> > Chief Scientist                             First Floor
> > dynamicsoft                                 East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> > http://www.dynamicsoft.com
> >  
> > 
> > > -----Original Message-----
> > > From: David Shrader [mailto:dshrader@master-consultant.com]
> > > Sent: Thursday, November 09, 2000 11:17 AM
> > > To: SIP List
> > > Subject: [SIP] Contact: header in re-INVITE
> > > 
> > > 
> > > Table 4 indicates that the Contact: header is mandatory "m" 
> > > in the INVITE
> > > request. Is this necessary for a re-INVITE message when the 
> > > two endpoints
> > > have already established communication.
> > > 
> > > I think this is similar to the Record-Route mechanism in 
> > > which the route is
> > > recorded once and applies to subsequent requests on the same 
> > > call leg. For
> > > that matter, if the Contact were to change. Do we change the
> > > Record-Route/Route as well?
> > > 
> > > David
> > > 
> > > ----------------------------
> > > David Shrader
> > > Master Consultant, Inc.
> > > dshrader@master-consultant.com
> > > http://www.EyeForTheFuture.com
> > > 
> > > 
> > > _______________________________________________
> > > SIP mailing list
> > > SIP@lists.bell-labs.com
> > > http://lists.bell-labs.com/mailman/listinfo/sip
> > > 
> > 
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> > 
> 

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Subject: [SIP] New SUBSCRIBE/NOTIFY draft available
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A new SUBSCRIBE/NOTIFY draft (-02) has been submitted. Until it
appears on the IETF website, you may retreive a copy from:

http://ia.yimg.com/a/5e379c0a/h/57a388ad/draft-roach-sip-subscribe-notify-02.txt

Please send comments and replies to the sip-events@egroups.com
mailing list.

I believe that this draft incorporates all of the concerns which have
been raised with the -01 draft; please inform me if I've overlooked
anything. Changes include:

     - Multiple contacts per SUBSCRIBE message disallowed.

     - Contact header now required in NOTIFY messages.

     - Distinction between third party/call member events removed.

     - Distinction between call-related/resource-related events removed.

     - Clarified that subscribers must expect NOTIFY messages before
       the SUBSCRIBE transaction completes

     - Added immediate NOTIFY message after successful SUBSCRIBE;
       this solves a myriad of issues, most having to do with forking.

     - Added discussion of "undefined state" (before a NOTIFY arrives).

     - Added mechanism for notifiers to shorten/cancel outstanding
       subscriptions.

     - Removed open issue about appropriateness of new "489" response.

     - Removed all discussion of out-of-band subscriptions.

     - Added brief discussion of event state polling.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA

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>http://ia.yimg.com/a/5e379c0a/h/57a388ad/draft-roach-sip-subscribe-notify-02.txt

It has been brought to my attention that this link doesn't always
work; you may instead need to select the draft from this page:

http://briefcase.yahoo.com/bc/adamr?c&.flabel=fld6&.src=bc

--
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA

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Hmm; I could not create the case either. The case that I thought would cause
a deadlock does not, in fact. 

Robert?

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Alexandre Charest [mailto:acharest@mediatrix.com]
> Sent: Wednesday, November 22, 2000 8:21 AM
> To: sip@lists.bell-labs.com
> Subject: [SIP] Multi-proxy authentication deadlock
> 
> 
> In (expired) ID draft-sparks-sip-multiproxy-auth-00:
> 
> "A UAC should be prepared to terminate the deadlock situation 
> caused by a
> proxy in the chain that expires a challenge after its first successful
> response."
> Can somebody give an example of call flow resulting in such a 
> deadlock? I
> tried many scenarios and I can't figure out what is meant.
> Thanks.
> 
> Alexandre Charest
> Software Designer
> Mediatrix Telecom Inc. (www.mediatrix.com)
> tél:(819) 829-8749 #275
> fax: (819) 829-5100
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
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> 

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To: "'Jo Hornsby'" <jhornsby@ubiquity.net>, hgs@cs.columbia.edu
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Subject: RE: [SIP] New Section 6.34 (was: Record-Route consensus)
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Date: Wed, 22 Nov 2000 17:52:34 -0500

Before we begin writing new text, lets look at the sub-problems and build up
a solution. I have another possibility that may be the best of all worlds.
I'll start with Roberts list:


1. non SIP URLs in request URI

We can either define record-route parameters, or else make sure that the URL
put into the record route is a sip url. I originally propose RR parameters,
but I think insisting on a SIP URL is backwards compatible and more
reasonable. That seems to be consensus.

2. transaction ID in the reverse direction

This was the problem that if there was a spiral in the forward direction,
and the current bis approach was used by the UAS to construct the Route
header, a request would arrive twice at the spiraled-at proxy. These could
be identified as separate transactions (as a result of Via header
differences), but there is no way to determine which of the two iterations
through it is. 

The solution is that the request URI need some info that the proxy created
and is returned to it. There seems to be little disagreement on that.

3. Proxy doesn't see URI parameters it inserted in RR, returned to it in
request URI

This is a consequence of the current bis text for building the route.
Proposed solution for this component is that the construction process at the
UAS has to copy the URI parameters from the RR rather than the Contact/From.

4. detecting new behavior

I don't think this is an issue, since we need a solution thats backwards
compatible.

5. (this one is my own) The request URI should refer to the resource for
whom the message is destined

The solutions proposed are either to have the UAS and UAC construct the
Route differently (as is done in the current bis), else Jo's text here where
this information is carried somehow as parameters. I think this is a big
hack, and quite a pain. When a proxy receives a re-INVITE, it will have a
hard time knowing which direction the re-INVITE is going, and consequently,
which of the pieces of info to extract for processing purposes.



Putting it all together, its clear from (1) that the record routes must
always be SIP URLs. From 2 and 3, at least the parameters placed into the RR
should be copied into the Route by both sides, and from 5, the URI should
contain information that identifies the recipient of the message, different
in both directions.

My new proposal does all of the above, and requires no mangling of RR in the
UA. Here's how it goes.

When a request arrives, and a proxy record-routes, it places a URI into the
RR which identifies the *calling UA* as a resource on this proxy. This URI
can be derived from the Contact/From, but should not be exactly equal to it.
This must be a SIP URL. THe UAS copies the RR set into the response. It also
constructs a Route header by taking the RR's in order, and adding the
Contact from the INVITE at the end. Now, here's the trick. As the response
traverses the proxies, any proxy that record-routed, finds its RR entry, and
*replaces it* with a new URI; this URI identifies the called party at this
server. Probably this is the request URI that was originally received in the
INVITE. The UAC then constructs the Route as defined in rfc2543.

Now, the result of this is that the URIs in the two Route sets are
different; they identify the actual recipient of the message, and they are
inserted by the proxy and can thus be set by the proxy. Seems to have
everything we want.

Here's an example. A calls B, through a record-routing Proxy P1. Here are
message snipits:

A->P1

INVITE sip:B@example.com
Contact: sip:A@host.example.com

P1->B
INVITE sip:B@host.example.com
Contact: sip:A@host.example.com
Record-Route: <sip:A@proxy.example.com>

B->P1
200 OK 
Contact: sip:B@host.example.com
Record-Route <sip:A@proxy.example.com>

P1->B
200 OK
Contact: sip:B@host.example.com
Record-Route <sip:B@proxy.example.com>

The Route constructed by A is:

sip:B@proxy.example.com
sip:B@host.example.com

and by B:

sip:A@proxy.example.com
sip:A@host.example.com


This is similar to Jo's proposal but maintains the property that a request
from A to B has a URI that always points to a resource that gets routed to
B, and vice a versa (in Jo's case, the URI is the same in both directions).
Thus, I think it satisfies everyone. Note that maddr is no longer needed so
long as the host port gets routed back to the same server; of course it can
put maddr if it likes.

The drawback is that the proxy needs to do a bit more work. Not a lot,
though.

Comments?

-Jonathan R.




---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Jo Hornsby [mailto:jhornsby@ubiquity.net]
> Sent: Wednesday, November 22, 2000 12:45 PM
> To: hgs@cs.columbia.edu
> Cc: sip@lists.bell-labs.com; Billy_Biggs@3com.com
> Subject: [SIP] New Section 6.34 (was: Record-Route consensus)
> 
> 
> Okay, here's a proposal for a new Section 6.34.  Hopefully, this
> is as backwardly-compatible (to 2543) as we could possibly be
> (that is, it should work with a 2543 proxy, and it even has a
> good chance of working with a 2543 UA, if it was the one calling).
> 
> Also, this should solve using Record-Route when a proxy receives
> a request with a non-SIP Request-URI.  The changes are basically
> now that a proxy is free to put just about any URI into a
> Record-Route value, as long as it's a UDP SIP URL; and the called
> UA no-longer does the nasty mangling when copying into the Route.
> 
> Anyway, without further ado...
> 
> 
> 
> 6.34 Record-Route
> 
> 
> 6.34.1 Operation
> 
>  The Record-Route request and response header field is added to a
> request by any proxy that insists on being in the path of subsequent
> requests for the same call leg.  A proxy SHOULD add it to any request
> for robustness, but a request route, once established, persists until
> the end of the call leg, regardless of whether the Record-Route header
> is present in subsequent requests.
> 
>     Some proxies, such as those controlling firewalls or in an
>     automatic call distribution (ACD) system, need to maintain call
>     state and thus need to receive any BYE, re-INVITE and ACK packets
>     for the call.
> 
>  Each Record-Route header field contains a globally reachable SIP URL
> that identifies a proxy server.  Any proxy that wishes to remain in
> the signalling path of a call leg adds its own SIP URL to the
> beginning of the list of these fields.  This SIP URL MUST represent an
> address that is reachable via UDP.  Requests between both user agents
> involved in the call leg, in either direction, traverse this route.
> 
>     Inclusion of, say, a TCP transport parameter may prevent a UA that
>     supports only UDP but reached the proxy inserting the Record-Route
>     via another proxy from reaching this proxy.
> 
>  The UAS copies the Record-Route request header field unchanged into
> the response.  (Record-Route is only relevant for 2xx responses and
> responses where the server can expect the client to retry for the same
> Call-ID, as in 401 (Unauthorized) or 484 (Address Incomplete).)
> 
>  *** Should we remove this bit about 401 and 484?  It makes things
>      more complicated, and also what if a proxy has collected a load
>      of WWW-Authenticate and Proxy-Authenticate headers into a
>      401 (Section 12.4 of bis2) -- the Record-Route could be 
> arbitrary.
> 
> 
> 6.34.2 Construction of Route header
> 
>  If a UAC finds a list of Record-Route header fields in a final
> response to the initial request in a call leg, it copies this list
> into Route header fields of all subsequent requests within the same
> call leg, reversing the order of fields, so that the first entry is
> the server closest to the UAC.
> 
>  *** I haven't said initial INVITE here, since wasn't there 
> some talk about
>      abstracting it for, say SUBSCRIBE/NOTIFY purposes?
> 
>  If a UAS finds a list of Record-Route header fields in the initial
> request in a call leg, it copies this list, maintaining its ordering,
> into the Route header fields of all subsequent requests within the
> same call leg.
> 
>  In either case, the UA MUST copy all SIP URLs verbatim.
> 
>     This allows a proxy to add a SIP URL parameter indicating a
>     request's original Request-URI, for instance.
> 
>  If the request/response featured a Contact header field, the 
> UA Contact
> header value is appended to the list of Route header fields.  If this
> Contact header value is found to have changed in a subsequent request
> or response, then the new value MUST then be used.  Note that the same
> is not true of changes to list of Record-Route header values.
> 
>  *** If no Contact, should add the From?
> 
> 
> 6.34.3 Request Destination
> 
> If a client finds a list of Route header fields in a request, it
> SHOULD send the request to the first SIP URL in the list.  A client
> MAY forward the request to a designated proxy instead, for example, if
> it lacks DNS resolution capability.  If a client uses the first Route
> entry to route the request, it removes it.
> 
> 
> 6.34.4 Syntax
> 
> The Record-Route header field has the following syntax:
> 
>     Record-Route = "Record-Route" ":" 1#(name-addr *(";" rr-param))
>         rr-param = generic-param
> 
> 
> 6.34.5 Example
> 
> Example for a request where the proxy servers at vanity-domain.com,
> ieee.org, and sunnydale-high.edu, in that order, insist on being part
> of subsequent request paths:
> 
>     Record-Route: <sip:buffy@sunnydale-high.edu>
>       <sip:ieee.org;uri=sip:jo@ieee.org>
>       <sip:sip.vanity-domain.com:6000;uri=tel:+1-972-555-2222>
> 
>  *** I took the liberty of making the example Record-Route slightly
>      more exotic. &;)
> 
> 
> _______________________________________________
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> 

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Date: Wed, 22 Nov 2000 17:09:47 -0600

Jonathan Rosenberg (jdrosen@dynamicsoft.com):

> 5. (this one is my own) The request URI should refer to the resource
> for whom the message is destined
> 
> The solutions proposed are either to have the UAS and UAC construct
> the Route differently (as is done in the current bis), else Jo's text
> here where this information is carried somehow as parameters. I think
> this is a big hack, and quite a pain. When a proxy receives a
> re-INVITE, it will have a hard time knowing which direction the
> re-INVITE is going, and consequently, which of the pieces of info to
> extract for processing purposes.


  The proxy should look at the To and the From to tell direction,
instead of doing this ugly hack! [1]

  The only note then is that if a UA calls itself through a
Record-Route'ing outbound proxy, then the tags it inserts had better be
different.  [I think you've heard that argument before!]


  Still, bis shouldn't talk about any of this.  Keep the spec really
simple, and let proxies which need to keep state figure out how to
record-route to their advantage.  Jo's text is pretty much on the mark.


 [1]  Or, it can always just look at which address is at the end of the
      Route...  Regardless, your suggestion is an ugly hack.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Wed, 22 Nov 2000 18:12:25 -0500



 

> -----Original Message-----
> From: Billy Biggs [mailto:Billy_Biggs@3com.com]
> Sent: Wednesday, November 22, 2000 6:10 PM
> To: Jonathan Rosenberg
> Cc: Jo Hornsby; hgs@cs.columbia.edu; sip@lists.bell-labs.com
> Subject: Re: [SIP] New Section 6.34 (was: Record-Route consensus)
> 
> 
> Jonathan Rosenberg (jdrosen@dynamicsoft.com):
> 
> > 5. (this one is my own) The request URI should refer to the resource
> > for whom the message is destined
> > 
> > The solutions proposed are either to have the UAS and UAC construct
> > the Route differently (as is done in the current bis), else 
> Jo's text
> > here where this information is carried somehow as 
> parameters. I think
> > this is a big hack, and quite a pain. When a proxy receives a
> > re-INVITE, it will have a hard time knowing which direction the
> > re-INVITE is going, and consequently, which of the pieces of info to
> > extract for processing purposes.
> 
> 
>   The proxy should look at the To and the From to tell direction,
> instead of doing this ugly hack! [1]
> 
>   The only note then is that if a UA calls itself through a
> Record-Route'ing outbound proxy, then the tags it inserts had 
> better be
> different.  [I think you've heard that argument before!]
> 
> 
>   Still, bis shouldn't talk about any of this.  Keep the spec really
> simple, and let proxies which need to keep state figure out how to
> record-route to their advantage.  Jo's text is pretty much on 
> the mark.
> 
> 
>  [1]  Or, it can always just look at which address is at the 
> end of the
>       Route...  Regardless, your suggestion is an ugly hack.

Billy, I prefer to hold list discussions based on technical issues rather
than name calling.

Proxies route based on request URIs. I'll say it again. Proxies route based
on request URIs. As such, it is really important that the request URI,
without needing to examine the To/From, by itself, indicate the resource to
whom the request is targeted. 

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Date: Wed, 22 Nov 2000 17:16:20 -0600

Billy Biggs (Billy_Biggs@3com.com):

>   The proxy should look at the To and the From to tell direction,
> instead of doing this ugly hack! [1]

  To clarify myself, I refer to Jonathan's hack of having the proxy
mangle the Record-Route in the response which comes back.

  I should have quoted his message better.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Wed, 22 Nov 2000 17:24:10 -0600

Jonathan Rosenberg (jdrosen@dynamicsoft.com):

> Billy, I prefer to hold list discussions based on technical issues
> rather than name calling.

  I apologize.  I was very, very unnecessarily harsh.

> Proxies route based on request URIs. I'll say it again. Proxies route
> based on request URIs. As such, it is really important that the
> request URI, without needing to examine the To/From, by itself,
> indicate the resource to whom the request is targeted. 

  Say my proxy inserts as a record route:
     <sip:Billy_Biggs@3com.com>

  When a re-INVITE comes in, is it really addressed to
Billy_Biggs@3com.com?  No.  The request is actually addressed to some
state value for the active call inside the 3com.com proxy.  To me, it
would be much better for the proxy to use:
     <sip:897fd897fdh@3com.com>

  Which correctly identifies this state key.  It's not like the proxy
needs to look up and see how to route the message to
Billy_Biggs@3com.com.  This routing is handled by the Route.  No?

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Igor Slepchin wrote:
> 

> 
> Sounds fine except that I don't think this is ever referred to in 2543bis.
> In fact, the grammar for SIP-Version explicitly disables these tricks:
> 
> SIP-Version="SIP/2.0"
> 
> Note that SIP-Version is a literal. I'm not sure what the value of allowing
> "SIP/02.00" is, but if that's the intention, it should probably be mentioned
> in the spec. BTW, the change would not be backwards compatible; if it's
> made, it should probably be mentioned that leading zeros should not be
> generated.

I agree that adding leading zeros gains nothing. Given the low
likelihood of version changes, it doesn't seem worth treating it as
anything other than a string. For consistency, I will add the CI
description:

The string is case-insensitive, but implementations {\MUST} use    
upper-case. 
\motivation{Unlike HTTP/1.1, SIP treats the version number as a literal
string.  In practice, this should make no difference.}



-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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From sip-admin@lists.bell-labs.com  Wed Nov 22 21:28:04 2000
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Date: Wed, 22 Nov 2000 18:26:48 -0800

hi,
   I don't know what the parser should do and it seems that the parser has muddled me. Can you help me? I think it should check the message byte-by-byte to ensure that every character has fallen into a specific field, such as unreserved filed, DIGIT filed etc. Also the parser can find the syntax error besides the characters falling into some fields. For example, the parser should find the errors such as "%4q", 101.3332.3.5 (the Ipv4address) and so on in the Request-URL, although the characters are allowed to appear in the message. 
   Is that true? Also I am not sure the @ character can be escaped. If it can be escaped, the parser would not have mechanism to distinguish the one that the Spec requests with the one that the user's own escaped character? 
   Thanks.
   Robert Kajun


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Date: Wed, 22 Nov 2000 23:00:03 -0500 (EST)

Dear All,

We want to know whether it is possible to run a SIP Proxy Server and a
SIP User Agent on the same machine. Has anyone ever tried to do this? 

Arun.




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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
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Date: Wed, 22 Nov 2000 20:01:42 -0800

Hi Jonathan,

That is a very interesting suggestion. I like the motivation to have the
Route's different in each direction.

You said that in one direction the Route was

Route: A@proxy.blah.com

and in the other direction

Route: B@proxy.blah.com

Only the proxy understands the significance of A and B so is there any point
in proscribing that it must be of the above form, for instance, can you see
any disadvantages of:

Route: proxy.blah.com;rstate=A 

and 

Route: proxy.blah.com;rstate=B

If the answer is that you would like the base URL to be different for
robustness, ok that's fine. In that case A and B could just as easily be a
unique token, eg "1" & "2" or literally the characters "A" and "B"
(regardless of the request URI). Should we just leave it up to the proxy ?
Does the user part need to be present at all? 

Additionally, if the proxy is keeping full call state, eg based on unique
Call-Id. Is it REQUIRED to mess with the Record-Route in the response.
Otherwise, adding a simple

Record-Route: proxy.blah.com


would be sufficient. [If the request spirals it will need to add something
different the second time but it still doesn't need to be different in each
direction.] What I am getting at is are you trying to make this a
requirement for the sake of protocol "invariant-ness" or merely as a helpful
recommendation for proxies not wanting to keep full state. ie, SHOULD or
MUST

Regardless of the above handling by the proxies, it seems that we have
reached a consensus that the UAS should no longer mangle the reverse Route
which I think is an important point. I believe it will be a lot easier to
change how proxies handle the Record-Route processing than the behaviour of
the UA's already out there - so the sooner the UA behaviour is settled the
better. It also makes the UA behaviour a lot simpler - which is always a
good thing. Agreed?

Regards,

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 



> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Thursday, November 23, 2000 7:12 AM
> To: 'Billy Biggs'; Jonathan Rosenberg
> Cc: Jo Hornsby; hgs@cs.columbia.edu; sip@lists.bell-labs.com
> Subject: RE: [SIP] New Section 6.34 (was: Record-Route consensus)
> 
> 
> 
> 
>  
> 
> > -----Original Message-----
> > From: Billy Biggs [mailto:Billy_Biggs@3com.com]
> > Sent: Wednesday, November 22, 2000 6:10 PM
> > To: Jonathan Rosenberg
> > Cc: Jo Hornsby; hgs@cs.columbia.edu; sip@lists.bell-labs.com
> > Subject: Re: [SIP] New Section 6.34 (was: Record-Route consensus)
> > 
> > 
> > Jonathan Rosenberg (jdrosen@dynamicsoft.com):
> > 
> > > 5. (this one is my own) The request URI should refer to 
> the resource
> > > for whom the message is destined
> > > 
> > > The solutions proposed are either to have the UAS and UAC 
> construct
> > > the Route differently (as is done in the current bis), else 
> > Jo's text
> > > here where this information is carried somehow as 
> > parameters. I think
> > > this is a big hack, and quite a pain. When a proxy receives a
> > > re-INVITE, it will have a hard time knowing which direction the
> > > re-INVITE is going, and consequently, which of the pieces 
> of info to
> > > extract for processing purposes.
> > 
> > 
> >   The proxy should look at the To and the From to tell direction,
> > instead of doing this ugly hack! [1]
> > 
> >   The only note then is that if a UA calls itself through a
> > Record-Route'ing outbound proxy, then the tags it inserts had 
> > better be
> > different.  [I think you've heard that argument before!]
> > 
> > 
> >   Still, bis shouldn't talk about any of this.  Keep the spec really
> > simple, and let proxies which need to keep state figure out how to
> > record-route to their advantage.  Jo's text is pretty much on 
> > the mark.
> > 
> > 
> >  [1]  Or, it can always just look at which address is at the 
> > end of the
> >       Route...  Regardless, your suggestion is an ugly hack.
> 
> Billy, I prefer to hold list discussions based on technical 
> issues rather
> than name calling.
> 
> Proxies route based on request URIs. I'll say it again. 
> Proxies route based
> on request URIs. As such, it is really important that the request URI,
> without needing to examine the To/From, by itself, indicate 
> the resource to
> whom the request is targeted. 
> 
> -Jonathan R.
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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> 
> A related question is whether any SIP implementations actually handle
> early media. It seems harmless to have in pure IP systems, if it gets
> never exercised by gateways.
>

Henning, you should be able to get this information from the SIP bake off
registrations, right?

From previous bakeoffs, I remember there is usually a number of companies
with which early media can be tested.

Regards,

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 

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> We want to know whether it is possible to run a SIP Proxy Server and a
> SIP User Agent on the same machine. Has anyone ever tried to do this? 

Yes -- it is possible.  In general they are going to have to listen
on different ports.  (I say in general because its possible to have an
application that functions as a UA for one transaction, and a
Proxy for another.)


 - Jo.


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Date: Thu, 23 Nov 2000 09:47:55 -0000

Hi,

Do any of you have any experience of the DynamicSoft SIP user agent?

-Paul

To iterate is human,
to recurse divine.
- L. Peter Deutsch 
 (author of Ghostscript)



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Jonathan Rosenberg wrote:
>
> Before we begin writing new text,

Ah, but I finally got you to bite. &;)

> else Jo's text here where this information is carried somehow as
> parameters. I think this is a big hack, and quite a pain.

I feel this is a little unfair; ultimately, I specified very
little about the Request-URI that the proxy inserted, beyond
it being a globally reachable address for the proxy.  It was
only a motivational item and the examples at the end that
suggested using a parameter to indicate the original
Request-URI.

> When a proxy receives a re-INVITE, it will have a hard time
> knowing which direction the re-INVITE is going, and
> consequently, which of the pieces of info to extract for
> processing purposes.

As Billy pointed out, making this determination isn't too
difficult.

> When a request arrives, and a proxy record-routes, it places a 
> URI into the RR which identifies the *calling UA* as a resource
> on this proxy. This URI can be derived from the Contact/From,
> but should not be exactly equal to it.  This must be a SIP URL.
> THe UAS copies the RR set into the response. It also constructs
> a Route header by taking the RR's in order, and adding the
> Contact from the INVITE at the end. Now, here's the trick. As
> the response traverses the proxies, any proxy that record-routed,
> finds its RR entry, and *replaces it* with a new URI; this URI
> identifies the called party at this server. Probably this is
> the request URI that was originally received in the INVITE.
> The UAC then constructs the Route as defined in rfc2543.

Very nice -- it's cool that you've found a way to maintain
"the Request-URI points to the ultimate destination" as an
invariant.

So instead of:

  Each Record-Route header field contains a globally reachable SIP URL
 that identifies a proxy server.  Any proxy that wishes to remain in
 the signalling path of a call leg adds its own SIP URL to the
 beginning of the list of these fields.  This SIP URL MUST represent an
 address that is reachable via UDP.  Requests between both user agents
 involved in the call leg, in either direction, traverse this route.

how about:

  Each Record-Route header field contains a globally reachable SIP URL
 that identifies a proxy server.  Any proxy that wishes to remain in
 the signalling path of a call leg adds its own SIP URL to the
 beginning of the list of these fields in a request.  This SIP URL
 MUST represent an address that is reachable via UDP, and SHOULD
 represent the calling UA (i.e., be derived from the Contact/From).
 In addition, a proxy SHOULD modify its URL in the corresponding
 response(s) to reflect the called UA (i.e., be derived from the
 original Request-URI).  Requests between both user agents
 involved in the call leg, in either direction, traverse this route.

could also perhaps add some motivational text indicating how this
behaviour helps to maintain the Request-URI semantic invariant.

Moving on, I wrote:
> >
> >  *** Should we remove this bit about 401 and 484?  It makes things
> >      more complicated, and also what if a proxy has collected a load
> >      of WWW-Authenticate and Proxy-Authenticate headers into a
> >      401 (Section 12.4 of bis2) -- the Record-Route could be 
> >      arbitrary.
[and]
> >  *** I haven't said initial INVITE here, since wasn't there 
> >      some talk about abstracting it for, say SUBSCRIBE/NOTIFY
> >      purposes?
[and]
> >  *** If no Contact, should add the From?

Now that we've got the overall operation sorted out &:), does
anybody have anything to contribute with regards to these
points?  Please? &:)

Cheers,


 - Jo.
 

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From: Markus.Isomaki@nokia.com
To: Billy_Biggs@3com.com, BMoloney@sonusnet.com
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Subject: RE: [SIP] Registration Question
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Date: Thu, 23 Nov 2000 12:48:44 +0200

Hi,

In 3GPP (third generation mobile networks standardisation) the situation is
that a UA always has an outbound proxy, who's address is learned using DHCP
(or some other methods). If the user is roaming, this proxy is located in
the visited network, and can offer local services (911 etc.) 

In addition each user has a "Serving proxy" (in 3GPP terminology Serving
Call State Control Function, S-CSCF), which is always situated in his home
network/domain. This proxy performs service control for both outgoing and
incoming sessions. So, each outgoing INVITE (actually any request) from a
user first traverses the outbound proxy and then the "serving proxy", which
then decides what to do with the request based on service control:  

UA--->Outboud proxy (visited nw)--->Serving proxy (home nw)--->

The serving proxy is assigned dynamically to the user during registration,
e.g. using a 3xx response to REGISTER. The reason might be different service
control preferences/capabilities or just load balancing.

The problem is the routing of requests from the outbound proxy (visited nw)
to serving proxy (home nw). Basically, the outbound proxy can check the
user's domain name in From: field and route the request to (some proxy in)
the correct home domain. In the home network a database lookup (or SIP
redirect) would then be used to determine which S-CSCF is currently
allocated to the user and forward the request there. (This is analogous to
what happens with incoming requests to the user in the home domain - S-CSCF
is located by a database query.)

UA--->Outbound proxy--->"Home domain proxy"--->DB lookup--->Serving proxy

However, there is a need to optimize the flow by allowing UA or the outbound
proxy to know the serving proxy's address directly. This could be done
during the registration. The serving proxy's address would need to be
carried in the 200 response to REGISTER, so that Outbound proxy and UA are
able to learn it. So the request routing would be directly:

UA--->Outbound proxy--->serving proxy

In my understanding SIP does not support this at the moment, so there have
been discussions whether to use Record-Route or some new header. So, there
would actually be some use for what Bill Moloney was asking, at least in
mobile networks.

Rgs,
	Markus

Billy Biggs:
> Moloney, Bill (BMoloney@sonusnet.com):
> 
> > I have a question on registration -
> > [...]
> > I believe that I could accomplish this by having B respond to the
> > registration with a 301 Moved Permanently, including a 
> Contact header
> > for C in the response.)
> > 
> > The question is this: Does this redirection only impact future
> > registration requests, or does it impact all future 
> requests - such as
> > INVITE?
> 
>   The redirection only impacts the specific registration 
> transaction and
> (optionally) future registration transactions.
> 
>   Requests such as INVITE go to their destination directly, and have
> nothing to do with the registration.
> 
> > What I am trying to accomplish is to have A direct all of its future
> > requests (not just registration) to the new proxy server 
> (C). Will the
> > redirect that I mention above accomplish this?
> 
>   This is called an outbound proxy, where even though the message is
> directed to, say, 'sip:5551212@sip.billybiggs.com', you send 
> the message
> to 'my.local.proxy.com' which then forwards it on.
> 
>   The outbound proxy is configured directly on the SIP end 
> device, or it
> can receive it from DHCP.  There is no way to signal using REGISTER or
> whatever that the proxy should be used as an outbound proxy, nor would
> it make sense to provide such signalling.
> 
> -- 
> Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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From sip-admin@lists.bell-labs.com  Thu Nov 23 06:27:06 2000
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Subject: [SIP] Inviting users to join a centralized conference
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Date: Thu, 23 Nov 2000 12:26:41 +0100

Hi,

In your I-D draft-rosenberg-sip-conferencing-models-00, you omitted to mention the possibility of inviting a user to join a centralized conference by sending the SIP REFER to the conference server. i.e. If user A wishes to invite user B to join, A could send the conference server a REFER :

REFER sip:conference34@servers.com SIP/2.0
From: sip:A@example.com
To: sip:conference34@servers.com
Refer-To: sip:B@example.com

The conference server can then decide whether to send an INVITE to user B depending on some criteria (eg. the number of participants, current resources, etc). In this way, we can avoid ringing user B if say the conference is already full.

Was this omission intentional ?

Ya-Ching Tan
________________________________________________________
ICM N MC MI E7		Tel.:	+49 89 722 36147
Hofmannstr. 51		Email:	ya-ching.tan@icn.siemens.de
81359 Munich		
Germany



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From sip-admin@lists.bell-labs.com  Thu Nov 23 07:17:09 2000
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From: "Hisham Khartabil" <hisham.khartabil@hotsip.com>
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Subject: [SIP] minor error in section 11.4 of bis
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Date: Thu, 23 Nov 2000 14:17:53 +0200
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Section 11.4 states in the third  bullet point about determining the network
destination and request-uri:

"Otherwise, if the UAC is configured with the address of an outbound proxy
server, the UAC sends
the request there. The Request-URI contains the same URL as the To header
field if the UAC is the
callee. It copies the URL and To header field from the caller's From header
if the UAC is the callee."

I believe the second sentence should say "The Request-URI contains the same
URL as the To header field if the UAC is the calleR".

Regards,
Hisham Khartabil
Hotsip Finland
www.hotsip.com


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From sip-admin@lists.bell-labs.com  Thu Nov 23 07:40:07 2000
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Date: Thu, 23 Nov 2000 04:39:35 -0800

Hi Jo,
 
> Very nice -- it's cool that you've found a way to maintain
> "the Request-URI points to the ultimate destination" as an
> invariant.

I still don't see how this is the case. You just have two different URL's
located at the proxy server. I don't see they point to the ultimate
destination (unless I missing something fundamental). 
 
>   Each Record-Route header field contains a globally reachable SIP URL
>  that identifies a proxy server.  Any proxy that wishes to remain in
>  the signalling path of a call leg adds its own SIP URL to the
>  beginning of the list of these fields in a request.  This SIP URL
>  MUST represent an address that is reachable via UDP, and SHOULD
>  represent the calling UA (i.e., be derived from the Contact/From).

As above, I don't understand how it "represents" the UA in any sense outside
the proxy's internal implementation. I am not saying that this is a bad idea
but I do think this wording will cause a lot of confusion. Do you derive it
from the just the user part of the Contact/From ? What if the user name is
the same (jack@A <-> jack@B?)? Are you suggestign we put an escaped copy of
the whole Contact/From into the username portion of the Route ? Why not just
say 

"This SIP URL MUST represent an address that is reachable via UDP, and the
user portion SHOULD be distinguishable by the proxy as being destined for
the calling UA. The URL SHOULD also be distguishable from any previous
Record-Route entries added by the proxy (e.g., due to spiraling routes)." 

>  In addition, a proxy SHOULD modify its URL in the corresponding
>  response(s) to reflect the called UA (i.e., be derived from the
>  original Request-URI).  Requests between both user agents
>  involved in the call leg, in either direction, traverse this route.

> could also perhaps add some motivational text indicating how this
> behaviour helps to maintain the Request-URI semantic invariant.
> 
> > >  *** Should we remove this bit about 401 and 484?  It makes things
> > >      more complicated, and also what if a proxy has 
> collected a load
> > >      of WWW-Authenticate and Proxy-Authenticate headers into a
> > >      401 (Section 12.4 of bis2) -- the Record-Route could be 
> > >      arbitrary.

I have no opinion on this one.

> > >  *** I haven't said initial INVITE here, since wasn't there 
> > >      some talk about abstracting it for, say SUBSCRIBE/NOTIFY
> > >      purposes?

I agree that we need to make the "pre- initial INVITE" behaviour very clear
because in mind the descriptions in Sections 11 & 12 are not so clear on
this.

> > >  *** If no Contact, should add the From?

Is there a reason why we should change the existing behaviour ?

Cheers,

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. --  

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Robert Fairlie-Cuninghame wrote:
> > Very nice -- it's cool that you've found a way to maintain
> > "the Request-URI points to the ultimate destination" as an
> > invariant.
>
> I still don't see how this is the case. You just have two different
> URL's located at the proxy server. I don't see they point to the
> ultimate destination (unless I missing something fundamental).

They might not point to the ultimate destination, per se; but
the Request-URI will be meaningful enough to the proxy such
that it could reapply its initial routing decisions, if
necessary.

> >   Each Record-Route header field contains a globally reachable SIP URL
> >  that identifies a proxy server.  Any proxy that wishes to remain in
> >  the signalling path of a call leg adds its own SIP URL to the
> >  beginning of the list of these fields in a request.  This SIP URL
> >  MUST represent an address that is reachable via UDP, and SHOULD
> >  represent the calling UA (i.e., be derived from the Contact/From).
>
> As above, I don't understand how it "represents" the UA in any
> sense outside the proxy's internal implementation. I am not saying
> that this is a bad idea but I do think this wording will cause a
> lot of confusion.

I see what you're saying -- my wording is confusing upon reading
it back.  Hmmm...

> Do you derive it from the just the user part of the Contact/From ?
> What if the user name is the same (jack@A <-> jack@B?)? Are you
> suggestign we put an escaped copy of the whole Contact/From into
> the username portion of the Route ?

This is probably what I would do, in fact.

> Why not just say
>
> "This SIP URL MUST represent an address that is reachable via UDP, and the
> user portion SHOULD be distinguishable by the proxy as being destined for
> the calling UA. The URL SHOULD also be distguishable from any previous
> Record-Route entries added by the proxy (e.g., due to spiraling routes)."

Basically because I'd rather we imply as little as possible about
the format of this URL; however, I think we're getting warmer.

How about:
    This SIP URL MUST represent an address that is reachable via
    UDP, and SHOULD somehow be meaningful to the proxy, such that
    it could map the URL back to the calling UA (e.g., by embedding
    the Contact/From ha a parameter).
?

> > > >  *** If no Contact, should add the From?
>
> Is there a reason why we should change the existing behaviour ?

I wasn't sure what the existing behaviour was...


 - Jo.


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From: Anders Kristensen <akristensen@dynamicsoft.com>
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: "'Jo Hornsby'" <jhornsby@ubiquity.net>, hgs@cs.columbia.edu,
        sip@lists.bell-labs.com, Billy_Biggs@3com.com
Subject: Re: [SIP] New Section 6.34 (was: Record-Route consensus)
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Date: Thu, 23 Nov 2000 16:21:08 +0100
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Jonathan Rosenberg wrote:
> 
> Before we begin writing new text, lets look at the sub-problems and build up
> a solution. I have another possibility that may be the best of all worlds.
> I'll start with Roberts list:
> 
> 1. non SIP URLs in request URI
> 
> We can either define record-route parameters, or else make sure that the URL
> put into the record route is a sip url. I originally propose RR parameters,
> but I think insisting on a SIP URL is backwards compatible and more
> reasonable. That seems to be consensus.
> 
> 2. transaction ID in the reverse direction
> 
> This was the problem that if there was a spiral in the forward direction,
> and the current bis approach was used by the UAS to construct the Route
> header, a request would arrive twice at the spiraled-at proxy. These could
> be identified as separate transactions (as a result of Via header
> differences), but there is no way to determine which of the two iterations
> through it is.
> 
> The solution is that the request URI need some info that the proxy created
> and is returned to it. There seems to be little disagreement on that.
> 
> 3. Proxy doesn't see URI parameters it inserted in RR, returned to it in
> request URI
> 
> This is a consequence of the current bis text for building the route.
> Proposed solution for this component is that the construction process at the
> UAS has to copy the URI parameters from the RR rather than the Contact/From.
> 
> 4. detecting new behavior
> 
> I don't think this is an issue, since we need a solution thats backwards
> compatible.
> 
> 5. (this one is my own) The request URI should refer to the resource for
> whom the message is destined
> 
> The solutions proposed are either to have the UAS and UAC construct the
> Route differently (as is done in the current bis), else Jo's text here where
> this information is carried somehow as parameters. I think this is a big
> hack, and quite a pain. When a proxy receives a re-INVITE, it will have a
> hard time knowing which direction the re-INVITE is going, and consequently,
> which of the pieces of info to extract for processing purposes.
> 
> Putting it all together, its clear from (1) that the record routes must
> always be SIP URLs. From 2 and 3, at least the parameters placed into the RR
> should be copied into the Route by both sides, and from 5, the URI should
> contain information that identifies the recipient of the message, different
> in both directions.
> 
> My new proposal does all of the above, and requires no mangling of RR in the
> UA. Here's how it goes.
> 
> When a request arrives, and a proxy record-routes, it places a URI into the
> RR which identifies the *calling UA* as a resource on this proxy. This URI
> can be derived from the Contact/From, but should not be exactly equal to it.
> This must be a SIP URL. THe UAS copies the RR set into the response. It also
> constructs a Route header by taking the RR's in order, and adding the
> Contact from the INVITE at the end. Now, here's the trick. As the response
> traverses the proxies, any proxy that record-routed, finds its RR entry, and
> *replaces it* with a new URI; this URI identifies the called party at this
> server. Probably this is the request URI that was originally received in the
> INVITE. The UAC then constructs the Route as defined in rfc2543.

Like Robert I struggle with this RR URI which "identifies the
[caller/callee] as a resource on this proxy". In the typical case only
one of the two parties would be known in any meaningful way to the
proxy.  The proxy record-routes using its own SIP URL and can derive a
user name from the caller/callee but they won't necessraily mean
anything to it, and it wouldn't necessarily be able to route messages on
it (and it won't have to - that's what the Route is for). This means [5]
isn't really satisfied anyway.

> ...
> This is similar to Jo's proposal but maintains the property that a request
> from A to B has a URI that always points to a resource that gets routed to
> B, and vice a versa (in Jo's case, the URI is the same in both directions).
> Thus, I think it satisfies everyone. Note that maddr is no longer needed so
> long as the host port gets routed back to the same server; of course it can
> put maddr if it likes.
> 
> The drawback is that the proxy needs to do a bit more work. Not a lot,
> though.

Well, until now proxies just had to pop the top Via and forward the
response, with this proposal they have to decode some or all
Record-Route headers, figure out which one it generated itself and
modify it.

It might be tempting for proxy writers NOT to put a user part into its
own Record-Route headers to begin with. That way there's less work on
the return path and routing works exactly the same.  Alternatively, as
has also been suggested, proxies can put whatever private state they
like in there and just consider it a piece of state which identifies the
call to that proxy.

It seems to me that Jonathans and Jo's proposals degenerate to the same
approach: always use SIP URLs in Record-Route and not make any
assumptions what goes into the user part.

Anders

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Date: Thu, 23 Nov 2000 10:38:35 -0500

I do :)

-Jonathan R.

---
Jonathan Rosenberg
dynamicsoft
jdrosen@dynamicsoft.com

-----Original Message-----
From: Paul Duncan <Paul.Duncan@LOSTWAX.COM>
To: sip@lists.bell-labs.com <sip@lists.bell-labs.com>
Sent: Thu Nov 23 04:47:55 2000
Subject: RE: [SIP] Help with sip

Hi,

Do any of you have any experience of the DynamicSoft SIP user agent?

-Paul

To iterate is human,
to recurse divine.
- L. Peter Deutsch 
 (author of Ghostscript)



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To: Hisham Khartabil <hisham.khartabil@hotsip.com>
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Subject: Re: [SIP] minor error in section 11.4 of bis
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The section was somewhat confusing, so I rewrote it for the next
edition.

Hisham Khartabil wrote:
> 
> Section 11.4 states in the third  bullet point about determining the network
> destination and request-uri:
> 
> "Otherwise, if the UAC is configured with the address of an outbound proxy
> server, the UAC sends
> the request there. The Request-URI contains the same URL as the To header
> field if the UAC is the
> callee. It copies the URL and To header field from the caller's From header
> if the UAC is the callee."
> 
> I believe the second sentence should say "The Request-URI contains the same
> URL as the To header field if the UAC is the calleR".

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From: Arun Kumar Chippada <chippada@cse.msu.edu>
To: Jo Hornsby <jhornsby@ubiquity.net>
Cc: sip@lists.bell-labs.com
Subject: RE: [SIP] Help with sip
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Date: Thu, 23 Nov 2000 14:19:14 -0500 (EST)

We are running the SIP Proxy Server and the SIP User Agent on the same
machine. We are using the Columbia University's sipd as our proxy server
and their sipua as the user agent. The sipd is listening on the default
port 5060. The sipua is started on port 5444. We verified that they are
listening properly like this on 2 different ports using "netstat -a"

Then we  tried to register a user with user name "chippada" and domain
name "cse.msu.edu" sending the register request from the sipua to
sipd. The machine that we are running the sipd as well as the sipua has IP
address "10.0.0.16" and name "johan.cse.msu.edu"

So we have the following session using the sipua and the output printed by
the sipua.



sipua> register sip:chippada@cse.msu.edu sip:10.0.0.16
Registering to server as sip:chippada@cse.msu.edu at sip:10.0.0.16 with
contact sip:chippada@johan.cse.msu.edu:5444
REGISTER sip:10.0.0.16 SIP/2.0
Via: SIP/2.0/UDP johan.cse.msu.edu:5444; branch=4044117737-0
From: sip:chippada@cse.msu.edu
To: sip:chippada@cse.msu.edu
CSeq: 1 REGISTER
Call-ID: 4163501107@johan.cse.msu.edu
Expires: 3600
Contact: sip:chippada@johan.cse.msu.edu:5444
Content-Length: 0


975005101.235 ReceiveUDP(): received UDP packet (length 299) from
10.0.0.16:1052:
SIP/2.0 482 Loop detected
Via: SIP/2.0/UDP johan.cse.msu.edu:5444; branch=4044117737-0
From: sip:chippada@cse.msu.edu
To: sip:chippada@cse.msu.edu
Call-ID: 4163501107@johan.cse.msu.edu
Cseq: 1 REGISTER
Date: Thu, 23 Nov 2000 18:45:01 GMT
Server: Columbia-SIP-Server/1.0
Content-Length: 0 

We are thinking that the sipua is able to contact the sipd but the sipd is
not registering the user  and instead sending the message "Loop Detected".
Does this have to do something with running the user agent and the
proxy server on the same machine? Or are we doing something wrong in the
register request?

Thankyou,
Arun.



On Thu, 23 Nov 2000, Jo Hornsby wrote:

> > We want to know whether it is possible to run a SIP Proxy Server and a
> > SIP User Agent on the same machine. Has anyone ever tried to do this? 
> 
> Yes -- it is possible.  In general they are going to have to listen
> on different ports.  (I say in general because its possible to have an
> application that functions as a UA for one transaction, and a
> Proxy for another.)
> 
> 
>  - Jo.
> 


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From: Kundan Singh <kns10@cs.columbia.edu>
To: Arun Kumar Chippada <chippada@cse.msu.edu>
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Subject: RE: [SIP] Help with sip
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Date: Thu, 23 Nov 2000 14:29:30 -0500 (EST)


Hi Arun,

We have found and fixed this problem a couple of weeks ago.
I will directly contact you soon regarding the new version.

Also, please do not use the SIP list for discussion
related to a specific implementation.

--
Kundan Singh     http://www.cs.columbia.edu/~kns10

On Thu, 23 Nov 2000, Arun Kumar Chippada wrote:

> We are running the SIP Proxy Server and the SIP User Agent on the same
> machine. We are using the Columbia University's sipd as our proxy server
> and their sipua as the user agent. The sipd is listening on the default
> port 5060. The sipua is started on port 5444. We verified that they are
> listening properly like this on 2 different ports using "netstat -a"
>
> Then we  tried to register a user with user name "chippada" and domain
> name "cse.msu.edu" sending the register request from the sipua to
> sipd. The machine that we are running the sipd as well as the sipua has IP
> address "10.0.0.16" and name "johan.cse.msu.edu"
>
> So we have the following session using the sipua and the output printed by
> the sipua.
>
>
>
> sipua> register sip:chippada@cse.msu.edu sip:10.0.0.16
> Registering to server as sip:chippada@cse.msu.edu at sip:10.0.0.16 with
> contact sip:chippada@johan.cse.msu.edu:5444
> REGISTER sip:10.0.0.16 SIP/2.0
> Via: SIP/2.0/UDP johan.cse.msu.edu:5444; branch=4044117737-0
> From: sip:chippada@cse.msu.edu
> To: sip:chippada@cse.msu.edu
> CSeq: 1 REGISTER
> Call-ID: 4163501107@johan.cse.msu.edu
> Expires: 3600
> Contact: sip:chippada@johan.cse.msu.edu:5444
> Content-Length: 0
>
>
> 975005101.235 ReceiveUDP(): received UDP packet (length 299) from
> 10.0.0.16:1052:
> SIP/2.0 482 Loop detected
> Via: SIP/2.0/UDP johan.cse.msu.edu:5444; branch=4044117737-0
> From: sip:chippada@cse.msu.edu
> To: sip:chippada@cse.msu.edu
> Call-ID: 4163501107@johan.cse.msu.edu
> Cseq: 1 REGISTER
> Date: Thu, 23 Nov 2000 18:45:01 GMT
> Server: Columbia-SIP-Server/1.0
> Content-Length: 0
>
> We are thinking that the sipua is able to contact the sipd but the sipd is
> not registering the user  and instead sending the message "Loop Detected".
> Does this have to do something with running the user agent and the
> proxy server on the same machine? Or are we doing something wrong in the
> register request?
>
> Thankyou,
> Arun.
>
>
>
> On Thu, 23 Nov 2000, Jo Hornsby wrote:
>
> > > We want to know whether it is possible to run a SIP Proxy Server and a
> > > SIP User Agent on the same machine. Has anyone ever tried to do this?
> >
> > Yes -- it is possible.  In general they are going to have to listen
> > on different ports.  (I say in general because its possible to have an
> > application that functions as a UA for one transaction, and a
> > Proxy for another.)
> >
> >
> >  - Jo.
> >
>
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>


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Subject: [SIP] Calling oneself
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Date: Thu, 23 Nov 2000 15:11:32 -0500
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I'm not sure, but it seems that there's a simple fix: simply have the UA
create *different* tags depending on whether this is for a From or To.
However, the tag is constant across calls.

The general matching rule is

- match incoming requests so that the To matches the locally generated
tag and
  the From matches the tag generated by the other side

Modifying the running example to use tag=5 for the To tag

> I decide to call myself.  I create a call-leg struct to represent it.
> Let's call it "Line 1".  I send out the initial INVITE:
> 
>   [From Line 1]
>   INVITE sip:me@proxy
>   Via: SIP/2.0/UDP my.ip
>   To: <sip:me@proxy>
>   From: <sip:me@proxy>;tag=7
>   Call-ID: 5
> 
>   This eventually loops back to me.  Now I see coming in:
> 
>   INVITE sip:me@proxy
>   Via: SIP/2.0/UDP some.ip
>   To: <sip:me@proxy>
>   From: <sip:me@proxy>;tag=7
>   Call-ID: 5
> 
>   I check and see there is no To-tag, so this is a new incoming call.  I
> create a call struct to represent it, called "Line 2".  Line 2 now
> begins ringing.  I respond with a 180 and the user switches lines and
> answers the incoming call:
> 
>   [From Line 2]
>   SIP/2.0 200 OK
>   Via: SIP/2.0/UDP some.ip
>   To: <sip:me@proxy>;tag=5

Modification here.

>   From: <sip:me@proxy>;tag=7
>   Call-ID: 5
> 
>   When I receive this response, I find the related pending transaction.
> Arguably, I could first find which call-leg it is destined to (and
> immediately have a problem), but say that I do transaction mapping first
> so we don't break yet.  I match the OK and send an ACK:
> 
>   [From Line 1]
>   ACK sip:me@contactaddr
>   Via: SIP/2.0/UDP my.ip
>   To: <sip:me@proxy>;tag=5
>   From: <sip:me@proxy>;tag=7
>   Call-ID: 5
> 
>   At which point the other pending transaction also correctly matches
> this ACK to it.
> 
>   Now, on Line 2, I decide to hang up.  I've had enough of talking to
> myself.  So, I send a BYE:
> 

As usuall, the callee simply flips To and From.


>   [From Line 2]
>   BYE sip:me@contact.addr
>   Via: SIP/2.0/UDP my.ip
>   To: <sip:me@proxy>;tag=7
>   From: <sip:me@proxy>;tag=5
>   Call-ID: 5
> 
>   I now see this incoming BYE:
> 
>   BYE sip:me@contact.addr
>   Via: SIP/2.0/UDP some.ip
>   To: <sip:me@proxy>;tag=7
>   From: <sip:me@proxy>;tag=5
>   Call-ID: 5
> 

Using the rule above, it matches the call on line 1, since 7 is the
locally-generated tag (stuck into From).

> 
>   Using unique tags per call avoids any confusion even in this case, and
> also protects the UA against clients which always use the same tag and
> happen to make two calls with the same Call-ID.

Since both Line1 and Line2 have the same call-ID (one incoming and one
outgoing), they would have the same locally generated tag according to
your rule, so I don't see how this helps.

Two calls with the same call-id can't happen, by definition. If they do,
the implementation is broken (and would presumably generate the "wrong"
tag, too, while it is at it).

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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
To: "'Tan Ya-Ching'" <Ya-Ching.Tan@icn.siemens.de>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        "'Henning G. Schulzrinne'" <hgs@cs.columbia.edu>
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Subject: RE: [SIP] Inviting users to join a centralized conference
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Date: Thu, 23 Nov 2000 16:04:47 -0500

No, the omission was not intentional, and can be done as well. I can include
in the next version of the document.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Tan Ya-Ching [mailto:Ya-Ching.Tan@icn.siemens.de]
> Sent: Thursday, November 23, 2000 6:27 AM
> To: 'Jonathan Rosenberg'; 'Henning G. Schulzrinne'
> Cc: 'sip@lists.bell-labs.com'
> Subject: [SIP] Inviting users to join a centralized conference
> 
> 
> Hi,
> 
> In your I-D draft-rosenberg-sip-conferencing-models-00, you 
> omitted to mention the possibility of inviting a user to join 
> a centralized conference by sending the SIP REFER to the 
> conference server. i.e. If user A wishes to invite user B to 
> join, A could send the conference server a REFER :
> 
> REFER sip:conference34@servers.com SIP/2.0
> From: sip:A@example.com
> To: sip:conference34@servers.com
> Refer-To: sip:B@example.com
> 
> The conference server can then decide whether to send an 
> INVITE to user B depending on some criteria (eg. the number 
> of participants, current resources, etc). In this way, we can 
> avoid ringing user B if say the conference is already full.
> 
> Was this omission intentional ?
> 
> Ya-Ching Tan
> ________________________________________________________
> ICM N MC MI E7		Tel.:	+49 89 722 36147
> Hofmannstr. 51		Email:	ya-ching.tan@icn.siemens.de
> 81359 Munich		
> Germany
> 
> 

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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: RE: [SIP] New Section 6.34 (was: Record-Route consensus)
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Date: Thu, 23 Nov 2000 16:27:23 -0500



 

> -----Original Message-----
> From: Anders Kristensen [mailto:akristensen@dynamicsoft.com]
> Sent: Thursday, November 23, 2000 10:21 AM
> To: Jonathan Rosenberg
> Cc: 'Jo Hornsby'; hgs@cs.columbia.edu; sip@lists.bell-labs.com;
> Billy_Biggs@3com.com
> Subject: Re: [SIP] New Section 6.34 (was: Record-Route consensus)
> 
> 
> Like Robert I struggle with this RR URI which "identifies the
> [caller/callee] as a resource on this proxy". In the typical case only
> one of the two parties would be known in any meaningful way to the
> proxy.  The proxy record-routes using its own SIP URL and can derive a
> user name from the caller/callee but they won't necessraily mean
> anything to it, and it wouldn't necessarily be able to route 
> messages on
> it (and it won't have to - that's what the Route is for). 
> This means [5]
> isn't really satisfied anyway.

My initial posting was not sufficiently detailed to explain this, so let me
clarify.

The URL inserted into the request must be something that (1) given the
client follows the server location process in draft-ietf-sip-srv-00, causes
the URL to be resolved to an address/port on the proxy, and (2) is a URL
that, if received by the proxy, the routing logic in the proxy would cause
the request to be forwarded to the Contact/From address of the calling
party. Similarly, the URL in the response must be something that (1) routes
to the proxy, and (2) is a URL whose routing logic in the proxy would cause
the request to be forwarded to the Contact/From address of the called party.

Since we don't dictate routing behavior, so too would the format of these
URLs not be specified. So, if we assume the caller is A@originating.com, and
the called party is B@terminating.com, one choice for the URL pair (request,
response) is:

(A@originating.com;maddr=proxy, B@terminating.com;maddr=proxy)

this tuple results in the same Route set getting constructed as gets
constructed by the current bis mechanism. The tuple satisfies the first
mandated property since the maddr causes the routing to go to the proxy. The
second is satisfied for any proxy that always forwards requests not in its
domain to the domain the R-URI, and for the originating.com proxy, knows how
to find A, and for terminating, knows how to find B.

Now, another way to structure the pair, that the terminating proxy could
use, is:

(A%64originating.com@server12.terminating.com, B@server12.terminating.com)

assuming its routing policy was that if the domain was its own, and the user
in the R-URI is unknown, the R-URI is URL-decoded. If it contains an @, the
resulting URL is used as the new request URI.

Now, consider the case of a proxy that has some kind of call state or
something, and a spiral comes through it, and it needs to differentiate each
pass through the spiral. In this case, the two tuples it could use are:

(A@originating.com;maddr=proxy;sessionid=76599,
B@terminating.com;maddr=proxy;sessionid=76599)

(A@originating.com;maddr=proxy;sessionid=876,
B@terminating.com;maddr=proxy;sessionid=876)

This satisfies that problem raised on the list with state and spirals.

Robert asked:

>Additionally, if the proxy is keeping full call state, eg based on unique
>Call-Id. Is it REQUIRED to mess with the Record-Route in the response.
>Otherwise, adding a simple

I think yes, since otherwise we won't achieve this property of the request
URI referring to a resource that would cause the request to be routed to the
appropriate party.

Now, things won't break if the proxies don't do this, but they become more
brittle, and various other subtle things might happen down the road. I had
come up with some other scenario where this property was needed, but don't
recall when it was.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Subject: [SIP] ACTION REQUESTED: Call for nominations for IAB/IESG
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The 2000 Nominations Committee is now
soliciting nominations for the open
slots on the IESG and IAB. 

Please take the time, even if it is
only a few moments, to send us your
thoughts. 

Members of the community are encouraged to 
nominate candidates for any of the positions 
which are up for re-filling: 

IESG Terms ending in March, 2001: 

Fred Baker		IETF Chair
Scott Bradner	Transport
Randy Bush		Operations & Management
Patrik Faltstrom	Applications
Thomas Narten	Internet
Dave Oran		Routing
Jeff Schiller	Security

IAB Terms ending in March, 2001: 

Harald Alvestrand 
Ran Atkinson
Rob Austein
Steve Deering
Tony Hain 
Geoff Huston

Members of the community are encouraged to contact the
nomcom to nominate someone for any of these positions, 
to provide input relating to potential candidates,
or to provide comments regarding the past performance
of incumbents, which will be considered
during the deliberations of the nominating committee. 

Comments sent to the nominations committee will be
kept in confidence. As noted in RFC 2727 "all 
deliberations and supporting information that 
relates to specific nominees, candidates, 
and confirmed candidates are confidential." 

Self-nominations are accepted. The deadline for
nominations is midnight Monday, December 18, 2000. 
Early input is desirable so that candidates may be 
interviewed during IETF 49. 

The members of the NomCom can be contacted individually  at
the addresses below or collectively at nomcom@ietf.org.  
Additionally, the members of the NomCom will be identified 
at the IETF by a marker (as yet undetermined) on their name 
tags.  People are encouraged to take advantage of the 
opportunity to talk to the members of the NomCom about 
nominations and other issues related to the IAB and IESG 
membership. 

The 2000 Nominations Committee

Phil Roberts <qa3445@email.mot.com>
Stuart Cheshire <cheshire@apple.com>
Perry E. Metzger <perry@piermont.com>
Bill Woodcock <woody@zocalo.net>
Hadriel Kaplan <hadriel@nortelnetworks.com>
Alain Durand <alain.durand@eng.sun.com>
Vijay Srinivasan <vijay.srinivasan@cosinecom.com>
Hal Sandick <hsandick@nortelnetworks.com>
David Robinson <david.robinson@ebay.sun.com>
David Allan <dallan@nortelnetworks.com>

Non-voting members:
Bernard Aboba <aboba@internaut.com> (Chair)
Avri Doria <avri@nortelnetworks.com> (Past Chair)
Leslie Daigle <leslie@thinkingcat.com> (IAB liason)
Allison Mankin <mankin@isi.edu> (IESG liason)

The role of non-voting members (from RFC 2727)

The nominations committee comprises at least a non-voting 
Chair, 10 voting volunteers, and 3 non-voting liaisons.
The sitting IAB and IESG members each appoint a non-voting
liaison to the nominating committee from their current
membership who are not sitting in an open position.
The Chair of the prior year's nominating committee also 
serves as a non-voting liaison.


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Subject: Re: [SIP] JAIN SIP - Issues to be resolved
References: <3A12B434.EE04C0E7@dynamicsoft.com> <3A130AB9.9A527F9E@nist.gov> <3A131F80.9060704@dynamicsoft.com> <3A16922C.EFDE9019@ericsson.com> <3A1A5BAC.A0975580@dynamicsoft.com>
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Date: Thu, 23 Nov 2000 20:30:37 -0500


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Chris, Erik,

While we are on the subject of Exceptions, why do null arguments throw runtime
exceptions when you do a get on them? For example, in the Accept interface,
getContentSubtype throws a IllegalArgumentException if  is null. IMHO, I think
it is valid to have a null subtype so why not just return null rather than throw
exception here?

I don't quite follow the argument that lazy parsing  "necessitates throwing a
parse exception for all get methods".   Here is what I had envisioned:


1 When a message comes in it will be separated into headers based on type.
2 The header would be returned with no field initialized except the text
corresponding to the message. (this would necessitate examining the first few
bytes of the message to determine the header type but the rest remains
unparsed.) You can also stow away the type information in the Header that is
returned as well to know which parser rule to invoke later.
3. Later, when the application wants to parse the header, it has the string
corresponding to the message and can hand it off to the parser, which can then
fill in the appropriate fields corresponding to the header.
4. Subsequently, the application can access the fields just as an eager parser
based application would do.

( Where am I missing the boat with this? )

It appears that what is being attempted here (please correct me if I am wrong)
is to use exceptions as a mechanism to catch un-initialized fields so the parser
can be invoked in the exception handler perhaps. This leads to a lot of overhead
in clunky code. I think it is a mistake to throw exceptions on every get to try
to protect the application from inadvertently accessing fields that have not
been initialized.  Why not just assume that the application must be smart enough
to parse the header text before accessing fields and not try to provide a
complex mechanism as above?


Regards,

Ranga.

Chris Harris wrote:

> Erik,
>
> The implication of what you say is that all set methods in the header
> interfaces which take String arguments must throw a checked
> JainSipParseException. I think this is reasonable given that different
> implementations will be more tolerant than others. However, in order to allow
> lazy parsing by an implementation, all get methods in the header interfaces
> (regardless of return type) must throw a checked parse exception. This is a
> lot of checked exceptions - and will make life more awkward for the
> application developer. It would certainly be much less effort to not allow
> lazy parsing - there would be no need to throw a parse exception for all get
> methods - greatly reducing the number of checked exceptions. Unfortunately it
> has been said that JAIN SIP support for lazy parsing is a must. If it is then
> I guess there is no other choice than to include the exceptions?
>
> Regards,
> Chris
>
> Erik Nissen wrote:
>
> > Chris Harris wrote:
> >
> > >
> > > We need to decide whether this should be a checked or runtime exception.
> >
> > The rule of thumb is to use:
> > * checked exceptions when the program should handle situations which
> > can/will occur.
> > * runtime exceptions when there are programming error (God forbid :-)
> >
> > So:
> > Parse errors should throw checked exceptions.
> > Becaurse parse errors are outside the control of the programmer
> > and he/she should be forced to handle such conditions in the code.
> >
> > Null/wrong arguments should throw runtime exceptions.
> >
> > If it is possible, reduce the number of methods throwing (checked)
> > exceptions
> > to a minimum.
> >
> > Best regards,
> > /Erik Nissen,
> > Ericsson Utveckling AB
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

--
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
Tel: 301 975 3664 Fax: 301 590 0932



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&nbsp;
<br>Chris, Erik,
<p>While we are on the subject of Exceptions, why do null arguments throw
runtime exceptions when you do a get on them? For example, in the Accept
interface, getContentSubtype throws a IllegalArgumentException if&nbsp;
is null. IMHO, I&nbsp;think it is valid to have a null subtype so why not
just return null rather than throw exception here?
<p>I&nbsp;don't quite follow the argument that lazy parsing&nbsp; "necessitates
throwing a parse exception for all get methods".&nbsp;&nbsp; Here is what
I&nbsp;had envisioned:
<br>&nbsp;
<p>1 When a message comes in it will be separated into headers based on
type.
<br>2 The header would be returned with no field initialized except the
text corresponding to the message. (this would necessitate examining the
first few bytes of the message to determine the header type but the rest
remains unparsed.) You can also stow away the type information in the Header
that is returned as well to know which parser rule to invoke later.
<br>3. Later, when the application wants to parse the header, it has the
string corresponding to the message and can hand it off to the parser,
which can then fill in the appropriate fields corresponding to the header.
<br>4. Subsequently, the application can access the fields just as an eager
parser based application would do.
<p>(&nbsp;Where am I&nbsp;missing the boat with this? )
<p>It appears that what is being attempted here (please correct me if I&nbsp;am
wrong) is to use exceptions as a mechanism to catch un-initialized fields
so the parser can be invoked in the exception handler perhaps. This leads
to a lot of overhead in clunky code. I&nbsp;think it is a mistake to throw
exceptions on every get to try to protect the application from inadvertently
accessing fields that have not been initialized.&nbsp; Why not just assume
that the application must be smart enough to parse the header text before
accessing fields and not try to provide a complex mechanism as above?
<br>&nbsp;
<p>Regards,
<p>Ranga.
<p>Chris Harris wrote:
<blockquote TYPE=CITE>Erik,
<p>The implication of what you say is that all set methods in the header
<br>interfaces which take String arguments must throw a checked
<br>JainSipParseException. I think this is reasonable given that different
<br>implementations will be more tolerant than others. However, in order
to allow
<br>lazy parsing by an implementation, all get methods in the header interfaces
<br>(regardless of return type) must throw a checked parse exception. This
is a
<br>lot of checked exceptions - and will make life more awkward for the
<br>application developer. It would certainly be much less effort to not
allow
<br>lazy parsing - there would be no need to throw a parse exception for
all get
<br>methods - greatly reducing the number of checked exceptions. Unfortunately
it
<br>has been said that JAIN SIP support for lazy parsing is a must. If
it is then
<br>I guess there is no other choice than to include the exceptions?
<p>Regards,
<br>Chris
<p>Erik Nissen wrote:
<p>> Chris Harris wrote:
<br>>
<br>> >
<br>> > We need to decide whether this should be a checked or runtime exception.
<br>>
<br>> The rule of thumb is to use:
<br>> * checked exceptions when the program should handle situations which
<br>> can/will occur.
<br>> * runtime exceptions when there are programming error (God forbid
:-)
<br>>
<br>> So:
<br>> Parse errors should throw checked exceptions.
<br>> Becaurse parse errors are outside the control of the programmer
<br>> and he/she should be forced to handle such conditions in the code.
<br>>
<br>> Null/wrong arguments should throw runtime exceptions.
<br>>
<br>> If it is possible, reduce the number of methods throwing (checked)
<br>> exceptions
<br>> to a minimum.
<br>>
<br>> Best regards,
<br>> /Erik Nissen,
<br>> Ericsson Utveckling AB
<p>_______________________________________________
<br>SIP mailing list
<br>SIP@lists.bell-labs.com
<br><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></blockquote>

<pre>--&nbsp;
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.&nbsp;
Tel: 301 975 3664 Fax: 301 590 0932</pre>
&nbsp;
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Hello!

This is a rather simple question. Of all the SIP headers in the bis
document, which can appear multiple times in a Request / Response ?
(For example,  VIA can appear multiple times, ACCEPT can appear multiple
times etc. but FROM can appear only once and TO can appear only once. )
If there is a table in the document giving this information  I have
missed,  please let me know (and apologies for redundant mail).

Thanks.

Regards

Ranga.

--
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
Tel: 301 975 3664 Fax: 301 590 0932



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Hello!
<p>This is a rather simple question. Of all the SIP&nbsp;headers in the
bis document, which can appear multiple times in a Request / Response ?&nbsp;
(For example,&nbsp; VIA&nbsp;can appear multiple times, ACCEPT&nbsp;can
appear multiple times etc. but FROM&nbsp;can appear only once and TO&nbsp;can
appear only once. ) If there is a table in the document giving this information&nbsp;
I&nbsp;have missed,&nbsp; please let me know (and apologies for redundant
mail).
<p>Thanks.
<p>Regards
<p>Ranga.
<pre>--&nbsp;
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.&nbsp;
Tel: 301 975 3664 Fax: 301 590 0932</pre>
&nbsp;
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Date: Thu, 23 Nov 2000 21:29:58 -0500

Since a comma separated list of values (indicated in BNF with #) is
identical to multiple headers of the same name, any header whose BNF starts
with # can appear more than once. 

-Jonathan R.
  
-----Original Message-----
>From: M. Ranganathan [mailto:mranga@nist.gov]
>Sent: Thursday, November 23, 2000 9:01 PM
>To: sip@lists.bell-labs.com
>Subject: [SIP] Multiple headers of a given type.
>
>
>Hello! 
>This is a rather simple question. Of all the SIP headers in the bis
document, which can 
>appear multiple times in a Request / Response ?  (For example,  VIA can
appear multiple 
>times, ACCEPT can appear multiple times etc. but FROM can appear only once
and TO can 
>appear only once. ) If there is a table in the document giving this
information  I have 
>missed,  please let me know (and apologies for redundant mail). 
>Thanks. 
>Regards 
>Ranga. 
-- 
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899. 
Tel: 301 975 3664 Fax: 301 590 0932
  

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
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jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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Date: Thu, 23 Nov 2000 22:36:47 -0500

Sorry for the delay in picking this thread up... I'm trying to finish a rev
on Caller Prefs, and there are some issues here.

First off, Eric raised a problem with my proposed solution. I'm replicating
it here since its been a while:

> If you remember, the problem was when doing call transfer 
> with consultation
> with the REFER request, how can we make sure that the INVITE 
> request (from
> the Transferee to the Transfer Target) is going to reach the 
> exact same
> User-Agent as the one that was consulted by the Transferor.  
> 
> The solution was to use the header "Accept-Contact" and 
> specify the contact
> of the transfer target.
> 
> I was taking a look at the caller preferences extension, thinking that
> "Accept-Contact" had solved our problem, but I think it does 
> not solves the
> problem.
> 
> Suppose we have the following participants:
> 
> Transferor (Tror)
> - Sip address: sip:tror@acme.com
> - Current location: sip:tror@office1.acme.com
> 
> Transferee (Tree)
> - Sip address: sip:tree@acme.com
> - Current location: sip:tree@office1.acme.com
>  
> Transfer-Target (TT)
> - Sip address: sip:tt@acme.com
> - Current location: sip:tt@office4.mediatrix.com
> 
> 
> Tror and Tree have established calls.  Tror contacts TT and 
> ask if he wishes
> to accept the call, which he does.  Tror sends the following 
> REFER to Tree:
> 
> REFER sip:tree@office1.acme.com SIP/2.0
> From: sip:tror@acme.com
> To: sip:tree@acme.com
> Refer-To: sip:tt@acme.com?Accept-Contact=sip:tt@office4.mediatrix.com
> ...
> 
> Which spawns the INVITE from Tree to a proxy server somewhere in ACME.
> 
> INVITE sip:tt@acme.com SIP/2.0
> From: sip:tree@acme.com
> To: sip:tt@acme.com
> Accept-Contact=sip:tt@office4.mediatrix.com
> Referred-By: ...
> ...
> 
> 
> The proxy receives this request and checks with the location 
> server to find
> where can tt@acme.com be found.  The list of contact returned 
> could look
> like the following:
> 
> Contact: sip:tt@office1.acme.com
> Contact: sip:tt@mediatrix.com
> Contact: sip:assistant@office1.acme.com
> 
> Now the proxy has these 3 contacts that do not match the 
> Accept-Contact URL,
> although the proxy at mediatrix.com would translate 
> tt@mediatrix.com to
> tt@office4.mediatrix.com.  Now the proxy does not really know 
> which contact
> is prefeered, and can try any of the three contacts, and 
> someone can happily
> answer at those addresses.
> 
> How can we avoid this?
> 
> First, I think that the Proxy should send the Accept-Contact in its
> downstream requests, in order to have the downstream proxies 
> be able to use
> the same mechanism.  But in truth, the proxy does not know if 
> the downstream
> request is going to be sent to a proxy or to a UA, so it is 
> possible for a
> UA to receive such a request.  Would it be a good idea to 
> have the UA take a
> look at the Accept-Contact header to know if the request is 
> really for him?
> I think that the current caller preferences draft states that 
> a UA should
> ignore these header if it receives it.

Eric is correct that even if none of the registered contacts matches those
in the Accept-Contact header, a proxy will forward the request, with
Accept-Contact intact. As a result, the request may ultimately arrive at
several UAs that don't match. Another related problem is that Accept-Contact
is not specified as a filter-all-but rule (partly for the reasons Eric
observes; a request would never get past the first proxy since the Contact
might be known only at a later proxy). The caller preferences specification
has a process for combining the q values. At best, it would effect ordering
of a sequential search. 

What we need here is a mechanism that says, "I only want to talk to you if
you are one of these Contacts". This rule can ONLY be enforced in the UAs.
This capability does not exist in the current caller preferences draft.
However, I think it will prove useful here (actually, it really is needed
for call transfer, more on that below), and other places too. So, I will add
an Accept-Only-Contact header to caller preferences, with this header only
being used by a UA.

Now, on to some other issues. Consider the scenario of A talking to B. A
calls C, and then A  wishes to transfer B to C, making sure that B gets the
same instance of C that A was talking to. Adam asked, why not have A refer B
to the Contact obtained from C:

> At the risk of sounding daft, I don't see why using the "Contact:"
> header as the new "INVITE" URI is considered Evil in this case.
> 
> When explaining the difference between, say, the "From:" (or "To:")
> field and the "Contact:" field, I generally think of "From:"/"To:" as
> being the persistent URI for contacting the appropriate resource in 
> an abstract sense (by resource, I mean "Adam Roach" or 
> "helpdesk staffer" 
> or "sales representative"), while the "Contact:" URI 
> indicates "where to
> get to me (as a particular instance of that resource) on this 
> particular
> device at this particular moment."

The answer is that URIs most definitely have a context; a place from which
they can be usefully used. In a perfect world with no NATs, firewalls, or
devices that want to know whats going on within the network, it would work
dandy. But thats not reality. The classic example is an ACD application,
where help@helpdesk.com can be routed to a variety of agents
(agent32@host32.helpdesk.com). Sophisticated software tracks the call status
of the various agents, how much volume they get, etc. Incoming calls MUST
come in through this helpdesk proxy to work. The proxy always record routes.
In fact, to enforce this, the proxy has access to a hosts file with the
host32.helpdesk.com address mapped to an IP address. Outside of this proxy,
that hostname is unroutable. So, if A called B, and then A called C, and C
was one of these operators, the URI returned by C
(agent32@host32.helpdesk.com) will definitely not be useful for B to use
directly.

Another issue was raised about the possibility of per-call Contacts
generated dynamically by a UA. Yikes. I don't know any UA that does this.
Unfortunately, if a UA provides no consistent form of identification to the
network or other users, it can't expect to be consistently transferred to in
this case. So, I think we can punt this case and declare it as "don't do
this".

Robert had proposed a hybrid approach:
> Given all that, and the likelyhood that for many (if not
> most) instances, the contact provided in the Contact URL
> _is_ routable, what problems exist with taking the following
> approach?
> 
>   1) REFER B directly to the contact that C provided to A.
>   2) If it fails, REFER B to C's well known URL with an Accept-Contact
>      header containing the contact that C provided to A.
> 

Hmm. Turns out you can construct cases where neither (1) nor (2) will work.

What if, when A called C, it wasn't really C that was called, but some
time-of-day routed 800 number that got forwarded to C. Now, while A talks to
C, the logic for the forwarding changes. Now, when A sends a REFER to B, the
URI in the Refer-To is that 800 number. It never even comes close to being
sent to a single UA that matches the Accept-Contact header. In this
scenario, sending directly to the Contact won't work either, since the
operator is not directly callable. Don't you love feature interaction
problems ;) ?

So, basically, referring to exactly the same person I'm talking to is hard
because (1) their unique address may not be directly routable, and (2) the
logic used to get to that person from a directly routable address may
change. If you ignore (1), you come up with the solution to (2) of sending
to the Contact. If you ignore (2), you come up with the Accept-Only-Contact
solution to deal with (1). If you want to deal with both (1) and (2), its
really rough. I thought about having A also include Route headers as header
params to the URI in the Refer-To, but this doesn't work either. A route
header set usable from A may not be usable from B, for many reasons, not the
least of which is problem (1) once more.

Perhaps these cases are so extreme its not worth worrying about. I wouldn't
be surprised if they don't work in the PSTN either. 

So, despite its complexity, I feel compelled to recommend that Robert's
hybrid approach be described in the REFER specification as the preferred way
to do transfer out of consultation, along with a nice long explanation of
why it needs to be so. I'm not sure about the order though. I suspect that
(2) will work in far more cases than (1) will. I welcome better ideas if
anyone has any...


Thanks,
Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

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To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: "'Jo Hornsby'" <jhornsby@ubiquity.net>, sip@lists.bell-labs.com
Subject: RE: [SIP] New Section 6.34 (was: Record-Route consensus)
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Date: Thu, 23 Nov 2000 19:52:28 -0800

Hi Jonathan,
> 
> Since we don't dictate routing behavior, so too would the 
> format of these
> URLs not be specified. So, if we assume the caller is 
> A@originating.com, and
> the called party is B@terminating.com, one choice for the URL 
> pair (request,
> response) is:

So you are saying that the Record-Route SHOULD at least contain the
user@host portions of the Contact or Request-URI in some form or another. We
should _suggest_  a method for acheiving this. For SIP URL's, I think the
first method (below) is the best for sustaining your protocol invariant
property. 
> 
> (A@originating.com;maddr=proxy, B@terminating.com;maddr=proxy)
>
The second method (inserting an escaped URL) is probably best recommended
when you are forced to construct a Record-ROute header from a non-SIP URL
(for generality).
> 
> (A%64originating.com@server12.terminating.com, 
> B@server12.terminating.com)
> 
Don't you mean 

(sip%3aA%64originating.com@proxy, sip%3aB%64server12.terminating.com@proxy)
?? 

If not, why not ? 

[I added the "sip:" to a) better identify the escaped URL (rather than just
relying on the "@") and  b) to differeniate non-SIP URL's which may not have
a "@" eg (tel%3a555-1234@proxy)]

Is it worthwhile suggesting different methods for SIP and non-SIP URL's ? Or
KISS and just stick with the latter, more general method?

Cheers,

Robert.


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From sip-admin@lists.bell-labs.com  Thu Nov 23 23:11:05 2000
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Date: Thu, 23 Nov 2000 20:10:40 -0800


I think we also need to clarify the use of angle brackets in the
Record-Route header. They may interfere with the inclusion on the route
parameters in the Request-URI (which cannot use angle brackets). Eg,

Record-Route: <sip:sip%3aA%64originating.com@proxy>;rstate=253878954375364

is "correct" but will probably cause problems in some proxies/UA's as only
the base url (<...>) will be copied into the Request-URI.

I think the best suggestion for proxies is to either a) not to use angle
brackets in Record-Route hoeaders or b) include all record-route parameters
inside the angle brackets. 

The latter is more general (KISS) but is a slight bending of the definition
of "record-route parameters". However, the parameters _are_ being treated as
URL parameters in the Request-URI so the behaviour makes sense from that
respect.

Agree?
 
Regards,

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 
> -----Original Message-----
> From: Jo Hornsby [mailto:jhornsby@ubiquity.net]
> Sent: Thursday, November 23, 2000 10:52 PM
> To: Fairlie-Cuninghame, Robert; Jonathan Rosenberg
> Cc: sip@lists.bell-labs.com
> Subject: RE: [SIP] New Section 6.34 (was: Record-Route consensus)
> 
> 
> Robert Fairlie-Cuninghame wrote:
> > > Very nice -- it's cool that you've found a way to maintain
> > > "the Request-URI points to the ultimate destination" as an
> > > invariant.
> >
> > I still don't see how this is the case. You just have two different
> > URL's located at the proxy server. I don't see they point to the
> > ultimate destination (unless I missing something fundamental).
> 
> They might not point to the ultimate destination, per se; but
> the Request-URI will be meaningful enough to the proxy such
> that it could reapply its initial routing decisions, if
> necessary.
> 
> > >   Each Record-Route header field contains a globally 
> reachable SIP URL
> > >  that identifies a proxy server.  Any proxy that wishes 
> to remain in
> > >  the signalling path of a call leg adds its own SIP URL to the
> > >  beginning of the list of these fields in a request.  This SIP URL
> > >  MUST represent an address that is reachable via UDP, and SHOULD
> > >  represent the calling UA (i.e., be derived from the 
> Contact/From).
> >
> > As above, I don't understand how it "represents" the UA in any
> > sense outside the proxy's internal implementation. I am not saying
> > that this is a bad idea but I do think this wording will cause a
> > lot of confusion.
> 
> I see what you're saying -- my wording is confusing upon reading
> it back.  Hmmm...
> 
> > Do you derive it from the just the user part of the Contact/From ?
> > What if the user name is the same (jack@A <-> jack@B?)? Are you
> > suggestign we put an escaped copy of the whole Contact/From into
> > the username portion of the Route ?
> 
> This is probably what I would do, in fact.
> 
> > Why not just say
> >
> > "This SIP URL MUST represent an address that is reachable 
> via UDP, and the
> > user portion SHOULD be distinguishable by the proxy as 
> being destined for
> > the calling UA. The URL SHOULD also be distguishable from 
> any previous
> > Record-Route entries added by the proxy (e.g., due to 
> spiraling routes)."
> 
> Basically because I'd rather we imply as little as possible about
> the format of this URL; however, I think we're getting warmer.
> 
> How about:
>     This SIP URL MUST represent an address that is reachable via
>     UDP, and SHOULD somehow be meaningful to the proxy, such that
>     it could map the URL back to the calling UA (e.g., by embedding
>     the Contact/From ha a parameter).
> ?
> 
> > > > >  *** If no Contact, should add the From?
> >
> > Is there a reason why we should change the existing behaviour ?
> 
> I wasn't sure what the existing behaviour was...
> 
> 
>  - Jo.
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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Can anyone help me to unsubscribe from the list.  I've sent 'unsubscribe'
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Date: Fri, 24 Nov 2000 01:15:50 -0800


Hi Neil,

> "Fairlie-Cuninghame, Robert" wrote:
> > 
> > The colon in the user part of Test 36 (the long telephone 
> subscriber) needs
> > to be quoted. Perhaps you could also put a similar long 
> telephone-subscriber
> > URL in the Contact - this would test whether the URL was correctly
> > reproduced if the call is accepted.
> 
> I think the ':' might be OK as it becomes part of a hname in 
> an escaped header as it follows a '?', i.e. from parsing it 
> into:
> 	%22Route:%20X%40Y%3bZ = W%22@gw1.wcom.com;user=phone
>         	       header = hvalue
> However this would then mean the following  ';' and '=' in the 
> hvalue need to be escaped.

No, this is not an escaped SIP header (per se), rather it is all just part
of a telephone subscriber user portion and as such it follows a different
set of rules.  Reread Section 2. You need to escape any characters that not
in the unreserved or user-unreserved sets (hname and hvalue have nothing to
do with it). Colon is such a character.

> Also we would back into the problem of escaped headers not being
> allowed in Request-URIs. I am a bit uncertain about this now.
> The spec indicates that they are not allowed (Table 2) and 
> should be removed by a proxy before futher processing (4.3).
> However there seems to be a feeling that this may not be
> desirable
> - should there be a spec change here?

Again, these are not escaped headers but merely part of the user part, they
are only inpreterpreted with respect to the tel: URL syntax and not the SIP
URL syntax.

I hope that clarifies things.

Robert.

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"Fairlie-Cuninghame, Robert" wrote:
> 
> Hi Neil,
> 
> > "Fairlie-Cuninghame, Robert" wrote:
> > >
> > > The colon in the user part of Test 36 (the long telephone
> > subscriber) needs
> > > to be quoted. Perhaps you could also put a similar long
> > telephone-subscriber
> > > URL in the Contact - this would test whether the URL was correctly
> > > reproduced if the call is accepted.
> >
> > I think the ':' might be OK as it becomes part of a hname in
> > an escaped header as it follows a '?', i.e. from parsing it
> > into:
> >       %22Route:%20X%40Y%3bZ = W%22@gw1.wcom.com;user=phone
> >                      header = hvalue
> > However this would then mean the following  ';' and '=' in the
> > hvalue need to be escaped.
>
> No, this is not an escaped SIP header (per se), rather it is all just part
> of a telephone subscriber user portion and as such it follows a different
> set of rules. 

Ooops I missed that rather important point, I'll change ':' ->
%3a.

Thanks,
Neil.
-- 
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Robert Fairlie-Cuninghame wrote:
>
> Jonathan Rosenberg wrote:
>
> > Since we don't dictate routing behavior, so too would the 
> > format of these URLs not be specified. So, if we assume the
> > caller is A@originating.com, and the called party is
> > B@terminating.com, one choice for the URL pair (request,
> > response) is:
> 
> So you are saying that the Record-Route SHOULD at least contain the
> user@host portions of the Contact or Request-URI in some form or 
> another. We should _suggest_  a method for acheiving this. For
> SIP URL's, I think the first method (below) is the best for
> sustaining your protocol invariant property. 
> > 
> > (A@originating.com;maddr=proxy, B@terminating.com;maddr=proxy)
> >

It works as long as the Request-URI doesn't already have an
maddr parameter. &:)

> The second method (inserting an escaped URL) is probably best recommended
> when you are forced to construct a Record-ROute header from a non-SIP URL
> (for generality).
> > 
> > (A%64originating.com@server12.terminating.com, 
> > B@server12.terminating.com)
> > 
> Don't you mean 
> 
> (sip%3aA%64originating.com@proxy, 
> sip%3aB%64server12.terminating.com@proxy)
> ?? 
> 
> If not, why not ? 
> 
> [I added the "sip:" to a) better identify the escaped URL (rather 
> than just relying on the "@") and  b) to differeniate non-SIP URL's
> which may not have a "@" eg (tel%3a555-1234@proxy)]
> 
> Is it worthwhile suggesting different methods for SIP and non-SIP 
> URL's ? Or KISS and just stick with the latter, more general method?

I would KISS. &:)  Personally, I prefer the
embed-the-uri-as-a-parameter method, but that's probably mainly
for aesthetical reasons (on simple URLs, one needs no escaping
at all).

One plus for parameters (although I guess the user part could
still be used, but...) that I just thought of allows proxy to
insert all the necessary information in the request only.  Say
I (a machine in sipfarm.net) receive the following:

    INVITE sip:buffy@sunnydale-high.edu SIP/2.0
    From: sip:willow@sunnydale-high.edu;tag=2718281828
    To: sip:buffy@sunnydale-high.edu
    Contact: sip:willow@machine35.cyber-cafe.com
    ...

I Record-Route and proxy downstream (illegal line wrapping for clarity):

    INVITE sip:angel@sunnydale.com SIP/2.0
    From: sip:willow@sunnydale-high.edu;tag=2718281828
    To: sip:buffy@sunnydale-high.edu
    Contact: sip:willow@machine35.cyber-cafe.com
    Record-Route: <sip:sipfarm.net;
      caller=sip:willow@machine35.cyber-cafe.com;
      callee=sip:buffy@sunnydale-high.edu>
    ...

At some point later, I see a re-INVITE:

    INVITE sip:sipfarm.net;
      caller=sip:willow@machine35.cyber-cafe.com;
      callee=sip:buffy@sunnydale-high.edu SIP/2.0
    From: sip:buffy@sunnydale-high.edu;tag=3141592654
    To: sip:willow@sunnydale-high.edu;tag=2718281828
    Route: sip:willow@machine35.cyber-cafe.com
    ...

Looking at the From and To, I can determine that this is from
called to caller.

Granted, it's a little more work initially, and potentially
in subsequent requests (because of the From/To decision), but
it seems a little easier than inspecting and modifying
Record-Routes in any responses.

Thoughts?


 - Jo.


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Robert Fairlie-Cuninghame wrote:
> I think we also need to clarify the use of angle brackets in the
> Record-Route header. They may interfere with the inclusion on the route
> parameters in the Request-URI (which cannot use angle brackets). Eg,
>
> Record-Route: <sip:sip%3aA%64originating.com@proxy>;rstate=253878954375364
>
> is "correct" but will probably cause problems in some proxies/UA's as only
> the base url (<...>) will be copied into the Request-URI.
>
> I think the best suggestion for proxies is to either a) not to use angle
> brackets in Record-Route hoeaders or b) include all record-route
> parameters inside the angle brackets.
>
> The latter is more general (KISS) but is a slight bending of the
> definition of "record-route parameters".

Indeed. &:)

> However, the parameters
> _are_ being treated as URL parameters in the Request-URI so the
> behaviour makes sense from that respect.

I'm not quite with you here, but I think I agree.  As things
currently stand, it doesn't look like Record-Route parameters
are particularly useful; however, this doesn't mean we won't
think of some funky new use in the future.

Therefore, at the moment, it might be wise to note that
proxies probably always want to use angle brackets.  Just to
be sure.

Cheers,


 - Jo.


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Date: Fri, 24 Nov 2000 03:19:05 -0800

Jo,

> > > 
> > > (A@originating.com;maddr=proxy, B@terminating.com;maddr=proxy)
> > >
> It works as long as the Request-URI doesn't already have an
> maddr parameter. &:)
> 

Request-URI's can't have maddr parameters but yes for the callee's Contact
this might be an issue. 

> > The second method (inserting an escaped URL) is probably 
> best recommended
> > when you are forced to construct a Record-ROute header from 
> a non-SIP URL
> > (for generality).
> > > 
> > (sip%3aA%64originating.com@proxy, 
> > sip%3aB%64server12.terminating.com@proxy)
> > ?? 
> > 
> 
> One plus for parameters (although I guess the user part could
> still be used, but...) that I just thought of allows proxy to
> insert all the necessary information in the request only.  Say
> I (a machine in sipfarm.net) receive the following:

There will also be problems when the URL contains escaped characters that
have significance in the original URL (eg @,=,;). You have no way of telling
why the character was escaped. In my opinion this is the sort of thing that
makes a behaviour brittle, eg, if the proxy actually relies on correctly
recovering the original uri for routing purposes. [so yes a parameter is
better in my view as well).

> 
>     INVITE sip:buffy@sunnydale-high.edu SIP/2.0
>     From: sip:willow@sunnydale-high.edu;tag=2718281828
>     To: sip:buffy@sunnydale-high.edu
>     Contact: sip:willow@machine35.cyber-cafe.com
>     ...
> 
> I Record-Route and proxy downstream (illegal line wrapping 
> for clarity):
> 
>     INVITE sip:angel@sunnydale.com SIP/2.0
>     From: sip:willow@sunnydale-high.edu;tag=2718281828
>     To: sip:buffy@sunnydale-high.edu
>     Contact: sip:willow@machine35.cyber-cafe.com
>     Record-Route: <sip:sipfarm.net;
>       caller=sip:willow@machine35.cyber-cafe.com;
>       callee=sip:buffy@sunnydale-high.edu>
>     ...
> 
> At some point later, I see a re-INVITE:
> 
>     INVITE sip:sipfarm.net;
>       caller=sip:willow@machine35.cyber-cafe.com;
>       callee=sip:buffy@sunnydale-high.edu SIP/2.0
>     From: sip:buffy@sunnydale-high.edu;tag=3141592654
>     To: sip:willow@sunnydale-high.edu;tag=2718281828
>     Route: sip:willow@machine35.cyber-cafe.com
>     ...
> 
> Looking at the From and To, I can determine that this is from
> called to caller.
> 
> Granted, it's a little more work initially, and potentially
> in subsequent requests (because of the From/To decision), but
> it seems a little easier than inspecting and modifying
> Record-Routes in any responses.

Only this doesn't provide you with the original request-uri/Contact if you
want to use it for routing purposes (as the caller and callee fields are
built up from the To and From).

I still maintain that embedding the URI into user part still doesn't
represent the calling or called UA's - to everyone else they are just
arbitrary user names located at the proxy. So how does that help with the
desired invariant property? I beleive using parameters would be better. Eg,
placing the following RR header in the forward request

Record-Route: <caller@proxy;rr-uri=sip:willow@machine35.cyber-cafe.com>

and this in the response 

Record-Route: <callee@proxy;rr-uri=sip:angel@sunnydale-high.edu>

is much less brittle than trying to embed the URL in the user part. Of
course, if the proxy is storing full state then rr-uri is not required.

As for maddr handling: If the proxy is required to add the rr-uri parameter,
then I think the proxy must also replace any maddr with a "rr-maddr"
parameter so that the outgoning request-URI is faithfully recreated. Of
course, once again this is only needed if the proxy is not keeping full
state. Ie, it is the responsibility of the actual implemenation to reproduce
the same routing action.

Eg, if your example was instead (with same routing 

>     INVITE sip:buffy@sunnydale-high.edu SIP/2.0
>     From: sip:willow@sunnydale-high.edu;tag=2718281828
>     To: sip:buffy@sunnydale-high.edu
>     Contact:
sip:willow@cyber-cafe.com;maddr=machine35.cyber-cafe.com;yadda=blah

Then I would suggest the following Forward RR could be generated

Record-Route: <callee@proxy;
			rr-loop=1;
			rr-uri=sip:willow@cyber-cafe.com;
			rr-maddr=machine35.cyber-cafe.com;
			yadda=blah> 

Once again, all we can do is provide a recommendation for proxies not want
to keep full state. So all we have to specify in the spec is characteristics
of what the base user@host is, right? 

Regards,

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 

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Date: Fri, 24 Nov 2000 03:31:41 -0800

> 
> > However, the parameters
> > _are_ being treated as URL parameters in the Request-URI so the
> > behaviour makes sense from that respect.
> 
> I'm not quite with you here, but I think I agree.  As things
> currently stand, it doesn't look like Record-Route parameters
> are particularly useful; however, this doesn't mean we won't
> think of some funky new use in the future.

Well to any other proxy than the generating/intended proxy, the extra
record-route related Request-URI parameters are simply treated as unknown
"other-param" parameters (as per Section 2 grammer definitions). 

Given that record-route parameters (ie, ones copied into the Request-URI)
are implementation specific, I think we need to recommend that all parameter
names should be of the form "rr-xxxxxx" (or something) so that we don't get
collisions in the future between new URL parameters and some proxy's
implementation specific record-route parameters. 

[Record-Route is a peculiar case, as I was saying, because we need to copy
the Record-Route parameters in to the Request-URI (which can't include angle
brackets).]

> Therefore, at the moment, it might be wise to note that
> proxies probably always want to use angle brackets.  Just to
> be sure.
> 
Agreed.

Robert.


-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. --  

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From: "Jo Hornsby" <jhornsby@ubiquity.net>
To: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: <sip@lists.bell-labs.com>
Subject: RE: [SIP] New Section 6.34 (was: Record-Route consensus)
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Robert wrote:
> > > > 
> > > > (A@originating.com;maddr=proxy, B@terminating.com;maddr=proxy)
> > > >
> > It works as long as the Request-URI doesn't already have an
> > maddr parameter. &:)
> > 
> 
> Request-URI's can't have maddr parameters but yes for the callee's
> Contact this might be an issue. 

Actually, this is no-longer true: Request-URIs are now allowed
to contain maddr; From/To are still not (see bis).

> > One plus for parameters (although I guess the user part could
> > still be used, but...) that I just thought of allows proxy to
> > insert all the necessary information in the request only.  Say
> > I (a machine in sipfarm.net) receive the following:
> > 
> >     INVITE sip:buffy@sunnydale-high.edu SIP/2.0
> >     From: sip:willow@sunnydale-high.edu;tag=2718281828
> >     To: sip:buffy@sunnydale-high.edu
> >     Contact: sip:willow@machine35.cyber-cafe.com
> >     ...
> > 
> > I Record-Route and proxy downstream (illegal line wrapping 
> > for clarity):
> > 
> >     INVITE sip:angel@sunnydale.com SIP/2.0
> >     From: sip:willow@sunnydale-high.edu;tag=2718281828
> >     To: sip:buffy@sunnydale-high.edu
> >     Contact: sip:willow@machine35.cyber-cafe.com
> >     Record-Route: <sip:sipfarm.net;
> >       caller=sip:willow@machine35.cyber-cafe.com;
> >       callee=sip:buffy@sunnydale-high.edu>
> >     ...
> > 
> > At some point later, I see a re-INVITE:
> > 
> >     INVITE sip:sipfarm.net;
> >       caller=sip:willow@machine35.cyber-cafe.com;
> >       callee=sip:buffy@sunnydale-high.edu SIP/2.0
> >     From: sip:buffy@sunnydale-high.edu;tag=3141592654
> >     To: sip:willow@sunnydale-high.edu;tag=2718281828
> >     Route: sip:willow@machine35.cyber-cafe.com
> >     ...
> > 
> > Looking at the From and To, I can determine that this is from
> > called to caller.
> > 
> > Granted, it's a little more work initially, and potentially
> > in subsequent requests (because of the From/To decision), but
> > it seems a little easier than inspecting and modifying
> > Record-Routes in any responses.
> 
> Only this doesn't provide you with the original request-uri/Contact
> if you want to use it for routing purposes (as the caller and
> callee fields are built up from the To and From).

My example was dull.  I had actually copied the Contact to the
callee parameter, and the Request-URI to the caller parameter;
unfortunately, the Request-URI happened to be the same as the
To.

> I still maintain that embedding the URI into user part still doesn't
> represent the calling or called UA's - to everyone else they are just
> arbitrary user names located at the proxy.

I so don't disagree.  Say we have the following scenario:

    A --- P1 --- P2 --- B

Even on the initial INVITE, the Request-URI that P2 sees maybe
completely different from Request-URI that that P1 sees (imagine
that I'm on holiday and thus I've set up call forwarding to a
close friend).  Thus from that perspective, an arbitrary
Record-Route URI that only makes sense to the proxy that inserted
it is no different.

> So how does that help with the desired invariant property? I
> beleive using parameters would be better.

I don't think it matters how the URI is formed, as typically
it's only going to make sense to the proxy that inserted it
(unless we come up with a BCP, or sommat).

I think when I talk about this invariant, I don't necessarily
mean that the Request-URI along the entire route would be
routable by an arbitrary proxy; what I mean is that the
Request-URI would be routable at the salient proxy.

I think what Jonathan didn't like -- and I wasn't happy about
it, either -- was that in the reverse direction, the Request-URI
could in no way ever have been useful in finding the calling
UA (save the case that the calling and called UA were the same,
perhaps).  With the sorts of things being proposed now --
having a different URI on request and response, or somehow
embedding the caller, say -- this is no longer the case.  Sure,
it's likely that only the proxy that inserted the URI will
understand it, but that's in no way atypical, and I feel
still satisfies the invariant.

> As for maddr handling: If the proxy is required to add the rr-uri 
> parameter, then I think the proxy must also replace any maddr with
> a "rr-maddr" parameter so that the outgoning request-URI is
> faithfully recreated.

I'm not sure why this is needed?  Couldn't we just do something
like:
  sip:giles@sunnydale-high.edu;maddr=library.sunnydale-high.edu
to
  sip:proxyfarm.net;caller=sip:giles@sunnydale-high.edu%3B
  maddr%3Dlibrary.sunnydale-hig.edu

Or is there some subtle parsing point that I'm missing?

> Once again, all we can do is provide a recommendation for proxies
> not want to keep full state. So all we have to specify in the spec is 
> characteristics of what the base user@host is, right?

I don't think we even need a user.


 - Jo.


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> > > However, the parameters
> > > _are_ being treated as URL parameters in the Request-URI so the
> > > behaviour makes sense from that respect.
> > 
> > I'm not quite with you here, but I think I agree.  As things
> > currently stand, it doesn't look like Record-Route parameters
> > are particularly useful; however, this doesn't mean we won't
> > think of some funky new use in the future.
> 
> Well to any other proxy than the generating/intended proxy, the extra
> record-route related Request-URI parameters are simply treated as unknown
> "other-param" parameters (as per Section 2 grammer definitions). 

Indeed.

> Given that record-route parameters (ie, ones copied into the Request-URI)
> are implementation specific, I think we need to recommend that 
> all parameter names should be of the form "rr-xxxxxx" (or something)
> so that we don't get collisions in the future between new URL
> parameters and some proxy's implementation specific record-route
> parameters.

I'm getting slightly confused by the term "record-route parameters",
I think.  Are you talking about the parameters that a proxy adds
to the SIP URL?

If the Contact, say, has all the `='s and `;'s escaped, and the
result of this is embedded as a parameter, does this solve the
collision problem?

> [Record-Route is a peculiar case, as I was saying, because we need
> to copy the Record-Route parameters in to the Request-URI (which
> can't include angle brackets).]

I think there's still a need to distinguish between Record-Route
parameters (that is, rr-params in section 6.34 of bis02) and the
parameters that a proxy adds to the SIP URL that it places in
a Record-Route value.  Record-Route params might still prove to
be useful; they could be used so that a proxy can identify the
Record-Route entry that it added, for instance.

Cheers,


 - Jo.


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Ranga,

See comments below.

Regards,
Chris

"M. Ranganathan" wrote:

>
> Chris, Erik,
>
> While we are on the subject of Exceptions, why do null arguments throw
> runtime exceptions when you do a get on them? For example, in the
> Accept interface, getContentSubtype throws a IllegalArgumentException
> if  is null. IMHO, I think it is valid to have a null subtype so why
> not just return null rather than throw exception here?

By a "doing a get on a null argument" here I assume you mean attempting
to get a non-existent optional field? If you do this a checked
ParameterNotSetException is thrown, not a runtime exception.

There are two approaches to handling optional header parts. One approach
is to allow the passing and returning of null objects from get and set
methods (implicit "remove" and "has"), and the other is to explicitly
add has and remove methods (and exceptions on the set and get methods).
As for mandatory header parts - I think passing null to a set method
should definitely throw an IllegalArgumentException.

(If it is considered acceptable to have a "null" subType then we should
make it optional)

>
>
> I don't quite follow the argument that lazy parsing  "necessitates
> throwing a parse exception for all get methods".   Here is what I had
> envisioned:
>
>
> 1 When a message comes in it will be separated into headers based on
> type.
> 2 The header would be returned with no field initialized except the
> text corresponding to the message. (this would necessitate examining
> the first few bytes of the message to determine the header type but
> the rest remains unparsed.) You can also stow away the type
> information in the Header that is returned as well to know which
> parser rule to invoke later.
> 3. Later, when the application wants to parse the header, it has the
> string corresponding to the message and can hand it off to the parser,
> which can then fill in the appropriate fields corresponding to the
> header.
> 4. Subsequently, the application can access the fields just as an
> eager parser based application would do.
>
> ( Where am I missing the boat with this? )

An application has no idea whether an implementation is using lazy or
eager parsing - it is simply passed a header object - and then if it is
interested in a particular field it calls the appropriate get method
(which would internally invoke the parser in a lazy parsing
implementation). So the application is completely oblivious to whether
or not this "optimisation" is used by the implementation.

If the field is optional and does not exist in the header a
ParameterNotSetException will be thrown by the implementation. If the
field is mandatory and does not exist, a JainSipParseException will be
thrown (header is malformed). If any header field is malformed a
JainSipParseException is thrown. If a JainSipParseException is thrown,
this indicates that the application must use the header's getValue
method to get a string representation of the header value and do what it
can with that.

>
>
> It appears that what is being attempted here (please correct me if I
> am wrong) is to use exceptions as a mechanism to catch un-initialized
> fields so the parser can be invoked in the exception handler perhaps.
> This leads to a lot of overhead in clunky code. I think it is a
> mistake to throw exceptions on every get to try to protect the
> application from inadvertently accessing fields that have not been
> initialized.  Why not just assume that the application must be smart
> enough to parse the header text before accessing fields and not try to
> provide a complex mechanism as above?

We are using ParameterNotSetException for the case of an optional header
not existing, and a JainSipParseException for the case of a malformed
header being received from the network.

If we assume the application is smart enough to do all header value
parsing, we reduce a header to just being two strings - the type and
value, which puts the burden of parsing on an application - this work
can be done by the underlying implementation - doesn't it make more
sense for an application to catch an exception than have to parse all
header value strings (especially since this functionality already exists
in the underlying stack)

>
>
>
> Regards,
>
> Ranga.
>
> Chris Harris wrote:
>
>> Erik,
>>
>> The implication of what you say is that all set methods in the
>> header
>> interfaces which take String arguments must throw a checked
>> JainSipParseException. I think this is reasonable given that
>> different
>> implementations will be more tolerant than others. However, in order
>> to allow
>> lazy parsing by an implementation, all get methods in the header
>> interfaces
>> (regardless of return type) must throw a checked parse exception.
>> This is a
>> lot of checked exceptions - and will make life more awkward for the
>> application developer. It would certainly be much less effort to not
>> allow
>> lazy parsing - there would be no need to throw a parse exception for
>> all get
>> methods - greatly reducing the number of checked exceptions.
>> Unfortunately it
>> has been said that JAIN SIP support for lazy parsing is a must. If
>> it is then
>> I guess there is no other choice than to include the exceptions?
>>
>> Regards,
>> Chris
>>
>> Erik Nissen wrote:
>>
>> > Chris Harris wrote:
>> >
>> > >
>> > > We need to decide whether this should be a checked or runtime
>> exception.
>> >
>> > The rule of thumb is to use:
>> > * checked exceptions when the program should handle situations
>> which
>> > can/will occur.
>> > * runtime exceptions when there are programming error (God forbid
>> :-)
>> >
>> > So:
>> > Parse errors should throw checked exceptions.
>> > Becaurse parse errors are outside the control of the programmer
>> > and he/she should be forced to handle such conditions in the code.
>>
>> >
>> > Null/wrong arguments should throw runtime exceptions.
>> >
>> > If it is possible, reduce the number of methods throwing (checked)
>>
>> > exceptions
>> > to a minimum.
>> >
>> > Best regards,
>> > /Erik Nissen,
>> > Ericsson Utveckling AB
>>
>> _______________________________________________
>> SIP mailing list
>> SIP@lists.bell-labs.com
>> http://lists.bell-labs.com/mailman/listinfo/sip
>
> --
> M.Ranganathan
> NIST Advanced Networking Technologies Group,
> 100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
> Tel: 301 975 3664 Fax: 301 590 0932
>
>

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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<body text="#000000" bgcolor="#FFFFFF" link="#0000FF" vlink="#FF0000" alink="#000088">
Ranga,
<p>See comments below.
<p>Regards,
<br>Chris
<p>"M. Ranganathan" wrote:
<blockquote TYPE=CITE>&nbsp;
<br>Chris, Erik,
<p>While we are on the subject of Exceptions, why do null arguments throw
runtime exceptions when you do a get on them? For example, in the Accept
interface, getContentSubtype throws a IllegalArgumentException if&nbsp;
is null. IMHO, I think it is valid to have a null subtype so why not just
return null rather than throw exception here?</blockquote>
By a "doing a get on a null argument" here I assume you mean attempting
to get a non-existent optional field? If you do this a checked ParameterNotSetException
is thrown, not a runtime exception.
<p>There are two approaches to handling optional header parts. One approach
is to allow the passing and returning of null objects from get and set
methods (implicit "remove" and "has"), and the other is to explicitly add
has and remove methods (and exceptions on the set and get methods). As
for mandatory header parts - I think passing null to a set method should
definitely throw an IllegalArgumentException.
<p>(If it is considered acceptable to have a "null" subType then we should
make it optional)
<blockquote TYPE=CITE>&nbsp;
<p>I don't quite follow the argument that lazy parsing&nbsp; "necessitates
throwing a parse exception for all get methods".&nbsp;&nbsp; Here is what
I had envisioned:
<br>&nbsp;
<p>1 When a message comes in it will be separated into headers based on
type.
<br>2 The header would be returned with no field initialized except the
text corresponding to the message. (this would necessitate examining the
first few bytes of the message to determine the header type but the rest
remains unparsed.) You can also stow away the type information in the Header
that is returned as well to know which parser rule to invoke later.
<br>3. Later, when the application wants to parse the header, it has the
string corresponding to the message and can hand it off to the parser,
which can then fill in the appropriate fields corresponding to the header.
<br>4. Subsequently, the application can access the fields just as an eager
parser based application would do.
<p>( Where am I missing the boat with this? )</blockquote>
An application has no idea whether an implementation is using lazy or eager
parsing - it is simply passed a header object - and then if it is interested
in a particular field it calls the appropriate get method (which would
internally invoke the parser in a lazy parsing implementation). So the
application is completely oblivious to whether or not this "optimisation"
is used by the implementation.
<p>If the field is optional and does not exist in the header a ParameterNotSetException
will be thrown by the implementation. If the field is mandatory and does
not exist, a JainSipParseException will be thrown (header is malformed).
If any header field is malformed a JainSipParseException is thrown. If
a JainSipParseException is thrown, this indicates that the application
must use the header's getValue method to get a string representation of
the header value and do what it can with that.
<blockquote TYPE=CITE>&nbsp;
<p>It appears that what is being attempted here (please correct me if I
am wrong) is to use exceptions as a mechanism to catch un-initialized fields
so the parser can be invoked in the exception handler perhaps. This leads
to a lot of overhead in clunky code. I think it is a mistake to throw exceptions
on every get to try to protect the application from inadvertently accessing
fields that have not been initialized.&nbsp; Why not just assume that the
application must be smart enough to parse the header text before accessing
fields and not try to provide a complex mechanism as above?</blockquote>

<p><br>We are using ParameterNotSetException for the case of an optional
header not existing, and a JainSipParseException for the case of a malformed
header being received from the network.
<p>If we assume the application is smart enough to do all header value
parsing, we reduce a header to just being two strings - the type and value,
which puts the burden of parsing on an application - this work can be done
by the underlying implementation - doesn't it make more sense for an application
to catch an exception than have to parse all header value strings (especially
since this functionality already exists in the underlying stack)
<blockquote TYPE=CITE>&nbsp;
<br>&nbsp;
<p>Regards,
<p>Ranga.
<p>Chris Harris wrote:
<blockquote TYPE=CITE>Erik,
<p>The implication of what you say is that all set methods in the header
<br>interfaces which take String arguments must throw a checked
<br>JainSipParseException. I think this is reasonable given that different
<br>implementations will be more tolerant than others. However, in order
to allow
<br>lazy parsing by an implementation, all get methods in the header interfaces
<br>(regardless of return type) must throw a checked parse exception. This
is a
<br>lot of checked exceptions - and will make life more awkward for the
<br>application developer. It would certainly be much less effort to not
allow
<br>lazy parsing - there would be no need to throw a parse exception for
all get
<br>methods - greatly reducing the number of checked exceptions. Unfortunately
it
<br>has been said that JAIN SIP support for lazy parsing is a must. If
it is then
<br>I guess there is no other choice than to include the exceptions?
<p>Regards,
<br>Chris
<p>Erik Nissen wrote:
<p>> Chris Harris wrote:
<br>>
<br>> >
<br>> > We need to decide whether this should be a checked or runtime exception.
<br>>
<br>> The rule of thumb is to use:
<br>> * checked exceptions when the program should handle situations which
<br>> can/will occur.
<br>> * runtime exceptions when there are programming error (God forbid
:-)
<br>>
<br>> So:
<br>> Parse errors should throw checked exceptions.
<br>> Becaurse parse errors are outside the control of the programmer
<br>> and he/she should be forced to handle such conditions in the code.
<br>>
<br>> Null/wrong arguments should throw runtime exceptions.
<br>>
<br>> If it is possible, reduce the number of methods throwing (checked)
<br>> exceptions
<br>> to a minimum.
<br>>
<br>> Best regards,
<br>> /Erik Nissen,
<br>> Ericsson Utveckling AB
<p>_______________________________________________
<br>SIP mailing list
<br>SIP@lists.bell-labs.com
<br><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></blockquote>

<pre>--&nbsp;
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.&nbsp;
Tel: 301 975 3664 Fax: 301 590 0932</pre>
&nbsp;</blockquote>

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Date: Fri, 24 Nov 2000 07:05:56 -0800

> 
> > So how does that help with the desired invariant property? I
> > beleive using parameters would be better.
> 
> I don't think it matters how the URI is formed, as typically
> it's only going to make sense to the proxy that inserted it
> (unless we come up with a BCP, or sommat).
> 
> I think when I talk about this invariant, I don't necessarily
> mean that the Request-URI along the entire route would be
> routable by an arbitrary proxy; what I mean is that the
> Request-URI would be routable at the salient proxy.
> 
> I think what Jonathan didn't like -- and I wasn't happy about
> it, either -- was that in the reverse direction, the Request-URI
> could in no way ever have been useful in finding the calling
> UA (save the case that the calling and called UA were the same,
> perhaps).  With the sorts of things being proposed now --
> having a different URI on request and response, or somehow
> embedding the caller, say -- this is no longer the case.  Sure,
> it's likely that only the proxy that inserted the URI will
> understand it, but that's in no way atypical, and I feel
> still satisfies the invariant.

I also like the idea. How about we express the invariant in the most simple
way possible: the user portion of the Record-Route indicates the direction
to the proxy (but it does not need to [necessarily] represent the original
Request-URI/Contact - allowing the proxy to rely on such "slightly mangled
original-uri" information, in my mind, is making things MORE brittle not
less [and complicated] )? 

Anything else should be left up to the internal implementation of the proxy.
eg including route state, original uri (if necessary), spiral tag (which
should be recommended), etc. The ammount of information included in the
Record-Route may be influenced by the ammount of information stored for the
call and how routing is performed (eg the Request-URI may not actually be
helpful in identifying the same routing destination for load-sharing or
time-dependent routes).

> > As for maddr handling: If the proxy is required to add the rr-uri 
> > parameter, then I think the proxy must also replace any maddr with
> > a "rr-maddr" parameter so that the outgoning request-URI is
> > faithfully recreated.
> 
> I'm not sure why this is needed?  Couldn't we just do something
> like:

Sorry, you're right I was getting confused - its not needed. 

> > Once again, all we can do is provide a recommendation for proxies
> > not want to keep full state. So all we have to specify in 
> the spec is 
> > characteristics of what the base user@host is, right?
> 
> I don't think we even need a user.
> 
I don't think it would hurt having a user - it makes the invariant easier to
express (as above). I don't know whether requiring a user part should be a
SHOULD or MUST but I feel a SHOULD is more appropriate.

[Boy, Jonathan's going to have a few emails waiting for him tomorrow.]

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 

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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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Hi,

A new version of the SDP media alignment draft has been released:
ftp://ftp.nordu.net/internet-drafts/draft-camarillo-sip-sdp-01.txt

This draft defines an SDP attrubite to be used for SDP media alignment
in SIP (MMUSIC & SIP WGs)

Regards,

Gonzalo
-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo

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Date: Fri, 24 Nov 2000 07:14:22 -0800

> 
> > > > However, the parameters
> > > > _are_ being treated as URL parameters in the Request-URI so the
> 
> I think there's still a need to distinguish between Record-Route
> parameters (that is, rr-params in section 6.34 of bis02) and the
> parameters that a proxy adds to the SIP URL that it places in
> a Record-Route value.  Record-Route params might still prove to
> be useful; they could be used so that a proxy can identify the
> Record-Route entry that it added, for instance.
> 
I agree. They should be distinguished and that's a very good suggestion for
the using the former (Record-Route URL parameters?). What do we call the
latter? Record-Route Request-URI parameters ?
As for any complex escaping schemes, I think that we should be aiming to
make Record-Route/Route as simple as possible - its complicated enough
already. :) [Thank god we can now at least get rid of route mangling in the
UA.]

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 

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I'm not sure, but it seems that there's a simple fix: simply have the UA
create *different* tags depending on whether this is for a From or To.
However, the tag is constant across calls and requests.

The general matching rule is

- match incoming requests so that the To matches the locally generated
tag and
  the From matches the tag generated by the other side

Modifying the running example to use tag=5 for the To tag

> I decide to call myself.  I create a call-leg struct to represent it.
> Let's call it "Line 1".  I send out the initial INVITE:
> 
>   [From Line 1]
>   INVITE sip:me@proxy
>   Via: SIP/2.0/UDP my.ip
>   To: <sip:me@proxy>
>   From: <sip:me@proxy>;tag=7
>   Call-ID: 5
> 
>   This eventually loops back to me.  Now I see coming in:
> 
>   INVITE sip:me@proxy
>   Via: SIP/2.0/UDP some.ip
>   To: <sip:me@proxy>
>   From: <sip:me@proxy>;tag=7
>   Call-ID: 5
> 
>   I check and see there is no To-tag, so this is a new incoming call.  I
> create a call struct to represent it, called "Line 2".  Line 2 now
> begins ringing.  I respond with a 180 and the user switches lines and
> answers the incoming call:
> 
>   [From Line 2]
>   SIP/2.0 200 OK
>   Via: SIP/2.0/UDP some.ip
>   To: <sip:me@proxy>;tag=5

Modification here.

>   From: <sip:me@proxy>;tag=7
>   Call-ID: 5
> 
>   When I receive this response, I find the related pending transaction.
> Arguably, I could first find which call-leg it is destined to (and
> immediately have a problem), but say that I do transaction mapping first
> so we don't break yet.  I match the OK and send an ACK:
> 
>   [From Line 1]
>   ACK sip:me@contactaddr
>   Via: SIP/2.0/UDP my.ip
>   To: <sip:me@proxy>;tag=5
>   From: <sip:me@proxy>;tag=7
>   Call-ID: 5
> 
>   At which point the other pending transaction also correctly matches
> this ACK to it.
> 
>   Now, on Line 2, I decide to hang up.  I've had enough of talking to
> myself.  So, I send a BYE:
> 

As usual, the callee simply flips To and From.


>   [From Line 2]
>   BYE sip:me@contact.addr
>   Via: SIP/2.0/UDP my.ip
>   To: <sip:me@proxy>;tag=7
>   From: <sip:me@proxy>;tag=5
>   Call-ID: 5
> 
>   I now see this incoming BYE:
> 
>   BYE sip:me@contact.addr
>   Via: SIP/2.0/UDP some.ip
>   To: <sip:me@proxy>;tag=7
>   From: <sip:me@proxy>;tag=5
>   Call-ID: 5
> 

Using the rule above, it matches the call on line 1, since 7 is the
locally-generated tag (stuck into From).

> 
>   Using unique tags per call avoids any confusion even in this case, and
> also protects the UA against clients which always use the same tag and
> happen to make two calls with the same Call-ID.

Since both Line1 and Line2 have the same call-ID (one incoming and one
outgoing), they would have the same locally generated tag according to
your rule, so I don't see how this helps.

Two calls with the same call-id can't happen, by definition. If they do,
the implementation is broken (and would presumably generate the "wrong"
tag, too, while it is at it).


-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Date: Fri, 24 Nov 2000 14:33:46 -0600

>> Like Robert I struggle with this RR URI which "identifies the
>> [caller/callee] as a resource on this proxy". In the typical case
>> only one of the two parties would be known in any meaningful way to
>> the proxy. [...]
>
> My initial posting was not sufficiently detailed to explain this, so
> let me clarify.
>
> The URL inserted into the request must be something that
> (1) given the client follows the server location process in
>     draft-ietf-sip-srv-00, causes the URL to be resolved to an
>     address/port on the proxy, and
> (2) is a URL that, if received by the proxy, the routing logic in the
>     proxy would cause the request to be forwarded to the Contact/From
>     address of the calling party.
>
> Similarly, the URL in the response must
> be something that
> (1) routes to the proxy, and
> (2) is a URL whose routing logic in the proxy would cause the request
>     to be forwarded to the Contact/From address of the called party.

  I'm ok with (1) on both of these.  (1) should be the same in both
directions, and the proxy must know this address itself.  If (1) isn't
the same in each direction, the proxy can insert two record-routes.

  I don't understand the reasoning behind (2).  If the message has a
Route, you do NOT want the routing logic in the proxy to be applied, as
this may have changed since the Route was set.  The request MUST be
routed according to the Route.

  Proxies which maintain state can of course put some magic parameter,
but I don't see why this must be unique in each direction.  However, if
some proxy requires it, then it too can put in two Record-Routes, and
then have enough smarts to remove both.

  I belive that all uses for Record-Route can be solved by using:

  <sip:proxy;state=cookie>

  With the exception of firewall proxies, where it is sufficient to use:

  <sip:firewall.proxy>

  If some middle-man snarfs the message, that's ok, because routing it
to the request-URI will get it to the correct destination: the proxy,
and that proxy can surely identify the message as being for it.  From
there, the proxy MUST use the Route to route it.


  Looking at your tuples:

  (A@originating.com;maddr=proxy, B@terminating.com;maddr=proxy)

  The only advantage I see is redundancy.  If proxy is down, then maybe
some clients might send the request to the rest of the address.
However, this will avoid all of the other proxies in the Route.  A
better idea would be for a smart node to instead drop this entry from
the Route and try the next URI.

  If this is purely information for the proxy to know, then fine
they're free to do it, but I don't see any reason why this needs to be
mentioned in the spec.  Seems like a special (and complicated) case to
me.  How is <sip:firewall.proxy> more brittle?  Trying to implement the
tuple scheme: finding an algorithm to copy the request-URI and add in an
maddr of the proxy, finding your record-route in the response and
mangling it, seems awfully brittle in itself.

  I have been told that using the maddr technique helps protect against
UAs which don't insert a Contact.  However, I can't come up with a
scenario, and now Contact is mandatory for INVITEs and 200-class
responses.

>> Additionally, if the proxy is keeping full call state, eg based on
>> unique Call-Id. Is it REQUIRED to mess with the Record-Route in the
>> response.  Otherwise, adding a simple
>
> I think yes, since otherwise we won't achieve this property of the
> request URI referring to a resource that would cause the request to be
> routed to the appropriate party.

  The appropriate party is the proxy.  After that, the appropriate party
is spelled out clearly by the Route.

--
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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From sip-admin@lists.bell-labs.com  Fri Nov 24 16:42:15 2000
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Subject: [SIP] Re: My comments on bis-02
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An updated version of -02 (to be submitted) is at the
www.cs.columbia.edu/sip location. I would appreciate if you could check
which of your remarks still apply. In terms of "fluff", the only way to
forward progress is if folks specifically identify redundant wording.
I'm all for shortening the spec, but not if it means that you have to
read FAQs, papers and mailing lists to start working on the
implementation.

Thanks for the comments.

Billy Biggs wrote:
> 
>   These are some of my (late) issues with bis-02.  BTW, It would help
> for commenting on the draft if there was a text file version.
> 
> Comments which may require discussion
> -------------------------------------
> 
> 1. In 4.2.1 there is discussion about how to handle INVITEs which cross
>    on the wire.  Are you worried about when two endpoints change their
>    contact address at the same time?  If you're just worried about the
>    session changing, it seems that it would be more
>    interoperability-friendly if implementations just re-re-INVITE'd when
>    they believe requests may have crossed, and use some internal random
>    timer.

This has been discussed a few times on the list, but may need to be
rolled up again.

> 
> 2. In 4.2.5, it is noted that CANCEL requests cannot have Route headers.
>    Why not?  Are you assuming that CANCEL is only useful for an initial
>    request?  This breaks since we need to be able to CANCEL a pending
>    REFER or other extension method which doesn't complete quickly, for
>    example.
> 

Removed.


> 3. Sections 6.25 and 10.1.1 (maybe others), as I mentioned in another
>    post to the list, imply that clients should choose a single local tag
>    and preserve this across calls.  This is a bad idea in cases when UAs
>    call themselves, may lead to UAs choosing the same tag value in some
>    cases, and also poses a security risk.
> 
>    I strongly recommend that the recommendation change to be that tags
>    should always be unique.  Apologies to those implementations that
>    want to try and preserve calls over reboots.  I don't see much point
>    to section 11.5 either.  Using the same tag doesn't solve all the
>    problems, so why bother trying?  It just makes things confusing and
>    prone to failure.

Obviously, separate discussion. Clarified as indicated in my email for
now.

> 
> 4. Regarding Authentication in 13.2, is it the responsibility of the
>    UAC to not use short form headers after the Authorization, or is it
>    the responsibility of the proxy to switch them when transforming the
>    message into canonical form?

Text says "Clients wishing to authenticate requests ... no short form
header fields". I'm not sure what's ambiguous here.

> 
>    I ask because there are extensions to SIP which propose new short
>    forms for headers that proxies will not know about.  An example of
>    this is the REFER draft.  Should the UAC not use these short forms,
>    or should extensions not propose them?
> 
> 5. Due to the proliferation of the Also header in BYE requests, I would
>    suggest that it be added to the bis draft, with its meaning
>    restricted to use with the BYE method.

Tentatively added.

> 
> 6. Mid-Call redirects.  The wording for the 305 response code seems to
>    indicate that it can be sent mid-call.  This breaks if you have a
>    Route.  I guess the only reasonable thing to do is say that if you
>    get redirected mid-call you should drop your route?  Or even better,
>    tell UAs that they cannot send a redirect response to a request with
>    a To tag?  Regardless, the bis draft should mention this case.

Any mid-call INVITE can return a 3xx. I can't see why 305 is special in
this regard. This does not necessarily mean that the route is bad, as
the Contact will just be added to the bottom. Yes, future requests may
travel strange hairpin paths, but, with care, a proxy can recuse itself
out of route.

> 
> 7. See my last post regarding the scope of the Call-ID.
> 
> Comments which are just bugs
> ----------------------------
> 
> 8. It would be nice in section 16 if all of the examples included from
>    tags.  As well, Figures 1 and 2 don't have tags.

Figure fixed; tags added.

> 
> 9. Section 4.2.6 explicitly mentions that REGISTER messages can be
>    redirected.  This seems redundant.  I also believe that there are
>    other areas of the spec where it is quite verbose and redundant.
>    While this is to the benefit of new readers in some cases, the draft
>    is also now quite large.  Can we maybe try and strip some of the
>    fluff out?

Section removed. Other suggestions welcome.

> 
> 10. Section 1.4.6 says Call-ID when it means "within the same call leg".
> 
> 11. In section 7.4.1, as was noted somewhere, "Missing Content-Length
>     header" is a bad example unless it is now mandatory.

Changed to Missing Call-ID header
> 
> 12. Section 1.4.2 implies that clients do not need to support UDP.  I
>     think this is a bug.
> 

Please see if this is still a problem.

> --
> Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Jonathan Rosenberg wrote:

> Before we begin writing new text, lets look at the sub-problems and build up
> a solution. I have another possibility that may be the best of all worlds.
> I'll start with Roberts list:
>
> 1. non SIP URLs in request URI
>
> We can either define record-route parameters, or else make sure that the URL
> put into the record route is a sip url. I originally propose RR parameters,
> but I think insisting on a SIP URL is backwards compatible and more
> reasonable. That seems to be consensus.

Please do not make this mistake. If we don't require a SIP URL in the
Request-URI,
the From,  the To, or even the Contact, then why mandate it in the
Record-Route:?

--
Sean Olson <sean.olson@ericsson.com>




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This is purely an implementation choice (obviously). There are so many
variables that enter into this choice that it is impossible to state
a definitive answer. I fall in the camp of doing as little as possible in
the parser, but there are good reasons for doing strict syntax checking
as you mention for some applications.

The "@" character should not be escaped. Escaping this character changes
the semantics of the URL.

My two cents
--
Sean Olson <sean.olson@ericsson.com>

kajun Robert wrote:

> hi,
>    I don't know what the parser should do and it seems that the parser has muddled me. Can you help me? I think it should check the message byte-by-byte to ensure that every character has fallen into a specific field, such as unreserved filed, DIGIT filed etc. Also the parser can find the syntax error besides the characters falling into some fields. For example, the parser should find the errors such as "%4q", 101.3332.3.5 (the Ipv4address) and so on in the Request-URL, although the characters are allowed to appear in the message.
>    Is that true? Also I am not sure the @ character can be escaped. If it can be escaped, the parser would not have mechanism to distinguish the one that the Spec requests with the one that the user's own escaped character?
>    Thanks.
>    Robert Kajun
>
> ________________________________________________________________________
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>
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Yes and Yes. You can choose different ports for each server or
you can do some clever magic to run both on the same port. I
leave it as an excercise for the reader to figure that one out.

--
Sean Olson <sean.olson@ericsson.com>

Arun Kumar Chippada wrote:

> Dear All,
>
> We want to know whether it is possible to run a SIP Proxy Server and a
> SIP User Agent on the same machine. Has anyone ever tried to do this?
>
> Arun.
>
> _______________________________________________
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Date: Fri, 24 Nov 2000 12:13:29 +0530
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My two cents of thoughts on this one... I don't know how feasible my
suggestion below is. The crux of this problem in my opinion here arises out
of the fact that the Transferor is first initiating an INVITE to check the
availability of the transfer target and depending upon the type of transfer,
the TT (or the transferree) initiates an INVITE to the transferree(or TT)
based either upon the request URL / contact URL received. This problem could
be solved if the transferee invoked the INVITE in all cases.
Venkatesh
----- Original Message -----
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
To: "'Billy Biggs'" <Billy_Biggs@3com.com>; "Robert Sparks"
<rsparks@dynamicsoft.com>
Cc: <sip@lists.bell-labs.com>
Sent: Friday, November 24, 2000 9:06 AM
Subject: RE: [SIP] Possible REFER problem?


> Sorry for the delay in picking this thread up... I'm trying to finish a
rev
> on Caller Prefs, and there are some issues here.
>
> First off, Eric raised a problem with my proposed solution. I'm
replicating
> it here since its been a while:
>
> > If you remember, the problem was when doing call transfer
> > with consultation
> > with the REFER request, how can we make sure that the INVITE
> > request (from
> > the Transferee to the Transfer Target) is going to reach the
> > exact same
> > User-Agent as the one that was consulted by the Transferor.
> >
> > The solution was to use the header "Accept-Contact" and
> > specify the contact
> > of the transfer target.
> >
> > I was taking a look at the caller preferences extension, thinking that
> > "Accept-Contact" had solved our problem, but I think it does
> > not solves the
> > problem.
> >
> > Suppose we have the following participants:
> >
> > Transferor (Tror)
> > - Sip address: sip:tror@acme.com
> > - Current location: sip:tror@office1.acme.com
> >
> > Transferee (Tree)
> > - Sip address: sip:tree@acme.com
> > - Current location: sip:tree@office1.acme.com
> >
> > Transfer-Target (TT)
> > - Sip address: sip:tt@acme.com
> > - Current location: sip:tt@office4.mediatrix.com
> >
> >
> > Tror and Tree have established calls.  Tror contacts TT and
> > ask if he wishes
> > to accept the call, which he does.  Tror sends the following
> > REFER to Tree:
> >
> > REFER sip:tree@office1.acme.com SIP/2.0
> > From: sip:tror@acme.com
> > To: sip:tree@acme.com
> > Refer-To: sip:tt@acme.com?Accept-Contact=sip:tt@office4.mediatrix.com
> > ...
> >
> > Which spawns the INVITE from Tree to a proxy server somewhere in ACME.
> >
> > INVITE sip:tt@acme.com SIP/2.0
> > From: sip:tree@acme.com
> > To: sip:tt@acme.com
> > Accept-Contact=sip:tt@office4.mediatrix.com
> > Referred-By: ...
> > ...
> >
> >
> > The proxy receives this request and checks with the location
> > server to find
> > where can tt@acme.com be found.  The list of contact returned
> > could look
> > like the following:
> >
> > Contact: sip:tt@office1.acme.com
> > Contact: sip:tt@mediatrix.com
> > Contact: sip:assistant@office1.acme.com
> >
> > Now the proxy has these 3 contacts that do not match the
> > Accept-Contact URL,
> > although the proxy at mediatrix.com would translate
> > tt@mediatrix.com to
> > tt@office4.mediatrix.com.  Now the proxy does not really know
> > which contact
> > is prefeered, and can try any of the three contacts, and
> > someone can happily
> > answer at those addresses.
> >
> > How can we avoid this?
> >
> > First, I think that the Proxy should send the Accept-Contact in its
> > downstream requests, in order to have the downstream proxies
> > be able to use
> > the same mechanism.  But in truth, the proxy does not know if
> > the downstream
> > request is going to be sent to a proxy or to a UA, so it is
> > possible for a
> > UA to receive such a request.  Would it be a good idea to
> > have the UA take a
> > look at the Accept-Contact header to know if the request is
> > really for him?
> > I think that the current caller preferences draft states that
> > a UA should
> > ignore these header if it receives it.
>
> Eric is correct that even if none of the registered contacts matches those
> in the Accept-Contact header, a proxy will forward the request, with
> Accept-Contact intact. As a result, the request may ultimately arrive at
> several UAs that don't match. Another related problem is that
Accept-Contact
> is not specified as a filter-all-but rule (partly for the reasons Eric
> observes; a request would never get past the first proxy since the Contact
> might be known only at a later proxy). The caller preferences
specification
> has a process for combining the q values. At best, it would effect
ordering
> of a sequential search.
>
> What we need here is a mechanism that says, "I only want to talk to you if
> you are one of these Contacts". This rule can ONLY be enforced in the UAs.
> This capability does not exist in the current caller preferences draft.
> However, I think it will prove useful here (actually, it really is needed
> for call transfer, more on that below), and other places too. So, I will
add
> an Accept-Only-Contact header to caller preferences, with this header only
> being used by a UA.
>
> Now, on to some other issues. Consider the scenario of A talking to B. A
> calls C, and then A  wishes to transfer B to C, making sure that B gets
the
> same instance of C that A was talking to. Adam asked, why not have A refer
B
> to the Contact obtained from C:
>
> > At the risk of sounding daft, I don't see why using the "Contact:"
> > header as the new "INVITE" URI is considered Evil in this case.
> >
> > When explaining the difference between, say, the "From:" (or "To:")
> > field and the "Contact:" field, I generally think of "From:"/"To:" as
> > being the persistent URI for contacting the appropriate resource in
> > an abstract sense (by resource, I mean "Adam Roach" or
> > "helpdesk staffer"
> > or "sales representative"), while the "Contact:" URI
> > indicates "where to
> > get to me (as a particular instance of that resource) on this
> > particular
> > device at this particular moment."
>
> The answer is that URIs most definitely have a context; a place from which
> they can be usefully used. In a perfect world with no NATs, firewalls, or
> devices that want to know whats going on within the network, it would work
> dandy. But thats not reality. The classic example is an ACD application,
> where help@helpdesk.com can be routed to a variety of agents
> (agent32@host32.helpdesk.com). Sophisticated software tracks the call
status
> of the various agents, how much volume they get, etc. Incoming calls MUST
> come in through this helpdesk proxy to work. The proxy always record
routes.
> In fact, to enforce this, the proxy has access to a hosts file with the
> host32.helpdesk.com address mapped to an IP address. Outside of this
proxy,
> that hostname is unroutable. So, if A called B, and then A called C, and C
> was one of these operators, the URI returned by C
> (agent32@host32.helpdesk.com) will definitely not be useful for B to use
> directly.
>
> Another issue was raised about the possibility of per-call Contacts
> generated dynamically by a UA. Yikes. I don't know any UA that does this.
> Unfortunately, if a UA provides no consistent form of identification to
the
> network or other users, it can't expect to be consistently transferred to
in
> this case. So, I think we can punt this case and declare it as "don't do
> this".
>
> Robert had proposed a hybrid approach:
> > Given all that, and the likelyhood that for many (if not
> > most) instances, the contact provided in the Contact URL
> > _is_ routable, what problems exist with taking the following
> > approach?
> >
> >   1) REFER B directly to the contact that C provided to A.
> >   2) If it fails, REFER B to C's well known URL with an Accept-Contact
> >      header containing the contact that C provided to A.
> >
>
> Hmm. Turns out you can construct cases where neither (1) nor (2) will
work.
>
> What if, when A called C, it wasn't really C that was called, but some
> time-of-day routed 800 number that got forwarded to C. Now, while A talks
to
> C, the logic for the forwarding changes. Now, when A sends a REFER to B,
the
> URI in the Refer-To is that 800 number. It never even comes close to being
> sent to a single UA that matches the Accept-Contact header. In this
> scenario, sending directly to the Contact won't work either, since the
> operator is not directly callable. Don't you love feature interaction
> problems ;) ?
>
> So, basically, referring to exactly the same person I'm talking to is hard
> because (1) their unique address may not be directly routable, and (2) the
> logic used to get to that person from a directly routable address may
> change. If you ignore (1), you come up with the solution to (2) of sending
> to the Contact. If you ignore (2), you come up with the
Accept-Only-Contact
> solution to deal with (1). If you want to deal with both (1) and (2), its
> really rough. I thought about having A also include Route headers as
header
> params to the URI in the Refer-To, but this doesn't work either. A route
> header set usable from A may not be usable from B, for many reasons, not
the
> least of which is problem (1) once more.
>
> Perhaps these cases are so extreme its not worth worrying about. I
wouldn't
> be surprised if they don't work in the PSTN either.
>
> So, despite its complexity, I feel compelled to recommend that Robert's
> hybrid approach be described in the REFER specification as the preferred
way
> to do transfer out of consultation, along with a nice long explanation of
> why it needs to be so. I'm not sure about the order though. I suspect that
> (2) will work in far more cases than (1) will. I welcome better ideas if
> anyone has any...
>
>
> Thanks,
> Jonathan R.
>
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>


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From sip-admin@lists.bell-labs.com  Sun Nov 26 19:42:49 2000
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Subject: RE: [SIP] Multiple headers of a given type.
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Date: Fri, 24 Nov 2000 08:17:01 -0500

1.  What is the preferred method (single header with multiple elements or
multiple headers with a single element).
2.  I assume you can get a mixture (multiple via headers, some with multiple
elements).
3.  Can a server, when resending, change the format (change the list format
to the multiple header format).  The server I'm writing decodes them but
encodes them the same way, regardless of how they were received.  I don't
keep track of how they were initially received.
4. Why support both methods in the first place.  Backward compatibility?



 

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Thursday, November 23, 2000 9:30 PM
To: 'mranga@nist.gov'; sip@lists.bell-labs.com
Subject: RE: [SIP] Multiple headers of a given type.


Since a comma separated list of values (indicated in BNF with #) is
identical to multiple headers of the same name, any header whose BNF starts
with # can appear more than once. 

-Jonathan R.
  
-----Original Message-----
>From: M. Ranganathan [mailto:mranga@nist.gov]
>Sent: Thursday, November 23, 2000 9:01 PM
>To: sip@lists.bell-labs.com
>Subject: [SIP] Multiple headers of a given type.
>
>
>Hello! 
>This is a rather simple question. Of all the SIP headers in the bis
document, which can 
>appear multiple times in a Request / Response ?  (For example,  VIA can
appear multiple 
>times, ACCEPT can appear multiple times etc. but FROM can appear only once
and TO can 
>appear only once. ) If there is a table in the document giving this
information  I have 
>missed,  please let me know (and apologies for redundant mail). 
>Thanks. 
>Regards 
>Ranga. 
-- 
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899. 
Tel: 301 975 3664 Fax: 301 590 0932
  

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Date: Sun, 26 Nov 2000 22:12:33 -0500

First off, questions on sip itself belong on the sip list
(sip@lists.bell-labs.com), not the sip forum. Answering your question:
  
-----Original Message-----
>From: Naresh Kumar Agarwal [mailto:naresh@mediaring.com.sg]
>Sent: Saturday, November 25, 2000 9:37 PM
>To: discussion@sipforum.org
>Subject: [SIPFORUM] Sending binary data instead of text
>
>
>Hi,
>
>Is it possible to send binary data in SIP at times apart from text.  

Yes, the body of SIP is 8 bit clean and can carry pure binary data.

>
>If yes, can we send the data in the binary form itself or does it needs to
be converted to 
>text first?

Pure binary.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
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Once again, please direct all sip technology questions to the sip list,
sip@lists.bell-labs.com.

  
-----Original Message-----
>From: Naresh Kumar Agarwal [mailto:naresh@mediaring.com.sg]
>Sent: Saturday, November 25, 2000 9:42 PM
>To: discussion@sipforum.org
>Subject: Re: [SIPFORUM] Queries
>
>
>Hi Jorgen,
>
>1. Hopefully, the following diagram should make things slightly clearer.
>
>
>SIP_CLIENT                     PROXY_LOCATOR                      SIP_PROXY
>|                                                   |
>|
>| 1)invite(client identifier, data1)       |
|
>|------------------------------------------------->|
>|
>| 2) 302 OK (SIP_PROXY_IP,data2) |
|
>|<-------------------------------------------------|
>|
>|                                                   |
>|
>| 3)invite(client identifier, data2)
|
>|--------------------------------------------------------------------------
---------------->--->|
>
>
>In this flow, data2 is something which is sent by the Proxy Locator upon
receiving the 
>invite from the SIP Client.  The Locator passes this data to the SIP Client
>but doesn't expect it to interpret it (it is intended for the destination
SIP Proxy).  But 
>at the same time, the SIP Client needs to interpret the IP address of the
SIP
>Proxy sent in the 302 OK.  

Your question came across as confusing, since there is no such thing as
"proxy locator" in the sip specifications or related documentations. This is
a redirect server.

The answer to your question, however, is yes. There is a way for the
redirect server to pass information to the client, which the client will
pass to the proxy. The client won't do anything with the data. It will be
placed in the request sent to the proxy. Of course, if this is non-standard
data, it will be meaningless to the proxy unless there is some kind of
specialized logic in the redirect server and proxy. Dangerous thing to do,
in general. If you need something from SIP thats not already there, you
should raise that issue instead of attempting non-standard extensions, which
is what this is.

ANyway, the way to send data is to encapsulate it as a header in the
redirected URL:

302 Try Here SIP/2.0
Contact: sip:user@host;maddr=proxy.me.com?newheader=foo

The client then sends an INVITE to proxy.me.com:

INVITE sip:user@host SIP/2.0
newheader: foo
From: sip:me@caller
.....



-Jonathan R.

----- Original Message ----- 
From: Jörgen Björkner 
To: discussion@sipforum.org 
Sent: Sunday, November 26, 2000 1:37 AM
Subject: RE: [SIPFORUM] Queries


Hi,
1) I am not sure that I really understand what you ask for. Could you
clarify with a small message flow diagram, since I am a but confused from
what you mean with client passing on data. 

2) I remember some discussion on the SIP list of using SIP for key exchange.
I think that the result was that there is a lot of things to think of when
designing a key interchange protocol, and therefore is it better to use one
of the already developed key exchange protocols for this purpose instead.
Although SIP provides digest authentication, but this requires a shared
secret between the parties. (Is this the right answer, should I add it to
the SIP FAQ?)

3) Yes, you can use PGP, for key distribution see answer 2)

/Jörgen
-----Original Message-----
From: owner-discussion@listserver.wineasy.se
[mailto:owner-discussion@listserver.wineasy.se]On Behalf Of Naresh Kumar
Agarwal
Sent: den 24 november 2000 17:49
To: discussion@sipforum.org
Subject: [SIPFORUM] Queries


Hi,

Would be grateful if any of you could help answer the following queries I
have:

1) During communication between a SIP client and a proxy locator (which is
used to find the location of a SIP proxy), is it possible for the proxy
locator to send some data (which is like a black box to the SIP client and
it does not need to interpret: something which it can simply pass on to the
SIP proxy later).  This data (black box) will be sent along with some other
data which the client might need to interpret.

Thus, what I want to find out is whether there is a SIP mechanism (some tag,
or something else) whereby a client can differentiate between data that it
needs to interpret and that it needs to simply pass on.

2) How do we do challenge in SIP?  Can we use the "challenge" mechanism for
key-exchange (for security reasons) in SIP?  If so, how?
3) I believe SIP supports PGP (Pretty Good Privacy).  In this case, how do
they actually get the key of the called-party?

Regards,

Naresh Agarwal.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Date: Sun, 26 Nov 2000 23:34:33 -0600

Jonathan Rosenberg (jdrosen@dynamicsoft.com):

> Anyway, the way to send data is to encapsulate it as a header in the
> redirected URL:
> 
> 302 Try Here SIP/2.0
> Contact: sip:user@host;maddr=proxy.me.com?newheader=foo
> 
> The client then sends an INVITE to proxy.me.com:
> 
> INVITE sip:user@host SIP/2.0
> newheader: foo
> From: sip:me@caller
> .....

  Picky point, but the start line should be:

  INVITE sip:user@host;maddr=proxy.me.com SIP/2.0

  Please be friendly to transparent proxies.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Mon, 27 Nov 2000 00:38:07 -0500



 

> -----Original Message-----
> From: Snyder, Duke [mailto:Duke.Snyder@inin.com]
> Sent: Friday, November 24, 2000 8:17 AM
> To: sip@lists.bell-labs.com
> Subject: RE: [SIP] Multiple headers of a given type.
> 
> 
> 1.  What is the preferred method (single header with multiple 
> elements or
> multiple headers with a single element).

There is none that is preferred. SIP does define a canonical form that is
used for authentication. My personal recommendation is to always generate
canonical form. Its in section 13.2 of rfc2543.

> 2.  I assume you can get a mixture (multiple via headers, 
> some with multiple
> elements).

Yes.

> 3.  Can a server, when resending, change the format (change 
> the list format
> to the multiple header format).  The server I'm writing 
> decodes them but
> encodes them the same way, regardless of how they were 
> received.  I don't
> keep track of how they were initially received.

Yes, it can change the format. However, if the request is authenticated,
you'd better spit it out in canonical form (and that had better be the form
it was received in).

> 4. Why support both methods in the first place.  Backward 
> compatibility?

We lifted much of SIPs operation from HTTP, which does this as well.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Date: Mon, 27 Nov 2000 00:05:08 -0600

Henning G. Schulzrinne (hgs@cs.columbia.edu):

> I'm not sure, but it seems that there's a simple fix: simply have the
> UA create *different* tags depending on whether this is for a From or
> To.  However, the tag is constant across calls and requests.

  Even safer to use "predictable-random", however:

  Choosing predictable tags does not make a UA crash-proof.  The SIP
spec should not recommend any behavior on generating tags.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Mon, 27 Nov 2000 00:15:09 -0600

Sean Olson (sean.olson@ericsson.com):

> > 1. non SIP URLs in request URI
> >
> > We can either define record-route parameters, or else make sure that
> > the URL put into the record route is a sip url. I originally propose
> > RR parameters, but I think insisting on a SIP URL is backwards
> > compatible and more reasonable. That seems to be consensus.
> 
> Please do not make this mistake. If we don't require a SIP URL in the
> Request-URI, the From,  the To, or even the Contact, then why mandate
> it in the Record-Route:?

  Do you want Record-Route and Contact header parameters indicating the
IP address and port to send the request?

  I don't mind non-SIP URLs, but I do mind new parameters: they just
provide a false sense of security that things will magically work.  If
you put a non-SIP URL in the Contact or Record-Route or Request-URI, you
had better know for sure that whomever is going to process the message
knows how to forward it on its own.

  Non-SIP URLs in the To/From don't matter because Contact is now
mandatory pretty much everywhere.  You don't need to look at the From to
route messages.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Mon, 27 Nov 2000 00:29:52 -0600

Henning G. Schulzrinne (hgs@cs.columbia.edu):

> An updated version of -02 (to be submitted) is at the
> www.cs.columbia.edu/sip location. I would appreciate if you could
> check which of your remarks still apply.

  Many thanks.  I will give more detailed comments soon.

> > 2. In 4.2.5, it is noted that CANCEL requests cannot have Route
> >    headers.  Why not?  Are you assuming that CANCEL is only useful
> >    for an initial request?  This breaks since we need to be able to
> >    CANCEL a pending REFER or other extension method which doesn't
> >    complete quickly, for example.
> 
> Removed.

  I think I was wrong on this one.  CANCEL is of hop-by-hop significance
within a transaction and doesn't use the Route.  Stateful proxies
instead remember and the UA sends the CANCEL just to the first entry in
the Route.

> > 6. Mid-Call redirects.  The wording for the 305 response code seems
> >    to indicate that it can be sent mid-call.  This breaks if you
> >    have a Route.  I guess the only reasonable thing to do is say
> >    that if you get redirected mid-call you should drop your route?
> >    Or even better, tell UAs that they cannot send a redirect
> >    response to a request with a To tag?  Regardless, the bis draft
> >    should mention this case.
> 
> Any mid-call INVITE can return a 3xx. I can't see why 305 is special
> in this regard. This does not necessarily mean that the route is bad,
> as the Contact will just be added to the bottom. Yes, future requests
> may travel strange hairpin paths, but, with care, a proxy can recuse
> itself out of route.

  I don't see how proxies can recuse themselves out of the Route.  It
has come up before on this list: once the Route is setup it may not be
modified, only the Contact at the end may change.

  Thank you for clarifying the meaning of a mid-call redirect.  Time to
fix code.

> > 12. Section 1.4.2 implies that clients do not need to support UDP.
> >     I think this is a bug.
> 
> Please see if this is still a problem.

  The text currently is:

  "If no protocol is specified, the client tries UDP (if UDP is
   supported).  If the attempt fails, or the client doesn't support UDP
   but supports other protocols, it tries those protocols in some
   unspecified order."

  This implies that clients do not need to support UDP.  I thought UDP
was a requirement, but it looks like it's only a SHOULD for clients.
MUST is only used in the section on a minimal implementation. 

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Subject: Re: [SIP] JAIN SIP - Issues to be resolved
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Date: Mon, 27 Nov 2000 15:05:02 +0000


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Ranga,

My comments are in red...

Chris


"M. Ranganathan" wrote:

> Chris,
>
> Thanks for the prompt response.  Hope you had a great Thanksgiving.

I wish there was such a thing as Thanksgiving in Ireland :(

>
>
> My followup is below. ( in blue)
>
> Regards
>
> Ranga.
>
> Chris Harris wrote:
>
>> Ranga,
>>
>> See comments below.
>>
>> Regards,
>> Chris
>>
>> "M. Ranganathan" wrote:
>>
>> >
>> > Chris, Erik,
>> >
>> > While we are on the subject of Exceptions, why do null arguments
>> > throw runtime exceptions when you do a get on them? For example, in
>> > the Accept interface, getContentSubtype throws a
>> > IllegalArgumentException if  is null. IMHO, I think it is valid to
>> > have a null subtype so why not just return null rather than throw
>> > exception here?
>>
>> By a "doing a get on a null argument" here I assume you mean
>> attempting to get a non-existent optional field? If you do this a
>> checked ParameterNotSetException is thrown, not a runtime exception.
>
>
> Yes. Thats what I had meant.  The word "argument" should have been
> "field".   Why do you need a  ParameterNotSet exception in this case?
> Are you trying to handle the case of basic types like integers, floats
> and so on ? For objects, null is synonymous with not set isnt it?

We discussed this matter in depth at the last JAIN SIP meeting in
October, and we came to the conclusion that getting and setting null
values is not a good idea for several reasons e.g. it is easier to debug
a ParameterNotSetException than searching for the source of a later
NullPointerException, and it forces the application to allow for the
fact that there is a reasonable chance that a particular field does not
exist. [Of course, if we did allow the use of null, we could cut down on
the number of methods and drop the ParameterNotSetException - but  is it
really worth sacrificing clarity and robustness for the sake of cutting
down on API size?]

>
>
>>
>>
>> There are two approaches to handling optional header parts. One
>> approach is to allow the passing and returning of null objects from
>> get and set methods (implicit "remove" and "has"), and the other is
>> to explicitly add has and remove methods (and exceptions on the set
>> and get methods). As for mandatory header parts - I think passing
>> null to a set method should definitely throw an
>> IllegalArgumentException.
>
> I have the following questions about setXXX methods:
>
> 1.  Why do you restrict throwing IllegalArgumentException to only null
> arguments in some of the headers? Why are you not checking for proper
> formatting of the passed arguments (by giving it to the parser for
> example and running the appropriate  parser rule).

In general, an IllegalArgumentException is thrown if an argument to a
set method fails basic validation (i.e. no implementation-specific
parsing is used). Note that an application can ensure that this
exception is not thrown by following the rules defined in the API . So
if an argument is a String, I throw an IllegalArgumentException if the
argument is null or zero-length.

Of course this is not sufficient validation and the string must be
parsed to check for whitespaces, newlines etc, and different
implementations will be more tolerant than others. This is why there is
a checked JainSipParseException - an application cannot tell in advance
if a particular implementation will accept a certain value, so it must
catch this exception.

I am proposing that this exception should also be thrown by get methods
(for all fields), because if an implementation uses lazy parsing, then
there is no guarantee that the header value will parse correctly upon
the invocation of the get method, and again this is
implementation-dependent.

>
>
> 2.  A more general question - why do you have set methods at all in
> the interfaces?  The class that implements the interface can have set
> methods but the interface itself does not need to have them.

I have set methods in the interfaces because the objects will not only
be coming up from the implementation - an application needs to be able
to create and send requests, and this means being able to set the
values.

>  Moreover, the parser has the job of setting up stuff in the returned
> object. It has a set of BNF rules to decide on the correctness of the
> arguments and it does not need for the class to re-implement the
> check. For example, I think it ought to be the parser's job to check
> if the q value is between 0 and 1.   (Is the purpose of the set method
> to allow for the application to do header formatting?)

I absolutely agree that the parser should perform validation (hence the
JainSipParseException) - but as I mentioned above there are certain
basic criteria that must be met by all implementations and applications
- I am using the IllegalArgumentException to cover these. Perhaps
certain cases go too far in this validation - perhaps the
IllegalArgumentException could be thrown only if a null argument is
passed and all other validation is left up to the implementation's
parser? (although a problem with certain cases such as PriorityHeader
comes to mind, where the header values are defined as constants in the
API and an IllegalArgumentException is thrown if any other value is
passed - should we allow any non-null String values for the priority? Or
to take your example of q-value above - should we just allow any
non-null Float and let the parser enforce the 0-1 range?


>
>
>
>>
>>
>> (If it is considered acceptable to have a "null" subType then we
>> should make it optional)
>>
>> >
>> >
>> > I don't quite follow the argument that lazy parsing  "necessitates
>> > throwing a parse exception for all get methods".   Here is what I
>> > had envisioned:
>> >
>> >
>> > 1 When a message comes in it will be separated into headers based
>> > on type.
>> > 2 The header would be returned with no field initialized except the
>> > text corresponding to the message. (this would necessitate
>> > examining the first few bytes of the message to determine the
>> > header type but the rest remains unparsed.) You can also stow away
>> > the type information in the Header that is returned as well to know
>> > which parser rule to invoke later.
>> > 3. Later, when the application wants to parse the header, it has
>> > the string corresponding to the message and can hand it off to the
>> > parser, which can then fill in the appropriate fields corresponding
>> > to the header.
>> > 4. Subsequently, the application can access the fields just as an
>> > eager parser based application would do.
>> >
>> > ( Where am I missing the boat with this? )
>>
>> An application has no idea whether an implementation is using lazy
>> or eager parsing - it is simply passed a header object - and then if
>> it is interested in a particular field it calls the appropriate get
>> method (which would internally invoke the parser in a lazy parsing
>> implementation). So the application is completely oblivious to
>> whether or not this "optimisation" is used by the implementation.
>
>
> Ah! ( Why is this hiding  of the implementation so important? )  Can
> this be handled in the following fashion by an implementation:
>
> 1. The implementation keeps a flag in the  class that implements the
> interface whether the parsing has been done or not (i.e. boolean
> parsingDone flag).
>
> 2. The Implementation keeps the entire string corresponding to the
> header in the object. So the parser has to peek into the message
> header to know what type the message is as soon as it has the  header
> string and then grabs the string and stores it away for lazy parsing.
>
>
> 3. The get Method  checks if the flag has been set, calls the parser
> to do the parse if the flag has not been set and then returns a
> parseException if the parse fails. This can be done transparently to
> the caller.

This is exactly what I would expect a lazy-parsing implementation to do
- the fact that lazy-parsing is hidden from the application (except that
the application must catch a JainSipParseException on the get methods to
allow for the possibility that an implementation uses lazy-parsing - so
technically the application will know that an implementation uses
lazy-parsing if this exception is thrown by a get method - but I think
the two cases are handled nicely by the addition of the exception)

>
>
>
>
>
>
>>
>>
>> If the field is optional and does not exist in the header a
>> ParameterNotSetException will be thrown by the implementation.
>
> A suggestion (shoot it down if it does not jive):
>
> 1. All fields are represented by objects (not simple types like int,
> use Integer instead).

I buy this part - then the IllegalArgumentException is only ever thrown
if an object parameter is null, and the JainSipParseException is thrown
for all other problems with the argument.

>
> 2. If the field is not set just return null

See above for null discussion,

>
>
>
>
>> If the field is mandatory and does not exist, a
>> JainSipParseException will be thrown (header is malformed). If any
>> header field is malformed a JainSipParseException is thrown. If a
>> JainSipParseException is thrown, this indicates that the application
>> must use the header's getValue method to get a string representation
>> of the header value and do what it can with that.
>
>
> Why the headers getValue ? Why is the string not part of the
> Exception?

Sure - we can say that the JainSipParseException's message must be the
string value of the header.

>
>
>
>>
>>
>> >
>> >
>> > It appears that what is being attempted here (please correct me if
>> > I am wrong) is to use exceptions as a mechanism to catch
>> > un-initialized fields so the parser can be invoked in the exception
>> > handler perhaps. This leads to a lot of overhead in clunky code. I
>> > think it is a mistake to throw exceptions on every get to try to
>> > protect the application from inadvertently accessing fields that
>> > have not been initialized.  Why not just assume that the
>> > application must be smart enough to parse the header text before
>> > accessing fields and not try to provide a complex mechanism as
>> > above?
>>
>>
>> We are using ParameterNotSetException for the case of an optional
>> header not existing, and a JainSipParseException for the case of a
>> malformed header being received from the network.
>>
>> If we assume the application is smart enough to do all header value
>> parsing, we reduce a header to just being two strings - the type and
>> value, which puts the burden of parsing on an application - this
>> work can be done by the underlying implementation - doesn't it make
>> more sense for an application to catch an exception than have to
>> parse all header value strings (especially since this functionality
>> already exists in the underlying stack)
>
>
> No that is not what I had intended. Parser functionality exists in the
> stack but has to be called explicitly by the implementation. Too much
> hiding of the implementation from the application has its own set of
> disadvantages (IMHO).

I wanted parser fuctionality to be invoked implicitly by an application
- from the API perspective it is an implementation detail - so the
application should not be aware of how or when the underlying parser is
invoked - it should just know that it is there in some form, and that if
it tries to get or set a value that the implementation doesn't like it
will get a JainSipParseException thrown back at it. This is important if
we want to maintain the three core values of JAIN - portability,
portability and portabilty ;)

>
>
>
>
>>
>>
>> >
>> >
>> >
>> > Regards,
>> >
>> > Ranga.
>> >
>> > Chris Harris wrote:
>> >
>> >>  Erik,
>> >>
>> >>  The implication of what you say is that all set methods in the
>> >>  header
>> >>  interfaces which take String arguments must throw a checked
>> >>  JainSipParseException. I think this is reasonable given that
>> >>  different
>> >>  implementations will be more tolerant than others. However, in
>> >>  order to allow
>> >>  lazy parsing by an implementation, all get methods in the header
>> >>  interfaces
>> >>  (regardless of return type) must throw a checked parse exception.
>> >>  This is a
>> >>  lot of checked exceptions - and will make life more awkward for
>> >>  the
>> >>  application developer. It would certainly be much less effort to
>> >>  not allow
>> >>  lazy parsing - there would be no need to throw a parse exception
>> >>  for all get
>> >>  methods - greatly reducing the number of checked exceptions.
>> >>  Unfortunately it
>> >>  has been said that JAIN SIP support for lazy parsing is a must.
>> >>  If it is then
>> >>  I guess there is no other choice than to include the exceptions?
>> >>
>> >>  Regards,
>> >>  Chris
>> >>
>> >>  Erik Nissen wrote:
>> >>
>> >>  > Chris Harris wrote:
>> >>  >
>> >>  > >
>> >>  > > We need to decide whether this should be a checked or runtime
>> >>  exception.
>> >>  >
>> >>  > The rule of thumb is to use:
>> >>  > * checked exceptions when the program should handle situations
>> >>  which
>> >>  > can/will occur.
>> >>  > * runtime exceptions when there are programming error (God
>> >>  forbid :-)
>> >>  >
>> >>  > So:
>> >>  > Parse errors should throw checked exceptions.
>> >>  > Becaurse parse errors are outside the control of the programmer
>> >>
>> >>  > and he/she should be forced to handle such conditions in the
>> >>  code.
>> >>  >
>> >>  > Null/wrong arguments should throw runtime exceptions.
>> >>  >
>> >>  > If it is possible, reduce the number of methods throwing
>> >>  (checked)
>> >>  > exceptions
>> >>  > to a minimum.
>> >>  >
>> >>  > Best regards,
>> >>  > /Erik Nissen,
>> >>  > Ericsson Utveckling AB
>> >>
>> >>  _______________________________________________
>> >>  SIP mailing list
>> >>  SIP@lists.bell-labs.com
>> >>  http://lists.bell-labs.com/mailman/listinfo/sip
>> >
>> > --
>> > M.Ranganathan
>> > NIST Advanced Networking Technologies Group,
>> > 100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
>> > Tel: 301 975 3664 Fax: 301 590 0932
>> >
>> >
>>
> --
> M.Ranganathan
> NIST Advanced Networking Technologies Group,
> 100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
> Tel: 301 975 3664 Fax: 301 590 0932
>
>

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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<body text="#000000" bgcolor="#FFFFFF" link="#0000FF" vlink="#FF0000" alink="#000088">
<font color="#FF0000">Ranga,</font><font color="#FF0000"></font>
<p><font color="#FF0000">My comments are in red...</font><font color="#FF0000"></font>
<p><font color="#FF0000">Chris</font>
<br>&nbsp;
<p>"M. Ranganathan" wrote:
<blockquote TYPE=CITE>Chris,
<p>Thanks for the prompt response.&nbsp; Hope you had a great Thanksgiving.</blockquote>
<font color="#FF0000">I wish there was such a thing as Thanksgiving in
Ireland :(</font>
<blockquote TYPE=CITE>&nbsp;
<p>My followup is below. ( in blue)
<p>Regards
<p>Ranga.
<p>Chris Harris wrote:
<blockquote TYPE=CITE>Ranga,
<p>See comments below.
<p>Regards,
<br>Chris
<p>"M. Ranganathan" wrote:
<blockquote TYPE=CITE>&nbsp;
<br>Chris, Erik,
<p>While we are on the subject of Exceptions, why do null arguments throw
runtime exceptions when you do a get on them? For example, in the Accept
interface, getContentSubtype throws a IllegalArgumentException if&nbsp;
is null. IMHO, I think it is valid to have a null subtype so why not just
return null rather than throw exception here?</blockquote>
By a "doing a get on a null argument" here I assume you mean attempting
to get a non-existent optional field? If you do this a checked ParameterNotSetException
is thrown, not a runtime exception.</blockquote>

<p><br><font color="#3333FF">Yes. Thats what I had meant.&nbsp; The word
"argument" should have been "field".&nbsp;&nbsp; Why do you need a&nbsp;
ParameterNotSet exception in this case?&nbsp; Are you trying to handle
the case of basic types like integers, floats and so on ? For objects,
null is synonymous with not set isnt it?</font></blockquote>
<font color="#FF0000">We discussed this matter in depth at the last JAIN
SIP meeting in October, and we came to the conclusion that getting and
setting null values is not a good idea for several reasons e.g. it is easier
to debug a ParameterNotSetException than searching for the source of a
later NullPointerException, and it forces the application to allow for
the fact that there is a reasonable chance that a particular field does
not exist. [Of course, if we did allow the use of null, we could cut down
on the number of methods and drop the ParameterNotSetException - but&nbsp;
is it really worth sacrificing clarity and robustness for the sake of cutting
down on API size?]</font>
<blockquote TYPE=CITE><font color="#3333FF"></font>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<p>There are two approaches to handling optional header parts. One approach
is to allow the passing and returning of null objects from get and set
methods (implicit "remove" and "has"), and the other is to explicitly add
has and remove methods (and exceptions on the set and get methods). As
for mandatory header parts - I think passing null to a set method should
definitely throw an IllegalArgumentException.</blockquote>
<font color="#3333FF">I have the following questions about setXXX methods:</font>
<p><font color="#3333FF">1.&nbsp; Why do you restrict throwing IllegalArgumentException
to only null arguments in some of the headers? Why are you not checking
for proper formatting of the passed arguments (by giving it to the parser
for example and running the appropriate&nbsp; parser rule).</font></blockquote>
<font color="#FF0000">In general, an IllegalArgumentException is thrown
if an argument to a set method fails basic validation (i.e. no implementation-specific
parsing is used). Note that an application can ensure that this exception
is not thrown by following the rules defined in the API . So if an argument
is a String, I throw an IllegalArgumentException if the argument is null
or zero-length.</font><font color="#FF0000"></font>
<p><font color="#FF0000">Of course this is not sufficient validation and
the string must be parsed to check for whitespaces, newlines etc, and different
implementations will be more tolerant than others. This is why there is
a checked JainSipParseException - an application cannot tell in advance
if a particular implementation will accept a certain value, so it must
catch this exception.</font><font color="#FF0000"></font>
<p><font color="#FF0000">I am proposing that this exception should also
be thrown by get methods (for all fields), because if an implementation
uses lazy parsing, then there is no guarantee that the header value will
parse correctly upon the invocation of the get method, and again this is
implementation-dependent.</font>
<blockquote TYPE=CITE><font color="#3333FF"></font>&nbsp;
<p><font color="#3333FF">2.&nbsp; A more general question - why do you
have set methods at all in the interfaces?&nbsp; The class that implements
the interface can have set methods but the interface itself does not need
to have them.</font></blockquote>
<font color="#FF0000">I have set methods in the interfaces because the
objects will not only be coming up from the implementation - an application
needs to be able to create and send requests, and this means being able
to set the values.</font>
<blockquote TYPE=CITE><font color="#3333FF">&nbsp;Moreover, the parser
has the job of setting up stuff in the returned object. It has a set of
BNF rules to decide on the correctness of the arguments and it does not
need for the class to re-implement the check. For example, I think it ought
to be the parser's job to check if the q value is between 0 and 1.&nbsp;&nbsp;
(Is the purpose of the set method to allow for the application to do header
formatting?)</font></blockquote>
<font color="#FF0000">I absolutely agree that the parser should perform
validation (hence the JainSipParseException) - but as I mentioned above
there are certain basic criteria that must be met by all implementations
and applications - I am using the IllegalArgumentException to cover these.
Perhaps certain cases go too far in this validation - perhaps the IllegalArgumentException
could be thrown only if a null argument is passed and all other validation
is left up to the implementation's parser? (although a problem with certain
cases such as PriorityHeader comes to mind, where the header values are
defined as constants in the API and an IllegalArgumentException is thrown
if any other value is passed - should we allow any non-null String values
for the priority? Or to take your example of q-value above - should we
just allow any non-null Float and let the parser enforce the 0-1 range?</font>
<br><font color="#FF0000"></font>&nbsp;
<blockquote TYPE=CITE><font color="#3333FF"></font>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<p>(If it is considered acceptable to have a "null" subType then we should
make it optional)
<blockquote TYPE=CITE>&nbsp;
<p>I don't quite follow the argument that lazy parsing&nbsp; "necessitates
throwing a parse exception for all get methods".&nbsp;&nbsp; Here is what
I had envisioned:
<br>&nbsp;
<p>1 When a message comes in it will be separated into headers based on
type.
<br>2 The header would be returned with no field initialized except the
text corresponding to the message. (this would necessitate examining the
first few bytes of the message to determine the header type but the rest
remains unparsed.) You can also stow away the type information in the Header
that is returned as well to know which parser rule to invoke later.
<br>3. Later, when the application wants to parse the header, it has the
string corresponding to the message and can hand it off to the parser,
which can then fill in the appropriate fields corresponding to the header.
<br>4. Subsequently, the application can access the fields just as an eager
parser based application would do.
<p>( Where am I missing the boat with this? )</blockquote>
An application has no idea whether an implementation is using lazy or eager
parsing - it is simply passed a header object - and then if it is interested
in a particular field it calls the appropriate get method (which would
internally invoke the parser in a lazy parsing implementation). So the
application is completely oblivious to whether or not this "optimisation"
is used by the implementation.</blockquote>

<p><br>Ah! ( Why is this hiding&nbsp; of the implementation so important?
)&nbsp; Can this be handled in the following fashion by an implementation:
<p>1. The implementation keeps a flag in the&nbsp; class that implements
the interface whether the parsing has been done or not (i.e. boolean parsingDone
flag).
<p>2. The Implementation keeps the entire string corresponding to the header
in the object. So the parser has to peek into the message header to know
what type the message is as soon as it has the&nbsp; header string and
then grabs the string and stores it away for lazy parsing.
<br>&nbsp;
<p>3. The get Method&nbsp; checks if the flag has been set, calls the parser
to do the parse if the flag has not been set and then returns a parseException
if the parse fails. This can be done transparently to the caller.</blockquote>
<font color="#FF0000">This is exactly what I would expect a lazy-parsing
implementation to do - the fact that lazy-parsing is hidden from the application
(except that the application must catch a JainSipParseException on the
get methods to allow for the possibility that an implementation uses lazy-parsing
- so technically the application will know that an implementation uses
lazy-parsing if this exception is thrown by a get method - but I think
the two cases are handled nicely by the addition of the exception)</font>
<blockquote TYPE=CITE>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<p>If the field is optional and does not exist in the header a ParameterNotSetException
will be thrown by the implementation.</blockquote>
A suggestion (shoot it down if it does not jive):
<p>1. All fields are represented by objects (not simple types like int,
use Integer instead).</blockquote>
<font color="#FF0000">I buy this part - then the IllegalArgumentException
is only ever thrown if an object parameter is null, and the JainSipParseException
is thrown for all other problems with the argument.</font>
<blockquote TYPE=CITE>&nbsp;
<br>2. If the field is not set just return null</blockquote>
<font color="#FF0000">See above for null discussion,</font>
<blockquote TYPE=CITE>&nbsp;
<br>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>If the field is mandatory and does not exist, a JainSipParseException
will be thrown (header is malformed). If any header field is malformed
a JainSipParseException is thrown. If a JainSipParseException is thrown,
this indicates that the application must use the header's getValue method
to get a string representation of the header value and do what it can with
that.</blockquote>

<p><br>Why the headers getValue ? Why is the string not part of the Exception?</blockquote>
<font color="#FF0000">Sure - we can say that the JainSipParseException's
message must be the string value of the header.</font>
<blockquote TYPE=CITE>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<p>It appears that what is being attempted here (please correct me if I
am wrong) is to use exceptions as a mechanism to catch un-initialized fields
so the parser can be invoked in the exception handler perhaps. This leads
to a lot of overhead in clunky code. I think it is a mistake to throw exceptions
on every get to try to protect the application from inadvertently accessing
fields that have not been initialized.&nbsp; Why not just assume that the
application must be smart enough to parse the header text before accessing
fields and not try to provide a complex mechanism as above?</blockquote>

<p><br>We are using ParameterNotSetException for the case of an optional
header not existing, and a JainSipParseException for the case of a malformed
header being received from the network.
<p>If we assume the application is smart enough to do all header value
parsing, we reduce a header to just being two strings - the type and value,
which puts the burden of parsing on an application - this work can be done
by the underlying implementation - doesn't it make more sense for an application
to catch an exception than have to parse all header value strings (especially
since this functionality already exists in the underlying stack)</blockquote>

<p><br><font color="#3333FF">No that is not what I had intended. Parser
functionality exists in the stack but has to be called explicitly by the
implementation. Too much hiding of the implementation from the application
has its own set of disadvantages (IMHO).</font></blockquote>
<font color="#FF0000">I wanted parser fuctionality to be invoked <i>implicitly</i>
by an application - from the API perspective it is an implementation detail
- so the application should not be aware of how or when the underlying
parser is invoked - it should just know that it is there in some form,
and that if it tries to get or set a value that the implementation doesn't
like it will get a JainSipParseException thrown back at it. This is important
if we want to maintain the three core values of JAIN - portability, portability
and portabilty ;)</font>
<blockquote TYPE=CITE><font color="#3333FF"></font>&nbsp;
<br>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<br>&nbsp;
<p>Regards,
<p>Ranga.
<p>Chris Harris wrote:
<blockquote TYPE=CITE>Erik,
<p>The implication of what you say is that all set methods in the header
<br>interfaces which take String arguments must throw a checked
<br>JainSipParseException. I think this is reasonable given that different
<br>implementations will be more tolerant than others. However, in order
to allow
<br>lazy parsing by an implementation, all get methods in the header interfaces
<br>(regardless of return type) must throw a checked parse exception. This
is a
<br>lot of checked exceptions - and will make life more awkward for the
<br>application developer. It would certainly be much less effort to not
allow
<br>lazy parsing - there would be no need to throw a parse exception for
all get
<br>methods - greatly reducing the number of checked exceptions. Unfortunately
it
<br>has been said that JAIN SIP support for lazy parsing is a must. If
it is then
<br>I guess there is no other choice than to include the exceptions?
<p>Regards,
<br>Chris
<p>Erik Nissen wrote:
<p>> Chris Harris wrote:
<br>>
<br>> >
<br>> > We need to decide whether this should be a checked or runtime exception.
<br>>
<br>> The rule of thumb is to use:
<br>> * checked exceptions when the program should handle situations which
<br>> can/will occur.
<br>> * runtime exceptions when there are programming error (God forbid
:-)
<br>>
<br>> So:
<br>> Parse errors should throw checked exceptions.
<br>> Becaurse parse errors are outside the control of the programmer
<br>> and he/she should be forced to handle such conditions in the code.
<br>>
<br>> Null/wrong arguments should throw runtime exceptions.
<br>>
<br>> If it is possible, reduce the number of methods throwing (checked)
<br>> exceptions
<br>> to a minimum.
<br>>
<br>> Best regards,
<br>> /Erik Nissen,
<br>> Ericsson Utveckling AB
<p>_______________________________________________
<br>SIP mailing list
<br>SIP@lists.bell-labs.com
<br><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></blockquote>

<pre>--&nbsp;
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.&nbsp;
Tel: 301 975 3664 Fax: 301 590 0932</pre>
&nbsp;</blockquote>
</blockquote>

<pre>--&nbsp;
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.&nbsp;
Tel: 301 975 3664 Fax: 301 590 0932</pre>
&nbsp;</blockquote>

</body>
</html>

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Billy Biggs wrote:

> Sean Olson (sean.olson@ericsson.com):
>
> > > 1. non SIP URLs in request URI
> > >
> > > We can either define record-route parameters, or else make sure that
> > > the URL put into the record route is a sip url. I originally propose
> > > RR parameters, but I think insisting on a SIP URL is backwards
> > > compatible and more reasonable. That seems to be consensus.
> >
> > Please do not make this mistake. If we don't require a SIP URL in the
> > Request-URI, the From,  the To, or even the Contact, then why mandate
> > it in the Record-Route:?
>
>   Do you want Record-Route and Contact header parameters indicating the
> IP address and port to send the request?

I would be happy with that.

>
>
>   I don't mind non-SIP URLs, but I do mind new parameters: they just
> provide a false sense of security that things will magically work.  If
> you put a non-SIP URL in the Contact or Record-Route or Request-URI, you
> had better know for sure that whomever is going to process the message
> knows how to forward it on its own.

Hmmm. Sort of like using a non-SIP URL in say, the To/From/Contact ...
If we can't use parameters for things like this, then why do we have
parameters
at all??? (I mean parameters on the RR)

>
>
>   Non-SIP URLs in the To/From don't matter because Contact is now
> mandatory pretty much everywhere.  You don't need to look at the From to
> route messages.
>

But the To/From is there for a reason. At a transaction level, I am all for
treating them as opaque strings. Higher up in the application strata,
these should have some meaning besides just strings -- at least there
should be the possibility for these to have meaning.

--
Sean Olson <sean.olson@ericsson.com>


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Sean Olson (sean.olson@ericsson.com):

>>>> 1. non SIP URLs in request URI
>>
>>   Do you want Record-Route and Contact header parameters indicating
>> the IP address and port to send the request?
> 
> I would be happy with that.
>
>>   I don't mind non-SIP URLs, but I do mind new parameters: they just
>> provide a false sense of security that things will magically work.
>> If you put a non-SIP URL in the Contact or Record-Route or
>> Request-URI, you had better know for sure that whomever is going to
>> process the message knows how to forward it on its own.
> 
> Hmmm. Sort of like using a non-SIP URL in say, the To/From/Contact ...
> If we can't use parameters for things like this, then why do we have
> parameters at all??? (I mean parameters on the RR)

  Sorry, let me be more specific.  Say I use a non-SIP URI in the RR (or
Contact) of:

  <tel:911>;raddr=udp/22.33.44.55:5060

  My only choice for request-URI is "tel:911", but this does not convey
the destination of the request.  Since you now can't route based on
request-URI, you can't use an outbound proxy, stuff breaks in general.

  Now, what if I have a cluster of proxies which all route messages
internally using tel URIs.  I have no problem with this, and placing
these non-SIP URIs in the RR totally makes sense.  But I see two issues:

  1. Should the spec instruct UAs to copy RR parameters into the Route,
     so that the raddr can be useful within this cluster?

     I think yes, definitely.  This is cheap and useful.

  2. Should we standardize an raddr RR parameter and allow this to be
     used for SIP URIs?

     I say no.  In general this technique breaks with outbound proxies
     or transparent proxies.  If necessary, this could be standardized
     outside of the SIP spec.


  I got the impression earlier that you were looking for point 2 to be
in the spec.

>>   Non-SIP URLs in the To/From don't matter because Contact is now
>> mandatory pretty much everywhere.  You don't need to look at the
>> From to route messages.
> 
> But the To/From is there for a reason. At a transaction level, I am
> all for treating them as opaque strings. Higher up in the application
> strata, these should have some meaning besides just strings -- at
> least there should be the possibility for these to have meaning.

  Totally, but a client should know to convert a tel URI to a SIP URI
like phonenumber@gateway.  I thought that was the point of them, not to
use them as a Request-URI.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Subject: [SIP] What SIP doesn't tell you how to do.
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Date: Mon, 27 Nov 2000 08:29:31 -0800 (PST)

Apparently SIP doesn't tell you how to do any of the
following as well:

1. Be professional.
2. Show others common courtesy.
3. Switch to decaf.

BTW- He never asked how to do anything, just if it
could be done.

-----Original Message-----
From: Jonathan Rosenberg
[mailto:jdrosen@dynamicsoft.com]
Sent: Wednesday, November 22, 2000 10:59 AM
To: 'Sanjiv Deshpande'; sip@lists.bell-labs.com
Subject: RE: [SIP] multiple location


SIP is just a protocol. Thats it. Its a tool. If you
want device A to call
device B, it tells you how to do that. Like all tools,
you can use them in
many different ways. When you buy a hammer in Home
Depot, does it tell you
whether to use your left hand or right? How many
fingers to grip with it?
The size of the nails to use with it? Of course not. 

However, for some reason that I cannot fathom, many
people seem to believe
the opposite is true for protocols; that is, protocols
need to insist on how
syntax and semantics AND how they are applied to solve
real world problems.
This approach just results in systems that are useful
to no one. Would you
buy a hammer that said "for nails of length 1 inch
only?" I doubt it.

So, to be specific, SIP doesn't tell you how to do any
of the following:

1. the color of the phones that use SIP
2. the GUI for SIP calls
3. how you build a network of proxy servers
4. whether a phone can make calls while its in a call
5. qualifications for human beings to make use of SIP
softphones (i.e., IQ
levels, height, or what have you).
6. how many simultaneous calls the device can be in

Thanks,
Jonathan R.



---
Jonathan D. Rosenberg                       72 Eagle
Rock Ave.
Chief Scientist                             First
Floor
dynamicsoft                                 East
Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:  
(973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE:
(973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Sanjiv Deshpande
[mailto:sdeshpan@telcordia.com]
> Sent: Wednesday, November 22, 2000 11:36 AM
> To: sip@lists.bell-labs.com
> Subject: [SIP] multiple location
> 
> 
> 
> 
> 
> 
> Q:
> INVITE Sent by UAC got forked and reached two UAS1
and UAS2 , 
>  the UAC decided
> to talk to UAS1. Is it possible that UAS2 (Assuming
it is a 
> client also) can
> make a call to someone else.
> 
> 
> What I wanted to know is, A user can be contacted at
multiple 
> location if one of
> the location answers a CALL is it possible for
somebody else 
> on behalf of that
> user make CALLS or receive CALLS....
> 
> 
> Thanks
> Sanjiv.
> 
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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>   Sorry, let me be more specific.  Say I use a non-SIP URI in the RR (or
> Contact) of:
>
>   <tel:911>;raddr=udp/22.33.44.55:5060
>
>   My only choice for request-URI is "tel:911", but this does not convey
> the destination of the request.  Since you now can't route based on
> request-URI, you can't use an outbound proxy, stuff breaks in general.

Why doesn't the outbound proxy RR?

>
>   Now, what if I have a cluster of proxies which all route messages
> internally using tel URIs.  I have no problem with this, and placing
> these non-SIP URIs in the RR totally makes sense.  But I see two issues:
>
>   1. Should the spec instruct UAs to copy RR parameters into the Route,
>      so that the raddr can be useful within this cluster?
>
>      I think yes, definitely.  This is cheap and useful.
>

Agreed

>
>   2. Should we standardize an raddr RR parameter and allow this to be
>      used for SIP URIs?
>
>      I say no.  In general this technique breaks with outbound proxies
>      or transparent proxies.  If necessary, this could be standardized
>      outside of the SIP spec.

Standardizing this as an extension doesn't really help. I'm curious
why transparent proxies would be in the signalling path.  Outbound
proxies should RR (am I missing something here?)

>
>   I got the impression earlier that you were looking for point 2 to be
> in the spec.

That was my intention.

>
>
> >>   Non-SIP URLs in the To/From don't matter because Contact is now
> >> mandatory pretty much everywhere.  You don't need to look at the
> >> From to route messages.
> >
> > But the To/From is there for a reason. At a transaction level, I am
> > all for treating them as opaque strings. Higher up in the application
> > strata, these should have some meaning besides just strings -- at
> > least there should be the possibility for these to have meaning.
>
>   Totally, but a client should know to convert a tel URI to a SIP URI
> like phonenumber@gateway.  I thought that was the point of them, not to
> use them as a Request-URI.

In this specific case, it may or may not be possible for the UAC to
translate the
tel: URI into a SIP URI. (How does the client know which "domain" owns a
specific
number?) This is where an outbound proxy may play a role. In any case, it
may
be useful in certain cases to use a Request-URI which is not directly
routable.
A proxy downstream can translate this into a Request-URI which is routable.
I see no good reason to make an arbitrary limitation of the RR to SIP URLs.
Is there an intractable problem here that I am missing? It is Monday after
all :)

> --
> Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

--
Sean Olson <sean.olson@ericsson.com>


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From: Billy Biggs <Billy_Biggs@3com.com>
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Date: Mon, 27 Nov 2000 11:28:50 -0600

Sean Olson (sean.olson@ericsson.com):

>>   Sorry, let me be more specific.  Say I use a non-SIP URI in the RR
>> (or Contact) of:
>>
>>   <tel:911>;raddr=udp/22.33.44.55:5060
>>
>>   My only choice for request-URI is "tel:911", but this does not
>> convey the destination of the request.  Since you now can't route
>> based on request-URI, you can't use an outbound proxy, stuff breaks
>> in general.
> 
> Why doesn't the outbound proxy RR?

  It does, but what if you call me directly and not through my outbound
proxy?  The INVITE transaction completes without my outbound proxy being
placed in the Route.  When I send a BYE or whatever, it will go to the
outbound proxy, and now your URI loses its significance.

  That's very brittle.

>>   2. Should we standardize an raddr RR parameter and allow this to
>>      be used for SIP URIs?
>>
>>      I say no.  In general this technique breaks with outbound
>>      proxies or transparent proxies.  If necessary, this could be
>>      standardized outside of the SIP spec.
> 
> Standardizing this as an extension doesn't really help. I'm curious
> why transparent proxies would be in the signalling path.  Outbound
> proxies should RR (am I missing something here?)

  Transparent proxies are useful when you want to divert SIP requests
going out to the internet to go to a special firewall proxy.  This is
common with HTTP.

  SIP and HTTP are elegant in their proxy-awareness.  Messages can be
forwarded between proxies without information loss.  However, with an
raddr or whatever, now the IP layer has significance to the meaning of
the message, which breaks the model.

>>   [...] but a client should know to convert a tel URI to a SIP URI
>> like phonenumber@gateway.  I thought that was the point of them,
>> not to use them as a Request-URI.
> 
> In this specific case, it may or may not be possible for the UAC to
> translate the tel: URI into a SIP URI. (How does the client know which
> "domain" owns a specific number?) This is where an outbound proxy may
> play a role. In any case, it may be useful in certain cases to use a
> Request-URI which is not directly routable.  A proxy downstream can
> translate this into a Request-URI which is routable.  [...]

  But now the client and the outbound proxy are together acting as a
single logical client.  This technique, translating a tel: URI into a
SIP URI, can only be done at the edges, and does not work unless you are
sure you are not going to hit some transparent proxy in the middle.

  My point is that URI schemes other than SIP are useful for internal
purposes.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Jonathan and Henning,

   I think this is a great shot at tunneling SIP and RTP through Firewalls,
though you may want to add the following caveats to section 6:

    1) Wrt corporate tunneling, high-bandwidth voice traffic has a distinct
network signature vs. (relatively) low-bandwidth Web traffic. If you blast
voice traffic, encrypted or unencrypted over Web ports, it's going to stick out
like a sore thumb even using the crudest network monitoring tools. ISS and the
other network intrusion detection packages may even flag it as a Netbus,
BackOrifice, or  Distributed Denial of Service attack signature. Most network
administrators will shut the originating host/port down immediately and ask
questions later.

    2) The Internet backbone itself has been heavily optimized (some might say
almost too optimized) for port 80 traffic, and is moving in that direction (as
much as is feasible) for port 443/HTTPS. Increasingly, ISPs are using Web
caching engines to limit this traffic across their backbones. Case in point--I
used Checkpoint's encrypted VPN product in a previous company to connect an
overseas location to our site (encrypted payloads, port numbers stay the same).
Our remote site wanted to access our internal Web server. Unfortunately, one of
the intermediate ISPs was using a Web caching product, which resulted in the
caching engine intercepting, then attempting to originate an (unauthorized)
port 80 access to our internal Web server on the remote end's behalf, with the
expected (non)result. The answer from the ISP was, of course, "well it works
fine for all our normal customers--EOM". I think you'd see some very
interesting  side effects if you applied a tunneled RTP stream to this or other
kinds of caching servers on port 80.
Granted, 443/SSL-caching is a much more difficult problem, equating to a
man-in-the- middle attack--maybe even unsolvable, but given the monetary
inceptive is huge to come up with a solution here too, partial or otherwise....



Howard Hart
ipDialog, Inc.

Jonathan Rosenberg wrote:

> I wanted to draw attention to the following draft which we have submitted.
> It tackles, and provides what we believe to be at least a workable solution,
> to the problem of SIP traversal through enterprise firewalls and NATs when
> there is no ALG at all associated with, or inside of, the firewall.
>
> Thanks,
> Jonathan R.
>
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
>
> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> Sent: Wednesday, November 22, 2000 6:13 AM
> To: IETF-Announce
> Cc: sip@lists.bell-labs.com
> Subject: I-D ACTION:draft-rosenberg-sip-entfw-00.txt
>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>
>         Title           : SIP Traversal through Residential and Enterprise
> NATs
>                           and Firewalls
>         Author(s)       : J. Rosenberg, H. Schulzrinne
>         Filename        : draft-rosenberg-sip-entfw-00.txt
>         Pages           : 15
>         Date            : 21-Nov-00
>
> In this draft, we discuss how SIP can traverse enterprise and
> residential firewalls and NATs. This environment is challenging
> because we assume here that the end user has little or no control
> over the firewall or NAT, and that the firewall or NAT is completely
> ignorant of SIP
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-rosenberg-sip-entfw-00.txt
>
> Internet-Drafts are also available by anonymous FTP. Login with the username
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>
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>
> Internet-Drafts can also be obtained by e-mail.
>
> Send a message to:
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>
> NOTE:   The mail server at ietf.org can return the document in
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> Subject:
> Date: Wed, 22 Nov 2000 14:05:37 -0500
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 Hello Everyone:

   I would like initiate some type of action in this WG on
 addressing security for SIP.  Everyone knows that its an
 important topic, but so far little has been done.  I have
 written a short and very high level problem statement to try
 and provide a start for these discussions.


 End-To-End Security For SIP

    SIP messages can/might traverse N proxies between the sender and the
 destination.  Each proxy along this path has "promised" to provide some
 service to messages it receives (We will address this idea of a promise
 in the lower paragraph).  However, in order to provide this service,
 the proxy might need to modify/access the SIP payload.  So while we the
 sender want to provide enough information to the proxy so that they can
 provide this service, we also do not want to expose our message in
entirety.
 How do we do this ?  The SIP draft has mentioned using underlying security
 mechanisms, but the adaptations of these are not discussed in enough
detail.
 For example, IPSec does not know how to operate on partial payloads.

 Proxy Services

    SIP proxies essentially provide a service to messages they receive.
 However, how do we know that these proxies are actually trusted ?  Or
 better yet, how do we know that a specific proxy is trusted to provide
 a certain service ?  The common answer would be to integrate PKI into the
 SIP infrastructure, but would that be deployable ?


 Plan ?

    Security is an issue which requires detailed addressing before
 mass acceptance.  Additional requirements would be compatibility
 with existing infrastructure, latency, costs, and possibly CALEA.  I would
 like to propose that at the next IETF, this body start addressing security
 seriously.  A trust model would be an ideal starting point.



 Cheers,
 - Tom Tang


Tom Tang
Systems Software Architect
IP Unity
1575 McCandless Drive
Milpitas, CA 95035
(408) 957-0800 x154 (w)
(408) 957-0823 (f)


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Date: Mon, 27 Nov 2000 13:41:37 -0600

Hi

1) Bis 4.2.1 says:

"A UAC MUST be prepared to receive media data according to the session description as soon as it sends an INVITE (or re-INVITE) "

2) According the 183 draft that there must be first a two way path established (or at least the UAC should get first the UAS RTP information) before the UAC is able to get Alerting/call treatment RTP packets.

I guess (2) overrides (1) (and is even included in 1 now)

My question is: Why is the need to have the two way path established for being able to convey the RTP information from the UAS to the UAC?

Thanks
Uri








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Date: Mon, 27 Nov 2000 15:26:47 -0500


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Chris,

Thanks for the prompt response. Not to drag this on for ever, but I have
a few more suggestions  in followup to your comments/clarifications  (in
blue below). As  you are in Ireland, your missiles cannot reach me so
I can be bold :)



Regards

Ranga.


A design issue not discussed below:

Whenever there is more than one header to be returned (via  headers for
example), you return an array.  An array is easy to walk through.
However,  I would think it more useful to return a collection of some
sort that supports additional methods such as checking if an address
exists in the list of headers returned etc. That might be of some use to
an implementation.  If you define the returned type as an array of
headers, there is no way to extend the class to support additional
functionality.  May I suggest adding a collection class interface for
collections of SIP Headers (and that can be specialized by
implementations to support additional query methods etc.), say something
like SIPHeaderList


Chris Harris wrote:

> Ranga,
>
> My comments are in red...
>
> Chris
>
>
> "M. Ranganathan" wrote:
>
>> Chris,
>>
>> Thanks for the prompt response.  Hope you had a great Thanksgiving.
>
> I wish there was such a thing as Thanksgiving in Ireland :(
>
>>
>>
>> My followup is below. ( in blue)
>>
>> Regards
>>
>> Ranga.
>>
>> Chris Harris wrote:
>>
>> > Ranga,
>> >
>> > See comments below.
>> >
>> > Regards,
>> > Chris
>> >
>> > "M. Ranganathan" wrote:
>> >
>> >>
>> >>  Chris, Erik,
>> >>
>> >>  While we are on the subject of Exceptions, why do null arguments
>> >>  throw runtime exceptions when you do a get on them? For example,
>> >>  in the Accept interface, getContentSubtype throws a
>> >>  IllegalArgumentException if  is null. IMHO, I think it is valid
>> >>  to have a null subtype so why not just return null rather than
>> >>  throw exception here?
>> >
>> > By a "doing a get on a null argument" here I assume you mean
>> > attempting to get a non-existent optional field? If you do this a
>> > checked ParameterNotSetException is thrown, not a runtime
>> > exception.
>>
>>
>> Yes. Thats what I had meant.  The word "argument" should have been
>> "field".   Why do you need a  ParameterNotSet exception in this
>> case?  Are you trying to handle the case of basic types like
>> integers, floats and so on ? For objects, null is synonymous with
>> not set isnt it?
>
> We discussed this matter in depth at the last JAIN SIP meeting in
> October, and we came to the conclusion that getting and setting null
> values is not a good idea for several reasons e.g. it is easier to
> debug a ParameterNotSetException than searching for the source of a
> later NullPointerException, and it forces the application to allow for
> the fact that there is a reasonable chance that a particular field
> does not exist. [Of course, if we did allow the use of null, we could
> cut down on the number of methods and drop the
> ParameterNotSetException - but  is it really worth sacrificing clarity
> and robustness for the sake of cutting down on API size?]

Well, I guess that is a judgement call.  With this design, you may have
code that is cluttered with a lot of exception handlers and the API is
quite big.  Checking for null is fairly standard  (am I giving my age
away ? :)  and I can see at least a few places in the base
JAVA API where null is returned to denote empty  and anyway, the app has
a good old stack trace to rely on to see the source of the exception
when null pointers are encountered.   Guarding an application from null
pointer exception is pretty hard anyway so why complicate the API to
handle this one case?  ( For collections of SIPHeaders, you can check
for empty collection and you will not have this problem. )



>
>
>>
>>
>> >
>> >
>> > There are two approaches to handling optional header parts. One
>> > approach is to allow the passing and returning of null objects from
>> > get and set methods (implicit "remove" and "has"), and the other is
>> > to explicitly add has and remove methods (and exceptions on the set
>> > and get methods). As for mandatory header parts - I think passing
>> > null to a set method should definitely throw an
>> > IllegalArgumentException.
>>
>> I have the following questions about setXXX methods:
>>
>> 1.  Why do you restrict throwing IllegalArgumentException to only
>> null arguments in some of the headers? Why are you not checking for
>> proper formatting of the passed arguments (by giving it to the
>> parser for example and running the appropriate  parser rule).
>
> In general, an IllegalArgumentException is thrown if an argument to a
> set method fails basic validation (i.e. no implementation-specific
> parsing is used). Note that an application can ensure that this
> exception is not thrown by following the rules defined in the API . So
> if an argument is a String, I throw an IllegalArgumentException if the
> argument is null or zero-length.
>
> Of course this is not sufficient validation and the string must be
> parsed to check for whitespaces, newlines etc, and different
> implementations will be more tolerant than others. This is why there
> is a checked JainSipParseException - an application cannot tell in
> advance if a particular implementation will accept a certain value, so
> it must catch this exception.
>
> I am proposing that this exception should also be thrown by get
> methods (for all fields), because if an implementation uses lazy
> parsing, then there is no guarantee that the header value will parse
> correctly upon the invocation of the get method, and again this is
> implementation-dependent.
>
>>
>>
>> 2.  A more general question - why do you have set methods at all in
>> the interfaces?  The class that implements the interface can have
>> set methods but the interface itself does not need to have them.
>
> I have set methods in the interfaces because the objects will not only
> be coming up from the implementation - an application needs to be able
> to create and send requests, and this means being able to set the
> values.

Ah! My understanding was correct then.

OK. Perhaps this would be too much of a change for you to consider but
let me make a suggestion anyway ....

I think formatting headers should be considered under a different set of
API.  Here is what I am suggesting

1. Get rid of the setXXX interfaces (the parser will have direct access
to the classes and does not need them and it will cut down the size of
the API).

2. Have a  SIPMessageFormatter interface who'se sole job it is to format
output messages.  This can have methods to format the different outgoing
messages and will throw parse exceptions if the strings that it gets are
incorrect. The MessageFormatter interface takes as parameter values only
Strings.

Concretely,  for example:

public interface SIPMessageFormatter {
        public String createFromMessage( String from ,  String tag)
throws SIPParseException;

    .....................

}

It would be slower for an implementation to extract the same information
out of a structure and hand it to a parser for validation (IMHO). A
"loose implementation" may not want to be bothered with validation at
all and leave it to the reciever to reject the mesage, thereby making
the implementation faster or it may do very cursory checking.






>
>
>>  Moreover, the parser has the job of setting up stuff in the
>> returned object. It has a set of BNF rules to decide on the
>> correctness of the arguments and it does not need for the class to
>> re-implement the check. For example, I think it ought to be the
>> parser's job to check if the q value is between 0 and 1.   (Is the
>> purpose of the set method to allow for the application to do header
>> formatting?)
>
> I absolutely agree that the parser should perform validation (hence
> the JainSipParseException) - but as I mentioned above there are
> certain basic criteria that must be met by all implementations and
> applications - I am using the IllegalArgumentException to cover these.
> Perhaps certain cases go too far in this validation - perhaps the
> IllegalArgumentException could be thrown only if a null argument is
> passed and all other validation is left up to the implementation's
> parser? (although a problem with certain cases such as PriorityHeader
> comes to mind, where the header values are defined as constants in the
> API and an IllegalArgumentException is thrown if any other value is
> passed - should we allow any non-null String values for the priority?
> Or to take your example of q-value above - should we just allow any
> non-null Float and let the parser enforce the 0-1 range?

I think I understand the point about checking for null.  I think, if you
are dead set on adopting the set method as a way of constructing
headers, one should validate arguments via the parser and throw
ParseException for anything other than null arguments. On the other
hand, I propose something entirely different for constructing headers
(above).

>
>
>
>>
>>
>>
>> >
>> >
>> > (If it is considered acceptable to have a "null" subType then we
>> > should make it optional)
>> >
>> >>
>> >>
>> >>  I don't quite follow the argument that lazy parsing
>> >>  "necessitates throwing a parse exception for all get methods".
>> >>  Here is what I had envisioned:
>> >>
>> >>
>> >>  1 When a message comes in it will be separated into headers based
>> >>  on type.
>> >>  2 The header would be returned with no field initialized except
>> >>  the text corresponding to the message. (this would necessitate
>> >>  examining the first few bytes of the message to determine the
>> >>  header type but the rest remains unparsed.) You can also stow
>> >>  away the type information in the Header that is returned as well
>> >>  to know which parser rule to invoke later.
>> >>  3. Later, when the application wants to parse the header, it has
>> >>  the string corresponding to the message and can hand it off to
>> >>  the parser, which can then fill in the appropriate fields
>> >>  corresponding to the header.
>> >>  4. Subsequently, the application can access the fields just as an
>> >>  eager parser based application would do.
>> >>
>> >>  ( Where am I missing the boat with this? )
>> >
>> > An application has no idea whether an implementation is using lazy
>> > or eager parsing - it is simply passed a header object - and then
>> > if it is interested in a particular field it calls the appropriate
>> > get method (which would internally invoke the parser in a lazy
>> > parsing implementation). So the application is completely oblivious
>> > to whether or not this "optimisation" is used by the
>> > implementation.
>>
>>
>> Ah! ( Why is this hiding  of the implementation so important? )  Can
>> this be handled in the following fashion by an implementation:
>>
>> 1. The implementation keeps a flag in the  class that implements the
>> interface whether the parsing has been done or not (i.e. boolean
>> parsingDone flag).
>>
>> 2. The Implementation keeps the entire string corresponding to the
>> header in the object. So the parser has to peek into the message
>> header to know what type the message is as soon as it has the
>> header string and then grabs the string and stores it away for lazy
>> parsing.
>>
>>
>> 3. The get Method  checks if the flag has been set, calls the parser
>> to do the parse if the flag has not been set and then returns a
>> parseException if the parse fails. This can be done transparently to
>> the caller.
>
> This is exactly what I would expect a lazy-parsing implementation to
> do - the fact that lazy-parsing is hidden from the application (except
> that the application must catch a JainSipParseException on the get
> methods to allow for the possibility that an implementation uses
> lazy-parsing - so technically the application will know that an
> implementation uses lazy-parsing if this exception is thrown by a get
> method - but I think the two cases are handled nicely by the addition
> of the exception)

>>
>>
>>
>>
>>
>>
>> >
>> >
>> > If the field is optional and does not exist in the header a
>> > ParameterNotSetException will be thrown by the implementation.
>>
>> A suggestion (shoot it down if it does not jive):
>>
>> 1. All fields are represented by objects (not simple types like int,
>> use Integer instead).
>
> I buy this part - then the IllegalArgumentException is only ever
> thrown if an object parameter is null, and the JainSipParseException
> is thrown for all other problems with the argument.

OK this jives and is easily defined.

>
>
>>
>> 2. If the field is not set just return null
>
> See above for null discussion,
>
>>
>>
>>
>>
>> > If the field is mandatory and does not exist, a
>> > JainSipParseException will be thrown (header is malformed). If any
>> > header field is malformed a JainSipParseException is thrown. If a
>> > JainSipParseException is thrown, this indicates that the
>> > application must use the header's getValue method to get a string
>> > representation of the header value and do what it can with that.
>>
>>
>> Why the headers getValue ? Why is the string not part of the
>> Exception?
>
> Sure - we can say that the JainSipParseException's message must be the
> string value of the header.

OK Glad you buy that one.


>

>
>
>>
>>
>>
>> >
>> >
>> >>
>> >>
>> >>  It appears that what is being attempted here (please correct me
>> >>  if I am wrong) is to use exceptions as a mechanism to catch
>> >>  un-initialized fields so the parser can be invoked in the
>> >>  exception handler perhaps. This leads to a lot of overhead in
>> >>  clunky code. I think it is a mistake to throw exceptions on every
>> >>  get to try to protect the application from inadvertently
>> >>  accessing fields that have not been initialized.  Why not just
>> >>  assume that the application must be smart enough to parse the
>> >>  header text before accessing fields and not try to provide a
>> >>  complex mechanism as above?
>> >
>> >
>> > We are using ParameterNotSetException for the case of an optional
>> > header not existing, and a JainSipParseException for the case of a
>> > malformed header being received from the network.
>> >
>> > If we assume the application is smart enough to do all header value
>> > parsing, we reduce a header to just being two strings - the type
>> > and value, which puts the burden of parsing on an application -
>> > this work can be done by the underlying implementation - doesn't it
>> > make more sense for an application to catch an exception than have
>> > to parse all header value strings (especially since this
>> > functionality already exists in the underlying stack)
>>
>>
>> No that is not what I had intended. Parser functionality exists in
>> the stack but has to be called explicitly by the implementation. Too
>> much hiding of the implementation from the application has its own
>> set of disadvantages (IMHO).
>
> I wanted parser fuctionality to be invoked implicitly by an
> application - from the API perspective it is an implementation detail
> - so the application should not be aware of how or when the underlying
> parser is invoked - it should just know that it is there in some form,
> and that if it tries to get or set a value that the implementation
> doesn't like it will get a JainSipParseException thrown back at it.
> This is important if we want to maintain the three core values of JAIN
> - portability, portability and portabilty ;)

The point about portability is a good one. Of course the important thing
about portability is to be able to maintain the same semantics of
exceptions as well.  For an eager parser,  there is no possibility of
throwing a parse  exception  on a get (i.e. parse exception can never
occur on a getXXX) so there will be some curious code in a getXXX
implementation  for an eager parser that has to throw an exception even
when it does not need to (just to keep the compiler happy).   Second,
the application has to be willing to catch such an exception to be able
to function both with a lazy and an eager implementation in a
transparent fashion and it may do different things when the
implementation is lazy, which exposes the underlying implementation
anyway as you have noted.

Explicit control by the application may give the application finer
grained control and eliminate some awkward exception handling that is
not meaningful for an eager parser .  This way, you can also have mixed
mode parsing, allowing the application to decide what headers it wants
to handle in a lazy fashion and which ones it wants to handle in an
eager fashion.

Another suggestion ( may be a bit radical) - would it make sense to have
extension interfaces (with enhanced getXXX  interfaces)  that are only
implemented by lazy parsers and make lazy parsing optional (so only
highly performance sensitive apps will  avail themselves of such
interfaces whereas most applications will use the eager parser
interfaces for getXXX).  ( May or may not make sense, I thought I would
mention it as it occured to me.  )

Anyway,  there is more ways than one to skin a cat. Just keep everything
as simple as possible but no simpler. I am sure you are eager to bring
this to closure so I will not belabor the point about eager parsing
versus lazy parsing  although, as a final point I would just like to
state that portability can be handled whether or not the parse is
explicitly or implicitly lazy. Enough said on this point.




Regards,

Ranga.





--
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
Tel: 301 975 3664 Fax: 301 590 0932



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<html>
<body text="#000000" bgcolor="#FFFFFF" link="#0000FF" vlink="#FF0000" alink="#000088">
Chris,
<p>Thanks for the prompt response. Not to drag this on for ever, but I&nbsp;have
a few more suggestions&nbsp; in followup to your comments/clarifications&nbsp;
(in blue below). As&nbsp; <font color="#000000">you are in Ireland, your
missiles cannot reach me so I&nbsp;can be bold :)</font>
<br><font color="#3333FF"></font>&nbsp;
<br>&nbsp;
<p>Regards
<p>Ranga.
<br>&nbsp;
<p><font color="#3333FF">A&nbsp;design issue not discussed below:</font><font color="#3333FF"></font>
<p><font color="#3333FF">Whenever there is more than one header to be returned
(via&nbsp; headers for example), you return an array.&nbsp; An array is
easy to walk through. However,&nbsp; I would think it more useful to return
a collection of some sort that supports additional methods such as checking
if an address exists in the list of headers returned etc. That might be
of some use to an implementation.&nbsp; If you define the returned type
as an array of headers, there is no way to extend the class to support
additional functionality.&nbsp; May I&nbsp;suggest adding a collection
class interface for collections of SIP&nbsp;Headers (and that can be specialized
by implementations to support additional query methods etc.), say something
like <i>SIPHeaderList</i></font>
<br><font color="#3333FF"></font>&nbsp;<font color="#3333FF"></font>
<p>Chris Harris wrote:
<blockquote TYPE=CITE><font color="#FF0000">Ranga,</font>
<p><font color="#FF0000">My comments are in red...</font>
<p><font color="#FF0000">Chris</font>
<br>&nbsp;
<p>"M. Ranganathan" wrote:
<blockquote TYPE=CITE>Chris,
<p>Thanks for the prompt response.&nbsp; Hope you had a great Thanksgiving.</blockquote>
<font color="#FF0000">I wish there was such a thing as Thanksgiving in
Ireland :(</font>
<blockquote TYPE=CITE>&nbsp;
<p>My followup is below. ( in blue)
<p>Regards
<p>Ranga.
<p>Chris Harris wrote:
<blockquote TYPE=CITE>Ranga,
<p>See comments below.
<p>Regards,
<br>Chris
<p>"M. Ranganathan" wrote:
<blockquote TYPE=CITE>&nbsp;
<br>Chris, Erik,
<p>While we are on the subject of Exceptions, why do null arguments throw
runtime exceptions when you do a get on them? For example, in the Accept
interface, getContentSubtype throws a IllegalArgumentException if&nbsp;
is null. IMHO, I think it is valid to have a null subtype so why not just
return null rather than throw exception here?</blockquote>
By a "doing a get on a null argument" here I assume you mean attempting
to get a non-existent optional field? If you do this a checked ParameterNotSetException
is thrown, not a runtime exception.</blockquote>

<p><br><font color="#3333FF">Yes. Thats what I had meant.&nbsp; The word
"argument" should have been "field".&nbsp;&nbsp; Why do you need a&nbsp;
ParameterNotSet exception in this case?&nbsp; Are you trying to handle
the case of basic types like integers, floats and so on ? For objects,
null is synonymous with not set isnt it?</font></blockquote>
<font color="#FF0000">We discussed this matter in depth at the last JAIN
SIP meeting in October, and we came to the conclusion that getting and
setting null values is not a good idea for several reasons e.g. it is easier
to debug a ParameterNotSetException than searching for the source of a
later NullPointerException, and it forces the application to allow for
the fact that there is a reasonable chance that a particular field does
not exist. [Of course, if we did allow the use of null, we could cut down
on the number of methods and drop the ParameterNotSetException - but&nbsp;
is it really worth sacrificing clarity and robustness for the sake of cutting
down on API size?]</font></blockquote>

<p><br><font color="#3333FF">Well, I&nbsp;guess that is a judgement call.&nbsp;
With this design, you may have code that is cluttered with a lot of exception
handlers and the API&nbsp;is quite big.&nbsp; Checking for null is fairly
standard&nbsp; (am I&nbsp;giving my age away ? :)&nbsp; and I&nbsp;can
see at least a few places in the base JAVA&nbsp;API&nbsp;where null is
returned to denote empty&nbsp; and anyway, the app has a good old stack
trace to rely on to see the source of the exception when null pointers
are encountered.&nbsp;&nbsp; Guarding an application from null pointer
exception is pretty hard anyway so why complicate the API&nbsp;to handle
this one case?&nbsp; ( For collections of SIPHeaders, you can check for
empty collection and you will not have this problem. )</font>
<br>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE><font color="#FF0000"></font>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<p>There are two approaches to handling optional header parts. One approach
is to allow the passing and returning of null objects from get and set
methods (implicit "remove" and "has"), and the other is to explicitly add
has and remove methods (and exceptions on the set and get methods). As
for mandatory header parts - I think passing null to a set method should
definitely throw an IllegalArgumentException.</blockquote>
<font color="#3333FF">I have the following questions about setXXX methods:</font>
<p><font color="#3333FF">1.&nbsp; Why do you restrict throwing IllegalArgumentException
to only null arguments in some of the headers? Why are you not checking
for proper formatting of the passed arguments (by giving it to the parser
for example and running the appropriate&nbsp; parser rule).</font></blockquote>
<font color="#FF0000">In general, an IllegalArgumentException is thrown
if an argument to a set method fails basic validation (i.e. no implementation-specific
parsing is used). Note that an application can ensure that this exception
is not thrown by following the rules defined in the API . So if an argument
is a String, I throw an IllegalArgumentException if the argument is null
or zero-length.</font>
<p><font color="#FF0000">Of course this is not sufficient validation and
the string must be parsed to check for whitespaces, newlines etc, and different
implementations will be more tolerant than others. This is why there is
a checked JainSipParseException - an application cannot tell in advance
if a particular implementation will accept a certain value, so it must
catch this exception.</font>
<p><font color="#FF0000">I am proposing that this exception should also
be thrown by get methods (for all fields), because if an implementation
uses lazy parsing, then there is no guarantee that the header value will
parse correctly upon the invocation of the get method, and again this is
implementation-dependent.</font>
<blockquote TYPE=CITE>&nbsp;
<p><font color="#3333FF">2.&nbsp; A more general question - why do you
have set methods at all in the interfaces?&nbsp; The class that implements
the interface can have set methods but the interface itself does not need
to have them.</font></blockquote>
<font color="#FF0000">I have set methods in the interfaces because the
objects will not only be coming up from the implementation - an application
needs to be able to create and send requests, and this means being able
to set the values.</font></blockquote>
<font color="#3333FF">Ah! My understanding was correct then.</font><font color="#3333FF"></font>
<p><font color="#3333FF">OK. Perhaps this would be too much of a change
for you to consider but let me make a suggestion anyway ....</font><font color="#3333FF"></font>
<p><font color="#3333FF">I&nbsp;think formatting headers should be considered
under a different set of API.&nbsp; Here is what I&nbsp;am suggesting</font><font color="#3333FF"></font>
<p><font color="#3333FF">1. Get rid of the setXXX interfaces (the parser
will have direct access to the classes and does not need them and it will
cut down the size of the API).</font><font color="#3333FF"></font>
<p><font color="#3333FF">2. Have a&nbsp; SIPMessageFormatter interface
who'se sole job it is to format output messages.&nbsp; This can have methods
to format the different outgoing messages and will throw parse exceptions
if the strings that it gets are incorrect. The MessageFormatter interface
takes as parameter values only Strings.</font><font color="#3333FF"></font>
<p><font color="#3333FF">Concretely,&nbsp; for example:</font><font color="#3333FF"></font>
<p><font color="#3333FF">public interface SIPMessageFormatter {</font>
<br><font color="#3333FF">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; public
String createFromMessage( String from ,&nbsp; String tag) throws SIPParseException;</font><font color="#3333FF"></font>
<p><font color="#3333FF">&nbsp;&nbsp;&nbsp; .....................</font><font color="#3333FF"></font>
<p><font color="#3333FF">}</font><font color="#3333FF"></font>
<p><font color="#3333FF">It would be slower for an implementation to extract
the same information out of a structure and hand it to a parser for validation
(IMHO). A "loose implementation" may not want to be bothered with validation
at all and leave it to the reciever to reject the mesage, thereby making
the implementation faster or it may do very cursory checking.</font>
<br><font color="#3333FF"></font>&nbsp;
<br><font color="#3333FF"></font>&nbsp;
<br><font color="#3333FF"></font>&nbsp;
<br><font color="#3333FF"></font>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE><font color="#FF0000"></font>&nbsp;
<blockquote TYPE=CITE><font color="#3333FF">&nbsp;Moreover, the parser
has the job of setting up stuff in the returned object. It has a set of
BNF rules to decide on the correctness of the arguments and it does not
need for the class to re-implement the check. For example, I think it ought
to be the parser's job to check if the q value is between 0 and 1.&nbsp;&nbsp;
(Is the purpose of the set method to allow for the application to do header
formatting?)</font></blockquote>
<font color="#FF0000">I absolutely agree that the parser should perform
validation (hence the JainSipParseException) - but as I mentioned above
there are certain basic criteria that must be met by all implementations
and applications - I am using the IllegalArgumentException to cover these.
Perhaps certain cases go too far in this validation - perhaps the IllegalArgumentException
could be thrown only if a null argument is passed and all other validation
is left up to the implementation's parser? (although a problem with certain
cases such as PriorityHeader comes to mind, where the header values are
defined as constants in the API and an IllegalArgumentException is thrown
if any other value is passed - should we allow any non-null String values
for the priority? Or to take your example of q-value above - should we
just allow any non-null Float and let the parser enforce the 0-1 range?</font></blockquote>

<p><br><font color="#3333FF">I&nbsp;think I&nbsp;understand the point about
checking for null.&nbsp; I&nbsp;think, if you are dead set on adopting
the set method as a way of constructing headers, one should validate arguments
via the parser and throw ParseException for anything other than null arguments.
On the other hand, I propose something entirely different for constructing
headers (above).</font>
<blockquote TYPE=CITE><font color="#FF0000"></font>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<p>(If it is considered acceptable to have a "null" subType then we should
make it optional)
<blockquote TYPE=CITE>&nbsp;
<p>I don't quite follow the argument that lazy parsing&nbsp; "necessitates
throwing a parse exception for all get methods".&nbsp;&nbsp; Here is what
I had envisioned:
<br>&nbsp;
<p>1 When a message comes in it will be separated into headers based on
type.
<br>2 The header would be returned with no field initialized except the
text corresponding to the message. (this would necessitate examining the
first few bytes of the message to determine the header type but the rest
remains unparsed.) You can also stow away the type information in the Header
that is returned as well to know which parser rule to invoke later.
<br>3. Later, when the application wants to parse the header, it has the
string corresponding to the message and can hand it off to the parser,
which can then fill in the appropriate fields corresponding to the header.
<br>4. Subsequently, the application can access the fields just as an eager
parser based application would do.
<p>( Where am I missing the boat with this? )</blockquote>
An application has no idea whether an implementation is using lazy or eager
parsing - it is simply passed a header object - and then if it is interested
in a particular field it calls the appropriate get method (which would
internally invoke the parser in a lazy parsing implementation). So the
application is completely oblivious to whether or not this "optimisation"
is used by the implementation.</blockquote>

<p><br>Ah! ( Why is this hiding&nbsp; of the implementation so important?
)&nbsp; Can this be handled in the following fashion by an implementation:
<p>1. The implementation keeps a flag in the&nbsp; class that implements
the interface whether the parsing has been done or not (i.e. boolean parsingDone
flag).
<p>2. The Implementation keeps the entire string corresponding to the header
in the object. So the parser has to peek into the message header to know
what type the message is as soon as it has the&nbsp; header string and
then grabs the string and stores it away for lazy parsing.
<br>&nbsp;
<p>3. The get Method&nbsp; checks if the flag has been set, calls the parser
to do the parse if the flag has not been set and then returns a parseException
if the parse fails. This can be done transparently to the caller.</blockquote>
<font color="#FF0000">This is exactly what I would expect a lazy-parsing
implementation to do - the fact that lazy-parsing is hidden from the application
(except that the application must catch a JainSipParseException on the
get methods to allow for the possibility that an implementation uses lazy-parsing
- so technically the application will know that an implementation uses
lazy-parsing if this exception is thrown by a get method - but I think
the two cases are handled nicely by the addition of the exception)</font></blockquote>

<blockquote TYPE=CITE>
<blockquote TYPE=CITE>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<p>If the field is optional and does not exist in the header a ParameterNotSetException
will be thrown by the implementation.</blockquote>
A suggestion (shoot it down if it does not jive):
<p>1. All fields are represented by objects (not simple types like int,
use Integer instead).</blockquote>
<font color="#FF0000">I buy this part - then the IllegalArgumentException
is only ever thrown if an object parameter is null, and the JainSipParseException
is thrown for all other problems with the argument.</font></blockquote>
<font color="#3333FF"></font>
<p><br><font color="#3333FF">OK this jives and is easily defined.</font>
<blockquote TYPE=CITE><font color="#FF0000"></font>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<br>2. If the field is not set just return null</blockquote>
<font color="#FF0000">See above for null discussion,</font>
<blockquote TYPE=CITE>&nbsp;
<br>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>If the field is mandatory and does not exist, a JainSipParseException
will be thrown (header is malformed). If any header field is malformed
a JainSipParseException is thrown. If a JainSipParseException is thrown,
this indicates that the application must use the header's getValue method
to get a string representation of the header value and do what it can with
that.</blockquote>

<p><br>Why the headers getValue ? Why is the string not part of the Exception?</blockquote>
<font color="#FF0000">Sure - we can say that the JainSipParseException's
message must be the string value of the header.</font></blockquote>
<font color="#3333FF"></font>
<p><br><font color="#3333FF">OK Glad you buy that one.</font>
<br><font color="#000099"></font>&nbsp;
<blockquote TYPE=CITE><font color="#FF0000"></font>&nbsp;</blockquote>

<blockquote TYPE=CITE><font color="#FF0000"></font>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<p>It appears that what is being attempted here (please correct me if I
am wrong) is to use exceptions as a mechanism to catch un-initialized fields
so the parser can be invoked in the exception handler perhaps. This leads
to a lot of overhead in clunky code. I think it is a mistake to throw exceptions
on every get to try to protect the application from inadvertently accessing
fields that have not been initialized.&nbsp; Why not just assume that the
application must be smart enough to parse the header text before accessing
fields and not try to provide a complex mechanism as above?</blockquote>

<p><br>We are using ParameterNotSetException for the case of an optional
header not existing, and a JainSipParseException for the case of a malformed
header being received from the network.
<p>If we assume the application is smart enough to do all header value
parsing, we reduce a header to just being two strings - the type and value,
which puts the burden of parsing on an application - this work can be done
by the underlying implementation - doesn't it make more sense for an application
to catch an exception than have to parse all header value strings (especially
since this functionality already exists in the underlying stack)</blockquote>

<p><br><font color="#3333FF">No that is not what I had intended. Parser
functionality exists in the stack but has to be called explicitly by the
implementation. Too much hiding of the implementation from the application
has its own set of disadvantages (IMHO).</font></blockquote>
<font color="#FF0000">I wanted parser fuctionality to be invoked <i>implicitly</i>
by an application - from the API perspective it is an implementation detail
- so the application should not be aware of how or when the underlying
parser is invoked - it should just know that it is there in some form,
and that if it tries to get or set a value that the implementation doesn't
like it will get a JainSipParseException thrown back at it. This is important
if we want to maintain the three core values of JAIN - portability, portability
and portabilty ;)</font></blockquote>

<p><br><font color="#3333FF">The point about portability is a good one.
Of course the important thing about portability is to be able to maintain
the same semantics of exceptions as well.&nbsp; For an eager parser,&nbsp;
there is no possibility of throwing a parse&nbsp; exception&nbsp; on a
get (i.e. parse exception can never occur on a getXXX) so there will be
some curious code in a getXXX&nbsp; implementation&nbsp; for an eager parser
that has to throw an exception even when it does not need to (just to keep
the compiler happy).&nbsp;&nbsp; Second,&nbsp; the application has to be
willing to catch such an exception to be able to function both with a lazy
and an eager implementation in a transparent fashion and it may do different
things when the implementation is lazy, which exposes the underlying implementation
anyway as you have noted.&nbsp;&nbsp;</font><font color="#3333FF"></font>
<p><font color="#3333FF">Explicit control by the application may give the
application finer grained control and eliminate some awkward exception
handling that is&nbsp; not meaningful for an eager parser .&nbsp; This
way, you can also have mixed mode parsing, allowing the application to
decide what headers it wants to handle in a lazy fashion and which ones
it wants to handle in an eager fashion.</font><font color="#3333FF"></font>
<p><font color="#3333FF">Another suggestion ( may be a bit radical) - would
it make sense to have extension interfaces (with enhanced getXXX&nbsp;
interfaces)&nbsp; that are only implemented by lazy parsers and make lazy
parsing optional (so only highly performance sensitive apps will&nbsp;
avail themselves of such interfaces whereas most applications will use
the eager parser interfaces for getXXX).&nbsp; (&nbsp;May or may not make
sense, I&nbsp;thought I&nbsp;would mention it as it occured to me.&nbsp;
)</font><font color="#3333FF"></font>
<p><font color="#3333FF">Anyway,&nbsp; there is more ways than one to skin
a cat. Just keep everything as simple as possible but no simpler. I&nbsp;am
sure you are eager to bring this to closure so I&nbsp;will not belabor
the point about eager parsing versus lazy parsing&nbsp; although, as a
final point&nbsp;I would just like to state that portability can be handled
whether or not the parse is explicitly or implicitly lazy. Enough said
on this point.</font>
<br><font color="#3333FF"></font>&nbsp;
<br><font color="#3333FF"></font>&nbsp;
<br><font color="#3333FF"></font>&nbsp;<font color="#3333FF"></font>
<p><font color="#3333FF">Regards,</font><font color="#3333FF"></font>
<p><font color="#3333FF">Ranga.</font>
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<pre>--&nbsp;
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.&nbsp;
Tel: 301 975 3664 Fax: 301 590 0932</pre>
&nbsp;
</body>
</html>

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Voravit:

You can check out www.vovida.org for opensource SDP APIs.

regards
sunitha




"Voravit H." wrote:

> Dear all
>
>         Is there any free SDP API for education purpose out there?
>         I am now trying to implement SIP but I stuck on SDP content.
>
> Mr. Voravit H.
>
>   ------------------------------------------------------------------------
>                   Name: winmail.dat
>    winmail.dat    Type: application/ms-tnef
>               Encoding: base64

--
Sunitha Kumar
http://www.cisco.com



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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Voravit:
<p>You can check out www.vovida.org for opensource SDP APIs.
<p>regards
<br>sunitha
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<p>"Voravit H." wrote:
<blockquote TYPE=CITE>Dear all
<p>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Is there any free SDP API
for education purpose out there?
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I am now trying to implement
SIP but I stuck on SDP content.
<p>Mr. Voravit H.
<p>&nbsp; ------------------------------------------------------------------------
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Name: winmail.dat
<br>&nbsp;&nbsp; winmail.dat&nbsp;&nbsp;&nbsp; Type: application/ms-tnef
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Encoding: base64</blockquote>

<pre>--&nbsp;
Sunitha Kumar
<A HREF="http://www.cisco.com">http://www.cisco.com</A></pre>
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From: Henry Sinnreich <Henry.Sinnreich@wcom.com>
Subject: RE: [SIP] What SIP doesn't tell you how to do.
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We should all thank Jonathan for answering countless questions on this
list - acting as a volunteer help desk for everybody!

Henry

-----Original Message-----
From: sip-admin@lists.bell-labs.com
[mailto:sip-admin@lists.bell-labs.com]On Behalf Of Brian S
Sent: Monday, November 27, 2000 10:30 AM
To: sip@lists.bell-labs.com
Subject: [SIP] What SIP doesn't tell you how to do.


Apparently SIP doesn't tell you how to do any of the
following as well:

1. Be professional.
2. Show others common courtesy.
3. Switch to decaf.

BTW- He never asked how to do anything, just if it
could be done.

-----Original Message-----
From: Jonathan Rosenberg
[mailto:jdrosen@dynamicsoft.com]
Sent: Wednesday, November 22, 2000 10:59 AM
To: 'Sanjiv Deshpande'; sip@lists.bell-labs.com
Subject: RE: [SIP] multiple location


SIP is just a protocol. Thats it. Its a tool. If you
want device A to call
device B, it tells you how to do that. Like all tools,
you can use them in
many different ways. When you buy a hammer in Home
Depot, does it tell you
whether to use your left hand or right? How many
fingers to grip with it?
The size of the nails to use with it? Of course not.

However, for some reason that I cannot fathom, many
people seem to believe
the opposite is true for protocols; that is, protocols
need to insist on how
syntax and semantics AND how they are applied to solve
real world problems.
This approach just results in systems that are useful
to no one. Would you
buy a hammer that said "for nails of length 1 inch
only?" I doubt it.

So, to be specific, SIP doesn't tell you how to do any
of the following:

1. the color of the phones that use SIP
2. the GUI for SIP calls
3. how you build a network of proxy servers
4. whether a phone can make calls while its in a call
5. qualifications for human beings to make use of SIP
softphones (i.e., IQ
levels, height, or what have you).
6. how many simultaneous calls the device can be in

Thanks,
Jonathan R.



---
Jonathan D. Rosenberg                       72 Eagle
Rock Ave.
Chief Scientist                             First
Floor
dynamicsoft                                 East
Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:
(973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE:
(973) 952-5000
http://www.dynamicsoft.com


> -----Original Message-----
> From: Sanjiv Deshpande
[mailto:sdeshpan@telcordia.com]
> Sent: Wednesday, November 22, 2000 11:36 AM
> To: sip@lists.bell-labs.com
> Subject: [SIP] multiple location
>
>
>
>
>
>
> Q:
> INVITE Sent by UAC got forked and reached two UAS1
and UAS2 ,
>  the UAC decided
> to talk to UAS1. Is it possible that UAS2 (Assuming
it is a
> client also) can
> make a call to someone else.
>
>
> What I wanted to know is, A user can be contacted at
multiple
> location if one of
> the location answers a CALL is it possible for
somebody else
> on behalf of that
> user make CALLS or receive CALLS....
>
>
> Thanks
> Sanjiv.
>
>
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>

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Date: Mon, 27 Nov 2000 17:33:18 -0800

Hi dear all,
I was trying to figure out what the 305 response gives additional info. to
the requester, what the 301 or 302 already does not. RFC explains that 305
means the requested resource should be accessed through a proxy, can not I
achieve the same by using 301 or 302?

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From: Billy Biggs <Billy_Biggs@3com.com>
To: "Emami-Nouri, Mohsen" <memami@nuera.com>
Cc: SIP List <sip@lists.bell-labs.com>
Subject: Re: [SIP] 305 response,  what for!
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Date: Mon, 27 Nov 2000 20:10:26 -0600

Emami-Nouri, Mohsen (memami@nuera.com):

> I was trying to figure out what the 305 response gives additional
> info. to the requester, what the 301 or 302 already does not. RFC
> explains that 305 means the requested resource should be accessed
> through a proxy, can not I achieve the same by using 301 or 302?

  The wording implies that if you get a 305 you should send only that
specific request through the proxy.  Future requests within the call-leg
should instead be directed at the old Contact address.

  I have doubts about mid-call redirects actually working at all, and I
suspect that most implementations handle all 3xx responses identically,
except maybe updating numbers when a 301 is received.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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From sip-admin@lists.bell-labs.com  Mon Nov 27 22:04:04 2000
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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
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Subject: RE: [SIP] New Section 6.34 (was: Record-Route consensus)
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Date: Mon, 27 Nov 2000 19:03:18 -0800

Hi guys,

This whole non-SIP Route concept is scaring me <grin>. I whole hearted agree
that when Route is used, the routing relationship is no longer just to the
original destination but rather to the network location of the next Route'd
proxy. [For this reason I am having misgivings about how worthwhile
Jonathan's invariant that the Request-URI in some form represents the final
UA is. If it just represents the direction ok I can live with that.] 

Take the tel case: you are not interested in routing to the tel: of the
final destintion but rather in routing to the next proxy that wanted to
remain in the path - through a globally udp-reachable address. For instance,
UAC->P1->P2->P3->P4->UAS. If P3 Record-Routes with a some URL scheme scheme
only known by adacent proxyies P2 and P4 then the call will fail because
either the UAC or P1 must also be able to route to P3 based on the Route.

So to patch that problem a raddr parameter is suggested - that all UA's and
proxies must understand. However, the raddr must still specify a globally
udp-reachable address for things to work (otherwise you run into the same
problem). So what's the point? Just KISS and stick with SIP URL's. I think
the simpler something is, the more likely it will be to work across
everyone's SIP devices - and this is somethign you want to work.

I would make the same argument for non-SIP Contacts (used for routing SIP
calls not for providing alias/alternate route possibilities). 

IMO Request-URI's are different due to that fact that the Request-URI can be
translated as it traverses a network - so you only have to know whether or
not an adjacent proxy/UA supports a certain (non-SIP) URL scheme. This is
NOT TRUE after the initial INVITE and you also no longer want to do
network-independent routing. 

As an aside, we were discussing earlier that about route parameters and
angle brackets. A couple of peopel agreed that for

Record-Route: <url>;rr-param

A UA or proxy does not need to include rr-param's when buidling the
Request-URI (ie, the Record-Route header parameters). If you want parameters
to be passed into the Request-URI, they should go _inside_ the angle
brackets(ie, Record-Route Request-URI parameters). This follows standard URL
rules and is intuitive given that the Request-URI cannot contain angle
brackets.

I look forward to hearing your point of view on this.

Regards,

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 


> -----Original Message-----
> From: Billy Biggs [mailto:Billy_Biggs@3com.com]
> Sent: Tuesday, November 28, 2000 1:29 AM
> To: Sean Olson
> Cc: Jonathan Rosenberg; Jo Hornsby; hgs@cs.columbia.edu;
> sip@lists.bell-labs.com
> Subject: Re: [SIP] New Section 6.34 (was: Record-Route consensus)
> 
> 
> Sean Olson (sean.olson@ericsson.com):
> 
> >>   Sorry, let me be more specific.  Say I use a non-SIP URI 
> in the RR
> >> (or Contact) of:
> >>
> >>   <tel:911>;raddr=udp/22.33.44.55:5060
> >>
> >>   My only choice for request-URI is "tel:911", but this does not
> >> convey the destination of the request.  Since you now can't route
> >> based on request-URI, you can't use an outbound proxy, stuff breaks
> >> in general.
> > 
> > Why doesn't the outbound proxy RR?
> 
>   It does, but what if you call me directly and not through 
> my outbound
> proxy?  The INVITE transaction completes without my outbound 
> proxy being
> placed in the Route.  When I send a BYE or whatever, it will go to the
> outbound proxy, and now your URI loses its significance.
> 
>   That's very brittle.
> 
> >>   2. Should we standardize an raddr RR parameter and allow this to
> >>      be used for SIP URIs?
> >>
> >>      I say no.  In general this technique breaks with outbound
> >>      proxies or transparent proxies.  If necessary, this could be
> >>      standardized outside of the SIP spec.
> > 
> > Standardizing this as an extension doesn't really help. I'm curious
> > why transparent proxies would be in the signalling path.  Outbound
> > proxies should RR (am I missing something here?)
> 
>   Transparent proxies are useful when you want to divert SIP requests
> going out to the internet to go to a special firewall proxy.  This is
> common with HTTP.
> 
>   SIP and HTTP are elegant in their proxy-awareness.  Messages can be
> forwarded between proxies without information loss.  However, with an
> raddr or whatever, now the IP layer has significance to the meaning of
> the message, which breaks the model.
> 
> >>   [...] but a client should know to convert a tel URI to a SIP URI
> >> like phonenumber@gateway.  I thought that was the point of them,
> >> not to use them as a Request-URI.
> > 
> > In this specific case, it may or may not be possible for the UAC to
> > translate the tel: URI into a SIP URI. (How does the client 
> know which
> > "domain" owns a specific number?) This is where an outbound 
> proxy may
> > play a role. In any case, it may be useful in certain cases to use a
> > Request-URI which is not directly routable.  A proxy downstream can
> > translate this into a Request-URI which is routable.  [...]
> 
>   But now the client and the outbound proxy are together acting as a
> single logical client.  This technique, translating a tel: URI into a
> SIP URI, can only be done at the edges, and does not work 
> unless you are
> sure you are not going to hit some transparent proxy in the middle.
> 
>   My point is that URI schemes other than SIP are useful for internal
> purposes.
> 
> -- 
> Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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From sip-admin@lists.bell-labs.com  Mon Nov 27 23:15:03 2000
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Date: Mon, 27 Nov 2000 23:17:16 -0500
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Hi All,

Need a clarification on the latest version of Manyfolks draft.

For the case where a UAS receives an Invite with confirm tag , the UAS
is expected
to send out a Comet to the originator if it has the ability to do so.

What should happen if after receiving an Invite with confirm tag the
call is negotiated to
a Best effort call and proceeds as Best effort. No reservations are
done. The UAS has the ability
to send a Comet. Question: Should a Comet be sent out by the UAS in this
case ?

Thanks
Ameet


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From sip-admin@lists.bell-labs.com  Mon Nov 27 23:46:02 2000
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From: Billy Biggs <Billy_Biggs@3com.com>
To: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
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        Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
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Date: Mon, 27 Nov 2000 22:45:25 -0600

  Some brief comments:

> Take the tel case: you are not interested in routing to the tel: of
> the final destintion but rather in routing to the next proxy that
> wanted to remain in the path - through a globally udp-reachable
> address. For instance, UAC->P1->P2->P3->P4->UAS. If P3 Record-Routes
> with a some URL scheme scheme only known by adacent proxyies P2 and P4
> then the call will fail because either the UAC or P1 must also be able
> to route to P3 based on the Route.

  If P2 and P4 Record-Route then this isn't a problem.

  Allowing unknown URI schemes in the RR/Contact/ReqURI is cheap.
However, they only work in a strictly engineered private network.

> IMO Request-URI's are different due to that fact that the Request-URI
> can be translated as it traverses a network - so you only have to know
> whether or not an adjacent proxy/UA supports a certain (non-SIP) URL
> scheme. This is NOT TRUE after the initial INVITE and you also no
> longer want to do network-independent routing. 

  There might be a transparent SIP proxy between you and this
destination which will snarf the message, at which point you lose the
ability to route based on request-URI.  This is unacceptable.

> As an aside, we were discussing earlier that about route parameters
> and angle brackets. A couple of peopel agreed that for
> 
> Record-Route: <url>;rr-param
> 
> A UA or proxy does not need to include rr-param's when buidling the
> Request-URI (ie, the Record-Route header parameters). If you want
> parameters to be passed into the Request-URI, they should go _inside_
> the angle brackets(ie, Record-Route Request-URI parameters). This
> follows standard URL rules and is intuitive given that the Request-URI
> cannot contain angle brackets.

  You are correct that rr-params don't get copied into the request-URI.
This is why placing routing information in rr-params is broken.

  This does not mean that rr-params are useless.  They may be useful for
extensions which require special behavior from the hop before.

  Note that parameters aren't standard across URI schemes.  As such, we
can't define, say, an "maddr" parameter and expect it to have
significance in a tel URI.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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        Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
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Date: Mon, 27 Nov 2000 21:34:19 -0800

Hi Billy, 

>   Some brief comments:
 Great.
 
> > Take the tel case: you are not interested in routing to the tel: of
> > the final destintion but rather in routing to the next proxy that
> > wanted to remain in the path - through a globally udp-reachable
> > address. For instance, UAC->P1->P2->P3->P4->UAS. If P3 Record-Routes
> > with a some URL scheme scheme only known by adacent 
> proxyies P2 and P4
> > then the call will fail because either the UAC or P1 must 
> also be able
> > to route to P3 based on the Route.
> 
>   If P2 and P4 Record-Route then this isn't a problem.
> 
>   Allowing unknown URI schemes in the RR/Contact/ReqURI is cheap.
> However, they only work in a strictly engineered private network.

I disagree that it is cheap. Sure its easy to just say "allow non-SIP URL's
in Record-ROutes" but in my mind it adds a lot of extra brittleness and
conceptual complexity that is not useful in anything _but_ strictly
engineered private networks. If the network is a strictly engineered private
network then surely the proxies can come up with a method of representing
the required information in a SIP URL? 

> 
> > IMO Request-URI's are different due to that fact that the 
> Request-URI
> > can be translated as it traverses a network - so you only 
> have to know
> > whether or not an adjacent proxy/UA supports a certain (non-SIP) URL
> > scheme. This is NOT TRUE after the initial INVITE and you also no
> > longer want to do network-independent routing. 
> 
>   There might be a transparent SIP proxy between you and this
> destination which will snarf the message, at which point you lose the
> ability to route based on request-URI.  This is unacceptable.

Ok, your right but I still purtain that this is different to placing a
non-SIP URL in a Route header. With the Request-URI, once the request hits a
proxy that converts the tel: URL to a sip: Request-URI then all further
proxies don't need to understand tel URL's (which is a feasible
requirement). If a proxy inserts a tel URL then all UA'a and proxies in the
path must potentially understand how to route tel URL's - which is not
feasible. 

As you argue, proxies could collude to "buffer" non-SIP Route URL's but will
probably result in assymetic Routes and all sorts of other trouble for what
gain?

> 
> > As an aside, we were discussing earlier that about route parameters
> > and angle brackets. A couple of peopel agreed that for
> > 
> > Record-Route: <url>;rr-param
> > 
> > 
> 
>   You are correct that rr-params don't get copied into the 
> request-URI.
> This is why placing routing information in rr-params is broken.

We are not suggesting this. We are saying that implementation specific
information (I think calling it routing information is confusing) can be
placed into the URL parameters -not header parameters.
> 
>   This does not mean that rr-params are useless.  They may be 
> useful for
> extensions which require special behavior from the hop before.

We agree on this.
> 
>   Note that parameters aren't standard across URI schemes.  
> As such, we
> can't define, say, an "maddr" parameter and expect it to have
> significance in a tel URI.

EXACTLY!! You don't have this problem if you don't try to put non-SIP URL's
in the Route header  Why should any other SIP entity have to handle this
information? Other entities only have to ensure that the message is routed
to the proxy. Are saying that this private network has no concepts of
routing via hostnames or addresses?

Regards, 

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 

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"Fairlie-Cuninghame, Robert" wrote:
> 
> Hi guys,
> 
> So to patch that problem a raddr parameter is suggested - that all UA's and
> proxies must understand. However, the raddr must still specify a globally
> udp-reachable address for things to work (otherwise you run into the same
> problem). So what's the point? Just KISS and stick with SIP URL's. I think
> the simpler something is, the more likely it will be to work across
> everyone's SIP devices - and this is somethign you want to work.
> 

Well the reason for the raddr parameter is the same as the
maddr parameter for SIP-URL. But having it as a RR parameter
makes it more general and independent of type of URI.
If you have a pool of machines building up a big SIP server 
you may want to make shure every new request for a certain
call-leg ends up on a specific machine in the pool (where
you have the states for this specific call-leg). This
is what I understand maddr was for. But since maddr only
exists for SIP-URL, how to do it for other types of URIs ?

This is the reason why I suggested the raddr parameter.
With the raddr parameter a server can specify which machine
it wants further requests to be sent to regardless of what
type of URI it has put into record-route. That was the idea
anyway. If people are happy with that you only can have
SIP-URLs in record-route then the raddr is not needed.
Personally I see a risk that some new type of URI pops
up in the future that SIP also needs to support in RR.

-- 
Bertil Engelholm
AXE Research and Development        voice : +46 8 727 3499
SIP Security                        Fax   : +46 8 647 8276
S-126 25 Stockholm Sweden           E-mail:
Bertil.Engelholm@uab.ericsson.se

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From sip-admin@lists.bell-labs.com  Tue Nov 28 03:59:12 2000
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Date: Tue, 28 Nov 2000 00:58:26 -0800

Hi Bertil,
> 
> Well the reason for the raddr parameter is the same as the
> maddr parameter for SIP-URL. But having it as a RR parameter
> makes it more general and independent of type of URI.
> If you have a pool of machines building up a big SIP server 
> you may want to make shure every new request for a certain
> call-leg ends up on a specific machine in the pool (where
> you have the states for this specific call-leg). This
> is what I understand maddr was for. But since maddr only
> exists for SIP-URL, how to do it for other types of URIs ?

Are you suggesting something of the form:

Record-Route: <tel:911>;raddr=emergency.sd.com

then the Request-URI will become simply tel:911 which sees potentially
brittle to me. 

Something of the form 

Record-Route: <sip:blah@proxy.com;blah2=blah3> 

where blah and blah2 represent implementation specific fields possibly
encoding tel:911 [or better yet, something more useful to the proxy] will
still always be routed to the proxy regardless of what happens in between. 
The field blah and blah2 only have significance to the proxy and hence don't
need to be mandated/examined/interpreted/understood by anyone but the proxy.

In this context, the SIP URL is essentially only describing the next hop
address - much like the Via field. I haven't heard requests (yet) to put tel
URL's into the Via fields becuase it is unneccessary - all you need is the
address and port - all other information is significant only to the
generating proxy and hence it can encode what it wants, right ?

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 

> This is the reason why I suggested the raddr parameter.
> With the raddr parameter a server can specify which machine
> it wants further requests to be sent to regardless of what
> type of URI it has put into record-route. That was the idea
> anyway. If people are happy with that you only can have
> SIP-URLs in record-route then the raddr is not needed.
> Personally I see a risk that some new type of URI pops
> up in the future that SIP also needs to support in RR.
> 
> -- 
> Bertil Engelholm
> AXE Research and Development        voice : +46 8 727 3499
> SIP Security                        Fax   : +46 8 647 8276
> S-126 25 Stockholm Sweden           E-mail:
> Bertil.Engelholm@uab.ericsson.se
> 

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From sip-admin@lists.bell-labs.com  Tue Nov 28 04:15:05 2000
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Date: Tue, 28 Nov 2000 10:14:31 +0100

Henry is right.
Jonathan is indeed answering a lot of questions. So Jonathan..thank you for your time and we are all looking forward for more "helpdesk" feedback. :-)

Arnoud van Wijk


-----Original Message-----
From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com]
Sent: maandag 27 nov 2000 22:34
To: Brian S; sip@lists.bell-labs.com
Subject: RE: [SIP] What SIP doesn't tell you how to do.


We should all thank Jonathan for answering countless questions on this
list - acting as a volunteer help desk for everybody!

Henry

-----Original Message-----
From: sip-admin@lists.bell-labs.com
[mailto:sip-admin@lists.bell-labs.com]On Behalf Of Brian S
Sent: Monday, November 27, 2000 10:30 AM
To: sip@lists.bell-labs.com
Subject: [SIP] What SIP doesn't tell you how to do.


Apparently SIP doesn't tell you how to do any of the
following as well:

1. Be professional.
2. Show others common courtesy.
3. Switch to decaf.

BTW- He never asked how to do anything, just if it
could be done.

-----Original Message-----
From: Jonathan Rosenberg
[mailto:jdrosen@dynamicsoft.com]
Sent: Wednesday, November 22, 2000 10:59 AM
To: 'Sanjiv Deshpande'; sip@lists.bell-labs.com
Subject: RE: [SIP] multiple location


SIP is just a protocol. Thats it. Its a tool. If you
want device A to call
device B, it tells you how to do that. Like all tools,
you can use them in
many different ways. When you buy a hammer in Home
Depot, does it tell you
whether to use your left hand or right? How many
fingers to grip with it?
The size of the nails to use with it? Of course not.

However, for some reason that I cannot fathom, many
people seem to believe
the opposite is true for protocols; that is, protocols
need to insist on how
syntax and semantics AND how they are applied to solve
real world problems.
This approach just results in systems that are useful
to no one. Would you
buy a hammer that said "for nails of length 1 inch
only?" I doubt it.

So, to be specific, SIP doesn't tell you how to do any
of the following:

1. the color of the phones that use SIP
2. the GUI for SIP calls
3. how you build a network of proxy servers
4. whether a phone can make calls while its in a call
5. qualifications for human beings to make use of SIP
softphones (i.e., IQ
levels, height, or what have you).
6. how many simultaneous calls the device can be in

Thanks,
Jonathan R.



---
Jonathan D. Rosenberg                       72 Eagle
Rock Ave.
Chief Scientist                             First
Floor
dynamicsoft                                 East
Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:
(973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE:
(973) 952-5000
http://www.dynamicsoft.com


> -----Original Message-----
> From: Sanjiv Deshpande
[mailto:sdeshpan@telcordia.com]
> Sent: Wednesday, November 22, 2000 11:36 AM
> To: sip@lists.bell-labs.com
> Subject: [SIP] multiple location
>
>
>
>
>
>
> Q:
> INVITE Sent by UAC got forked and reached two UAS1
and UAS2 ,
>  the UAC decided
> to talk to UAS1. Is it possible that UAS2 (Assuming
it is a
> client also) can
> make a call to someone else.
>
>
> What I wanted to know is, A user can be contacted at
multiple
> location if one of
> the location answers a CALL is it possible for
somebody else
> on behalf of that
> user make CALLS or receive CALLS....
>
>
> Thanks
> Sanjiv.
>
>
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>

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Date: Tue, 28 Nov 2000 10:31:54 +0000


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Ranga,

I think I'll go for green this time...

Regards,
Chris


"M. Ranganathan" wrote:

> Chris,
>
> Thanks for the prompt response. Not to drag this on for ever, but I
> have a few more suggestions  in followup to your
> comments/clarifications  (in blue below). As  you are in Ireland, your
> missiles cannot reach me so I can be bold :)
>
>
>
> Regards
>
> Ranga.
>
>
> A design issue not discussed below:
>
> Whenever there is more than one header to be returned (via  headers
> for example), you return an array.  An array is easy to walk through.
> However,  I would think it more useful to return a collection of some
> sort that supports additional methods such as checking if an address
> exists in the list of headers returned etc. That might be of some use
> to an implementation.  If you define the returned type as an array of
> headers, there is no way to extend the class to support additional
> functionality.  May I suggest adding a collection class interface for
> collections of SIP Headers (and that can be specialized by
> implementations to support additional query methods etc.), say
> something like SIPHeaderList

I initially wanted to return a Vector, but was told that an array was
preferable for API's. Besides, any addition of functionality would be
purely for the implementor's benefit and would not be available (without
using reflection anway) to the application.

>
>
>
> Chris Harris wrote:
>
>> Ranga,
>>
>> My comments are in red...
>>
>> Chris
>>
>>
>> "M. Ranganathan" wrote:
>>
>> > Chris,
>> >
>> > Thanks for the prompt response.  Hope you had a great Thanksgiving.
>>
>> I wish there was such a thing as Thanksgiving in Ireland :(
>>
>> >
>> >
>> > My followup is below. ( in blue)
>> >
>> > Regards
>> >
>> > Ranga.
>> >
>> > Chris Harris wrote:
>> >
>> >>  Ranga,
>> >>
>> >>  See comments below.
>> >>
>> >>  Regards,
>> >>  Chris
>> >>
>> >>  "M. Ranganathan" wrote:
>> >>
>> >> >
>> >> > Chris, Erik,
>> >> >
>> >> > While we are on the subject of Exceptions, why do null
>> >> > arguments throw runtime exceptions when you do a get on them?
>> >> > For example, in the Accept interface, getContentSubtype throws
>> >> > a IllegalArgumentException if  is null. IMHO, I think it is
>> >> > valid to have a null subtype so why not just return null rather
>> >> > than throw exception here?
>> >>
>> >>  By a "doing a get on a null argument" here I assume you mean
>> >>  attempting to get a non-existent optional field? If you do this a
>> >>  checked ParameterNotSetException is thrown, not a runtime
>> >>  exception.
>> >
>> >
>> > Yes. Thats what I had meant.  The word "argument" should have been
>> > "field".   Why do you need a  ParameterNotSet exception in this
>> > case?  Are you trying to handle the case of basic types like
>> > integers, floats and so on ? For objects, null is synonymous with
>> > not set isnt it?
>>
>> We discussed this matter in depth at the last JAIN SIP meeting in
>> October, and we came to the conclusion that getting and setting null
>> values is not a good idea for several reasons e.g. it is easier to
>> debug a ParameterNotSetException than searching for the source of a
>> later NullPointerException, and it forces the application to allow
>> for the fact that there is a reasonable chance that a particular
>> field does not exist. [Of course, if we did allow the use of null,
>> we could cut down on the number of methods and drop the
>> ParameterNotSetException - but  is it really worth sacrificing
>> clarity and robustness for the sake of cutting down on API size?]
>
>
> Well, I guess that is a judgement call.  With this design, you may
> have code that is cluttered with a lot of exception handlers and the
> API is quite big.  Checking for null is fairly standard  (am I giving
> my age away ? :)  and I can see at least a few places in the base JAVA
> API where null is returned to denote empty  and anyway, the app has a
> good old stack trace to rely on to see the source of the exception
> when null pointers are encountered.   Guarding an application from
> null pointer exception is pretty hard anyway so why complicate the API
> to handle this one case?  ( For collections of SIPHeaders, you can
> check for empty collection and you will not have this problem. )
>

I guess it is a judgement call. I actually approached this topic at the
meeting from your perspective (devil's advocate? :) ) There were a few
who agreed (who admitted that their opinion was due to their C
background!), but the majority preferred the explicit handling of the
ParameterNotSetException and the use of has and remove methods to be on
the safe side. I hear what you are saying about null being sprinkled
about in the core Java APIs - but even Sun can make mistakes :)

>
>
>
>>
>>
>> >
>> >
>> >>
>> >>
>> >>  There are two approaches to handling optional header parts. One
>> >>  approach is to allow the passing and returning of null objects
>> >>  from get and set methods (implicit "remove" and "has"), and the
>> >>  other is to explicitly add has and remove methods (and exceptions
>> >>  on the set and get methods). As for mandatory header parts - I
>> >>  think passing null to a set method should definitely throw an
>> >>  IllegalArgumentException.
>> >
>> > I have the following questions about setXXX methods:
>> >
>> > 1.  Why do you restrict throwing IllegalArgumentException to only
>> > null arguments in some of the headers? Why are you not checking for
>> > proper formatting of the passed arguments (by giving it to the
>> > parser for example and running the appropriate  parser rule).
>>
>> In general, an IllegalArgumentException is thrown if an argument to
>> a set method fails basic validation (i.e. no implementation-specific
>> parsing is used). Note that an application can ensure that this
>> exception is not thrown by following the rules defined in the API .
>> So if an argument is a String, I throw an IllegalArgumentException
>> if the argument is null or zero-length.
>>
>> Of course this is not sufficient validation and the string must be
>> parsed to check for whitespaces, newlines etc, and different
>> implementations will be more tolerant than others. This is why there
>> is a checked JainSipParseException - an application cannot tell in
>> advance if a particular implementation will accept a certain value,
>> so it must catch this exception.
>>
>> I am proposing that this exception should also be thrown by get
>> methods (for all fields), because if an implementation uses lazy
>> parsing, then there is no guarantee that the header value will parse
>> correctly upon the invocation of the get method, and again this is
>> implementation-dependent.
>>
>> >
>> >
>> > 2.  A more general question - why do you have set methods at all in
>> > the interfaces?  The class that implements the interface can have
>> > set methods but the interface itself does not need to have them.
>>
>> I have set methods in the interfaces because the objects will not
>> only be coming up from the implementation - an application needs to
>> be able to create and send requests, and this means being able to
>> set the values.
>
> Ah! My understanding was correct then.
>
> OK. Perhaps this would be too much of a change for you to consider but
> let me make a suggestion anyway ....
>
> I think formatting headers should be considered under a different set
> of API.  Here is what I am suggesting
>
> 1. Get rid of the setXXX interfaces (the parser will have direct
> access to the classes and does not need them and it will cut down the
> size of the API).
>
> 2. Have a  SIPMessageFormatter interface who'se sole job it is to
> format output messages.  This can have methods to format the different
> outgoing messages and will throw parse exceptions if the strings that
> it gets are incorrect. The MessageFormatter interface takes as
> parameter values only Strings.
>
> Concretely,  for example:
>
> public interface SIPMessageFormatter {
>         public String createFromMessage( String from ,  String tag)
> throws SIPParseException;
>
>     .....................
>
> }
>
> It would be slower for an implementation to extract the same
> information out of a structure and hand it to a parser for validation
> (IMHO). A "loose implementation" may not want to be bothered with
> validation at all and leave it to the reciever to reject the mesage,
> thereby making the implementation faster or it may do very cursory
> checking.
>

This is a major change at this stage (public review technically closes
today)  - I personally don't think it's very OO, but I would be
interested to hear other people's opinons.

>
>
>
>
>
>>
>>
>> >  Moreover, the parser has the job of setting up stuff in the
>> > returned object. It has a set of BNF rules to decide on the
>> > correctness of the arguments and it does not need for the class to
>> > re-implement the check. For example, I think it ought to be the
>> > parser's job to check if the q value is between 0 and 1.   (Is the
>> > purpose of the set method to allow for the application to do header
>> > formatting?)
>>
>> I absolutely agree that the parser should perform validation (hence
>> the JainSipParseException) - but as I mentioned above there are
>> certain basic criteria that must be met by all implementations and
>> applications - I am using the IllegalArgumentException to cover
>> these. Perhaps certain cases go too far in this validation - perhaps
>> the IllegalArgumentException could be thrown only if a null argument
>> is passed and all other validation is left up to the
>> implementation's parser? (although a problem with certain cases such
>> as PriorityHeader comes to mind, where the header values are defined
>> as constants in the API and an IllegalArgumentException is thrown if
>> any other value is passed - should we allow any non-null String
>> values for the priority? Or to take your example of q-value above -
>> should we just allow any non-null Float and let the parser enforce
>> the 0-1 range?
>
>
> I think I understand the point about checking for null.  I think, if
> you are dead set on adopting the set method as a way of constructing
> headers, one should validate arguments via the parser and throw
> ParseException for anything other than null arguments. On the other
> hand, I propose something entirely different for constructing headers
> (above).
>
>>
>>
>>
>> >
>> >
>> >
>> >>
>> >>
>> >>  (If it is considered acceptable to have a "null" subType then we
>> >>  should make it optional)
>> >>
>> >> >
>> >> >
>> >> > I don't quite follow the argument that lazy parsing
>> >> > "necessitates throwing a parse exception for all get
>> >> > methods".   Here is what I had envisioned:
>> >> >
>> >> >
>> >> > 1 When a message comes in it will be separated into headers
>> >> > based on type.
>> >> > 2 The header would be returned with no field initialized except
>> >> > the text corresponding to the message. (this would necessitate
>> >> > examining the first few bytes of the message to determine the
>> >> > header type but the rest remains unparsed.) You can also stow
>> >> > away the type information in the Header that is returned as
>> >> > well to know which parser rule to invoke later.
>> >> > 3. Later, when the application wants to parse the header, it
>> >> > has the string corresponding to the message and can hand it off
>> >> > to the parser, which can then fill in the appropriate fields
>> >> > corresponding to the header.
>> >> > 4. Subsequently, the application can access the fields just as
>> >> > an eager parser based application would do.
>> >> >
>> >> > ( Where am I missing the boat with this? )
>> >>
>> >>  An application has no idea whether an implementation is using
>> >>  lazy or eager parsing - it is simply passed a header object - and
>> >>  then if it is interested in a particular field it calls the
>> >>  appropriate get method (which would internally invoke the parser
>> >>  in a lazy parsing implementation). So the application is
>> >>  completely oblivious to whether or not this "optimisation" is
>> >>  used by the implementation.
>> >
>> >
>> > Ah! ( Why is this hiding  of the implementation so important? )
>> > Can this be handled in the following fashion by an implementation:
>> >
>> > 1. The implementation keeps a flag in the  class that implements
>> > the interface whether the parsing has been done or not (i.e.
>> > boolean parsingDone flag).
>> >
>> > 2. The Implementation keeps the entire string corresponding to the
>> > header in the object. So the parser has to peek into the message
>> > header to know what type the message is as soon as it has the
>> > header string and then grabs the string and stores it away for lazy
>> > parsing.
>> >
>> >
>> > 3. The get Method  checks if the flag has been set, calls the
>> > parser to do the parse if the flag has not been set and then
>> > returns a parseException if the parse fails. This can be done
>> > transparently to the caller.
>>
>> This is exactly what I would expect a lazy-parsing implementation to
>> do - the fact that lazy-parsing is hidden from the application
>> (except that the application must catch a JainSipParseException on
>> the get methods to allow for the possibility that an implementation
>> uses lazy-parsing - so technically the application will know that an
>> implementation uses lazy-parsing if this exception is thrown by a
>> get method - but I think the two cases are handled nicely by the
>> addition of the exception)
>
>> >
>> >
>> >
>> >
>> >
>> >
>> >>
>> >>
>> >>  If the field is optional and does not exist in the header a
>> >>  ParameterNotSetException will be thrown by the implementation.
>> >
>> > A suggestion (shoot it down if it does not jive):
>> >
>> > 1. All fields are represented by objects (not simple types like
>> > int, use Integer instead).
>>
>> I buy this part - then the IllegalArgumentException is only ever
>> thrown if an object parameter is null, and the JainSipParseException
>> is thrown for all other problems with the argument.
>
>
> OK this jives and is easily defined.
>
>>
>>
>> >
>> > 2. If the field is not set just return null
>>
>> See above for null discussion,
>>
>> >
>> >
>> >
>> >
>> >>  If the field is mandatory and does not exist, a
>> >>  JainSipParseException will be thrown (header is malformed). If
>> >>  any header field is malformed a JainSipParseException is thrown.
>> >>  If a JainSipParseException is thrown, this indicates that the
>> >>  application must use the header's getValue method to get a string
>> >>  representation of the header value and do what it can with that.
>> >
>> >
>> > Why the headers getValue ? Why is the string not part of the
>> > Exception?
>>
>> Sure - we can say that the JainSipParseException's message must be
>> the string value of the header.
>
>
> OK Glad you buy that one.

I was just wondering what the message of a JainSipParseException thrown
from a set method should contain - the value passed in? Perhaps we
should add a string field to the JainSipParseException (say unparsable)
which can be whatever the parser was unable to parse? Then the message
can be left for explanatory comments.

>
>
>
>>
>
>>
>>
>> >
>> >
>> >
>> >>
>> >>
>> >> >
>> >> >
>> >> > It appears that what is being attempted here (please correct me
>> >> > if I am wrong) is to use exceptions as a mechanism to catch
>> >> > un-initialized fields so the parser can be invoked in the
>> >> > exception handler perhaps. This leads to a lot of overhead in
>> >> > clunky code. I think it is a mistake to throw exceptions on
>> >> > every get to try to protect the application from inadvertently
>> >> > accessing fields that have not been initialized.  Why not just
>> >> > assume that the application must be smart enough to parse the
>> >> > header text before accessing fields and not try to provide a
>> >> > complex mechanism as above?
>> >>
>> >>
>> >>  We are using ParameterNotSetException for the case of an optional
>> >>  header not existing, and a JainSipParseException for the case of
>> >>  a malformed header being received from the network.
>> >>
>> >>  If we assume the application is smart enough to do all header
>> >>  value parsing, we reduce a header to just being two strings - the
>> >>  type and value, which puts the burden of parsing on an
>> >>  application - this work can be done by the underlying
>> >>  implementation - doesn't it make more sense for an application to
>> >>  catch an exception than have to parse all header value strings
>> >>  (especially since this functionality already exists in the
>> >>  underlying stack)
>> >
>> >
>> > No that is not what I had intended. Parser functionality exists in
>> > the stack but has to be called explicitly by the implementation.
>> > Too much hiding of the implementation from the application has its
>> > own set of disadvantages (IMHO).
>>
>> I wanted parser fuctionality to be invoked implicitly by an
>> application - from the API perspective it is an implementation
>> detail - so the application should not be aware of how or when the
>> underlying parser is invoked - it should just know that it is there
>> in some form, and that if it tries to get or set a value that the
>> implementation doesn't like it will get a JainSipParseException
>> thrown back at it. This is important if we want to maintain the
>> three core values of JAIN - portability, portability and portabilty
>> ;)
>
>
> The point about portability is a good one. Of course the important
> thing about portability is to be able to maintain the same semantics
> of exceptions as well.  For an eager parser,  there is no possibility
> of throwing a parse  exception  on a get (i.e. parse exception can
> never occur on a getXXX) so there will be some curious code in a
> getXXX  implementation  for an eager parser that has to throw an
> exception even when it does not need to (just to keep the compiler
> happy).   Second,  the application has to be willing to catch such an
> exception to be able to function both with a lazy and an eager
> implementation in a transparent fashion and it may do different things
> when the implementation is lazy, which exposes the underlying
> implementation anyway as you have noted.
>
> Explicit control by the application may give the application finer
> grained control and eliminate some awkward exception handling that is
> not meaningful for an eager parser .  This way, you can also have
> mixed mode parsing, allowing the application to decide what headers it
> wants to handle in a lazy fashion and which ones it wants to handle in
> an eager fashion.

The application does not get to decide which method is used by an
implementation - an implementation could be lazy, eager or, as you point
out, a mixture of both. An application has to accomodate this fact to be
portable.

>
>
> Another suggestion ( may be a bit radical) - would it make sense to
> have extension interfaces (with enhanced getXXX  interfaces)  that are
> only implemented by lazy parsers and make lazy parsing optional (so
> only highly performance sensitive apps will  avail themselves of such
> interfaces whereas most applications will use the eager parser
> interfaces for getXXX).  ( May or may not make sense, I thought I
> would mention it as it occured to me.  )

Again, the application does not choose - the implementator does.

>
>
> Anyway,  there is more ways than one to skin a cat. Just keep
> everything as simple as possible but no simpler. I am sure you are
> eager to bring this to closure so I will not belabor the point about
> eager parsing versus lazy parsing  although, as a final point I would
> just like to state that portability can be handled whether or not the
> parse is explicitly or implicitly lazy. Enough said on this point.

Thanks for all your input Ranga!

>
>
>
>
>
> Regards,
>
> Ranga.
>
>
>
>
>
> --
> M.Ranganathan
> NIST Advanced Networking Technologies Group,
> 100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
> Tel: 301 975 3664 Fax: 301 590 0932
>
>

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<font color="#009900">Ranga,</font><font color="#009900"></font>
<p><font color="#009900">I think I'll go for green this time...</font><font color="#009900"></font>
<p><font color="#009900">Regards,</font>
<br><font color="#009900">Chris</font>
<br>&nbsp;
<p>"M. Ranganathan" wrote:
<blockquote TYPE=CITE>Chris,
<p>Thanks for the prompt response. Not to drag this on for ever, but I
have a few more suggestions&nbsp; in followup to your comments/clarifications&nbsp;
(in blue below). As&nbsp; <font color="#000000">you are in Ireland, your
missiles cannot reach me so I can be bold :)</font>
<br>&nbsp;
<br>&nbsp;
<p>Regards
<p>Ranga.
<br>&nbsp;
<p><font color="#3333FF">A design issue not discussed below:</font>
<p><font color="#3333FF">Whenever there is more than one header to be returned
(via&nbsp; headers for example), you return an array.&nbsp; An array is
easy to walk through. However,&nbsp; I would think it more useful to return
a collection of some sort that supports additional methods such as checking
if an address exists in the list of headers returned etc. That might be
of some use to an implementation.&nbsp; If you define the returned type
as an array of headers, there is no way to extend the class to support
additional functionality.&nbsp; May I suggest adding a collection class
interface for collections of SIP Headers (and that can be specialized by
implementations to support additional query methods etc.), say something
like <i>SIPHeaderList</i></font></blockquote>
<font color="#009900">I initially wanted to return a Vector, but was told
that an array was preferable for API's. Besides, any addition of functionality
would be purely for the implementor's benefit and would not be available
(without using reflection anway) to the application.</font>
<blockquote TYPE=CITE><i><font color="#3333FF"></font></i>&nbsp;
<br>&nbsp;
<p>Chris Harris wrote:
<blockquote TYPE=CITE><font color="#FF0000">Ranga,</font>
<p><font color="#FF0000">My comments are in red...</font>
<p><font color="#FF0000">Chris</font>
<br>&nbsp;
<p>"M. Ranganathan" wrote:
<blockquote TYPE=CITE>Chris,
<p>Thanks for the prompt response.&nbsp; Hope you had a great Thanksgiving.</blockquote>
<font color="#FF0000">I wish there was such a thing as Thanksgiving in
Ireland :(</font>
<blockquote TYPE=CITE>&nbsp;
<p>My followup is below. ( in blue)
<p>Regards
<p>Ranga.
<p>Chris Harris wrote:
<blockquote TYPE=CITE>Ranga,
<p>See comments below.
<p>Regards,
<br>Chris
<p>"M. Ranganathan" wrote:
<blockquote TYPE=CITE>&nbsp;
<br>Chris, Erik,
<p>While we are on the subject of Exceptions, why do null arguments throw
runtime exceptions when you do a get on them? For example, in the Accept
interface, getContentSubtype throws a IllegalArgumentException if&nbsp;
is null. IMHO, I think it is valid to have a null subtype so why not just
return null rather than throw exception here?</blockquote>
By a "doing a get on a null argument" here I assume you mean attempting
to get a non-existent optional field? If you do this a checked ParameterNotSetException
is thrown, not a runtime exception.</blockquote>

<p><br><font color="#3333FF">Yes. Thats what I had meant.&nbsp; The word
"argument" should have been "field".&nbsp;&nbsp; Why do you need a&nbsp;
ParameterNotSet exception in this case?&nbsp; Are you trying to handle
the case of basic types like integers, floats and so on ? For objects,
null is synonymous with not set isnt it?</font></blockquote>
<font color="#FF0000">We discussed this matter in depth at the last JAIN
SIP meeting in October, and we came to the conclusion that getting and
setting null values is not a good idea for several reasons e.g. it is easier
to debug a ParameterNotSetException than searching for the source of a
later NullPointerException, and it forces the application to allow for
the fact that there is a reasonable chance that a particular field does
not exist. [Of course, if we did allow the use of null, we could cut down
on the number of methods and drop the ParameterNotSetException - but&nbsp;
is it really worth sacrificing clarity and robustness for the sake of cutting
down on API size?]</font></blockquote>

<p><br><font color="#3333FF">Well, I guess that is a judgement call.&nbsp;
With this design, you may have code that is cluttered with a lot of exception
handlers and the API is quite big.&nbsp; Checking for null is fairly standard&nbsp;
(am I giving my age away ? :)&nbsp; and I can see at least a few places
in the base JAVA API where null is returned to denote empty&nbsp; and anyway,
the app has a good old stack trace to rely on to see the source of the
exception when null pointers are encountered.&nbsp;&nbsp; Guarding an application
from null pointer exception is pretty hard anyway so why complicate the
API to handle this one case?&nbsp; ( For collections of SIPHeaders, you
can check for empty collection and you will not have this problem. )</font>
<br>&nbsp;</blockquote>
<font color="#009900">I guess it is a judgement call. I actually approached
this topic at the meeting from your perspective (devil's advocate? :) )
There were a few who agreed (who admitted that their opinion was due to
their C background!), but the majority preferred the explicit handling
of the ParameterNotSetException and the use of has and remove methods to
be on the safe side. I hear what you are saying about null being sprinkled
about in the core Java APIs - but even Sun can make mistakes :)</font>
<blockquote TYPE=CITE>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<p>There are two approaches to handling optional header parts. One approach
is to allow the passing and returning of null objects from get and set
methods (implicit "remove" and "has"), and the other is to explicitly add
has and remove methods (and exceptions on the set and get methods). As
for mandatory header parts - I think passing null to a set method should
definitely throw an IllegalArgumentException.</blockquote>
<font color="#3333FF">I have the following questions about setXXX methods:</font>
<p><font color="#3333FF">1.&nbsp; Why do you restrict throwing IllegalArgumentException
to only null arguments in some of the headers? Why are you not checking
for proper formatting of the passed arguments (by giving it to the parser
for example and running the appropriate&nbsp; parser rule).</font></blockquote>
<font color="#FF0000">In general, an IllegalArgumentException is thrown
if an argument to a set method fails basic validation (i.e. no implementation-specific
parsing is used). Note that an application can ensure that this exception
is not thrown by following the rules defined in the API . So if an argument
is a String, I throw an IllegalArgumentException if the argument is null
or zero-length.</font>
<p><font color="#FF0000">Of course this is not sufficient validation and
the string must be parsed to check for whitespaces, newlines etc, and different
implementations will be more tolerant than others. This is why there is
a checked JainSipParseException - an application cannot tell in advance
if a particular implementation will accept a certain value, so it must
catch this exception.</font>
<p><font color="#FF0000">I am proposing that this exception should also
be thrown by get methods (for all fields), because if an implementation
uses lazy parsing, then there is no guarantee that the header value will
parse correctly upon the invocation of the get method, and again this is
implementation-dependent.</font>
<blockquote TYPE=CITE>&nbsp;
<p><font color="#3333FF">2.&nbsp; A more general question - why do you
have set methods at all in the interfaces?&nbsp; The class that implements
the interface can have set methods but the interface itself does not need
to have them.</font></blockquote>
<font color="#FF0000">I have set methods in the interfaces because the
objects will not only be coming up from the implementation - an application
needs to be able to create and send requests, and this means being able
to set the values.</font></blockquote>
<font color="#3333FF">Ah! My understanding was correct then.</font>
<p><font color="#3333FF">OK. Perhaps this would be too much of a change
for you to consider but let me make a suggestion anyway ....</font>
<p><font color="#3333FF">I think formatting headers should be considered
under a different set of API.&nbsp; Here is what I am suggesting</font>
<p><font color="#3333FF">1. Get rid of the setXXX interfaces (the parser
will have direct access to the classes and does not need them and it will
cut down the size of the API).</font>
<p><font color="#3333FF">2. Have a&nbsp; SIPMessageFormatter interface
who'se sole job it is to format output messages.&nbsp; This can have methods
to format the different outgoing messages and will throw parse exceptions
if the strings that it gets are incorrect. The MessageFormatter interface
takes as parameter values only Strings.</font>
<p><font color="#3333FF">Concretely,&nbsp; for example:</font>
<p><font color="#3333FF">public interface SIPMessageFormatter {</font>
<br><font color="#3333FF">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; public
String createFromMessage( String from ,&nbsp; String tag) throws SIPParseException;</font>
<p><font color="#3333FF">&nbsp;&nbsp;&nbsp; .....................</font>
<p><font color="#3333FF">}</font>
<p><font color="#3333FF">It would be slower for an implementation to extract
the same information out of a structure and hand it to a parser for validation
(IMHO). A "loose implementation" may not want to be bothered with validation
at all and leave it to the reciever to reject the mesage, thereby making
the implementation faster or it may do very cursory checking.</font>
<br>&nbsp;</blockquote>
<font color="#009900">This is a major change at this stage (public review
technically closes today)&nbsp; - I personally don't think it's very OO,
but I would be interested to hear other people's opinons.&nbsp;</font>
<blockquote TYPE=CITE>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<blockquote TYPE=CITE><font color="#3333FF">&nbsp;Moreover, the parser
has the job of setting up stuff in the returned object. It has a set of
BNF rules to decide on the correctness of the arguments and it does not
need for the class to re-implement the check. For example, I think it ought
to be the parser's job to check if the q value is between 0 and 1.&nbsp;&nbsp;
(Is the purpose of the set method to allow for the application to do header
formatting?)</font></blockquote>
<font color="#FF0000">I absolutely agree that the parser should perform
validation (hence the JainSipParseException) - but as I mentioned above
there are certain basic criteria that must be met by all implementations
and applications - I am using the IllegalArgumentException to cover these.
Perhaps certain cases go too far in this validation - perhaps the IllegalArgumentException
could be thrown only if a null argument is passed and all other validation
is left up to the implementation's parser? (although a problem with certain
cases such as PriorityHeader comes to mind, where the header values are
defined as constants in the API and an IllegalArgumentException is thrown
if any other value is passed - should we allow any non-null String values
for the priority? Or to take your example of q-value above - should we
just allow any non-null Float and let the parser enforce the 0-1 range?</font></blockquote>

<p><br><font color="#3333FF">I think I understand the point about checking
for null.&nbsp; I think, if you are dead set on adopting the set method
as a way of constructing headers, one should validate arguments via the
parser and throw ParseException for anything other than null arguments.
On the other hand, I propose something entirely different for constructing
headers (above).</font>
<blockquote TYPE=CITE>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<p>(If it is considered acceptable to have a "null" subType then we should
make it optional)
<blockquote TYPE=CITE>&nbsp;
<p>I don't quite follow the argument that lazy parsing&nbsp; "necessitates
throwing a parse exception for all get methods".&nbsp;&nbsp; Here is what
I had envisioned:
<br>&nbsp;
<p>1 When a message comes in it will be separated into headers based on
type.
<br>2 The header would be returned with no field initialized except the
text corresponding to the message. (this would necessitate examining the
first few bytes of the message to determine the header type but the rest
remains unparsed.) You can also stow away the type information in the Header
that is returned as well to know which parser rule to invoke later.
<br>3. Later, when the application wants to parse the header, it has the
string corresponding to the message and can hand it off to the parser,
which can then fill in the appropriate fields corresponding to the header.
<br>4. Subsequently, the application can access the fields just as an eager
parser based application would do.
<p>( Where am I missing the boat with this? )</blockquote>
An application has no idea whether an implementation is using lazy or eager
parsing - it is simply passed a header object - and then if it is interested
in a particular field it calls the appropriate get method (which would
internally invoke the parser in a lazy parsing implementation). So the
application is completely oblivious to whether or not this "optimisation"
is used by the implementation.</blockquote>

<p><br>Ah! ( Why is this hiding&nbsp; of the implementation so important?
)&nbsp; Can this be handled in the following fashion by an implementation:
<p>1. The implementation keeps a flag in the&nbsp; class that implements
the interface whether the parsing has been done or not (i.e. boolean parsingDone
flag).
<p>2. The Implementation keeps the entire string corresponding to the header
in the object. So the parser has to peek into the message header to know
what type the message is as soon as it has the&nbsp; header string and
then grabs the string and stores it away for lazy parsing.
<br>&nbsp;
<p>3. The get Method&nbsp; checks if the flag has been set, calls the parser
to do the parse if the flag has not been set and then returns a parseException
if the parse fails. This can be done transparently to the caller.</blockquote>
<font color="#FF0000">This is exactly what I would expect a lazy-parsing
implementation to do - the fact that lazy-parsing is hidden from the application
(except that the application must catch a JainSipParseException on the
get methods to allow for the possibility that an implementation uses lazy-parsing
- so technically the application will know that an implementation uses
lazy-parsing if this exception is thrown by a get method - but I think
the two cases are handled nicely by the addition of the exception)</font></blockquote>

<blockquote TYPE=CITE>
<blockquote TYPE=CITE>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<p>If the field is optional and does not exist in the header a ParameterNotSetException
will be thrown by the implementation.</blockquote>
A suggestion (shoot it down if it does not jive):
<p>1. All fields are represented by objects (not simple types like int,
use Integer instead).</blockquote>
<font color="#FF0000">I buy this part - then the IllegalArgumentException
is only ever thrown if an object parameter is null, and the JainSipParseException
is thrown for all other problems with the argument.</font></blockquote>

<p><br><font color="#3333FF">OK this jives and is easily defined.</font>
<blockquote TYPE=CITE>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<br>2. If the field is not set just return null</blockquote>
<font color="#FF0000">See above for null discussion,</font>
<blockquote TYPE=CITE>&nbsp;
<br>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>If the field is mandatory and does not exist, a JainSipParseException
will be thrown (header is malformed). If any header field is malformed
a JainSipParseException is thrown. If a JainSipParseException is thrown,
this indicates that the application must use the header's getValue method
to get a string representation of the header value and do what it can with
that.</blockquote>

<p><br>Why the headers getValue ? Why is the string not part of the Exception?</blockquote>
<font color="#FF0000">Sure - we can say that the JainSipParseException's
message must be the string value of the header.</font></blockquote>

<p><br><font color="#3333FF">OK Glad you buy that one.</font></blockquote>
<font color="#009900">I was just wondering what the message of a JainSipParseException
thrown from a set method should contain - the value passed in? Perhaps
we should add a string field to the JainSipParseException (say unparsable)
which can be whatever the parser was unable to parse? Then the message
can be left for explanatory comments.</font>
<blockquote TYPE=CITE><font color="#3333FF"></font>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;</blockquote>

<blockquote TYPE=CITE>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<p>It appears that what is being attempted here (please correct me if I
am wrong) is to use exceptions as a mechanism to catch un-initialized fields
so the parser can be invoked in the exception handler perhaps. This leads
to a lot of overhead in clunky code. I think it is a mistake to throw exceptions
on every get to try to protect the application from inadvertently accessing
fields that have not been initialized.&nbsp; Why not just assume that the
application must be smart enough to parse the header text before accessing
fields and not try to provide a complex mechanism as above?</blockquote>

<p><br>We are using ParameterNotSetException for the case of an optional
header not existing, and a JainSipParseException for the case of a malformed
header being received from the network.
<p>If we assume the application is smart enough to do all header value
parsing, we reduce a header to just being two strings - the type and value,
which puts the burden of parsing on an application - this work can be done
by the underlying implementation - doesn't it make more sense for an application
to catch an exception than have to parse all header value strings (especially
since this functionality already exists in the underlying stack)</blockquote>

<p><br><font color="#3333FF">No that is not what I had intended. Parser
functionality exists in the stack but has to be called explicitly by the
implementation. Too much hiding of the implementation from the application
has its own set of disadvantages (IMHO).</font></blockquote>
<font color="#FF0000">I wanted parser fuctionality to be invoked <i>implicitly</i>
by an application - from the API perspective it is an implementation detail
- so the application should not be aware of how or when the underlying
parser is invoked - it should just know that it is there in some form,
and that if it tries to get or set a value that the implementation doesn't
like it will get a JainSipParseException thrown back at it. This is important
if we want to maintain the three core values of JAIN - portability, portability
and portabilty ;)</font></blockquote>

<p><br><font color="#3333FF">The point about portability is a good one.
Of course the important thing about portability is to be able to maintain
the same semantics of exceptions as well.&nbsp; For an eager parser,&nbsp;
there is no possibility of throwing a parse&nbsp; exception&nbsp; on a
get (i.e. parse exception can never occur on a getXXX) so there will be
some curious code in a getXXX&nbsp; implementation&nbsp; for an eager parser
that has to throw an exception even when it does not need to (just to keep
the compiler happy).&nbsp;&nbsp; Second,&nbsp; the application has to be
willing to catch such an exception to be able to function both with a lazy
and an eager implementation in a transparent fashion and it may do different
things when the implementation is lazy, which exposes the underlying implementation
anyway as you have noted.</font>
<p><font color="#3333FF">Explicit control by the application may give the
application finer grained control and eliminate some awkward exception
handling that is&nbsp; not meaningful for an eager parser .&nbsp; This
way, you can also have mixed mode parsing, allowing the application to
decide what headers it wants to handle in a lazy fashion and which ones
it wants to handle in an eager fashion.</font></blockquote>
<font color="#009900">The application does not get to decide which method
is used by an implementation - an implementation could be lazy, eager or,
as you point out, a mixture of both. An application has to accomodate this
fact to be portable.</font>
<blockquote TYPE=CITE><font color="#3333FF"></font>&nbsp;
<p><font color="#3333FF">Another suggestion ( may be a bit radical) - would
it make sense to have extension interfaces (with enhanced getXXX&nbsp;
interfaces)&nbsp; that are only implemented by lazy parsers and make lazy
parsing optional (so only highly performance sensitive apps will&nbsp;
avail themselves of such interfaces whereas most applications will use
the eager parser interfaces for getXXX).&nbsp; ( May or may not make sense,
I thought I would mention it as it occured to me.&nbsp; )</font></blockquote>
<font color="#009900">Again, the application does not choose - the implementator
does.</font>
<blockquote TYPE=CITE><font color="#3333FF"></font>&nbsp;
<p><font color="#3333FF">Anyway,&nbsp; there is more ways than one to skin
a cat. Just keep everything as simple as possible but no simpler. I am
sure you are eager to bring this to closure so I will not belabor the point
about eager parsing versus lazy parsing&nbsp; although, as a final point
I would just like to state that portability can be handled whether or not
the parse is explicitly or implicitly lazy. Enough said on this point.</font></blockquote>
<font color="#009900">Thanks for all your input Ranga!</font>
<blockquote TYPE=CITE><font color="#3333FF"></font>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<p><font color="#3333FF">Regards,</font>
<p><font color="#3333FF">Ranga.</font>
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<pre>--&nbsp;
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.&nbsp;
Tel: 301 975 3664 Fax: 301 590 0932</pre>
&nbsp;</blockquote>

</body>
</html>

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Date: Tue, 28 Nov 2000 12:48:15 +0200


This is quite true that we should distinguish between the protocol and the
use made of it.

On the other hand, the sticker on the ladder does say "don't use this
ladder as scaffolding" because some uses for tools are inappropriate, even
if they are possible. \\For example, we can do some weird hacks with HTTP
or TLS to tunnel through firewalls, but people who are getting started with
these protocols vshould be informed that these are not the ideal uses of
the protocol. (The wonderful SIP-firewall-tunneling draft gets the balance
right, by in explaining a clever hack for SIP-over-TLS to port 443, while
also noting that this is an awful hack.)


> SIP is just a protocol. Thats it. Its a tool. If you
> want device A to call
> device B, it tells you how to do that. Like all tools,
> you can use them in
> many different ways. When you buy a hammer in Home
> Depot, does it tell you
> whether to use your left hand or right? How many
> fingers to grip with it?
> The size of the nails to use with it? Of course not.





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Subject: [SIP] call-flow; each user has 2 UAs
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Date: Tue, 28 Nov 2000 12:50:21 +0200

I'd like to ask a call-flow question. Excuse me if I asked a similar
question before, but perhaps I did not state it clearly earlier.

What happens if two people EACH have two UA's, each on a different media.
This is a reasonable scenario, since people may buy multiple devices or
applications, and system integrators may wish to integrate separate
applications without altering their software.

As a simple example, say that the two medias are audio and video (although
in real life, of course, an app/device with video will also likely have
audio).

Jack                         Jill
Audio  <-------------------------------------------------> Audio
Video  <-------------------------------------------------> Video

The important point is that the Audio and Video devices are NOT integrated
and have separate UA's, yet we want all four apps/devices to be joined in
one "session", with Jack's audio talking to Jill's audio, and Jack's video
talking to Jill's video. Perhaps we can assume that Jack's audio and video
both have the same SIP URL.

We need some sort of four-way session with proper use of SDP to distinguish
media types. To add two UA's on the UAS side, we need forking with
non-exclusive response from two UA's. We also have to decide how to bring
in two UA's on the UAC side.


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Subject: Re: [SIP] call-flow; each user has 2 UAs
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Date: Tue, 28 Nov 2000 08:44:36 -0600

Joshua_Fox@vocaltec.com (Joshua_Fox@vocaltec.com):

> As a simple example, say that the two medias are audio and video
> (although in real life, of course, an app/device with video will also
> likely have audio).
> 
> Jack                         Jill
> Audio  <-------------------------------------------------> Audio
> Video  <-------------------------------------------------> Video
> 
> The important point is that the Audio and Video devices are NOT
> integrated and have separate UA's, yet we want all four apps/devices
> to be joined in one "session", with Jack's audio talking to Jill's
> audio, and Jack's video talking to Jill's video. Perhaps we can assume
> that Jack's audio and video both have the same SIP URL.

  The session you describe is an abstraction layer above a SIP session.

  I see two ways of handling this:

        Remote               The first method is to have a SIP call
	  |  SIP             between the remote side and some
	Master               controller which communicates to the A and
       /      \  ?           V apps and incorporates their addresses
    Video    Audio           into a common SDP.  The master can be a
			     3pcc server if you use SIP to talk to A/V.

                             The second method is simply to have two
         Remote              calls up, and use some other means of
        |      | SIP         association higher up.  This may simply be
      Video   Audio          Jack telling Jill about the other session,
                             or some new header which flags the app
                             telling it to associate the two calls.

> We need some sort of four-way session with proper use of SDP to
> distinguish media types. To add two UA's on the UAS side, we need
> forking with non-exclusive response from two UA's. We also have to
> decide how to bring in two UA's on the UAC side.

  You don't need to get so complicated.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Howard Hart wrote:
> 
> Jonathan and Henning,
> 
>    I think this is a great shot at tunneling SIP and RTP through Firewalls,
> though you may want to add the following caveats to section 6:
> 
>     1) Wrt corporate tunneling, high-bandwidth voice traffic has a distinct
> network signature vs. (relatively) low-bandwidth Web traffic. If you blast
> voice traffic, encrypted or unencrypted over Web ports, it's going to stick out
> like a sore thumb even using the crudest network monitoring tools. ISS and the
> other network intrusion detection packages may even flag it as a Netbus,
> BackOrifice, or  Distributed Denial of Service attack signature. Most network
> administrators will shut the originating host/port down immediately and ask
> questions later.

I'm not sure whether that's a problem, as voice bandwidth is generally
much lower than web retrievals. (As low as 8 kb/s vs. a few hundred
kilo*bytes*/second on my desktop to many web servers. If I download a
piece of software, it will easily consume as much bandwidth as a phone
call: 3 minute call = 180 seconds = 180 kBytes, which is smaller than
many individual web pages.)

> 
>     2) The Internet backbone itself has been heavily optimized (some might say
> almost too optimized) for port 80 traffic, and is moving in that direction (as

This is bogus, except for the web caching that some ISPs (but,
fortunately, by no means all ISPs) do. Routers don't know about port 80.


> much as is feasible) for port 443/HTTPS. Increasingly, ISPs are using Web
> caching engines to limit this traffic across their backbones. Case in point--I
> used Checkpoint's encrypted VPN product in a previous company to connect an
> overseas location to our site (encrypted payloads, port numbers stay the same).
> Our remote site wanted to access our internal Web server. Unfortunately, one of
> the intermediate ISPs was using a Web caching product, which resulted in the
> caching engine intercepting, then attempting to originate an (unauthorized)
> port 80 access to our internal Web server on the remote end's behalf, with the
> expected (non)result. The answer from the ISP was, of course, "well it works
> fine for all our normal customers--EOM". I think you'd see some very
> interesting  side effects if you applied a tunneled RTP stream to this or other
> kinds of caching servers on port 80.
> Granted, 443/SSL-caching is a much more difficult problem, equating to a
> man-in-the- middle attack--maybe even unsolvable, but given the monetary
> inceptive is huge to come up with a solution here too, partial or otherwise....

Much of 443 traffic is inherently uncacheable, as it is for things like
forms responses from web orders. I doubt that even the looniest ISP
would want to return your airline booking confirmation page or bank
balance to me. Also, unless the cache has the certificate of the origin
server, intercept is just not possible. I doubt that Citibank, say,
would hand out their private key to random ISPs.

> 
> Howard Hart
> ipDialog, Inc.
> 

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Date: Mon, 27 Nov 2000 21:54:25 -0500


>   Regardless, lets just talk about the assumptions we're making about
>the capability of the clients.
>
>   I don't expect all clients to be able to support:
>
>   - sending audio to multiple locations, and
>   - mixing audio from multiple sources.
>
>   I don't think your draft should either.  Is that unreasonable?

Protocols should not be designed with the "probability of implementation" 
in mind when the motivation for those probabilities is are based on 
"assumptions" about likely products to be built under market conditions of 
the moment.  Remember Moore's law -- the DSP and memory resources available 
to vendors (of client products) will change much faster than the protocol 
changes.

Protocols should be designed to specify behavior in ways that are 
technically logical, defensible, as-minimal-as-should-be, and yet provide 
flexibility and extensiblility so that they can have a long life, 
especially if these behaviors "advance the ball" (vs. replicate the 
status-quo).  There should also be an understood, graceful fallback / 
failure mechanism for what behavior happens if certain protocol features 
are not implemented in a participating system.

IMHO, the requirements placed on clients by the multiparty conferencing I-D 
meet these requirements.

If implementors belive that implementing a component of the protocol falls 
outside the functional and market constraints for a product being offered 
at a particular moment in time (given target market, price point, etc.), 
the implementor can choose to act according to the fallback / failure 
mechanism.  This behavior simply becomes a momentary component of 
differentiation between competing products.

But protocol designers should not make consideration of this choice a 
determining factor for inclusion in the protocol.

If "likelihood of implementation" were a factor in the specification of 
HTTP, MIME, RFC822, etc., we would have a much less interesting world now 
than we do.  Design for the future -- not the present, keeping Moore's (and 
Gilder's) laws in mind.  Designing for the lowest common denominator is boring.

Cheers
-jb

----------
Jay Batson
Pingtel Corp.
jbatson@pingtel.com

781-938-5306 (office)
781-938-9650 (fax)


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From sip-admin@lists.bell-labs.com  Tue Nov 28 10:46:03 2000
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Dear all,

One thing which has been bothering me is having all the create methods
for messages and headers on the concrete JainSipFactory. These methods
all require that message and header implementations have an empty
constructor and then the JainSipFactory  invokes all the required set
methods with the create method's parameters.

I think that this could be quite restrictive for an implementor, and I
suggest that the create methods should be defined in interfaces rather
than the JainSipFactory class. The JainSipFactory could have
createJainSipMessageFactory and createJainSipHeaderFactory methods which
create particular implementations of JainSipMessageFactory and
JainSipHeaderFactory. [Or we could put these two methods in the
JainSipStack.] The implementations of these abstract factories are free
to use any constructors they like, and potentially could provide
proprietary optimisations such as object pooling. What do people think
about this?

Another thing about defining messages and headers as interfaces rather
than classes, is that the arguments to methods are defined in terms of
interfaces. Do we not really want to have an application mixing
different implementations? - I don't believe so. Then how can we enforce
that say XXX's implementation of the JainSipProvider can only take XXX's
implementations of the JAIN SIP interfaces as arguments? Use reflection
and throw an exception if the class of the object received doesn't match
the expected class? Again your opinions are welcomed

Regards,
Chris

PS - The Public Review period for the JAIN SIP Specification closes
today, so I would be grateful if you could send me any additional
comments ASAP. All going well, I hope to have the First Public Release
available at the SIP Bakeoff next month.


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Subject: [SIP] multiple calls for IP telephony client
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Date: Tue, 28 Nov 2000 17:19:28 +0100

While I was thinking about functionality of the User Agents for IP telephony. I got myself tangled in the following:

A call is one or more sessions ----> no problem, all sessions have the same Call_ID
This means that if A initiates a call, and B receives the call and B will then add 2 minutes later another session (a text session for relay purposes or a seperate video stream), it will be considered to be the same call? I know that SIP can just do this within the same call_ID via a re-invite.
But if B wants the text stream or video stream  
But if B's session is seen as a new call, does it mean that the IP telephony client is handling 2 seperate calls? I know that technically this does not matter for the client, but practically this may confuse the clients, since most people will deal with one call at the time and not simultaneously.

Maybe I am overlooking some things. I hope that you understand my question. :-)

A second question is:
Can the originator of a session be found in the re-invite?
If A calls B with session 1 and B comes then with session 2, the re-invite (adding session 2 to session 1) has then from: B@buffy.com?
even that originally the invite (session 1) was reading from: A@Angel.com (A started the call)?
I do believe so.

(been a long day:-)

Thanks for your time.


Arnoud

_______________________________________________________________
ERICSSON EUROLABS NETHERLANDS BV
Arnoud van Wijk
ABACUS Lab
Research & Development (ELN/V)
Fax: +31-161 247569
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From sip-admin@lists.bell-labs.com  Tue Nov 28 11:32:16 2000
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Date: Tue, 28 Nov 2000 10:31:40 -0600

  Our situation is as follows:

             A
            / \
           B   C

  A is in a 3-way call with B and C and wishes to be excused.  If C
called A, then A has no 'well-known URI' for C.  The best address is the
Contact.  The From is not useful because it may be an unknown URI
scheme, or C might be a 3pcc server, or the URI may be stale (or
fork)...

  In our implementation, we use the Contact exclusively.  I expect other
implementations will also generalize transfer with consultation by
instead implementing call handoff, and act similarily.  If we REFER B to
C and it fails, maybe we'll instead REFER C to B.  If some server really
needs to be in the signalling path, it should do 3pcc, or mangle the
Contact, or act as a transparent SIP proxy...

  Jonathan discussed why using the Contact is dangerous:

> The answer is that URIs most definitely have a context; a place from
> which they can be usefully used. In a perfect world with no NATs,
> firewalls, or devices that want to know whats going on within the
> network, it would work dandy. But thats not reality. The classic
> example is an ACD application, where help@helpdesk.com can be routed
> to a variety of agents (agent32@host32.helpdesk.com). Sophisticated
> software tracks the call status of the various agents, how much volume
> they get, etc. Incoming calls MUST come in through this helpdesk proxy
> to work. The proxy always record routes.  In fact, to enforce this,
> the proxy has access to a hosts file with the host32.helpdesk.com
> address mapped to an IP address. Outside of this proxy, that hostname
> is unroutable. So, if A called B, and then A called C, and C was one
> of these operators, the URI returned by C
> (agent32@host32.helpdesk.com) will definitely not be useful for B to
> use directly.

  With an ACD application, in order to enforce all calls through the
server, the server must perform 3pcc and only SIPly expose itself, or
have all of the clients collude with the server and redirect all
incoming direct-dialed calls.  I don't think a Record-Route'ing proxy
with some unroutable client addresses is clean, and I think it is
reasonable to have attended transfer fail in these scenarios.

  If the Contact address is not generally routable (firewalls and NATs),
then it should be mangled if the user wishes attended transfer to work,
or have calls go through some helper 3pcc box.  I see this as the only
reasonable solution.

> Perhaps these cases are so extreme its not worth worrying about. I
> wouldn't be surprised if they don't work in the PSTN either. 

  The PSTN has no concept of a REFER, which is where all this breaks
down.  Again, the 3-way call:

             A
            / \
           B   C

  A cannot excuse itself from the call in the PSTN.  It can only keep
the bridge active.  This is acceptable because A is a PBX or similar
server.  There is no concept of signalling a transfer over the PSTN
where the transferred party makes a new call.

> So, despite its complexity, I feel compelled to recommend that
> Robert's hybrid approach be described in the REFER specification as
> the preferred way to do transfer out of consultation, along with a
> nice long explanation of why it needs to be so. I'm not sure about the
> order though. I suspect that (2) will work in far more cases than (1)
> will. I welcome better ideas if anyone has any...

  It might be better to remove transfer out of consultation from the
REFER draft, and maybe do a new draft on attended transfer, discussing
these options and also introducing the replaces header from
draft-biggs-sip-replaces-00.

  I really want to be able to have A silently excuse itself from the
conference.  This will help with many services, including moving
transparently to a conference bridge, call park with server, ...

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Tue, 28 Nov 2000 12:11:28 -0500


--------------25E0FFD89B17F900BAF69875
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Hi All,

Need a clarification on the latest version of Manyfolks draft.

What should happen if the UAS receiving an Invite with confirm tag does
not have
the ability to do QoS. Should a Comet be sent out by the UAS in this
case ? The call has proceeded as Best effort and the 183 sent out by the
UAS had
no "a" line in its SDP body.

Thanks
Ameet

--
******************************
AMEET KHER, Software Engineer
Cisco Systems
Tel : (919) 392-8431



--------------25E0FFD89B17F900BAF69875
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Hi All,
<p>Need a clarification on the latest version of Manyfolks draft.
<p>What should happen if the UAS receiving an Invite with confirm tag does
not have
<br>the ability to do QoS. Should a Comet be sent out by the UAS in this
<br>case ? The call has proceeded as Best effort and the 183 sent out by
the UAS had
<br>no "a" line in its SDP body.
<p>Thanks
<br>Ameet
<pre>--&nbsp;
******************************
AMEET KHER, Software Engineer
Cisco Systems
Tel : (919) 392-8431</pre>
&nbsp;</html>

--------------25E0FFD89B17F900BAF69875--


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From: Billy Biggs <Billy_Biggs@3com.com>
To: Robert Sparks <rsparks@dynamicsoft.com>
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Subject: [SIP] Multiple-Transaction REFER
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Date: Tue, 28 Nov 2000 12:31:10 -0600

  As pointed out earlier by Jonathan, the single-transaction REFER
mechanism fails as the REFER may time out.  To clean things up, the
suggestion is to split up REFER into a sequence of transactions.

  To start discussion, here are some ideas:

1. REFER looks the same, but it is responded with a 2xx if the far end
   accepts the REFER and intends to act upon it.

2. A REFERDONE request which signals back that the spawned INVITE
   completed.  Success or failure is indicated in some header (or in the
   body?).  This request requires a header, Refer-CSeq or similar, to
   indicate which REFER transaction is being completed.

  Optionally:

3. A REFERSTATUS request which can signal back the call setup progress.
   Requires a Refer-CSeq header.  Optional support by both UAs.

4. A REFERCANCEL request which can attempt to cancel a pending REFER.
   Requires a Refer-CSeq header.  Optional support by both UAs.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Tue, 28 Nov 2000 10:17:44 +0530



Hi All !
     I had a query regarding SDP message bodies.

Is it possible to have multiple SDP bodies in a single SIP message. If so,
where would this be applicable ? (Could someone give a scenario/example)
Eg:
<SIP msg>
<SDP body 1>
<SDP body 2>
...
...
<SDP body n>

Thanks in advance
Subhash Nayak
Hughes Software Systems



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Hi Henning,

   See responses below:

"Henning G. Schulzrinne" wrote:

> Howard Hart wrote:
> >
> > Jonathan and Henning,
> >
> >    I think this is a great shot at tunneling SIP and RTP through Firewalls,
> > though you may want to add the following caveats to section 6:
> >
> >     1) Wrt corporate tunneling, high-bandwidth voice traffic has a distinct
> > network signature vs. (relatively) low-bandwidth Web traffic. If you blast
> > voice traffic, encrypted or unencrypted over Web ports, it's going to stick out
> > like a sore thumb even using the crudest network monitoring tools. ISS and the
> > other network intrusion detection packages may even flag it as a Netbus,
> > BackOrifice, or  Distributed Denial of Service attack signature. Most network
> > administrators will shut the originating host/port down immediately and ask
> > questions later.
>
> I'm not sure whether that's a problem, as voice bandwidth is generally
> much lower than web retrievals. (As low as 8 kb/s vs. a few hundred
> kilo*bytes*/second on my desktop to many web servers. If I download a
> piece of software, it will easily consume as much bandwidth as a phone
> call: 3 minute call = 180 seconds = 180 kBytes, which is smaller than
> many individual web pages.)

If the average Web page retrieval required 100+ kbytes/second, I don't think the
average user on their 28.8 baud, 3-5 kbytes/second (w/VJ compression) modem would do
much web surfing :-), but I get your point. My contention is divergence from Web
traffic signatures will be highly dependent on a number of cumulative factors, from
your locally provisioned bandwidth, to your usage patterns, to the number of times
you access the same server, to the sensitivity of your intrusion detection package,
to how long you stay on the phone, etc. If you throw in sustained, encrypted RTP
packet rate signatures at 25-50 pps over several minutes with the added burden that
only a small percentage of Web traffic is encrypted/port 443, it will probably
trigger the alarms, but your mileage will vary depending on the level of paranoia at
your local site.

>
>
> >
> >     2) The Internet backbone itself has been heavily optimized (some might say
> > almost too optimized) for port 80 traffic, and is moving in that direction (as
>
> This is bogus, except for the web caching that some ISPs (but,
> fortunately, by no means all ISPs) do. Routers don't know about port 80.

Agreed, my wording was poor. More accurately, ISPs are deploying web caching engines
at their edges to eliminate/optimize redundant HTTP/port 80 traffic over their
backbones, especially when they're being used for transit vs. direct access to Web
servers within their networks.

>
>
> > much as is feasible) for port 443/HTTPS. Increasingly, ISPs are using Web
> > caching engines to limit this traffic across their backbones. Case in point--I
> > used Checkpoint's encrypted VPN product in a previous company to connect an
> > overseas location to our site (encrypted payloads, port numbers stay the same).
> > Our remote site wanted to access our internal Web server. Unfortunately, one of
> > the intermediate ISPs was using a Web caching product, which resulted in the
> > caching engine intercepting, then attempting to originate an (unauthorized)
> > port 80 access to our internal Web server on the remote end's behalf, with the
> > expected (non)result. The answer from the ISP was, of course, "well it works
> > fine for all our normal customers--EOM". I think you'd see some very
> > interesting  side effects if you applied a tunneled RTP stream to this or other
> > kinds of caching servers on port 80.
> > Granted, 443/SSL-caching is a much more difficult problem, equating to a
> > man-in-the- middle attack--maybe even unsolvable, but given the monetary
> > inceptive is huge to come up with a solution here too, partial or otherwise....
>
> Much of 443 traffic is inherently uncacheable, as it is for things like
> forms responses from web orders. I doubt that even the looniest ISP
> would want to return your airline booking confirmation page or bank
> balance to me. Also, unless the cache has the certificate of the origin
> server, intercept is just not possible. I doubt that Citibank, say,
> would hand out their private key to random ISPs.

True, and as I stated above, maybe unsolvable, but for a sample of where pieces of
this might come into play later on, check out
http://www.f5.com/f5products/bigip/ecommerce, one of the more popular Web
load-balancers. The advantage here is you offload encryption and key negotiation
from individual servers in a large server farm and place the burden in hardware on
the load-balancer. The question then becomes an ISP/SysAdmin policy issue as to
whether you can locate your remote tunneling Proxy in front of or behind the
load-balancer, and what's the impact when the load-balancer assumes all traffic on
port 443 is HTTP only.

NOTE: This is obviously NOT an issue for your scheme today. Just something to
consider or maybe even take advantage of in the future.

Howard Hart
ipDialog, Inc.


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Date: Tue, 28 Nov 2000 16:39:03 -0500

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I would like to draw your attention to this new I-D. All comments and
questions are welcomed.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-hamer-sip-session-auth-00.txt

ABSTRACT

Current proposals dealing with authorization of media streams for 
multimedia services like IP telephony and video assume a pre- 
established relationship between elements of the network (e.g. 
session managers, edge routers, policy servers and end hosts). In 
some environments, however, such pre-established relationships may 
not exist either due to the complexity of creating these 
associations a priori (e.g. in a network with many elements), or due 
to the different business entities involved (e.g. service provider 
and access provider), or due to the dynamic nature of these 
associations (e.g. in a mobile environment). 
In this document, we describe scenarios where there is no pre-
established relationship between entities and describe mechanisms 
for exchanging information between network elements in order to 
authorize the use of resources for a service and to co-ordinate 
actions between the session and bearer control planes.


Regards,
Louis-Nicolas Hamer

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5.5.2652.35">
<TITLE>I-D ACTION:draft-hamer-sip-session-auth-00.txt</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2 FACE=3D"Arial">I would like to draw your attention to =
this new I-D. All comments and questions are welcomed.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">A URL for this Internet-Draft =
is:</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-hamer-sip-session-auth=
-00.txt" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-hamer-sip-se=
ssion-auth-00.txt</A></FONT></U>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">ABSTRACT</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Current proposals dealing with =
authorization of media streams for </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">multimedia services like IP telephony =
and video assume a pre- </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">established relationship between =
elements of the network (e.g. </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">session managers, edge routers, =
policy servers and end hosts). In </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">some environments, however, such =
pre-established relationships may </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">not exist either due to the =
complexity of creating these </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">associations a priori (e.g. in a =
network with many elements), or due </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">to the different business entities =
involved (e.g. service provider </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">and access provider), or due to the =
dynamic nature of these </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">associations (e.g. in a mobile =
environment). </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">In this document, we describe =
scenarios where there is no pre-</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">established relationship between =
entities and describe mechanisms </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">for exchanging information between =
network elements in order to </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">authorize the use of resources for a =
service and to co-ordinate </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">actions between the session and =
bearer control planes.</FONT>
</P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Times New Roman">Regards,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Times New Roman">Louis-Nicolas Hamer</FONT>
</P>

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</HTML>
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Chris,

A few closing comments on this string of exchanges (edited out all but
the relevant parts to keep the reading simple).

Regards

Ranga.

  ------------------------------------------------------------------------

>> A design issue not discussed below:
>>
>> Whenever there is more than one header to be returned (via  headers
>> for example), you return an array.  An array is easy to walk
>> through. However,  I would think it more useful to return a
>> collection of some sort that supports additional methods such as
>> checking if an address exists in the list of headers returned etc.
>> That might be of some use to an implementation.  If you define the
>> returned type as an array of headers, there is no way to extend the
>> class to support additional functionality.  May I suggest adding a
>> collection class interface for collections of SIP Headers (and that
>> can be specialized by implementations to support additional query
>> methods etc.), say something like SIPHeaderList
>
> I initially wanted to return a Vector, but was told that an array was
> preferable for API's. Besides, any addition of functionality would be
> purely for the implementor's benefit and would not be available
> (without using reflection anway) to the application.

My point was that if you returned an array, you could not extend it so
IMHO,  Vector would work better (you can extend  the class and add
whatever functionality you want). This would make an implementer's life
easier and the implementation more elegant.


>> >
>>
>> Ah! My understanding was correct then.
>>
>> OK. Perhaps this would be too much of a change for you to consider
>> but let me make a suggestion anyway ....
>>
>> I think formatting headers should be considered under a different
>> set of API.  Here is what I am suggesting
>>
>> 1. Get rid of the setXXX interfaces (the parser will have direct
>> access to the classes and does not need them and it will cut down
>> the size of the API).
>>
>> 2. Have a  SIPMessageFormatter interface who'se sole job it is to
>> format output messages.  This can have methods to format the
>> different outgoing messages and will throw parse exceptions if the
>> strings that it gets are incorrect. The MessageFormatter interface
>> takes as parameter values only Strings.
>>
>> Concretely,  for example:
>>
>> public interface SIPMessageFormatter {
>>         public String createFromMessage( String from ,  String tag)
>> throws SIPParseException;
>>
>>     .....................
>>
>> }
>>
>> It would be slower for an implementation to extract the same
>> information out of a structure and hand it to a parser for
>> validation (IMHO). A "loose implementation" may not want to be
>> bothered with validation at all and leave it to the reciever to
>> reject the mesage, thereby making the implementation faster or it
>> may do very cursory checking.
>>
>
> This is a major change at this stage (public review technically closes
> today)  - I personally don't think it's very OO, but I would be
> interested to hear other people's opinons.

I agree that it is maybe not OOcentric and that it is a late change but
it  does follow the end to end argument:  i.e.  the sender does not
necessarily have to check for correctness of the input  because it can
rely on the receiver doing so.  I think this is a strong argument in its
favor.  It also cuts down on the number of APIs substantially and
removes the overhead of formatting inputs to Strings  from other types
of inputs and checking these with the parser. I think it can simplify
things and give you the functionality you need.




>> OK Glad you buy that one.
>
> I was just wondering what the message of a JainSipParseException
> thrown from a set method should contain - the value passed in? Perhaps
> we should add a string field to the JainSipParseException (say
> unparsable) which can be whatever the parser was unable to parse? Then
> the message can be left for explanatory comments.

I think the portion of the header that the set method was unable to
parse would be the relevant information to return. However, if you buy
into eliminating setXXX methods and having a MessageFormatter instead,
the point is moot :)

Regards
Ranga

--
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
Tel: 301 975 3664 Fax: 301 590 0932





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<html>
<body text="#000000" bgcolor="#FFFFFF" link="#0000FF" vlink="#FF0000" alink="#000088">
&nbsp;
<br>&nbsp;
<p><font color="#3333FF">Chris,</font>
<p><font color="#3333FF">A few closing comments on this string of exchanges
(edited out all but the relevant parts to keep the reading simple).</font>
<p><font color="#3333FF">Regards</font>
<p><font color="#3333FF">Ranga.</font>
<p>
<hr WIDTH="100%">
<blockquote TYPE=CITE>
<blockquote TYPE=CITE><font color="#3333FF">A design issue not discussed
below:</font>
<p><font color="#3333FF">Whenever there is more than one header to be returned
(via&nbsp; headers for example), you return an array.&nbsp; An array is
easy to walk through. However,&nbsp; I would think it more useful to return
a collection of some sort that supports additional methods such as checking
if an address exists in the list of headers returned etc. That might be
of some use to an implementation.&nbsp; If you define the returned type
as an array of headers, there is no way to extend the class to support
additional functionality.&nbsp; May I suggest adding a collection class
interface for collections of SIP Headers (and that can be specialized by
implementations to support additional query methods etc.), say something
like <i>SIPHeaderList</i></font></blockquote>
<font color="#009900">I initially wanted to return a Vector, but was told
that an array was preferable for API's. Besides, any addition of functionality
would be purely for the implementor's benefit and would not be available
(without using reflection anway) to the application.</font></blockquote>

<p><br><font color="#3333FF">My point was that if you returned an array,
you could not extend it so IMHO,&nbsp; Vector would work better (you can
extend&nbsp; the class and add whatever functionality you want). This would
make an implementer's life easier and the implementation more elegant.</font>
<br>&nbsp;
<blockquote TYPE=CITE>
<blockquote TYPE=CITE>
<blockquote TYPE=CITE>&nbsp;</blockquote>
<font color="#3333FF">Ah! My understanding was correct then.</font>
<p><font color="#3333FF">OK. Perhaps this would be too much of a change
for you to consider but let me make a suggestion anyway ....</font>
<p><font color="#3333FF">I think formatting headers should be considered
under a different set of API.&nbsp; Here is what I am suggesting</font>
<p><font color="#3333FF">1. Get rid of the setXXX interfaces (the parser
will have direct access to the classes and does not need them and it will
cut down the size of the API).</font>
<p><font color="#3333FF">2. Have a&nbsp; SIPMessageFormatter interface
who'se sole job it is to format output messages.&nbsp; This can have methods
to format the different outgoing messages and will throw parse exceptions
if the strings that it gets are incorrect. The MessageFormatter interface
takes as parameter values only Strings.</font>
<p><font color="#3333FF">Concretely,&nbsp; for example:</font>
<p><font color="#3333FF">public interface SIPMessageFormatter {</font>
<br><font color="#3333FF">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; public
String createFromMessage( String from ,&nbsp; String tag) throws SIPParseException;</font>
<p><font color="#3333FF">&nbsp;&nbsp;&nbsp; .....................</font>
<p><font color="#3333FF">}</font>
<p><font color="#3333FF">It would be slower for an implementation to extract
the same information out of a structure and hand it to a parser for validation
(IMHO). A "loose implementation" may not want to be bothered with validation
at all and leave it to the reciever to reject the mesage, thereby making
the implementation faster or it may do very cursory checking.</font>
<br>&nbsp;</blockquote>
<font color="#009900">This is a major change at this stage (public review
technically closes today)&nbsp; - I personally don't think it's very OO,
but I would be interested to hear other people's opinons.</font></blockquote>

<p><br><font color="#3333FF">I agree that it is maybe not OOcentric and
that it is a late change but it&nbsp; does follow the end to end argument:&nbsp;
i.e.&nbsp; the sender does not necessarily have to check for correctness
of the input&nbsp; because it can rely on the receiver doing so.&nbsp;
I think this is a strong argument in its favor.&nbsp; It also cuts down
on the number of APIs substantially and removes the overhead of formatting
inputs to Strings&nbsp; from other types of inputs and checking these with
the parser. I think it can simplify things and give you the functionality
you need.</font>
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>
<blockquote TYPE=CITE><font color="#3333FF">OK Glad you buy that one.</font></blockquote>
<font color="#009900">I was just wondering what the message of a JainSipParseException
thrown from a set method should contain - the value passed in? Perhaps
we should add a string field to the JainSipParseException (say unparsable)
which can be whatever the parser was unable to parse? Then the message
can be left for explanatory comments.</font></blockquote>
<font color="#3333FF">I think the portion of the header that the set method
was unable to parse would be the relevant information to return. However,
if you buy into eliminating setXXX methods and having a MessageFormatter
instead, the point is moot :)</font>
<p><font color="#3333FF">Regards</font>
<br><font color="#3333FF">Ranga</font>
<pre>--&nbsp;
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.&nbsp;
Tel: 301 975 3664 Fax: 301 590 0932</pre>
&nbsp;
<p>&nbsp;
</body>
</html>

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Chris Harris wrote:

> Dear all,
>
> One thing which has been bothering me is having all the create methods
> for messages and headers on the concrete JainSipFactory. These methods
> all require that message and header implementations have an empty
> constructor and then the JainSipFactory  invokes all the required set
> methods with the create method's parameters.

There is probably a very good reason for these API but I have not
understood the motivation . How do you envision these API being used (When
would an application  want to create Headers in this fashion?  Are these to
be interfaces to a parser?  )

>
>
> I think that this could be quite restrictive for an implementor, and I
> suggest that the create methods should be defined in interfaces rather
> than the JainSipFactory class.

> The JainSipFactory could have
> createJainSipMessageFactory and createJainSipHeaderFactory methods which
> create particular implementations of JainSipMessageFactory and
> JainSipHeaderFactory. [Or we could put these two methods in the
> JainSipStack.] The implementations of these abstract factories are free
> to use any constructors they like, and potentially could provide
> proprietary optimisations such as object pooling. What do people think
> about this?

If you need to create Headers in this fashion, I would agree with your
suggestion above.

>
>
> Another thing about defining messages and headers as interfaces rather
> than classes, is that the arguments to methods are defined in terms of
> interfaces. Do we not really want to have an application mixing
> different implementations? - I don't believe so.

In the utopian world of true portability, this would work but in reality, I
suppose this would be a bad idea.



> Then how can we enforce
> that say XXX's implementation of the JainSipProvider can only take XXX's
> implementations of the JAIN SIP interfaces as arguments? Use reflection
> and throw an exception if the class of the object received doesn't match
> the expected class? Again your opinions are welcomed

Reflection would be a good way of checking this (or one could use something
like a UUID I suppose).


>
>
> Regards,
> Chris
>
> PS - The Public Review period for the JAIN SIP Specification closes
> today, so I would be grateful if you could send me any additional
> comments ASAP. All going well, I hope to have the First Public Release
> available at the SIP Bakeoff next month.
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

--
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.
Tel: 301 975 3664 Fax: 301 590 0932



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Chris Harris wrote:
<blockquote TYPE=CITE>Dear all,
<p>One thing which has been bothering me is having all the create methods
<br>for messages and headers on the concrete JainSipFactory. These methods
<br>all require that message and header implementations have an empty
<br>constructor and then the JainSipFactory&nbsp; invokes all the required
set
<br>methods with the create method's parameters.</blockquote>

<p><br>There is probably a very good reason for these API but I&nbsp;have
not understood the motivation . How do you envision these API&nbsp;being
used (When would an application&nbsp; want to create Headers in this fashion?
&nbsp;Are these to be interfaces to a parser?&nbsp; )
<blockquote TYPE=CITE>&nbsp;
<p>I think that this could be quite restrictive for an implementor, and
I
<br>suggest that the create methods should be defined in interfaces rather
<br>than the JainSipFactory class.</blockquote>

<blockquote TYPE=CITE>The JainSipFactory could have
<br>createJainSipMessageFactory and createJainSipHeaderFactory methods
which
<br>create particular implementations of JainSipMessageFactory and
<br>JainSipHeaderFactory. [Or we could put these two methods in the
<br>JainSipStack.] The implementations of these abstract factories are
free
<br>to use any constructors they like, and potentially could provide
<br>proprietary optimisations such as object pooling. What do people think
<br>about this?</blockquote>

<p><br>If you need to create Headers in this fashion, I&nbsp;would agree
with your suggestion above.
<blockquote TYPE=CITE>&nbsp;
<p>Another thing about defining messages and headers as interfaces rather
<br>than classes, is that the arguments to methods are defined in terms
of
<br>interfaces. Do we not really want to have an application mixing
<br>different implementations? - I don't believe so.</blockquote>

<p><br>In the utopian world of true portability, this would work but in
reality, I suppose this would be a bad idea.
<br>&nbsp;
<br>&nbsp;
<blockquote TYPE=CITE>Then how can we enforce
<br>that say XXX's implementation of the JainSipProvider can only take
XXX's
<br>implementations of the JAIN SIP interfaces as arguments? Use reflection
<br>and throw an exception if the class of the object received doesn't
match
<br>the expected class? Again your opinions are welcomed</blockquote>

<p><br>Reflection would be a good way of checking this (or one could use
something like a UUID I&nbsp;suppose).
<br>&nbsp;
<blockquote TYPE=CITE>&nbsp;
<p>Regards,
<br>Chris
<p>PS - The Public Review period for the JAIN SIP Specification closes
<br>today, so I would be grateful if you could send me any additional
<br>comments ASAP. All going well, I hope to have the First Public Release
<br>available at the SIP Bakeoff next month.
<p>_______________________________________________
<br>SIP mailing list
<br>SIP@lists.bell-labs.com
<br><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></blockquote>

<pre>--&nbsp;
M.Ranganathan
NIST Advanced Networking Technologies Group,
100 Bureau Drive, Stop 8920, Gaithersburg, MD 20899.&nbsp;
Tel: 301 975 3664 Fax: 301 590 0932</pre>
&nbsp;
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Subject: [SIP] RECORD-ROUTE/ROUTE requirements
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This discussion seems endless 8-). 

There are a lot of implementation details discussed.
What I wonder is what the requirements are on ROUTE
RECORD-ROUTE. What's the "really" purpose with these 
headers ? What do they have to support ?

If we can agree on the requirements for these
headers maybe it's easier to find the best way
to "implement" this in the specification ?

The main purpose with RECORD-ROUTE/ROUTE as I 
understand it is that a server can make shure it will 
get all new requests for a certain call-leg by adding 
a RECORD-ROUTE to the first (and consequtive) request. 

The first requirement on ROUTE is that it has to include 
an address so that a SIP server (or client) can send the
next request to the correct server. The first issue I
see is how specific this address should be ? 

Should it specify a specific host (like maddr in a
SIP-URL) or is it enough to find one of the hosts
in the server and then it's up to the implementation
of that server to find the right host for this call ?

The way I see it is that this is implementation
dependent. It's up to each server implementation to
decide how specific the address is that it puts into
RECORD-ROUTE/ROUTE. So depending on how robust a 
server wants to be it might implement this in 
different ways. Should the specification require
a certain robustness in a server ?

The next issue I see is port number. Today you have 
to use the same port number from both caller and callee
because the same ROUTE is used in both directions.
Should it be possible to use different port numbers ?
I would say yes to this. If a server wants to have
different port numbers it should have the possibility
to do so.

A similar issue is transport. Should it be possible to
have different transport types in the two directions ?
If a proxy is doing transport conversion I think it
would be nice if you can specify the different types
dependent on direction of the request.

    ----     ----     ----     ----  
---| P1 |---| P2 |---| P3 |---| P4 |---
    ----     ----     ----     ----  

What I mean by this is that if P3 uses UDP towards P2 
and TCP towards P4 maybe it wants to tell P2 and P4
what they should use in the next request. Of course
P2 and P4 could remember this but if e.g. P2 does
not RECORD-ROUTE P1 will not know what transport to 
use towards P3 unless this is said in ROUTE.

Another issue is identification. A server have to be
able to put some identification in RECORD-ROUTE/ROUTE
so that when the next request arrives it can find which
call states to use for this request. Certain server
implementations maybe find it more optimal to put
the state information in the message instead of an
identifier. So I beleave both possibilities needs to
be supported.

Have I forgot something ??

So from this reasoning I draw the conclusion that a
server needs the possibility to put in an address 
(host + port), a transport and an identification/state 
in RECORD-ROUTE/ROUTE. Since the address and transport
part have to be used by other servers (clients) this
information have to be detailed specified in the SIP
specification so that each server knows where to find
these parts. 
How the identification/state part looks like is 
implementation dependent and does not need to be 
as detailed as the other parts. It only need to be so
detailed so that the identification/state part can be 
sent to the next server by the previous server in the 
ROUTE chain. How a server use this part is up to each 
server and don't need to be specified (maybe as examples).  

So if I now go down to details I see different ways of
"implementing" the "requirements" in the specification.

I think I can atleast say that the identification part
MUST be a URI since it's going to be sent to the next
server as a request URI. But what a server have put
into the URI is up to him. 

The more controversial parts I guess is the address 
and the transport. Should they be RECORD-ROUTE/ROUTE 
parameters or URI parameters ? Personally I prefere
RECORD-ROUTE/ROUTE parameters because that means I
don't have to look inside the ROUTE-URI and that makes 
me more future proof towards possibly new types of URIs. 
The other possibility that has been discussed is to
only allow SIP-URL as URI in ROUTE and put the address 
in there as today (maddr). I don't like this solution 
because if a new type of URI pops up in the future that 
a server needs to put in as identification it can't do
it directly. Instead it has to convert it into a SIP-URL 
in some way.

The possibility to have different portnumbers and/or
transport in the two directions I beleave can be solved
as Jonathan described. _IF_ a server wants to use different
address (port) or transport in the two directions it has
to modify it's own RECORD-ROUTE when the response is sent
back. Since this can be done already today I wonder if
the specification needs to say much about this (maybe as
examples). However I see a risk in this behaviour and that
is that a server might modify the wrong RECORD-ROUTE by
mistake and it will be very tricky to find which server
it was that did the modification unless you have logs
from all servers.

That's all for now folks. Maybe someone should set up
a discussion about this during the bakeoff ??


-- 
Bertil Engelholm
AXE Research and Development        voice : +46 8 727 3499
SIP Security                        Fax   : +46 8 647 8276
S-126 25 Stockholm Sweden           E-mail:
Bertil.Engelholm@uab.ericsson.se

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Date: Wed, 29 Nov 2000 06:31:49 -0500

--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: SIP Session Timer
	Author(s)	: S. Donovan, J. Rosenberg
	Filename	: draft-ietf-sip-session-timer-04.txt
	Pages		: 24
	Date		: 28-Nov-00
	
This document proposes an extension to the Session Initiation
Protocol (SIP). This extension allows for a periodic refresh of SIP
sessions through a re-INVITE. The refresh allows both user agents and
proxies to determine if the SIP session is still active. The
extension defines a new general header, Session-Expires, which
conveys the lifetime of the session.

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From sip-admin@lists.bell-labs.com  Wed Nov 29 09:38:50 2000
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Date: Wed, 29 Nov 2000 06:31:53 -0500

--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: SIP Telephony Call Flow Examples
	Author(s)	: A. Johnston, S. Donovan, R. Sparks, 
                          C. Cunningham, D. Willis, J. Rosenberg,
                          K. Summers, H. Schulzrinne
	Filename	: draft-ietf-sip-call-flows-02.txt
	Pages		: 202
	Date		: 28-Nov-00
	
This document gives examples of SIP (Session Initiation Protocol)
call flows for IP telephony.  Elements in these
call flows include SIP User Agents and Clients, SIP Proxy and
Redirect Servers, and Gateways to the PSTN (Public Switch Telephone
Network).  IP telephony scenarios include SIP Registration, SIP to
SIP calling, SIP to Gateway, Gateway to SIP, and Gateway to Gateway
via SIP.  Call flow diagrams and message details are shown.  PSTN
telephony protocols are illustrated using ISDN (Integrated Services
Digital Network), ANSI ISUP (ISDN User Part), and FGB (Feature Group
B) circuit associated signaling.  PSTN calls are illustrated using
global telephone numbers from the PSTN and private extensions served
on by a PBX (Private Branch Exchange).  Example SIP messages used for
testing during SIP 'bakeoff' events include SIP 'torture test'
messages, and messages with invalid parameters, methods, and tags.

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From sip-admin@lists.bell-labs.com  Wed Nov 29 09:46:10 2000
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From: "Jo Hornsby" <jhornsby@ubiquity.net>
To: "Bertil Engelholm" <Bertil.Engelholm@uab.ericsson.se>,
        <SIP@lists.bell-labs.com>
Subject: RE: [SIP] RECORD-ROUTE/ROUTE requirements
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> This discussion seems endless 8-). 

Indeed.  And there I was hoping that we would have this
sorted for the Bake-Off.  Oh well.

> The main purpose with RECORD-ROUTE/ROUTE as I 
> understand it is that a server can make shure it will 
> get all new requests for a certain call-leg by adding 
> a RECORD-ROUTE to the first (and consequtive) request. 

Exactly.  And that is why I think we should mandate (i.e.,
MUST) that it is a SIP URL with either no transport
parameter, or a transport parameter that matches /UDP/i.
If these conditions are not met, then we cannot guarantee
that a client will be able to route successfully.

As an analogy, consider the fact that UDP is a MUST.
I wouldn't be surprised if someone writes something that
doesn't do UDP, because TLS is essential (for some reason).
Now they can't expect to be able to interoperate, but
doubtless that won't matter, since the parameters of
the network this non-UDP-supporting-something is going
to be running in will ensure that it can interoperate
with whatever it needs to.

In the same way, someone might one day insert a non-SIP
URI into a Record-Route, but they can't expect it to
work unless they know that the devices in the network
are controlled such that they'll understand this non-SIP
URI.

> The first requirement on ROUTE is that it has to include 
> an address so that a SIP server (or client) can send the
> next request to the correct server. The first issue I
> see is how specific this address should be ? 

It MUST be specific enough such that 1.4.2 lookup
procedures _and nothing else_ will resolve to that same
(or similar in a farm, or whatever) proxy.

> The next issue I see is port number. Today you have 
> to use the same port number from both caller and callee
> because the same ROUTE is used in both directions.
> Should it be possible to use different port numbers ?
> I would say yes to this. If a server wants to have
> different port numbers it should have the possibility
> to do so.

Well, if a proxy is prepared to modify the Record-Route
in the response, then it is free to expose a different
port to callee/caller.

> A similar issue is transport. Should it be possible to
> have different transport types in the two directions ?

No, it MUST be UDP.

> Another issue is identification. A server have to be
> able to put some identification in RECORD-ROUTE/ROUTE
> so that when the next request arrives it can find which
> call states to use for this request. Certain server
> implementations maybe find it more optimal to put
> the state information in the message instead of an
> identifier. So I beleave both possibilities needs to
> be supported.

Indeed.  But one by no means rules out the other.

> Have I forgot something ??

No, I think you got it pretty much covered. &:)

> The more controversial parts I guess is the address 
> and the transport. Should they be RECORD-ROUTE/ROUTE 
> parameters or URI parameters ? Personally I prefere
> RECORD-ROUTE/ROUTE parameters because that means I
> don't have to look inside the ROUTE-URI and that makes 
> me more future proof towards possibly new types of URIs. 
>
> The other possibility that has been discussed is to
> only allow SIP-URL as URI in ROUTE and put the address 
> in there as today (maddr). I don't like this solution 
> because if a new type of URI pops up in the future that 
> a server needs to put in as identification it can't do
> it directly. Instead it has to convert it into a SIP-URL 
> in some way.

If a proxy needs to embed exotic URIs (or whatever) in
Record-Route, then I think it should have to suffer all
the pain involved.

If we use anything other than UDP SIP URLs, I can't see
how we can guarantee interoperability.

Cheers,


 - Jo.


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Folks,

I wanted to call your attention to this updated 3pcc document. It deals with
implementation issues that have come up since the publication of the
original draft. Note that, specifically, it no longer recommends carrying
SDP in the ACK, but rather using a re-INVITE. It talks about QoS
integration, as discussed by Gonzalo at past meetings. It also includes some
notes at the end on SIP features that you really, really, *really* want to
support so that your phone, PC, or whatever, can receive features and
services provided by a controller. 

As such, I would strongly recommend that people who build UAs take a look at
this draft, and plan on supporting the features discussed there. If anyone
has questions on the value of doing this, feel free to contact me.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

-----Original Message-----
From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
Sent: Friday, November 24, 2000 6:01 PM
To: IETF-Announce
Subject: I-D ACTION:draft-rosenberg-sip-3pcc-01.txt


A New Internet-Draft is available from the on-line Internet-Drafts
directories.


	Title		: Third Party Call Control in SIP
	Author(s)	: J. Rosenberg, J. Peterson, H. Schulzrinne, 
                          G. Camarillo
	Filename	: draft-rosenberg-sip-3pcc-01.txt
	Pages		: 15
	Date		: 22-Nov-00
	
This document discusses the usage of the Session Initiation Protocol
(SIP) for third party call control. Third party call control refers
to the ability of one entity to create a call in which communications
is actually between other parties. We present a SIP mechanism for
accomplishing third party call control that does not require any
extensions or changes to SIP.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-rosenberg-sip-3pcc-01.txt

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Date: Wed, 29 Nov 2000 16:43:20 +0000



Is it now generally agreed that the requirement in 6.43.2 for clients to set up
the Route name-addr using Contact/From headers (excluding maddr and port) should
be removed ?  This seems to be the fundamental problem.

K . Robinson



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From: Billy Biggs <Billy_Biggs@3com.com>
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Date: Wed, 29 Nov 2000 10:47:02 -0600

  The sip-peer-3pcc draft discusses providing PHONECTL functionality
using REFER.

  To send a BYE on a specific call leg, the draft uses a Refer-To URI
of: <sip:bob@foo.com;method=BYE?i=102@phone.foo.com>

  To indicate a request to REGISTER as a specific user, the draft
uses: <sip:user@foo.com;method=REGISTER>

  I strongly believe that this sort of behavior is outside the scope of
REFER.  The meaning of the request is hidden in an obfuscated URI, which
cannot provide enough information to identify which call-leg is being
referenced. [1]

  I am rewriting the PHONECTL draft, splitting the commands into new
methods.  For example, LOGIN (ask the UA to REGISTER) and HANGUP (ask
the UA to leave a session).  This keeps the intent clear, and helps the
UA maintain a seperate authentication model for these device control
extensions.

  I hope to have a draft ready this week, but would be willing to set up
a mailing list or something if there was interest in this work.


[1] The original INVITE may have resulted in two calls being setup, and
    no mechanism of determining the tags used is available.  As well,
    you cannot specify which tags to use in a URI.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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> Is it now generally agreed that the requirement in 6.43.2 for 
> clients to set up the Route name-addr using Contact/From
> headers (excluding maddr and port) should be removed ?  This
> seems to be the fundamental problem.

I think so.

*duck and cover*

&:)


 - Jo.


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From sip-admin@lists.bell-labs.com  Wed Nov 29 12:01:34 2000
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From: "Robert Sparks" <rsparks@dynamicsoft.com>
To: "SIP List" <sip@lists.bell-labs.com>
Subject: RE: [SIP] Multiple-Transaction REFER
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Lets revisit the root of the time-out problem mentioned below.

REFER as currently written waits until the referenced action
completes before returning its final response. This runs into
trouble if any of the participating UAs (including any proxies
in the path of the request) decide the request has been pending
too long and do whatever they do on a time-out.

I've been reviewing 10.4.1 and 10.4.2 and currently I can't see
where that time-out (for non-INVITE messages) is defined by the
protocol. It seems to be left up to the application.

10.4.1 specifies when the request will stop being retransmitted
in absence of a final response, but doesn't say when the client
gives up on the request after the last retransmission.

10.4.2 even more explicitly leaves this in the applications hands:
  Clients using a reliable transport protocol such as TCP, SCTP
  or TLS do not need to retransmit requests, but MAY give up
  after receiving no response for an extended period of time.

Perhaps we need clarification of behavior here. Would it be worth
standardizing a default interpretation of Expires (using the
INVITE behavior) for all requests where it doesn't have specialized
meaning (like REGISTER).

If we went down that path, we could also recommend behavior for how
long to wait after that last retransmission. In particular, recommend
extending that timeout in the face of provisional responses.

There are places other than REFER where this issue will cause grief.

The response to the sip-events SUBSCRIBE request might need time to
obtain authorizations. Processing QAUTH in sip-for-presence may involve
user interaction.

Comments?

RjS



> -----Original Message-----
> From: sip-admin@lists.bell-labs.com
> [mailto:sip-admin@lists.bell-labs.com]On Behalf Of Billy Biggs
> Sent: Tuesday, November 28, 2000 12:31 PM
> To: Robert Sparks
> Cc: SIP List
> Subject: [SIP] Multiple-Transaction REFER
> 
> 
>   As pointed out earlier by Jonathan, the single-transaction REFER
> mechanism fails as the REFER may time out.  To clean things up, the
> suggestion is to split up REFER into a sequence of transactions.
> 
>   To start discussion, here are some ideas:
> 
> 1. REFER looks the same, but it is responded with a 2xx if the far end
>    accepts the REFER and intends to act upon it.
> 
> 2. A REFERDONE request which signals back that the spawned INVITE
>    completed.  Success or failure is indicated in some header (or in the
>    body?).  This request requires a header, Refer-CSeq or similar, to
>    indicate which REFER transaction is being completed.
> 
>   Optionally:
> 
> 3. A REFERSTATUS request which can signal back the call setup progress.
>    Requires a Refer-CSeq header.  Optional support by both UAs.
> 
> 4. A REFERCANCEL request which can attempt to cancel a pending REFER.
>    Requires a Refer-CSeq header.  Optional support by both UAs.
> 
> -- 
> Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com
> 
> _______________________________________________
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> 

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Date: Wed, 29 Nov 2000 11:09:44 -0600

  In sip-service-examples-00, you discuss performing music on hold using
REFER.  A shortened call flow is below:

  On hold:

         User A        User B       Music Server
          |              |              |
          |   REFER Refer-To: sip:music@server.com
          |<-------------|              |
          | INVITE/200 OK/ACK           |
          |<--------------------------->|
          |  200 OK      |              |
          |------------->|              |
          |              |              |

  Off hold:

          |              |              |
          |   REFER Refer-To: music@server.com;method=BYE
          |<-------------|              |
          |  BYE/200 OK  |              |
          |<--------------------------->|
          |  200 OK      |              |
          |------------->|              |
          |              |              |

  I have two problems with this method:

  1. In our implementation, User A will hang up the call to User B after
     the REFER completes successfully.  This is to protect ourselves
     (User B will also send a BYE), however, I can imagine most
     implementations will be similar.

  2. As mentioned in my previous email on remote device control using
     REFER, the URI music@server.com;method=BYE is insufficient to
     indicate to the UA which call should be disconnected.  It's also
     overloading REFER and obfuscating its meaning.

  There are more appropriate ways to implement music on hold.  One is to
have User B call up the music server itself and swap the SDP (3pcc).
Another is to have User A call up the music on hold server when it
receives on-hold SDP.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Wed, 29 Nov 2000 11:20:02 -0600

Robert Sparks (rsparks@dynamicsoft.com):

> I've been reviewing 10.4.1 and 10.4.2 and currently I can't see where
> that time-out (for non-INVITE messages) is defined by the protocol. It
> seems to be left up to the application.
> 
> 10.4.1 specifies when the request will stop being retransmitted in
> absence of a final response, but doesn't say when the client gives up
> on the request after the last retransmission.

  If the retransmissions stop, there is no way to reliably receive the
response.

> Perhaps we need clarification of behavior here. Would it be worth
> standardizing a default interpretation of Expires (using the INVITE
> behavior) for all requests where it doesn't have specialized meaning
> (like REGISTER).

  I like Expires how it is.  For SUBSCRIBE, Expires dictates how long
the subscription should last.  I'm sure new extensions will have uses
for Expires.

  In my multi-transaction proposal, an expires in the original REFER
could mean the time in which it must receive a REFERDONE.  This seems
more clean.

> There are places other than REFER where this issue will cause grief.
> 
> The response to the sip-events SUBSCRIBE request might need time to
> obtain authorizations. Processing QAUTH in sip-for-presence may
> involve user interaction.

  I don't like having transactions outstanding for too long.  The
retransmission model breaks for everything except INVITE.  If half a
minute doesn't seem reasonable for these authorizations or user
interaction, then the process must be extended to multiple transactions.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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From: Billy Biggs <Billy_Biggs@3com.com>
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Date: Wed, 29 Nov 2000 11:24:18 -0600

Jo Hornsby (jhornsby@ubiquity.net):

>> Is it now generally agreed that the requirement in 6.43.2 for clients
>> to set up the Route name-addr using Contact/From headers (excluding
>> maddr and port) should be removed ?  This seems to be the fundamental
>> problem.
> 
> I think so.

  I think so too.


  That said, I have always thought that this line was a bug: "Each such
proxy server adds the Request-URI of the _incoming_request_ to the
beginning of the list."

  s/incoming request/proxy/

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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Date: Wed, 29 Nov 2000 12:52:04 -0600

Lyndon,

Was it the word "bake=off"?  Or was it our occasional
references to Henry as the "SIP Dough-Boy" which really
got them going?

;)

Lyndon Ong wrote:                         (Wed, 29 Nov 2000 10:27:07)
>
> Folks,
> 
> For those who asked about the origin of the term "bake-off", you might
> be interested in the discussion on the ietf mailing list where
> apparently Pillsbury has sent a "cease and desist" letter to Henning
> Schulzrinne on using the term "bake-off" for the SIP bake-offs.  The
> original email is dated Nov 6th.
> 
> Are there any non-English terms that might be equivalent in meaning?
> This could be a way around the Pillsbury legal team ;o)
> 
> Cheers,
> 
> L. Ong

-- 
Brian F. G. Bidulock
bidulock@dallas.net

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From: Alan Johnston <alan.johnston@wcom.com>
Subject: Re: [SIP] Music on Hold in ietf-sip-service-examples-00
To: Billy Biggs <Billy_Biggs@3com.com>
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Billy,

So your interpretation of the REFER request is that a successful completion
of the request always automatically generates a BYE request?  I'd be
interested to hear if other implementors have made this assumption as well.

This seems to me like mixing a service with the primitive used to implement
the service.

I thought this was why the original name of the method was changed from
TRANSFER (which is the name of a service and does imply a disconnect after
successful completion) to REFER, which can be used to implement a transfer
service, but just has the semantic to request establishment of a new session
without affecting the existing session.

Is your proposal for Music On Hold  for User B to proxy media from the Music
On Hold Server, or is it to send a re-INVITE to User A using SDP from the
Music On Hold Server?  Your other approach is for User A to choose its own
Music On Hold server without assistance from User B?

I'm always interested in seeing new call flows, but I'm not sure that these
are better approaches.

Thanks,
Alan.


Billy Biggs wrote:

>   In sip-service-examples-00, you discuss performing music on hold using
> REFER.  A shortened call flow is below:
>
>   On hold:
>
>          User A        User B       Music Server
>           |              |              |
>           |   REFER Refer-To: sip:music@server.com
>           |<-------------|              |
>           | INVITE/200 OK/ACK           |
>           |<--------------------------->|
>           |  200 OK      |              |
>           |------------->|              |
>           |              |              |
>
>   Off hold:
>
>           |              |              |
>           |   REFER Refer-To: music@server.com;method=BYE
>           |<-------------|              |
>           |  BYE/200 OK  |              |
>           |<--------------------------->|
>           |  200 OK      |              |
>           |------------->|              |
>           |              |              |
>
>   I have two problems with this method:
>
>   1. In our implementation, User A will hang up the call to User B after
>      the REFER completes successfully.  This is to protect ourselves
>      (User B will also send a BYE), however, I can imagine most
>      implementations will be similar.
>
>   2. As mentioned in my previous email on remote device control using
>      REFER, the URI music@server.com;method=BYE is insufficient to
>      indicate to the UA which call should be disconnected.  It's also
>      overloading REFER and obfuscating its meaning.
>
>   There are more appropriate ways to implement music on hold.  One is to
> have User B call up the music server itself and swap the SDP (3pcc).
> Another is to have User A call up the music on hold server when it
> receives on-hold SDP.
>
> --
> Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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Date: Wed, 29 Nov 2000 15:49:45 -0600

Alan Johnston (alan.johnston@wcom.com):

> So your interpretation of the REFER request is that a successful
> completion of the request always automatically generates a BYE
> request?

  Implementation not interpretation.

> [...] to REFER, which can be used to implement a transfer service, but
> just has the semantic to request establishment of a new session
> without affecting the existing session.

  The REFER transaction itself does not affect the old session.
However, to keep our phone UI simple, we disconnect the old call.  Users
can't understand it otherwise.

  Yes, we are using REFER to implement transfer.  If there are seperate
requirements, such as requesting to add a user in a 3-way call, or
asking the remote phone to dial another user seperately and be able to
remotely control the call, then these should be seperate methods or with
a Requires or something.  PHONECTL was designed (and implemented) to do
much of this, with an appropriate authentication structure.

> Is your proposal for Music On Hold  for User B to proxy media from the
> Music On Hold Server, or is it to send a re-INVITE to User A using SDP
> from the Music On Hold Server?  Your other approach is for User A to
> choose its own Music On Hold server without assistance from User B?

  As with most telephony features and SIP, there are many options.  It
is up to the implementor to pick one.  My problem with your Music on
Hold proposal is that it isn't interoperability-friendly.

  A more important problem is:

>>   2. As mentioned in my previous email on remote device control using
>>      REFER, the URI music@server.com;method=BYE is insufficient to
>>      indicate to the UA which call should be disconnected.  It's also
>>      overloading REFER and obfuscating its meaning.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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http://www.isi.edu/in-notes/iana/assignments/sip-parameters

Please send any corrections to me for the time being.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

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Date: Wed, 29 Nov 2000 21:05:26 -0600
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I thought I had done this a few weeks ago, but the email seems to have not
happened. My apologies.

I hereby issue a WG last call on Jonathan's informational-track "Guidelines
to the Authors of SIP Extensions", closing December 24, 2000. My apologies
on the overlap with our December meeting, but the document has been around
for many months without further comment and I should have closed it sooner.

The document can be retrieved from:

http://search.ietf.org/internet-drafts/draft-ietf-sip-guidelines-00.txt

--
Dean Willis



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Date: Wed, 29 Nov 2000 22:26:05 -0500

A few corrections here; I submitted a -01 which should appear shortly. In
the mean time:
http://www.cs.columbia.edu/~jdrosen/papers/draft-ietf-sip-guidelines-01.txt

Secondly, I believe this is more appropriate as a BCP. The "guidelines for
authors of RTP payload formats", which is a similar document, is a BCP.

-Jonathan R.

 

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Wednesday, November 29, 2000 10:05 PM
> To: IETF SIP (E-mail)
> Subject: [SIP] WG Last Call for draft-ietf-sip-guidelines-00.txt
> 
> 
> I thought I had done this a few weeks ago, but the email 
> seems to have not
> happened. My apologies.
> 
> I hereby issue a WG last call on Jonathan's 
> informational-track "Guidelines
> to the Authors of SIP Extensions", closing December 24, 2000. 
> My apologies
> on the overlap with our December meeting, but the document 
> has been around
> for many months without further comment and I should have 
> closed it sooner.
> 
> The document can be retrieved from:
> 
> http://search.ietf.org/internet-drafts/draft-ietf-sip-guidelin
> es-00.txt
> 
> --
> Dean Willis
> 
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Date: Thu, 30 Nov 2000 00:22:54 -0500

I think this all has to be taken within the context that the entfw draft is
NOT meant to be a long term solution. It is a short term hack, nothing more.
As such, I am not really worried about potential HTTPS caching proxies (if
even possible) that might arise one or two years from now. 

The draft aims really to solve a chicken-and-egg problem. No enterprise IT
admin is going to let SIP (let alone UDP based RTP) in or out of the
firewall until there is overwhelming demand and strong reasons for them to
do so. These incentives come when enough internal users are using services
and deriving business value from them. Unfortunately, that won't ever happen
since the firewalls block SIP/RTP. So you see the problem. Through what is,
in the best case, an ugly hack, we can get some usage going within
enterprises, and then the demand can grow, and then IT departments will
change policies so that users can stop doing horrible things like port 443
tunneling, and rather use SIP the original way it was designed to be used.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Howard Hart [mailto:hch@ipdialog.com]
> Sent: Tuesday, November 28, 2000 3:09 PM
> To: Henning G. Schulzrinne
> Cc: Jonathan Rosenberg; 'sip@lists.bell-labs.com'
> Subject: Re: [SIP] FW: I-D ACTION:draft-rosenberg-sip-entfw-00.txt
> 
> 
> Hi Henning,
> 
>    See responses below:
> 
> "Henning G. Schulzrinne" wrote:
> 
> > Howard Hart wrote:
> > >
> > > Jonathan and Henning,
> > >
> > >    I think this is a great shot at tunneling SIP and RTP 
> through Firewalls,
> > > though you may want to add the following caveats to section 6:
> > >
> > >     1) Wrt corporate tunneling, high-bandwidth voice 
> traffic has a distinct
> > > network signature vs. (relatively) low-bandwidth Web 
> traffic. If you blast
> > > voice traffic, encrypted or unencrypted over Web ports, 
> it's going to stick out
> > > like a sore thumb even using the crudest network 
> monitoring tools. ISS and the
> > > other network intrusion detection packages may even flag 
> it as a Netbus,
> > > BackOrifice, or  Distributed Denial of Service attack 
> signature. Most network
> > > administrators will shut the originating host/port down 
> immediately and ask
> > > questions later.
> >
> > I'm not sure whether that's a problem, as voice bandwidth 
> is generally
> > much lower than web retrievals. (As low as 8 kb/s vs. a few hundred
> > kilo*bytes*/second on my desktop to many web servers. If I 
> download a
> > piece of software, it will easily consume as much bandwidth 
> as a phone
> > call: 3 minute call = 180 seconds = 180 kBytes, which is 
> smaller than
> > many individual web pages.)
> 
> If the average Web page retrieval required 100+ 
> kbytes/second, I don't think the
> average user on their 28.8 baud, 3-5 kbytes/second (w/VJ 
> compression) modem would do
> much web surfing :-), but I get your point. My contention is 
> divergence from Web
> traffic signatures will be highly dependent on a number of 
> cumulative factors, from
> your locally provisioned bandwidth, to your usage patterns, 
> to the number of times
> you access the same server, to the sensitivity of your 
> intrusion detection package,
> to how long you stay on the phone, etc. If you throw in 
> sustained, encrypted RTP
> packet rate signatures at 25-50 pps over several minutes with 
> the added burden that
> only a small percentage of Web traffic is encrypted/port 443, 
> it will probably
> trigger the alarms, but your mileage will vary depending on 
> the level of paranoia at
> your local site.
> 
> >
> >
> > >
> > >     2) The Internet backbone itself has been heavily 
> optimized (some might say
> > > almost too optimized) for port 80 traffic, and is moving 
> in that direction (as
> >
> > This is bogus, except for the web caching that some ISPs (but,
> > fortunately, by no means all ISPs) do. Routers don't know 
> about port 80.
> 
> Agreed, my wording was poor. More accurately, ISPs are 
> deploying web caching engines
> at their edges to eliminate/optimize redundant HTTP/port 80 
> traffic over their
> backbones, especially when they're being used for transit vs. 
> direct access to Web
> servers within their networks.
> 
> >
> >
> > > much as is feasible) for port 443/HTTPS. Increasingly, 
> ISPs are using Web
> > > caching engines to limit this traffic across their 
> backbones. Case in point--I
> > > used Checkpoint's encrypted VPN product in a previous 
> company to connect an
> > > overseas location to our site (encrypted payloads, port 
> numbers stay the same).
> > > Our remote site wanted to access our internal Web server. 
> Unfortunately, one of
> > > the intermediate ISPs was using a Web caching product, 
> which resulted in the
> > > caching engine intercepting, then attempting to originate 
> an (unauthorized)
> > > port 80 access to our internal Web server on the remote 
> end's behalf, with the
> > > expected (non)result. The answer from the ISP was, of 
> course, "well it works
> > > fine for all our normal customers--EOM". I think you'd 
> see some very
> > > interesting  side effects if you applied a tunneled RTP 
> stream to this or other
> > > kinds of caching servers on port 80.
> > > Granted, 443/SSL-caching is a much more difficult 
> problem, equating to a
> > > man-in-the- middle attack--maybe even unsolvable, but 
> given the monetary
> > > inceptive is huge to come up with a solution here too, 
> partial or otherwise....
> >
> > Much of 443 traffic is inherently uncacheable, as it is for 
> things like
> > forms responses from web orders. I doubt that even the looniest ISP
> > would want to return your airline booking confirmation page or bank
> > balance to me. Also, unless the cache has the certificate 
> of the origin
> > server, intercept is just not possible. I doubt that Citibank, say,
> > would hand out their private key to random ISPs.
> 
> True, and as I stated above, maybe unsolvable, but for a 
> sample of where pieces of
> this might come into play later on, check out
> http://www.f5.com/f5products/bigip/ecommerce, one of the more 
> popular Web
> load-balancers. The advantage here is you offload encryption 
> and key negotiation
> from individual servers in a large server farm and place the 
> burden in hardware on
> the load-balancer. The question then becomes an ISP/SysAdmin 
> policy issue as to
> whether you can locate your remote tunneling Proxy in front 
> of or behind the
> load-balancer, and what's the impact when the load-balancer 
> assumes all traffic on
> port 443 is HTTP only.
> 
> NOTE: This is obviously NOT an issue for your scheme today. 
> Just something to
> consider or maybe even take advantage of in the future.
> 
> Howard Hart
> ipDialog, Inc.
> 

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Date: Thu, 30 Nov 2000 00:25:23 -0500

Multiple SDP is not used within SIP. Whilst its not explicitly prohibited,
the rfc implicitly discusses a single SDP payload and describes usage only
for this case.
I'll add a note saying just one SDP.

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: sunayak@hss.hns.com [mailto:sunayak@hss.hns.com]
> Sent: Monday, November 27, 2000 11:48 PM
> To: sip@lists.bell-labs.com
> Subject: [SIP] Multiple SDP bodies in a SIP message
> 
> 
> 
> 
> Hi All !
>      I had a query regarding SDP message bodies.
> 
> Is it possible to have multiple SDP bodies in a single SIP 
> message. If so,
> where would this be applicable ? (Could someone give a 
> scenario/example)
> Eg:
> <SIP msg>
> <SDP body 1>
> <SDP body 2>
> ...
> ...
> <SDP body n>
> 
> Thanks in advance
> Subhash Nayak
> Hughes Software Systems
> 
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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Date: Thu, 30 Nov 2000 00:34:44 -0500



 

> -----Original Message-----
> From: Billy Biggs [mailto:Billy_Biggs@3com.com]
> Sent: Wednesday, November 29, 2000 4:50 PM
> To: Alan Johnston
> Cc: Robert Sparks; SIP List
> Subject: Re: [SIP] Music on Hold in ietf-sip-service-examples-00
> 
> 
> Alan Johnston (alan.johnston@wcom.com):
> 
> > So your interpretation of the REFER request is that a successful
> > completion of the request always automatically generates a BYE
> > request?
> 
>   Implementation not interpretation.
> 
> > [...] to REFER, which can be used to implement a transfer 
> service, but
> > just has the semantic to request establishment of a new session
> > without affecting the existing session.
> 
>   The REFER transaction itself does not affect the old session.
> However, to keep our phone UI simple, we disconnect the old 
> call.  Users
> can't understand it otherwise.

You are most definitely confusing a primitive operation (REFER), with a
specific service, which is transfer.

As Alan correctly pointed out, REFER has other applications beyyond
transfer. Assuming that everytime the device sends a REFER, its a transfer,
is both an incorrect interpretation and an incorrect implementation. Now, it
may be that when a user presses "transfer" on your phone, it sends a REFER
then a BYE. Thats fine; in this flow here, the user would not press
"transfer", since this is not a transfer service. 


> 
>   Yes, we are using REFER to implement transfer.  If there 
> are seperate
> requirements, such as requesting to add a user in a 3-way call, or
> asking the remote phone to dial another user seperately and be able to
> remotely control the call, then these should be seperate 
> methods or with
> a Requires or something.  PHONECTL was designed (and 
> implemented) to do
> much of this, with an appropriate authentication structure.

Absolutely not. Putting a Require header in to specify each service for
which REFER might be used is a bad thing. Its the same as writing a formal
protocol spec for each specific service. Our aim here was to build primitive
operation which could be used to build many services, without every party
knowing the service. Just the invoker needs to know how to map the services
into protocol primitives.

-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com

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From: "Dean Willis" <dean.willis@softarmor.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
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Subject: Re: [SIP] WG Last Call for draft-ietf-sip-guidelines-00.txt
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Date: Thu, 30 Nov 2000 00:01:46 -0600
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Thanks. I stand corrected (actually, I sit asleep and typing)  The working
group last call is on the -01 version of the draft, which can be retrieved
from:

http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-guidelines-01.txt

This document is intended for publication as a BCP.

Good night.

--
Dean Willis

----- Original Message -----
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>; "IETF SIP (E-mail)"
<sip@lists.bell-labs.com>
Sent: Wednesday, November 29, 2000 9:26 PM
Subject: RE: [SIP] WG Last Call for draft-ietf-sip-guidelines-00.txt


> A few corrections here; I submitted a -01 which should appear shortly. In
> the mean time:
>
http://www.cs.columbia.edu/~jdrosen/papers/draft-ietf-sip-guidelines-01.txt
>
> Secondly, I believe this is more appropriate as a BCP. The "guidelines for
> authors of RTP payload formats", which is a similar document, is a BCP.
>
> -Jonathan R.
>
>
>
> > -----Original Message-----
> > From: Dean Willis [mailto:dean.willis@softarmor.com]
> > Sent: Wednesday, November 29, 2000 10:05 PM
> > To: IETF SIP (E-mail)
> > Subject: [SIP] WG Last Call for draft-ietf-sip-guidelines-00.txt
> >
> >
> > I thought I had done this a few weeks ago, but the email
> > seems to have not
> > happened. My apologies.
> >
> > I hereby issue a WG last call on Jonathan's
> > informational-track "Guidelines
> > to the Authors of SIP Extensions", closing December 24, 2000.
> > My apologies
> > on the overlap with our December meeting, but the document
> > has been around
> > for many months without further comment and I should have
> > closed it sooner.
> >
> > The document can be retrieved from:
> >
> > http://search.ietf.org/internet-drafts/draft-ietf-sip-guidelin
> > es-00.txt
> >
> > --
> > Dean Willis
> >
> >
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> >
>
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>


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Well, it looks like I let another draft get a little long iin the tooth
without care and feeding. There has been no discussion for several months,
and all threads appear to have been successfully answered on the list. I am
therefore issuing a WG last call on the Caller Preferences draft,
retrievable from:

http://www.ietf.org/internet-drafts/draft-ietf-sip-callerprefs-02.txt

This last call will close December 24. I want to send this puppy to the IESG
for New Years.

--
Dean


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Date: Thu, 30 Nov 2000 01:07:17 -0500

Once again, I must unfortunately correct our over-zealous working group
chair.

There is an updated version of this document in the I-D editor pipe. Until
it appears, you can pick up a copy at:

http://www.cs.columbia.edu/~jdrosen/papers/draft-ietf-sip-callerprefs-03.txt

Thanks,
Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Thursday, November 30, 2000 12:49 AM
> To: IETF SIP (E-mail)
> Cc: Joerg Ott (E-mail); Brian Rosen (E-mail); Jonathan Rosenberg
> (E-mail); Henning Schulzrinne (E-mail)
> Subject: WG Last Call on Caller Preferences
> 
> 
> Well, it looks like I let another draft get a little long iin 
> the tooth
> without care and feeding. There has been no discussion for 
> several months,
> and all threads appear to have been successfully answered on 
> the list. I am
> therefore issuing a WG last call on the Caller Preferences draft,
> retrievable from:
> 
> http://www.ietf.org/internet-drafts/draft-ietf-sip-callerprefs-02.txt
> 
> This last call will close December 24. I want to send this 
> puppy to the IESG
> for New Years.
> 
> --
> Dean
> 

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From: Billy Biggs <Billy_Biggs@3com.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: Re: [SIP] Music on Hold in ietf-sip-service-examples-00
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Date: Thu, 30 Nov 2000 00:14:36 -0600

Jonathan Rosenberg (jdrosen@dynamicsoft.com):

>>   The REFER transaction itself does not affect the old session.
>> However, to keep our phone UI simple, we disconnect the old call.
>> Users can't understand it otherwise.
> 
> You are most definitely confusing a primitive operation (REFER), with
> a specific service, which is transfer.

  Understandably we need some primitive.  I think we agree this
primitive is "initiate a session with the Refer-To address".

  Due to the design constraints of our device, we cannot keep around
both calls.  I suspect that a PSTN gateway would be in a similar
situation, since there is no way for the POTS phone to be able to
indicate this.  When the new session is active, we disconnect the old.

  Yes, we are unable to use services which require the old call to stay
active.  It's a constraint.  But our implementation is completely
reasonable and we do follow the spec.


  So far, the only service I've halted seems to be Music-On-Hold, which
requires that the far end have some crazy logic to send a BYE based on a
REFER anyways.  I'm sure there will be other services that this device
won't be able to handle, but I'm not worried yet.

  I suspect that other SIP devices which try to act like phones will
behave similarily, which is why I suggested to change the music call
flow to be more interoperability-friendly.

> [...] Putting a Require header in to specify each service for which
> REFER might be used is a bad thing. Its the same as writing a formal
> protocol spec for each specific service. Our aim here was to build
> primitive operation which could be used to build many services,
> without every party knowing the service. Just the invoker needs to
> know how to map the services into protocol primitives.

  Yes, but there are two actors here.  The sender of the REFER (who
knows the service it wants to provide) and the receiver of the REFER
(who doesn't).  At least in this example, if we knew that the sender
really needed the call to stay up after the REFER completes, then we
could at least reject the REFER.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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From: "Dean Willis" <dean.willis@softarmor.com>
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        "IETF SIP (E-mail)" <sip@lists.bell-labs.com>
Cc: "Joerg Ott (E-mail)" <jo@tzi.uni-bremen.de>,
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Date: Thu, 30 Nov 2000 00:15:37 -0600
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Yes indeed, I am underslept and continue to be amazed by colleaugue
Jonathan's forebearance.

The correct document for WG last call ending 24Dec is

http://www.softarmor.com/sipwg/drafts/draft-ietf-sip-callerprefs-03.txt

--
Dean

----- Original Message -----
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>; "IETF SIP (E-mail)"
<sip@lists.bell-labs.com>
Cc: "Joerg Ott (E-mail)" <jo@tzi.uni-bremen.de>; "Brian Rosen (E-mail)"
<Brian.Rosen@marconi.com>; "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>;
"Henning Schulzrinne (E-mail)" <schulzrinne@cs.columbia.edu>
Sent: Thursday, November 30, 2000 12:07 AM
Subject: RE: WG Last Call on Caller Preferences


> Once again, I must unfortunately correct our over-zealous working group
> chair.
>
> There is an updated version of this document in the I-D editor pipe. Until
> it appears, you can pick up a copy at:
>
>
http://www.cs.columbia.edu/~jdrosen/papers/draft-ietf-sip-callerprefs-03.txt
>
> Thanks,
> Jonathan R.
>
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
>
> > -----Original Message-----
> > From: Dean Willis [mailto:dean.willis@softarmor.com]
> > Sent: Thursday, November 30, 2000 12:49 AM
> > To: IETF SIP (E-mail)
> > Cc: Joerg Ott (E-mail); Brian Rosen (E-mail); Jonathan Rosenberg
> > (E-mail); Henning Schulzrinne (E-mail)
> > Subject: WG Last Call on Caller Preferences
> >
> >
> > Well, it looks like I let another draft get a little long iin
> > the tooth
> > without care and feeding. There has been no discussion for
> > several months,
> > and all threads appear to have been successfully answered on
> > the list. I am
> > therefore issuing a WG last call on the Caller Preferences draft,
> > retrievable from:
> >
> > http://www.ietf.org/internet-drafts/draft-ietf-sip-callerprefs-02.txt
> >
> > This last call will close December 24. I want to send this
> > puppy to the IESG
> > for New Years.
> >
> > --
> > Dean
> >
>


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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: "'Jo Hornsby'" <jhornsby@ubiquity.net>,
        Bertil Engelholm <Bertil.Engelholm@uab.ericsson.se>,
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Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Subject: RE: [SIP] RECORD-ROUTE/ROUTE requirements
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Date: Wed, 29 Nov 2000 22:27:53 -0800

Comments below.

> 
> > This discussion seems endless 8-). 
> 
> Indeed.  And there I was hoping that we would have this
> sorted for the Bake-Off.  Oh well.
> 

I still think we can have a proposal before the bake-off [and unfotunately I
can't make the bake-off]. At least that gives a starting point.

> > The main purpose with RECORD-ROUTE/ROUTE as I 
> > understand it is that a server can make shure it will 
> > get all new requests for a certain call-leg by adding 
> > a RECORD-ROUTE to the first (and consequtive) request. 
> 
> Exactly.  And that is why I think we should mandate (i.e.,
> MUST) that it is a SIP URL with either no transport
> parameter, or a transport parameter that matches /UDP/i.
> If these conditions are not met, then we cannot guarantee
> that a client will be able to route successfully.
> 
> As an analogy, consider the fact that UDP is a MUST.
> I wouldn't be surprised if someone writes something that
> doesn't do UDP, because TLS is essential (for some reason).
> Now they can't expect to be able to interoperate, but
> doubtless that won't matter, since the parameters of
> the network this non-UDP-supporting-something is going
> to be running in will ensure that it can interoperate
> with whatever it needs to.
> 
> In the same way, someone might one day insert a non-SIP
> URI into a Record-Route, but they can't expect it to
> work unless they know that the devices in the network
> are controlled such that they'll understand this non-SIP
> URI.
> 
Yes! yes! yes! I agree entirely. 


> > The next issue I see is port number. Today you have 
> > to use the same port number from both caller and callee
> > because the same ROUTE is used in both directions.
> > Should it be possible to use different port numbers ?
> > I would say yes to this. If a server wants to have
> > different port numbers it should have the possibility
> > to do so.
> 
> Well, if a proxy is prepared to modify the Record-Route
> in the response, then it is free to expose a different
> port to callee/caller.
> 
Right, its all up to the proxy as to what it puts in there. We just have to
say what is the minimum that it should put in there.

> > The more controversial parts I guess is the address 
> > and the transport. Should they be RECORD-ROUTE/ROUTE 
> > parameters or URI parameters ? Personally I prefere
> > RECORD-ROUTE/ROUTE parameters because that means I
> > don't have to look inside the ROUTE-URI and that makes 
> > me more future proof towards possibly new types of URIs. 
> >
> > The other possibility that has been discussed is to
> > only allow SIP-URL as URI in ROUTE and put the address 
> > in there as today (maddr). I don't like this solution 
> > because if a new type of URI pops up in the future that 
> > a server needs to put in as identification it can't do
> > it directly. Instead it has to convert it into a SIP-URL 
> > in some way.
> 
> If a proxy needs to embed exotic URIs (or whatever) in
> Record-Route, then I think it should have to suffer all
> the pain involved.
> 
> If we use anything other than UDP SIP URLs, I can't see
> how we can guarantee interoperability.
> 
Exactly. I think this is more a matter of confusion than controversy. Bertil
said:

> > The more controversial parts I guess is the address 
> > and the transport.

So a SIP URL is adquate for all host based networks and UDP transport is
required - so why make life hard. Just because _the original_ call routing
is done on mailto, tel or whatever esoteric URI scheme doesn't impact the
fact that a Route header MUST be sent to the specified UDP reachable host.

As you say Jo, if someone wants to use non-SIP Route URL's that's fine but
you can't guarantee interoperability so it should not be put in the SIP
spec.

Ok so as I see it:

-Record-Route headers must be SIP URL's. 
-The address in the SIP URL must be the UDP reachable address of the proxy
inserting the route.
-The username and any parameters added to the URL to the header are
implementation defined. 
-The Record-Route should use angle brackets. 
Ie, Record-Route: <xxxxx@proxyaddress:port;yyyyy=yyyyy>;zzzzz=zzzzz
where y are the implementation defined Record-Route URL parameters and zzzz
are the implementation defined Record-Route header parameters. xxxxx may be
empty.
-If multiple Record-Routes are added for the same request (ie, spiral
route), then each Record-Route added SHOULD be different so that they can be
distinguished.
-In the response to a Record-Route'd request, a proxy MAY change any
Record-Route header that the it originally generated. 
- A UAS must copy the Record-Route unchanged into the Route header; a UAC
must copy the Record-Route unchanged into the Route header, reversing the
order. A Route header is added if a Contact is added (as before). 
- If a proxy inserts Record-Route header into a request without Contact or
Record-Route headers, then the Proxy SHOULD insert a Contact into the
request with the From URL information and a maddr of the source address.
[This ia KISS option. Contacts are now mandatory and so this is only
required for supporting older UA's and why don't we keep the proxy behaviour
as simple as possible. Can anyone see any reasons why this would cause
problems?]
- If a proxy receives a response to a request in which it added a
Record-Route header AND the request does not contain a COntact header AND no
new headers have been added by other proxies (ie is last Record-Route), then
the proxy SHOULD add a Contact header field to the response with the
original Request-URI and an additional maddr parameter with the forward
address (to replicate the original routing action).
[This ia KISS option. Contacts are now mandatory and so this is only
required for supporting older UA's and why don't we keep the proxy behaviour
as simple as possible (please!!). Can anyone see any reasons why this would
cause problems?]

In my mind, this allows the proxy the most freedom of how much information
is placed in the Record-Route and how to encode it. The backwards compat
case where there is no Contact added is handled in the simplest way possible
that should give a good chance of success. 

Jonathan, do you still want to argue for more stringent restrictions that
the username cannot be empty and that the proxy MUST change the Record-Route
in the response ?

Comments or omissions?

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 

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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Date: Thu, 30 Nov 2000 03:06:05 -0500

So, figuring we didn't all have enough email on the sip list to read (I am
personally hopelessly behind), let me stir things around some more with
another proposal for something to ditch from the spec.

Right now, you can carry SDP in the INV/200/ACK exchange in two ways. You
can send nothing in the INVITE, followed by SDP in the 200, and then SDP in
ACK (and fortunately I don't know anyone who actively does this). Or, you
can send SDP in INVITE, SDP in 200, and nothing in ACK. I also think there
are several confused implementations that likely send it in all three. 

Add to this the unfortunate PRACK mess, which also has been used to send SDP
in order to support the manyfolks resource work when used with 3pcc, but
which is yet unimplemented.

It all adds up to a messy confusion. Its not clear what it means to send SDP
in all these places, and how it works.

So, my proposal is that we simplify. We only allow SDP in INVITE and then
provisionals/200 OK. No SDP in ACK. No SDP in PRACK. (sounds like a Dr.
Seuss rhyme, doesn't it?)

The latest 3pcc draft no longer recommends using the SDP-in-ACK, since it
has problems. Given that there don't seem to be useful applications of it
any longer, and given that, to my knowledge, it is not sent by anyone (at
least not in any commercially shipping boxes, I hope), I'd like to yank it.

There is a consequence, and that is dealing with the manyfolks requirement
for sending an updated SDP to deal with manyfolks+3pcc. My proposal for that
is to use a re-invite, rahter than PRACK, to carry the updated SDP. This
does mean that we need to allow, in SIP, a re-INVITE to arrive while an
initial INVITE is in progress. Allowing this would also address some of the
concerns raised in previous threads about handling requirements for changes
in early media.

I don't think its hard to deal with the re-INVITE while INVITE-is-active
case. The simplest solution is to reject the first INVITE and continue
processing with the second one, using the new SDP and all.

Anyway, before getting into those details, what does the group think about
dropping the SDP-in-ACK and SDP-in-PRACK stuff?

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

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Date: Thu, 30 Nov 2000 14:00:53 +0530




SDP in PRACK I have no problems with.

However ,removing SDP in ACK will break some SIP-H.323 implementations
which send media
on ACK. There were two approaches to this, one of which was SDP in ACK. So
those who took this latter
approach will suffer (although I dont know how many commercial deployments
exists - I guess for most
it would just be another evening or two  of re-working in their labs)

As far as receiving multiple INVITE before each trans. end is concerned, at
first thought this seems messier
for the developer. All this while , if you received T2 while T1 was in
progress (T=Transaction) from the same
endpoint, you could send a bad message with a retry after. This made
implementations simple, but now with
multipe INVITEs, the receiving side has to remember:
     * CSeq values of all INVITEs and respond with app. 200 OKs
     * remember possible diff. media parameters for each recvd. INVITE and
respond with OK
     * know how to correlate an ACK recvd afer it sends back Ok for one of
the overlapped invites.
     <etc ?>
All this can be surely implemented, but at first thought, it looks dirty.
Agreed though that it solves some problems
of efficiency (eg overlapped dialing).  This entire thread of record-route
reworking along with now multiple INVITEs
before they complete seem to be slowly making life that much more difficult
:-)

Thoughts ?

Regds
Arjun

--
Arjun Roychowdhury @ Hughes Software Systems







Jonathan Rosenberg <jdrosen@dynamicsoft.com> on 11/30/2000 01:36:05 PM

To:   "'sip@lists.bell-labs.com'" <sip@lists.bell-labs.com>
cc:

Subject:  [SIP] as if there wasn't enough traffic on this darn list...




So, figuring we didn't all have enough email on the sip list to read (I am
personally hopelessly behind), let me stir things around some more with
another proposal for something to ditch from the spec.

Right now, you can carry SDP in the INV/200/ACK exchange in two ways. You
can send nothing in the INVITE, followed by SDP in the 200, and then SDP in
ACK (and fortunately I don't know anyone who actively does this). Or, you
can send SDP in INVITE, SDP in 200, and nothing in ACK. I also think there
are several confused implementations that likely send it in all three.

Add to this the unfortunate PRACK mess, which also has been used to send
SDP
in order to support the manyfolks resource work when used with 3pcc, but
which is yet unimplemented.

It all adds up to a messy confusion. Its not clear what it means to send
SDP
in all these places, and how it works.

So, my proposal is that we simplify. We only allow SDP in INVITE and then
provisionals/200 OK. No SDP in ACK. No SDP in PRACK. (sounds like a Dr.
Seuss rhyme, doesn't it?)

The latest 3pcc draft no longer recommends using the SDP-in-ACK, since it
has problems. Given that there don't seem to be useful applications of it
any longer, and given that, to my knowledge, it is not sent by anyone (at
least not in any commercially shipping boxes, I hope), I'd like to yank it.

There is a consequence, and that is dealing with the manyfolks requirement
for sending an updated SDP to deal with manyfolks+3pcc. My proposal for
that
is to use a re-invite, rahter than PRACK, to carry the updated SDP. This
does mean that we need to allow, in SIP, a re-INVITE to arrive while an
initial INVITE is in progress. Allowing this would also address some of the
concerns raised in previous threads about handling requirements for changes
in early media.

I don't think its hard to deal with the re-INVITE while INVITE-is-active
case. The simplest solution is to reject the first INVITE and continue
processing with the second one, using the new SDP and all.

Anyway, before getting into those details, what does the group think about
dropping the SDP-in-ACK and SDP-in-PRACK stuff?

-Jonathan R.

---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com


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Subject: RE: [SIP] RECORD-ROUTE/ROUTE requirements
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Date: Thu, 30 Nov 2000 10:34:51 -0000
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Robert wrote:
> So a SIP URL is adquate for all host based networks and UDP
> transport is required - so why make life hard. Just because
> _the original_ call routing is done on mailto, tel or whatever
> esoteric URI scheme doesn't impact the fact that a Route
> header MUST be sent to the specified UDP reachable host.

Very much so.

> As you say Jo, if someone wants to use non-SIP Route URL's
> that's fine but you can't guarantee interoperability so it
> should not be put in the SIP spec.

Indeed.

> Ok so as I see it:
> 
> -Record-Route headers must be SIP URL's. 
> -The address in the SIP URL must be the UDP reachable address
> of the proxy inserting the route.
> -The username and any parameters added to the URL to the header
> are implementation defined. 
> -The Record-Route should use angle brackets. 
> Ie, Record-Route: <xxxxx@proxyaddress:port;yyyyy=yyyyy>;zzzzz=zzzzz
> where y are the implementation defined Record-Route URL 
> parameters and zzzz are the implementation defined Record-Route
> header parameters. xxxxx may be empty.

(as may be yyyyy / zzzzz)

> -If multiple Record-Routes are added for the same request
> (ie, spiral route), then each Record-Route added SHOULD be
> different so that they can be distinguished.
> -In the response to a Record-Route'd request, a proxy MAY
> change any Record-Route header that the it originally
> generated. 
> - A UAS must copy the Record-Route unchanged into the Route
> header; a UAC must copy the Record-Route unchanged into the
> Route header, reversing the order. A Route header is added
> if a Contact is added (as before).

No arguments here. &:)

> - If a proxy inserts Record-Route header into a request
> without Contact or Record-Route headers, then the Proxy SHOULD
> insert a Contact into the request with the From URL information
> and a maddr of the source address.  [This ia KISS option.
> Contacts are now mandatory and so this is only required for
> supporting older UA's and why don't we keep the proxy behaviour
> as simple as possible. Can anyone see any reasons why this
> would cause problems?]
> - If a proxy receives a response to a request in which it
> added a Record-Route header AND the request does not contain
> a COntact header AND no new headers have been added by other
> proxies (ie is last Record-Route), then the proxy SHOULD add
> a Contact header field to the response with the original
> Request-URI and an additional maddr parameter with the forward
> address (to replicate the original routing action).
> [This ia KISS option. Contacts are now mandatory and so this
> is only required for supporting older UA's and why don't we
> keep the proxy behaviour as simple as possible (please!!).
> Can anyone see any reasons why this would cause problems?]

I don't think that proxies should start second-guessing UAS.
I would say that the simplest thing is just to say that when the
UA is building the Route header list, it should append the
Contact, if present; failing that, the From.

Really old UAs might lose, but they were always going to do
that, anyway[0].

Cheers,


 - Jo.

[0] I'm obviously a bis snob.


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people,

i'm slightly concerened (maybe un-justifably) about the re-invite
working of the 3PCC draft.

my concern is that it is possible (is it not?) for the user agent
receiving the re-invite to change its SDP settings in its response as
follows:

 A                Controller            B
 |  INV held SDP     |                  |
 |<------------------|                  |
 |                   |                  |
 |  200 SDP A1       |                  |
 |-----------------> |  INV SDP A1      |
 |  ACK              |----------------->|
 |<----------------- |                  |
 |                   |  200 SDP B1      |
 |                   |<-----------------|
 |                   |                  |
 |                   |  ACK             |
 |  INV SDP B1       |----------------->|
 |<------------------|                  |
 |  200 OK SDP A2    |                  |
 |------------------>|                  |
 |  ACK              |                  |
 |<------------------|                  |
 |                                      |
 |               BROKEN RTP             |
 |                                      |
 |                                      |
 |                                      |

NOTE: A has returned different SDP in its two different 200 OK messages.

Why might it do this?  It might not but it is possible is it not.

perhaps we should mandate that an agent can only change its SDP for a
session in an INVITE not in a response (after the initial
INVITE/RESPONSE exchange of course)

what do we think?

-- 
Gethin Liddell
Ubiquity Software Corporation

http://www.ubiquity.net
mailto:gethin@ubiquity.net

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Hello Jonathan,

Hmm yes,it is a good idea to keep things clear.
I agree that it is a good idea to keep the ACK free of SDP and if you want to change the session, then to send a re-invite.

Another advantage is that in this case, the UAs can already act on the 200 OK SDP message without waiting for the ACK.
(saves some time).

cheers

Arnoud

_______________________________________________________________
ERICSSON EUROLABS NETHERLANDS BV
Arnoud van Wijk
ABACUS Lab
Research & Development (ELN/V)
Fax: +31-161 247569
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Jonathan Rosenberg wrote:
> 
> So, figuring we didn't all have enough email on the sip list to read (I am
> personally hopelessly behind), let me stir things around some more with
> another proposal for something to ditch from the spec.
[...]
> So, my proposal is that we simplify. We only allow SDP in INVITE and then
> provisionals/200 OK. No SDP in ACK. No SDP in PRACK. (sounds like a Dr.
> Seuss rhyme, doesn't it?)

Did you just get done watching "The Grinch" :-)

[...]
> I don't think its hard to deal with the re-INVITE while INVITE-is-active
> case. The simplest solution is to reject the first INVITE and continue
> processing with the second one, using the new SDP and all.
> 
> Anyway, before getting into those details, what does the group think about
> dropping the SDP-in-ACK and SDP-in-PRACK stuff?

Personally, I do not have any problems dropping SDP-in-ACK and SDP-in-PRACK 
(modulo Arjun's note that dropping SDP-in-ACK may break SIP-H.323 
implementations).

However, getting a re-INVITE while processing an INVITE (for the same call)
may get messy.  My first impulse it to say that, from a developer's view-
point, there is enough complexity while processing a normal INVITE: 
detecting and handling spiralled loops, request merging, forking, 
correlating response to multiple requests, etc.  Is the added complexity 
of handling re-INVITEs worth it?  Furthermore, will this break older 
implementations which at best may send a 4xx response with a Retry-After 
header, or at worst will not know what to do with the re-INVITE.  I don't 
think RFC 2543 dealt with re-INVITEs.

Just my $0.02.

Best regards,

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Internet Software and eServices Group 
Lucent Technologies/Bell Labs Innovations 263 Shuman Blvd., Rm 1A-413
Naperville, Illinois 60566     Voice: +1 630 224 0216   Fax: +1 630 713 0184

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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        "'Howard Hart'" <hch@ipdialog.com>,
        "Henning G. Schulzrinne" <hgs@cs.columbia.edu>
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Subject: RE: [SIP] FW: I-D ACTION:draft-rosenberg-sip-entfw-00.txt
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Date: Thu, 30 Nov 2000 10:02:36 -0500

There are a couple of mitigating factors.

One is that the big firewall vendors are starting to
commit to supporting SIP. No names, sorry, NDA issues!

Another is that commercial SIP firewall "side cars" are appearing.

I had an enlightening conversation with my internal security folks
about the relative merits of "in the firewall" and "in parallel with
the regular firewall" approaches.  They seemed content with the argument
that a parallel approach would be best if traffic volume of RTP was
high, but "in the existing firewall" was best otherwise.

I remember arguing for what we called an "interim AH" security scheme
in Megaco, because at the time, the embedded OSs did not support
IPSEC, and it was pretty hard to add it unless you had source code
(since the AH header goes between IP and anything else).  We used
a header in the Megaco payload.  By the time the protocol came out,
the OSs had been upgraded, so that most now have IPSEC, if not from
the vendor, then a 3rd party.  I wonder if we would have the same
result here.

Brian

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Thursday, November 30, 2000 12:23 AM
> To: 'Howard Hart'; Henning G. Schulzrinne
> Cc: Jonathan Rosenberg; 'sip@lists.bell-labs.com'
> Subject: RE: [SIP] FW: I-D ACTION:draft-rosenberg-sip-entfw-00.txt
> 
> 
> I think this all has to be taken within the context that the 
> entfw draft is
> NOT meant to be a long term solution. It is a short term 
> hack, nothing more.
> As such, I am not really worried about potential HTTPS 
> caching proxies (if
> even possible) that might arise one or two years from now. 
> 
> The draft aims really to solve a chicken-and-egg problem. No 
> enterprise IT
> admin is going to let SIP (let alone UDP based RTP) in or out of the
> firewall until there is overwhelming demand and strong 
> reasons for them to
> do so. These incentives come when enough internal users are 
> using services
> and deriving business value from them. Unfortunately, that 
> won't ever happen
> since the firewalls block SIP/RTP. So you see the problem. 
> Through what is,
> in the best case, an ugly hack, we can get some usage going within
> enterprises, and then the demand can grow, and then IT 
> departments will
> change policies so that users can stop doing horrible things 
> like port 443
> tunneling, and rather use SIP the original way it was 
> designed to be used.
> 
> -Jonathan R.
> 
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>  
> 
> > -----Original Message-----
> > From: Howard Hart [mailto:hch@ipdialog.com]
> > Sent: Tuesday, November 28, 2000 3:09 PM
> > To: Henning G. Schulzrinne
> > Cc: Jonathan Rosenberg; 'sip@lists.bell-labs.com'
> > Subject: Re: [SIP] FW: I-D ACTION:draft-rosenberg-sip-entfw-00.txt
> > 
> > 
> > Hi Henning,
> > 
> >    See responses below:
> > 
> > "Henning G. Schulzrinne" wrote:
> > 
> > > Howard Hart wrote:
> > > >
> > > > Jonathan and Henning,
> > > >
> > > >    I think this is a great shot at tunneling SIP and RTP 
> > through Firewalls,
> > > > though you may want to add the following caveats to section 6:
> > > >
> > > >     1) Wrt corporate tunneling, high-bandwidth voice 
> > traffic has a distinct
> > > > network signature vs. (relatively) low-bandwidth Web 
> > traffic. If you blast
> > > > voice traffic, encrypted or unencrypted over Web ports, 
> > it's going to stick out
> > > > like a sore thumb even using the crudest network 
> > monitoring tools. ISS and the
> > > > other network intrusion detection packages may even flag 
> > it as a Netbus,
> > > > BackOrifice, or  Distributed Denial of Service attack 
> > signature. Most network
> > > > administrators will shut the originating host/port down 
> > immediately and ask
> > > > questions later.
> > >
> > > I'm not sure whether that's a problem, as voice bandwidth 
> > is generally
> > > much lower than web retrievals. (As low as 8 kb/s vs. a 
> few hundred
> > > kilo*bytes*/second on my desktop to many web servers. If I 
> > download a
> > > piece of software, it will easily consume as much bandwidth 
> > as a phone
> > > call: 3 minute call = 180 seconds = 180 kBytes, which is 
> > smaller than
> > > many individual web pages.)
> > 
> > If the average Web page retrieval required 100+ 
> > kbytes/second, I don't think the
> > average user on their 28.8 baud, 3-5 kbytes/second (w/VJ 
> > compression) modem would do
> > much web surfing :-), but I get your point. My contention is 
> > divergence from Web
> > traffic signatures will be highly dependent on a number of 
> > cumulative factors, from
> > your locally provisioned bandwidth, to your usage patterns, 
> > to the number of times
> > you access the same server, to the sensitivity of your 
> > intrusion detection package,
> > to how long you stay on the phone, etc. If you throw in 
> > sustained, encrypted RTP
> > packet rate signatures at 25-50 pps over several minutes with 
> > the added burden that
> > only a small percentage of Web traffic is encrypted/port 443, 
> > it will probably
> > trigger the alarms, but your mileage will vary depending on 
> > the level of paranoia at
> > your local site.
> > 
> > >
> > >
> > > >
> > > >     2) The Internet backbone itself has been heavily 
> > optimized (some might say
> > > > almost too optimized) for port 80 traffic, and is moving 
> > in that direction (as
> > >
> > > This is bogus, except for the web caching that some ISPs (but,
> > > fortunately, by no means all ISPs) do. Routers don't know 
> > about port 80.
> > 
> > Agreed, my wording was poor. More accurately, ISPs are 
> > deploying web caching engines
> > at their edges to eliminate/optimize redundant HTTP/port 80 
> > traffic over their
> > backbones, especially when they're being used for transit vs. 
> > direct access to Web
> > servers within their networks.
> > 
> > >
> > >
> > > > much as is feasible) for port 443/HTTPS. Increasingly, 
> > ISPs are using Web
> > > > caching engines to limit this traffic across their 
> > backbones. Case in point--I
> > > > used Checkpoint's encrypted VPN product in a previous 
> > company to connect an
> > > > overseas location to our site (encrypted payloads, port 
> > numbers stay the same).
> > > > Our remote site wanted to access our internal Web server. 
> > Unfortunately, one of
> > > > the intermediate ISPs was using a Web caching product, 
> > which resulted in the
> > > > caching engine intercepting, then attempting to originate 
> > an (unauthorized)
> > > > port 80 access to our internal Web server on the remote 
> > end's behalf, with the
> > > > expected (non)result. The answer from the ISP was, of 
> > course, "well it works
> > > > fine for all our normal customers--EOM". I think you'd 
> > see some very
> > > > interesting  side effects if you applied a tunneled RTP 
> > stream to this or other
> > > > kinds of caching servers on port 80.
> > > > Granted, 443/SSL-caching is a much more difficult 
> > problem, equating to a
> > > > man-in-the- middle attack--maybe even unsolvable, but 
> > given the monetary
> > > > inceptive is huge to come up with a solution here too, 
> > partial or otherwise....
> > >
> > > Much of 443 traffic is inherently uncacheable, as it is for 
> > things like
> > > forms responses from web orders. I doubt that even the 
> looniest ISP
> > > would want to return your airline booking confirmation 
> page or bank
> > > balance to me. Also, unless the cache has the certificate 
> > of the origin
> > > server, intercept is just not possible. I doubt that 
> Citibank, say,
> > > would hand out their private key to random ISPs.
> > 
> > True, and as I stated above, maybe unsolvable, but for a 
> > sample of where pieces of
> > this might come into play later on, check out
> > http://www.f5.com/f5products/bigip/ecommerce, one of the more 
> > popular Web
> > load-balancers. The advantage here is you offload encryption 
> > and key negotiation
> > from individual servers in a large server farm and place the 
> > burden in hardware on
> > the load-balancer. The question then becomes an ISP/SysAdmin 
> > policy issue as to
> > whether you can locate your remote tunneling Proxy in front 
> > of or behind the
> > load-balancer, and what's the impact when the load-balancer 
> > assumes all traffic on
> > port 443 is HTTP only.
> > 
> > NOTE: This is obviously NOT an issue for your scheme today. 
> > Just something to
> > consider or maybe even take advantage of in the future.
> > 
> > Howard Hart
> > ipDialog, Inc.
> > 
> 
> _______________________________________________
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From sip-admin@lists.bell-labs.com  Thu Nov 30 10:19:06 2000
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Subject: [SIP] Processing BYE request at Proxy
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I have questions about BYE request.. My questions are two

<First Question>
I am a Proxy.
And I have received INVITE request from client and forked that to 
downstream.(now there is server and client call state in my proxy)
And then I have receive BYE request which has the same call-leg with previous INVITE request but has lower CSeq than previous INVITE request.

If I were UAS, I could reply 400(bad request).
But I am proxy, So What Can I do in this case?(According to SPEC, Proxy cannot generate final response to BYE request)

Of course, this situation is exceptional, But one of possible situation.

<Second Question>
I am a proxy.
And I have received INVITE request from client and forked that to down stream..
And then I have received none-200 final responses from all branches.... so I have forwarded lowest numbered final response to upstream.
But before I receive ACK for final response from upstream, I receive BYE request which has same call leg with previous INVITE request and has higher CSeq than previous INVITE request.
According to SPEC, Proxy cannot generate final response to BYE request.. So there is no way but to simply forking BYE request to downstream. I think this generates unnecessary network traffic.
I think If proxy could generated final response to BYE request without forking BYE request to downstream in this special case, there would be network bandwidth savings without malfunction.
I want your opinion about this issue   


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From sip-admin@lists.bell-labs.com  Thu Nov 30 10:33:11 2000
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> Jonathan Rosenberg (jdrosen@dynamicsoft.com):
> 
> >>   The REFER transaction itself does not affect the old session.
> >> However, to keep our phone UI simple, we disconnect the old call.
> >> Users can't understand it otherwise.
> > 
> > You are most definitely confusing a primitive operation (REFER), with
> > a specific service, which is transfer.
> 
>   Understandably we need some primitive.  I think we agree this
> primitive is "initiate a session with the Refer-To address".

Actually, it's do whatever makes sense in your UA for the URL.
Keep in mind that non-SIP URLs are valid here. 

> 
>   Due to the design constraints of our device, we cannot keep around
> both calls.  I suspect that a PSTN gateway would be in a similar
> situation, since there is no way for the POTS phone to be able to
> indicate this.  When the new session is active, we disconnect the old.
> 
>   Yes, we are unable to use services which require the old call to stay
> active.  It's a constraint.  But our implementation is completely
> reasonable and we do follow the spec.
> 
> 
>   So far, the only service I've halted seems to be Music-On-Hold, which
> requires that the far end have some crazy logic to send a BYE based on a
> REFER anyways.  I'm sure there will be other services that this device
> won't be able to handle, but I'm not worried yet.

What crazy logic? Building the request based on the method parameter and
possibly header parameter in the URL? 

> 
>   I suspect that other SIP devices which try to act like phones will
> behave similarily, which is why I suggested to change the music call
> flow to be more interoperability-friendly.
> 
> > [...] Putting a Require header in to specify each service for which
> > REFER might be used is a bad thing. Its the same as writing a formal
> > protocol spec for each specific service. Our aim here was to build
> > primitive operation which could be used to build many services,
> > without every party knowing the service. Just the invoker needs to
> > know how to map the services into protocol primitives.
> 
>   Yes, but there are two actors here.  The sender of the REFER (who
> knows the service it wants to provide) and the receiver of the REFER
> (who doesn't).  At least in this example, if we knew that the sender
> really needed the call to stay up after the REFER completes, then we
> could at least reject the REFER.
> 
> -- 
> Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com
> 

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From sip-admin@lists.bell-labs.com  Thu Nov 30 10:56:05 2000
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Date: Thu, 30 Nov 2000 09:50:35 -0600

You bring up a good point IMO.  Currently, a controller 
needs to check that A didn't change its SDP and if so, 
re-invite B.  Could end up in an awful loop with both UAs.

I agree that we need a way to guard against this situation. 
Your proposal seems reasonable to me.

Regards,

Bert

> -----Original Message-----
> From: Gethin Liddell [mailto:gethin@ubiquity.net]
> Sent: Thursday, November 30, 2000 7:19 AM
> To: Sip Mail List
> Subject: [SIP] 3PCC and the re-invite response
> 
> 
> 
> people,
> 
> i'm slightly concerened (maybe un-justifably) about the re-invite
> working of the 3PCC draft.
> 
> my concern is that it is possible (is it not?) for the user agent
> receiving the re-invite to change its SDP settings in its response as
> follows:
> 
>  A                Controller            B
>  |  INV held SDP     |                  |
>  |<------------------|                  |
>  |                   |                  |
>  |  200 SDP A1       |                  |
>  |-----------------> |  INV SDP A1      |
>  |  ACK              |----------------->|
>  |<----------------- |                  |
>  |                   |  200 SDP B1      |
>  |                   |<-----------------|
>  |                   |                  |
>  |                   |  ACK             |
>  |  INV SDP B1       |----------------->|
>  |<------------------|                  |
>  |  200 OK SDP A2    |                  |
>  |------------------>|                  |
>  |  ACK              |                  |
>  |<------------------|                  |
>  |                                      |
>  |               BROKEN RTP             |
>  |                                      |
>  |                                      |
>  |                                      |
> 
> NOTE: A has returned different SDP in its two different 200 OK
> messages.
> 
> Why might it do this?  It might not but it is possible is it not.
> 
> perhaps we should mandate that an agent can only change its SDP for a
> session in an INVITE not in a response (after the initial
> INVITE/RESPONSE exchange of course)
> 
> what do we think?
> 
> -- 
> Gethin Liddell
> Ubiquity Software Corporation
> 
> http://www.ubiquity.net
> mailto:gethin@ubiquity.net
> 
> _______________________________________________
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> 

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From sip-admin@lists.bell-labs.com  Thu Nov 30 11:04:33 2000
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From: Billy Biggs <Billy_Biggs@3com.com>
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Date: Thu, 30 Nov 2000 10:03:43 -0600

Robert Sparks (rsparks@dynamicsoft.com):

>>   So far, the only service I've halted seems to be Music-On-Hold,
>> which requires that the far end have some crazy logic to send a BYE
>> based on a REFER anyways.  I'm sure there will be other services
>> that this device won't be able to handle, but I'm not worried yet.
> 
> What crazy logic? Building the request based on the method parameter
> and possibly header parameter in the URL? 

  I assume you want the BYE to be associated with the music call.
Otherwise, that BYE would be sent with a new From tag and no To tag (a
new call leg), and would not end the old call.

  Aside from that matching, there are also security implications of
being able to tell a UA to send any SIP request, especially a BYE.  I
don't want a remote UA to tell me to hang up a call.  So if some service
requires this, then we have to do some logic to see if the user
previously asked us to make a call such that we would allow them to send
a BYE for it.  Ugh.

  So, our security is simple:

  - Remote UA must have a call active to us (no out-of-call REFERs).
  - We will only allow REFERs which ask us to INVITE.
  - We will only allow being REFERed once (REFER is at the cost of the
    old session).

  More priviledges are available through something stronger (PHONECTL),
which requires authentication.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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> -----Original Message-----
> From: Billy Biggs [mailto:Billy_Biggs@3com.com]
> Sent: Thursday, November 30, 2000 10:04 AM
> To: Robert Sparks
> Cc: Jonathan Rosenberg; Alan Johnston; SIP List
> Subject: Re: [SIP] Music on Hold in ietf-sip-service-examples-00
> 
> 
> Robert Sparks (rsparks@dynamicsoft.com):
> 
> >>   So far, the only service I've halted seems to be Music-On-Hold,
> >> which requires that the far end have some crazy logic to send a BYE
> >> based on a REFER anyways.  I'm sure there will be other services
> >> that this device won't be able to handle, but I'm not worried yet.
> > 
> > What crazy logic? Building the request based on the method parameter
> > and possibly header parameter in the URL? 
> 
>   I assume you want the BYE to be associated with the music call.
> Otherwise, that BYE would be sent with a new From tag and no To tag (a
> new call leg), and would not end the old call.

Hence To: and From: headers in the URL.

> 
>   Aside from that matching, there are also security implications of
> being able to tell a UA to send any SIP request, especially a BYE.  I
> don't want a remote UA to tell me to hang up a call.  So if some service
> requires this, then we have to do some logic to see if the user
> previously asked us to make a call such that we would allow them to send
> a BYE for it.  Ugh.

So your UA has a policy to always 503 sip: BYE URLs. 

> 
>   So, our security is simple:
> 
>   - Remote UA must have a call active to us (no out-of-call REFERs).
>   - We will only allow REFERs which ask us to INVITE.
>   - We will only allow being REFERed once (REFER is at the cost of the
>     old session).
> 
>   More priviledges are available through something stronger (PHONECTL),
> which requires authentication.

You could require that your REFERs be SIP authenticated. 

> 
> -- 
> Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
> http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com
> 

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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: RE: [SIP] 3PCC and the re-invite response
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Date: Thu, 30 Nov 2000 11:23:20 -0500

This is a recognized problem. Note the following section in the document:

   In addition, note that in Figure 1, the controller sends a re-INVITE
   to A with the SDP from B. The response to this re-INVITE is a 200 OK
   that contains "SDP A". If the SDP returned in the re-INVITE response
   were not the same, the controller would need to initiate a re-INVITE
   to B with that new SDP. This means that a UA which returns a



Rosenberg/Peterson/Schulzrinne/Camarillo                     [Page 12]

Internet Draft                    3pcc                 November 22, 2000


   different SDP (for example, by changing the ports) in the response to
   every re-INVITE will trigger an infinite re-INVITE loop from the
   controller to each controller entity. As such, it is STRONGLY
   RECOMMENDED that if a re-INVITE does not require a UAS to modify the
   formulation of the SDP description for a specific stream in the
   response, the SDP description of that stream in the response MUST be
   the same. In other words, if a UA was in a session with a single
   stream, using codec A, and it received a re-INVITE modifying the port
   to send to, the re-INVITE response MUST be the same as it was to the
   initial request. However, if the re-INVITE forces the UAS to change
   codecs, it is acceptable in that case to use a different port in the
   SDP in the response.



-Jonathan R.
---
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
http://www.dynamicsoft.com
 

> -----Original Message-----
> From: Gethin Liddell [mailto:gethin@ubiquity.net]
> Sent: Thursday, November 30, 2000 7:19 AM
> To: Sip Mail List
> Subject: [SIP] 3PCC and the re-invite response
> 
> 
> 
> people,
> 
> i'm slightly concerened (maybe un-justifably) about the re-invite
> working of the 3PCC draft.
> 
> my concern is that it is possible (is it not?) for the user agent
> receiving the re-invite to change its SDP settings in its response as
> follows:
> 
>  A                Controller            B
>  |  INV held SDP     |                  |
>  |<------------------|                  |
>  |                   |                  |
>  |  200 SDP A1       |                  |
>  |-----------------> |  INV SDP A1      |
>  |  ACK              |----------------->|
>  |<----------------- |                  |
>  |                   |  200 SDP B1      |
>  |                   |<-----------------|
>  |                   |                  |
>  |                   |  ACK             |
>  |  INV SDP B1       |----------------->|
>  |<------------------|                  |
>  |  200 OK SDP A2    |                  |
>  |------------------>|                  |
>  |  ACK              |                  |
>  |<------------------|                  |
>  |                                      |
>  |               BROKEN RTP             |
>  |                                      |
>  |                                      |
>  |                                      |
> 
> NOTE: A has returned different SDP in its two different 200 
> OK messages.
> 
> Why might it do this?  It might not but it is possible is it not.
> 
> perhaps we should mandate that an agent can only change its SDP for a
> session in an INVITE not in a response (after the initial
> INVITE/RESPONSE exchange of course)
> 
> what do we think?
> 
> -- 
> Gethin Liddell
> Ubiquity Software Corporation
> 
> http://www.ubiquity.net
> mailto:gethin@ubiquity.net
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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From: Billy Biggs <Billy_Biggs@3com.com>
To: Robert Sparks <rsparks@dynamicsoft.com>
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Subject: Re: [SIP] Music on Hold in ietf-sip-service-examples-00
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Date: Thu, 30 Nov 2000 10:47:11 -0600

Robert Sparks (rsparks@dynamicsoft.com):

>>>>   So far, the only service I've halted seems to be Music-On-Hold,
>>>> which requires that the far end have some crazy logic to send a BYE
>>>> based on a REFER anyways.  I'm sure there will be other services
>>>> that this device won't be able to handle, but I'm not worried yet.
>>> 
>>> What crazy logic? Building the request based on the method parameter
>>> and possibly header parameter in the URL? 
>> 
>>   I assume you want the BYE to be associated with the music call.
>> Otherwise, that BYE would be sent with a new From tag and no To tag
>> (a new call leg), and would not end the old call.
> 
> Hence To: and From: headers in the URL.

  How can the referrer know what tag the far-end UA used?  As well,
using the supplied to/from is another security no-no.

  REFER should not be a control protocol.  It does not supply enough
information to properly orchestrate these operations.  If you need this
sort of functionality, let's keep it seperate, like PHONECTL.

>>   Aside from that matching, there are also security implications of
>> being able to tell a UA to send any SIP request, especially a BYE.  I
>> don't want a remote UA to tell me to hang up a call.  So if some
>> service requires this, then we have to do some logic to see if the
>> user previously asked us to make a call such that we would allow them
>> to send a BYE for it.  Ugh.
> 
> So your UA has a policy to always 503 sip: BYE URLs. 

  Right now we treat everything as an INVITE, but the plan would be to
send a 403.

>>   So, our security is simple:
>> 
>>   - Remote UA must have a call active to us (no out-of-call REFERs).
>>   - We will only allow REFERs which ask us to INVITE.
>>   - We will only allow being REFERed once (REFER is at the cost of
>>     the old session).
>> 
>>   More priviledges are available through something stronger
>> (PHONECTL), which requires authentication.
> 
> You could require that your REFERs be SIP authenticated. 

  Which would be the same as PHONECTL.  The remote UA needs to know the
phone's password to to do remote phone control operations.

  It doesn't seem reasonable for me to tell my phone's password to
everyone who wants to play me music-on-hold.

-- 
Billy Biggs, 3Com               Email: Billy_Biggs@3com.com
http://www.div8.net/billy/      Phone: Billy_Biggs@sip.3com.com

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"Rosen, Brian" wrote:
> 
> There are a couple of mitigating factors.
> 
> One is that the big firewall vendors are starting to
> commit to supporting SIP. No names, sorry, NDA issues!

For example, Cisco claims to have SIP support in their
latest PIX firewalls 5.2:
http://www.cisco.com/univercd/cc/td/doc/product/iaabu/pix/pix_v52/pixrn521.htm#xtocid2350933

FYI: The "in parallel with the regular firewall" topic
will be discussed at the MidCom BOF

Jiri

-- 
Jiri Kuthan         http://www.fokus.gmd.de/usr/kuthan
		    tel:+49-30-34637271
Internet Telephony: http://www.fokus.gmd.de/glone/projects/ipt

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From: "Tom Tang" <ttang@ipunity.com>
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        "Rosen, Brian" <Brian.Rosen@marconi.com>
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 Jiri :

   Can you elaborate on what operations they support for SIP ?
 I looked through that link and on the Cisco website, but
 nada. Thanks.

 - Tom


-----Original Message-----
From: sip-admin@lists.bell-labs.com
[mailto:sip-admin@lists.bell-labs.com]On Behalf Of Jiri Kuthan
Sent: Thursday, November 30, 2000 9:46 AM
To: Rosen, Brian
Cc: 'Jonathan Rosenberg'; 'Howard Hart'; Henning G. Schulzrinne;
'sip@lists.bell-labs.com'
Subject: Re: [SIP] FW: I-D ACTION:draft-rosenberg-sip-entfw-00.txt


"Rosen, Brian" wrote:
>
> There are a couple of mitigating factors.
>
> One is that the big firewall vendors are starting to
> commit to supporting SIP. No names, sorry, NDA issues!

For example, Cisco claims to have SIP support in their
latest PIX firewalls 5.2:
http://www.cisco.com/univercd/cc/td/doc/product/iaabu/pix/pix_v52/pixrn521.h
tm#xtocid2350933

FYI: The "in parallel with the regular firewall" topic
will be discussed at the MidCom BOF

Jiri

--
Jiri Kuthan         http://www.fokus.gmd.de/usr/kuthan
		    tel:+49-30-34637271
Internet Telephony: http://www.fokus.gmd.de/glone/projects/ipt

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BroadSoft uses the re-INVITE without SDP and ACK
with SDP to avoid the possibility of both sides getting
into the mentioned SDP change loop.

This is why BroadSoft would like to keep the use of
ACK with SDP at least with regard to re-INVITEs.

This is also why it would be nice for rfc2543 to mention
that an INVITE without SDP can be used as a request
to pull the receiver off of hold.  This has been discussed
at the backoff and on the news group, but is currently
not explicitly mentioned in rfc2543.

----- Original Message -----
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
To: 'Gethin Liddell' <gethin@ubiquity.net>; Sip Mail List
<sip@lists.bell-labs.com>
Sent: Thursday, November 30, 2000 11:23 AM
Subject: RE: [SIP] 3PCC and the re-invite response


> This is a recognized problem. Note the following section in the document:
>
>    In addition, note that in Figure 1, the controller sends a re-INVITE
>    to A with the SDP from B. The response to this re-INVITE is a 200 OK
>    that contains "SDP A". If the SDP returned in the re-INVITE response
>    were not the same, the controller would need to initiate a re-INVITE
>    to B with that new SDP. This means that a UA which returns a
>
>
>
> Rosenberg/Peterson/Schulzrinne/Camarillo                     [Page 12]
>
> Internet Draft                    3pcc                 November 22, 2000
>
>
>    different SDP (for example, by changing the ports) in the response to
>    every re-INVITE will trigger an infinite re-INVITE loop from the
>    controller to each controller entity. As such, it is STRONGLY
>    RECOMMENDED that if a re-INVITE does not require a UAS to modify the
>    formulation of the SDP description for a specific stream in the
>    response, the SDP description of that stream in the response MUST be
>    the same. In other words, if a UA was in a session with a single
>    stream, using codec A, and it received a re-INVITE modifying the port
>    to send to, the re-INVITE response MUST be the same as it was to the
>    initial request. However, if the re-INVITE forces the UAS to change
>    codecs, it is acceptable in that case to use a different port in the
>    SDP in the response.
>
>
>
> -Jonathan R.
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
>
> > -----Original Message-----
> > From: Gethin Liddell [mailto:gethin@ubiquity.net]
> > Sent: Thursday, November 30, 2000 7:19 AM
> > To: Sip Mail List
> > Subject: [SIP] 3PCC and the re-invite response
> >
> >
> >
> > people,
> >
> > i'm slightly concerened (maybe un-justifably) about the re-invite
> > working of the 3PCC draft.
> >
> > my concern is that it is possible (is it not?) for the user agent
> > receiving the re-invite to change its SDP settings in its response as
> > follows:
> >
> >  A                Controller            B
> >  |  INV held SDP     |                  |
> >  |<------------------|                  |
> >  |                   |                  |
> >  |  200 SDP A1       |                  |
> >  |-----------------> |  INV SDP A1      |
> >  |  ACK              |----------------->|
> >  |<----------------- |                  |
> >  |                   |  200 SDP B1      |
> >  |                   |<-----------------|
> >  |                   |                  |
> >  |                   |  ACK             |
> >  |  INV SDP B1       |----------------->|
> >  |<------------------|                  |
> >  |  200 OK SDP A2    |                  |
> >  |------------------>|                  |
> >  |  ACK              |                  |
> >  |<------------------|                  |
> >  |                                      |
> >  |               BROKEN RTP             |
> >  |                                      |
> >  |                                      |
> >  |                                      |
> >
> > NOTE: A has returned different SDP in its two different 200
> > OK messages.
> >
> > Why might it do this?  It might not but it is possible is it not.
> >
> > perhaps we should mandate that an agent can only change its SDP for a
> > session in an INVITE not in a response (after the initial
> > INVITE/RESPONSE exchange of course)
> >
> > what do we think?
> >
> > --
> > Gethin Liddell
> > Ubiquity Software Corporation
> >
> > http://www.ubiquity.net
> > mailto:gethin@ubiquity.net
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> >
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>
>


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BroadSoft uses the re-INVITE without SDP and ACK
with SDP to avoid the possibility of both sides getting
into a SDP-change-loop discussed in thread
"[SIP] 3PCC and the re-invite response".

This is why BroadSoft would like to keep the use of
ACK with SDP at least with regard to re-INVITEs.


----- Original Message -----
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
To: <sip@lists.bell-labs.com>
Sent: Thursday, November 30, 2000 3:06 AM
Subject: [SIP] as if there wasn't enough traffic on this darn list...


> So, figuring we didn't all have enough email on the sip list to read (I am
> personally hopelessly behind), let me stir things around some more with
> another proposal for something to ditch from the spec.
>
> Right now, you can carry SDP in the INV/200/ACK exchange in two ways. You
> can send nothing in the INVITE, followed by SDP in the 200, and then SDP
in
> ACK (and fortunately I don't know anyone who actively does this). Or, you
> can send SDP in INVITE, SDP in 200, and nothing in ACK. I also think there
> are several confused implementations that likely send it in all three.
>
> Add to this the unfortunate PRACK mess, which also has been used to send
SDP
> in order to support the manyfolks resource work when used with 3pcc, but
> which is yet unimplemented.
>
> It all adds up to a messy confusion. Its not clear what it means to send
SDP
> in all these places, and how it works.
>
> So, my proposal is that we simplify. We only allow SDP in INVITE and then
> provisionals/200 OK. No SDP in ACK. No SDP in PRACK. (sounds like a Dr.
> Seuss rhyme, doesn't it?)
>
> The latest 3pcc draft no longer recommends using the SDP-in-ACK, since it
> has problems. Given that there don't seem to be useful applications of it
> any longer, and given that, to my knowledge, it is not sent by anyone (at
> least not in any commercially shipping boxes, I hope), I'd like to yank
it.
>
> There is a consequence, and that is dealing with the manyfolks requirement
> for sending an updated SDP to deal with manyfolks+3pcc. My proposal for
that
> is to use a re-invite, rahter than PRACK, to carry the updated SDP. This
> does mean that we need to allow, in SIP, a re-INVITE to arrive while an
> initial INVITE is in progress. Allowing this would also address some of
the
> concerns raised in previous threads about handling requirements for
changes
> in early media.
>
> I don't think its hard to deal with the re-INVITE while INVITE-is-active
> case. The simplest solution is to reject the first INVITE and continue
> processing with the second one, using the new SDP and all.
>
> Anyway, before getting into those details, what does the group think about
> dropping the SDP-in-ACK and SDP-in-PRACK stuff?
>
> -Jonathan R.
>
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>
>


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A while back we had a vigorous discussion going on how to implement the
PBX/Centrex "single line extension" service or get behavior similar to US
residentil "party lines".

I don't believe I've seen any discussion on this topic for a long time.
Anybody still working on it, or can we declare the effort dead for now?

--
Dean


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From sip-admin@lists.bell-labs.com  Thu Nov 30 14:53:14 2000
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From: "Rosen, Brian" <Brian.Rosen@marconi.com>
To: "'Dean Willis'" <dean.willis@softarmor.com>
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Subject: RE: [SIP] Single Line Extension work
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Date: Thu, 30 Nov 2000 14:52:31 -0500

While I was one of the folks who are supposed to be working on it,
I haven't been able to, so for now, I'd consider it dead.

What I would say is until and unless someone brings at least a half baked
draft explaining the issues, and proposing a reasonable mechanism, we
declare it to be not an active WG item.

Brian

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: Thursday, November 30, 2000 2:42 PM
> To: IETF SIP (E-mail)
> Subject: [SIP] Single Line Extension work
> 
> 
> 
> A while back we had a vigorous discussion going on how to 
> implement the
> PBX/Centrex "single line extension" service or get behavior 
> similar to US
> residentil "party lines".
> 
> I don't believe I've seen any discussion on this topic for a 
> long time.
> Anybody still working on it, or can we declare the effort 
> dead for now?
> 
> --
> Dean
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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Subject: RE: [SIP] as if there wasn't enough traffic on this darn list...
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Hmmm, I would simplify by going the OTHER direction and allowing SDP in any leg of an INVITE transaction. The meaning would be coupled with the state the transaction is in. This allows for 3-way negotiation of SDP, and allows very simple rules: the SDP you operate on, and reply to is the one you received in the last Request (INVITE or PRACK).

The simple call scenarios don't get any more complicated: you use the most recent SDP sent, otherwise the most recent one you received.

That gives much more flexibility and eliminates the special case rules as well.

Is the liquid sloshing out of the pot yet?

Dave.

> -----Original Message-----
> From: sip-admin@lists.bell-labs.com 
> [mailto:sip-admin@lists.bell-labs.com]On Behalf Of Jonathan Rosenberg
> Sent: Thursday, November 30, 2000 3:06 AM
> To: 'sip@lists.bell-labs.com'
> Subject: [SIP] as if there wasn't enough traffic on this darn list...
> 
> 
> So, figuring we didn't all have enough email on the sip list 
> to read (I am personally hopelessly behind), let me stir 
> things around some more with another proposal for something 
> to ditch from the spec.
> 
> Right now, you can carry SDP in the INV/200/ACK exchange in 
> two ways. You can send nothing in the INVITE, followed by SDP 
> in the 200, and then SDP in ACK (and fortunately I don't know 
> anyone who actively does this). Or, you can send SDP in 
> INVITE, SDP in 200, and nothing in ACK. I also think there 
> are several confused implementations that likely send it in 
> all three. 
> 
> Add to this the unfortunate PRACK mess, which also has been 
> used to send SDP in order to support the manyfolks resource 
> work when used with 3pcc, but which is yet unimplemented.
> 
> It all adds up to a messy confusion. Its not clear what it 
> means to send SDP in all these places, and how it works.
> 
> So, my proposal is that we simplify. We only allow SDP in 
> INVITE and then provisionals/200 OK. No SDP in ACK. No SDP in 
> PRACK. (sounds like a Dr. Seuss rhyme, doesn't it?)
> 
> The latest 3pcc draft no longer recommends using the 
> SDP-in-ACK, since it has problems. Given that there don't 
> seem to be useful applications of it any longer, and given 
> that, to my knowledge, it is not sent by anyone (at least not 
> in any commercially shipping boxes, I hope), I'd like to yank it.
> 
> There is a consequence, and that is dealing with the 
> manyfolks requirement for sending an updated SDP to deal with 
> manyfolks+3pcc. My proposal for that is to use a re-invite, 
> rahter than PRACK, to carry the updated SDP. This does mean 
> that we need to allow, in SIP, a re-INVITE to arrive while an 
> initial INVITE is in progress. Allowing this would also 
> address some of the concerns raised in previous threads about 
> handling requirements for changes in early media.
> 
> I don't think its hard to deal with the re-INVITE while 
> INVITE-is-active case. The simplest solution is to reject the 
> first INVITE and continue processing with the second one, 
> using the new SDP and all.
> 
> Anyway, before getting into those details, what does the 
> group think about dropping the SDP-in-ACK and SDP-in-PRACK stuff?
> 
> -Jonathan R.
> 
> ---
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>  
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com 
> http://lists.bell-> labs.com/mailman/listinfo/sip
> 
> 


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From: "Dean Willis" <dean.willis@softarmor.com>
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Subject: [SIP] Discussion of Presentations at Upcoming Meeting
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Date: Thu, 30 Nov 2000 16:20:03 -0600
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Folks, the last two meetings have been less satisfying than they might have
been. I tried to wedge too many presentations into too little time and make
everybody happy, and I did not leave enough time for discussion of important
issues.

This leads us to where we are -- behind on our deliverables, uncertain of
what they are or should be, and faced with over a hundred active drafts and
over twenty-five slot requests for the upcoming meetings, totaling a
requested four hours not including any discussion time.

I've heard several requests that that we allocate only one day to the
"presentations" of drafts. This includes any new drafts that have had no
previous discussion on the lists or are not work items of the group. We wold
focus the other day on resolving open issues, the VAST majority of which
have been brought up on the list against the bis draft such as
record-routing in the reverse direction.

We have two sessions -- Monday 1300-1500, and Tuesday 0900-1130. This gives
us four and a half hours. I think we need to make some hard editorial
decions according to the standard priorities:

    Priority 1: Originally chartered work.
    Priority 2: Work we have agreed to add to our charter.
    Priority 3: Everything else.

I propose that we schedule Monday afternoon for discussion and Tuesday
morning for presentations of drafts. I will prioritize the time Tuesday
based on the above priorities, then by date of request arrival.


--
Dean Willis


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From sip-admin@lists.bell-labs.com  Thu Nov 30 17:44:54 2000
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From: "Dean Willis" <dean.willis@softarmor.com>
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Subject: [SIP] Guidelines for Talks and Presentations at IETF 49
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Date: Thu, 30 Nov 2000 16:27:58 -0600
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We're going to have an incredibly packed schedule at IETF 49. It will help
if we can all minimize our presentations.

Here are some suggestions on getting the most out of short presentations.

For "new" material (00 drafts):
    -- Describe the basic problem
    -- Describe the proposed solution briefly
    -- Discuss known issues or stumbling points
    -- Propose further steps

For "updated" material (0[~0]) drafts:
    -- Describe the changes from the previous draft
    -- Summarize issues or discussions
    -- Propose further steps

Thanks.

--
Dean


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From sip-admin@lists.bell-labs.com  Thu Nov 30 20:53:04 2000
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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: "'David Oran'" <oran@cisco.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
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Subject: RE: [SIP] as if there wasn't enough traffic on this darn list...
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Date: Thu, 30 Nov 2000 17:52:08 -0800

> 
> Hmmm, I would simplify by going the OTHER direction and 
> allowing SDP in any leg of an INVITE transaction. The meaning 
> would be coupled with the state the transaction is in. This 
> allows for 3-way negotiation of SDP, and allows very simple 
> rules: the SDP you operate on, and reply to is the one you 
> received in the last Request (INVITE or PRACK).
> 
> The simple call scenarios don't get any more complicated: you 
> use the most recent SDP sent, otherwise the most recent one 
> you received.
> 
> That gives much more flexibility and eliminates the special 
> case rules as well.

It also removes the problem whereby one side doesn't support
unannounced/arbitrary/intermixed vocoder changes or assymetric vocoder
selections.

eg 
INVITE has SDP with m=audio 12150 RTP/AVP 0 2 4 8
200 OK selects m=audio 12150 RTP/AVP 0 2 but caller's UA doesn't support
multiple codecs or wants to use ony one vocoder, so send
ACK re-selecting m=audio 12150 RTP/AVP 0

There _are_ implementations that are commercially shippped which support for
SDP-in-ACK and people have tested this functionality at previous interop
events (even tested with the Dynamicsoft UA). As H.323-SIP gateways start
appearing on the market I think you will find more and more SIP user-agents
wanting to support this function as they will want to interoperate with the
all the Microsoft Netmeetings and existing H.323 equipment. 

I think there will be a lot of hidden problems and complexity with the
suggested re-INVITE before INVITE completes. Implementation complexity (I
feel) will be significantly higher and there great number of different
message permutations that will need to be tested. I can't see it being very
backwardly compatiable either. 

I do believe we need to fixed the re-INVITE glare mechanism which is (IMO)
currently unworkable, this however is a separate problem.

Regards,

Robert.

-- My opinions are my own. I tried selling them once but everybody
	seems to already have one. -- 

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From sip-admin@lists.bell-labs.com  Thu Nov 30 23:50:09 2000
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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: "'Jo Hornsby'" <jhornsby@ubiquity.net>, SIP@lists.bell-labs.com
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Subject: RE: [SIP] RECORD-ROUTE/ROUTE requirements
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Date: Thu, 30 Nov 2000 20:48:38 -0800

> 
> I don't think that proxies should start second-guessing UAS.
> I would say that the simplest thing is just to say that when the
> UA is building the Route header list, it should append the
> Contact, if present; failing that, the From.
> 
> Really old UAs might lose, but they were always going to do
> that, anyway[0].
> 
What about in the other direction? The UAC will need to add a Record-Route
for the _original_ Request-URI - which may in fact be entirely unhelpful. If
the proxy were to add the extra Record-Route I think that is much safer as
it knows what its own behaviour will be when it receives the request..

I guess, I was also trying to stick with the existing RFC2543 and bis UA
behaviour (minus route mangling). Ie, the UA doesn't add an extra
Route-Header for the original From or Request-URI - its only added if there
is a Contact (which I think is a good thing !!). 

I think we can do better than saying RFC2543 doesn't work. 

- If a proxy inserts Record-Route header into a request without neither a
Contact nor Record-Route header, then the Proxy MAY add a second
Record-Route header to the end of the R-R list using the From URL address
with the source address as an maddr parameter.

- If a proxy receives a response to a request in which it added a
Record-Route header AND the response does not contain a COntact header AND
no new headers have been added by other proxies (ie proxy is final
Record-Route'd proxy), then the proxy MAY add another preceeding
Record-Route header to the start of the response's R-R list using the
request's request-URI with the request's forward address as an maddr
parameter.

This should cover the majority of RFC2543 scenarios without too much fuss
for the proxy or UA - I think it is better than trying to embed using r-r
parameters as this just gets messya nd brittle. As I said before, I think it
is better that the proxy does this as it understand that it will use the
added Record-Route. If the UA tries this, it has little chance of success. A
UA has never (if I remember correctly) added extra R-R headers if there is
no Contact (and shouldn't matter even if it did).

What do you think ?

Robert.

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