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From: Michael Teroerde <michael.teroerde@gmx.de>
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Hello !!

I have searched your Email-list for Questions like mine, but there
wasn't
a posting which answerd all my questions.

I would know something about the REGISTER- Request in the SIP-Language.

1. Could a Server send me an De-Register message if he want (e.g. if he
wants to reboot)??
2. Could a Server change the "Expires"- Header field (and therefore
 change the Register- Time)??
3. Did the Server send somthing as "Keep Alive" to the registerd
 UserAgent??(or did he make a "ping" to the IP or something else to
determine that my UA is still active)
4. What happens if my UserAgent hangs up??
5. Is there a generelly timout when the Server delets the
register-entry   (e.g. after 48 hours) ??
6. Did the server MUST send a "200 OK"- Response if the UserAgent send a
 De-register(REGISTER with Expires=0) ??

Please help me !!!
If you know an interesting Paper about generelly Sever behavior, please
tell it to me.


Thanks for your effort

Michael Teroerde

(instead of posting it on the List , you can also send it to my
NEC-Email-Adress)
----------------------------------------------------------------
Michael Teroerde
Student at NEC- Heidelberg
michael.teroerde@ccrle.nec.de

-- 
Sent through Global Message Exchange - http://www.gmx.net




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1. Could a Server send me an De-Register message if he want (e.g. if he
wants to reboot)??

Arjun> Actually this issue did come up before and I think it was decided
then
Arjun> that its not important (however, I would have preferred if there
were a well stated
Arjun> way, which you could use if you want)
Arjun> I have a question- since INFO can carry any information, can I use
INFO for this purpose ?

2. Could a Server change the "Expires"- Header field (and therefore
 change the Register- Time)??

Arjun> Yes, the Registrar can select a higher value of Expires, but not a
lower one. (4.2.6, RFC 2543)

3. Did the Server send somthing as "Keep Alive" to the registerd
 UserAgent??(or did he make a "ping" to the IP or something else to
determine that my UA is still active)

Arjun> Is there any defined message for this from server-client ? Or is it
that inactivity
Arjun> can be figured out by other means (eg no RTP packets during call,
timeout, etc ?)

4. What happens if my UserAgent hangs up??

Arjun> In what state ?

5. Is there a generelly timout when the Server delets the
register-entry   (e.g. after 48 hours) ??

Arjun>  If nothing is specified in the REGISTER request, 60 mins is the
default (4.2.6, RFC 2543)

6. Did the server MUST send a "200 OK"- Response if the UserAgent send a
 De-register(REGISTER with Expires=0) ??

Arjun> Yes - This OK is to tell the sender that it received the REGISTER
message.
Arjun> You are probably wondering what happens if the client sends and
immediately exits
Arjun> without waiting for a response - then registrar transmits 200OK once
and stops anyway


Regds
Arjun
--
Arjun Roychowdhury @ Hughes Software Systems





From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb  1 04:57:48 2000
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From: "Christian Jansson" <Christian.Jansson@hotsip.com>
To: "Michael Teroerde" <michael.teroerde@gmx.de>,
        <sip@lists.research.bell-labs.com>
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Read the section about the REGISTER request, or have a look at my
answers below.

>1. Could a Server send me an De-Register message if he want (e.g. if he
>wants to reboot)??

No.

>2. Could a Server change the "Expires"- Header field (and therefore
> change the Register- Time)??

Yes, if it is done while handeling a register request. "A server SHOULD NOT
use a higher lifetime than the one requested, but MAY use a lower one.
"As the server SOULD return the current list of registrations in the 200
response,
 the client can find out what Expire value the server chose.
If the question is, "Can the server change the Expires value _after_ the 200
response has been sent?", the answer is no.

>3. Did the Server send somthing as "Keep Alive" to the registerd
> UserAgent??(or did he make a "ping" to the IP or something else to
>determine that my UA is still active)

No.

>4. What happens if my UserAgent hangs up??

Depends on your implementation of "hang up". If you mean that the UserAgent
abruptly dies, the location server will not get any notification of that.

>5. Is there a generelly timout when the Server delets the
>register-entry   (e.g. after 48 hours) ??

If no Expires header or expires Contact parameter is present, one hour is
assumed
by the server.

>6. Did the server MUST send a "200 OK"- Response if the UserAgent send a
> De-register(REGISTER with Expires=0) ??

Yes.

>
>Please help me !!!
>If you know an interesting Paper about generelly Sever behavior, please
>tell it to me.
>
>
>Thanks for your effort
>
>Michael Teroerde
>

--
Christian Jansson (christian@hotsip.com)
HotSip AB
phone:  +46 (0)8 555 10 704
mobile: +46 (0)70 558 1038








From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb  1 07:32:01 2000
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From: "Gazal, Elly" <Elly_Gazal@icomverse.com>
To: sip@lists.research.bell-labs.com
Subject: Media payload type
Date: Tue, 1 Feb 2000 14:27:49 +0200 
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Does anyone know where can I find the media payload type list/table (0=mlaw
PCM, .....,3=GSM, etc.)?


Elly Gazal
System Engineer

Signaling R&D Group
Comverse Network Systems Ltd.

Tel:  972-3-6458994
Fax: 972-3-6458915
Email: Elly_Gazal@icomverse.com




From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb  1 07:37:58 2000
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To: "Gazal, Elly" <Elly_Gazal@icomverse.com>
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Subject: Re: Media payload type
In-Reply-To: Message from "Gazal, Elly" <Elly_Gazal@icomverse.com> of "Tue, 01 Feb 2000 14:27:49 +0200." <CE835E918749D21191B10060084C377E016D1E75@ismail1.icomverse.com>
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--> "Gazal, Elly" writes:
>Does anyone know where can I find the media payload type list/table (0=mlaw
>PCM, .....,3=GSM, etc.)?

See RFC 1890. This question is best asked on the AVT working group's list
rem-conf@es.net since it's not directly SIP related.

Colin



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archow@hss.hns.com wrote:
> 2. Could a Server change the "Expires"- Header field (and therefore
>  change the Register- Time)??
> 
> Arjun> Yes, the Registrar can select a higher value of Expires, but not a
> lower one. (4.2.6, RFC 2543)

It's the other way around (a quote from 4.3.6 below):

A server SHOULD NOT use a higher lifetime than the one requested, but
MAY use a lower one.

---
Igor Slepchin



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb  1 14:48:10 2000
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Date: Tue, 01 Feb 2000 14:45:58 -0500
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To add a few comments to what has been said already:

Michael Teroerde wrote:
> 
> Hello !!
> 
> I have searched your Email-list for Questions like mine, but there
> wasn't
> a posting which answerd all my questions.
> 
> I would know something about the REGISTER- Request in the SIP-Language.
> 
> 1. Could a Server send me an De-Register message if he want (e.g. if he
> wants to reboot)??

The decision was made not to include such a function as it can be
accomplished in many other ways. For example, by changing the DNS
entries for the server, or by mandating multicast for registrations. The
latter makes reboot handling really easy. Adding a de-register message
has other problems. To what clients is it sent? What about clients that
aren't registered right now but are going to do so after the server
reboots? You need to tell them to try a different server also.


> 2. Could a Server change the "Expires"- Header field (and therefore
>  change the Register- Time)??

This has been answered by others.

> 3. Did the Server send somthing as "Keep Alive" to the registerd
>  UserAgent??(or did he make a "ping" to the IP or something else to
> determine that my UA is still active)

Registrations are kept alive through refresh. The server can reduce the
refresh interval by placing it in the 200 response to the registration,
as has been mentioned.

SIP sessions, right now, have no keepalive. It is done at the moment
through the media session itself. However, a specification is in
development for a keepalive within SIP itself:

http://www.ietf.org/internet-drafts/draft-ietf-sip-session-timer-00.txt

Note that the call is always up so long as the end systems believe it
is. If a proxy uses the session timer, and the call times out, the call
may still be up if the end systems think it is.

> 4. What happens if my UserAgent hangs up??

Like others, I don't understand this question.

> 5. Is there a generelly timout when the Server delets the
> register-entry   (e.g. after 48 hours) ??

Been answered by others.

> 6. Did the server MUST send a "200 OK"- Response if the UserAgent send a
>  De-register(REGISTER with Expires=0) ??

All requests generate responses. 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb  1 17:28:08 2000
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>> However, if  6.6 states it as a MUST, which implies that a person who
sends
>> CR orLF only is not SIP compliant.

> It's one of the overarching philosophies of IETF protocols:
> be strict in what you generate, but lenient in what you accept.
> That way, for two implementations of a protocol to be
> non-interoperable, they must *both* be broken.

Or one of them must be broken in a way other than one of the few specific
minor ways in which it's "OK" to be broken, despite the fact that it's not
compliant to be broken.

> This is one of the reasons occasionally cited why IETF protocols
> tend to interoperate more sucessfully than those of other
> standards bodies.

Can't argue with that, though I have a lot of sympathy for the other side.
If I can't trust my counterpart to be able to follow the standard well
enough to send the proper CRLF (even though I can cover for it), can I trust
it to do anything right?

Tim.



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There are pros and cons. The one and only pro is saving space in the
message.

The cons are more architectural in nature. It creates a special case for
naming conventions, creates an implicit namespace for this first class
set of extensions. Henning pointed out to me another interesting issue -
lets say RTSP wants to use some new header, foo, defined for SIP. Would
this extension be referred to as foo within RTSP, or org.ietf.sip.foo?

-Jonathan R.

Michelle Kitchings wrote:
> 
> Jonathan,
> 
> I agree with presence-imply and wasn't attempting to re-open that particular
> issue.
> 
> But, the whole discussion made me think of the idea of an implied
> "org.ietf.sip" in general.  Do you think the idea has merit as a means to
> shortening messages that have Require, Proxy-require, Supported, and
> Unsupported?  Like someone said in another post, we could end up having dozens
> of IETF extensions.
> 
> Cheers,
> Michelle :^)
> 
> Jonathan Rosenberg wrote:
> >
> > I think we have clear consensus for having the presence of Supported
> > imply support for org.ietf.sip.supported.
> >
> > Issue closed.
> >
> > -Jonathan R.
> >
> > > Michelle Kitchings wrote:
> > > >
> ...
> > > > Here is my first post, so bear with me.  :)
> > > >
> > > > If the motivation for making "Supported:" imply "org.ietf.sip.supported" is
> > > > because of the extensive length of the feature name, couldn't the feature
> > > > lists be interpreted such that any feature without a reverse domain is assumed
> > > > to be "org.ietf.sip"?  This would mean that
> > > >
> > > >   supported = org.ietf.sip.supported
> > > >   100rel = org.ietf.sip.100rel
> > > >   info = org.ietf.sip.info
> > > >   etc. ...
> > > >
> > > > This mechanism makes any/all IETF extension cheaper in terms of bandwidth.
> > > >
> > > > The downfall to my idea is that if a provider (eg, Ericsson) defines an
> > > > extension called "info" without using a reverse domain, then the servers could
> > > > inadvertantly interpet that as the wrong kind of info--which would happen
> > > > anyway if some other provider (eg, MCI) was also using info without reverse
> > > > domain.  Perhaps this default domain interpretation would "blackmail"
> > > > providers into making sure that they fully-qualify all features?
> > > >
> > > > Cheers,
> > > > Michelle :^)
> > > >
> > > > --
> > > > Michelle Kitchings | Ph:  +1 972 583 7101 | 1010 E. Arapaho, MS L-04
> > > > Ericsson, Inc.     | Fax: +1 972 669 0154 | Richardson, TX 75081 USA

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Hi,
I had posted this some time earlier  - never received my own mail thru the
list - maybe it got swallowed somewhere.
Regds
Arjun

---------------------- Forwarded by Arjun Roychowdhury/HSSBLR on 02/02/2000
08:55 AM ---------------------------


Arjun Roychowdhury
01/31/2000 07:05 PM

To:   sip@lists.research.bell-labs.com
cc:

Subject:  Encryption of From Header

Hi,
Please refer to section 13.1.1 of RFC 2543

"The From header field SHOULD
   normally remain in the clear, but MAY be encrypted if required, in
   which case some proxies MAY return a 401 (Unauthorized) status if
   they require a From field."

The above indicates that its possible for FROM Header to be encrypted if so
 desired.

Now, please refer to Section 6 Table 4 which clearly mentions From (and
others) to have an "n" status in the enc colum implying
they MUST NOT be encrypted.

So which is the correct statememt ?

Regds
Arjun
--
Arjun Roychowdhury @ Hughes Software Systems








From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb  2 05:54:13 2000
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Right, so there's one pro and one cons. Neither is entirely compelling
in itself. I would point out, though, that the large majority of
extensions to come into wide use will likely become standardized under
the org.ietf.sip namespace. I believe this is true for all extensions
currently planned and this IMHO somewhat weakens the cons argument.

Jonathan Rosenberg wrote:
> 
> There are pros and cons. The one and only pro is saving space in the
> message.
> 
> The cons are more architectural in nature. It creates a special case for
> naming conventions, creates an implicit namespace for this first class
> set of extensions. Henning pointed out to me another interesting issue -
> lets say RTSP wants to use some new header, foo, defined for SIP. Would
> this extension be referred to as foo within RTSP, or org.ietf.sip.foo?

Presumably, if an extension *would* have fallen under the org.ietf.sip
namespace under the proposed scheme it would fall under the "default"
namespace.

Anders

-- 
Anders Kristensen <ak@hplb.hpl.hp.com>,
http://www-uk.hpl.hp.com/people/ak/
Hewlett-Packard Labs, Bristol, UK



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb  2 08:22:13 2000
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From: "Masanori Inui" <inui@iwatsu.co.jp>
To: "SIP-MailingList" <sip@lists.research.bell-labs.com>
Subject: Regarding the reliability via UDP
Date: Wed, 2 Feb 2000 22:17:56 +0900
Organization: IWATSU Electric CO.,LTD.
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Hi,

The spec. of SIP define how to ensure the reliability, if the entity is
using UDP.
My understanding of reliability is bellow. Please see those diagrams.
So, I have questions.

-Is my understanding correct?
-On (2),(3) and (4),the server goes to initial state(no cache the result)
 ,once the server send a 1st response.If the server receive the
re-transmitted request,
 the server send a 481 final response.Is the SIP allow this implementation?

I worry that the server can not accept a new call depend on cache space(real
memory),
because, I think that the cache time is too long.

(1)200 or 3xx response for INVITE

UAC          UAS
 ---INVITE---->
 <--100--------
 <--180--------
 <--200--------  + start of the timer for reliability
                 | (T1 * 2^n)
 <--200--------  + if UAS has not received ACK, BYE or CANCEL then
retransmit
                 |
                 |
                 + if UAS has not received ACK, BYE or CANCEL then
retransmit
                 |
                 v and, so on until the response has been transmitted 7times

(2)no-200 or non-3xx final response for INVITE

UAC          UAS
 ---INVITE---->
 <--100--------
 <--180--------
 <--xxx-------- + start of the timer for reliability
                | (T1 * 2^n)
 <--xxx-------- + if UAS has not received ACK, BYE or CANCEL then retransmit
                |
                |
                + if UAS has not received ACK, BYE or CANCEL then retransmit
                |
                v and, so on until the response has been transmitted 7times


(3)200 response for BYE,CANCEL,OPTIONS,REGISTER

UAC          SERVER
 ---BYE------->
 <--100--------
 <--200--------  + start of the timer for reliability
                 | (T2 * 10)
 ---BYE------->  + if SERVER has received BYE(re-transmit), then retransmit
 <--200--------  |
                 |
                 v and, so on until the response has been elapsed T2*10

(4)no-200 final response for BYE,CANCEL,OPTIONS,REGISTER

The same behavior as (3)

Best regards,

===========================
Masanori Inui
IWATSU Electric CO.,LTD
Project Team No.2
Solution Engineering Dept.
Engineering Division
---------------------------
1-7-41 Kugayama Suginami-ku
Tokyo, 168-8501, Japan
---------------------------
Tel.+81-3-5370-5211
Fax.+81-3-5370-5193
Email inui@iwatsu.co.jp
===========================





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Date: Wed, 02 Feb 2000 05:45:50 -0800
From: "Liberati, Ernest" <Ernest.Liberati@GD-ES.COM>
Subject: Call legs and Cseq
To: "'sip@lists.research.bell-labs.com'" <sip@lists.research.bell-labs.com>
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Hi,

For a SIP call that involves phone with UAC, host with UAS, and Proxy Server
components, shown below, I need some help to clear my confusion on call legs
and Cseq.



1. Do call legs exist between all adjacent components (e.g., A - B, B - C,
...)?

2. Does a call leg include the requests in both directions (e.g., B - C and
C - B)?

3. The most recent definition of a call leg, that I have found, is a
combination of local- and remote-addresses (including tags) and the Call-ID.
How do the Request URI, To, and From address relate to local- and
remote-addresses?

4. Within a call leg, is the Cseq incremented for each transaction in each
direction?


Thanks in advance,
Ernest Liberati
General Dynamics Electronic Systems
301 310-0344
ernest.liberati@gd-es.com



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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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I'm trying to reach closure on a minor syntax issue.

Currently, the "tag=" parameter is a hexadecimal string (plus dash).

It has been proposed

Advantage of the change:
- one fewer syntax type to worry about;
- currently, the similar-in-spirit Via branch parameter is a token, so
this would unify the two;
- makes it easy to encode information strings as base64;
- higher information density (base64 vs. base16)
- it's easy to forget this little hex limitation, so implementations may
need to be "forgiving" in any event if one major one gets it wrong;

Disadvantage:
- it's a change from RFC 2543 and not backwards-compatible.

Questions:

1) If you are an implementor: does your implementation in any way depend
on tag being hexadecimal?

2) Any other opinions?

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



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To: "Liberati, Ernest" <Ernest.Liberati@GD-ES.COM>
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Subject: Re: Call legs and Cseq
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Ernest,

Your confusion is quite understandable.

A SIP call leg denotes the (abstract) notion of an end-to-end
connection, regardless of whether any proxies are on the signaling path.
This means that (in SIP terminology) there are no A-B and B-C legs but
only an A-C leg. Obviously a (forking) proxy needs to know about those
other "leg segments" but they are not considered legs as such. This is
different from the PSTN switch notion of legs.

As for the CSeq number, the UAs involved in a leg choose CSeq numbers in
increasing order for different transactions but independent of each
other. The fact that they can initiate transactions at overlapping
points in time implies that the CSeq number spaces are separate in the
two directions, i.e. the transaction {i,A,C,42} is *not* the same
transaction as {i,C,A,42} even though the legs {i,A,C} is the same as
the leg {i,C,A} where i is the Call-ID and 42 is the CSeq.

"Liberati, Ernest" wrote:
> 
> Hi,
> 
> For a SIP call that involves phone with UAC, host with UAS, and Proxy Server
> components, shown below, I need some help to clear my confusion on call legs
> and Cseq.
> 
> 1. Do call legs exist between all adjacent components (e.g., A - B, B - C,
> ...)?
> 
> 2. Does a call leg include the requests in both directions (e.g., B - C and
> C - B)?
> 
> 3. The most recent definition of a call leg, that I have found, is a
> combination of local- and remote-addresses (including tags) and the Call-ID.
> How do the Request URI, To, and From address relate to local- and
> remote-addresses?
> 
> 4. Within a call leg, is the Cseq incremented for each transaction in each
> direction?
> 

-- 
Anders Kristensen <ak@hplb.hpl.hp.com>,
http://www-uk.hpl.hp.com/people/ak/
Hewlett-Packard Labs, Bristol, UK



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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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Henning Schulzrinne wrote before having his first dose of caffeine:

> 
> Currently, the "tag=" parameter is a hexadecimal string (plus dash).
> 
> It has been proposed
> 
and forgot to add:

to change this to "token" instead (which is a superset of the
hexadecimal + hyphen syntax currently allowed).

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



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Hi,
My 2c :

Id prefer token for tag. Advantages: as you mentioned
Let the originator of the tag decide how he wants to correlate - I dont see
any reason why it _must_ be hex - infact its easy to simply
apply functions of say getpid()+time() or something and generate a
corresponding tag correlator


hgs> 1) If you are an implementor: does your implementation in any way
depend
hgs> on tag being hexadecimal?

No, it does not.

Regds
Arjun
--
Arjun Roychowdhury @ Hughes Software Systems





Henning Schulzrinne <schulzrinne@cs.columbia.edu> on 02/02/2000 07:37:50 PM

To:   sip@lists.research.bell-labs.com
cc:

Subject:  From Tag syntax




I'm trying to reach closure on a minor syntax issue.

Currently, the "tag=" parameter is a hexadecimal string (plus dash).

It has been proposed

Advantage of the change:
- one fewer syntax type to worry about;
- currently, the similar-in-spirit Via branch parameter is a token, so
this would unify the two;
- makes it easy to encode information strings as base64;
- higher information density (base64 vs. base16)
- it's easy to forget this little hex limitation, so implementations may
need to be "forgiving" in any event if one major one gets it wrong;

Disadvantage:
- it's a change from RFC 2543 and not backwards-compatible.

Questions:

1) If you are an implementor: does your implementation in any way depend
on tag being hexadecimal?

2) Any other opinions?

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








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I also want to add that we'll be taking a poll on this at the next
bakeoff. We don't want to make this change if implementations are
adversely affected. So long as none are, I think this change is good
(our implementation doesn't depend on hex either).

-Jonathan R.

Henning Schulzrinne wrote:
> 
> Questions:
> 
> 1) If you are an implementor: does your implementation in any way depend
> on tag being hexadecimal?
> 
> 2) Any other opinions?
> 
> Thanks.
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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        "'sip@lists.research.bell-labs.com'" <sip@lists.research.bell-labs.com>
Subject: Re: Call legs and Cseq
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Anders has it right. Some additional clarifications:

Anders Kristensen wrote:
> 
> > 2. Does a call leg include the requests in both directions (e.g., B - C and
> > C - B)?

As Anders has pointed out, call legs really refer to end to end
connections. So, B and C are user agents. Given that, the call leg just
refers to the relationship between them. Within a call leg, there are
numerous transactions in both directions. 

> >
> > 3. The most recent definition of a call leg, that I have found, is a
> > combination of local- and remote-addresses (including tags) and the Call-ID.
> > How do the Request URI, To, and From address relate to local- and
> > remote-addresses?

The request URI does not used in call leg identification.

The To and From field relate to local and remote in the following way.
WHen I send a request on a call leg, the From field contains the local
address, and the To field the remote address. When a request is
received, the To field is matched to the local address, and the From
field to the remote address.

> >
> > 4. Within a call leg, is the Cseq incremented for each transaction in each
> > direction?

As Anders has pointed out, the CSeq space in the two directions are
independent. Within a single direction (A to B), they do increment for
each transaction.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Masanori Inui wrote:
> 
> Hi,
> 
> The spec. of SIP define how to ensure the reliability, if the entity is
> using UDP.
> My understanding of reliability is bellow. Please see those diagrams.
> So, I have questions.
> 
> -Is my understanding correct?

Generally looks right. There is no difference in user agent behavior for
the differing final responses to INVITE and non-INVITE requests (you
have two diagrams for INVITE, one for 200 or 3xx, and another for
everything else, but they are the same). Your diagrams don't show it,
but for INVITE the requests are retransmitted with an exponentially
increasing interval until a response is received (provisional or
otherwise).

Also, whilst the spec talks about stopping retransmissions of the INVITE
response if a BYE is received, I think the next revision of the spec
will not recommend this. The problem is for proxies. Since they don't
necessarily have call state, they will end up having two transactions,
one of which never completes. This causes stale transaction state to be
held in a proxy for longer than needed. As much as possible, I'd like to
keep transactions independent of each other. 

For a similar reason, the new recommended behavior for receiving CANCEL
is to keep retransmitting the final response if you've already sent it.
Otherwise, if the final response and the CANCEL cross on the wire, but
the final response is lost, the server thinks the call is set up, but
the client won't know since it will never receive a response to the
INVITE. Similarly, if the server hasn't sent a final response to an
INVITE, the server should send a 487 response to the INVITE when a
CANCEL comes. This ensures that the original INVITE transaction
completes like a normal transaction; CANCEL just speeds it up. 


> -On (2),(3) and (4),the server goes to initial state(no cache the result)
>  ,once the server send a 1st response.If the server receive the
> re-transmitted request,
>  the server send a 481 final response.Is the SIP allow this implementation?

For a user agent server, no. It must cache the response (I think for 60
seconds), or regenerate the same response (possible if the initial
response was some kind of error because the request was bad). Otherwise,
the reliability doesn't work.

For a proxy or redirect server which generates its own response, its the
same as UAS. 

For a stateful proxy that forwards a response it received, it must also
cache the response.

A stateless proxy need not cache a response. When a request
retransmission is received, it forwards it as if it was the initial
request (it can't tell the difference between a retransmission and the
original request).

> 
> I worry that the server can not accept a new call depend on cache space(real
> memory),
> because, I think that the cache time is too long.

What kind of server? For a UAS, you still need to store call state
anyway. You will need to store things like CSeq numbers, routes,
credentials, To and From fields, and so on. I can't imagine that for a
UAS handling high call volumes (a gateway typically) the memory
requirements for the responses are a bottleneck. 

If you are worried about memory requirements in a proxy, use a stateless
proxy. They don't need to cache anything.


> 
> (1)200 or 3xx response for INVITE
> 
> UAC          UAS
>  ---INVITE---->
>  <--100--------
>  <--180--------
>  <--200--------  + start of the timer for reliability
>                  | (T1 * 2^n)
>  <--200--------  + if UAS has not received ACK, BYE or CANCEL then
> retransmit
>                  |
>                  |
>                  + if UAS has not received ACK, BYE or CANCEL then
> retransmit
>                  |
>                  v and, so on until the response has been transmitted 7times
> 
> (2)no-200 or non-3xx final response for INVITE
> 
> UAC          UAS
>  ---INVITE---->
>  <--100--------
>  <--180--------
>  <--xxx-------- + start of the timer for reliability
>                 | (T1 * 2^n)
>  <--xxx-------- + if UAS has not received ACK, BYE or CANCEL then retransmit
>                 |
>                 |
>                 + if UAS has not received ACK, BYE or CANCEL then retransmit
>                 |
>                 v and, so on until the response has been transmitted 7times
> 
> (3)200 response for BYE,CANCEL,OPTIONS,REGISTER
> 
> UAC          SERVER
>  ---BYE------->
>  <--100--------
>  <--200--------  + start of the timer for reliability
>                  | (T2 * 10)
>  ---BYE------->  + if SERVER has received BYE(re-transmit), then retransmit
>  <--200--------  |
>                  |
>                  v and, so on until the response has been elapsed T2*10
> 
> (4)no-200 final response for BYE,CANCEL,OPTIONS,REGISTER
> 
> The same behavior as (3)
> 
> Best regards,
> 
> ===========================
> Masanori Inui
> IWATSU Electric CO.,LTD
> Project Team No.2
> Solution Engineering Dept.
> Engineering Division
> ---------------------------
> 1-7-41 Kugayama Suginami-ku
> Tokyo, 168-8501, Japan
> ---------------------------
> Tel.+81-3-5370-5211
> Fax.+81-3-5370-5193
> Email inui@iwatsu.co.jp
> ===========================

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb  2 11:03:05 2000
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Date: Wed, 02 Feb 2000 16:58:52 +0100
From: Rabbe Fogelholm <Rabbe.Fogelholm@uab.ericsson.se>
Reply-To: Rabbe Fogelholm <Rabbe.Fogelholm@ericsson.com>
To: sip@lists.research.bell-labs.com, sip@cslab.ericsson.se,
        sipfactory@uab.ericsson.se
Cc: schulzrinne@cs.columbia.edu
Subject: New public SIP server
Message-ID: <322959594.949510732@upc1390.uab.ericsson.se>
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This message is to announce a "new" public SIP server (actually it is a
reincarnation of the Ellemtel SIP server that used to be running at
sip.pcs.ellemtel.net:5060 in 1999). Apologies if you receive this
message in duplicate.

Henning, here are data for your SIP server page
(http://www.cs.columbia.edu/~hgs/sip/servers.html):

Organization:
  Ericsson (http://www.ericsson.se)

SIP URL:
  sip:Robot@iponax.com

SRV records:
  iponax.com

Transport:
  UDP, TCP

Modes:
  redirect

Accept registrations from other domains?
  yes

Remarks:
  The sip:Robot@iponax.com accepts INVITEs to VoIP sessions.
  It will play a voice announcement of about 30 s length.

Comments etc are welcome. Enjoy,

Rabbe.Fogelholm@ericsson.com



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb  2 11:49:54 2000
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From: Itamar Gilad <ItamarG@tlv.radvision.com>
To: "'Anders Kristensen'" <ak@hplb.hpl.hp.com>
Cc: "'sip@lists.research.bell-labs.com'" <sip@lists.research.bell-labs.com>
Subject: RE: Call legs and Cseq
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Just to make sure:

reqMsg1 == reqMsg2 (duplicate requests) IFF the To, From, Call-ID and CSeq
headers are identical (addresses and tags only in the To and From headers).

msg1 and msg2 belong to the same Call-leg if 
    the Call-ID is the same and 
    (msg1.To == msg2.To && msg1.From == msg2.From) || 
    (msg1.To == msg2.From && msg1.From == msg2.To)

( ==  means the addresses and tags are the same)

Is that right? 

BTW, you cannot identify duplicate responses using the first rule.

Itamar

> -----Original Message-----
> From: Anders Kristensen [mailto:ak@hplb.hpl.hp.com]
> Sent: Wednesday, February 02, 2000 3:17 PM
> To: Liberati, Ernest
> Cc: 'sip@lists.research.bell-labs.com'
> Subject: Re: Call legs and Cseq
> 
> 
> Ernest,
> 
> Your confusion is quite understandable.
> 
> A SIP call leg denotes the (abstract) notion of an end-to-end
> connection, regardless of whether any proxies are on the 
> signaling path.
> This means that (in SIP terminology) there are no A-B and B-C legs but
> only an A-C leg. Obviously a (forking) proxy needs to know about those
> other "leg segments" but they are not considered legs as such. This is
> different from the PSTN switch notion of legs.
> 
> As for the CSeq number, the UAs involved in a leg choose CSeq 
> numbers in
> increasing order for different transactions but independent of each
> other. The fact that they can initiate transactions at overlapping
> points in time implies that the CSeq number spaces are separate in the
> two directions, i.e. the transaction {i,A,C,42} is *not* the same
> transaction as {i,C,A,42} even though the legs {i,A,C} is the same as
> the leg {i,C,A} where i is the Call-ID and 42 is the CSeq.
> 
> "Liberati, Ernest" wrote:
> > 
> > Hi,
> > 
> > For a SIP call that involves phone with UAC, host with UAS, 
> and Proxy Server
> > components, shown below, I need some help to clear my 
> confusion on call legs
> > and Cseq.
> > 
> > 1. Do call legs exist between all adjacent components 
> (e.g., A - B, B - C,
> > ...)?
> > 
> > 2. Does a call leg include the requests in both directions 
> (e.g., B - C and
> > C - B)?
> > 
> > 3. The most recent definition of a call leg, that I have found, is a
> > combination of local- and remote-addresses (including tags) 
> and the Call-ID.
> > How do the Request URI, To, and From address relate to local- and
> > remote-addresses?
> > 
> > 4. Within a call leg, is the Cseq incremented for each 
> transaction in each
> > direction?
> > 
> 
> -- 
> Anders Kristensen <ak@hplb.hpl.hp.com>,
> http://www-uk.hpl.hp.com/people/ak/
> Hewlett-Packard Labs, Bristol, UK
> 



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Subject: Re: From Tag syntax
To: schulzrinne@cs.columbia.edu (Henning Schulzrinne)
Date: Wed, 2 Feb 2000 11:02:20 -0600 (CST)
Cc: sip@lists.research.bell-labs.com
In-Reply-To: <38983A36.27CBA54F@cs.columbia.edu> from "Henning Schulzrinne" at Feb 02, 2000 09:07:50 AM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>1) If you are an implementor: does your implementation in any way depend
>on tag being hexadecimal?

Ours does not. While helping another team to debug what was going
on at the most recent bakeoff, we discovered that at least one
implementation sends back a 500 class message when they receive
a tag which contains non-hex. However, it is also worth noting
that this problem surfaced with *only* one implementation -- so,
we can probably assume that all of the implementations at the
bakeoff except for one accepted non-hex tags.

I'm all for allowing the tags to be tokens.

-- 
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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Itamar Gilad wrote:
> 
> Just to make sure:
> 
> reqMsg1 == reqMsg2 (duplicate requests) IFF the To, From, Call-ID and CSeq
> headers are identical (addresses and tags only in the To and From headers).

Yes. Obviously, reqMsg1 and reqMsg2 may in fact differ in other headers
(although this should never happen), but for the purpose of identifying
and filtering out retransmissions they're the same.

> 
> msg1 and msg2 belong to the same Call-leg if
>     the Call-ID is the same and
>     (msg1.To == msg2.To && msg1.From == msg2.From) ||
>     (msg1.To == msg2.From && msg1.From == msg2.To)
> 
> ( ==  means the addresses and tags are the same)
> 
> Is that right?

Yes. 

> 
> BTW, you cannot identify duplicate responses using the first rule.

No, not quite. If the request was sent without a To tag the UAC needs to
take into account the possibility that a UAS has added this to the
response, and must also be prepared to handle multiple responses with
different To tags (these belonging to different legs).

-- 
Anders Kristensen <ak@hplb.hpl.hp.com>,
http://www-uk.hpl.hp.com/people/ak/
Hewlett-Packard Labs, Bristol, UK



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To: "'Anders Kristensen'" <ak@hplb.hpl.hp.com>,
        Itamar Gilad <ItamarG@tlv.radvision.com>
Cc: "'sip@lists.research.bell-labs.com'" <sip@lists.research.bell-labs.com>
Subject: RE: Call legs and Cseq
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> -----Original Message-----
> From: Anders Kristensen [mailto:ak@hplb.hpl.hp.com]
> Sent: Wednesday, February 02, 2000 5:59 PM
> To: Itamar Gilad
> Cc: 'Anders Kristensen'; 'sip@lists.research.bell-labs.com'
> Subject: Re: Call legs and Cseq
> 
> 
> 
> Itamar Gilad wrote:
> > 
> > Just to make sure:
> > 
> > reqMsg1 == reqMsg2 (duplicate requests) IFF the To, From, 
> Call-ID and CSeq
> > headers are identical (addresses and tags only in the To 
> and From headers).
> 
> Yes. Obviously, reqMsg1 and reqMsg2 may in fact differ in 
> other headers
> (although this should never happen), but for the purpose of 
> identifying
> and filtering out retransmissions they're the same.
> 
> > 
> > msg1 and msg2 belong to the same Call-leg if
> >     the Call-ID is the same and
> >     (msg1.To == msg2.To && msg1.From == msg2.From) ||
> >     (msg1.To == msg2.From && msg1.From == msg2.To)
> > 
> > ( ==  means the addresses and tags are the same)
> > 
> > Is that right?
> 
> Yes. 
> 
> > 
> > BTW, you cannot identify duplicate responses using the first rule.
> 
> No, not quite. If the request was sent without a To tag the 
> UAC needs to
> take into account the possibility that a UAS has added this to the
> response, and must also be prepared to handle multiple responses with
> different To tags (these belonging to different legs).
> 

Right, but what I meant is that even if two responses have the same To,
From, Call-ID and CSeq values (including tags where applicable) it still
doesn't mean these are identical messages.   The Status line may be
different (and usually is).

Cheers
  
  Itamar



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Itamar Gilad wrote:
> 
> > >
> > > BTW, you cannot identify duplicate responses using the first rule.
> >
> > No, not quite. If the request was sent without a To tag the
> > UAC needs to
> > take into account the possibility that a UAS has added this to the
> > response, and must also be prepared to handle multiple responses with
> > different To tags (these belonging to different legs).
> >
> 
> Right, but what I meant is that even if two responses have the same To,
> From, Call-ID and CSeq values (including tags where applicable) it still
> doesn't mean these are identical messages.   The Status line may be
> different (and usually is).

Right, of course. You don't want to filter out those final responses ;-)

-- 
Anders Kristensen <ak@hplb.hpl.hp.com>,
http://www-uk.hpl.hp.com/people/ak/
Hewlett-Packard Labs, Bristol, UK



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Subject: Re: From Tag syntax
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Henning Schulzrinne wrote:

[snip]

>
> Questions:
>
> 1) If you are an implementor: does your implementation in any way depend
> on tag being hexadecimal?
>

Not at all in our case.  So, a token is perfect for us too.

Manu.

>
> 2) Any other opinions?
>
> Thanks.
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

--
Emmanuel Bertrand
mailto:eb@indigo-software.com
http://www.indigo-software.com





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Our implementation does not have  the hex limitation, for the Tag field.

regards
sunitha


Henning Schulzrinne wrote:

> I'm trying to reach closure on a minor syntax issue.
>
> Currently, the "tag=" parameter is a hexadecimal string (plus dash).
>
> It has been proposed
>
> Advantage of the change:
> - one fewer syntax type to worry about;
> - currently, the similar-in-spirit Via branch parameter is a token, so
> this would unify the two;
> - makes it easy to encode information strings as base64;
> - higher information density (base64 vs. base16)
> - it's easy to forget this little hex limitation, so implementations may
> need to be "forgiving" in any event if one major one gets it wrong;
>
> Disadvantage:
> - it's a change from RFC 2543 and not backwards-compatible.
>
> Questions:
>
> 1) If you are an implementor: does your implementation in any way depend
> on tag being hexadecimal?
>
> 2) Any other opinions?
>
> Thanks.
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

--
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374



--------------03AF9C00D67D15270E01F98C
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Our implementation does not have&nbsp; the hex limitation, for the Tag
field.
<p>regards
<br>sunitha
<br>&nbsp;
<p>Henning Schulzrinne wrote:
<blockquote TYPE=CITE>I'm trying to reach closure on a minor syntax issue.
<p>Currently, the "tag=" parameter is a hexadecimal string (plus dash).
<p>It has been proposed
<p>Advantage of the change:
<br>- one fewer syntax type to worry about;
<br>- currently, the similar-in-spirit Via branch parameter is a token,
so
<br>this would unify the two;
<br>- makes it easy to encode information strings as base64;
<br>- higher information density (base64 vs. base16)
<br>- it's easy to forget this little hex limitation, so implementations
may
<br>need to be "forgiving" in any event if one major one gets it wrong;
<p>Disadvantage:
<br>- it's a change from RFC 2543 and not backwards-compatible.
<p>Questions:
<p>1) If you are an implementor: does your implementation in any way depend
<br>on tag being hexadecimal?
<p>2) Any other opinions?
<p>Thanks.
<br>--
<br>Henning Schulzrinne&nbsp;&nbsp; <a href="http://www.cs.columbia.edu/~hgs">http://www.cs.columbia.edu/~hgs</a></blockquote>

<pre>--&nbsp;
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374</pre>
&nbsp;</html>

--------------03AF9C00D67D15270E01F98C--




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From: Francois Menard List Account <fm-listproc@fmmo.ca>
To: Rabbe Fogelholm <Rabbe.Fogelholm@ericsson.com>
Cc: sip@lists.research.bell-labs.com, sip@cslab.ericsson.se,
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Does IPonAX means that Ericsson is playing with putting TCP/IP in an AXE ?

-=Francois=-


On Wed, 2 Feb 2000, Rabbe Fogelholm wrote:

> This message is to announce a "new" public SIP server (actually it is a
> reincarnation of the Ellemtel SIP server that used to be running at
> sip.pcs.ellemtel.net:5060 in 1999). Apologies if you receive this
> message in duplicate.
> 
> Henning, here are data for your SIP server page
> (http://www.cs.columbia.edu/~hgs/sip/servers.html):
> 
> Organization:
>   Ericsson (http://www.ericsson.se)
> 
> SIP URL:
>   sip:Robot@iponax.com
> 
> SRV records:
>   iponax.com
> 
> Transport:
>   UDP, TCP
> 
> Modes:
>   redirect
> 
> Accept registrations from other domains?
>   yes
> 
> Remarks:
>   The sip:Robot@iponax.com accepts INVITEs to VoIP sessions.
>   It will play a voice announcement of about 30 s length.
> 
> Comments etc are welcome. Enjoy,
> 
> Rabbe.Fogelholm@ericsson.com
> 




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Subject: Re: Regarding the reliability via UDP
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Hi, Jonathan,
Thank you, for your reply,

I almost understood your explanation.
I will implement which the server remember the result(what kind of response)
, and retransmit it when the server needs to send a response.

Further more, I would like to know how the server treat a request for new
call
when the server cache space is empty.

My idea is:
(1)The server send a 500 response(Server Internal Error) with Retry-After.
     In this case the server is not cache.
(2)The server silently drop a request for new call.
     In this case, the client will retransmit a request.

What do you think about these?
If you have any idea, please tell me them.

Thanks,
===========================
Masanori Inui
IWATSU Electric CO.,LTD
Project Team No.2
Solution Engineering Dept.
Engineering Division
---------------------------
1-7-41 Kugayama Suginami-ku
Tokyo, 168-8501, Japan
---------------------------
Tel.+81-3-5370-5211
Fax.+81-3-5370-5193
Email inui@iwatsu.co.jp
WEB http://www.iwatsu.co.jp
===========================





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Masanori Inui wrote:
> 
> Further more, I would like to know how the server treat a request for new
> call
> when the server cache space is empty.
> 
> My idea is:
> (1)The server send a 500 response(Server Internal Error) with Retry-After.
>      In this case the server is not cache.
> (2)The server silently drop a request for new call.
>      In this case, the client will retransmit a request.

The decision is yours. Its an implementation choice.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Anders Kristensen wrote:
> 
> Itamar Gilad wrote:
> >
> > Just to make sure:
> >
> > reqMsg1 == reqMsg2 (duplicate requests) IFF the To, From, Call-ID and CSeq
> > headers are identical (addresses and tags only in the To and From headers).
> 
> Yes. Obviously, reqMsg1 and reqMsg2 may in fact differ in other headers
> (although this should never happen), but for the purpose of identifying
> and filtering out retransmissions they're the same.

This is the case as specified in RFC2543, but two issues have been
discovered since, and discussed on the list, which probably require two
more fields thrown in there.

First is loop detection. As we discussed on the list, there are valid
cases where a message is not a loop even though it hits the same server
twice; namely, when the request URI is different between the two. In
this case, the request and its looped version will differ by the request
URI (and parts of the Via too). Here, the fact that the request URI is
different is important. 

Another issue is request merging (the reverse of forking). In this case,
two messages arrive from different upstream servers, with the same
request URI, To, From, Call-ID and CSeq. Whilst it is OK to effectively
ignore this problem, and treat the two as retransmissions, better
performance is achieved by considering them as separate requests. See:

http://www.cs.columbia.edu/~jdrosen/sip/multiple_requests.txt

for a discussion of the options and their various consequences. Thus,
one could include the topmost Via in the determination of equality.

> 
> >
> > msg1 and msg2 belong to the same Call-leg if
> >     the Call-ID is the same and
> >     (msg1.To == msg2.To && msg1.From == msg2.From) ||
> >     (msg1.To == msg2.From && msg1.From == msg2.To)
> >
> > ( ==  means the addresses and tags are the same)
> >
> > Is that right?

Almost. One would not compare two messages to determine if they are part
of the same call-leg. Rather, when an INVITE is sent and/or received for
the first time, the call leg is established by extracting the local and
remote addresses. Whether they are extracted from the To or From depends
on whether the INVITE was sent or received. Then, subsequent messages
are determined to be part of the call leg by matching the To  (From) to
the remote (local) participant if the message is a request (response),
and the From (To) to the local (remote) participate if the message is a
response (request). 


> >
> > BTW, you cannot identify duplicate responses using the first rule.
> 
> No, not quite. If the request was sent without a To tag the UAC needs to
> take into account the possibility that a UAS has added this to the
> response, and must also be prepared to handle multiple responses with
> different To tags (these belonging to different legs).

For a proxy, the branch ID parameter will also need to be looked at if
the proxy forked, to determine which forked request the response
corresponds to.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb  3 01:39:40 2000
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Message-ID: <38959310.A7C9728B@ccrle.nec.de>
Date: Mon, 31 Jan 2000 14:50:08 +0100
From: Michael =?iso-8859-1?Q?Ter=F6rde?= <michael.teroerde@ccrle.nec.de>
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Hello !!

I have searched your Email-list for Questions like mine, but there
wasn't
a posting which answerd all my questions.

I woul know something about the REGISTER- Request in the SIP-Language.

1. Could a Server send me an De-Register message if he want (e.g. if he
wants to reboot)??
2. Could a Server change the "Expires"- Header field (and therefore
 change the Register- Time)??
3. Did the Server send somthing as "Keep Alive" to the registerd
 UserAgent??(or did he make a "ping" to the IP or something else to
determine that my UA is still active)
4. What happens if my UserAgent hangs up??
5. Is there a generelly timout when the Server delets the
register-entry   (e.g. after 48 hours) ??
6. Did the server MUST send a "200 OK"- Response if the UserAgent send a
 De-register(REGISTER with Expires=0) ??

Please help me !!!
If you know an interesting Paper about generelly Sever behavior, please
tell it to me.


Thanks for your effort

Michael Teroerde

(instead of posting it on the List , you can also send it to my
NEC-Email-Adress)
----------------------------------------------------------------
Michael Teroerde
Student at NEC- Heidelberg
michael.teroerde@ccrle.nec.de



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb  3 06:27:56 2000
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Message-ID: <389964F7.56D4EB68@8x8.com>
Date: Thu, 03 Feb 2000 11:22:31 +0000
From: Marc Petit-Huguenin <petithug@8x8.com>
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Hi all,

First, thank for all your responses to my previous answers.

I have a question about sending subsequent requests from the callee.

Section 11.4 in the SIP draft give the following rules:
- If the previous request contains a Record-Route, use the infos in this
header.
- Otherwise use the previous Contact received (if existing)
- Otherwise use the provisioned proxy server (if provisioned)
- Otherwise use the value in the From header field.

As the rules 1, 2 and 3 can fail, we must be sure that the rule 4 works. 
The problem is that the "transport=" parameter is forbidden in a From
header field, so there is no way to known which transport protocol to
use.
So do we must use :
- Any protocol?
- The same protocol that was use for the last received request?

Thank you for your answers.

Regards,

-- 
Marc Petit-Huguenin
Home: mailto:petithug@cyberservices.com
Work: mailto:petithug@8x8.com
"Everything I love is either immoral, illegal or makes you fat"



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb  3 10:43:57 2000
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Date: Thu, 03 Feb 2000 16:39:38 +0100
From: Rabbe Fogelholm <rabbe.fogelholm@uab.ericsson.se>
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        sipfactory@uab.ericsson.se, schulzrinne@cs.columbia.edu
Subject: Re: New public SIP server
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The story behind the domain name is that it was to be read "IP
Only Access". The domain was acquired for a different project,
and we have started using it mostly since it was there and
unused.

--Rabbe Fogelholm


Francois Menard List Account wrote:
> 
> Does IPonAX means that Ericsson is playing with putting TCP/IP in an AXE ?
> 
> -=Francois=-
> 
> On Wed, 2 Feb 2000, Rabbe Fogelholm wrote:
> 
> > This message is to announce a "new" public SIP server (actually it is a
> > reincarnation of the Ellemtel SIP server that used to be running at
> > sip.pcs.ellemtel.net:5060 in 1999). Apologies if you receive this
> > message in duplicate.
> >
> > Henning, here are data for your SIP server page
> > (http://www.cs.columbia.edu/~hgs/sip/servers.html):
> >
> > Organization:
> >   Ericsson (http://www.ericsson.se)
> >
> > SIP URL:
> >   sip:Robot@iponax.com
> >
> > SRV records:
> >   iponax.com
> >
> > Transport:
> >   UDP, TCP
> >
> > Modes:
> >   redirect
> >
> > Accept registrations from other domains?
> >   yes
> >
> > Remarks:
> >   The sip:Robot@iponax.com accepts INVITEs to VoIP sessions.
> >   It will play a voice announcement of about 30 s length.
> >
> > Comments etc are welcome. Enjoy,
> >
> > Rabbe.Fogelholm@ericsson.com
> >



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb  3 10:48:20 2000
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Date: Thu, 03 Feb 2000 10:41:41 -0500
From: Igor Slepchin <islepchin@dynamicsoft.com>
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Marc Petit-Huguenin wrote:
> As the rules 1, 2 and 3 can fail, we must be sure that the rule 4 works.
> The problem is that the "transport=" parameter is forbidden in a From
> header field, so there is no way to known which transport protocol to
> use.
> So do we must use :
> - Any protocol?
> - The same protocol that was use for the last received request?

You should use the same algorithm you used for the first request, see
1.4.2 and appendix D for details. If the other side wants to be
contacted differently (different transport, requestURI, port etc.), it
must specify so in the Contact field.

---
Igor Slepchin



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb  3 15:13:54 2000
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Date: Thu, 03 Feb 2000 15:10:45 -0500
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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Dear SIP bakers:

The 4th SIP bake-off will be held in Rolling Meadows, Illinois, hosted
by 3Com, from April 17 to 19, 2000. Further logistical details will be
available shortly at http://www.sipbakeoff.org/

Registration is set to begin around March 1st.

Please contact Jennifer Wade at Jennifer_Wade@3com.com for further
details.

Regards,

Henning

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



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb  4 11:20:53 2000
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Date: Fri, 04 Feb 2000 11:17:45 -0500
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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Gautam Nair and I have submitted an Internet draft for using DHCP to
configure SIP clients with the address of outbound SIP servers.

Until the draft appears in the I-D directories, it can also be found at
http://www.cs.columbia.edu/sip/drafts/draft-nair-sip-dhcp-00.txt

(For the DHCP experts: SIP, the Session Initiation Protocol, is used for
setting up, modifying and terminating multimedia sessions, including
Internet telephone calls. It is an IETF Proposed Standard document in
RFC 2543, with information available at http://www.cs.columbia.edu/sip)

We have documented a number of approaches in the draft, some of which
may well be dropped if deemed to be redundant. A particular concern of
ours are low-complexity embedded systems (such as
http://www.cs.columbia.edu/~hgs/ephone) that may not support DNS and do
not have the computational resources to host an SLP implementation.
(Indeed, the outbound SIP server may well handle DNS resolution for the
client.)

Since SIP domains can be rather large (a whole ISP), we considered it
important to provide a load balancing mechanism. Since this operates in
a telephony context, reliability is very important, leading us to
include the notion of backup servers and the ability to operate even
when local DNS service is interrupted. There may well be better
mechanisms to address either of these concerns.

Feedback and comments are appreciated.

Henning



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Folks,

The SIP wg meeting in IETF 47 in Adelaide has been tentatively scheduled
for Monday, March 27, 9:30 - 11:30 and Tuesday, March 28, 1-3:15.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com
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Working Group and BOF Chairs,

Here is the current Agenda including all of the requests I have
received as of today, February 3, 2000.

All scheduling is tentative and subject to change.  If a change needs 
to be made to your working group, I will notify you.  All attempts will 
be made to keep the working group as close as possible to its current 
scheduled time.

Please submit an agenda for your working group as soon as possible to:
agenda@ietf.org
The deadline for submitting an agenda is March 17 at 1200 ET.

               Draft Agenda of the Forty-seventh IETF
                         March 26-31, 2000
                           As of 2/3/00

SUNDAY, March 26, 2000
1200-1900  Registration - 
1530-1600  Newcomer's Orientation - 
1600-1630  IETF Standards Process Orientation - 
1700-1900  Welcome Reception - 

MONDAY, March 27, 2000
0800-1930  IETF Registration - 
0800-0900  Continental Breakfast 
0900-0930  Opening Plenary Session - 
0930-1130  Morning Sessions
	APP  apparea	Applications Area Meeting
	INT  frnetmib	Frame Relay Service MIB WG
	SEC  cat	Common Authentication Technology WG
	TSV  sip	Session Initiation Protocol WG *

1130-1300  Break
1300-1500  Afternoon Sessions I
	APP  ldapext	LDAP Extension WG
	RTG  manet	Mobile Ad-hoc Networks WG
	SEC  xmldsig	XML Digital Signatures WG
	TSV  nat	Network Address Translators WG
	USV  uswg	User Services WG

1500-1530  Break (Refreshments provided) - 
1530-1730  Afternoon Sessions II
	APP  conneg	Content Negotiation WG
	INT  ipcdn	IP over Cable Data Networks WG
	OPS  mboned	MBONE Deployment WG *
	RTG  mobileip	IP Routing for Wireless/Mobile Hosts WG
	TSV  diffserv	Differentiated Services WG *
	TSV  spirits	Service in the PSTN/IN Requesting Internet Service WG

1730-1930  Break
1930-2200  Evening Sessions
	OPS  ngtrans	Next Generation Transition WG
	RTG  mpls	Multiprotocol Label Switching WG
	TSV  iptel	IP Telephony WG *
	TSV  nfsv4	Network File System Version 4 WG
			
* Designates Multicast Sessions


TUESDAY, March 28, 2000
0800-1700  IETF Registration - 
0800-0900  Continental Breakfast - 
0900-1130  Morning Sessions
	APP  calsch	Calendaring and Scheduling WG
	INT  dnsext	DNS Extensions WG *
	OPS  rap	Resource Allocation Protocol WG
	RTG  mobileip	IP Routing for Wireless/Mobile Hosts WG
	TSV  enum	Telephone Number Mapping WG
	TSV  megaco	Media Gateway Control WG

1130-1300  Break
1300-1400  Afternoon Sessions I
	TSV  diffserv	Differentiated Services WG *
	TSV  sip	Session Initiation Protocol WG *

1415-1515  Afternoon Sessions II
	RTG  vrrp	Virtual Router Redundancy Protocol WG
	TSV  nfsv4	Network File System Version 4
	TSV  sip	Session Initiation Protocol WG *

1515-1545  Break (Refreshments provided) - 
1545-1645  Afternoon Sessions III
	OPS  snmpconf	Configuration Management with SNMP WG
	TSV  rmt	Reliable Multicast Transport WG *

1700-1800  Afternoon Sessions IV
	OPS  snmpconf	Configuration Management with SNMP WG
	TSV  pilc	Performance Implications of Link Characteristics WG
	TSV  rmt	Reliable Multicast Transport WG *

* Designates Multicast Sessions


WEDNESDAY, March 29, 2000
0800-1700  IETF Registration - 
0800-0900  Continental Breakfast - 
0900-1130  Morning Sessions
	APP  trade	Internet Open Trading Protocol WG
	INT  ipngwg	IPNG WG *
	OPS  bmwg	Benchmarking Methodology WG
	RTG  mpls	Multiprotocol Label Switching WG
	TSV  rmt	Reliable Multicast Transport WG *
	TSV  sigtran	Signaling Transport WG

1130-1300  Break
1300-1500  Afternoon Sessions I
	INT  ipfc	IP Over Fibre Channel WG
	OPS  ngtrans	Next Generation Transition WG
	RTG  msdp	Multicast Source Discovery Protocol WG *
	SEC  pkix	Public-Key Infrastructure (X.509) WG
	TSV  tsvwg	Transport Area Working Group WG

1500-1530  Break (Refreshments provided) - 
1530-1730  Afternoon Sessions II
	INT  l2tpext	Layer Two Tunneling Protocol Extensions WG
	RTG  rtgarea	Routing Area Meeting
	SEC  smime	S/MIME Mail Security WG
	TSV  avt	Audio/Video Transport WG *

1730-1930  Break
1930-2200  Open Plenary -  
        o Welcome - Fred Baker, IETF Chair
        o IAB Open Plenary
        o IESG Open Plenary

* Designates Multicast Sessions


THURSDAY, March 30, 2000
0800-1700  IETF Registration - 
0800-0900  Continental Breakfast - 
0900-1130  Morning Sessions
	APP  blocks	An Application Protocol Framework & A Model Application BOF
	OPS  snmpconf	Configuration Management with SNMP WG
	TSV  malloc	Multicast-Address Allocation WG *

1130-1300  Break
1300-1500  Afternoon Sessions I
	OPS  snmpv3	SNMP Version 3
	SEC  saag	Open Security Area Directorate Meeting
	TSV  avt	Audio/Video Transport WG *

1500-1530  Break (Refreshments provided) - 
1530-1730  Afternoon Sessions II
	SEC  idwg	Intrusion Detection Exchange Format WG
	TSV  mmusic	Multiparty Multimedia Session Control WG

FRIDAY, March 31, 2000
0800-1000  IETF Registration - 
0800-0900  Continental Breakfast - 
0900-1130  Morning Sessions
	SEC  idwg	Intrusion Detection Exchange Format WG
	TSV  ecm	Endpoint Congestion Management WG

* Designates Multicast Sessions


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From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb  7 03:12:05 2000
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From: "Christian Jansson" <Christian@hotsip.com>
To: <sip@lists.research.bell-labs.com>
Subject: Question about REGISTER
Date: Mon, 7 Feb 2000 09:07:27 +0100
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From 4.2.6:

All current registrations MUST share the same action value. Registrations
that have a different action than current registrations for the same user
MUST be rejected with status of 409 (Conflict).


1.What about a REGISTER request that does not have an action value:
   Does that match any of "redirect" or "proxy"?
   Is it then up to the server to chose the action, and indicate the choice
   in the response?
   Or is it the first REGISTER with an action that sets the rule (also for 
   earlier REGISTERS that did not have an action value)?

2. Why do they have to share the same action at all?

3. A server that only supports one of the redirect/proxy behaviors receives
    a REGISTER with the unsupported action indicated. What is it supposed
    to do, send a 409 response?

--
Christian Jansson (christian@hotsip.com)
HotSip AB
phone:  +46 (0)8 555 10 704
mobile: +46 (0)70 558 1038




From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb  7 05:53:55 2000
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From: Michael Teroerde <michael.teroerde@gmx.de>
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Hello !!

If I want to make an "third-party registration" could I write another
domain
in the From-header field as in the To- header ???

Example:
...
To: alice@acme.com
From: michael@nec.de
...

(In the RFC- Example are the domains equal)

If it is possible, is the OK- response send to alice@nec.de or
michael@acme.com or to both people?

If it is impossible, is there another way to register a user from another
Domain as where the UA is running (and get all Responses regarding the
registration, but no Message about the Call)?

Thank you !!

Michael Teroerde

(instead of posting it on the List , you can also send it to my
NEC-Email-Adress)
----------------------------------------------------------------
Michael Teroerde
Student at NEC- Heidelberg
michael.teroerde@ccrle.nec.de

-- 
Sent through Global Message Exchange - http://www.gmx.net




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Michael Teroerde wrote:
> 
> Hello !!
> 
> If I want to make an "third-party registration" could I write another
> domain
> in the From-header field as in the To- header ???

Absolutely. 

> 
> Example:
> ...
> To: alice@acme.com
> From: michael@nec.de
> ...
> 
> (In the RFC- Example are the domains equal)
> 
> If it is possible, is the OK- response send to alice@nec.de or
> michael@acme.com or to both people?

Neither. Like all SIP requests, the response is sent to the address in
the Via header.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb  7 09:58:00 2000
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Christian Jansson wrote:
> 
> >From 4.2.6:
> 
> All current registrations MUST share the same action value. Registrations
> that have a different action than current registrations for the same user
> MUST be rejected with status of 409 (Conflict).
> 
> 1.What about a REGISTER request that does not have an action value:
>    Does that match any of "redirect" or "proxy"?
>    Is it then up to the server to chose the action, and indicate the choice
>    in the response?
>    Or is it the first REGISTER with an action that sets the rule (also for
>    earlier REGISTERS that did not have an action value)?

If no action is present in the initial registration, the server does
whatever it likes and returns that in the response. From the spec:

>  action: The "action" parameter is used only when registering with the
>         REGISTER request. It indicates whether the client wishes that
>         the server proxy or redirect future requests intended for the
>         client. If this parameter is not specified the action taken
>         depends on server configuration. In its response, the registrar
>         SHOULD indicate the mode used. This parameter is ignored for
>         other requests.


Subsequent registrations with no action will take the same value.

> 
> 2. Why do they have to share the same action at all?

It doesn't make sense for them not to. If a server has two URLs, one
with proxy, and one with redirect, and then an INVITE comes, what does
it do? Does it ignore the proxy and then redirect? Would it redirect to
one or the other or both URLS? Would it first proxy, and then on
failure, redirect? These are the kinds of things CPL can specify. Rather
than build arbitrary and complex meaning into the action parameter, we
kept it simple.

> 
> 3. A server that only supports one of the redirect/proxy behaviors receives
>     a REGISTER with the unsupported action indicated. What is it supposed
>     to do, send a 409 response?

403 Forbidden is probably better. It doesn't much matter, unless you
want to have the client resubmit the request with a different action
tag. There is currently no way to do this; I'm not sure its a big deal
(tend to think its not generally needed)

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb  7 10:22:58 2000
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From: "Christian Jansson" <Christian@hotsip.com>
To: <sip@lists.research.bell-labs.com>
Subject: Question about REGISTER
Date: Fri, 4 Feb 2000 16:51:43 +0100
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>From 4.2.6:

All current registrations MUST share the same action value. Registrations
that have a different action than current registrations for the same user
MUST be rejected with status of 409 (Conflict).


1.What about a REGISTER request that does not have an action value:
    Does that match any of "redirect" or "proxy"?
    Is it then up to the server to chose the action, and indicate the choice
    in the response? Or is it the first REGISTER with an action that sets
    the rule (also for earlier REGISTERS that did not have an action value)?

2. Why do they have to share the same action at all?

3. A server that only supports one of the redirect/proxy behaviors receives
    a REGISTER with the unsupported action indicated. What does it do,
    send a 409 response?

--
Christian Jansson (christian@hotsip.com)
HotSip AB
phone:  +46 (0)8 555 10 704
mobile: +46 (0)70 558 1038




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From: Michael Teroerde <michael.teroerde@gmx.de>
To: sip@lists.research.bell-labs.com
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Subject: Via-Header in a REGISTER-request
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Hello !!!

If I send an REGISTER and add an additional Via-header. Does the Server
remember the Via headers for sending an INVITE trough the same route ??

For example:

REGISTER sip:tu-berlin.de SIP/2.0
Via: SIP/2.0/UDP bell.de
Via: SIP/2.0/UDP nec.de
From: sip:michael@nec.de
.....

Did the proxy forward an incomming INVITE for michael@nec.de to bell.de or

did he tries to connect directly to nec.de ??

Thanks for your effort

Michael Teroerde
----------------------------------------------------------------
Michael Teroerde
Student at NEC- Heidelberg
michael.teroerde@ccrle.nec.de

-- 
Sent through Global Message Exchange - http://www.gmx.net




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Date: Tue, 08 Feb 2000 15:24:16 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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To: michael.teroerde@gmx.de
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Michael,

michael.teroerde@gmx.de wrote:
> 
> Hello !!!
> 
> If I send an REGISTER and add an additional Via-header. Does the Server
> remember the Via headers for sending an INVITE trough the same route ??
> 

Via headers are used just for routing responses. The INVITE is a
request.

> For example:
> 
> REGISTER sip:tu-berlin.de SIP/2.0
> Via: SIP/2.0/UDP bell.de
> Via: SIP/2.0/UDP nec.de
> From: sip:michael@nec.de
> ......
> 
> Did the proxy forward an incomming INVITE for michael@nec.de to bell.de or
> 
> did he tries to connect directly to nec.de ??

The proxy will use the Contact header in the REGISTER.

Regards,

Gonzalo

> 
> Thanks for your effort
> 
> Michael Teroerde
> ----------------------------------------------------------------
> Michael Teroerde
> Student at NEC- Heidelberg
> michael.teroerde@ccrle.nec.de
> 
> --
> Sent through Global Message Exchange - http://www.gmx.net

-- 
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



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To: sip@lists.research.bell-labs.com
Subject: SIP Stacks
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Can somebody tell me who are the SIP Vendors??



Thanks 

Denis



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To: sip@lists.research.bell-labs.com
Subject: SIP Stacks
Date: Thu, 3 Feb 2000 12:45:48 -0600 
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I need to know, who has SIP stacks (available) that include User Agents and
Proxies


Denis Dagenais



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb  8 11:15:34 2000
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From: "Christian Jansson" <Christian@hotsip.com>
To: <sip@lists.research.bell-labs.com>
Subject: Question about REGISTER
Date: Thu, 3 Feb 2000 11:57:24 +0100
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>From 4.2.6:

All current registrations MUST share the same action value. Registrations
that have a different action than current registrations for the same user
MUST be rejected with status of 409 (Conflict).


1.What about a REGISTER request that does not have an action value:
    Does that match any of "redirect" or "proxy"?
    Is it then up to the server to chose the action, and indicate the choice
    in the response? Or is it the first REGISTER with an action that sets
    the rule (also for earlier REGISTERS that did not have an action value)?

2. Why do they have to share the same action at all?

3. A server that only supports one of the redirect/proxy behaviors receives
    a REGISTER with the unsupported action indicated. What does it do,
    send a 409 response?

--
Christian Jansson (christian@hotsip.com)
HotSip AB
phone:  +46 (0)8 555 10 704
mobile: +46 (0)70 558 1038




From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb  8 11:18:46 2000
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Date: Tue, 08 Feb 2000 17:12:24 +0100
From: Jiri Kuthan <kuthan@fokus.gmd.de>
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To: Denis.Dagenais@nokia.com
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Subject: Re: SIP Stacks
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Check Henning's SIP site [1] and my iptel products page [2].
The former includes SIP products only, the latter both
SIP and H.323.

Jiri

[1] http://www.cs.columbia.edu/~hgs/sip/implementations.html
[2] http://www.fokus.gmd.de/research/cc/glone/projects/ipt/products.php3

Denis.Dagenais@nokia.com wrote:
> 
> Can somebody tell me who are the SIP Vendors??
> 
> Thanks
> 
> Denis

-- 
Jiri Kuthan		  Phone: ++49 / (0)30 / 3463 - 7 271
GMD FOKUS                 Fax  : ++49 / (0)30 / 3463 - 8 271
Kaiserin-Augusta-Allee 31 E-Mail : kuthan@fokus.gmd.de
D-10589 Berlin, Germany   WWW    : http://www.fokus.gmd.de/usr/kuthan



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb  8 12:38:04 2000
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From: Igor Slepchin <islepchin@dynamicsoft.com>
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Christian Jansson wrote:
> 
> 1.What about a REGISTER request that does not have an action value:
>     Does that match any of "redirect" or "proxy"?
>     Is it then up to the server to chose the action, and indicate the choice
>     in the response? Or is it the first REGISTER with an action that sets
>     the rule (also for earlier REGISTERS that did not have an action value)?

From 6.13:

If this parameter (i.e., action) is not specified the action taken
depends on server configuration.

I guess it makes sense to assume the action parameter that does not
conflict with the existing registrations for the same address of record.
 
> 2. Why do they have to share the same action at all?

Consider the situation when john.doe@dynamicsoft.com is registered with
john.doe@ietf.org;action=proxy and john.doe@acm.org;action=redirect. The
proxy receives an INVITE for john.doe@dynamicsoft.com; what should it do
now? It cannot both redirect and proxy at the same time; sending a
redirect immediately would mean that the second address is not even
tried; proxying first means that the redirect is at least delayed.
Generally, there is a conflict between desired actions for
john.doe@dynamicsoft.com between registering entities.
 
> 3. A server that only supports one of the redirect/proxy behaviors receives
>     a REGISTER with the unsupported action indicated. What does it do,
>     send a 409 response?

This would work. Another option is something like 501 Not implemented:
action=proxy .

---
Igor Slepchin



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Date: Tue, 08 Feb 2000 12:21:06 -0600
From: Henry Sinnreich <henry.sinnreich@wcom.com>
Subject: RE: SIP Stacks
In-reply-to: <38A04068.A61F74FD@fokus.gmd.de>
To: Jiri Kuthan <kuthan@fokus.gmd.de>, Denis.Dagenais@nokia.com
Cc: sip@lists.research.bell-labs.com
Message-id: <NDBBLDFFOKEECMNDFGLCOEJIFFAA.henry.sinnreich@wcom.com>
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>http://www.fokus.gmd.de/research/cc/glone/projects/ipt/
>products.php3
Is a good site for products. Small nit:
Have also seen public demos and SIP bake-off tests of SIP phones
from 3Com, Nortel (Etherset cluster) and Pingtel.

Henry

>-----Original Message-----
>From: owner-sip@lists.research.bell-labs.com
>[mailto:owner-sip@lists.research.bell-labs.com]On
>Behalf Of Jiri Kuthan
>Sent: Tuesday, February 08, 2000 10:12 AM
>To: Denis.Dagenais@nokia.com
>Cc: sip@lists.research.bell-labs.com
>Subject: Re: SIP Stacks
>
>
>Check Henning's SIP site [1] and my iptel products page [2].
>The former includes SIP products only, the latter both
>SIP and H.323.
>
>Jiri
>
>[1] http://www.cs.columbia.edu/~hgs/sip/implementations.html
>[2]
>http://www.fokus.gmd.de/research/cc/glone/projects/ipt/
>products.php3
>
>Denis.Dagenais@nokia.com wrote:
>>
>> Can somebody tell me who are the SIP Vendors??
>>
>> Thanks
>>
>> Denis
>
>--
>Jiri Kuthan		  Phone: ++49 / (0)30 / 3463 - 7 271
>GMD FOKUS                 Fax  : ++49 / (0)30 / 3463 - 8 271
>Kaiserin-Augusta-Allee 31 E-Mail : kuthan@fokus.gmd.de
>D-10589 Berlin, Germany   WWW    :
>http://www.fokus.gmd.de/usr/kuthan
>




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From: "Andrew Chalupka" <andcha@nortelnetworks.com>
To: "'SIP List'" <sip@lists.research.bell-labs.com>
Subject: Question about media desinations
Date: Tue, 8 Feb 2000 13:55:25 -0500
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Hi all,

Apologies if this question isn't appropriate for the SIP list...

Assume a user, "caller", initiates a SIP call with a second user, "callee",
through the usual means (INVITE, 200 OK, ACK).  During this message
exchange, each end speficies the IP address and UDP port number(s) to which
the other end should send his audio stream.  Would it be a violation of the
SIP standard for one of the participants to send his audio stream to a third
party (say to a server which might transcode the sender's audio packets into
a format the other end can understand) and then have that third party
forward the packets on to the appropriate destination?

Section B.2 of RFC 2543 (pg 138) mentions...

"For receive-only and send-or-receive streams, the port number and address
in the session description indicate where the media stream should be sent to
by the recipient of the session description, either caller or callee."

...which isn't as strong as saying the port-address pair supplied by the
other end *must* be used.  

Any clarifications?

- Andrew
 



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<P><FONT SIZE=3D2 FACE=3D"Arial">Hi all,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Apologies if this question isn't =
appropriate for the SIP list...</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Assume a user, &quot;caller&quot;, =
initiates a SIP call with a second user, &quot;callee&quot;, through =
the usual means (INVITE, 200 OK, ACK).&nbsp; During this message =
exchange, each end speficies the IP address and UDP port number(s) to =
which the other end should send his audio stream.&nbsp; Would it be a =
violation of the SIP standard for one of the participants to send his =
audio stream to a third party (say to a server which might transcode =
the sender's audio packets into a format the other end can understand) =
and then have that third party forward the packets on to the =
appropriate destination?</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Section B.2 of RFC 2543 (pg 138) =
mentions...</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">&quot;For receive-only and =
send-or-receive streams, the port number and address in the session =
description indicate where the media stream should be sent to by the =
recipient of the session description, either caller or =
callee.&quot;</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">...which isn't as strong as saying the =
port-address pair supplied by the other end *must* be used.&nbsp; =
</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Any clarifications?</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">- Andrew</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&nbsp;</FONT>
</P>
<BR>

</BODY>
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From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb  8 14:25:58 2000
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Date: Tue, 08 Feb 2000 20:21:32 +0100
From: Jiri Kuthan <kuthan@fokus.gmd.de>
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Subject: Re: SIP Stacks
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My website is open and the vendors are invited to
send me links to their SIP phones webpages. Unfortunately,
I was not able to find the webpages on the particular 
products.

Regards, Jiri


Henry Sinnreich wrote:
> 
> >http://www.fokus.gmd.de/research/cc/glone/projects/ipt/
> >products.php3
> Is a good site for products. Small nit:
> Have also seen public demos and SIP bake-off tests of SIP phones
> from 3Com, Nortel (Etherset cluster) and Pingtel.
> 
> Henry
> 

-- 
Jiri Kuthan		  Phone: ++49 / (0)30 / 3463 - 7 271
GMD FOKUS                 Fax  : ++49 / (0)30 / 3463 - 8 271
Kaiserin-Augusta-Allee 31 E-Mail : kuthan@fokus.gmd.de
D-10589 Berlin, Germany   WWW    : http://www.fokus.gmd.de/usr/kuthan



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb  8 14:34:04 2000
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From: "Paul Defrain" <defrain@nortelnetworks.com>
To: "Andrew Chalupka" <andcha@nortelnetworks.com>,
        "'SIP List'" <sip@lists.research.bell-labs.com>
Subject: RE: Question about media desinations
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Andrew, 

Our application operates in the manner you suggest. The bearer is not
terminated to both participants in the SIP session (the endpoints of the
respective SDP and SIP sessions are not necessarilly the same). This is not
a violation of the standard as I've interpreted it. In some cases it is
absolutely necessary for the SDP participants to be different than the SIP
participants which furthers my conviction on this point. I'm interested to
hear of any other opinions concerning this matter.

Paul DeFrain
Nortel Networks, IP Services


> -----Original Message-----
> From:	Chalupka, Andrew [CAR:6M42:EXCH] 
> Sent:	Tuesday, February 08, 2000 12:55 PM
> To:	'SIP List'
> Subject:	Question about media desinations
> 
> Hi all, 
> 
> Apologies if this question isn't appropriate for the SIP list... 
> 
> Assume a user, "caller", initiates a SIP call with a second user,
> "callee", through the usual means (INVITE, 200 OK, ACK).  During this
> message exchange, each end speficies the IP address and UDP port number(s)
> to which the other end should send his audio stream.  Would it be a
> violation of the SIP standard for one of the participants to send his
> audio stream to a third party (say to a server which might transcode the
> sender's audio packets into a format the other end can understand) and
> then have that third party forward the packets on to the appropriate
> destination?
> 
> Section B.2 of RFC 2543 (pg 138) mentions... 
> 
> "For receive-only and send-or-receive streams, the port number and address
> in the session description indicate where the media stream should be sent
> to by the recipient of the session description, either caller or callee."
> 
> ...which isn't as strong as saying the port-address pair supplied by the
> other end *must* be used.  
> 
> Any clarifications? 
> 
> - Andrew 
>   
> 
> 

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<TITLE>RE: Question about media desinations</TITLE>
</HEAD>
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<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Andrew, </FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Our application =
operates in the manner you suggest. The bearer is not terminated to =
both participants in the SIP session (the endpoints of the respective =
SDP and SIP sessions are not necessarilly the same). This is not a =
violation of the standard as I've interpreted it. In some cases it is =
absolutely necessary for the SDP participants to be different than the =
SIP participants which furthers my conviction on this point. I'm =
interested to hear of any other opinions concerning this =
matter.</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Paul DeFrain</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Nortel Networks, IP =
Services</FONT>
</P>
<BR>
<UL>
<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">Chalupka, Andrew [CAR:6M42:EXCH] </FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Tuesday, February 08, 2000 12:55 PM</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">'SIP List'</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">Question about media =
desinations</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Hi all,</FONT><FONT FACE=3D"Arial"> =
</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Apologies if this question isn't =
appropriate for the SIP list...</FONT><FONT FACE=3D"Arial"> </FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Assume a user, &quot;caller&quot;, =
initiates a SIP call with a second user, &quot;callee&quot;, through =
the usual means (INVITE, 200 OK, ACK).=A0 During this message exchange, =
each end speficies the IP address and UDP port number(s) to which the =
other end should send his audio stream.=A0 Would it be a violation of =
the SIP standard for one of the participants to send his audio stream =
to a third party (say to a server which might transcode the sender's =
audio packets into a format the other end can understand) and then have =
that third party forward the packets on to the appropriate =
destination?</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Section B.2 of RFC 2543 (pg 138) =
mentions...</FONT><FONT FACE=3D"Arial"> </FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">&quot;For receive-only and =
send-or-receive streams, the port number and address in the session =
description indicate where the media stream should be sent to by the =
recipient of the session description, either caller or =
callee.&quot;</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">...which isn't as strong as saying the =
port-address pair supplied by the other end *must* be used.=A0</FONT>=20
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Any clarifications?</FONT><FONT =
FACE=3D"Arial"> </FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">- Andrew</FONT><FONT =
FACE=3D"Arial"><BR>
</FONT><FONT SIZE=3D2 FACE=3D"Arial">=A0</FONT><FONT FACE=3D"Arial"> =
</FONT>
</P>
<BR>
</UL>
</BODY>
</HTML>
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Date: Tue, 8 Feb 2000 14:06:07 -0600
From: Billy Biggs <vektor@div8.net>
To: Andrew Chalupka <andcha@nortelnetworks.com>
Cc: "'SIP List'" <sip@lists.research.bell-labs.com>
Subject: Re: Question about media desinations
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Andrew Chalupka (andcha@nortelnetworks.com):

> [...]  Would it be a violation of the SIP standard for one of the
> participants to send his audio stream to a third party (say to a server which
> might transcode the sender's audio packets into a format the other end can
> understand) and then have that third party forward the packets on to the
> appropriate destination?
> 
> Section B.2 of RFC 2543 (pg 138) mentions...
> 
> "For receive-only and send-or-receive streams, the port number and address
> in the session description indicate where the media stream should be sent to
> by the recipient of the session description, either caller or callee."
> 
> ...which isn't as strong as saying the port-address pair supplied by the
> other end *must* be used.  

  Maybe I am misunderstanding your questions, but we frequently send SIP to a
proxy which is different then the UA we exchange media with.  So, yes.

  This brings up an interesting issue that I think is relevant to the sip list.
Is there any sane policy that a client can enforce to prevent unsolicited audio
to be played?

  One policy we came up with was that a client should only accept audio from
whomever it is sending audio to.  This policy seems to be broken.  I think the
example Andrew gives above is I think close.  What if we're receiving audio
locally, but our audio is sent to a third party and then sent to the far end.

  Given that there is no required method of dictating where audio will come
from, a client is therefore in my eyes forced to accept and play whatever audio
receives.

  Following on that, once audio is being received, in my mind there isn't
anything stopping someone from blabbing into an existing conversation.  Since
you can't be assured about where audio is to come from, is a sane policy that
once you start receiving audio from a foreign party that it must always come
from that host?

  I don't think that works either.  Why can't you have hold music come from a
different host while still receiving audio?  No re-INVITE required...  In that
case, you can't even be assured of a common SSRC.

  So, what's stopping a haxor like me from strobbing a gateway with random
audio... ??

-- 
Billy Biggs                         vektor@div8.net
http://www.div8.net/billy       wbiggs@uwaterloo.ca



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>   So, what's stopping a haxor like me from strobbing a gateway with random
> audio... ??
>
Crypto.
 




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Date: Tue, 8 Feb 2000 14:47:18 -0600
From: Billy Biggs <vektor@div8.net>
To: David Oran <oran@cisco.com>
Cc: Andrew Chalupka <andcha@nortelnetworks.com>,
        "'SIP List'" <sip@lists.research.bell-labs.com>
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David Oran (oran@cisco.com):

> >   So, what's stopping a haxor like me from strobbing a gateway with random
> > audio... ??
>
> Crypto.

  Ok, so lets think about how that's going to work.  It's well defined how you
encrypt and authenticate your SIP messages, no big deal.  But what about media?

  SDP provides a k= line to convey an encryption key, but it does not describe
how to practically make use of it.

  I admit I'm not up to speed on this, but is there a draft on how to
interoperably initiate an encrypted audio stream using G.711 encoding?  Is the
vagueness of draft-ietf-avt-profile-new-08.txt what I should be going on?

  Regardless, if I find the IP address of a SIP-PSTN gateway, it's bound to
also support unencrypted connections.  Sending audio probes at it seems to be a
completely reasonable tactic as far as I'm concerned.

-- 
Billy Biggs                         vektor@div8.net
http://www.div8.net/billy       wbiggs@uwaterloo.ca



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Billy Biggs wrote:
> 
>   SDP provides a k= line to convey an encryption key, but it does not describe
> how to practically make use of it.

True enough. The original idea is that the SIP INVITE would be
encrypted, and the body contains the key for the media session. The key
is used for RTP level encryption. No particular algorithm is specified
for SIP, a hole we are working to plug.

As an alternative, one can set up an ipsec association for the media.
This just requires the SDP to somehow indicate that ipsec is desired;
the rest is up to ike, ipsec and its related friends. This is described
in:

http://www.normos.org/ietf/draft/draft-ietf-mmusic-sdp-qos-00.txt

although the ipsec/ike stuff is not described, just the SIP/SDP
component.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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From: Dean Willis <dean.willis@wcom.com>
Subject: RE: Question about media desinations
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>   So, what's stopping a haxor like me from strobbing a gateway with random
> audio... ??

This came up in the SIP security design team.

We have some hope that the receiver can pay attention to the SSRC and
discard your spam. Of course, if you can guess the SSRC, you can get in.

I once pointed out that one solution would be to extend SDP to include the
source address. Of course, this could always be spoofed.

--
Dean




From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb  8 16:41:51 2000
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> Andrew Chalupka wrote:
> 
> Assume a user, "caller", initiates a SIP call with a second user,
> "callee", through the usual means (INVITE, 200 OK, ACK).  During this
> message exchange, each end speficies the IP address and UDP port
> number(s) to which the other end should send his audio stream.  Would
> it be a violation of the SIP standard for one of the participants to
> send his audio stream to a third party (say to a server which might
> transcode the sender's audio packets into a format the other end can
> understand) and then have that third party forward the packets on to
> the appropriate destination?

No, its not a violation at all. If security is being used for the media
streams, there are key distribution issues, but there is nothing that
stops you from doing this. In fact, sending media to a different IP
address than the one which terminates the SIP is needed for MGCP/megaco
- perhaps that is what you are talking about.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Date: Tue, 8 Feb 2000 15:42:05 -0600
From: Billy Biggs <vektor@div8.net>
To: Dean Willis <dean.willis@wcom.com>
Cc: Andrew Chalupka <andcha@nortelnetworks.com>,
        "'SIP List'" <sip@lists.research.bell-labs.com>
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Dean Willis (dean.willis@wcom.com):

> >   So, what's stopping a haxor like me from strobbing a gateway with random
> > audio... ??
> 
> This came up in the SIP security design team.
> 
> We have some hope that the receiver can pay attention to the SSRC and
> discard your spam. Of course, if you can guess the SSRC, you can get in.

  Well, I don't think you can count on the SSRC being constant throughout the
call.  Why can't I stop sending audio from one source and begin sending it from
another with a different SSRC?  It wouldn't violate the context of the SIP call
if I'm still receiving media at the address I specified in my SDP.

> I once pointed out that one solution would be to extend SDP to include the
> source address. Of course, this could always be spoofed.

  Which SDP?  Would my sipcaster, which just sends media, have to send SDP just
to indicate where media would come from?  A field like that wouldn't make sense
in the context of SDP.

  It also wouldn't make sense in the context of SIP though, since SIP isn't
meant to discuss the media at all.

  So, you're kinda screwed there too...  There really isn't any place that you
should put it...  The only solution is crypto, which at this point seems too
undefined for practical use.

-- 
Billy Biggs                         vektor@div8.net
http://www.div8.net/billy       wbiggs@uwaterloo.ca



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Billy Biggs wrote:
> 

> 
>   So, you're kinda screwed there too...  There really isn't any place that you
> should put it...  The only solution is crypto, which at this point seems too
> undefined for practical use.

RTP crypto is sufficiently defined to have been interworking since the
early 90's, as long as you stick to encrypting the whole RTP packet
using DES or 3DES, not just the payload. Indeed, vat, rat and nevot have
used encryption for at least five years. (Just encrypting the payload is
useless for preventing attacks since one can't tell whether the
decryption transformation yielded a valid packet or not, at least for
codecs whose bit streams are not heavily structured. This includes G.711
and probably most other audio codecs.)

RTP-level encryption has the advantage that it doesn't inflate the
packet size and doesn't require OS support.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



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Date: Tue, 8 Feb 2000 17:17:37 -0600
From: Billy Biggs <vektor@div8.net>
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Cc: Dean Willis <dean.willis@wcom.com>,
        "'SIP List'" <sip@lists.research.bell-labs.com>
Subject: Re: Question about media desinations
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Henning Schulzrinne (schulzrinne@cs.columbia.edu):

> Billy Biggs wrote:
> > 
> >   So, you're kinda screwed there too...  There really isn't any place that you
> > should put it...  The only solution is crypto, which at this point seems too
> > undefined for practical use.
> 
> RTP crypto is sufficiently defined to have been interworking since the
> early 90's, as long as you stick to encrypting the whole RTP packet
> using DES or 3DES, not just the payload. Indeed, vat, rat and nevot have
> used encryption for at least five years. (Just encrypting the payload is
> useless for preventing attacks since one can't tell whether the
> decryption transformation yielded a valid packet or not, at least for
> codecs whose bit streams are not heavily structured. This includes G.711
> and probably most other audio codecs.)

  So, are you suggesting that for building a SIP client that receives encrypted
media it should attempt to decrypt every incomming packet and see if it
'works'?  I guess this should be only attempted if an encryption key is
specified inside the SDP, or if the user explicitly sets a shared secret key...

  Assuming that the k= line will be encrypted inside a secure SIP message, we
still need to specify at some point the algorithm to be used for the
encryption.  As far as I can tell, this is unspecified.  So, again, should a
client keep trying supported algorithms until it 'works'?

-- 
Billy Biggs                         vektor@div8.net
http://www.div8.net/billy       wbiggs@uwaterloo.ca



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Subject: Re: Question about media desinations
References: <20000208140607.A17103@div8.net> <000301bf727b$358f7080$ad9423a6@mcit.com> <20000208154204.A17839@div8.net> <38A0940F.EF3DEE1F@cs.columbia.edu> <20000208171737.A18193@div8.net>
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Billy Biggs wrote:
> 

> 
>   So, are you suggesting that for building a SIP client that receives encrypted
> media it should attempt to decrypt every incomming packet and see if it
> 'works'?  I guess this should be only attempted if an encryption key is
> specified inside the SDP, or if the user explicitly sets a shared secret key...

Indeed, this assumes that the key is known to the other side.

RFC 1890 describes how the key and algorithm are encoded, with the name
space given in RFC 1423. This string is then included in SDP.

> 
>   Assuming that the k= line will be encrypted inside a secure SIP message, we
> still need to specify at some point the algorithm to be used for the
> encryption.  As far as I can tell, this is unspecified.  So, again, should a
> client keep trying supported algorithms until it 'works'?

No. Please reread RFC 1890.


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



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb  8 20:54:04 2000
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Hi Denis:

Vovida has SIP stacks, with user Agents, available as open source on our web
site. Hope that helps.

regards
sunitha



Denis.Dagenais@nokia.com wrote:

> I need to know, who has SIP stacks (available) that include User Agents and
> Proxies
>
> Denis Dagenais






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Hi :

does anybody know of a mailing list for the SIP security design

Thanks.

--
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374



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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Hi :
<p>does anybody know of a mailing list for the SIP security design
<p>Thanks.
<pre>--&nbsp;
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374</pre>
&nbsp;</html>

--------------F00F3151697AE01B955A3BCB--




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Hi :

In this case, should the 200 OK response be generated locally by each
proxy, and then the BYE be forwarded.
Or, should the proxy in concern wait for the 200 OK from the far end,
and then propagate this on.

Thanks much!

--
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374



--------------894AF97565BB8729EDBC55F6
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Hi :
<p>In this case, should the 200 OK response be generated locally by each
proxy, and then the BYE be forwarded.
<br>Or, should the proxy in concern wait for the 200 OK from the far end,
and then propagate this on.
<p>Thanks much!
<pre>--&nbsp;
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374</pre>
&nbsp;</html>

--------------894AF97565BB8729EDBC55F6--




From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb  8 23:53:59 2000
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The proxy waits for the 200 OK and then propagates that on. The response
to BYE (and any other request) may not be a 200 OK. The request needs to
go to the UAS where it can be answered correctly.

-Jonathan R.

Sunitha Kumar wrote:
> 
> Hi :
> 
> In this case, should the 200 OK response be generated locally by each
> proxy, and then the BYE be forwarded.
> Or, should the proxy in concern wait for the 200 OK from the far end,
> and then propagate this on.
> 
> Thanks much!
> 
> --
> Sunitha Kumar
> Software Engineer
> Vovida Networks
> (408) 957 - 6374
> 
> 

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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> >   So, what's stopping a haxor like me from strobbing a gateway with random
>> audio... ??
>>
>
>Crypto.

I'm ignorant of the capabilities of the specific relevant protocols here, but isn't (simpler and cheaper) authentication, rather than encryption, called for here?

/Jonathan Lundell.



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Date: Tue, 08 Feb 2000 23:42:06 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: Question about media desinations
In-reply-to: <20000208154204.A17839@div8.net>
To: Billy Biggs <vektor@div8.net>
Cc: Andrew Chalupka <andcha@nortelnetworks.com>,
        "'SIP List'" <sip@lists.research.bell-labs.com>
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>   Which SDP?  Would my sipcaster, which just sends media, have to
> send SDP just
> to indicate where media would come from?  A field like that
> wouldn't make sense
> in the context of SDP.

Actually, I would propose the answer to that question should be "yes". And
if the source changes, a reinvite is needed to change the describing SDP.

Here's why -- lots of people are trying to build dynamic firewall rules.
They'd like to be able to inspect (not interact with) the SIP packets going
past and be able to deduce the minimal access lists that would allow the
associated RTP/UDP streams to pass. With the current usage, we have only
three elements of the 5-tuple (source address, source port, destination
address, destination port, protocol) that describes each flow (one 5-tuple
for each direction or media half-leg). We have protocol, destination
address, and destination port number. One really needs at least the source
IP address to make a reasonable ACL entry, and that doesn't protect calls
originating from multisuer machines from being jammed by other users on that
machine. Ideally, we should have the whole 5-tuple, or multiple 5-tuples
that describe additional peerings such as might be experienced from a
multihomed host that load balances its transmitting interfaces.

So, how do the firewall people work around it? They start making
assumptions, such as that each endpoint will send and receive media from the
same IP address. This is potentially a very bad assumption, but it appears
to be likely to be the state-of-the-art in SIP-aware packet filtering
firewalls for some time. I'd rather give the the data to do the job closer
to right.

The good news is that even at its worst, SIP is still easier to deal with in
the firewall than is H.323.

--
Dean




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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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        sip@lists.research.bell-labs.com
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Hi, 

I guess the question in the first mail of this thread was something
like:

A sends an INVITE to B, and B answers with a 200 OK. There is no codec
that both can support. A transcoding unit is necessary.

1) A proxy server can monitor the SDPs of both parties, and modify them
in order the media stream to go though the transcoding point. For doing
that, the proxy should be able to understand the SDP (decrypt it).
Firewalls work like this. See example "Successful SIP to SIP through SIP
Firewall Proxy" in page 34 of
http://www.cs.columbia.edu/sip/drafts/draft-johnston-sip-call-flows-00.txt

2) A could INVITE the trasnscoding point (SIP capable) using an "also"
header for B. This way the user is in control and knows that transcoding
is being peformed.

About encrypting RTP streams:
Henning wrote the following when
http://www.cs.columbia.edu/sip/drafts/draft-nair-sip-dhcp-00.txt was
released:
"A particular concern of
ours are low-complexity embedded systems (such as
http://www.cs.columbia.edu/~hgs/ephone) that may not support DNS and do
not have the computational resources to host an SLP implementation."
Probably low-processing power devices will not be able to cope with RTP
encryption/decryption in real time.

Best regards,

Gonzalo


dean.willis@wcom.com wrote:
> 
> >   Which SDP?  Would my sipcaster, which just sends media, have to
> > send SDP just
> > to indicate where media would come from?  A field like that
> > wouldn't make sense
> > in the context of SDP.
> 
> Actually, I would propose the answer to that question should be "yes". And
> if the source changes, a reinvite is needed to change the describing SDP.
> 
> Here's why -- lots of people are trying to build dynamic firewall rules.
> They'd like to be able to inspect (not interact with) the SIP packets going
> past and be able to deduce the minimal access lists that would allow the
> associated RTP/UDP streams to pass. With the current usage, we have only
> three elements of the 5-tuple (source address, source port, destination
> address, destination port, protocol) that describes each flow (one 5-tuple
> for each direction or media half-leg). We have protocol, destination
> address, and destination port number. One really needs at least the source
> IP address to make a reasonable ACL entry, and that doesn't protect calls
> originating from multisuer machines from being jammed by other users on that
> machine. Ideally, we should have the whole 5-tuple, or multiple 5-tuples
> that describe additional peerings such as might be experienced from a
> multihomed host that load balances its transmitting interfaces.
> 
> So, how do the firewall people work around it? They start making
> assumptions, such as that each endpoint will send and receive media from the
> same IP address. This is potentially a very bad assumption, but it appears
> to be likely to be the state-of-the-art in SIP-aware packet filtering
> firewalls for some time. I'd rather give the the data to do the job closer
> to right.
> 
> The good news is that even at its worst, SIP is still easier to deal with in
> the firewall than is H.323.
> 
> --
> Dean

-- 
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



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From: Itamar Gilad <ItamarG@tlv.radvision.com>
To: "'Dean Willis'" <dean.willis@wcom.com>
Cc: "'SIP List'" <sip@lists.research.bell-labs.com>
Subject: RE: Question about media desinations
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> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@wcom.com]
> Sent: Wednesday, February 09, 2000 6:45 AM
> To: Billy Biggs
> Cc: Andrew Chalupka; 'SIP List'
> Subject: RE: Question about media desinations
> 

> 
> The good news is that even at its worst, SIP is still easier 
> to deal with in
> the firewall than is H.323.
> 
> --
> Dean
> 
> 

If I'm not mistaken RTP is as much of an headache in both cases.  Leaving
the H.323 vs. SIP debate aside, I'm not sure I understand how SIP is
supposed to traverse a Firewall. I found some scattered references to this
in  RFC2543 and also something in Henning's FAQ, but nothing concrete enough
to be used as a guideline for SIP/firewall implementors.  Anyone knows of
other relevant documents on this subject?

  Itamar



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Date: Wed, 9 Feb 2000 14:00:55 +0100 (CET)
From: Lars Berggren <lars@intertex.se>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: Billy Biggs <vektor@div8.net>, David Oran <oran@cisco.com>,
        Andrew Chalupka <andcha@nortelnetworks.com>,
        "'SIP List'" <sip@lists.research.bell-labs.com>
Subject: Re: Question about media desinations
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On Tue, 8 Feb 2000, Jonathan Rosenberg wrote:

> 
> 
> Billy Biggs wrote:
> > 
> >   SDP provides a k= line to convey an encryption key, but it does not describe
> > how to practically make use of it.
> 
> True enough. The original idea is that the SIP INVITE would be
> encrypted, and the body contains the key for the media session. The key
> is used for RTP level encryption. No particular algorithm is specified
> for SIP, a hole we are working to plug.
> 

If the SIP INVITE is encrypted using the callee's public key things will
not work when traversing a firewall proxy or any other SIP-hop that needs
to change the SDP addresses in order to proxy media streams.

To support RTP level encryption through a firewall proxy (FW), is the
caller supposed to encrypt the SIP INVITE using the FW's public key so
that the FW can decrypt, change SDP, and encrypt again using the callee's
public key? It doesn't seem realistic to do this amount of processing in
the FW, I think.

I'm not sure how RTP works, but assuming the FW does not need to
inspect/change RTP packets going through it in order for RTP-things to
work, RTP level encryption through firewalls can be supported if the
encryption key can be transferred encrypted without having to encrypt the
whole SDP. I think some kind of mechanism for transferring the 
RTP-encryption-key encrypted without having the whole SDP encrypted would
be useful.

The kind of mechanisms I can come up with are:
1) The value of k= in SDP is encrypted using the recipients public key.
2) A SIP-header for transferring the RTP-key encrypted.
3) An additional body with the RTP-key encrypted.
4) Some kind of change to the SIP-end-to-end-encryption mechanism can
solve the problems of encryption through firewalls in a more general way.
5) Extra SIP re-INVITE round-trip with SDP k= only.
6) Using k=uri or k=prompt to obtain the key. This is already specified in
SDP.

Or will ipsec behave well through firewalls?

Comments?

/Lars.

> As an alternative, one can set up an ipsec association for the media.
> This just requires the SDP to somehow indicate that ipsec is desired;
> the rest is up to ike, ipsec and its related friends. This is described
> in:
> 
> http://www.normos.org/ietf/draft/draft-ietf-mmusic-sdp-qos-00.txt
> 
> although the ipsec/ike stuff is not described, just the SIP/SDP
> component.
> 
> -Jonathan R.
> 
> 
> -- 
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120 
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 
> 

Lars Berggren       <mailto:lars.berggren@intertex.se>
Intertex Data AB    tel: +46-8-6282828
Sundbyberg, Sweden  fax: +46-8-6286414




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Itamar Gilad wrote:
> 
> If I'm not mistaken RTP is as much of an headache in both cases.  Leaving
> the H.323 vs. SIP debate aside, I'm not sure I understand how SIP is
> supposed to traverse a Firewall. I found some scattered references to this
> in  RFC2543 and also something in Henning's FAQ, but nothing concrete enough
> to be used as a guideline for SIP/firewall implementors.  Anyone knows of
> other relevant documents on this subject?

Actually, yes. I am finishing writing up, along with some colleagues, a
comprehensive guide on getting SIP through NATs and firewalls. Expect an
I-D in the next few weeks.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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From: richard.swale@bt.com
To: ItamarG@tlv.radvision.com, dean.willis@wcom.com
Cc: sip@lists.research.bell-labs.com
Subject: RE: Question about media desinations
Date: Wed, 9 Feb 2000 13:14:12 -0000 
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Itamar,

You might like to look at a paper I wrote for the ETSI TIPHON project and
the associated presentation. Althought the presentation is predicated on
H.323 the problem is in fact not.

The documents are:-

16TD115
16TD137

Both of these documents can be downloaded from the ETSI website at:-

http://docbox.etsi.org/Tech-Org/TIPHON/Document/tiphon/ARCHIVES/1999/05-9912
-San_Diego/

regards

Richard
> -----Original Message-----
> From:	Itamar Gilad [SMTP:ItamarG@tlv.radvision.com]
> Sent:	Wednesday, February 09, 2000 08:20
> To:	'Dean Willis'
> Cc:	'SIP List'
> Subject:	RE: Question about media desinations
> 
> 
> > -----Original Message-----
> > From: Dean Willis [mailto:dean.willis@wcom.com]
> > Sent: Wednesday, February 09, 2000 6:45 AM
> > To: Billy Biggs
> > Cc: Andrew Chalupka; 'SIP List'
> > Subject: RE: Question about media desinations
> > 
> 
> > 
> > The good news is that even at its worst, SIP is still easier 
> > to deal with in
> > the firewall than is H.323.
> > 
> > --
> > Dean
> > 
> > 
> 
> If I'm not mistaken RTP is as much of an headache in both cases.  Leaving
> the H.323 vs. SIP debate aside, I'm not sure I understand how SIP is
> supposed to traverse a Firewall. I found some scattered references to this
> in  RFC2543 and also something in Henning's FAQ, but nothing concrete
> enough
> to be used as a guideline for SIP/firewall implementors.  Anyone knows of
> other relevant documents on this subject?
> 
>   Itamar



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Date: Wed, 09 Feb 2000 08:47:54 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: Lars Berggren <lars@intertex.se>
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        Andrew Chalupka <andcha@nortelnetworks.com>,
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Subject: Re: Question about media desinations
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Lars Berggren wrote:
> 
> If the SIP INVITE is encrypted using the callee's public key things will
> not work when traversing a firewall proxy or any other SIP-hop that needs
> to change the SDP addresses in order to proxy media streams.

Thats correct. 

> 
> To support RTP level encryption through a firewall proxy (FW), is the
> caller supposed to encrypt the SIP INVITE using the FW's public key so
> that the FW can decrypt, change SDP, and encrypt again using the callee's
> public key? It doesn't seem realistic to do this amount of processing in
> the FW, I think.

No. How would the caller know it needs to do this? Rather, I believe the
right solution is one of the following:

1. Don't encrypt the SDP. Use some new paramters in an SDP extension to
do a diffie helman exchange to arrive at a shared session key. Then, use
RTP encryption for the media. Benefits: still works through firewalls,
no ipsec overheads. Drawbacks: yet another SDP extension, may introduce
security holes by doing diffie helman ourselves.

2. Don't encrypt SDP or use it for key exchange. Based on
http://www.cs.columbia.edu/~jdrosen/papers/draft-ietf-mmusic-sdp-qos-00.txt,
make security an optional precondition for call setup. This will result
in ike/ipsec stuff to exchange keys and set up an ipsec connection for
the media. Benefits: also works through firewalls, uses standard ipsec
stuff. Drawbacks: ipsec, especially tunnel mode, adds a lot of overhead
to already small RTP packets.

> Or will ipsec behave well through firewalls?

Firewalls can be traversed. NATs are not a good story. NATs rewrite
headers which are protected by ipsec.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Date: Wed, 09 Feb 2000 09:19:51 -0500
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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        David Oran <oran@cisco.com>,
        Andrew Chalupka <andcha@nortelnetworks.com>,
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Subject: Re: Question about media desinations
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Jonathan Rosenberg wrote:
> 

> 
> No. How would the caller know it needs to do this? Rather, I believe the
> right solution is one of the following:
> 
> 1. Don't encrypt the SDP. Use some new paramters in an SDP extension to
> do a diffie helman exchange to arrive at a shared session key. Then, use
> RTP encryption for the media. Benefits: still works through firewalls,
> no ipsec overheads. Drawbacks: yet another SDP extension, may introduce
> security holes by doing diffie helman ourselves.
> 
> 2. Don't encrypt SDP or use it for key exchange. Based on
> http://www.cs.columbia.edu/~jdrosen/papers/draft-ietf-mmusic-sdp-qos-00.txt,
> make security an optional precondition for call setup. This will result
> in ike/ipsec stuff to exchange keys and set up an ipsec connection for
> the media. Benefits: also works through firewalls, uses standard ipsec
> stuff. Drawbacks: ipsec, especially tunnel mode, adds a lot of overhead
> to already small RTP packets.

A third solution which is far simpler: Don't encrypt inside the firewall
- we don't do this for other internal traffic anyway. The FW (more
precisely, the outbound proxy) encrypts the SIP message using the
recipient's public key, after modifying the SDP IP address
appropriately. This even allows the FW to insist on media encryption
between it and the outside world, even if the inside device is not
encryption-capable (as might be the case for embedded devices, although
DES support is far simpler than a complete SLP implementation, in terms
of code-space...). Having simple end systems implement IKE is, in my
view, pretty hopeless, given its complexity.

DH is subject to man-in-the-middle attacks, unless the parties have a
public number that the other party can look up securely. Not that
different from a public key in that case.


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



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What is Voicemail packaging? 
Are SIP+ and SIP-TGCP related to VM packaging? if so, how?

Please help!


Regards,

Elly Gazal
System Engineer





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The packetcable security spec specifies a 2 or 4
byte MAC (using MMH) to be appended to the RTP
packet itself. I have argued that RFC 2198 might
actually be a cleaner way to append the MAC
itself, but so long as the parties agree to the
payload format, a relatively weak MAC should make
barge in sufficiently difficult that even if an
attacker managed to get through every once in a
while, they probably wouldn't do anything
especially exciting.

		Mike

Dean Willis writes:
 > 
 > >   So, what's stopping a haxor like me from strobbing a gateway with random
 > > audio... ??
 > 
 > This came up in the SIP security design team.
 > 
 > We have some hope that the receiver can pay attention to the SSRC and
 > discard your spam. Of course, if you can guess the SSRC, you can get in.
 > 
 > I once pointed out that one solution would be to extend SDP to include the
 > source address. Of course, this could always be spoofed.
 > 
 > --
 > Dean
 > 
 > 
 > 
 > 



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb  9 13:38:13 2000
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Michael Thomas wrote:
> 
> The packetcable security spec specifies a 2 or 4
> byte MAC (using MMH) to be appended to the RTP

Is this simply hash(RTP packet | shared secret)?

Could the bad guy record a conversation, including the hash, and then
mix in the old data with the "real" one?

> packet itself. I have argued that RFC 2198 might
> actually be a cleaner way to append the MAC
> itself, but so long as the parties agree to the
> payload format, a relatively weak MAC should make
> barge in sufficiently difficult that even if an
> attacker managed to get through every once in a
> while, they probably wouldn't do anything
> especially exciting.

Since this is an RTP issue, I wonder if this shouldn't be written up for
avt. Another possibility is an RTP header extension. That, at least,
would be backward compatible with non-MMH implementations.

> 

>  >
>  > >   So, what's stopping a haxor like me from strobbing a gateway with random
>  > > audio... ??

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



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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Cc: Michael Thomas <mat@cisco.com>,
        "'SIP List'" <sip@lists.research.bell-labs.com>, rem-conf@es.net
Subject: Re: Question about media desinations
In-reply-to: Your message of "Wed, 09 Feb 2000 12:27:20 EST." <38A1A378.469B6FE1@cs.columbia.edu>
Date: Wed, 09 Feb 2000 18:03:27 +0000
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--> Henning Schulzrinne writes:
>Michael Thomas wrote:
>> I do sort of like the idea of the MAC being a
>> payload not entirely unlike a codec payload,
>> though I'm not wedded to the notion. It does have
>> the advantage that we can offer up the MAC "codec"
>> up in SDP m= as being something that the endpoint
>> is willing to receive. With RFC 2198, the sender
>> would then oblige by sending several different
>> "codecs", namely the real codec and the MAC at the
>> same time.
>
>My major concern with this is that there's a fair amount of overhead
>just to get 2 or 4 bytes across.

It's one more byte than using the RTP header extension.....

Still, it may be better to define a new RTP profile which includes the 
MAC as part of the fixed RTP header. That's probably the solution with 
the least overhead, and the protocol definition can be straightforward
RFC1890 + 4 bytes of MAC header.

Colin



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Michael Thomas wrote:
> 

>  > Since this is an RTP issue, I wonder if this shouldn't be written up for
>  > avt. Another possibility is an RTP header extension. That, at least,
>  > would be backward compatible with non-MMH implementations.
> 
> Whatever way has the least number of bytes is
> probably the right answer. Packetcable's only
> incurs the cost of the MAC itself since there is
> an agreement between the sender and receiver that
> there be the MAC tacked on to the end. Since it's
> a trailer, it ought not effect a naive
> implementation (though it would certainly violate
> the "conservative in what you send" principle).

Since RTP doesn't have a length field, a "naive" implementation would
treat the MAC bytes as G.711 samples. Not good.

> 
> I do sort of like the idea of the MAC being a
> payload not entirely unlike a codec payload,
> though I'm not wedded to the notion. It does have
> the advantage that we can offer up the MAC "codec"
> up in SDP m= as being something that the endpoint
> is willing to receive. With RFC 2198, the sender
> would then oblige by sending several different
> "codecs", namely the real codec and the MAC at the
> same time.

My major concern with this is that there's a fair amount of overhead
just to get 2 or 4 bytes across.

> 
>                Mike

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



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From: Michael Thomas <mat@cisco.com>
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To: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
Cc: Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
        Michael Thomas <mat@cisco.com>,
        "'SIP List'" <sip@lists.research.bell-labs.com>, rem-conf@es.net
Subject: Re: Question about media desinations
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Colin Perkins writes:
 > --> Henning Schulzrinne writes:
 > >Michael Thomas wrote:
 > >> I do sort of like the idea of the MAC being a
 > >> payload not entirely unlike a codec payload,
 > >> though I'm not wedded to the notion. It does have
 > >> the advantage that we can offer up the MAC "codec"
 > >> up in SDP m= as being something that the endpoint
 > >> is willing to receive. With RFC 2198, the sender
 > >> would then oblige by sending several different
 > >> "codecs", namely the real codec and the MAC at the
 > >> same time.
 > >
 > >My major concern with this is that there's a fair amount of overhead
 > >just to get 2 or 4 bytes across.
 > 
 > It's one more byte than using the RTP header extension.....
 > 
 > Still, it may be better to define a new RTP profile which includes the 
 > MAC as part of the fixed RTP header. That's probably the solution with 
 > the least overhead, and the protocol definition can be straightforward
 > RFC1890 + 4 bytes of MAC header.

Would that not lead to a combinatorial explosion
of name space with rtpmap in SDP? Ie, G711-MMH2,
etc?  I'm all for having zero overhead, but not at
the expense of making a mess of SDP. That is sort
of why I was attracted to RFC 2198. If somebody
else has a better way of doing the same thing
without requiring, for example, structured names
in SDP that would be great too.
  
	Mike



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb  9 14:07:33 2000
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Subject: using RADIUS authentication for SIP requests
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Does anyone know of a way to authenticate SIP requests using a RADIUS server?

Using a RADIUS server (which uses CHAP) to authenticate SIP requests that use
BASIC authentication is not difficult.  The SIP Server (user agent or proxy) can
simply spoof a CHAP challenge and response based on the username and password
contained in the SIP request's Authorization (or Proxy-Authorization) header and
submit it to the RADIUS server.  However, BASIC authentication is quite weak and
therefore many implementations may not support it.

Using a RADIUS server to authenticate SIP requests that use DIGEST
authentication is impossible.  This is because the method of producing the
DIGEST response is very different from the method of producing the CHAP
response.  Therefore, a SIP Server (user agent or proxy) which receives a DIGEST
challenge and response cannot spoof a CHAP challenge and response to a RADIUS
server.

Does anyone have a solution to this problem?  Perhaps a new authentication
mechanism (such as CHAP, in addition to BASIC and DIGEST) is in order?

-Jerry






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Cc: Michael Thomas <mat@cisco.com>,
        "'SIP List'" <sip@lists.research.bell-labs.com>, rem-conf@es.net
Subject: Re: Question about media desinations
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Henning Schulzrinne writes:
 > Michael Thomas wrote:
 > > 
 > > The packetcable security spec specifies a 2 or 4
 > > byte MAC (using MMH) to be appended to the RTP
 > 
 > Is this simply hash(RTP packet | shared secret)?

Yes. In packetcable, the payload is encrypted with
RC4, but the MAC is applied to the entire RTP
packet.
 
 > Could the bad guy record a conversation, including the hash, and then
 > mix in the old data with the "real" one?

Not if you use the RTP timestamp. Built in
replay protection :-)

 > > packet itself. I have argued that RFC 2198 might
 > > actually be a cleaner way to append the MAC
 > > itself, but so long as the parties agree to the
 > > payload format, a relatively weak MAC should make
 > > barge in sufficiently difficult that even if an
 > > attacker managed to get through every once in a
 > > while, they probably wouldn't do anything
 > > especially exciting.
 > 
 > Since this is an RTP issue, I wonder if this shouldn't be written up for
 > avt. Another possibility is an RTP header extension. That, at least,
 > would be backward compatible with non-MMH implementations.

Whatever way has the least number of bytes is
probably the right answer. Packetcable's only
incurs the cost of the MAC itself since there is
an agreement between the sender and receiver that
there be the MAC tacked on to the end. Since it's
a trailer, it ought not effect a naive
implementation (though it would certainly violate
the "conservative in what you send" principle).

I do sort of like the idea of the MAC being a
payload not entirely unlike a codec payload,
though I'm not wedded to the notion. It does have
the advantage that we can offer up the MAC "codec"
up in SDP m= as being something that the endpoint
is willing to receive. With RFC 2198, the sender
would then oblige by sending several different
"codecs", namely the real codec and the MAC at the
same time.

	       Mike



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Subject: Re: Question about media desinations
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Michael Thomas wrote:
> 

> Would that not lead to a combinatorial explosion
> of name space with rtpmap in SDP? Ie, G711-MMH2,

No, not really, this designation would replace the RTP/AVP in SDP with
something like RTP/AVP-MMH. The codec names would stay the same, since
they are unaffected by this change. .

> etc?  I'm all for having zero overhead, but not at
> the expense of making a mess of SDP. That is sort
> of why I was attracted to RFC 2198. If somebody
> else has a better way of doing the same thing
> without requiring, for example, structured names
> in SDP that would be great too.
> 
>         Mike

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



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb  9 16:19:12 2000
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From: "'Billy Biggs'" <vektor@div8.net>
To: Robert Sparks <Robert.Sparks@wcom.com>
Cc: "'Dean Willis'" <dean.willis@wcom.com>,
        "'SIP List'" <sip@lists.research.bell-labs.com>
Subject: Specifying media source
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Robert Sparks (Robert.Sparks@wcom.com):

> Billy - I agree with Dean that you should issue a new INVITE when you change
> your source port/address. This only makes sense of course, if the INVITE
> contains information about that source port/address. You claim such
> information doesn't make sense in the context of SDP. Why not? I can see (yet
> another) simple field addition to cover that for unicast sessions. I'm not
> proposing that right now btw - but I do strongly believe that we need to make
> the media source part of session negotiaton. This should be addressed now, in
> parallel with the important encryption discussion.

  My understanding of SIP's use of SDP is that it is used strictly as an
announcement of what you are willing to receive.  To indicate any other
information would be premature, since one can't speak objectively about the
call in full until all signalling has been completed.

  I personally think it's important that SDP not be used as a negotiation
protocol, since that would be well beyond the scope of the protocol.

  SIP definitely allows changing the media source address without updating
either of the session descriptions, since they each only discuss the receiving
end of the conversation.  Therefore, the media source can change mid-call
without an INVITE.  This seems completely valid, and even 'clean'.

  Regardless, the purpose of indicating the source address of where media is
invalidated by the simplicity of spoofing RTP.

> Billy Biggs (vektor@div8.net) :
> > Dean Willis (dean.willis@wcom.com):
> > 
> > > I once pointed out that one solution would be to extend SDP to include
> > > the source address. Of course, this could always be spoofed.
> > 
> >   Which SDP?  Would my sipcaster, which just sends media, have to send SDP
> > just to indicate where media would come from?  A field like that wouldn't
> > make sense in the context of SDP.
> > 
> >   It also wouldn't make sense in the context of SIP though, since SIP isn't
> > meant to discuss the media at all.
> > 
> >   So, you're kinda screwed there too...  There really isn't any place that
> > you should put it...  The only solution is crypto, which at this point
> > seems too undefined for practical use.

-- 
Billy Biggs                         vektor@div8.net
http://www.div8.net/billy       wbiggs@uwaterloo.ca



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Subject: RE: Question about media desinations
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The SSRC may change even when the source does not. An RTP
implementation is required to be prepared to do so in the
case of SSRC collision detection. IIRC, there's nothing
in the RTP spec to prevent an implementation from whimsically
changing to a new SSRC at any time.

Billy - I agree with Dean that you should issue a new INVITE when
you change your source port/address. This only makes sense
of course, if the INVITE contains information about that source
port/address. You claim such information doesn't make sense
in the context of SDP. Why not? I can see (yet another) simple field
addition to cover that for unicast sessions. I'm not proposing that
right now btw - but I do strongly believe that we need to make the
media source part of session negotiaton. This should be addressed
now, in parallel with the important encryption discussion.

RjS

> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of 
> Billy Biggs
> Sent: Tuesday, February 08, 2000 3:42 PM
> To: Dean Willis
> Cc: Andrew Chalupka; 'SIP List'
> Subject: Re: Question about media desinations
> 
> 
> Dean Willis (dean.willis@wcom.com):
> 
> > >   So, what's stopping a haxor like me from strobbing a 
> gateway with random
> > > audio... ??
> > 
> > This came up in the SIP security design team.
> > 
> > We have some hope that the receiver can pay attention to 
> the SSRC and
> > discard your spam. Of course, if you can guess the SSRC, 
> you can get in.
> 
>   Well, I don't think you can count on the SSRC being 
> constant throughout the
> call.  Why can't I stop sending audio from one source and 
> begin sending it from
> another with a different SSRC?  It wouldn't violate the 
> context of the SIP call
> if I'm still receiving media at the address I specified in my SDP.
> 
> > I once pointed out that one solution would be to extend SDP 
> to include the
> > source address. Of course, this could always be spoofed.
> 
>   Which SDP?  Would my sipcaster, which just sends media, 
> have to send SDP just
> to indicate where media would come from?  A field like that 
> wouldn't make sense
> in the context of SDP.
> 
>   It also wouldn't make sense in the context of SIP though, 
> since SIP isn't
> meant to discuss the media at all.
> 
>   So, you're kinda screwed there too...  There really isn't 
> any place that you
> should put it...  The only solution is crypto, which at this 
> point seems too
> undefined for practical use.
> 
> -- 
> Billy Biggs                         vektor@div8.net
> http://www.div8.net/billy       wbiggs@uwaterloo.ca
> 



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Robert Sparks wrote:
> 
> The SSRC may change even when the source does not. An RTP
> implementation is required to be prepared to do so in the
> case of SSRC collision detection. IIRC, there's nothing
> in the RTP spec to prevent an implementation from whimsically
> changing to a new SSRC at any time.
> 
> Billy - I agree with Dean that you should issue a new INVITE when
> you change your source port/address. This only makes sense
> of course, if the INVITE contains information about that source
> port/address. You claim such information doesn't make sense
> in the context of SDP. Why not? I can see (yet another) simple field
> addition to cover that for unicast sessions. I'm not proposing that
> right now btw - but I do strongly believe that we need to make the
> media source part of session negotiaton. This should be addressed
> now, in parallel with the important encryption discussion.

I'm not clear on whether the source port and address are really needed.
I thought that firewalls allowed rules without exact values for the
5-tuples. Certainly you'll need that for other applications - for
example, getting http out through a firewall via an internal proxy. In
that case, your rule only specifies a destination port and source IP
address.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb  9 19:34:09 2000
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Good Morning!

When and where Is Going to appear new version of
draft-ietf-mmusic-sip-new-00.ps?
I'v got it and Date of Expiration is February 2000 ?

Best Wishes, Piotr Kossowski



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb  9 20:22:05 2000
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Jonathan Rosenberg wrote:
> 
> Robert Sparks wrote:
> >
> > The SSRC may change even when the source does not. An RTP
> > implementation is required to be prepared to do so in the
> > case of SSRC collision detection. IIRC, there's nothing
> > in the RTP spec to prevent an implementation from whimsically
> > changing to a new SSRC at any time.
> >
> > Billy - I agree with Dean that you should issue a new INVITE when
> > you change your source port/address. This only makes sense
> > of course, if the INVITE contains information about that source
> > port/address. You claim such information doesn't make sense
> > in the context of SDP. Why not? I can see (yet another) simple field
> > addition to cover that for unicast sessions. I'm not proposing that
> > right now btw - but I do strongly believe that we need to make the
> > media source part of session negotiaton. This should be addressed
> > now, in parallel with the important encryption discussion.
> 
> I'm not clear on whether the source port and address are really needed.
> I thought that firewalls allowed rules without exact values for the
> 5-tuples. Certainly you'll need that for other applications - for
> example, getting http out through a firewall via an internal proxy. In
> that case, your rule only specifies a destination port and source IP
> address.
> 
> -Jonathan R.

Jonathan,

it is not a matter of what the firewall allows you to do but what you
want them to do.

(Most state-of-the-art routers support arbitrary filtering policies. The
filtering rules may be based on any subset of src/dst port/address,
protocol and even more. I guess that you can even parse deeper and
for example, investigate RTP sequence numbers. Of course, the processing
overhead grows with number of filtering rules and number of investigated
header fields.)

Dean's motivation behind indicating sender's port numbers in SIP/SDP
(sorry if I misunderstood) is the most restrictive filtering policy 
'default-deny-explicit-permit', where the explicit permissions are 
dynamically controlled by a SIP/SDP-aware entity and describe an
authorized 
IP stream by the 5-tuples. (BTW, could anyone point out a study on how
the processing overhead scales with the number of rules?)

If the source port number and address of the session flows
entering the firewall from the Internet are unknown to the SIP/SDP
entity, they cannot be included in the filtering rule and then
the firewall allows packets with any source addresses/ports
(i.e. even those that do not belong to the session) to reach the 
receiver.

This can be fixed by including source address/port in SIP/SDP, although
as Billy pointed out, packets with spoofed src address/port may still
fool the firewall.

Jiri


-- 
Jiri Kuthan		  Phone: ++49 / (0)30 / 3463 - 7 271
GMD FOKUS                 Fax  : ++49 / (0)30 / 3463 - 8 271
Kaiserin-Augusta-Allee 31 E-Mail : kuthan@fokus.gmd.de
D-10589 Berlin, Germany   WWW    : http://www.fokus.gmd.de/usr/kuthan



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From: Francois Menard List Account <fm-listproc@fmmo.ca>
To: Denis.Dagenais@nokia.com
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Subject: Re: SIP Stacks
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Mediatrix

-=Francois=-

On Thu, 3 Feb 2000 Denis.Dagenais@nokia.com wrote:

> I need to know, who has SIP stacks (available) that include User Agents and
> Proxies
> 
> 
> Denis Dagenais
> 




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Subject: Re: Question about media desinations
References: <003a01bf7334$f0d5cba0$8a9223a6@mcit.com> <38A203C0.9CCDF597@dynamicsoft.com> <38A211DC.2E6734B4@fokus.gmd.de>
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Jiri Kuthan wrote:
> 
> If the source port number and address of the session flows
> entering the firewall from the Internet are unknown to the SIP/SDP
> entity, they cannot be included in the filtering rule and then
> the firewall allows packets with any source addresses/ports
> (i.e. even those that do not belong to the session) to reach the
> receiver.
> 
> This can be fixed by including source address/port in SIP/SDP, although
> as Billy pointed out, packets with spoofed src address/port may still
> fool the firewall.

I see.

Another difficulty with sending the source IP address in SDP is that we
break one of the very  nice features of RTP, the independence of user
association from IP address. 

I recognize the desire for increased security (especially in light of
the recent attacks
against ebay, yahoo, etc.), but I am wondering if adding this to SDP
really buys you
much. Without this, an attacker must still guess the destination IP
address for media and
the destination port and a time during which a call is active. Adding
the source means it must also guess that. Is it enough extra?

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Subject: Re: When to send 183?
To: ssubbara@cisco.com (Srinath Subbarayan)
Date: Thu, 10 Feb 2000 09:28:47 -0600 (CST)
Cc: hgs@cs.columbia.edu, jdrosen@dynamicsoft.com,
        sip@lists.research.bell-labs.com
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>In "draft-ietf-sip-183-00.txt", one of the sections reads as follows:
>
>"5.11.6. Callee Needs Early Media
>
>   When the called UA receives an(d) INVITE message that results in the
>   need to report on the status of the media setup through a media
>   stream, the called UA has the option to send a 183 message with a
>   session description to the calling UA."
>
>If the called UA happens to be a PSTN gateway, on what basis will it
>determine
>if there is a need to report on the status of the media setup (and
>hence, send a
>183, as opposed to sending the usual 180 ringing)?

You would use the 183 only if you:

a) Are able to determine that the audio being generated
   is certainly *not* ringing (e.g. "comfort tone" or
   "pay tone" as defined in E.18x), or

b) Are unable to definitively determine that alerting
   is occuring. This really should only happen with older
   CAS protocols. ISUP and ISDN have sufficient information
   to determine what is happening on the far end.

You can also use 183 if you are able to determine that an error
has occured, but that there is a tone or announcement accompanying
it (e.g. an ACM with a cause code present), you can send a 183
to set up the media for the announcement (ideally with the
announcement text as the text string), wait for a timer (on the 
order of ~30 seconds), and then send an appropriate SIP error 
message.

-- 
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: Thu, 10 Feb 2000 09:42:10 -0600
From: Robert Sparks <Robert.Sparks@wcom.com>
Subject: 481 response to an unknown tag
To: "'SIP List' (E-mail)" <sip@lists.research.bell-labs.com>
Cc: "Henning Schulzrinne (E-mail)" <schulzrinne@cs.columbia.edu>
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I want to call implementers attention situations like this. User agent A
receives the following message having no previous knowledge of the contained
Call-ID, To, or From.

                             Agent
                               A
    INVITE A@there             |
    To: A@there;tag=abcd       |
    From: B@here               |
    Call-ID: 42@here           |
    -------------------------->|

A MUST respond to this with a 481 Call Leg/Transaction Does Not Exist.

A MUST NOT create a new call leg using the given tag (or multiple endpoints
of a forked request will end up sharing a tag, defeating its purpose).
Implementers should also take care not to blindly append a tag to the To:
line when receiving a request for an unknown Call-ID (rendering something
like To: A@there;tag=abcd;tag=1234 in the response).

I propose this clarification to the last paragraph of section 10.1.1:

"If the Call-ID was not found, the To: field is inspected for a tag
parameter. The server rejects the request if a tag is present. Otherwise,
a new call leg is created, with entries for the To, From and Call-ID
headers. The server returns a response containing the same To value, but
with a unique tag added. The tag MAY be omitted if the request contained
only one Via header field."





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Date: Thu, 10 Feb 2000 09:41:09 -0600
From: Robert Sparks <Robert.Sparks@wcom.com>
Subject: UAC use of tags received in responses
To: "'SIP List' (E-mail)" <sip@lists.research.bell-labs.com>
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Currently, a proxy is required to add a tag to a response it generates
locally if the request did not contain one. A UAC is required to use
tags it sees in final responses (creating legs for each unique tag
received) in subsequent requests to that leg.

Taken together, these requirements destroy call progress in the case
of a 407 (and several other responses a proxy might generate that 
shouldn't necessarily clobber call progress, such as 500, 503, 504,
305, 400, 411, 413, maybe even 408).

Consider the following:

  UA A           Proxy        UA B
   |  INVITE       |            |
   |  To: B        |            |
   |  From: A      |            |
   |  CallID: 1    |            |
   |  CSeq: 1 INVITE            |
   |-------------->|            |
   |               |            |
   |  407          |            |
   |  To: B;tag=1  |            |
   |  From: A      |            |
   |  CallID: 1    |            |
   |  CSeq: 1 INVITE            |
   |<--------------|            |
   |               |            |
   |  ACK          |            |
   |  To: B;tag=1  |            |
   |  From: A      |            |
   |  CallID: 1    |            |
   |  CSeq: 1 ACK  |            |
   |-------------->|            |
   |               |            |
   |  INVITE       |            |
   |  To: B;tag=1  |            |
   |  From:A       |            |
   |  (appropriate credentials) |
   |  CallID: 1    |            |
   |  CSeq: 2 INVITE            |
   |-------------->|            |
   |               |            |
   |               | INVITE     |
   |               | To: B;tag=1|
   |               | From: A    |
   |               | CallID: 1  |
   |               | CSeq: 2 INVITE
   |               |----------->|

B MUST respond now with a 481 Call Leg/Transaction Does Not Exist -
There is no legal way for the session to be successfully established.

One of the two requirements above must be modified to allow for this
scenario. Changing the requirement on the proxy, having it not add tags
for some or all locally generated responses, leads to problems determining
the appropriate destination for the subsequent ACK (pointed out by Jonathan
Rosenberg). The better solution then (suggested by both Jonathan and Alan
Johnston) is to modify the behavior of the UAC such that:

   A UAC only copies a tag into subsequent requests if it arrives in a
   200 OK to an INVITE.

So, in the above sequence, the CSeq: 2 INVITE would have no tag.

Unless discussion here uncovers a problem with this approach, I propose
adding the above sentence to section 11.3 and changing the sentence in
Section 6.41 To:
from 
   All subsequent transactions between two entities MUST include
   the "tag" parameter, as described in Section 11.
to
   Section 11 details when the "tag" parameter MUST appear in subsequent 
   requests.

NOTE that even though a UAC doesn't use a tag it sees in non-200 responses
in subsequent requests, an implementation may need to keep them as part
of transaction state. An example is being able to associate 183s with
200s on a forked INVITE.

RjS




From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 10 11:28:04 2000
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Cc: Srinath Subbarayan <ssubbara@cisco.com>, hgs@cs.columbia.edu,
        jdrosen@dynamicsoft.com, sip@lists.research.bell-labs.com
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I agree with a) and b).

Forgive me as I have not kept up with the lastest on the 183
draft, but I would argue that if it is known that an error has
occured, that a 3xx, 4xx or 5xx code should be used.  Not telling the
UA of a known error, prevents it from handling the call in a means
best suited for the user.  If we need a two stage error to frame
the error media stream then, I would argue that we need a 18x
status that means an error is about to occur.

183 should only be used when it is impossible to tell the out come
of the call setup.  Using 183 when a known error has occured
blinds the UA unnecessarily.


"Adam B. Roach" wrote:

> >In "draft-ietf-sip-183-00.txt", one of the sections reads as follows:
> >
> >"5.11.6. Callee Needs Early Media
> >
> >   When the called UA receives an(d) INVITE message that results in the
> >   need to report on the status of the media setup through a media
> >   stream, the called UA has the option to send a 183 message with a
> >   session description to the calling UA."
> >
> >If the called UA happens to be a PSTN gateway, on what basis will it
> >determine
> >if there is a need to report on the status of the media setup (and
> >hence, send a
> >183, as opposed to sending the usual 180 ringing)?
>
> You would use the 183 only if you:
>
> a) Are able to determine that the audio being generated
>    is certainly *not* ringing (e.g. "comfort tone" or
>    "pay tone" as defined in E.18x), or
>
> b) Are unable to definitively determine that alerting
>    is occuring. This really should only happen with older
>    CAS protocols. ISUP and ISDN have sufficient information
>    to determine what is happening on the far end.
>
> You can also use 183 if you are able to determine that an error
> has occured, but that there is a tone or announcement accompanying
> it (e.g. an ACM with a cause code present), you can send a 183
> to set up the media for the announcement (ideally with the
> announcement text as the text string), wait for a timer (on the
> order of ~30 seconds), and then send an appropriate SIP error
> message.
>
> --
> 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




From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 10 12:06:00 2000
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Cc: Srinath Subbarayan <ssubbara@cisco.com>, sip@lists.research.bell-labs.com
Subject: Re: When to send 183?
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"Adam B. Roach" wrote:
> 

> You would use the 183 only if you:
> 
> a) Are able to determine that the audio being generated
>    is certainly *not* ringing (e.g. "comfort tone" or
>    "pay tone" as defined in E.18x), or
> 
> b) Are unable to definitively determine that alerting
>    is occuring. This really should only happen with older
>    CAS protocols. ISUP and ISDN have sufficient information
>    to determine what is happening on the far end.
> 
> You can also use 183 if you are able to determine that an error
> has occured, but that there is a tone or announcement accompanying
> it (e.g. an ACM with a cause code present), you can send a 183
> to set up the media for the announcement (ideally with the
> announcement text as the text string), wait for a timer (on the
> order of ~30 seconds), and then send an appropriate SIP error
> message.

There seem to be two somewhat separate cases:

- progress tones and announcements

- failure tones and announcements

In either case, the receiver may or may not care to hear them. (I don't
think current telemarketing systems are smart enough to have a voice
recognition system translate "The number you have dialed is no longer in
service. The new number is ..." into a database update...)

Particularly for failure indications, it would seem to be preferable to
simply say "give me the error code (even if it's just "failure for
unknown reason") and spare me the phone lady". I suppose one way to
solve this is to only include the "Supported" header for this feature
for non-robots.

I believe we had discussed as one alternative for the failure case a 20x
variation that says "you haven't reached the real person, but here's a
robot; BYE it if you'd rather not talk to one". A potential problem with
the current solution is that some messages are infinite-length
(repeating), say for number change messages. You don't just want to cut
off after 30 seconds, since I may need to run to fetch a pencil... 


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



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Subject: Re: When to send 183?
To: dpetrie@pingtel.com (Daniel G. Petrie)
Date: Thu, 10 Feb 2000 11:57:12 -0600 (CST)
Cc: ssubbara@cisco.com (Srinath Subbarayan), hgs@cs.columbia.edu,
        jdrosen@dynamicsoft.com, sip@lists.research.bell-labs.com
In-Reply-To: <38A2E5D2.ECE703C4@pingtel.com> from "Daniel G. Petrie" at Feb 10, 2000 11:22:42 AM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>Forgive me as I have not kept up with the lastest on the 183
>draft, but I would argue that if it is known that an error has
>occured, that a 3xx, 4xx or 5xx code should be used.  Not telling the
>UA of a known error, prevents it from handling the call in a means
>best suited for the user.  If we need a two stage error to frame
>the error media stream then, I would argue that we need a 18x
>status that means an error is about to occur.
>
>183 should only be used when it is impossible to tell the out come
>of the call setup.  Using 183 when a known error has occured
>blinds the UA unnecessarily.

But it's not a violation of the protocol. It's an implementation
issue. You can design your gateways as you see fit, for the purposes
of your customers. You can even make it configurable or based on
the particular call (e.g. send pre-failure media only for queries
from hosts who have sent me ISUP bodies).

Addressing the utility of pre-failure media: if the call is going 
from PSTN to PSTN with a SIP transit network (possibly providing 
services), the *media* indication of failure is very important.
I don't want a reorder tone when some machine off in the network 
is helpfully announcing my friend's new phone number. On the other
hand, If you're terminating at a native client, they'll want an error 
code instead (or possibly a 301, as in the above case).

My intention was not to tell anyone how the protocol must be used;
I meant to inform what it allows you to do. I assume that anyone
designing a PSTN gateway has done enough analysis to determine what
behavior is appropriate to meet the requirements of their particular
customers.

-- 
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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Subject: Re: When to send 183?
To: schulzrinne@cs.columbia.edu (Henning Schulzrinne)
Date: Thu, 10 Feb 2000 12:09:42 -0600 (CST)
Cc: ssubbara@cisco.com (Srinath Subbarayan), sip@lists.research.bell-labs.com
In-Reply-To: <38A2EE7C.B35AF762@cs.columbia.edu> from "Henning Schulzrinne" at Feb 10, 2000 11:59:40 AM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>There seem to be two somewhat separate cases:
>
>- progress tones and announcements
>
>- failure tones and announcements
>
>In either case, the receiver may or may not care to hear them. (I don't
>think current telemarketing systems are smart enough to have a voice
>recognition system translate "The number you have dialed is no longer in
>service. The new number is ..." into a database update...)

They wouldn't necessarily need to. For ISUP, the ACM message in these 
circumstances contains a release code of 22 (Number Changed), along
with a new number. Whether current telemarketing systems utilize this 
information is another issue altogether -- one that I have no knowledge 
about.

Getting back on topic...

It seems a bit odd that a caller wouldn't want to know about
the progress tones (e.g. ringing, "your call has been queued,"
"your call is being forwarded to a phone outside company premises"
[which I beleive may be a legal requirement in some circumstances]).

I agree that native clients may not want to know about failure 
announcements (and certainly wouldn't want to hear failure tones).

However, as I argued in my previous message, there is a definite
utility to transiting this information between gateways under
certain circumstances. The protocol and 183 extension don't
currently prohibit this behavior; doing so would cripple the 
extension for a whole class of PSTN interoperation issues.

-- 
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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I've updated the draft accordingly, with minor editorial changes.
Thanks.

Robert Sparks wrote:
> 
> I want to call implementers attention situations like this. User agent A
> receives the following message having no previous knowledge of the contained
> Call-ID, To, or From.
> 
>                              Agent
>                                A
>     INVITE A@there             |
>     To: A@there;tag=abcd       |
>     From: B@here               |
>     Call-ID: 42@here           |
>     -------------------------->|
> 
> A MUST respond to this with a 481 Call Leg/Transaction Does Not Exist.
> 
> A MUST NOT create a new call leg using the given tag (or multiple endpoints
> of a forked request will end up sharing a tag, defeating its purpose).
> Implementers should also take care not to blindly append a tag to the To:
> line when receiving a request for an unknown Call-ID (rendering something
> like To: A@there;tag=abcd;tag=1234 in the response).
> 
> I propose this clarification to the last paragraph of section 10.1.1:
> 
> "If the Call-ID was not found, the To: field is inspected for a tag
> parameter. The server rejects the request if a tag is present. Otherwise,
> a new call leg is created, with entries for the To, From and Call-ID
> headers. The server returns a response containing the same To value, but
> with a unique tag added. The tag MAY be omitted if the request contained
> only one Via header field."

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



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Subject: INFO message
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I have two questions with regard to the propsed INFO message.  Let's say A and B
are two UAs.

1.  A calls B
2.  A sends B an Info message. Its callleg looks like the following:

        From: A
        To: B
        Call-ID: xxxxxx
3.   B receives the info message, compares the callleg with existing callleg and
accepts the info message

4.  B then sends A an info message that looks like the following:

        From: B
        To: A
        Call-ID: xxxxxx

5. A receives the message and compares the callleg.  The existing callleg known
to A has the To and From field fliped from the To and From field in the info
message A received from B.  In this case, the INFO message spec requires A to
reject B's info message with a "481 Call Leg/Transaction Does Not Exit" response.

My questions are:

a.  is my interpretation of the INFO message spec correct?
b.  If my interpretation is correct, then how do I send INFO message from B to A?

Thanks in advance for any help.

Best Regards,

Jeff Gao




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        "'Dean Willis'" <dean.willis@wcom.com>,
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Subject: Re: Question about media desinations
References: <003a01bf7334$f0d5cba0$8a9223a6@mcit.com> <38A203C0.9CCDF597@dynamicsoft.com> <38A211DC.2E6734B4@fokus.gmd.de> <38A2D1A9.BBB6C889@dynamicsoft.com>
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Jonathan Rosenberg wrote:
> Another difficulty with sending the source IP address in SDP is that we
> break one of the very  nice features of RTP, the independence of user
> association from IP address.
> 
> I recognize the desire for increased security (especially in light of
> the recent attacks
> against ebay, yahoo, etc.), but I am wondering if adding this to SDP
> really buys you
> much. Without this, an attacker must still guess the destination IP
> address for media and
> the destination port and a time during which a call is active. Adding
> the source means it must also guess that. Is it enough extra?

In general, packet filtering reduces the security risks. The more
precise
the rules the lower the risk. However, even if very precise rules are
deployed, spoofing cannot be prevented unless a kind of packet 
authorization based on a cryptographic method is used. For example, 
a MAC covering the relevant header fields would have to be included 
in every packet. The MAC secret key could be signaled with SIP.

The questions I see are:
1) Does the benefit of more precise rules justify including the source
   address/port in SDP? (rewording of what Jonathan asked)
2) Does the packet authorization make sense? Can its computational
   and bandwidth overhead be kept low? Can be some existing
   work reused? Are we going to focus on this problem?

Jiri



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Subject: Re: INFO message
To: jgao@vovida.com (Jeff Gao)
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Cc: sip@lists.research.bell-labs.com
In-Reply-To: <38A312DB.C0588855@vovida.com> from "Jeff Gao" at Feb 10, 2000 11:34:51 AM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>5. A receives the message and compares the callleg.  The existing callleg known
>to A has the To and From field fliped from the To and From field in the info
>message A received from B.  In this case, the INFO message spec requires A to
>reject B's info message with a "481 Call Leg/Transaction Does Not Exit" 
>response.

This has been hashed over *many* times on the list:
A call leg is identified by the tuple of (Call-ID, Remote-URI, Local-URI).

-- 
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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Subject: Re: Question about media desinations
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Jiri Kuthan wrote:
> 
> In general, packet filtering reduces the security risks. The more
> precise
> the rules the lower the risk. However, even if very precise rules are
> deployed, spoofing cannot be prevented unless a kind of packet
> authorization based on a cryptographic method is used. For example,
> a MAC covering the relevant header fields would have to be included
> in every packet. The MAC secret key could be signaled with SIP.
> 
> The questions I see are:
> 1) Does the benefit of more precise rules justify including the source
>    address/port in SDP? (rewording of what Jonathan asked)
> 2) Does the packet authorization make sense? Can its computational
>    and bandwidth overhead be kept low? Can be some existing
>    work reused? Are we going to focus on this problem?

Using a MAC solves a different problem. The firewall stops the attack
that has recently been launched - just flooding the host with packets.
These packets may fail authentication, but their mere presence may cause
the host to be overloaded just verifying and discarding them.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Message-ID: <86256881.007466E1.00@mwgate02.mw.3com.com>
Date: Thu, 10 Feb 2000 15:08:04 -0600
Subject: Re: Comments on draft-camarillo-mmusic-sip-isup-bcp-00.txt
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For the solution discussed below for overlap signaling, if the proxy is
stateless, how do we gaurantee that subsequent INVITE messages (with same
CallIdentifier but different call-leg with a different To header) reach the same
egress GW? Particularly, this is a problem when multiple egress GWs can be used
to reach a particular destination and there is some way of balancing the load on
egress GWs.

- hariram
(3Com)





Jonathan Rosenberg <jdrosen@dynamicsoft.com> on 11/18/99 09:57:31 AM

Sent by:  Jonathan Rosenberg <jdrosen@dynamicsoft.com>


To:   Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
cc:   Adam.Roach@ericsson.com, bartw@nortelnetworks.com,
      Gonzalo.Camarillo@ericsson.com, sip@lists.research.bell-labs.com (Ravandhu
      Hariram/MW/US/3Com)
Subject:  Re: Comments on draft-camarillo-mmusic-sip-isup-bcp-00.txt



What if, rather than encapsulating subsequent digits into INFO, they are
added to the request URI in an INVITE? A local proxy for the gateway
will respond with 484 to all those that don't have enough digits until
it gets one that does have enough. So:


Ingress GW            P             Egress GW

INV tel:1 ------------>
CSeq: 1

INV tel:12 ----------->
CSeq: 2


<---------484 --------
          CSeq: 1

INV tel:121 --------->
CSeq: 3

<---------484 -------
          CSeq: 2

INV tel:1212 --------> ---------------------->   IAM 1212---------->
CSeq: 4

<--------------484----
               CSeq: 3

INV tel:12125---------> ----------------------> SAM 5 ------------->
CSeq: 5

INV tel:121255 -------> ----------------------> SAM 5 ------------->
CSeq: 6

INV tel:1212555 ------> -----------------------> SAM 5 ------------>
CSeq: 7

.......(same thing for digits 1, 2, 1)

INV tel:12125551212----> ---------------------> SAM 2 -------------->
CSeq: 11                                        <---------ACM-------

                         <------484-----------
                                CSeq: 5
                         <------484-----------
                                CSeq: 6
                               .......
                         <------484
                                CSeq: 10
                         <---183--------------
                                CSeq: 11


Ok, so let me explain. Proxies do routing just on longest prefix matches
for INVITE's. So, proxy P can be stateless actually, as it just forwards
INVITEs that match 1212. The ingress gateway sends a re-INVITE (possibly
without waiting for a response to the previous INVITE) every time a new
digit is received from the PSTN. Once the proxy has enough digits to
route, it won't send 484 anymore, and forwards it to the egress gateway.
The egress gateway is smart, and creates an IAM from the first INVITE.
Subsequent re-INVITEs cause SAMs to be sent, containing the difference
in the set of digits from the previous INVITE. Once the egress gateway
receives an ACM (address complete, right?), it responds with a 484 to
all the INVITEs but the last, and the final INVITE with a 183. Things
progress normally from there.

Implications:

1. proxies don't have to know anything about INFO or ISUP. They just
have to know longest prefix match routing for phone numbers. They can
even be stateless.

2. egress gateways that know ISUP effectively re-generate the SAMs

3. if the egress point is not an ISUP gateway, but a PC terminal or
something that happens to have a phone number as its name, it will have
to know enough to reject each INVITE until it gets one with a request
URI/To field that matches its own number. This, BTW, is normal behavior.
A UAS rejects an INVITE if the URI is for an address thats not its own.

4. This works *even* if there is misordering of re-INVITEs. The egress
gateway computes the SAMs based on differentials. So, lets say it
receives the INVITE with 1555, then the INVITE with 155512. It then
sends two SAMs, the first with 1, the next with 2. Lets say, then, that
the INVITE with 15551 comes. Based on CSeq, the egress gateway knows its
old and rejects it.

5. The ingress gateway doesn't depend on special messages or a
completion character, like #, to send an INVITE.




Its very possible I've completely missed something, as I'm hardly an
ISUP expert. Perhaps there is additional data in SAM such that it MUST
be sent across, rather than being regenerated. But, the above solution
seems nice, as it doesn't break anything, works with non-ISUP
terminating points, doesn't require users to dial a # at the end of the
digit string, etc.

Comments?

-Jonathan R.


Gonzalo Camarillo wrote:
>
> It is nicer to send complete URIs in the "to" header. If the i-gateway
> sends the INVITE before receiveing all the digits, the "to" header is
> not complete. It would contain just a few digits of the whole telephone
> number (this assumes that with just some digits the i-gateway figures
> out where to send the INVITE).
>
> Gonzalo
>
> Adam.Roach@ericsson.com wrote:
> >
> > >The use of a timer to force enbloc signaling is not acceptable. In the case
> > >of SIP bridging, subscribers originating calls from the PSTN will not
accept
> > >an artificial post-dial delay (even if it is just a few seconds long).
> > >Overlap signaling must be supported. It is essential that the overlap
> > >process takes place within the context of a single session - using the
> > >suggested INVITE/CANCEL/INVITE approach would result in multiple abortive
> > >calls being delivered to the terminating agent, which would not be
> > >acceptable.
> > >
> > >Why not send an INVITE message when the ISUP IAM is received and then send
> > >subsequent INFO messages for each ISUP SAM? I understand that "pure" SIP
> > >agents will expect the INVITE to contain the complete address of the called
> > >party. This could be handled by including support for overlap in the
> > >"Require" extensions (draft-roach-mmusic-sip-pstn-require-header-00.txt).
If
> > >the terminator did not support the PSTN extensions, the originator could
> > >then switch to enbloc and wait for the full address to be collected.
> >
> > There are a variety of problems here. The first is determining how
> > intervening proxies will react. If, for example, a proxy is making
> > decisions about where to forward messages based on a called party URI,
> > your suggestion would require it to understand ISUP (for interpretation
> > of SAM message). It also requires it to have number analysis code and
> > dialing plan information (potentially for the entire globe).
> >
> > The problem gets even stranger for redirect servers, which expect to
> > be able to send an immediate response to an INVITE -- not sit around
> > collecting digits.
> >
> > Basically, your suggestion forces overlapped dialing deep into each
> > and every SIP node in the network that is expected to receive calls
> > originating from the PSTN network.
> >
> > I agree that this is a problem that needs to be solved if we intend
> > to use SIP between two PSTN phones; but we need to find a better
> > solution than tunelling SAMs all over the place.
> >
> > We shouldn't overlook human factors: it could be that customers are
> > willing to, for example, be trained to press "#" at the end of a
> > phone number to skip the post-dial-delay.
> >
> > --
> > 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
>
> --
> 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

--
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com









From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 10 17:12:01 2000
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Robert Sparks wrote:
> 
> One of the two requirements above must be modified to allow for this
> scenario. Changing the requirement on the proxy, having it not add tags
> for some or all locally generated responses, leads to problems determining
> the appropriate destination for the subsequent ACK (pointed out by Jonathan
> Rosenberg). The better solution then (suggested by both Jonathan and Alan
> Johnston) is to modify the behavior of the UAC such that:
> 
>    A UAC only copies a tag into subsequent requests if it arrives in a
>    200 OK to an INVITE.
> 

I would further clarify this:

     A UAC copies the tag from the response into the ACK, but it MUST
NOT
     copy the tag into any subsequent requests unless the response was
     a 200 class response to an INVITE.

Since ACK is a "subsequent" request after all.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 10 17:16:21 2000
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To: "Piotr S. Kossowski" <P.Kossowski@elka.pw.edu.pl>
Cc: sip@lists.research.bell-labs.com
Subject: Re: Expiration of draft-ietf-mmusic-sip-new-00.ps
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"Piotr S. Kossowski" wrote:
> 
> Good Morning!
> 
> When and where Is Going to appear new version of
> draft-ietf-mmusic-sip-new-00.ps?
> I'v got it and Date of Expiration is February 2000 ?

Well, the expiration date is really irrelevant since the draft was never
even submitted. We have been accumulating fixes and clarifications from
the list and bakeoffs into this draft. It might be a good time to
release it to officially get people looking at some of them. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 10 17:17:42 2000
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Robert Sparks wrote:
> 
> I propose this clarification to the last paragraph of section 10.1.1:
> 
> "If the Call-ID was not found, the To: field is inspected for a tag
> parameter. The server rejects the request if a tag is present. Otherwise,
> a new call leg is created, with entries for the To, From and Call-ID
> headers. The server returns a response containing the same To value, but
> with a unique tag added. The tag MAY be omitted if the request contained
> only one Via header field."


Actually, I'd prefer to drop this last sentence. I hate special cases
like this. I'd much rather write my client not having to consider that a
tag might not appear.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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--------------98451DCD1F943F6EBA040569
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Hi:

Wanted to clarify that the To field can contain other-parms in its
SipUrl.
Also, the grammar for other parm is

other-parm = (token | token "=" (token | quoted_string)))

What do token stand for ? Is this something that the caller , and callee
agree prior to starting their transaction?

Thanks much!

--
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374



--------------98451DCD1F943F6EBA040569
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Hi:
<p>Wanted to clarify that the To field can contain other-parms in its SipUrl.
<br>Also, the grammar for other parm is
<p>other-parm = (token | token "=" (token | quoted_string)))
<p>What do token stand for ? Is this something that the caller , and callee
agree prior to starting their transaction?
<p>Thanks much!
<pre>--&nbsp;
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374</pre>
&nbsp;</html>

--------------98451DCD1F943F6EBA040569--




From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 10 20:09:55 2000
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Subject: Re: Question about Call comparaison
References: <65256848.006D9D53.00@sampark.hss.hns.com> <38588347.9C3F1347@dynamicsoft.com> <3858FB96.EDD557C3@cs.columbia.edu> <3859D17D.64A211F@dynamicsoft.com>
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Hi:

Was any consensus reached on the URI equality.
also, when we say the optional parameters of the URI,  what are they? Do they
include other-parms?

Thanks!
sunitha


Jonathan Rosenberg wrote:

> Henning Schulzrinne wrote:
> >
> > Jonathan Rosenberg wrote:
> > >
> > > Yes, I agree that these are the same. When matching the To and From
> > > field for call legs, I would say that:
> > >
> > > 1. the display name is not included in the matching
> >
> > > 2. the matching includes tags
> > > 3. the matching be based on the user, password, host and port components
> > > of the URI, in addition to any parameters. The matching is semantic
> > > based, so that if a default component is omitted, it still matches.
> > > 4. string comparisons are case insensitive
> > > 5. an IP address that is the result of a DNS lookup on a hostname does
> > > NOT match that hostname.
> > >
> > > Matching based on URI parameters is debatable. But, we need at least the
> > > user param, and rather than special case things, I'd like to make the
> > > rule simple and consistent. So, lets include all parameters.
> > >
> >
> > This also agrees with RFC 2616 (http), which says:
> >
> > 3.2.3 URI Comparison
> >
> >    When comparing two URIs to decide if they match or not, a client
> >    SHOULD use a case-sensitive octet-by-octet comparison of the entire
> >    URIs, with these exceptions:
> >
> >       - A port that is empty or not given is equivalent to the default
> >         port for that URI-reference;
> >
> >         - Comparisons of host names MUST be case-insensitive;
> >
> >         - Comparisons of scheme names MUST be case-insensitive;
>
> Hmm, so this means that comparisons on the user and password are case
> SENSITIVE, and that comparisons on parameters are also case SENSITIVE.
> Its a bit yucky with these special cases, but I guess we should be
> consistent with http and rfc2396. In that case, I think we should
> explicitly say that the username, password, and parameters comparisons
> are case sensitive.
>
> -Jonathan R.
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com




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Its used for extensions. token is a standard construction in BNF. If you
see one and you don't know what it is for, ignore it.

-Jonathan R.

Sunitha Kumar wrote:
> 
> Hi:
> 
> Wanted to clarify that the To field can contain other-parms in its
> SipUrl.
> Also, the grammar for other parm is
> 
> other-parm = (token | token "=" (token | quoted_string)))
> 
> What do token stand for ? Is this something that the caller , and
> callee agree prior to starting their transaction?
> 
> Thanks much!
> 
> --
> Sunitha Kumar
> Software Engineer
> Vovida Networks
> (408) 957 - 6374
> 
> 

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Date: Thu, 10 Feb 2000 23:13:11 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: Question about media desinations
In-reply-to: <38A2D1A9.BBB6C889@dynamicsoft.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
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Jonathan asked:
> I recognize the desire for increased security (especially in light of
> the recent attacks
> against ebay, yahoo, etc.), but I am wondering if adding this to SDP
> really buys you
> much. Without this, an attacker must still guess the destination IP
> address for media and
> the destination port and a time during which a call is active. Adding
> the source means it must also guess that. Is it enough extra?

Without source-address rules in the firewall, an attacker can simply scan
for addresses that don't return an "ICMP administratively denied", then scan
each such address for ports that don't return "ICMP port unreachable". If
the port isn't one normally used for something else, chances are its a media
port and can be spammed with an RTP stream. This is pretty easy.

With source-address rules in the firewall, the intruder most know enough to
launch a scan with spoofed source addresses from a host likely to be in
session with a host behind the firewall, and must be able to contrive to
receive return packets to the spoofed address to detect success. This is
fairly hard.

--
Dean




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Date: Fri, 11 Feb 2000 00:10:08 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: When to send 183?
In-reply-to: <200002101528.JAA02712@b04a24.exu.ericsson.se>
To: "Adam B. Roach" <Adam.Roach@Ericsson.com>,
        Srinath Subbarayan <ssubbara@cisco.com>
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Adam wrote:
> You would use the 183 only if you:
>
> a) Are able to determine that the audio being generated
>    is certainly *not* ringing (e.g. "comfort tone" or
>    "pay tone" as defined in E.18x), or
>
> b) Are unable to definitively determine that alerting
>    is occuring. This really should only happen with older
>    CAS protocols. ISUP and ISDN have sufficient information
>    to determine what is happening on the far end.
>

I would argue that a PSTN gateway which is talking to a native UAS (rather
than another ss7 gateway) should ALWAYS return 183 upon entering a
call-pending state (such as when receiving an ACM from an egress switch).
This behavior is required for:

1) Specialized ringing tones, and
2) Network announcements, such as "The number you have reached has been
disconnected".

The early media session MUST be available BEFORE the announcement is played.
You can't wait for audio detection before setting it up, or the announcement
would be clipped.


In fact, I would go so far as to say that a gateway that does NOT implement
this feature is pretty close to useless for IP-PSTN call routing. Such a
gateway would only be useful for the PSTN-IP-PSTN role being pursued by
alternate-route carriers where SS7 release codes are tunneled back to the
originating switch (and I'm frankly not sure it even works then). This needs
to be fixed in the SIP-T work if we ever want to talk to REAL internet
devices, instead of just tunnelling calls between gateways. I don't believe
any real telco would buy an IP-PSTN gateway that doesn't support 183 with
early media -- they just wouldn't be useful.

One might argue that a UAS which knows it is placing a call thru an IP-PSTN
gateway might want to state a requirement for 183 support in its initial
INVITE. This approach would allow a gateway to decide whether to 183 based
on the initiating INVITE, thereby allowing a gateway to server pstn-ip-pstn
and ip-pstn transparently.

One might also wonder how a forking proy would deal with 183 messages
relating to a call that has been forked.

--
Dean




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"Gazal, Elly" wrote:
> 
> What is Voicemail packaging?
> Are SIP+ and SIP-TGCP related to VM packaging? if so, how?

I have no idea what voicemail packaging is. Nor do I know what SIP-TGCP
is.

SIP+ and SIP-T (aka SIP-T-BCP) is a SIP extension plus some interworking
guidlines that specify how ISUP messages can be carried in SIP to
provide ISUP transparency. It has nothing to do with voicemail so far as
I know.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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From: "Rosen, Brian" <brosen@fore.com>
To: "Gazal, Elly" <Elly_Gazal@icomverse.com>
Cc: "'SIP_ML'" <sip@lists.research.bell-labs.com>
Subject: RE: voicemail packaging
Date: Fri, 11 Feb 2000 10:12:02 -0500
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Aaah, perhaps this is a misdirected query.

There is an effort to define an MGCP Voicemail package in ISC.

If that is what the enquiry is about, join the ISC's MGCP list.

Brian

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Friday, February 11, 2000 1:32 AM
> To: Gazal, Elly
> Cc: 'SIP_ML'
> Subject: Re: voicemail packaging
> 
> 
> 
> 
> "Gazal, Elly" wrote:
> > 
> > What is Voicemail packaging?
> > Are SIP+ and SIP-TGCP related to VM packaging? if so, how?
> 
> I have no idea what voicemail packaging is. Nor do I know 
> what SIP-TGCP
> is.
> 
> SIP+ and SIP-T (aka SIP-T-BCP) is a SIP extension plus some 
> interworking
> guidlines that specify how ISUP messages can be carried in SIP to
> provide ISUP transparency. It has nothing to do with 
> voicemail so far as
> I know.
> 
> -Jonathan R.
> 
> -- 
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120 
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 



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Subject: Re: When to send 183?
To: dean.willis@wcom.com (Dean Willis)
Date: Fri, 11 Feb 2000 09:16:57 -0600 (CST)
Cc: ssubbara@cisco.com (Srinath Subbarayan), hgs@cs.columbia.edu,
        jdrosen@dynamicsoft.com, sip@lists.research.bell-labs.com
In-Reply-To: <002301bf7456$a5c4a280$1af523a6@mcit.com> from "Dean Willis" at Feb 11, 2000 12:10:08 AM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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Dean Willis writes:
>Adam wrote:
>> You would use the 183 only if you:
>>
>> a) Are able to determine that the audio being generated
>>    is certainly *not* ringing (e.g. "comfort tone" or
>>    "pay tone" as defined in E.18x), or
>>
>> b) Are unable to definitively determine that alerting
>>    is occuring. This really should only happen with older
>>    CAS protocols. ISUP and ISDN have sufficient information
>>    to determine what is happening on the far end.
>>
>
>I would argue that a PSTN gateway which is talking to a native UAS (rather
>than another ss7 gateway) should ALWAYS return 183 upon entering a
>call-pending state (such as when receiving an ACM from an egress switch).

I would argue that this is only appropriate when no other 1xx code
will adequately describe the state. 

>This behavior is required for:
>
>1) Specialized ringing tones, and

If you're providing a specialized ring tone, I would argue that you
should send a 180 *with an SDP body* to indicate that you're providing 
the ringtone yourself. This way, the user hears the ringtone you want
him to hear, and the phone (and any other signal-level node) knows
that the call is ringing. My hope for 183 is that it does not get
used as a generic "ACM -- set up media" type of message that depricates
180-182. It is a suppliment to fill in the gaps that 180-182
leave.

>2) Network announcements, such as "The number you have reached has been
>disconnected".

Wouldn't a "410 The number you have reached has been disconnected"
message serve the same purpose for native clients? I sure would
prefer to read the text on a display than suffer through the
recording. I beleive this is a more appropriate approach if
you *know* for a fact that you're terminating on a native client.
If you don't know whether you're hitting a PSTN gateway, I'd
argue for playing the announcement.

This is a fairly trivial task to accomplish by doing a simple
mapping of release codes to appropriate SIP codes in the
PSTN gateway; e.g.:

ISUP    SIP
1       410 Number Disconnected
2       404 No route to network
3       404 No route to destination
8       503 Call has been preempted

etc.

My contributions to the call flows draft will go over this in
more detail.

-- 
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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Subject: Re: Question about Call comparaison
References: <65256848.006D9D53.00@sampark.hss.hns.com> <38588347.9C3F1347@dynamicsoft.com> <3858FB96.EDD557C3@cs.columbia.edu> <3859D17D.64A211F@dynamicsoft.com> <38A360A6.92852E3C@vovida.com>
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Yes, consensus was reached. The rules are:

1. matching over domain names are case insensitive, all others are case
sensitive. 
2. The matching is over all fields: user, password, domain, port AND all
parameters, including the optional and extension parameters
3. If a parameter is not present, but it has a default value, the
matching is based on the default (i.e., a URL without a port and a URL
with port 5060 match)
4. A URL with a domain name does not match a URL with an IP address,
where that address is the result of a DNS lookup of the domain name.

-Jonathan R.

Sunitha Kumar wrote:
> 
> Hi:
> 
> Was any consensus reached on the URI equality.
> also, when we say the optional parameters of the URI,  what are they? Do they
> include other-parms?
> 
> Thanks!
> sunitha
> 
> Jonathan Rosenberg wrote:
> 
> > Henning Schulzrinne wrote:
> > >
> > > Jonathan Rosenberg wrote:
> > > >
> > > > Yes, I agree that these are the same. When matching the To and From
> > > > field for call legs, I would say that:
> > > >
> > > > 1. the display name is not included in the matching
> > >
> > > > 2. the matching includes tags
> > > > 3. the matching be based on the user, password, host and port components
> > > > of the URI, in addition to any parameters. The matching is semantic
> > > > based, so that if a default component is omitted, it still matches.
> > > > 4. string comparisons are case insensitive
> > > > 5. an IP address that is the result of a DNS lookup on a hostname does
> > > > NOT match that hostname.
> > > >
> > > > Matching based on URI parameters is debatable. But, we need at least the
> > > > user param, and rather than special case things, I'd like to make the
> > > > rule simple and consistent. So, lets include all parameters.
> > > >
> > >
> > > This also agrees with RFC 2616 (http), which says:
> > >
> > > 3.2.3 URI Comparison
> > >
> > >    When comparing two URIs to decide if they match or not, a client
> > >    SHOULD use a case-sensitive octet-by-octet comparison of the entire
> > >    URIs, with these exceptions:
> > >
> > >       - A port that is empty or not given is equivalent to the default
> > >         port for that URI-reference;
> > >
> > >         - Comparisons of host names MUST be case-insensitive;
> > >
> > >         - Comparisons of scheme names MUST be case-insensitive;
> >
> > Hmm, so this means that comparisons on the user and password are case
> > SENSITIVE, and that comparisons on parameters are also case SENSITIVE.
> > Its a bit yucky with these special cases, but I guess we should be
> > consistent with http and rfc2396. In that case, I think we should
> > explicitly say that the username, password, and parameters comparisons
> > are case sensitive.
> >
> > -Jonathan R.
> > --
> > Jonathan D. Rosenberg                       200 Executive Drive
> > Chief Scientist                             Suite 120
> > dynamicsoft                                 West Orange, NJ 07052
> > jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> > http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> > http://www.dynamicsoft.com

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Dean Willis wrote:
> 
> Jonathan asked:
> > I recognize the desire for increased security (especially in light of
> > the recent attacks
> > against ebay, yahoo, etc.), but I am wondering if adding this to SDP
> > really buys you
> > much. Without this, an attacker must still guess the destination IP
> > address for media and
> > the destination port and a time during which a call is active. Adding
> > the source means it must also guess that. Is it enough extra?
> 
> Without source-address rules in the firewall, an attacker can simply scan
> for addresses that don't return an "ICMP administratively denied", then scan
> each such address for ports that don't return "ICMP port unreachable". If
> the port isn't one normally used for something else, chances are its a media
> port and can be spammed with an RTP stream. This is pretty easy.

This presumes a firewall sends an ICMP error message when it discards
the packet. I would
think they would blackhole them, which will eliminate the ability to
conduct this attack
you describe.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 11 11:52:15 2000
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I think it must not be encrypted. This is because (1) call leg
identification may need to be done in a proxy, (2) its possible with the
call control spec that there are two transactions that have the same To,
Call-ID, CSeq but different From, in which case the From is needed for
transaction identification. 

We should therefore remove this tidbit from 13.1.1.

Thanks,
Jonathan R.

archow@hss.hns.com wrote:
> 
> Hi,
> Please refer to section 13.1.1 of RFC 2543
> 
> "The From header field SHOULD
>    normally remain in the clear, but MAY be encrypted if required, in
>    which case some proxies MAY return a 401 (Unauthorized) status if
>    they require a From field."
> 
> The above indicates that its possible for FROM Header to be encrypted if so
>  desired.
> 
> Now, please refer to Section 6 Table 4 which clearly mentions From (and
> others) to have an "n" status in the enc colum implying
> they MUST NOT be encrypted.
> 
> So which is the correct statememt ?
> 
> Regds
> Arjun
> --
> Arjun Roychowdhury @ Hughes Software Systems

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Subject: Re: Question about Call comparaison
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Jonathan Rosenberg wrote:
> 
> Yes, consensus was reached. The rules are:
> 
> 1. matching over domain names are case insensitive, all others are case
> sensitive.

I believe we agreed that parameter names are CI, also. Otherwise
implementations would have to remember not only, say, that a To tag had
value "xyz" but also the tag string itself just to get the case right.
Not very good.

-- 
Anders Kristensen <ak@hplb.hpl.hp.com>,
http://www-uk.hpl.hp.com/people/ak/
Hewlett-Packard Labs, Bristol, UK



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Jonathan Rosenberg wrote:
> 
> I think it must not be encrypted. This is because (1) call leg
> identification may need to be done in a proxy, (2) its possible with the
> call control spec that there are two transactions that have the same To,
> Call-ID, CSeq but different From, in which case the From is needed for
> transaction identification.
> 
> We should therefore remove this tidbit from 13.1.1.

The spec has been updated accordingly.



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 11 14:15:58 2000
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I agree that having a 4xx/5xx immediately is preferable. The only
strange thing is that if the announcement is long (including infinite -
I don't know how many times the number change or area code change
announcement is repeated in the PSTN), the other side hangs up, but
can't really send a BYE, since there never was a successful call to
begin with. When should the calling gateway stop sending audio?

Termination audio seems much closer to a success, from a UA perspective.
If there's a human or PSTN gateway at the calling end, it seems like the
behavior is closer to success, including hanging up to terminate the
announcement, than a failure. Clearly, even for humans, some distinct
status code (other than 200) should be provided.



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Subject: Re: Question about Call comparaison
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Anders Kristensen wrote:
> 
> Jonathan Rosenberg wrote:
> >
> > Yes, consensus was reached. The rules are:
> >
> > 1. matching over domain names are case insensitive, all others are case
> > sensitive.
> 
> I believe we agreed that parameter names are CI, also. Otherwise
> implementations would have to remember not only, say, that a To tag had
> value "xyz" but also the tag string itself just to get the case right.
> Not very good.

Note that the "tag" is not part of the URL. Thus, call matching = URL
matching for From/To + tag matching for To + Call-ID matching.

I've added wording on call leg matching in the From header section.



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Henning Schulzrinne wrote:
> 
> Anders Kristensen wrote:
> >
> > Jonathan Rosenberg wrote:
> > >
> > > Yes, consensus was reached. The rules are:
> > >
> > > 1. matching over domain names are case insensitive, all others are case
> > > sensitive.
> >
> > I believe we agreed that parameter names are CI, also. Otherwise
> > implementations would have to remember not only, say, that a To tag had
> > value "xyz" but also the tag string itself just to get the case right.
> > Not very good.
> 
> Note that the "tag" is not part of the URL. Thus, call matching = URL
> matching for From/To + tag matching for To + Call-ID matching.

Right you are, so substitute "tag" for "transport" and the comment
holds.

> 
> I've added wording on call leg matching in the From header section.

-- 
Anders Kristensen <ak@hplb.hpl.hp.com>,
http://www-uk.hpl.hp.com/people/ak/
Hewlett-Packard Labs, Bristol, UK



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You are right on the mark here.  Comments below.

"Adam B. Roach" wrote:

> Dean Willis writes:
> >Adam wrote:
> >> You would use the 183 only if you:
> >>
> >> a) Are able to determine that the audio being generated
> >>    is certainly *not* ringing (e.g. "comfort tone" or
> >>    "pay tone" as defined in E.18x), or
> >>
> >> b) Are unable to definitively determine that alerting
> >>    is occuring. This really should only happen with older
> >>    CAS protocols. ISUP and ISDN have sufficient information
> >>    to determine what is happening on the far end.
> >>
> >
> >I would argue that a PSTN gateway which is talking to a native UAS (rather
> >than another ss7 gateway) should ALWAYS return 183 upon entering a
> >call-pending state (such as when receiving an ACM from an egress switch).
>
> I would argue that this is only appropriate when no other 1xx code
> will adequately describe the state.

Yes!

>
>
> >This behavior is required for:
> >
> >1) Specialized ringing tones, and
>
> If you're providing a specialized ring tone, I would argue that you
> should send a 180 *with an SDP body* to indicate that you're providing
> the ringtone yourself. This way, the user hears the ringtone you want
> him to hear, and the phone (and any other signal-level node) knows
> that the call is ringing. My hope for 183 is that it does not get
> used as a generic "ACM -- set up media" type of message that depricates
> 180-182. It is a suppliment to fill in the gaps that 180-182
> leave.

Absolutely!  Using 183 when a more descriptive status is available
looses information. That can only serve to limit the knowledge that the
UA has and subsequently diminish what it can do to improve the user
experience.

>
>
> >2) Network announcements, such as "The number you have reached has been
> >disconnected".
>
> Wouldn't a "410 The number you have reached has been disconnected"
> message serve the same purpose for native clients? I sure would
> prefer to read the text on a display than suffer through the
> recording. I beleive this is a more appropriate approach if
> you *know* for a fact that you're terminating on a native client.
> If you don't know whether you're hitting a PSTN gateway, I'd
> argue for playing the announcement.
>
> This is a fairly trivial task to accomplish by doing a simple
> mapping of release codes to appropriate SIP codes in the
> PSTN gateway; e.g.:
>
> ISUP    SIP
> 1       410 Number Disconnected
> 2       404 No route to network
> 3       404 No route to destination
> 8       503 Call has been preempted
>
> etc.
>
> My contributions to the call flows draft will go over this in
> more detail.
>
> --
> 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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It seems that your argument is for early media, not necessarily
for a 183 status.  The point I have been trying to make is
that the status 183 is too generic if the outcome is known
to be a call setup failure. One should avoid using the status
183 if a more descriptive status can be provided.

Early media makes sense with many different status.  I think
that we need to be very careful in keeping these issues
separate.


Dean Willis wrote:

> Adam wrote:
> > You would use the 183 only if you:
> >
> > a) Are able to determine that the audio being generated
> >    is certainly *not* ringing (e.g. "comfort tone" or
> >    "pay tone" as defined in E.18x), or
> >
> > b) Are unable to definitively determine that alerting
> >    is occuring. This really should only happen with older
> >    CAS protocols. ISUP and ISDN have sufficient information
> >    to determine what is happening on the far end.
> >
>
> I would argue that a PSTN gateway which is talking to a native UAS (rather
> than another ss7 gateway) should ALWAYS return 183 upon entering a
> call-pending state (such as when receiving an ACM from an egress switch).
> This behavior is required for:
>
> 1) Specialized ringing tones, and
> 2) Network announcements, such as "The number you have reached has been
> disconnected".
>
> The early media session MUST be available BEFORE the announcement is played.
> You can't wait for audio detection before setting it up, or the announcement
> would be clipped.
>
> In fact, I would go so far as to say that a gateway that does NOT implement
> this feature is pretty close to useless for IP-PSTN call routing. Such a
> gateway would only be useful for the PSTN-IP-PSTN role being pursued by
> alternate-route carriers where SS7 release codes are tunneled back to the
> originating switch (and I'm frankly not sure it even works then). This needs
> to be fixed in the SIP-T work if we ever want to talk to REAL internet
> devices, instead of just tunnelling calls between gateways. I don't believe
> any real telco would buy an IP-PSTN gateway that doesn't support 183 with
> early media -- they just wouldn't be useful.
>
> One might argue that a UAS which knows it is placing a call thru an IP-PSTN
> gateway might want to state a requirement for 183 support in its initial
> INVITE. This approach would allow a gateway to decide whether to 183 based
> on the initiating INVITE, thereby allowing a gateway to server pstn-ip-pstn
> and ip-pstn transparently.
>
> One might also wonder how a forking proy would deal with 183 messages
> relating to a call that has been forked.
>
> --
> Dean




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Anders Kristensen wrote:
> 
> > 1. matching over domain names are case insensitive, all others are case
> > sensitive.
> 
> I believe we agreed that parameter names are CI, also. Otherwise
> implementations would have to remember not only, say, that a To tag had
> value "xyz" but also the tag string itself just to get the case right.
> Not very good.

I'm confused; having it CS is easier. I'll store the tag as a string,
and do a normal
string comparison on it to "get the case right". Whats the difficulty?

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Date: Sat, 12 Feb 2000 00:04:19 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: When to send 183?
In-reply-to: <200002111516.JAA05863@b04a24.exu.ericsson.se>
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>
> Wouldn't a "410 The number you have reached has been disconnected"
> message serve the same purpose for native clients? I sure would
> prefer to read the text on a display than suffer through the
> recording. I beleive this is a more appropriate approach if
> you *know* for a fact that you're terminating on a native client.
> If you don't know whether you're hitting a PSTN gateway, I'd
> argue for playing the announcement.

As I understand it, the SS7 release code is often quite vague, and certainly
doesn't convey the "NEW NUMBER IS . . . " information contained in the voice
recording.

AS for 180 with ring-SDP -- how does the network gateway anticipate the ring
treatment that will be applied by a domestic terminating switch? Can't, can
it?

I'd suggest that the easiest general solution is to send a 183 with media
following the ACM. This gets the RTP stream established in time for network
call treatment.

I don't think I think 183 is an edge condition -- I think it is far more
lilkely to be the norm for PSTN interworking.

--
Dean




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Date: Sat, 12 Feb 2000 00:15:45 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: Question about media desinations
In-reply-to: <38A43AD6.B436A9F0@dynamicsoft.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Try scanning the net at 63.81.221.0/24, for example by telneting to port 122
on 63.81.221.234 -- you'll get back lots of "packet filtered" messages. The
default behavior in most systems is to return such a message. Some firewall
implementors turn them off. Most don't.

--
Dean

Jonathan asked:
> This presumes a firewall sends an ICMP error message when it discards
> the packet. I would
> think they would blackhole them, which will eliminate the ability to
> conduct this attack
> you describe.




From owner-sip-outgoing@lists.research.bell-labs.com  Sat Feb 12 09:03:59 2000
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Subject: Re: Question about Call comparaison
References: <65256848.006D9D53.00@sampark.hss.hns.com> <38588347.9C3F1347@dynamicsoft.com> <3858FB96.EDD557C3@cs.columbia.edu> <3859D17D.64A211F@dynamicsoft.com> <38A360A6.92852E3C@vovida.com> <38A42D91.49442FF8@dynamicsoft.com> <38A45338.3FD08932@hpl.hp.com> <38A4978D.50F2D34F@dynamicsoft.com>
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Jonathan Rosenberg wrote:
> 
> Anders Kristensen wrote:
> >
> > > 1. matching over domain names are case insensitive, all others are case
> > > sensitive.
> >
> > I believe we agreed that parameter names are CI, also. Otherwise
> > implementations would have to remember not only, say, that a To tag had
> > value "xyz" but also the tag string itself just to get the case right.
> > Not very good.
> 
> I'm confused; having it CS is easier. I'll store the tag as a string,
> and do a normal
> string comparison on it to "get the case right".

Right, actually it depends on the implementation. Currently our SipURL
class has attributes for RFC 2543 defined SIP URL parameters: user,
maddr ttl, transport, and method, whereas other-params are stored in a
map.

There are advantages and disadvantages to this approach. Amongst the
advantages is that it saves us from having to remember the actual
attribute name used on the wire for these "well-defined" attributes, and
that retrieval is faster than having to lookup in a hashtable or what
have you.

My posting was predicated on the assumption that people would naturally
want to store rfc 2543 SIP URL parameters different from other-params,
which I realize now isn't necessarily the case.

Anyway, I dont really feel very strongly about it. CS parameter names is
fine with me.

Anders

-- 
Anders Kristensen <ak@hplb.hpl.hp.com>,
http://www-uk.hpl.hp.com/people/ak/
Hewlett-Packard Labs, Bristol, UK



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Subject: Re: Question about Call comparaison
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> Jonathan Rosenberg wrote:
> > 
> > Anders Kristensen wrote:
> > >
> > > > 1. matching over domain names are case insensitive, all others are case
> > > > sensitive.
> > >
> > > I believe we agreed that parameter names are CI, also. Otherwise
> > > implementations would have to remember not only, say, that a To tag had
> > > value "xyz" but also the tag string itself just to get the case right.
> > > Not very good.
> > 

> > I'm confused; having it CS is easier. I'll store the tag as a string,
> > and do a normal
> > string comparison on it to "get the case right".
> 
Is this an argument for CS names or CS values?

> Right, actually it depends on the implementation. Currently our SipURL
> class has attributes for RFC 2543 defined SIP URL parameters: user,
> maddr ttl, transport, and method, whereas other-params are stored in a
> map.
> 

We take a very similar approach.

> There are advantages and disadvantages to this approach. Amongst the
> advantages is that it saves us from having to remember the actual
> attribute name used on the wire for these "well-defined" attributes, and
> that retrieval is faster than having to lookup in a hashtable or what
> have you.
> 

Exactly.

> My posting was predicated on the assumption that people would naturally
> want to store rfc 2543 SIP URL parameters different from other-params,
> which I realize now isn't necessarily the case.
> 

I too think it is a very natural approach, however, some might want just
to deal with simple strings.

> Anyway, I dont really feel very strongly about it. CS parameter names is
> fine with me.
> 

If parameter names are case sensitive, then what is the "correct" case for
the standard SIP URL parameters (is it "Transport", "TRANSPORT", "transport", 
etc?). Or are you saying that a SIP URL with a "transport=tcp" parameter
does not match a SIP URL with a "Transport=tcp" parameter (where both cases
would be interpreted as a correctly formatted transport parameter)?

> Anders
> 
> -- 
> Anders Kristensen <ak@hplb.hpl.hp.com>,
> http://www-uk.hpl.hp.com/people/ak/
> Hewlett-Packard Labs, Bristol, UK
> 

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



From owner-sip-outgoing@lists.research.bell-labs.com  Sun Feb 13 12:12:47 2000
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Subject: Re: Question about media desinations
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This raises an important point. I don't want to make changes to SDP and
its usage within SIP to fix a problem that can be prevented by an
administrative change in firewall behavior. If the firewalls did not
return these error messages, do you still believe including the source
port and address in the SDP is useful? 

-Jonathan R.

Dean Willis wrote:
> 
> Try scanning the net at 63.81.221.0/24, for example by telneting to port 122
> on 63.81.221.234 -- you'll get back lots of "packet filtered" messages. The
> default behavior in most systems is to return such a message. Some firewall
> implementors turn them off. Most don't.
> 
> --
> Dean
> 
> Jonathan asked:
> > This presumes a firewall sends an ICMP error message when it discards
> > the packet. I would
> > think they would blackhole them, which will eliminate the ability to
> > conduct this attack
> > you describe.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Sun Feb 13 12:15:34 2000
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Sean Olson wrote:
> 
> > > I'm confused; having it CS is easier. I'll store the tag as a string,
> > > and do a normal
> > > string comparison on it to "get the case right".
> >
> Is this an argument for CS names or CS values?

Both are case sensitive.

> 
> > Right, actually it depends on the implementation. Currently our SipURL
> > class has attributes for RFC 2543 defined SIP URL parameters: user,
> > maddr ttl, transport, and method, whereas other-params are stored in a
> > map.
> >
> 
> We take a very similar approach.
> 
> > There are advantages and disadvantages to this approach. Amongst the
> > advantages is that it saves us from having to remember the actual
> > attribute name used on the wire for these "well-defined" attributes, and
> > that retrieval is faster than having to lookup in a hashtable or what
> > have you.
> >
> 
> Exactly.

Thats fine, I still don't see why case insensitivity helps you. Consider
the tag parameter. The parameter is case sensistive, which means "tag"
is the one defined in rfc2543, and "TAG" is an other-param. So, when you
need to generate the message, read your tag parameter attribute from the
SipURL class, and then spit out "tag=" + myURL.tag(); or whatever. 

I can understand why you'd prefer case insensitive if "tag" and "TAG"
both refer to the same thing, but you needed to remember which was
received.

> If parameter names are case sensitive, then what is the "correct" case for
> the standard SIP URL parameters (is it "Transport", "TRANSPORT", "transport",
> etc?). Or are you saying that a SIP URL with a "transport=tcp" parameter
> does not match a SIP URL with a "Transport=tcp" parameter (where both cases
> would be interpreted as a correctly formatted transport parameter)?

Yes, that is what I'm saying. I believe this is what is implied in http
1.1 also.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Sun Feb 13 12:25:46 2000
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Generally speaking, unless there's a strong reason not to, CS comparison
is far preferable, since with UTF-8, doing CI comparison is non-trivial.
In full fledged OS environments, somebody probably has already taken
care of all the hassle of deciding what the upper case equivalents of
different characters are, but in an embedded environment, most people
are going to deal correctly with ASCII, if we're lucky. Thus, to avoid
I18N surprises later, byte-by-byte comparison (i.e, CS) seems generally
preferable unless we are forced by convention, e.g., for hostnames
(where currently non-ASCII letters are less of an issue) and for header
names (where non-ASCII choices for standardized headers are not likely).
CS comparison is also somewhat slower.



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Date: Wed, 09 Feb 2000 14:01:39 -0800
From: Paul Krumviede <paul@mci.net>
Subject: Re: using RADIUS authentication for SIP requests
In-reply-to: <86256880.005CE532.00@mwgate02.mw.3com.com>
To: Jerry Mahler <Jerry_Mahler@mw.3com.com>, sip@lists.research.bell-labs.com
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--On Wednesday, 09 February, 2000 10:51 -0600 Jerry Mahler 
<Jerry_Mahler@mw.3com.com> wrote:

> Does anyone know of a way to authenticate SIP requests using a RADIUS
> server?
>
> Using a RADIUS server (which uses CHAP) to authenticate SIP requests
> that use BASIC authentication is not difficult.  The SIP Server (user
> agent or proxy) can simply spoof a CHAP challenge and response based on
> the username and password contained in the SIP request's Authorization
> (or Proxy-Authorization) header and submit it to the RADIUS server.
> However, BASIC authentication is quite weak and therefore many
> implementations may not support it.
>
> Using a RADIUS server to authenticate SIP requests that use DIGEST
> authentication is impossible.  This is because the method of producing
> the DIGEST response is very different from the method of producing the
> CHAP response.  Therefore, a SIP Server (user agent or proxy) which
> receives a DIGEST challenge and response cannot spoof a CHAP challenge
> and response to a RADIUS server.
>
> Does anyone have a solution to this problem?  Perhaps a new
> authentication mechanism (such as CHAP, in addition to BASIC and DIGEST)
> is in order?

Note that in the case of CHAP, a RADIUS server has no way of authenticating
the requestor (other than by source IP address, which isn't very strong). 
Running
such an authentication protocol over something that did provide such 
authentication
might be useful. An alternative would be to look at ways of specifying a 
way to
carry digest authentication in RADIUS; a motivation for doing so might be 
the
desire to use the HMAC construction in HTTP digest rather than the 
construction
in CHAP.

-paul




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Message-ID: <013901bf7715$d1583260$4834c526@rbadriw.santera.com>
From: "Raja Badri" <raja.badri@santera.com>
To: "sip ML" <sip@lists.research.bell-labs.com>
Subject: SIP BCP-T question
Date: Mon, 14 Feb 2000 12:03:36 -0600
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This is a multi-part message in MIME format.

------=_NextPart_000_0136_01BF76E3.860520C0
Content-Type: text/plain;
	charset="iso-8859-1"
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Hi,

I have one question regarding the SIP BCP-T.
INFO message outlines the Mid-Call ISUP messages.
What is the reason behind not including the ISUP maintenance=20
messages like Block, Unblock and Reset ?

Thanks,
Raja Badri
Santera Systems Inc.


------=_NextPart_000_0136_01BF76E3.860520C0
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD W3 HTML//EN">
<HTML>
<HEAD>

<META content=3Dtext/html;charset=3Diso-8859-1 =
http-equiv=3DContent-Type>
<META content=3D'"MSHTML 4.72.3110.7"' name=3DGENERATOR>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT color=3D#000000 size=3D2>Hi,</FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000000 size=3D2>I have one question regarding the =
SIP=20
BCP-T.</FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2></FONT><FONT size=3D2>INFO message =
outlines the=20
Mid-Call ISUP messages.</FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2>What is the reason behind not =
including the ISUP=20
maintenance </FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2>messages like Block, Unblock and =
Reset=20
?</FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#000000 size=3D2>Thanks,</FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2>Raja Badri</FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2>Santera Systems Inc.</FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2></FONT>&nbsp;</DIV></BODY></HTML>

------=_NextPart_000_0136_01BF76E3.860520C0--




From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 14 13:25:57 2000
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From: Aparna.Vemuri@Level3.com
To: raja.badri@santera.com, sip@lists.research.bell-labs.com
Subject: RE: SIP BCP-T question
Date: Mon, 14 Feb 2000 11:20:26 -0700
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Raja,
Blocking/ resetting of circuits and circuit-groups is not necessary between
MGCs, circuits here are 'paths' over IP.
 
Thanks,
Aparna

Aparna V. 
Level (3) Communications.


-----Original Message-----
From: Raja Badri [ mailto:raja.badri@santera.com
<mailto:raja.badri@santera.com> ]
Sent: Monday, February 14, 2000 11:04 AM
To: sip ML
Subject: SIP BCP-T question


Hi,
 
I have one question regarding the SIP BCP-T.
INFO message outlines the Mid-Call ISUP messages.
What is the reason behind not including the ISUP maintenance 
messages like Block, Unblock and Reset ?
 
Thanks,
Raja Badri
Santera Systems Inc.
 




From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 14 15:04:02 2000
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I have a couple of suggestions wrt. the serverfeatures draft:

  1. Define Supported in the base spec.
  2. Do *not* let presence of the Supported header implicitly
     signal support for the extension negotiation mechanism.
  3. Drop the reversed domain name extension naming convention.
  4. Rename the negotiation extension.
  5. Define "all-extensions-listed" tag to be used in Supported
     header.

I'll go through these in turns.

  1. Define Supported in the base spec.

The draft can be seen as having two parts: the definition of the
Supported header and a proposal for a feature negotiation mechanism. I
think the two should be separated. It makes perfectly good sense for a
server to signal supported features in an OPTIONs response without
having to implement the negotiation mechanism, and likewise, clients
should be allowed to indicate that they support, say, reliable 1xx's
without having to implement the negotiation mechanism.

So assuming the two should be split, the Supported header must be
defined elsewhere. I think it belongs in the base spec because it's the
direct equivalent to Unsupported which is in the base spec, it's
universally useful, and it's really simple enough to go in there.

  2. Do *not* let presence of the Supported header implicitly
     signal support for the extension negotiation mechanism.

This is a corollory of the previous argument. If UAs can list supported
features independent of the negotiation mechanism then presence of the
Supported header cannot be taken to implicitly indicate support for the
latter.

  3. Drop the reversed domain name feature naming convention.

It makes messages longer and is unnecessary.

The (primary) justifiction for the extension negotiation mechanism is
that servers need to know what features are supported by clients. It
seems to me that the simplest way to meet this requirement is for
clients to always tell servers what features they support (and BTW this
is another argument for defining Supported in the base spec).

Now, the reason why this answer is not enough must be the potentially
large overhead of listing all these extensions all the time. Hence SIZE
MATTERS! Anything we can do to minimize the size overhead of signaling
supported features makes it more practical for us to use the simpler and
more efficient mechanism of just listing all supported features always. 
The use of the short header form and the proposed implicit serverfeature
support is further testament to the need to keep messages short. There
may be times where it really isn't practical to list all supported
extensions, and for these cases we still need the negotiation.

For this reason, and *because it's unnecessary*, I propose we drop the
convention of using reversed domain name prefixes as namespace
identifiers for extension tags.  First, as I've mused on a previous
occasion, the vast majority (if not all) extensions to come into
widespread, regular use will belong to the org.ietf.sip namespace
anyway, making this prefix redundant, and secondly, I don't even think
we need these namespaces anyway. We don't have namespaces for header
names or status codes, and MIME has gotten along just fine for years
without namespaces for its media types (of which there will probably
always be many more than there will be SIP extensions). We could use the
"x-" prefix for experimental or proprietary extensions or continue to
use the domain namespaces for this purpose. 

  4. Rename the negotiation extension.

Follows from 1. and 4. Maybe just call the extension "negotiate".

  5. Define "all-extensions-listed" tag to be used in Supported
     header.

In order to avoid unnecessary round-trips it would be good if clients
can indicate that they have listed *all* supported extensions in the
Supported header. This can be done by defining a new token with that
meaning. I'd propose using "." since the semantics are kind of like that
of "."s use in natural written language (and tokens doesn't come much
shorter).  So, for example

  Supported: negotiate, 100rel, .

would indicate that the sender supports the 100rel and negotiate
extensions and *no* other extension. Hence there is no point for the
server to ask whether extension "foo" is supported - it isn't. This tag
can be seen as just identifying an extension of its own, and could be
defined in its own document, or it could go into the base spec. I'd
prefer the latter, since it seems like a generally very useful feature.

Of course the same mechanism can be used by servers.

So to summarize: the serverfeatures draft is a fine piece of work, but
it just makes sense to try and minimize the need for extra
"multiple-roundtrip" negotiation. Hence

  - UAs should be allowed to signal what features are supported
    without supporting negotiation
  - feature tags should be kept shortish
  - UAs should be able to say *all* features have been listed

My 2 pence worths.

Anders

-- 
Anders Kristensen <ak@hplb.hpl.hp.com>,
http://www-uk.hpl.hp.com/people/ak/
Hewlett-Packard Labs, Bristol, UK



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From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb 15 09:50:10 2000
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Date: Tue, 15 Feb 2000 09:45:30 -0500
From: Gautam Nair <gnair@cs.columbia.edu>
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A new version of the Internet-Draft - DHCP Option for SIP Servers - is
available.
The significant changes over the previous draft are:
The number of sub-options have been reduced to 2. 
The DNS A record sub-option has been removed since it was redundant.
The DNS SRV record has been effectively duplicated in the IP address
list sub-option.
Feedback and comments are appreciated.
Gautam Nair.

-------- Original Message --------
Subject: I-D ACTION:draft-nair-sip-dhcp-01.txt
Date: Tue, 15 Feb 2000 08:36:20 -0500
From: Internet-Drafts@ietf.org (by way of Ralph Droms
<droms@bucknell.edu>)
Reply-To: Internet-Drafts@ietf.org
To: DHCPv4 discussion list <dhcp-v4@bucknell.edu>

A New Internet-Draft is available from the on-line Internet-Drafts
directories.


	Title		: DHCP Option for SIP Servers
	Author(s)	: G. Nair, H. Schulzrinne
	Filename	: draft-nair-sip-dhcp-01.txt
	Pages		: 8
	Date		: 14-Feb-00
	
This document defines a DHCP option that contains one or more
pointers to one or more SIP servers . This enables a SIP client to
obtain the addresses of the SIP servers during bootup.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-nair-sip-dhcp-01.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-nair-sip-dhcp-01.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.
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Subject: Re: When to send 183?
To: dean.willis@wcom.com (Dean Willis)
Date: Tue, 15 Feb 2000 11:01:57 -0600 (CST)
Cc: ssubbara@cisco.com (Srinath Subbarayan), hgs@cs.columbia.edu,
        jdrosen@dynamicsoft.com, sip@lists.research.bell-labs.com
In-Reply-To: <001001bf751f$00325dc0$feba2ca6@dwillispc8> from "Dean Willis" at Feb 12, 2000 12:04:19 AM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>> Wouldn't a "410 The number you have reached has been disconnected"
>> message serve the same purpose for native clients? I sure would
>> prefer to read the text on a display than suffer through the
>> recording. I beleive this is a more appropriate approach if
>> you *know* for a fact that you're terminating on a native client.
>> If you don't know whether you're hitting a PSTN gateway, I'd
>> argue for playing the announcement.
>
>As I understand it, the SS7 release code is often quite vague, and certainly
>doesn't convey the "NEW NUMBER IS . . . " information contained in the voice
>recording.

Grab Q.850. On page Page 5, cause code 22 -- what is the information
conveyed in the diagnostic field? Regardless of how often this is used,
my argument is that, if it is present, it would be appropriate to return
it as a 410 message to the client instead of reading the announcement.

In the cases that the release code is too vague to determine this
information, you are absolutely correct: a 1xx early media message 
is the only available method for conveying the error information
to the user (unless you want to set up a full call, which may
have billing implications, depending on your network architecture).

>AS for 180 with ring-SDP -- how does the network gateway anticipate the ring
>treatment that will be applied by a domestic terminating switch? Can't, can
>it?
>
>I'd suggest that the easiest general solution is to send a 183 with media
>following the ACM. This gets the RTP stream established in time for network
>call treatment.

You're right. My mistake (for some reason, I was thinking that ACM had
event code as an optional parameter). I'd argue that this situation would 
be best handled by the 183 for early media, followed by an SDP-less 180 
when and if a CPG arrives with an event code of 1 (alerting) present.

It's far less elegant than I was hoping, but it works.

-- 
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: James Kempf <James.Kempf@Eng.Sun.COM>
Reply-To: James Kempf <James.Kempf@Eng.Sun.COM>
Subject: Discovering SIP Servers
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Attached is an individual submission on how to find SIP servers using
SLP, including a service template for SIP. The service template defines
the SLP attributes advertised by a SIP server for discovery. The
draft may be of interest to members of the group, which is why 
I've posted it.

As an aside, there has been some concern expressed on other lists about
the size of an SLP implementation, which might impact the usability
of SLP in small devices, such as phones. We have done a client side 
implementation of the minimal SLP UA specified by RFC 2608 in 5K of uncompressed 
Java bytecodes.

Please send any comments, questions, and criticisms


		jak

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Session Initiation Protocol                                James Kempf
INTERNET DRAFT                                             Sun Microsystems
Feburary 8, 2000                                           Jonathan Rosenberg
                                                           dynamicsoft


                        Finding a SIP Server With SLP
                       draft-kempf-sip-findsrv-00.txt


Status of This Memo

This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.  Internet-Drafts are work-
ing documents of the Internet Engineering Task Force (IETF), its
areas, and its working groups.  Note that other groups may also dis-
tribute working documents as Internet-Drafts.

Internet-Drafts are draft documents valid for a maximum of six
months and may be updated, replaced, or obsoleted by other documents
at any time.  It is inappropriate to use Internet-Drafts as refer-
ence material or to cite them other than as "work in progress."

The list of current Internet-Drafts can be accessed at:

http://www.ietf.org/ietf/1id-abstracts.txt

The list of Internet-Draft Shadow Directories can be accessed at:

http://www.ietf.org/shadow.html.

Abstract

The Session Initiation Protocol (SIP) allows a user to communicate
with a local SIP proxy and registrar, primarily for registration
services and the execution of certain features. Currently, SIP
allows for discovery of these servers through multicast or
through static configuration. In many cases, it may be more
useful for a SIP client to discover local SIP servers based
on features supported by those servers. In this document, an alternate
technique is specified based on Service Location Protocol (SLP).
The document contains a short description of how to use SLP for service
discovery, and a SIP service type template. The service type template
defines the attributes and service URL for a SIP server advertisement,
suitable for SIP clients to discover.

1.  Introduction




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The Session Initiation Protocol (SIP) [1] allows a user to communicate
with a local SIP proxy and registrar. The server provides registration
services and the execution of certain features. SIP allows for the
discovery of servers through multicast and static configuration. The
multicast mechanism allows a SIP client, or UAC, to send a REGISTER
message to a well-known SIP multicast address. A registrar that is wil-
ling to serve the user can then answer the request. Similarly, a UAC
can send an INVITE message via multicast in order to determine its
local outbound proxy.

This approach is limiting, however, if many SIP servers are distributed
throughout the domain. Different servers may have differing capabili-
ties. Some technique is required whereby a UAC can find a server that
supports the particular characteristics it needs.

In this document, we describe how to use Service Location Protocol
(SLP)[2] for configuring a UAC with a SIP proxy or registrar server.
SLP is a new protocol for dynamically discovering contact information
for a service. SLP is specifically designed for co-operatively managed
networks, and not for the Internet, so the techniques described in
RFC 2543 would still apply if the SIP client is required to directly
connect with a server on the Internet.

Since SLP supports querying for services based on attributes, using
SLP allows a SIP client to find a locally available SIP  server
meeting its requirements with regard to transport protocol and
other characteristics. Querying by characteristic reduces the
amount of network traffic involved in discovering a server, and the
amount of computation the client is required to perform before
starting the SIP session. Section 8 contains a service
type template [3] describing the attributes advertised by a SIP
server and the format of the service URL returned as a result of an
SLP query.

2.  Notation Conventions

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED",  "MAY", and "OPTIONAL" in
this document are to be interpreted as described in RFC 2119 [5].

3.  Terminology

We reproduce here some SLP terminology from RFC 2608 for readers
unfamilar with SLP.

        User Agent (SLP UA)- A process working on the user's behalf to
        establish contact with some service. The SLP UA retrieves service
        information from the Service Agents, using multicast, or



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        Directory Agents, using unicast.

        Service Agent(SA)- A process working on behalf of one or more
        services to advertise the services and their capabilites. The
        SA listens for multicast SLP UA requests, and registers its
        advertisements with DAs if any DAs are available.

        Directory Agent(DA)- A process which collects service
        advertisements.  A DA acts as a cache to consolidate service
        advertisements so a SLP UA can use unicast instead of multicast to
        obtain service advertisements.

        Scope - A set of services, typically making up a logical
        administrative group.

        Service Advertisement - A URL, attributes, and a lifetime
        (indicating how long the advertisement is valid), providing
        service access information and capabilities description for a
        particular service.

4.  Using SLP for SIP Server Discovery

SLP provides the framework in which a SIP client finds an appropri-
ate local  registrar or proxy/redirect server. Here is a description of
how an SIP client finds its  server using SLP:

        1. The SIP  server implements an SLP SA while the UAC
        implements an SLP UA.

        2. The SIP SA constructs a service advertisment consisting of a
        service URL, attributes and a lifetime. The URL has an
        appropriate service type, either "service:sip:proxy" if the
        server is a proxy/redirect server or "service:sip:registrar"
        if the server is a registrar server, and attributes defined
        according to the template in Section 8.

        3. When the SIP client requires contact information for a SIP
        server, the SLP UA either contacts the DA using unicast or the SA
        using multicast. The SLP UA includes a query based on the attributes
        to indicate the characteristics of the server it requires.

        4. Once the SLP UA has the contact information for the SIP server,
        the UAC can either send a registration (if the UAC is seeking a
        registrar server) or an INVITE (if the UAC is seeking a local
        outbound proxy/redirect server). The service URL returned from
        the SLP query includes the IP address or host name, and,
        optionally, the port number of the SIP  server. If the port
        number is not included, the SIP client uses the IANA-assigned



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        SIP port, 5060.

This procedure is exactly the same for any client/server pair imple-
menting SLP and is not specific to SIP.

SLP provides a number of advantages over SIP-specific multicast
and static configuration. The major advantage of SLP is that
clients can specify the attributes describing the desired server.
The UAC can find a server that supports exactly the features
it needs without requiring every server to support all SIP features.
With SLP, an enterprise can support a heterogeneous mix of
servers and UACs can find those that best suit their needs.
Additionally, SLP contains a number of scaling mechanisms (DAs,
scopes), that facilitate deployment in large enterprise networks, as
well as in smaller networks.

5.  Scalability

SLP contains a mechanism, called scopes, for grouping service
advertisements. For example, a network administrator could arrange
for all users in the Marketing Department to share access to a col-
lection of service advertisements by putting their SLP UAs, SAs, and
DAs into the same scope. Unless another deparment is in the same
scope, their SLP UAs, SAs and DAs would not have access to service
advertisements. Scopes don't control access to the services
themselves, however, just to the advertisements. Scope configuration
is not required, because SLP specifies a default scope that is used
if no other scope is available. Scopes are one important scalability
mechanism included in the SLP design.

The other scaling mechanism is DAs. In small/home office networks, SLP
UAs use multicast to contact SLP SAs. In larger, enterprise networks,
the network can be configured with DAs to cache advertisements
and reduce the amount of multicast traffic on the network. Again, DAs
are not a required part of SLP but can be configured if necessary.

Scope and DA information can be preconfigured for SLP agents by
including them in the DHCP option record for SLP [4]. This allows the
SLP agents to be configured without any local information. DHCP con-
figuration is not necessary, however. SLP UAs can determine their scope
and DA information dynamically, and DAs and SAs use the default
scope if no other scope is configured.

6. Finding the Right Proxy for an Incoming Call

Without SLP, SIP servers are typically organized into a hierarchy
according to the hierarchy of DNS domain names. For example, Acme, Inc.
has one top level domain and three second level domains, corresponding



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to the Sales, Marketing, and Engineering departments. The domain names
are:

        acme.com - Top level domain
        sales.acme.com - Sales department
        mkt.acme.com - Marketing department
        eng.acme.com - Engineering department

In a statically configured SIP system, each domain has one or
several proxy servers per domain. The names of these servers are
available from DNS, either through resolving a name directly or through
using the DNS SRV record. If multiple proxies are configured to support
a single domain (using DNS load balancing techniques), all of those
servers contain the registration for a user. This is accomplished
through some back end replication protocol, or through SIP multicast
registrations.

A call coming through the proxy server for the top level domain is
resolved by determining in which domain the called party is
located. The request is proxied to any proxy server in the called
party's domain. In our previous example, if a call comes in for Joe
User in the Engineering department, the Acme top level proxy can
easily determine which which proxy to use by looking up the user in
some corporate directory, as in the following figure:


                        |
                        | joe.user@acme.com
                        |
                        |
                        v
                +----------------+        +------------------+
                |                |        |                  |
                | acme.com proxy | <----> | Backend Database |
                |                |        |                  |
                +----------------+        +------------------+
                             ^
                             |
                             v
  +--------------+   +--------------+   +----------------+
  |              |   |              |   |                |
  | mkt.acme.com |   | eng.acme.com |   | sales.acme.com |
  |  registrar   |   |  registrar   |   |   registrar    |
  |              |   |              |   |                |
  +--------------+   |     joe.user |   +----------------+
                     +--------------+





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If SLP is in use, a SIP system can be configured so that different
registrar servers in the same domain support different capabilities.
Thus, from the example, the Engineering domain might contain one
registrar server that supports CPL [6] processing and another that
supports CGI processing.

A UAC can use SLP to discover a SIP registrar server supporting
a particular set of capabilities that it needs. For example, suppose
that Joe User wants to upload a CPL program to his SIP registrar.
Joe can specify that his UAC use SLP to find a registrar that supports
CPL [7] and register with it. As a consequence, the registration will
not be available to other registrars/proxies.

From the previous figure, the proxy server for acme.com can't simply
use a static back end database to determine the next hop proxy,
because a dynamically chosen one has a registration for the user:


                        |
                        | joe.user@acme.com
                        |
                        |
                        v
                +----------------+
                |                |
                | acme.com proxy |
                |                |
                +----------------+

                ?????

  +------------------+   +------------------+
  |                  |   |                  |
  | cgi.eng.acme.com |   | cpl.eng.acme.com |
  |     registrar    |   |     registrar    |
  |                  |   |                  |
  +------------------+   |         joe.user |
                         +------------------+



There are two ways this problem can be handled:

        1. The top level proxy server can use SIP "forking"
        to try more than one next hop proxy server at once.
        Whichever proxy has a registration for the user will
        redirect or proxy there, and all others will return a 404
        Not Found. This approach has the drawback of requiring the



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        top level proxy to have a list of all possible proxies the user
        may have registered at. It could discover these using SLP (see
        Section 7 on non-SIP-UAs as SLP UAs), or be configured with
        them.

        2. The top level proxy server multicasts the invitation
        to the SIP multicast address. The registrar with the
        registration will proxy the call to the user. All others
        will not respond. This approach has the advantage of requiring
        less configuration.


7. SIP Servers as SLP UAs

Although SLP is primarily of interest by UACs for finding servers
having particular characteristics, it may also be used by SIP
servers for finding other SIP servers. As an example, in the previous
section, the top level acme.com SIP proxy server could use SLP to
locate the registrar servers in the eng domain. Although this does
not help in finding which registrar server has the registration,
it does allow the network of SIP servers to autoconfigure, in the
event a server goes down or a new one is added, without requiring
DNS updates or changing static configurations.

In general, a SIP server locating other SIP servers does
not include particular attributes in its query, other than the
"transport" attribute that is required in every query. And,
since SLP is only of use on cooperatively managed networks,
a SIP proxy server in one top level domain can't use SLP over
the Internet to locate the proxy in another. But a server MAY
include a query if it requires that the contacted servers
support particular characteristics. An example here is IP-level
security.

8.  The SIP Service Type Template

SIP defines two different kinds of servers: registrars and proxy/
redirect servers. The features of both are different enough that
separate service type templates are required. We define an abstract
service  type [3], with a concrete service type for the two
different kinds of servers.

8.1 SIP Abstract Service Type

Name of submitters: James Kempf <james.kempf@sun.com>
                    Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Language of service template: en
Security Considerations:



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SIP clients can use Service Location Protocol to find SIP servers
having particular security characteristics. If secure access to
such information is required, SLP security should be used.

Template text:

----------------------template begins here ------------------------

template-type = sip

template-version = 0.0

template-description=
The abstract service:sip type provides advertisements for clients
seeing Session Initiation Protocol (SIP) service. Concrete types
specialize the service:sip type for proxy/redirect and
registrar servers.

template-url-syntax= transport-attr-list
transport-attr-list = ';transport=udp' / ';transport=tcp' /
                      ';transport=udp,tcp'
;The SIP service URL includes the transport protocol or
;protocols that the server supports. While the client must include
;the protocol type in the query and therefore does not need this
;information, the transport protocol attribute is included in
;the URL in case the client passes the URL along to a third
;party, for example, by way of a SIP Request-URI.
;An example SIP service URL (for a proxy server) is:
;      service:sip:proxy://telsrv:2021;transport=udp.
;

transport = STRING X L M
udp
#Transport protocol used to contact the server. Servers can support
#UDP, TCP, or both. This attribute is REQUIRED in every query.
udp,tcp

domain = STRING M L O
#The name of the domain represented by the server.

ip-security-mechanisms = STRING M L O
#IP-level security mechanisms supported by the server.
#Allowed values are:
#
# ipsec - The server supports IP level security (IPSEC).
# tls - The server supports transport level security (TLS).
#
ipsec,tls



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sip-security-mechanisms = STRING M L O
#SIP-level security mechanisms supported by the server.
#Allowed values are:
#
# basic - The server supports basic authentication.
# digest - The server supports digest authentication.
# pgp - The server supports PGP authentication.
#
basic,digest,pgp

options = STRING M L O
#A collection of SIP option tags supported by the server. The tags
#are appropriate for inclusion in the REQUIRE or OPTIONS
#SIP request.

extensions = STRING M L O
#A list of extensions supported by the server. UACs use
#this attribute to identify whether a server supports
#a particular extension.

----------------------template ends here --------------------------

8.2 SIP Proxy/Redirect Concrete Service Type

Name of submitters: James Kempf <james.kempf@sun.com>
                    Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Language of service template: en
Security Considerations:
SIP clients can use Service Location Protocol to find SIP proxy/redirect
servers having particular security characteristics. If secure access to
such information is required, SLP security should be used.

Template text:

----------------------template begins here ------------------------

template-type = sip:proxy

template-version = 0.0

template-description=
The service:sip:proxy type provides advertisements for clients
seeking Session Initiation Protocol (SIP) proxy/redirect servers.

template-url-syntax = ;inherited from abstract type.

features = STRING M L O
#A list of features supported by the server. Allowed values are:



Kempf, Rosenberg          expires August 2000                   [Page 9]

INTERNET DRAFT                                             Feburary 2000


#
# recursion - The server can recursively try addresses returned to
#             it in a redirect request.
# forking - The proxy may fork off multiple requests in response to
#           a client request.
# stateful - The proxy remembers incoming request that generated
#            outgoing requests and the corresponding outgoing
#            requests.
#
recursion,forking,stateful

caller-preferences = BOOLEAN O
#True if the server supports caller preferences.

handles-e.164-addresses = BOOLEAN O
#True if the server handles E.164 (telephone number)
#addresses. An example is:
#       tel:5551234

----------------------template ends here --------------------------

8.3 SIP Registrar Concrete Service Type

Name of submitters: James Kempf <james.kempf@sun.com>
                    Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Language of service template: en
Security Considerations:
SIP clients can use Service Location Protocol to find SIP registrar
servers having particular security characteristics. If secure access to
such information is required, SLP security should be used.

Template text:

----------------------template begins here ------------------------

template-type = sip:registrar

template-version = 0.0

template-description=
The service:sip:registrar type provides advertisements for clients
seeking Session Initiation Protocol (SIP) registrar servers.

template-url-syntax = ;inherited from abstract type.

app-services = STRING M L O
#Application services supported by the server. Allowed values are:
#



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# cpl - The server supports Call Processing Language (CPL) upload.
# cgi - The server supports CGI upload.
#
cpl,cgi

non-sip-urls = BOOLEAN O
#True if the server supports nonSIP URLs in Contacts.

----------------------template ends here --------------------------

9.  An Example SLP Query

From the example in Section 8, suppose Joe User's UAC is looking
for a registrar server that supports CPL. The following SLP query
can be used by the UAC to find a registrar server that supports
CPL:

        (&(transport=udp)(app-services=cpl))

The transport type is required in all queries (indicated by the 'X'
flag in the template). In this case, the UAC requires UDP.

10.  Security Considerations

Service type templates provide information that is used to inter-
pret information obtained by clients through SLP.  If the SIP tem-
plate is modified or if a false template is distributed, SIP servers
may not correctly register themselves, and SIP clients may not be
able to interpret service information.

SLP provides an authentication mechanism for SLP UAs to assure that ser-
vice advertisments only come from trusted SAs [2]. If trust is an
issue, particularly with respect to the information  sought by the
client about security mechanisms, then SLP authentication should be
enabled in the network.

11.  Summary

This document describes how to use SLP to discover a SIP server. SLP
allows the SIP client to specify characteristics of the SIP server
that it requires, such as transport protocol type, SIP services sup-
plied, application services supplied, etc. A service type template
is defined that contains the attributes advertised by SIP servers.

12.  Full Copyright Statement

Copyright (C) The Internet Society (1997).  All Rights Reserved.




Kempf, Rosenberg          expires August 2000                  [Page 11]

INTERNET DRAFT                                             Feburary 2000


This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph
are included on all such copies and derivative works.  However, this
document itself may not be modified in any way, such as by removing
the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of devel-
oping Internet standards in which case the procedures for copy-
rights defined in the Internet Standards process must be followed,
or as required to translate it into languages other than English.

The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns.

This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE."

13.  References

[1] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg."SIP:
Session Initiation Protocol".RFC 2543, March, 1999.

[2] E. Guttman, C. Perkins, J. Veizades, and M. Day. "Service Loca-
tion Protocol". RFC 2608, July, 1999.

[3] E. Guttman, C. Perkins and J. Kempf. "Service Templates and ser-
vice: Schemes", RFC 2609, July, 1999.

[4] C. Perkins and E. Guttman. "DHCP Options for Service Location
Protocol". RFC 2610, July, 1999.

[5] S. Bradner. "Key words for use in RFCs to indicate requirement
levels", BCP 14,  RFC 2119, March 1997.

[6] J. Lennox and H. Schulzrinne. "Call Processing Language Framework
and Requirements". draft-ietf-iptel-cpl-framework-01.txt. A work
in progress.

14.  Author's Addresses

James Kempf                                     Jonathan Rosenberg
Sun Microsystems                                dynamicsoft



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INTERNET DRAFT                                             Feburary 2000


901 San Antonio Rd.                             200 Executive Drive
UMPK15-214                                      Suite 120
Palo Alto, CA, 94040                            West Orange, NJ, 07052
USA             USA
+1 650 786 5890                                 +1 732 741 7244
+1 650 786 6445(fax)                            +1 732 741 4778(fax)
james.kempf@sun.com                             jdrosen@dynamicsoft.com












































Kempf, Rosenberg          expires August 2000                  [Page 13]


--Gang_of_Elks_029_000--



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb 15 15:27:57 2000
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Date: Tue, 15 Feb 2000 15:16:49 -0500 (EST)
From: Francois Menard List Account <fm-listproc@fmmo.ca>
To: James Kempf <James.Kempf@Eng.Sun.COM>
Cc: sip@lists.research.bell-labs.com, etremblay@mediatrix.com
Subject: Re: Discovering SIP Servers
In-Reply-To: <200002151920.LAA11743@nasnfs.eng.sun.com>
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Excellent!

I welcome this initiative.

Will Sun release the source code of the minimal UA implementation ?

-=Francois=-

On Tue, 15 Feb 2000, James Kempf wrote:

> Attached is an individual submission on how to find SIP servers using
> SLP, including a service template for SIP. The service template defines
> the SLP attributes advertised by a SIP server for discovery. The
> draft may be of interest to members of the group, which is why 
> I've posted it.
> 
> As an aside, there has been some concern expressed on other lists about
> the size of an SLP implementation, which might impact the usability
> of SLP in small devices, such as phones. We have done a client side 
> implementation of the minimal SLP UA specified by RFC 2608 in 5K of uncompressed 
> Java bytecodes.
> 
> Please send any comments, questions, and criticisms
> 
> 
> 		jak
> 




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Date: Tue, 15 Feb 2000 12:35:09 -0800 (PST)
From: James Kempf <James.Kempf@eng.sun.com>
Reply-To: James Kempf <James.Kempf@eng.sun.com>
Subject: Re: Discovering SIP Servers
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We're working on it.

		jak


>Date: Tue, 15 Feb 2000 15:16:49 -0500 (EST)
>From: Francois Menard List Account <fm-listproc@fmmo.ca>
>To: James Kempf <James.Kempf@eng.sun.com>
>cc: sip@lists.research.bell-labs.com, etremblay@mediatrix.com
>Subject: Re: Discovering SIP Servers
>MIME-Version: 1.0
>
>
>Excellent!
>
>I welcome this initiative.
>
>Will Sun release the source code of the minimal UA implementation ?
>
>-=Francois=-
>
>On Tue, 15 Feb 2000, James Kempf wrote:
>
>> Attached is an individual submission on how to find SIP servers using
>> SLP, including a service template for SIP. The service template defines
>> the SLP attributes advertised by a SIP server for discovery. The
>> draft may be of interest to members of the group, which is why 
>> I've posted it.
>> 
>> As an aside, there has been some concern expressed on other lists about
>> the size of an SLP implementation, which might impact the usability
>> of SLP in small devices, such as phones. We have done a client side 
>> implementation of the minimal SLP UA specified by RFC 2608 in 5K of 
uncompressed 
>> Java bytecodes.
>> 
>> Please send any comments, questions, and criticisms
>> 
>> 
>> 		jak
>> 
>




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Subject: RE: When to send 183?
Date: Tue, 15 Feb 2000 16:51:48 -0600
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From: Dean Willis [mailto:dean.willis@wcom.com]
>>
As I understand it, the SS7 release code is often quite vague, and certainly
doesn't convey the "NEW NUMBER IS . . . " information contained in the voice
recording.
<<

It looks to me like the ACM message gets cause code 22, and the diagnostic
field may *optionally* contain the new number.  Also, it is not required
that cause code 22 be supported, in which case cause code 1 is used instead.
So, it could work well, but almost certainly wouldn't consistently work
well.

Tim Schroeder



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Date: Tue, 15 Feb 2000 18:47:40 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: When to send 183?
In-reply-to: <38A47BD6.235031C9@pingtel.com>
To: "Daniel G. Petrie" <dpetrie@pingtel.com>
Cc: "Adam B. Roach" <Adam.Roach@Ericsson.com>,
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My argument is that you MUST send a 18x with early media immediately after
receiving an ACM in order to have a media path established JUST IN CASE
there is an unsignaled in-band call treatment. Every time, all the time, in
a PSTN gateway.

I believe this should be a 183 rather than a 180 due to the semantic
difference -- a 180 means that something is being alerted, whereas a 183
just means that something is happening. This distinction allows UACes and
proxies to be a little smarter about how the use the 18x message.

Where some sort of release code can be determined, I firmly concur that the
call should be appropriately finalized.

--
Dean

> -----Original Message-----
> From: Daniel G. Petrie [mailto:dpetrie@pingtel.com]
> Sent: Friday, February 11, 2000 3:15 PM
> To: Dean Willis
> Cc: Adam B. Roach; Srinath Subbarayan; hgs@cs.columbia.edu;
> jdrosen@dynamicsoft.com; sip@lists.research.bell-labs.com
> Subject: Re: When to send 183?
>
>
> It seems that your argument is for early media, not necessarily
> for a 183 status.  The point I have been trying to make is
> that the status 183 is too generic if the outcome is known
> to be a call setup failure. One should avoid using the status
> 183 if a more descriptive status can be provided.
>
> Early media makes sense with many different status.  I think
> that we need to be very careful in keeping these issues
> separate.
>




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Date: Tue, 15 Feb 2000 18:51:17 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: Question about media desinations
In-reply-to: <38A6E5B6.F0C3E412@dynamicsoft.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: IETF SIP <sip@lists.research.bell-labs.com>
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Yes. Having both endpoints allows the firewall to construct a far more
effective dynamic ACL. It also allows UAs to selectively bind their RTP
streams to a specific partner, rather than waiting open-eared for a download
of spam.

--
Dean

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Sunday, February 13, 2000 11:11 AM
> To: Dean Willis
> Cc: 'SIP List'
> Subject: Re: Question about media desinations
>
>
> This raises an important point. I don't want to make changes to SDP and
> its usage within SIP to fix a problem that can be prevented by an
> administrative change in firewall behavior. If the firewalls did not
> return these error messages, do you still believe including the source
> port and address in the SDP is useful?
>
> -Jonathan R.
>
> Dean Willis wrote:
> >
> > Try scanning the net at 63.81.221.0/24, for example by
> telneting to port 122
> > on 63.81.221.234 -- you'll get back lots of "packet filtered"
> messages. The
> > default behavior in most systems is to return such a message.
> Some firewall
> > implementors turn them off. Most don't.
> >
> > --
> > Dean
> >
> > Jonathan asked:
> > > This presumes a firewall sends an ICMP error message when it discards
> > > the packet. I would
> > > think they would blackhole them, which will eliminate the ability to
> > > conduct this attack
> > > you describe.
>
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
>




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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		: The SIP INFO Method
	Author(s)	: S. Donovan
	Filename	: draft-ietf-sip-info-method-02.txt
	Pages		: 10
	Date		: 15-Feb-00
	
This document proposes an extension to the Session Initiation
Protocol (SIP).  This extension adds the INFO method to the SIP
protocol.  The intent of the INFO method is to allow for the carrying
of session related control information that is generated during a
session.  One example of such session control information is ISUP and
ISDN signaling messages used to control telephony call services.
This and other example uses of the INFO method may be standardized in
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From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb 16 14:26:19 2000
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Date: Wed, 16 Feb 2000 13:18:07 -0600
From: Robert Sparks <Robert.Sparks@wcom.com>
Subject: RE: UAC use of tags received in responses
In-reply-to: <38A33790.5A052AC6@dynamicsoft.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: "''SIP List' (E-mail)'" <sip@lists.research.bell-labs.com>,
        "'Henning Schulzrinne (E-mail)'" <schulzrinne@cs.columbia.edu>
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Since there have been no other comments, the suggestion now
is to add the following to '11.3 Caller Receives Response to 
Initial Request':

   A UAC MUST copy the tag from a final response into the ACK of that
   response. The UAC MUST include that tag into all other subsequent
   requests if, and only if, the response was in the 200 class. The
   UAC MUST NOT include that tag in any subsequent requests beyond
   the ACK if the response was not in the 200 class.

and changing the sentence in Section 6.41 To:
from 
   All subsequent transactions between two entities MUST include
   the "tag" parameter, as described in Section 11.
to
   Section 11 details when the "tag" parameter MUST appear in subsequent 
   requests.

RjS

> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Jonathan
> Rosenberg
> Sent: Thursday, February 10, 2000 4:11 PM
> To: Robert.Sparks@wcom.com
> Cc: 'SIP List' (E-mail); Henning Schulzrinne (E-mail)
> Subject: Re: UAC use of tags received in responses
> 
> 
> 
> 
> Robert Sparks wrote:
> > 
> > One of the two requirements above must be modified to allow for this
> > scenario. Changing the requirement on the proxy, having it 
> not add tags
> > for some or all locally generated responses, leads to 
> problems determining
> > the appropriate destination for the subsequent ACK (pointed 
> out by Jonathan
> > Rosenberg). The better solution then (suggested by both 
> Jonathan and Alan
> > Johnston) is to modify the behavior of the UAC such that:
> > 
> >    A UAC only copies a tag into subsequent requests if it 
> arrives in a
> >    200 OK to an INVITE.
> > 
> 
> I would further clarify this:
> 
>      A UAC copies the tag from the response into the ACK, but it MUST
> NOT
>      copy the tag into any subsequent requests unless the response was
>      a 200 class response to an INVITE.
> 
> Since ACK is a "subsequent" request after all.
> 
> -Jonathan R.
> 
> -- 
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120 
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb 16 15:11:59 2000
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Robert Sparks wrote:
> 
> Since there have been no other comments, the suggestion now
> is to add the following to '11.3 Caller Receives Response to
> Initial Request':
>
> <...>
> The UAC MUST include that tag into all other subsequent
>    requests if, and only if, the response was in the 200 class. 

CANCEL is an exception from that rule. People also argued for allowing
BYE requests with no tag (though I am not very happy with the idea).

---
Igor Slepchin



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb 16 18:20:04 2000
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Date: Wed, 16 Feb 2000 17:12:47 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: 183 vs 180, When and Where, a Proposal
In-reply-to: <4.2.0.58.20000216105725.00a1b5c0@farley.cisco.com>
To: IETF SIP <sip@lists.research.bell-labs.com>
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Taichi Fu and I have been having a discussion on 18x provisional messages
with media that may be worth sharing. Here's what I think we agreed to.

For PSTN interworking, you really have to "listen" to the bearer channel to
figure out what's going on (see previous discussion on release codes). Note
that this is really quite different from a pure-IP scenario.

If we don't have PSTN interworking, then a 180 means that a user is being
alerted, and media on a 180 clearly indicates a custom message that can be
played to the caller. For example, a French user might want to play back the
distinctive French double-ring, and a Trekky might want to
play a clip of Lt. Uhuru saying "Captain, there is an incoming call".  We
might also want to include a text message for the audio-impaired device or
user, or present a visual notification such as a corporate logo.

On the other hand, many UASes (and proxies) MAY choose to ignore the media
on a 180. This can avoid very complex mixing scenarios in a forking
environment.

Devices MAY also choose to ignore the media on a 183, but this MUST be done
very carefully. Media on a 183 for PSTN interworking is far more critical
than that on a 180 -- it may provide the only useful information available
to the caller for determining the cause of a call failure. Think very
carefully before ignoring a 183.

We suggest that a straight IP system, especially a UAS, SHOULD send 180
(which MAY have media) when alerting a user. A PSTN gateway SHOULD send a
183 with media when "launching" a call onto the PSTN. Devices MAY decide to
ignore the media of a 180. Devices SHOULD present the media of a 183 to an
end user, and MUST deal with the results of not doing so.

A key point:

Some have argued that 183 also has uses outside of PSTN interworking,
whenever there is some status information describing a call which needs to
be presented "in band", but there has been no actual notification delivered
to a destination.

For example, consider a proxy (P) that implements a forking search. An
INVITE for "Joe" is delivered to P. P might, in parallel:
	1) respond with a 183 containing media in two parts.
		part 1 is an html object with
		text="Your call is being forwarded by Joe's Finder Service".
		part 2 is an SDP object describing a media stream with a spoken
		rendition of the same text
	2) send INVITES to all of the contacts Joe has on his contact list.

This is probably a BAD usage of 183 -- the status information, while useful
and entertaining, is not critical to the successful interworking of the
call. Its presence could obfuscate an upstream forking system, and would
provide no additional value to that system. This example SHOULD be
implemented using a 180, not a 183.

--
Dean




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Date: Wed, 16 Feb 2000 18:59:33 -0800
To: Dean Willis <dean.willis@wcom.com>
From: Taichi Fu <tfu@cisco.com>
Subject: Re: 183 vs 180, When and Where, a Proposal
Cc: IETF SIP <sip@lists.research.bell-labs.com>
In-Reply-To: <000001bf78d3$56db2be0$ad9423a6@mcit.com>
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Dean,

Thanks for the summary. I add a small note below the paragraph 6
from my understanding.

Regards,
-Taichi

At 05:12 PM 2/16/00 -0600, Dean Willis wrote:

>Taichi Fu and I have been having a discussion on 18x provisional messages
>with media that may be worth sharing. Here's what I think we agreed to.
>
>For PSTN interworking, you really have to "listen" to the bearer channel to
>figure out what's going on (see previous discussion on release codes). Note
>that this is really quite different from a pure-IP scenario.
>
>If we don't have PSTN interworking, then a 180 means that a user is being
>alerted, and media on a 180 clearly indicates a custom message that can be
>played to the caller. For example, a French user might want to play back the
>distinctive French double-ring, and a Trekky might want to
>play a clip of Lt. Uhuru saying "Captain, there is an incoming call".  We
>might also want to include a text message for the audio-impaired device or
>user, or present a visual notification such as a corporate logo.
>
>On the other hand, many UASes (and proxies) MAY choose to ignore the media
>on a 180. This can avoid very complex mixing scenarios in a forking
>environment.
>
>Devices MAY also choose to ignore the media on a 183, but this MUST be done
>very carefully. Media on a 183 for PSTN interworking is far more critical
>than that on a 180 -- it may provide the only useful information available
>to the caller for determining the cause of a call failure. Think very
>carefully before ignoring a 183.
>
>We suggest that a straight IP system, especially a UAS, SHOULD send 180
>(which MAY have media) when alerting a user. A PSTN gateway SHOULD send a
>183 with media when "launching" a call onto the PSTN. Devices MAY decide to
>ignore the media of a 180. Devices SHOULD present the media of a 183 to an
>end user, and MUST deal with the results of not doing so.

That means, for SIP and PSTN interworking (a PSTN gateway converting
between SIP messages and SS7 messages):

(1) For a call from SIP to PSTN, a PSTN gateway SHOULD respond with 183 , and
(2) For a call from PSTN to SIP, a PSTN gateway SHOULD expect to receive 180.

An asymmetry.

>A key point:
>
>Some have argued that 183 also has uses outside of PSTN interworking,
>whenever there is some status information describing a call which needs to
>be presented "in band", but there has been no actual notification delivered
>to a destination.
>
>For example, consider a proxy (P) that implements a forking search. An
>INVITE for "Joe" is delivered to P. P might, in parallel:
>         1) respond with a 183 containing media in two parts.
>                 part 1 is an html object with
>                 text="Your call is being forwarded by Joe's Finder Service".
>                 part 2 is an SDP object describing a media stream with a 
> spoken
>                 rendition of the same text
>         2) send INVITES to all of the contacts Joe has on his contact list.
>
>This is probably a BAD usage of 183 -- the status information, while useful
>and entertaining, is not critical to the successful interworking of the
>call. Its presence could obfuscate an upstream forking system, and would
>provide no additional value to that system. This example SHOULD be
>implemented using a 180, not a 183.
>
>--
>Dean
>




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Igor Slepchin wrote:
> 
> Robert Sparks wrote:
> >
> > Since there have been no other comments, the suggestion now
> > is to add the following to '11.3 Caller Receives Response to
> > Initial Request':
> >
> > <...>
> > The UAC MUST include that tag into all other subsequent
> >    requests if, and only if, the response was in the 200 class.
> 
> CANCEL is an exception from that rule. 

In theory, yes. However, the copy-the-tag-into-subsequent-requests rule
applies only after a 200 OK to INVITE has come. At that point, sending a
CANCEL is guaranteed to do nothing. So, there is no point in sending
one. If you did, you are right it should not contain a tag.

> People also argued for allowing
> BYE requests with no tag (though I am not very happy with the idea).

The same thing is true here with BYE. The issue is that people will be
sending BYE before the 200 OK (and thus the tag) has been established. 

Igor has a good point, though, that we should clarify that obviously
tags cannot be inserted into requests sent before the 200 OK to INVITE
comes.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Subject: Re: 183 vs 180, When and Where, a Proposal
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I'm not sure I like making this kind of distinction. Why not simply say
that a UAC always pay attention to media it is asked to receive via a
provisional response, and that pure IP UASs are advised not to send SDP
in provisional responses, but rather encode the data in the response
itself, via reason phrases or content. 

-Jonathan R.

Dean Willis wrote:
> 
> Taichi Fu and I have been having a discussion on 18x provisional messages
> with media that may be worth sharing. Here's what I think we agreed to.
> 
> For PSTN interworking, you really have to "listen" to the bearer channel to
> figure out what's going on (see previous discussion on release codes). Note
> that this is really quite different from a pure-IP scenario.
> 
> If we don't have PSTN interworking, then a 180 means that a user is being
> alerted, and media on a 180 clearly indicates a custom message that can be
> played to the caller. For example, a French user might want to play back the
> distinctive French double-ring, and a Trekky might want to
> play a clip of Lt. Uhuru saying "Captain, there is an incoming call".  We
> might also want to include a text message for the audio-impaired device or
> user, or present a visual notification such as a corporate logo.
> 
> On the other hand, many UASes (and proxies) MAY choose to ignore the media
> on a 180. This can avoid very complex mixing scenarios in a forking
> environment.
> 
> Devices MAY also choose to ignore the media on a 183, but this MUST be done
> very carefully. Media on a 183 for PSTN interworking is far more critical
> than that on a 180 -- it may provide the only useful information available
> to the caller for determining the cause of a call failure. Think very
> carefully before ignoring a 183.
> 
> We suggest that a straight IP system, especially a UAS, SHOULD send 180
> (which MAY have media) when alerting a user. A PSTN gateway SHOULD send a
> 183 with media when "launching" a call onto the PSTN. Devices MAY decide to
> ignore the media of a 180. Devices SHOULD present the media of a 183 to an
> end user, and MUST deal with the results of not doing so.
> 
> A key point:
> 
> Some have argued that 183 also has uses outside of PSTN interworking,
> whenever there is some status information describing a call which needs to
> be presented "in band", but there has been no actual notification delivered
> to a destination.
> 
> For example, consider a proxy (P) that implements a forking search. An
> INVITE for "Joe" is delivered to P. P might, in parallel:
>         1) respond with a 183 containing media in two parts.
>                 part 1 is an html object with
>                 text="Your call is being forwarded by Joe's Finder Service".
>                 part 2 is an SDP object describing a media stream with a spoken
>                 rendition of the same text
>         2) send INVITES to all of the contacts Joe has on his contact list.
> 
> This is probably a BAD usage of 183 -- the status information, while useful
> and entertaining, is not critical to the successful interworking of the
> call. Its presence could obfuscate an upstream forking system, and would
> provide no additional value to that system. This example SHOULD be
> implemented using a 180, not a 183.
> 
> --
> Dean

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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>> subject: "Check this"
>> body: "Have fun with these links. Bye."
>> attachment: "LINKS.VBS"
>>
>> Do not open the attachment! Delete it immediatly and DO NOT open it! It
>> contains a visual basic script that sends all your friends (at your
>address
>> book) the same email for them to open. It might also install a virus on
>your
>> computer.
>>
>>
>
>
>






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Jonathan Rosenberg wrote:
> 
> I'm not sure I like making this kind of distinction. Why not simply say
> that a UAC always pay attention to media it is asked to receive via a
> provisional response, and that pure IP UASs are advised not to send SDP
> in provisional responses, but rather encode the data in the response
> itself, via reason phrases or content.

Beyond this, I still have two major concerns/requirements:

- The caller needs to be able to express that it doesn't want audio.
Yes, it can ignore it, but the traffic may interfere with other on-going
communications.

- If this is an error condition, i.e., no call can ensue, when is the
UAS going to send a 4xx or 5xx? There is no right time, since it either
prematurely shuts off the explanation or it delays sequential searches
in proxies for very long time periods.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 17 11:44:05 2000
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Subject: Re: 183 vs 180, When and Where, a Proposal
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Henning Schulzrinne wrote:
> 
> Jonathan Rosenberg wrote:
> >
> > I'm not sure I like making this kind of distinction. Why not simply say
> > that a UAC always pay attention to media it is asked to receive via a
> > provisional response, and that pure IP UASs are advised not to send SDP
> > in provisional responses, but rather encode the data in the response
> > itself, via reason phrases or content.
> 
> Beyond this, I still have two major concerns/requirements:
> 
> - The caller needs to be able to express that it doesn't want audio.
> Yes, it can ignore it, but the traffic may interfere with other on-going
> communications.

I'm not sure this is a NEED so much as something that might be a nice
feature. Actually
this seems something appropriate for caller preferences.


> 
> - If this is an error condition, i.e., no call can ensue, when is the
> UAS going to send a 4xx or 5xx? There is no right time, since it either
> prematurely shuts off the explanation or it delays sequential searches
> in proxies for very long time periods.

My guess is that in practice, there will seldom be forking with gateways
involved. As such, reasonable behavior might be to send the audio,
potentially delaying the unlikely sequential search.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Jonathan Rosenberg wrote:
> 
> Henning Schulzrinne wrote:
> >
> > Jonathan Rosenberg wrote:
> > >
> > > I'm not sure I like making this kind of distinction. Why not simply say
> > > that a UAC always pay attention to media it is asked to receive via a
> > > provisional response, and that pure IP UASs are advised not to send SDP
> > > in provisional responses, but rather encode the data in the response
> > > itself, via reason phrases or content.
> >
> > Beyond this, I still have two major concerns/requirements:
> >
> > - The caller needs to be able to express that it doesn't want audio.
> > Yes, it can ignore it, but the traffic may interfere with other on-going
> > communications.
> 
> I'm not sure this is a NEED so much as something that might be a nice
> feature. Actually
> this seems something appropriate for caller preferences.

Agreed, or one can argue that it is a feature, described via Supported,
since not all end systems will even be able to accept voice ahead of 200
(e.g., they may only start the media tools after the 200 has been
received).

> 
> >
> > - If this is an error condition, i.e., no call can ensue, when is the
> > UAS going to send a 4xx or 5xx? There is no right time, since it either
> > prematurely shuts off the explanation or it delays sequential searches
> > in proxies for very long time periods.
> 
> My guess is that in practice, there will seldom be forking with gateways
> involved. As such, reasonable behavior might be to send the audio,
> potentially delaying the unlikely sequential search.

I'm not sure that's true. I get the voice announcement if a cell phone
(Sprint PCS) is not answering. If I route to both a home phone and cell
number, I don't want the announcement to delay ringing the other phone.

> 

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



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References: <003201bf78b2$8f59c4c0$8a9223a6@mcit.com> <38AB0319.B3E2AF52@dynamicsoft.com> <38AB99A8.D33E74FB@dynamicsoft.com> <38AB9F58.6889F1B7@dynamicsoft.com>
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Igor Slepchin wrote:
> 
> > > CANCEL is an exception from that rule.
> >
> > In theory, yes. However, the copy-the-tag-into-subsequent-requests rule
> > applies only after a 200 OK to INVITE has come. At that point, sending a
> > CANCEL is guaranteed to do nothing. So, there is no point in sending
> > one. If you did, you are right it should not contain a tag.
> 
> One could theoretically CANCEL any request. A more practical example is
> a re-INVITE. Besides, UAC might want to send a CANCEL even after it got
> a 200 OK to cancel the search on forked branches that might still be
> outstanding.

Fair enough. However, I think that in the case of cancelling a
re-INVITE,
you do actually want the tag in the To field. I believe the CANCEL needs
to
match the request its cancelling, in terms of the To, From, Call-ID and
CSeq,
including tags. In the case of multiple call legs, if a re-INVITE were
sent on
each call leg, and then cancelled on each call leg, you might not be
able to distinguish 
the CANCELs except for the tag in the To field.

So, if this is OK, the proposed addition is:

"The exception to this is CANCEL. The CANCEL always contains exactly the
same value of the To field as the request it is cancelling."


-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 17 14:10:04 2000
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Hmmm. Actually, placing the source address in the SDP solves another
problem. When multiple 200 OK (or 183 or whatever) arrive with SDP, the
caller is going to receive audio all on the same port, with no way to
figure out which RTP stream corresponds to which call leg. By adding the
source address, the UAC will be able to correlate them.

I guess the next step is to write up the SDP extension as a small draft
and submit it to mmusic, as its really an mmusic issue and not a sip
one.

-Jonathan R.

Dean Willis wrote:
> 
> Yes. Having both endpoints allows the firewall to construct a far more
> effective dynamic ACL. It also allows UAs to selectively bind their RTP
> streams to a specific partner, rather than waiting open-eared for a download
> of spam.
> 
> --
> Dean
> 
> > -----Original Message-----
> > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > Sent: Sunday, February 13, 2000 11:11 AM
> > To: Dean Willis
> > Cc: 'SIP List'
> > Subject: Re: Question about media desinations
> >
> >
> > This raises an important point. I don't want to make changes to SDP and
> > its usage within SIP to fix a problem that can be prevented by an
> > administrative change in firewall behavior. If the firewalls did not
> > return these error messages, do you still believe including the source
> > port and address in the SDP is useful?
> >
> > -Jonathan R.
> >
> > Dean Willis wrote:
> > >
> > > Try scanning the net at 63.81.221.0/24, for example by
> > telneting to port 122
> > > on 63.81.221.234 -- you'll get back lots of "packet filtered"
> > messages. The
> > > default behavior in most systems is to return such a message.
> > Some firewall
> > > implementors turn them off. Most don't.
> > >
> > > --
> > > Dean
> > >
> > > Jonathan asked:
> > > > This presumes a firewall sends an ICMP error message when it discards
> > > > the packet. I would
> > > > think they would blackhole them, which will eliminate the ability to
> > > > conduct this attack
> > > > you describe.
> >
> > --
> > Jonathan D. Rosenberg                       200 Executive Drive
> > Chief Scientist                             Suite 120
> > dynamicsoft                                 West Orange, NJ 07052
> > jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> > http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> > http://www.dynamicsoft.com
> >

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 17 14:13:23 2000
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Hi:

I had sent this request on Feb 8th. But, unfortunately got no reply
back.
So, i am resending it.

The question was;

where can i find any information in SIP security mailing list or
discussions going on in this group.

Thanks much!

--
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374



--------------7017333F4C4B969150960DF3
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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Hi:
<p>I had sent this request on Feb 8th. But, unfortunately got no reply
back.
<br>So, i am resending it.
<p>The question was;
<p>where can i find any information in SIP security mailing list or discussions
going on in this group.
<p>Thanks much!
<pre>--&nbsp;
Sunitha Kumar
Software Engineer
Vovida Networks
(408) 957 - 6374</pre>
&nbsp;</html>

--------------7017333F4C4B969150960DF3--




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Date: Thu, 17 Feb 2000 13:39:14 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: 183 vs 180, When and Where, a Proposal
In-reply-to: <38AB9C3A.442D3AC7@dynamicsoft.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: IETF SIP <sip@lists.research.bell-labs.com>, Taichi Fu <tfu@cisco.com>
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Well, some marketing folks I know got all gooey over custom ringback, which
I see implemented as 180 with SDP pointing to a streaming audio source. I
suppose one could provide this with a back-to-back UA and a transfer
operation, but the signaling semantics would be much more annoying if the
UAC proved to be a gateway.

--
dean

> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Jonathan
> Rosenberg
> Sent: Thursday, February 17, 2000 12:59 AM
> To: Dean Willis
> Cc: IETF SIP; Taichi Fu
> Subject: Re: 183 vs 180, When and Where, a Proposal
>
>
> I'm not sure I like making this kind of distinction. Why not simply say
> that a UAC always pay attention to media it is asked to receive via a
> provisional response, and that pure IP UASs are advised not to send SDP
> in provisional responses, but rather encode the data in the response
> itself, via reason phrases or content.
>
> -Jonathan R.
>
> Dean Willis wrote:
> >
> > Taichi Fu and I have been having a discussion on 18x
> provisional messages
> > with media that may be worth sharing. Here's what I think we agreed to.
> >
> > For PSTN interworking, you really have to "listen" to the
> bearer channel to
> > figure out what's going on (see previous discussion on release
> codes). Note
> > that this is really quite different from a pure-IP scenario.
> >
> > If we don't have PSTN interworking, then a 180 means that a
> user is being
> > alerted, and media on a 180 clearly indicates a custom message
> that can be
> > played to the caller. For example, a French user might want to
> play back the
> > distinctive French double-ring, and a Trekky might want to
> > play a clip of Lt. Uhuru saying "Captain, there is an incoming
> call".  We
> > might also want to include a text message for the
> audio-impaired device or
> > user, or present a visual notification such as a corporate logo.
> >
> > On the other hand, many UASes (and proxies) MAY choose to
> ignore the media
> > on a 180. This can avoid very complex mixing scenarios in a forking
> > environment.
> >
> > Devices MAY also choose to ignore the media on a 183, but this
> MUST be done
> > very carefully. Media on a 183 for PSTN interworking is far
> more critical
> > than that on a 180 -- it may provide the only useful
> information available
> > to the caller for determining the cause of a call failure. Think very
> > carefully before ignoring a 183.
> >
> > We suggest that a straight IP system, especially a UAS, SHOULD send 180
> > (which MAY have media) when alerting a user. A PSTN gateway
> SHOULD send a
> > 183 with media when "launching" a call onto the PSTN. Devices
> MAY decide to
> > ignore the media of a 180. Devices SHOULD present the media of
> a 183 to an
> > end user, and MUST deal with the results of not doing so.
> >
> > A key point:
> >
> > Some have argued that 183 also has uses outside of PSTN interworking,
> > whenever there is some status information describing a call
> which needs to
> > be presented "in band", but there has been no actual
> notification delivered
> > to a destination.
> >
> > For example, consider a proxy (P) that implements a forking search. An
> > INVITE for "Joe" is delivered to P. P might, in parallel:
> >         1) respond with a 183 containing media in two parts.
> >                 part 1 is an html object with
> >                 text="Your call is being forwarded by Joe's
> Finder Service".
> >                 part 2 is an SDP object describing a media
> stream with a spoken
> >                 rendition of the same text
> >         2) send INVITES to all of the contacts Joe has on his
> contact list.
> >
> > This is probably a BAD usage of 183 -- the status information,
> while useful
> > and entertaining, is not critical to the successful interworking of the
> > call. Its presence could obfuscate an upstream forking system, and would
> > provide no additional value to that system. This example SHOULD be
> > implemented using a 180, not a 183.
> >
> > --
> > Dean
>
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
>




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Date: Thu, 17 Feb 2000 13:39:16 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: 183 vs 180, When and Where, a Proposal
In-reply-to: <4.2.0.58.20000216174540.00d15450@farley.cisco.com>
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Thanks for the clarification. That's what I meant.

--
Dean

Taichi Fu said:
> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Taichi Fu
> Sent: Wednesday, February 16, 2000 9:00 PM
> To: Dean Willis
> Cc: IETF SIP
> Subject: Re: 183 vs 180, When and Where, a Proposal
>
>
> Dean,
>
> Thanks for the summary. I add a small note below the paragraph 6
> from my understanding.
>
> Regards,
> -Taichi
>
> At 05:12 PM 2/16/00 -0600, Dean Willis wrote:
>
> >Taichi Fu and I have been having a discussion on 18x provisional messages
> >with media that may be worth sharing. Here's what I think we agreed to.
> >
> >For PSTN interworking, you really have to "listen" to the bearer
> channel to
> >figure out what's going on (see previous discussion on release
> codes). Note
> >that this is really quite different from a pure-IP scenario.
> >
> >If we don't have PSTN interworking, then a 180 means that a user is being
> >alerted, and media on a 180 clearly indicates a custom message
> that can be
> >played to the caller. For example, a French user might want to
> play back the
> >distinctive French double-ring, and a Trekky might want to
> >play a clip of Lt. Uhuru saying "Captain, there is an incoming call".  We
> >might also want to include a text message for the audio-impaired
> device or
> >user, or present a visual notification such as a corporate logo.
> >
> >On the other hand, many UASes (and proxies) MAY choose to ignore
> the media
> >on a 180. This can avoid very complex mixing scenarios in a forking
> >environment.
> >
> >Devices MAY also choose to ignore the media on a 183, but this
> MUST be done
> >very carefully. Media on a 183 for PSTN interworking is far more critical
> >than that on a 180 -- it may provide the only useful information
> available
> >to the caller for determining the cause of a call failure. Think very
> >carefully before ignoring a 183.
> >
> >We suggest that a straight IP system, especially a UAS, SHOULD send 180
> >(which MAY have media) when alerting a user. A PSTN gateway SHOULD send a
> >183 with media when "launching" a call onto the PSTN. Devices
> MAY decide to
> >ignore the media of a 180. Devices SHOULD present the media of a
> 183 to an
> >end user, and MUST deal with the results of not doing so.
>
> That means, for SIP and PSTN interworking (a PSTN gateway converting
> between SIP messages and SS7 messages):
>
> (1) For a call from SIP to PSTN, a PSTN gateway SHOULD respond
> with 183 , and
> (2) For a call from PSTN to SIP, a PSTN gateway SHOULD expect to
> receive 180.
>
> An asymmetry.
>
> >A key point:
> >
> >Some have argued that 183 also has uses outside of PSTN interworking,
> >whenever there is some status information describing a call
> which needs to
> >be presented "in band", but there has been no actual
> notification delivered
> >to a destination.
> >
> >For example, consider a proxy (P) that implements a forking search. An
> >INVITE for "Joe" is delivered to P. P might, in parallel:
> >         1) respond with a 183 containing media in two parts.
> >                 part 1 is an html object with
> >                 text="Your call is being forwarded by Joe's
> Finder Service".
> >                 part 2 is an SDP object describing a media
> stream with a
> > spoken
> >                 rendition of the same text
> >         2) send INVITES to all of the contacts Joe has on his
> contact list.
> >
> >This is probably a BAD usage of 183 -- the status information,
> while useful
> >and entertaining, is not critical to the successful interworking of the
> >call. Its presence could obfuscate an upstream forking system, and would
> >provide no additional value to that system. This example SHOULD be
> >implemented using a 180, not a 183.
> >
> >--
> >Dean
> >
>
>




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Date: Thu, 17 Feb 2000 13:49:21 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: 183 vs 180, When and Where, a Proposal
In-reply-to: <38AC2581.C87C7B4D@cs.columbia.edu>
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Henning wrote:
> Agreed, or one can argue that it is a feature, described via Supported,
> since not all end systems will even be able to accept voice ahead of 200
> (e.g., they may only start the media tools after the 200 has been
> received).

Then they're going to have a hard time making calls to the PSTN, where
"early media" is a fact of life.


> > > - If this is an error condition, i.e., no call can ensue, when is the
> > > UAS going to send a 4xx or 5xx? There is no right time, since
> it either
> > > prematurely shuts off the explanation or it delays sequential searches
> > > in proxies for very long time periods.
> >
> > My guess is that in practice, there will seldom be forking with gateways
> > involved. As such, reasonable behavior might be to send the audio,
> > potentially delaying the unlikely sequential search.

The UAS sends a 4xx on teardown-with-cause from the PSTN side. This is AFTER
the message has been played. Can't do it sooner and still work.

> I'm not sure that's true. I get the voice announcement if a cell phone
> (Sprint PCS) is not answering. If I route to both a home phone and cell
> number, I don't want the announcement to delay ringing the other phone.

It may be necessary to consider things like a "forking proxy" that can fork,
absorb the 183(s), record the media attached to it(them), notify the caller,
then play the media back on demand. This lets a user do things like "Hmm,
that call failed, I wonder why. Oh, my message light is blinking, there must
be a telco message from that call." I call this "thinking carefully about
ignoring the provisional media". Save this, it's "prior art".

--
Dean





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Dean Willis wrote:
> 
> Henning wrote:
> > Agreed, or one can argue that it is a feature, described via Supported,
> > since not all end systems will even be able to accept voice ahead of 200
> > (e.g., they may only start the media tools after the 200 has been
> > received).
> 
> Then they're going to have a hard time making calls to the PSTN, where
> "early media" is a fact of life.

Not for devices like autodialers. They want to get off the line as
quickly as possible, however useful the phonelady spiel.


> 
> The UAS sends a 4xx on teardown-with-cause from the PSTN side. This is AFTER
> the message has been played. Can't do it sooner and still work.

The problem is that AFTER is ill-defined. As far as I know, the "The
number you have dialed has been changed. The new number is ..." repeats
indefinitely or at least a very long time.

> 
> > I'm not sure that's true. I get the voice announcement if a cell phone
> > (Sprint PCS) is not answering. If I route to both a home phone and cell
> > number, I don't want the announcement to delay ringing the other phone.
> 
> It may be necessary to consider things like a "forking proxy" that can fork,
> absorb the 183(s), record the media attached to it(them), notify the caller,
> then play the media back on demand. This lets a user do things like "Hmm,
> that call failed, I wonder why. Oh, my message light is blinking, there must
> be a telco message from that call." I call this "thinking carefully about
> ignoring the provisional media". Save this, it's "prior art".
> 
> --
> Dean

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



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Subject: Re: Question about media desinations
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Jonathan Rosenberg wrote:
> 
> Hmmm. Actually, placing the source address in the SDP solves another
> problem. When multiple 200 OK (or 183 or whatever) arrive with SDP, the
> caller is going to receive audio all on the same port, with no way to
> figure out which RTP stream corresponds to which call leg. By adding the
> source address, the UAC will be able to correlate them.

While this is helpful, it is not foolproof, since RTP translators will
mess up the IP address. I think RTCP-based means are preferable. Also,
for multi-homed hosts, it's not always easy to know at the application
layer, which source address is going to be used, since that can depend
on routing. While this is probably infrequent, relying on it may be
suboptimal. Maybe RTCP SDES should contain the appropriate identifier,
including tag.

>

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



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Date: Thu, 17 Feb 2000 15:35:04 -0600
From: Robert Sparks <Robert.Sparks@wcom.com>
Subject: RE: Question about media desinations
In-reply-to: <38AC5EA9.AFC3FD93@cs.columbia.edu>
To: "'Henning Schulzrinne'" <schulzrinne@cs.columbia.edu>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
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How would an RTP translator get into the media path of a
SIP initiated session without one of the SIP participants
being aware of it during the setup?

Multi-homed SIP entities already have to discover which
interfaces are going to participate in a given SIP 
transaction (for proper construction of Vias, Contacts,
Record-Routes, and possibly the receiving address in
session description).

In the context of an RTCP-based solution and the case that
Jonathan describes, what would the client do with RTP
packets received before the first associated RTCP packet
arrives?

RjS

Henning Schulzrinne writes:
> Jonathan Rosenberg wrote:
> > 
> > Hmmm. Actually, placing the source address in the SDP solves another
> > problem. When multiple 200 OK (or 183 or whatever) arrive 
> with SDP, the
> > caller is going to receive audio all on the same port, with 
> no way to
> > figure out which RTP stream corresponds to which call leg. 
> By adding the
> > source address, the UAC will be able to correlate them.
> 
> While this is helpful, it is not foolproof, since RTP translators will
> mess up the IP address. I think RTCP-based means are preferable. Also,
> for multi-homed hosts, it's not always easy to know at the application
> layer, which source address is going to be used, since that can depend
> on routing. While this is probably infrequent, relying on it may be
> suboptimal. Maybe RTCP SDES should contain the appropriate identifier,
> including tag.




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Robert Sparks wrote:
> 
> How would an RTP translator get into the media path of a
> SIP initiated session without one of the SIP participants
> being aware of it during the setup?

Never mind...

> 
> Multi-homed SIP entities already have to discover which
> interfaces are going to participate in a given SIP
> transaction (for proper construction of Vias, Contacts,
> Record-Routes, and possibly the receiving address in
> session description).

Not really, since if the receiver binds on INADDR_ANY, it doesn't matter
which interface the SIP packet comes in on.

> 
> In the context of an RTCP-based solution and the case that
> Jonathan describes, what would the client do with RTP
> packets received before the first associated RTCP packet
> arrives?

Mix it and play it out. Streams can be distinguished based on SSRC, they
just can't be identified with some blinking icon which says "Joe is
talking right now".



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



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Matthias.Lang@ericsson.com wrote:
> 
> Jonathan Rosenberg writes:
>  > Hmmm. Actually, placing the source address in the SDP solves another
>  > problem. When multiple 200 OK (or 183 or whatever) arrive with SDP, the
>  > caller is going to receive audio all on the same port, with no way to
>  > figure out which RTP stream corresponds to which call leg. By adding the
>  > source address, the UAC will be able to correlate them.
> 
> The easy way to figure out which RTP stream corresponds to which SIP
> call is to use different destination port numbers. I don't think
> you can always distinguish calls based on source address, I don't
> think there's anything in the RTP spec which says I can't send two
> different RTP streams from the same port.

Clearly thats the preferred solution, but it doesn't work for multiple
200 OK - thats my point. The INVITE contains the port I expect to
receive media on. If two gateways receive that, and each send a 183 and
then send media, they will end up sending media to that same port. Thus,
I cannot use the port to differentiate. Of course, I will realize there
are two separate streams. These two streams will generally have
different SSRC, so I can separate them. However, there is no way to
correlate a particular stream (identified by its SSRC) with the SIP call
leg.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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From: "Daniel G. Petrie" <dpetrie@pingtel.com>
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        sip@lists.research.bell-labs.com
Subject: Re: UAC use of tags received in responses
References: <003201bf78b2$8f59c4c0$8a9223a6@mcit.com> <38AB0319.B3E2AF52@dynamicsoft.com> <38AB99A8.D33E74FB@dynamicsoft.com> <38AB9F58.6889F1B7@dynamicsoft.com> <38AC2DE1.4E5B10B0@dynamicsoft.com>
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Sorry I could not resist these two questions.

What would one expect to happen when sending
a CANCEL to a ReINVITE:
  Cancel the transaction, leave the call up at the previously negotiated
capabilities.
 or
  Cancel the transaction, drop the call.

Why would someone want to do this?
Cheers,
Dan


Jonathan Rosenberg wrote:

>
>
> Fair enough. However, I think that in the case of cancelling a
> re-INVITE,
> you do actually want the tag in the To field. I believe the CANCEL needs
> to
> match the request its cancelling, in terms of the To, From, Call-ID and
> CSeq,
> including tags. In the case of multiple call legs, if a re-INVITE were
> sent on
> each call leg, and then cancelled on each call leg, you might not be
> able to distinguish
> the CANCELs except for the tag in the To field.
>
> So, if this is OK, the proposed addition is:
>
> "The exception to this is CANCEL. The CANCEL always contains exactly the
> same value of the To field as the request it is cancelling."
>
> -Jonathan R.
>
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com

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From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 17 20:49:52 2000
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Subject: Re: UAC use of tags received in responses
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"Daniel G. Petrie" wrote:
> 
> What would one expect to happen when sending
> a CANCEL to a ReINVITE:
>   Cancel the transaction, leave the call up at the previously negotiated
> capabilities.
>  or
>   Cancel the transaction, drop the call.

The former.
 
> Why would someone want to do this?

Media renegotiation might involve some user interaction and the caller
may want to give up on the re-INVITE if it does not complete within
certain time. For example, a re-INVITE may be sent when one of the call
participants wants to start video exchange. If the other side is capable
of supporting it, the user there might be prompted to see if he/she
agrees to activate video.

---
Igor Slepchin



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 18 09:30:07 2000
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Folks,

Many of you may have seen the recent draft presenting DHCP option codes
for SIP:

http://search.ietf.org/internet-drafts/draft-nair-sip-dhcp-01.txt

The consensus among the authors and wg chairs is that this is a clearly
useful item, and we would like to move it forward. In the interests of
speed, we are proposing a "fast track" approach for this document.
Basically, this message will serve as announcement of our intention to
adopt this as a work item. If no one objects to this procedure, in one
weeks time, we will issue an extended working group last call on it, and
that call will last 3 weeks. Comments will be collected, and then we
will send the document to IESG for proposed.

I believe its critical that we do whatever we can to speed up the
process of delivering documents to the community. This is especially the
case for small, largely uncontroversial documents such as this. Rather
than languish for months, getting this from concept to submission to
IESG in a matter of a single month seems more desirable.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 18 09:43:21 2000
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This is a question regarding UDP connections and the received tag in
Via header fields (6.40.2 of the draft I believe.)  I did not see this
answered off hand in the clarifications or elsewhere.

Section 6.40.2 states that a proxy should check that the address in 
the last Via: header matches the address that the request was actually
received from.  If the addresses do not match, the proxy must add the
received tag with the address that the request was actually received
from.  

My problem is with what happens if the address/hostname in the Via
header matches several IP addresses, including the one actually used
to send it, but not all of the addresses are routable, preferred, or
public.

That is, suppose a server receives a Via header like:

  Via: SIP/2.0/UDP aka.generic.com:5060

in a UDP packet with an originating address of 10.100.100.100.
The server looks up the IP addresses for aka.generic.com to see
whether or not 10.100.100.100 is one of its addresses.  The
returned addresses for aka.generic.com are:

  192.168.128.1, 192.168.128.2, 192.168.128.3, 192.168.128.3, and
  10.100.100.100.

So, since the 10.100.100.100 address is present, no received tag is
added to the Via field.  Now, when the response comes back through
the proxy, and the proxy tries to forward it back along the Via
lines, which IP address should the proxy choose to send the UDP
message to?  (Of course, the 192.* addresses might be public
interfaces, and the 10.100.100.100 address is on a private network
to keep call information private, etc.)  Could a recieved= tag 
be added to the Via: header anyways to be used to tell the proxy
exactly which IP address the response should be sent to?  

(Hmmm. This almost but does not quite feel like a security issue.
If a target server has an IP address of 10.10.10.10, an attacker
sets up a DNS entry for bogus.someplace.com that has IP addresses
10.10.10.10 and 20.20.20.20, then sends a packet from 20.20.20.20
with a Via line specifying bogus.someplace.com as the return hop.)

Also, I think I may have seen this question earlier, but I cannot
find it now (is there a mailing list archive somewhere?) -- how does
the Via:/received affect loop detection?  The example given in 
the draft indicates that one use of the received tag was for when
a SIP request is passing through NAT.  When checking for loops
and a received tag is present, should only the address in the 
received tag be checked?  (For instance, one organization is 
using the 10.* network behind NAT, and a SIP request is sent
across the public network to a second organization which also uses
the 10.* network behind NAT.  The SIP servers in the second 
organization see a Via header:

  Via: SIP/2.0/UDP 10.10.10.10:5060; received=200.200.200.200

If the second organization using the 10.* network also has a SIP
server at 10.10.10.10, will this show up as an accidental loop 
detection?

Thanks,
Kim Liu



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Igor Slepchin wrote:
> 
> "Daniel G. Petrie" wrote:
> >
> > What would one expect to happen when sending
> > a CANCEL to a ReINVITE:
> >   Cancel the transaction, leave the call up at the previously negotiated
> > capabilities.
> >  or
> >   Cancel the transaction, drop the call.
> 
> The former.
> 
> > Why would someone want to do this?
> 
> Media renegotiation might involve some user interaction and the caller
> may want to give up on the re-INVITE if it does not complete within
> certain time. For example, a re-INVITE may be sent when one of the call
> participants wants to start video exchange. If the other side is capable
> of supporting it, the user there might be prompted to see if he/she
> agrees to activate video.

I'd also like to add to this that whilst CANCEL is most useful for
INVITE, it is a generic mechanism that can be applied to any request
(excepting CANCEL itself and ACK). In general, if you get a CANCEL for a
request for which you haven't sent a response, you should send a 487
response to the original request, 200 OK to the CANCEL, and then behave
as if the original request were never received.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Henning Schulzrinne wrote:
> 
> > In the context of an RTCP-based solution and the case that
> > Jonathan describes, what would the client do with RTP
> > packets received before the first associated RTCP packet
> > arrives?
> 
> Mix it and play it out. Streams can be distinguished based on SSRC, they
> just can't be identified with some blinking icon which says "Joe is
> talking right now".

The bigger problem is firewalls. If we presume that the firewall won't
let things through unless it has the source address at least (source
port as well I guess would be desirable), the RTCP will never get
through in the first place. 

I'm not 100% convinced of putting the source address/port in the SDP yet
- still trying to understand the pros and cons.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Date: Fri, 18 Feb 2000 08:57:03 -0600
From: Robert Sparks <Robert.Sparks@wcom.com>
Subject: RE: Question about media desinations
In-reply-to: <38AC724F.969A824@cs.columbia.edu>
To: "'Henning Schulzrinne'" <schulzrinne@cs.columbia.edu>
Cc: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        "'IETF SIP'" <sip@lists.research.bell-labs.com>
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Henning Schulzrinne wrote:
> > 
> > Multi-homed SIP entities already have to discover which
> > interfaces are going to participate in a given SIP
> > transaction (for proper construction of Vias, Contacts,
> > Record-Routes, and possibly the receiving address in
> > session description).
> 
> Not really, since if the receiver binds on INADDR_ANY, it 
> doesn't matter
> which interface the SIP packet comes in on.

If the interfaces are to discrete networks, there could be
no valid response path to a request with a Via indicating
the wrong interface.

RjS



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References: <000501bf798e$db4348c0$8a9223a6@mcit.com> <38AC724F.969A824@cs.columbia.edu> <38AD5DBA.483CD19D@dynamicsoft.com>
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> The bigger problem is firewalls. If we presume that the firewall won't
> let things through unless it has the source address at least (source
> port as well I guess would be desirable), the RTCP will never get
> through in the first place.
> 
> I'm not 100% convinced of putting the source address/port in the SDP yet
> - still trying to understand the pros and cons.

I wonder how firewalls handle current RealAudio streams. I gather that
most do allow Real UDP packets through, although I haven't checked this.



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 18 10:34:10 2000
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To: Dean Willis <dean.willis@wcom.com>
Cc: IETF SIP <sip@lists.research.bell-labs.com>, Taichi Fu <tfu@cisco.com>
Subject: Re: 183 vs 180, When and Where, a Proposal
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Thats fine. My suggestion was that if possible, to encode reasons in
responses rather than use provisional media. But, that UACs still pay
attention to it if provisional media is sent, for something like
distinctive ringing.

-Jonathan R.

Dean Willis wrote:
> 
> Well, some marketing folks I know got all gooey over custom ringback, which
> I see implemented as 180 with SDP pointing to a streaming audio source. I
> suppose one could provide this with a back-to-back UA and a transfer
> operation, but the signaling semantics would be much more annoying if the
> UAC proved to be a gateway.
> 
> --
> dean
> 
> > -----Original Message-----
> > From: owner-sip@lists.research.bell-labs.com
> > [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Jonathan
> > Rosenberg
> > Sent: Thursday, February 17, 2000 12:59 AM
> > To: Dean Willis
> > Cc: IETF SIP; Taichi Fu
> > Subject: Re: 183 vs 180, When and Where, a Proposal
> >
> >
> > I'm not sure I like making this kind of distinction. Why not simply say
> > that a UAC always pay attention to media it is asked to receive via a
> > provisional response, and that pure IP UASs are advised not to send SDP
> > in provisional responses, but rather encode the data in the response
> > itself, via reason phrases or content.
> >
> > -Jonathan R.
> >
> > Dean Willis wrote:
> > >
> > > Taichi Fu and I have been having a discussion on 18x
> > provisional messages
> > > with media that may be worth sharing. Here's what I think we agreed to.
> > >
> > > For PSTN interworking, you really have to "listen" to the
> > bearer channel to
> > > figure out what's going on (see previous discussion on release
> > codes). Note
> > > that this is really quite different from a pure-IP scenario.
> > >
> > > If we don't have PSTN interworking, then a 180 means that a
> > user is being
> > > alerted, and media on a 180 clearly indicates a custom message
> > that can be
> > > played to the caller. For example, a French user might want to
> > play back the
> > > distinctive French double-ring, and a Trekky might want to
> > > play a clip of Lt. Uhuru saying "Captain, there is an incoming
> > call".  We
> > > might also want to include a text message for the
> > audio-impaired device or
> > > user, or present a visual notification such as a corporate logo.
> > >
> > > On the other hand, many UASes (and proxies) MAY choose to
> > ignore the media
> > > on a 180. This can avoid very complex mixing scenarios in a forking
> > > environment.
> > >
> > > Devices MAY also choose to ignore the media on a 183, but this
> > MUST be done
> > > very carefully. Media on a 183 for PSTN interworking is far
> > more critical
> > > than that on a 180 -- it may provide the only useful
> > information available
> > > to the caller for determining the cause of a call failure. Think very
> > > carefully before ignoring a 183.
> > >
> > > We suggest that a straight IP system, especially a UAS, SHOULD send 180
> > > (which MAY have media) when alerting a user. A PSTN gateway
> > SHOULD send a
> > > 183 with media when "launching" a call onto the PSTN. Devices
> > MAY decide to
> > > ignore the media of a 180. Devices SHOULD present the media of
> > a 183 to an
> > > end user, and MUST deal with the results of not doing so.
> > >
> > > A key point:
> > >
> > > Some have argued that 183 also has uses outside of PSTN interworking,
> > > whenever there is some status information describing a call
> > which needs to
> > > be presented "in band", but there has been no actual
> > notification delivered
> > > to a destination.
> > >
> > > For example, consider a proxy (P) that implements a forking search. An
> > > INVITE for "Joe" is delivered to P. P might, in parallel:
> > >         1) respond with a 183 containing media in two parts.
> > >                 part 1 is an html object with
> > >                 text="Your call is being forwarded by Joe's
> > Finder Service".
> > >                 part 2 is an SDP object describing a media
> > stream with a spoken
> > >                 rendition of the same text
> > >         2) send INVITES to all of the contacts Joe has on his
> > contact list.
> > >
> > > This is probably a BAD usage of 183 -- the status information,
> > while useful
> > > and entertaining, is not critical to the successful interworking of the
> > > call. Its presence could obfuscate an upstream forking system, and would
> > > provide no additional value to that system. This example SHOULD be
> > > implemented using a 180, not a 183.
> > >
> > > --
> > > Dean
> >
> > --
> > Jonathan D. Rosenberg                       200 Executive Drive
> > Chief Scientist                             Suite 120
> > dynamicsoft                                 West Orange, NJ 07052
> > jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> > http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> > http://www.dynamicsoft.com
> >

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 18 10:37:53 2000
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Date: Fri, 18 Feb 2000 09:29:01 -0600
From: Robert Sparks <Robert.Sparks@wcom.com>
Subject: Cancelled, Decllined, or Failed ReINVITES (was RE: UAC use of tags
 received in responses)
In-reply-to: <38ACA3CC.78FD2652@dynamicsoft.com>
To: "'Igor Slepchin'" <islepchin@dynamicsoft.com>,
        "'Daniel G. Petrie'" <dpetrie@pingtel.com>
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This is like the case when the reINVITE ultimately
receives a non-200 final response (not including 401, 407
or responses that induce Retry-Afters leading in the end
to a reINVITE receiving a 200). 

I think that case has been discussed here (I know it has
been discussed at the bake-offs). Returning (if possible)
to the previous call state is the right approach. UA
implementations must be careful, however, with how they get
back to that state. A failed HOLD, for example, better not 
leave the user thinking the call is on hold while media is
still being sent to the other end. 

RjS

ps. More at the edge, if you have a reINVITE/200/ACK sequence
where the 200 contains "bad" SDP, yet another reINVITE will 
have to be issued to restore the call to the previous state (which
should have a high probability of success since the UAs agreed
to get that call into that state once before).

> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Igor
> Slepchin
> Sent: Thursday, February 17, 2000 7:44 PM
> To: Daniel G. Petrie
> Cc: Jonathan Rosenberg; sip@lists.research.bell-labs.com
> Subject: Re: UAC use of tags received in responses
> 
> 
> "Daniel G. Petrie" wrote:
> > 
> > What would one expect to happen when sending
> > a CANCEL to a ReINVITE:
> >   Cancel the transaction, leave the call up at the 
> previously negotiated
> > capabilities.
> >  or
> >   Cancel the transaction, drop the call.
> 
> The former.
>  
> > Why would someone want to do this?
> 
> Media renegotiation might involve some user interaction and the caller
> may want to give up on the re-INVITE if it does not complete within
> certain time. For example, a re-INVITE may be sent when one 
> of the call
> participants wants to start video exchange. If the other side 
> is capable
> of supporting it, the user there might be prompted to see if he/she
> agrees to activate video.
> 
> ---
> Igor Slepchin
> 



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 18 10:41:42 2000
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Dean Willis wrote:
> 
> > I'm not sure that's true. I get the voice announcement if a cell phone
> > (Sprint PCS) is not answering. If I route to both a home phone and cell
> > number, I don't want the announcement to delay ringing the other phone.
> 
> It may be necessary to consider things like a "forking proxy" that can fork,
> absorb the 183(s), record the media attached to it(them), notify the caller,
> then play the media back on demand. This lets a user do things like "Hmm,
> that call failed, I wonder why. Oh, my message light is blinking, there must
> be a telco message from that call." I call this "thinking carefully about
> ignoring the provisional media". Save this, it's "prior art".

You can certainly do this, but I definitely don't want to mandate this
kind of behavior in a normal proxy. 

-Jonathan R.


-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Date: Fri, 18 Feb 2000 19:36:30 +0200
From: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
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Hi,

I have a question about the syntax of the display-name.

According to the draft, the syntax is:

display-name = *token | quoted-string

My question concerns the case when we do NOT use quotes. Then, we should
only be allowed to use token characters, which don't allow space
characters, but in many of the examples I've seen we have, for example,
a name, a space(!), and a family name - and this without quotes.

eg.

To: T.<SP>Watson ...



Regards,

Christer Holmberg
Ericsson Finland



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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: sip@lists.research.bell-labs.com
Subject: DHCP option tags for SIP
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I'd like to throw out one wet blanket: getting the
name of a first hop proxy is all well and good,
but it hardly solves the larger provisioning
problem, especially for "simple" phones and other
devices which don't have an easy way to be
configured by a user, or want/need to be centrally
managed. Trying to cram bootstrap provisioning for
N different kinds of protocols/devices into one
MTU sized packet seems like a losing situation and
symptomatic that DHCP is being used for way more
than it can deliver in the long run.

It seems to me that it would be far better to
create some sort of indirection method within DHCP
which says "here's the URL if you want to get SIP
related parameters" (ad naseum). 

But all that's being asked for is "just" the
address of the first hop proxy... sigh.

	   Mike


Jonathan Rosenberg writes:
 > Folks,
 > 
 > Many of you may have seen the recent draft presenting DHCP option codes
 > for SIP:
 > 
 > http://search.ietf.org/internet-drafts/draft-nair-sip-dhcp-01.txt
 > 
 > The consensus among the authors and wg chairs is that this is a clearly
 > useful item, and we would like to move it forward. In the interests of
 > speed, we are proposing a "fast track" approach for this document.
 > Basically, this message will serve as announcement of our intention to
 > adopt this as a work item. If no one objects to this procedure, in one
 > weeks time, we will issue an extended working group last call on it, and
 > that call will last 3 weeks. Comments will be collected, and then we
 > will send the document to IESG for proposed.
 > 
 > I believe its critical that we do whatever we can to speed up the
 > process of delivering documents to the community. This is especially the
 > case for small, largely uncontroversial documents such as this. Rather
 > than languish for months, getting this from concept to submission to
 > IESG in a matter of a single month seems more desirable.
 > 
 > -Jonathan R.
 > -- 
 > Jonathan D. Rosenberg                       200 Executive Drive
 > Chief Scientist                             Suite 120 
 > dynamicsoft                                 West Orange, NJ 07052
 > jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
 > http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
 > http://www.dynamicsoft.com
 > 
 > 
 > 



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From: James Kempf <James.Kempf@Eng.Sun.COM>
Reply-To: James Kempf <James.Kempf@Eng.Sun.COM>
Subject: Re: DHCP option tags for SIP
To: jdrosen@dynamicsoft.com, mat@cisco.com
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Well, at the risk of prolonging a controversy, SLP was designed to
fix exactly the problems you outline with DHCP.

		jak

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>Delivered-To: sip-local@paperless.dnrc.bell-labs.com
>From: Michael Thomas <mat@cisco.com>
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>Date: Fri, 18 Feb 2000 09:41:42 -0800 (PST)
>To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
>Cc: sip@lists.research.bell-labs.com
>Subject: DHCP option tags for SIP
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bUcI'1!
>
>
>I'd like to throw out one wet blanket: getting the
>name of a first hop proxy is all well and good,
>but it hardly solves the larger provisioning
>problem, especially for "simple" phones and other
>devices which don't have an easy way to be
>configured by a user, or want/need to be centrally
>managed. Trying to cram bootstrap provisioning for
>N different kinds of protocols/devices into one
>MTU sized packet seems like a losing situation and
>symptomatic that DHCP is being used for way more
>than it can deliver in the long run.
>
>It seems to me that it would be far better to
>create some sort of indirection method within DHCP
>which says "here's the URL if you want to get SIP
>related parameters" (ad naseum). 
>
>But all that's being asked for is "just" the
>address of the first hop proxy... sigh.
>
>	   Mike
>
>
>Jonathan Rosenberg writes:
> > Folks,
> > 
> > Many of you may have seen the recent draft presenting DHCP option codes
> > for SIP:
> > 
> > http://search.ietf.org/internet-drafts/draft-nair-sip-dhcp-01.txt
> > 
> > The consensus among the authors and wg chairs is that this is a clearly
> > useful item, and we would like to move it forward. In the interests of
> > speed, we are proposing a "fast track" approach for this document.
> > Basically, this message will serve as announcement of our intention to
> > adopt this as a work item. If no one objects to this procedure, in one
> > weeks time, we will issue an extended working group last call on it, and
> > that call will last 3 weeks. Comments will be collected, and then we
> > will send the document to IESG for proposed.
> > 
> > I believe its critical that we do whatever we can to speed up the
> > process of delivering documents to the community. This is especially the
> > case for small, largely uncontroversial documents such as this. Rather
> > than languish for months, getting this from concept to submission to
> > IESG in a matter of a single month seems more desirable.
> > 
> > -Jonathan R.
> > -- 
> > Jonathan D. Rosenberg                       200 Executive Drive
> > Chief Scientist                             Suite 120 
> > dynamicsoft                                 West Orange, NJ 07052
> > jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> > http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> > http://www.dynamicsoft.com
> > 
> > 
> > 
>




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To: Christer Holmberg <christer.holmberg@lmf.ericsson.se>
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Christer Holmberg wrote:
> 
> Hi,
> 
> I have a question about the syntax of the display-name.
> 
> According to the draft, the syntax is:
> 
> display-name = *token | quoted-string
> 
> My question concerns the case when we do NOT use quotes. Then, we should
> only be allowed to use token characters, which don't allow space
> characters, but in many of the examples I've seen we have, for example,
> a name, a space(!), and a family name - and this without quotes.

*token matches zero or more tokens. The grammar allows whitespace
between these tokens - see the comments on implied LWS in appendix C of
rfc 2543.

Anders

-- 
Anders Kristensen <ak@hplb.hpl.hp.com>,
http://www-uk.hpl.hp.com/people/ak/
Hewlett-Packard Labs, Bristol, UK



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 18 13:25:57 2000
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Michael Thomas wrote:
> 
> I'd like to throw out one wet blanket: getting the
> name of a first hop proxy is all well and good,
> but it hardly solves the larger provisioning
> problem, especially for "simple" phones and other
> devices which don't have an easy way to be
> configured by a user, or want/need to be centrally
> managed. Trying to cram bootstrap provisioning for
> N different kinds of protocols/devices into one
> MTU sized packet seems like a losing situation and
> symptomatic that DHCP is being used for way more
> than it can deliver in the long run.

As far as I can tell from our implementation experience, this is pretty
much (*) the last piece of information needed to configure a phone. At
the DHCP level, we currently use for a basic phone:

- IP address, DNS server address, default gateway (standard DHCP fare)
- time server address
- outbound SIP proxy server
- normal boot image information for phones with downloadable code

We have to implement DHCP in any event, so the incremental cost of
providing this is close to zero.

There are other SIP-level configurations that are inappropriate, in my
view, for DHCP, things like speed dial configurations or other UI
assignments, any end system CPL, etc.

> 
> It seems to me that it would be far better to
> create some sort of indirection method within DHCP
> which says "here's the URL if you want to get SIP
> related parameters" (ad naseum).
> 
> But all that's being asked for is "just" the
> address of the first hop proxy... sigh.

As I said, this is necessary and sufficient to get working phones, from
what I can tell.

> 
>            Mike



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 18 14:44:42 2000
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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There is definitely room for both. DHCP is appropriate where a simple
"gimme the closest proxy" will do. SLP is better when the selection
process is more complicated, and depends on attributes of the proxy.

-Jonathan R.

James Kempf wrote:
> 
> Well, at the risk of prolonging a controversy, SLP was designed to
> fix exactly the problems you outline with DHCP.
> 
>                 jak
> 

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Hi:

Can the proxy tag the From or the To fields.

Thanks!

Sunitha Kumar




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Sunitha Kumar wrote:
> 
> Hi:
> 
> Can the proxy tag the From or the To fields.

Proxy should never touch the From field. A proxy should put a tag in the
To field of final responses it generates for the requests with no tag in
the To.

---
Igor Slepchin



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To: Michael Thomas <mat@cisco.com>
Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        sip@lists.research.bell-labs.com
Subject: Re: DHCP option tags for SIP
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The option is SLP I think .... DHCP as a bootstrap protocol doesn't ring
too well with me.

-=Francois=-

On Fri, 18 Feb 2000, Michael Thomas wrote:

> 
> I'd like to throw out one wet blanket: getting the
> name of a first hop proxy is all well and good,
> but it hardly solves the larger provisioning
> problem, especially for "simple" phones and other
> devices which don't have an easy way to be
> configured by a user, or want/need to be centrally
> managed. Trying to cram bootstrap provisioning for
> N different kinds of protocols/devices into one
> MTU sized packet seems like a losing situation and
> symptomatic that DHCP is being used for way more
> than it can deliver in the long run.
> 
> It seems to me that it would be far better to
> create some sort of indirection method within DHCP
> which says "here's the URL if you want to get SIP
> related parameters" (ad naseum). 
> 
> But all that's being asked for is "just" the
> address of the first hop proxy... sigh.
> 
> 	   Mike
> 
> 
> Jonathan Rosenberg writes:
>  > Folks,
>  > 
>  > Many of you may have seen the recent draft presenting DHCP option codes
>  > for SIP:
>  > 
>  > http://search.ietf.org/internet-drafts/draft-nair-sip-dhcp-01.txt
>  > 
>  > The consensus among the authors and wg chairs is that this is a clearly
>  > useful item, and we would like to move it forward. In the interests of
>  > speed, we are proposing a "fast track" approach for this document.
>  > Basically, this message will serve as announcement of our intention to
>  > adopt this as a work item. If no one objects to this procedure, in one
>  > weeks time, we will issue an extended working group last call on it, and
>  > that call will last 3 weeks. Comments will be collected, and then we
>  > will send the document to IESG for proposed.
>  > 
>  > I believe its critical that we do whatever we can to speed up the
>  > process of delivering documents to the community. This is especially the
>  > case for small, largely uncontroversial documents such as this. Rather
>  > than languish for months, getting this from concept to submission to
>  > IESG in a matter of a single month seems more desirable.
>  > 
>  > -Jonathan R.
>  > -- 
>  > Jonathan D. Rosenberg                       200 Executive Drive
>  > Chief Scientist                             Suite 120 
>  > dynamicsoft                                 West Orange, NJ 07052
>  > jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
>  > http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
>  > http://www.dynamicsoft.com
>  > 
>  > 
>  > 
> 




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So, let me poke at this a bit: 

1) if I have a residential gateway, say, which runs 
   SIP on its two POTS lines, with SLP can I find 
   the (different) first hop proxies for each port? 
2) Suppose I have some IP phones at home which are
   gateway'd through my home SIP proxy which allow
   me to set up local dial plans. Would SLP be adequate
   to configure the 1) phones 2) my home proxy for learning
   their next hope proxy?

I'm not trying to diss the idea, but SLP seems to
have been written with enterprises in mind --
which is fine -- but the more complicated
situations will probably need to be considered
where we start splitting up the monolithicly
administered network into various administrative
domains, and differently administered services.

	     Mike

Francois Menard List Account writes:
 > 
 > 
 > The option is SLP I think .... DHCP as a bootstrap protocol doesn't ring
 > too well with me.
 > 
 > -=Francois=-
 > 
 > On Fri, 18 Feb 2000, Michael Thomas wrote:
 > 
 > > 
 > > I'd like to throw out one wet blanket: getting the
 > > name of a first hop proxy is all well and good,
 > > but it hardly solves the larger provisioning
 > > problem, especially for "simple" phones and other
 > > devices which don't have an easy way to be
 > > configured by a user, or want/need to be centrally
 > > managed. Trying to cram bootstrap provisioning for
 > > N different kinds of protocols/devices into one
 > > MTU sized packet seems like a losing situation and
 > > symptomatic that DHCP is being used for way more
 > > than it can deliver in the long run.
 > > 
 > > It seems to me that it would be far better to
 > > create some sort of indirection method within DHCP
 > > which says "here's the URL if you want to get SIP
 > > related parameters" (ad naseum). 
 > > 
 > > But all that's being asked for is "just" the
 > > address of the first hop proxy... sigh.
 > > 
 > > 	   Mike
 > > 
 > > 
 > > Jonathan Rosenberg writes:
 > >  > Folks,
 > >  > 
 > >  > Many of you may have seen the recent draft presenting DHCP option codes
 > >  > for SIP:
 > >  > 
 > >  > http://search.ietf.org/internet-drafts/draft-nair-sip-dhcp-01.txt
 > >  > 
 > >  > The consensus among the authors and wg chairs is that this is a clearly
 > >  > useful item, and we would like to move it forward. In the interests of
 > >  > speed, we are proposing a "fast track" approach for this document.
 > >  > Basically, this message will serve as announcement of our intention to
 > >  > adopt this as a work item. If no one objects to this procedure, in one
 > >  > weeks time, we will issue an extended working group last call on it, and
 > >  > that call will last 3 weeks. Comments will be collected, and then we
 > >  > will send the document to IESG for proposed.
 > >  > 
 > >  > I believe its critical that we do whatever we can to speed up the
 > >  > process of delivering documents to the community. This is especially the
 > >  > case for small, largely uncontroversial documents such as this. Rather
 > >  > than languish for months, getting this from concept to submission to
 > >  > IESG in a matter of a single month seems more desirable.
 > >  > 
 > >  > -Jonathan R.
 > >  > -- 
 > >  > Jonathan D. Rosenberg                       200 Executive Drive
 > >  > Chief Scientist                             Suite 120 
 > >  > dynamicsoft                                 West Orange, NJ 07052
 > >  > jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
 > >  > http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
 > >  > http://www.dynamicsoft.com
 > >  > 
 > >  > 
 > >  > 
 > > 
 > 
 > 
 > 
 > 



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Subject: Re: DHCP option tags for SIP
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Henning Schulzrinne writes:
 > As far as I can tell from our implementation experience, this is pretty
 > much (*) the last piece of information needed to configure a phone. At
 > the DHCP level, we currently use for a basic phone:
 > 
 > - IP address, DNS server address, default gateway (standard DHCP fare)
 > - time server address
 > - outbound SIP proxy server
 > - normal boot image information for phones with downloadable code

   Dumb question: would a phone need its E.164 address(es)
   when registering, etc?

 > We have to implement DHCP in any event, so the incremental cost of
 > providing this is close to zero.

   Sure, but that wasn't my point of course.
 
 > There are other SIP-level configurations that are inappropriate, in my
 > view, for DHCP, things like speed dial configurations or other UI
 > assignments, any end system CPL, etc.

   This, however, was my point. If you believe in directories
   for configless widgets, it would make a lot more sense to
   dump all of this information there and have a pointer to
   the directory in DHCP. If the sum total of configuration 
   that a configurationless device is greater than just the
   address of the first hop proxy, then all this new
   option is doing is solving a small part of a previously 
   unsolved problem.

		Mike



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Michael Thomas wrote:
> 
> Henning Schulzrinne writes:
>  > As far as I can tell from our implementation experience, this is pretty
>  > much (*) the last piece of information needed to configure a phone. At
>  > the DHCP level, we currently use for a basic phone:
>  >
>  > - IP address, DNS server address, default gateway (standard DHCP fare)
>  > - time server address
>  > - outbound SIP proxy server
>  > - normal boot image information for phones with downloadable code
> 
>    Dumb question: would a phone need its E.164 address(es)
>    when registering, etc?

Actually, an interesting question. A phone would need the identity with
which it should register. From what I can tell, PBX will probably have
two identities for their phones, one "legacy" (E.164) identity and one
or more SIP identities (email address). It could be more than one if two
people share a phone, as our grad students do.

It might even have a third identity reflecting the phone's location,
e.g., "Room2F524@somewhere.com". The first two types of identities would
change if a person moved into a different office (assuming that she
leaves her phone behind), while the latter would stay constant.
Conversely, if people take their phones with them, only the last
identity would change.

Since these identities don't generally change all that often, this could
be hard-coded. (Fancier mechanisms, such as identities conveyed by
PalmPilots, have been discussed elsewhere.) Alternatively, a few of us
have discussed that there might well be a need for an additional
configuration parameter, namely a list of "identities" that the phone
should register with. Pretty straightforward to define, but I kind of
wanted to get the server issue out of the way first.


> 
>  > There are other SIP-level configurations that are inappropriate, in my
>  > view, for DHCP, things like speed dial configurations or other UI
>  > assignments, any end system CPL, etc.
> 
>    This, however, was my point. If you believe in directories
>    for configless widgets, it would make a lot more sense to
>    dump all of this information there and have a pointer to
>    the directory in DHCP. If the sum total of configuration
>    that a configurationless device is greater than just the
>    address of the first hop proxy, then all this new
>    option is doing is solving a small part of a previously
>    unsolved problem.

Our current intent is to put some of this into the REGISTER response,
using an appropriate MIME type, since it avoids having to define yet
another directory protocol, more servers, etc. (Putting LDAP into an
embedded system strikes me as challenging, to put it mildly.)

However, the outbound proxy can be different from my registration
server.

Indeed, in order to find my non-multicast registration server, I need to
know my identity (see discussion above).

Thus, overall, I see DHCP used for things that are needed to bootstrap
into the SIP infrastructure with minimum cost and without creating
seventeen new directory servers. Almost all SOHO-style integrated
devices support DHCP, for obvious reasons, while installing and
configuring LDAP, say, seems like a stretch for this type of
application. (You don't mention a specific directory service, so pardon
me if I'm misinterpreting your remarks.)

Overall, the larger the number of servers, the higher the probability
that one of the alphabet soup servers is down, misconfigured. In my
view, it would be nice if I can install a basic Internet PBX with two
servers, a DHCP server and a SIP server. These seem to be required,
adding anything else as a basic requirement to make phones ring seems
like it needs a fair amount of justification for the complexity. This
does not mean that more complicated scenarios can't make good use of SLP
or LDAP. (We use LDAP in our SIP server and PC client, for example.)

> 
>                 Mike

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



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From: James Kempf <James.Kempf@Eng.Sun.COM>
Reply-To: James Kempf <James.Kempf@Eng.Sun.COM>
Subject: Re: DHCP option tags for SIP
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>1) if I have a residential gateway, say, which runs 
>   SIP on its two POTS lines, with SLP can I find 
>   the (different) first hop proxies for each port? 

I'm not quite sure I understand the scenerio (I'm new to SIP, sorry).
What is looking for the first hop proxy, the residental gateway?
My feeling is that the answer to this must be "no" because
if the gateway is sitting on a POTS line, then the protocol on
the POTS line must be something other than IP. Or do you mean
having a phone in the house find the first hop proxy which would
be the residential gateway? If that's the case, then because
the scenerio states that there are different first hop proxies,
there must be two residential gateways, or? If that's the case,
then the phone could select a gateway using SLP, if the service
advertisements for the gateways distinguished them in some way.

>2) Suppose I have some IP phones at home which are
>   gateway'd through my home SIP proxy which allow
>   me to set up local dial plans. Would SLP be adequate
>   to configure the 1) phones 2) my home proxy for learning
>   their next hope proxy?
>

Let me speculate about the intent of this question (please correct
me if I'm wrong). Is the intent of this question to see whether
SLP can configure a home phone with a custom dial plan? If so,
then I would say that that is probably not the right use for
SLP. I would venture to guess that would involve some kind of
negotiation with the telephone company's subscriber database.
I would presume a AAA protocol would get involved somehow, to
authenticate your phone for access to the dial plan.
As for 1), this depends on what you want to configure the phones
with. You can't configure them with an IP address, because you
need an IP address to use SLP. You could configure them with
basic stuff like outbound proxy address. Similarly for 2), you could
have the proxy find the next hop proxy using SLP if you wanted.

>I'm not trying to diss the idea, but SLP seems to
>have been written with enterprises in mind --
>which is fine -- but the more complicated
>situations will probably need to be considered
>where we start splitting up the monolithicly
>administered network into various administrative
>domains, and differently administered services.
>

There is nothing specific to the enterprise in SLP's design, though
it is not designed to be used on the Internet (large "I" here).
SLP has some features for home-style networks, like using multicast
in the simplest case. As Jonathan points out, it is primarily of
value when the client has choices for a server. If you take a look
at the SLP template draft we submitted the other day, you can get
an idea of the kinds of attributes we are proposing for distinguishing
servers.

		jak




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Date: Mon, 21 Feb 2000 18:10:32 +0530
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hi All,

The document SIP Caller Preferences and Callee Capabilities -
draft-ietf-sipcallerprefs-00.txt
gives the grammer for Accept-Contact Header as

Accept-Contact   =  "Accept-Contact" ":" 1# rule
   rule             =  ( name-addr | addr-spec )
                       [ *( ";" (cp-params | extension-param | q-param) ) ]
   q-param          =  "q" "=" qvalue
   extension-param  =  extension-name "=" extension-value
   extension-name   =  token
   extension-value  =  token | quoted-string

and the example given is

Accept-Contact: sip:sales@acme.com ;q=0,
     ;media="!video" ;q=0.1,
     ;mobility="fixed"  ;q=0.6,
     ;mobility="!fixed" ;q=0.4

According to my understanding, in this example each of the comma  separated
values is a separate Accept Contact rule,
And the parameters of a paticular accept contact rule are separated by ";"
semicolon.
Here each of the q-params are followed by a comma and then a semi-colon .

I cannot understand whether to consider this as
1) a single accept contact rule with all the entries following
sip:sales@acme.com as parameters in which case I dont see a need for a
comma and semi-colon as separaters between parameters.

2) 4 rules where-in the 2nd 3rd and 4th do not have a nameaddr|addrspec
entry which is also an error, becos the rule mandates the presence of
nameaddr|addr-spec.

Can someone explain this to me.

Regards,
Preethy







From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 21 08:19:49 2000
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From: Francois Menard List Account <fm-listproc@fmmo.ca>
To: James Kempf <James.Kempf@Eng.Sun.COM>
Cc: mat@cisco.com, sip@lists.research.bell-labs.com, etremblay@mediatrix.com
Subject: Re: DHCP option tags for SIP
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The question has to do with generic means for capabilities exchange
between two devices, using a common set of semantics.

I can easily see how an XML DTD for a specific industry (i.e. in some
cases Internet Telephony for SIP), could be used to agree on capabilities  
parameters of other UA's, Proxy Servers, PSTN Gateways & RTP Mixers.

In this sense, and to answer some private concerns of others, I feel that
SLP is a better end-to-end protocol than DHCP in order to perform these
tasks.  I dont like the approach, of "if all you want to do is pass on the
IP address of the first Proxy Server, then DHCP is fine". To me this
violates the notion of end-to-end capabilities exchange (and gives all the
power to a mighty-god Proxy server). 

To this end, my proposal for an SLP DTD for SIP appliances could 
resembles the Call Processing Language.  Maybe the XML DTD
for CPL (with some tweaks) could be used for caps exchange using SLP.

For example, Mediatrix has a device which is very unique, i.e. a VoIP
gateway with more RTP mixing capabilties than end-points, 3 fxs ports and
one fxs/fxo port.  Being able to advertise this functionality through SLP
is a very unique situation.

Hence my warm welcome of SLP to the SIP scene.  Its simply better than all
those nasty broadcast packets on LANs using DHCP.

-=Francois=-


On Sun, 20 Feb 2000, James Kempf wrote:

> 
> >1) if I have a residential gateway, say, which runs 
> >   SIP on its two POTS lines, with SLP can I find 
> >   the (different) first hop proxies for each port? 
> 
> I'm not quite sure I understand the scenerio (I'm new to SIP, sorry).
> What is looking for the first hop proxy, the residental gateway?
> My feeling is that the answer to this must be "no" because
> if the gateway is sitting on a POTS line, then the protocol on
> the POTS line must be something other than IP. Or do you mean
> having a phone in the house find the first hop proxy which would
> be the residential gateway? If that's the case, then because
> the scenerio states that there are different first hop proxies,
> there must be two residential gateways, or? If that's the case,
> then the phone could select a gateway using SLP, if the service
> advertisements for the gateways distinguished them in some way.
> 
> >2) Suppose I have some IP phones at home which are
> >   gateway'd through my home SIP proxy which allow
> >   me to set up local dial plans. Would SLP be adequate
> >   to configure the 1) phones 2) my home proxy for learning
> >   their next hope proxy?
> >
> 
> Let me speculate about the intent of this question (please correct
> me if I'm wrong). Is the intent of this question to see whether
> SLP can configure a home phone with a custom dial plan? If so,
> then I would say that that is probably not the right use for
> SLP. I would venture to guess that would involve some kind of
> negotiation with the telephone company's subscriber database.
> I would presume a AAA protocol would get involved somehow, to
> authenticate your phone for access to the dial plan.
> As for 1), this depends on what you want to configure the phones
> with. You can't configure them with an IP address, because you
> need an IP address to use SLP. You could configure them with
> basic stuff like outbound proxy address. Similarly for 2), you could
> have the proxy find the next hop proxy using SLP if you wanted.
> 
> >I'm not trying to diss the idea, but SLP seems to
> >have been written with enterprises in mind --
> >which is fine -- but the more complicated
> >situations will probably need to be considered
> >where we start splitting up the monolithicly
> >administered network into various administrative
> >domains, and differently administered services.
> >
> 
> There is nothing specific to the enterprise in SLP's design, though
> it is not designed to be used on the Internet (large "I" here).
> SLP has some features for home-style networks, like using multicast
> in the simplest case. As Jonathan points out, it is primarily of
> value when the client has choices for a server. If you take a look
> at the SLP template draft we submitted the other day, you can get
> an idea of the kinds of attributes we are proposing for distinguishing
> servers.
> 
> 		jak
> 
> 




From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 21 10:40:28 2000
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Cc: James Kempf <James.Kempf@Eng.Sun.COM>, mat@cisco.com,
        sip@lists.research.bell-labs.com, etremblay@mediatrix.com
Subject: Re: DHCP option tags for SIP
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I'd like to try and keep this discussion focused in order to quickly
answer the following question:

  Should we "fast track" a DHCP options for finding a SIP server
document?

Neither DHCP nor SLP are capable of providing complicated application
layer configuration parameters, such as dialing plans or addresses.
Neither are good for capabilities exchange, which is something different
altogether IMHO. With DHCP, you can at least find the server to register
to, and that may provide much of this information in registration
responses. Certainly, it doesn't preclude using DHCP to point to a
specialized configuration server down the road either.

Both DHCP and SLP will do the same thing - get you the adress of a
server. With SLP, you can be picky about which server you want, based on
its capabilities (like ability to support CPL). I see roles for both.

-Jonathan R.

Francois Menard List Account wrote:
> 
> The question has to do with generic means for capabilities exchange
> between two devices, using a common set of semantics.
> 
> I can easily see how an XML DTD for a specific industry (i.e. in some
> cases Internet Telephony for SIP), could be used to agree on capabilities
> parameters of other UA's, Proxy Servers, PSTN Gateways & RTP Mixers.
> 
> In this sense, and to answer some private concerns of others, I feel that
> SLP is a better end-to-end protocol than DHCP in order to perform these
> tasks.  I dont like the approach, of "if all you want to do is pass on the
> IP address of the first Proxy Server, then DHCP is fine". To me this
> violates the notion of end-to-end capabilities exchange (and gives all the
> power to a mighty-god Proxy server).
> 
> To this end, my proposal for an SLP DTD for SIP appliances could
> resembles the Call Processing Language.  Maybe the XML DTD
> for CPL (with some tweaks) could be used for caps exchange using SLP.
> 
> For example, Mediatrix has a device which is very unique, i.e. a VoIP
> gateway with more RTP mixing capabilties than end-points, 3 fxs ports and
> one fxs/fxo port.  Being able to advertise this functionality through SLP
> is a very unique situation.
> 
> Hence my warm welcome of SLP to the SIP scene.  Its simply better than all
> those nasty broadcast packets on LANs using DHCP.
> 
> -=Francois=-
> 
> On Sun, 20 Feb 2000, James Kempf wrote:
> 
> >
> > >1) if I have a residential gateway, say, which runs
> > >   SIP on its two POTS lines, with SLP can I find
> > >   the (different) first hop proxies for each port?
> >
> > I'm not quite sure I understand the scenerio (I'm new to SIP, sorry).
> > What is looking for the first hop proxy, the residental gateway?
> > My feeling is that the answer to this must be "no" because
> > if the gateway is sitting on a POTS line, then the protocol on
> > the POTS line must be something other than IP. Or do you mean
> > having a phone in the house find the first hop proxy which would
> > be the residential gateway? If that's the case, then because
> > the scenerio states that there are different first hop proxies,
> > there must be two residential gateways, or? If that's the case,
> > then the phone could select a gateway using SLP, if the service
> > advertisements for the gateways distinguished them in some way.
> >
> > >2) Suppose I have some IP phones at home which are
> > >   gateway'd through my home SIP proxy which allow
> > >   me to set up local dial plans. Would SLP be adequate
> > >   to configure the 1) phones 2) my home proxy for learning
> > >   their next hope proxy?
> > >
> >
> > Let me speculate about the intent of this question (please correct
> > me if I'm wrong). Is the intent of this question to see whether
> > SLP can configure a home phone with a custom dial plan? If so,
> > then I would say that that is probably not the right use for
> > SLP. I would venture to guess that would involve some kind of
> > negotiation with the telephone company's subscriber database.
> > I would presume a AAA protocol would get involved somehow, to
> > authenticate your phone for access to the dial plan.
> > As for 1), this depends on what you want to configure the phones
> > with. You can't configure them with an IP address, because you
> > need an IP address to use SLP. You could configure them with
> > basic stuff like outbound proxy address. Similarly for 2), you could
> > have the proxy find the next hop proxy using SLP if you wanted.
> >
> > >I'm not trying to diss the idea, but SLP seems to
> > >have been written with enterprises in mind --
> > >which is fine -- but the more complicated
> > >situations will probably need to be considered
> > >where we start splitting up the monolithicly
> > >administered network into various administrative
> > >domains, and differently administered services.
> > >
> >
> > There is nothing specific to the enterprise in SLP's design, though
> > it is not designed to be used on the Internet (large "I" here).
> > SLP has some features for home-style networks, like using multicast
> > in the simplest case. As Jonathan points out, it is primarily of
> > value when the client has choices for a server. If you take a look
> > at the SLP template draft we submitted the other day, you can get
> > an idea of the kinds of attributes we are proposing for distinguishing
> > servers.
> >
> >               jak
> >
> >

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Thanks for pointing this out. The bnf appears to be wrong, as it
mandates a valid name-addr or addr-spec. The document clearly discusses
the capability of not specifying one. I will fix in the next version
(due out before the draft deadline). The BNF should be:

Accept-Contact = "Accept-Contact" ":" 1#rule
rule   =  (name-addr | addr-spec | "*")
....


So, we'll use the * for anything, aligning with current Contact usage.

-Jonathan R.

rpreethy@hss.hns.com wrote:
> 
> hi All,
> 
> The document SIP Caller Preferences and Callee Capabilities -
> draft-ietf-sipcallerprefs-00.txt
> gives the grammer for Accept-Contact Header as
> 
> Accept-Contact   =  "Accept-Contact" ":" 1# rule
>    rule             =  ( name-addr | addr-spec )
>                        [ *( ";" (cp-params | extension-param | q-param) ) ]
>    q-param          =  "q" "=" qvalue
>    extension-param  =  extension-name "=" extension-value
>    extension-name   =  token
>    extension-value  =  token | quoted-string
> 
> and the example given is
> 
> Accept-Contact: sip:sales@acme.com ;q=0,
>      ;media="!video" ;q=0.1,
>      ;mobility="fixed"  ;q=0.6,
>      ;mobility="!fixed" ;q=0.4
> 
> According to my understanding, in this example each of the comma  separated
> values is a separate Accept Contact rule,
> And the parameters of a paticular accept contact rule are separated by ";"
> semicolon.
> Here each of the q-params are followed by a comma and then a semi-colon .
> 
> I cannot understand whether to consider this as
> 1) a single accept contact rule with all the entries following
> sip:sales@acme.com as parameters in which case I dont see a need for a
> comma and semi-colon as separaters between parameters.
> 
> 2) 4 rules where-in the 2nd 3rd and 4th do not have a nameaddr|addrspec
> entry which is also an error, becos the rule mandates the presence of
> nameaddr|addr-spec.
> 
> Can someone explain this to me.
> 
> Regards,
> Preethy

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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From: "Glenn Morrow" <gmorrow@nortelnetworks.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        Francois Menard List Account <fm-listproc@fmmo.ca>
Cc: James Kempf <James.Kempf@Eng.Sun.COM>, mat@cisco.com,
        sip@lists.research.bell-labs.com, etremblay@mediatrix.com
Subject: RE: DHCP option tags for SIP
Date: Mon, 21 Feb 2000 10:15:50 -0600
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Why can't the downloaded UAS, by bootp or other means, simply have the
configured proxy address? The configuration can be changed later with SIP
signaling without reboot for the in-service cases.

This approach requires no changes to DHCP or SIP and appears to meet the
requirements. 

If we do go forward with these changes, what do think SG-16 will want to do
in another month?

- another damp blanket.


> -----Original Message-----
> From:	Jonathan Rosenberg [SMTP:jdrosen@dynamicsoft.com]
> Sent:	Monday, February 21, 2000 9:36 AM
> To:	Francois Menard List Account
> Cc:	James Kempf; mat@cisco.com; sip@lists.research.bell-labs.com;
> etremblay@mediatrix.com
> Subject:	Re: DHCP option tags for SIP
> 
> I'd like to try and keep this discussion focused in order to quickly
> answer the following question:
> 
>   Should we "fast track" a DHCP options for finding a SIP server
> document?
> 
> Neither DHCP nor SLP are capable of providing complicated application
> layer configuration parameters, such as dialing plans or addresses.
> Neither are good for capabilities exchange, which is something different
> altogether IMHO. With DHCP, you can at least find the server to register
> to, and that may provide much of this information in registration
> responses. Certainly, it doesn't preclude using DHCP to point to a
> specialized configuration server down the road either.
> 
> Both DHCP and SLP will do the same thing - get you the adress of a
> server. With SLP, you can be picky about which server you want, based on
> its capabilities (like ability to support CPL). I see roles for both.
> 
> -Jonathan R.
> 
> Francois Menard List Account wrote:
> > 
> > The question has to do with generic means for capabilities exchange
> > between two devices, using a common set of semantics.
> > 
> > I can easily see how an XML DTD for a specific industry (i.e. in some
> > cases Internet Telephony for SIP), could be used to agree on
> capabilities
> > parameters of other UA's, Proxy Servers, PSTN Gateways & RTP Mixers.
> > 
> > In this sense, and to answer some private concerns of others, I feel
> that
> > SLP is a better end-to-end protocol than DHCP in order to perform these
> > tasks.  I dont like the approach, of "if all you want to do is pass on
> the
> > IP address of the first Proxy Server, then DHCP is fine". To me this
> > violates the notion of end-to-end capabilities exchange (and gives all
> the
> > power to a mighty-god Proxy server).
> > 
> > To this end, my proposal for an SLP DTD for SIP appliances could
> > resembles the Call Processing Language.  Maybe the XML DTD
> > for CPL (with some tweaks) could be used for caps exchange using SLP.
> > 
> > For example, Mediatrix has a device which is very unique, i.e. a VoIP
> > gateway with more RTP mixing capabilties than end-points, 3 fxs ports
> and
> > one fxs/fxo port.  Being able to advertise this functionality through
> SLP
> > is a very unique situation.
> > 
> > Hence my warm welcome of SLP to the SIP scene.  Its simply better than
> all
> > those nasty broadcast packets on LANs using DHCP.
> > 
> > -=Francois=-
> > 
> > On Sun, 20 Feb 2000, James Kempf wrote:
> > 
> > >
> > > >1) if I have a residential gateway, say, which runs
> > > >   SIP on its two POTS lines, with SLP can I find
> > > >   the (different) first hop proxies for each port?
> > >
> > > I'm not quite sure I understand the scenerio (I'm new to SIP, sorry).
> > > What is looking for the first hop proxy, the residental gateway?
> > > My feeling is that the answer to this must be "no" because
> > > if the gateway is sitting on a POTS line, then the protocol on
> > > the POTS line must be something other than IP. Or do you mean
> > > having a phone in the house find the first hop proxy which would
> > > be the residential gateway? If that's the case, then because
> > > the scenerio states that there are different first hop proxies,
> > > there must be two residential gateways, or? If that's the case,
> > > then the phone could select a gateway using SLP, if the service
> > > advertisements for the gateways distinguished them in some way.
> > >
> > > >2) Suppose I have some IP phones at home which are
> > > >   gateway'd through my home SIP proxy which allow
> > > >   me to set up local dial plans. Would SLP be adequate
> > > >   to configure the 1) phones 2) my home proxy for learning
> > > >   their next hope proxy?
> > > >
> > >
> > > Let me speculate about the intent of this question (please correct
> > > me if I'm wrong). Is the intent of this question to see whether
> > > SLP can configure a home phone with a custom dial plan? If so,
> > > then I would say that that is probably not the right use for
> > > SLP. I would venture to guess that would involve some kind of
> > > negotiation with the telephone company's subscriber database.
> > > I would presume a AAA protocol would get involved somehow, to
> > > authenticate your phone for access to the dial plan.
> > > As for 1), this depends on what you want to configure the phones
> > > with. You can't configure them with an IP address, because you
> > > need an IP address to use SLP. You could configure them with
> > > basic stuff like outbound proxy address. Similarly for 2), you could
> > > have the proxy find the next hop proxy using SLP if you wanted.
> > >
> > > >I'm not trying to diss the idea, but SLP seems to
> > > >have been written with enterprises in mind --
> > > >which is fine -- but the more complicated
> > > >situations will probably need to be considered
> > > >where we start splitting up the monolithicly
> > > >administered network into various administrative
> > > >domains, and differently administered services.
> > > >
> > >
> > > There is nothing specific to the enterprise in SLP's design, though
> > > it is not designed to be used on the Internet (large "I" here).
> > > SLP has some features for home-style networks, like using multicast
> > > in the simplest case. As Jonathan points out, it is primarily of
> > > value when the client has choices for a server. If you take a look
> > > at the SLP template draft we submitted the other day, you can get
> > > an idea of the kinds of attributes we are proposing for distinguishing
> > > servers.
> > >
> > >               jak
> > >
> > >
> 
> -- 
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120 
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 

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<TITLE>RE: DHCP option tags for SIP</TITLE>
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<BODY>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Why can't the =
downloaded UAS, by bootp or other means, simply have the configured =
proxy address? The configuration can be changed later with SIP =
signaling without reboot for the in-service cases.</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">This approach =
requires no changes to DHCP or SIP and appears to meet the =
requirements. </FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">If we do go forward =
with these changes, what do think SG-16 will want to do in another =
month?</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">- another damp =
blanket.</FONT>
</P>
<BR>
<UL>
<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">Jonathan Rosenberg =
[SMTP:jdrosen@dynamicsoft.com]</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Monday, February 21, 2000 9:36 AM</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">Francois Menard List Account</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Cc:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">James Kempf; mat@cisco.com; =
sip@lists.research.bell-labs.com; etremblay@mediatrix.com</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">Re: DHCP option tags for SIP</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I'd like to try and keep this =
discussion focused in order to quickly</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">answer the following question:</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&nbsp; Should we &quot;fast =
track&quot; a DHCP options for finding a SIP server</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">document?</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Neither DHCP nor SLP are capable of =
providing complicated application</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">layer configuration parameters, such =
as dialing plans or addresses.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Neither are good for capabilities =
exchange, which is something different</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">altogether IMHO. With DHCP, you can =
at least find the server to register</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">to, and that may provide much of this =
information in registration</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">responses. Certainly, it doesn't =
preclude using DHCP to point to a</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">specialized configuration server down =
the road either.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Both DHCP and SLP will do the same =
thing - get you the adress of a</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">server. With SLP, you can be picky =
about which server you want, based on</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">its capabilities (like ability to =
support CPL). I see roles for both.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">-Jonathan R.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Francois Menard List Account =
wrote:</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; The question has to do with =
generic means for capabilities exchange</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; between two devices, using a =
common set of semantics.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; I can easily see how an XML DTD =
for a specific industry (i.e. in some</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; cases Internet Telephony for =
SIP), could be used to agree on capabilities</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; parameters of other UA's, Proxy =
Servers, PSTN Gateways &amp; RTP Mixers.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; In this sense, and to answer =
some private concerns of others, I feel that</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; SLP is a better end-to-end =
protocol than DHCP in order to perform these</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; tasks.&nbsp; I dont like the =
approach, of &quot;if all you want to do is pass on the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; IP address of the first Proxy =
Server, then DHCP is fine&quot;. To me this</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; violates the notion of =
end-to-end capabilities exchange (and gives all the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; power to a mighty-god Proxy =
server).</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; To this end, my proposal for an =
SLP DTD for SIP appliances could</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; resembles the Call Processing =
Language.&nbsp; Maybe the XML DTD</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; for CPL (with some tweaks) could =
be used for caps exchange using SLP.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; For example, Mediatrix has a =
device which is very unique, i.e. a VoIP</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; gateway with more RTP mixing =
capabilties than end-points, 3 fxs ports and</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; one fxs/fxo port.&nbsp; Being =
able to advertise this functionality through SLP</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; is a very unique =
situation.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Hence my warm welcome of SLP to =
the SIP scene.&nbsp; Its simply better than all</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; those nasty broadcast packets on =
LANs using DHCP.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; -=3DFrancois=3D-</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; On Sun, 20 Feb 2000, James Kempf =
wrote:</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;1) if I have a =
residential gateway, say, which runs</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;&nbsp;&nbsp; SIP on its =
two POTS lines, with SLP can I find</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;&nbsp;&nbsp; the =
(different) first hop proxies for each port?</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; I'm not quite sure I =
understand the scenerio (I'm new to SIP, sorry).</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; What is looking for the =
first hop proxy, the residental gateway?</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; My feeling is that the =
answer to this must be &quot;no&quot; because</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; if the gateway is sitting =
on a POTS line, then the protocol on</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; the POTS line must be =
something other than IP. Or do you mean</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; having a phone in the house =
find the first hop proxy which would</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; be the residential gateway? =
If that's the case, then because</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; the scenerio states that =
there are different first hop proxies,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; there must be two =
residential gateways, or? If that's the case,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; then the phone could select =
a gateway using SLP, if the service</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; advertisements for the =
gateways distinguished them in some way.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;2) Suppose I have some =
IP phones at home which are</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;&nbsp;&nbsp; gateway'd =
through my home SIP proxy which allow</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;&nbsp;&nbsp; me to set =
up local dial plans. Would SLP be adequate</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;&nbsp;&nbsp; to =
configure the 1) phones 2) my home proxy for learning</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;&nbsp;&nbsp; their next =
hope proxy?</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; Let me speculate about the =
intent of this question (please correct</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; me if I'm wrong). Is the =
intent of this question to see whether</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; SLP can configure a home =
phone with a custom dial plan? If so,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; then I would say that that =
is probably not the right use for</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; SLP. I would venture to =
guess that would involve some kind of</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; negotiation with the =
telephone company's subscriber database.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; I would presume a AAA =
protocol would get involved somehow, to</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; authenticate your phone for =
access to the dial plan.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; As for 1), this depends on =
what you want to configure the phones</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; with. You can't configure =
them with an IP address, because you</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; need an IP address to use =
SLP. You could configure them with</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; basic stuff like outbound =
proxy address. Similarly for 2), you could</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; have the proxy find the =
next hop proxy using SLP if you wanted.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;I'm not trying to diss =
the idea, but SLP seems to</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;have been written with =
enterprises in mind --</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;which is fine -- but =
the more complicated</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;situations will =
probably need to be considered</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;where we start =
splitting up the monolithicly</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;administered network =
into various administrative</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;domains, and =
differently administered services.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; There is nothing specific =
to the enterprise in SLP's design, though</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; it is not designed to be =
used on the Internet (large &quot;I&quot; here).</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; SLP has some features for =
home-style networks, like using multicast</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; in the simplest case. As =
Jonathan points out, it is primarily of</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; value when the client has =
choices for a server. If you take a look</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; at the SLP template draft =
we submitted the other day, you can get</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; an idea of the kinds of =
attributes we are proposing for distinguishing</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; servers.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp; jak</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt;</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">-- </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Jonathan D. =
Rosenberg&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
200 Executive Drive</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Chief =
Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Suite 120 </FONT>
<BR><FONT SIZE=3D2 =
FACE=3D"Arial">dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; West Orange, NJ 07052</FONT>
<BR><FONT SIZE=3D2 =
FACE=3D"Arial">jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp; FAX:&nbsp;&nbsp; (732) 741-4778</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://www.cs.columbia.edu/~jdrosen" =
TARGET=3D"_blank">http://www.cs.columbia.edu/~jdrosen</A></FONT></U><FON=
T SIZE=3D2 =
FACE=3D"Arial">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: =
(732) 741-7244</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://www.dynamicsoft.com" =
TARGET=3D"_blank">http://www.dynamicsoft.com</A></FONT></U>
</P>
</UL>
</BODY>
</HTML>
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To: Glenn Morrow <gmorrow@nortelnetworks.com>
Cc: Francois Menard List Account <fm-listproc@fmmo.ca>,
        James Kempf <James.Kempf@Eng.Sun.COM>, mat@cisco.com,
        sip@lists.research.bell-labs.com, etremblay@mediatrix.com
Subject: Re: DHCP option tags for SIP
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> Glenn Morrow wrote:
> 
> Why can't the downloaded UAS, by bootp or other means, simply have the
> configured proxy address? The configuration can be changed later with
> SIP signaling without reboot for the in-service cases.

The place I download my client from is generally unrelated to my service
provider (I downloaded Netscape from Netscapes site... has nothing to do
with my ISP). 

Even presuming you did this, how does one change this later with SIP
signaling? Changing this is exactly what DHCP or SLP is desired for...

> If we do go forward with these changes, what do think SG-16 will want
> to do in another month?

I'm not sure I understand this... is SG16 planning some kind of activity
in this area (SIP configuration)?

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 21 11:47:36 2000
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, rpreethy@hss.hns.com
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Subject: Re: Accept-Contact Header
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Just wanted to point out another error in this draft:

The proposed matching algorithm is not correct. Consider for example if
we have a contact entry that is language="en,ge" and the rule is
language="en,nl". Certainly we want this to be a match, since both
parties speak one common language. But the rule is clearly not a subset
of the entry . In the draft there is an example, which is not accurate
following the above algorithm:

;duplex="full,half"
matches the contact entry
;duplex="full"


As we see the rule set is clearly not a subset of the entry set (rather
a superset) and according to the proposed algorithm there should not be
a match. But we want it to be a match (as stated in the example, and in
the text above). The algorithm may instead be changed to the following:

- The parameter name is first checked. If it is not defined in the
contact entry, the   parameter is ignored.
- Parameter values are treated as sets, so the intersection of the rule
set and the entry set   for this parameter is checked.
- If the intersection is non-empty and the rule set is not negated we
have a match.
- If the intersection is empty and the rule set is negated we also have
a match.


regards
//Anders 

Jonathan Rosenberg wrote:
> 
> Thanks for pointing this out. The bnf appears to be wrong, as it
> mandates a valid name-addr or addr-spec. The document clearly discusses
> the capability of not specifying one. I will fix in the next version
> (due out before the draft deadline). The BNF should be:
> 
> Accept-Contact = "Accept-Contact" ":" 1#rule
> rule   =  (name-addr | addr-spec | "*")
> ....
> 
> So, we'll use the * for anything, aligning with current Contact usage.
> 
> -Jonathan R.
> 
> rpreethy@hss.hns.com wrote:
> >
> > hi All,
> >
> > The document SIP Caller Preferences and Callee Capabilities -
> > draft-ietf-sipcallerprefs-00.txt
> > gives the grammer for Accept-Contact Header as
> >
> > Accept-Contact   =  "Accept-Contact" ":" 1# rule
> >    rule             =  ( name-addr | addr-spec )
> >                        [ *( ";" (cp-params | extension-param | q-param) ) ]
> >    q-param          =  "q" "=" qvalue
> >    extension-param  =  extension-name "=" extension-value
> >    extension-name   =  token
> >    extension-value  =  token | quoted-string
> >
> > and the example given is
> >
> > Accept-Contact: sip:sales@acme.com ;q=0,
> >      ;media="!video" ;q=0.1,
> >      ;mobility="fixed"  ;q=0.6,
> >      ;mobility="!fixed" ;q=0.4
> >
> > According to my understanding, in this example each of the comma  separated
> > values is a separate Accept Contact rule,
> > And the parameters of a paticular accept contact rule are separated by ";"
> > semicolon.
> > Here each of the q-params are followed by a comma and then a semi-colon .
> >
> > I cannot understand whether to consider this as
> > 1) a single accept contact rule with all the entries following
> > sip:sales@acme.com as parameters in which case I dont see a need for a
> > comma and semi-colon as separaters between parameters.
> >
> > 2) 4 rules where-in the 2nd 3rd and 4th do not have a nameaddr|addrspec
> > entry which is also an error, becos the rule mandates the presence of
> > nameaddr|addr-spec.
> >
> > Can someone explain this to me.
> >
> > Regards,
> > Preethy
> 
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com

-- 
--------------------------------------------------------------------------
Anders Byttner 				Email:    <andbyt@hplb.hpl.hp.com>
Hewlett-Packard Laboratories		Cellular: +44-(0)7901-805 479
Internet Communication 			Work:     +44-(0)117-312 96 03

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To: gmorrow@nortelnetworks.com, jdrosen@dynamicsoft.com
Subject: Re: DHCP option tags for SIP
Cc: fm-listproc@fmmo.ca, James.Kempf@Eng.Sun.COM, mat@cisco.com,
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> > Glenn Morrow wrote:
> > 
> > Why can't the downloaded UAS, by bootp or other means, simply have the
> > configured proxy address? The configuration can be changed later with
> > SIP signaling without reboot for the in-service cases.
> 

Hey, we could even use a multicast/broadcast SIP OPTIONS request to query
local SIP servers and part of the response could be the parameters people
have been discussing like first SIP proxy hop. There are a lot of 
possibilities here. I haven't seen a good argument against simply including
the SIP proxy as a DHCP option yet. If you are allowing for BOOTP, it's a
short step to go to DHCP and if you allow DHCP then why not exploit its
capabilities? I see the advantage of SLP over DHCP, but its clearly overkill
for a simple SIP phone especially if you need to implement DHCP anyways.

my two cents
sean
--
Sean Olson <sean.olson@ericsson.com>
Ericsson Inc.
tel:972-583-5472





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From: "Ashutosh Dutta" <adutta@telcordia.com>
To: Sean Olson <eussean@exu.ericsson.se>
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For a wireless roaming user (e.g. moving between cells, subnets, and domains)
where fast handoff is an issue during an ongoing call, DHCP option tag for
finding out the nearest SIP proxy would be also very useful.

Thanks
Ashutosh





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Sean Olson wrote:
> 
> > > Glenn Morrow wrote:
> > >
> > > Why can't the downloaded UAS, by bootp or other means, simply have the
> > > configured proxy address? The configuration can be changed later with
> > > SIP signaling without reboot for the in-service cases.
> >
> 
> Hey, we could even use a multicast/broadcast SIP OPTIONS request to query
> local SIP servers and part of the response could be the parameters people
> have been discussing like first SIP proxy hop. There are a lot of
> possibilities here. I haven't seen a good argument against simply including
> the SIP proxy as a DHCP option yet. If you are allowing for BOOTP, it's a
> short step to go to DHCP and if you allow DHCP then why not exploit its
> capabilities? I see the advantage of SLP over DHCP, but its clearly overkill
> for a simple SIP phone especially if you need to implement DHCP anyways.

While I agree that multicast is a nice mechanism for some
autoconfig/discovery, the problem is that in many cases the outbound
proxy server may not be part of the local network. I don't think, for
example, that you'd want to have all 20,000 phones at Columbia send out
multicast requests, with all other phones having to listen to them,
particularly if they happen to reboot at roughly the same time, as after
a power outage.

Otherwise, we seem to be in agreement...

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



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Anders Kristensen wrote:
> 
> I'll go through these in turns.
> 
>   1. Define Supported in the base spec.
> 
> The draft can be seen as having two parts: the definition of the
> Supported header and a proposal for a feature negotiation mechanism. I
> think the two should be separated. It makes perfectly good sense for a
> server to signal supported features in an OPTIONs response without
> having to implement the negotiation mechanism, and likewise, clients
> should be allowed to indicate that they support, say, reliable 1xx's
> without having to implement the negotiation mechanism.

I agree that Supported has uses outside of the negotiation mechanism. In
fact,
rfc2543bis has a definition of Supported.

However, you cannot separate the header from the protocol semantics and
exchanges it is used in. To date, there are two applications:

1. Returned in response to OPTIONS to indicate capabilities (this is
what is described in rfc2543bis)
2. the negotiation mechanism in the draft

So, I would argue that (1) is something which might well be included in
the next version of the base spec - its completely backwards compatible
and easy, and well in line with OPTIONS operation now. The other use,
(2), is new and different. So, where does the SYNTAX itself of Supported
get defined? Doesn't really matter... may very well be both.


>   2. Do *not* let presence of the Supported header implicitly
>      signal support for the extension negotiation mechanism.
> 
> This is a corollory of the previous argument. If UAs can list supported
> features independent of the negotiation mechanism then presence of the
> Supported header cannot be taken to implicitly indicate support for the
> latter.

Again, you can't separate the header from the protocol behavior. I would
agree that adding Supported to an OPTIONS response does not imply
understanding of the negotiation mechanism. However, placement of
Supported in a request is strongly coupled with the negotiation
mechanism, and I would be reluctant to allow it without the full thing.
So, in the interests of short fields (which you argue for below), I
would prefer to allow the presence of Supported alone in a request be
used to indicate this.

> 
>   3. Drop the reversed domain name feature naming convention.
> 
> It makes messages longer and is unnecessary.
> 
> The (primary) justifiction for the extension negotiation mechanism is
> that servers need to know what features are supported by clients. It
> seems to me that the simplest way to meet this requirement is for
> clients to always tell servers what features they support (and BTW this
> is another argument for defining Supported in the base spec).
> 
> Now, the reason why this answer is not enough must be the potentially
> large overhead of listing all these extensions all the time. Hence SIZE
> MATTERS! Anything we can do to minimize the size overhead of signaling
> supported features makes it more practical for us to use the simpler and
> more efficient mechanism of just listing all supported features always.
> The use of the short header form and the proposed implicit serverfeature
> support is further testament to the need to keep messages short. There
> may be times where it really isn't practical to list all supported
> extensions, and for these cases we still need the negotiation.
> 
> For this reason, and *because it's unnecessary*, I propose we drop the
> convention of using reversed domain name prefixes as namespace
> identifiers for extension tags.  First, as I've mused on a previous
> occasion, the vast majority (if not all) extensions to come into
> widespread, regular use will belong to the org.ietf.sip namespace
> anyway, making this prefix redundant, and secondly, I don't even think
> we need these namespaces anyway. We don't have namespaces for header
> names or status codes, and MIME has gotten along just fine for years
> without namespaces for its media types (of which there will probably
> always be many more than there will be SIP extensions). We could use the
> "x-" prefix for experimental or proprietary extensions or continue to
> use the domain namespaces for this purpose.

OK.... I'll buy on this one. However, I'd still like the base spec to
define usage of the domain-based namespaces for unregistered extensions.

> 
>   4. Rename the negotiation extension.
> 
> Follows from 1. and 4. Maybe just call the extension "negotiate".

Rename the draft? The feature tag?

> 
>   5. Define "all-extensions-listed" tag to be used in Supported
>      header.
> 
> In order to avoid unnecessary round-trips it would be good if clients
> can indicate that they have listed *all* supported extensions in the
> Supported header. This can be done by defining a new token with that
> meaning. I'd propose using "." since the semantics are kind of like that
> of "."s use in natural written language (and tokens doesn't come much
> shorter).  So, for example
> 
>   Supported: negotiate, 100rel, .
> 
> would indicate that the sender supports the 100rel and negotiate
> extensions and *no* other extension. Hence there is no point for the
> server to ask whether extension "foo" is supported - it isn't. This tag
> can be seen as just identifying an extension of its own, and could be
> defined in its own document, or it could go into the base spec. I'd
> prefer the latter, since it seems like a generally very useful feature.

Very good idea. I like this a lot. The period is nice too since its got
that
email connotation of "thats all". We must make sure implementations are
prepared
to not receive it last, though, since ordering is not important among
feature tags.


> My 2 pence worths.
> 

I thank you for your insightful comments. I apologize for the delay in
responding... I'd like to get this whole thing closed soon so we can
move on to other things.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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From: "Glenn Morrow" <gmorrow@nortelnetworks.com>
To: "'Henning Schulzrinne'" <schulzrinne@cs.columbia.edu>,
        Sean Olson <eussean@exu.ericsson.se>
Cc: sip@lists.research.bell-labs.com
Subject: RE: DHCP option tags for SIP
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While much of the free router code and some early commercial implementations
only allow for local interfaces for multicast addresses, many new commercial
implementations offer more route type options for these addresses. 

The local route concern is not an issue for commercial implementations and
there are also ways to scale in an n to n fashion using any type of address
(multicast, anycast or even unicast). 

These methods are needed to account for maintenance and other service
disruptions, as well.


Thank You

Glenn

> -----Original Message-----
> From:	Henning Schulzrinne [SMTP:schulzrinne@cs.columbia.edu]
> Sent:	Monday, February 21, 2000 12:50 PM
> To:	Sean Olson
> Cc:	sip@lists.research.bell-labs.com
> Subject:	Re: DHCP option tags for SIP
> 
> Sean Olson wrote:
> > 
> > > > Glenn Morrow wrote:
> > > >
> > > > Why can't the downloaded UAS, by bootp or other means, simply have
> the
> > > > configured proxy address? The configuration can be changed later
> with
> > > > SIP signaling without reboot for the in-service cases.
> > >
> > 
> > Hey, we could even use a multicast/broadcast SIP OPTIONS request to
> query
> > local SIP servers and part of the response could be the parameters
> people
> > have been discussing like first SIP proxy hop. There are a lot of
> > possibilities here. I haven't seen a good argument against simply
> including
> > the SIP proxy as a DHCP option yet. If you are allowing for BOOTP, it's
> a
> > short step to go to DHCP and if you allow DHCP then why not exploit its
> > capabilities? I see the advantage of SLP over DHCP, but its clearly
> overkill
> > for a simple SIP phone especially if you need to implement DHCP anyways.
> 
> While I agree that multicast is a nice mechanism for some
> autoconfig/discovery, the problem is that in many cases the outbound
> proxy server may not be part of the local network. I don't think, for
> example, that you'd want to have all 20,000 phones at Columbia send out
> multicast requests, with all other phones having to listen to them,
> particularly if they happen to reboot at roughly the same time, as after
> a power outage.
> 
> Otherwise, we seem to be in agreement...
> 
> -- 
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> 

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charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2651.65">
<TITLE>RE: DHCP option tags for SIP</TITLE>
</HEAD>
<BODY>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">While much of the =
free router code and some early commercial implementations only allow =
for local interfaces for multicast addresses, many new commercial =
implementations offer more route type options for these addresses. =
</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">The local route =
concern is not an issue for commercial implementations and there are =
also ways to scale in an n to n fashion using any type of address =
(multicast, anycast or even unicast). </FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">These methods are =
needed to account for maintenance and other service disruptions, as =
well.</FONT>
</P>
<BR>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Thank You</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Glenn</FONT>
</P>
<UL>
<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">Henning Schulzrinne =
[SMTP:schulzrinne@cs.columbia.edu]</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Monday, February 21, 2000 12:50 PM</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">Sean Olson</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Cc:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">sip@lists.research.bell-labs.com</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">Re: DHCP option tags for SIP</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Sean Olson wrote:</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; Glenn Morrow =
wrote:</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; Why can't the =
downloaded UAS, by bootp or other means, simply have the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; configured proxy =
address? The configuration can be changed later with</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; SIP signaling without =
reboot for the in-service cases.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Hey, we could even use a =
multicast/broadcast SIP OPTIONS request to query</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; local SIP servers and part of =
the response could be the parameters people</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; have been discussing like first =
SIP proxy hop. There are a lot of</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; possibilities here. I haven't =
seen a good argument against simply including</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; the SIP proxy as a DHCP option =
yet. If you are allowing for BOOTP, it's a</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; short step to go to DHCP and if =
you allow DHCP then why not exploit its</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; capabilities? I see the =
advantage of SLP over DHCP, but its clearly overkill</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; for a simple SIP phone =
especially if you need to implement DHCP anyways.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">While I agree that multicast is a nice =
mechanism for some</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">autoconfig/discovery, the problem is =
that in many cases the outbound</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">proxy server may not be part of the =
local network. I don't think, for</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">example, that you'd want to have all =
20,000 phones at Columbia send out</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">multicast requests, with all other =
phones having to listen to them,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">particularly if they happen to reboot =
at roughly the same time, as after</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">a power outage.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Otherwise, we seem to be in =
agreement...</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">-- </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Henning Schulzrinne&nbsp;&nbsp;<U> =
</U></FONT><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://www.cs.columbia.edu/~hgs" =
TARGET=3D"_blank">http://www.cs.columbia.edu/~hgs</A></FONT></U>
</P>
</UL>
</BODY>
</HTML>
------_=_NextPart_001_01BF7CA1.0709CD48--



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> Glenn Morrow wrote:
> 
> While much of the free router code and some early commercial
> implementations only allow for local interfaces for multicast
> addresses, many new commercial implementations offer more route type
> options for these addresses.

I'm not concerned about availability of multicast, but its scalability
in the current setup. Inherently, the SIP multicast model implies that
every device has to listen to all multicast packets, which just doesn't
scale beyond a fairly small region. In my view, outbound proxy server
often cover a larger region.

> 
> The local route concern is not an issue for commercial implementations
> and there are also ways to scale in an n to n fashion using any type
> of address (multicast, anycast or even unicast).

I'm afraid I lost you somewhere in this sentence.



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



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Jonathan Rosenberg wrote:
> 

> 
> However, you cannot separate the header from the protocol semantics and
> exchanges it is used in. To date, there are two applications:
> 
> 1. Returned in response to OPTIONS to indicate capabilities (this is
> what is described in rfc2543bis)

I see the ability to declare capabilities in *all* requests as an
alternative model that is also backwards-compatible, as long as we
define it before the first capability extension is standardized. (All we
would need to say is that an implementation that supports "foo" also has
to support the "Supported" header, which seems trivial enough.)

There are two models, it seems:

- Reveal all, one round-trip: The client lists all capabilities that it
wants the server to use. Asking again is not going to elicit additional
capabilities.

  +: lot simpler, still backward compatible with existing systems
  +: uniform call setup delay
  -: longer headers (but see below)

- Selective disclosure: Be coy and wait to be asked for particular
features.

  The trade-off is pretty much the opposite as above.

The draft mainly deals with the shy persons. Maybe giving SIP
implementations some powdermilk biscuits might help (don't ask).

In terms of efficiency, it depends on how often a second round-trip is
required. If it is even 10% of the time, I would guess that declaring
all capabilities ahead of time is a good move, both in terms of time and
space (since you need a lot of options to compensate for the 500 bytes
of repeated request, even ignoring the crucial issue of call setup
delay).

Given that, I'm somewhat tempted to explore a different, simpler avenue
initially: assume that the client declares all beyond-RFC2543
capabilities, possibly allow short-hand feature names (as Anders is
suggesting) and wait until we have a better idea as to whether we'll
have 10 or 100 extensions. Clients that have 100 extensions, can offer
the

  btw-I-know-more-than-this-please-ask-again

extension in Supported (with a somewhat shorter name, like "shy"), which
is the current model.

If 100 turns out to be closer to the mark, it might well be that there
will be agreed-upon groups of capabilities, similar to "Host
Requirements" (see the TIA work), shrinking that number to a smaller
number of aliases again.


> 2. the negotiation mechanism in the draft
> 
> So, I would argue that (1) is something which might well be included in
> the next version of the base spec - its completely backwards compatible
> and easy, and well in line with OPTIONS operation now. The other use,
> (2), is new and different. So, where does the SYNTAX itself of Supported
> get defined? Doesn't really matter... may very well be both.

I'd rather have one place. If timing dictates this, we can have a
Proposed Standard for the Supported header syntax and use (nothing else)
pushed out in a matter of weeks, which would then be labeled as
"obsoleted by" RFC2543bis whenever it reaches Draft.


> 
> OK.... I'll buy on this one. However, I'd still like the base spec to
> define usage of the domain-based namespaces for unregistered extensions.

I would turn the "we don't do 'com.hp.header: foo' argument around and
say that if you see a header that looks unfamiliar (and it's important,
i.e., you can't just ignore it), it should show up in the Supported
header, giving others a much smaller number of places to ask and look
around for where this new unfamiliar header might be defined.

Thus, header names are not so much to avoid collisions (those haven't
really been a major problem in HTTP or email - these usually get
discovered early and can be fixed before major damage occurs), but
rather the ability to see who's responsible for a particular feature.

The x- mechanism, by the way, is not very helpful. It tends to stick
around forever (even if the header gets widely used), since nobody wants
to change all those implementations that caused it to be widely use. 

Thus, my overall modified agreement with Anders: org.ietf.sip is implied
(but can be spelled out?) and proprietary extensions need the reverse
path. If the length penalty encourages people to write informational or
standards-track RFCs to describe the feature, I wouldn't mind.

> 
> Very good idea. I like this a lot. The period is nice too since its got
> that
> email connotation of "thats all". We must make sure implementations are
> prepared
> to not receive it last, though, since ordering is not important among
> feature tags.

I would argue that the "I have declared everything unless I put in some
special token" as being a better option, until we have a better idea
what's happening and whether we need a relatively complicated mechanism
to solve what turns out to be a non-problem.

At the rate that we are pushing out extensions, we're a long way of
having to worry about listing them all.



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



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 21 14:58:51 2000
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From: Linden deCarmo <ldeCarmo@netspeak.com>
To: sip@lists.research.bell-labs.com
Subject: Aurthorization & Proxy-Authorization URIs
Date: Mon, 21 Feb 2000 14:53:22 -0500
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The BNF for digest-uri defined by "HTTP Authentication: Basic and Digest
Access Authentication" for use in Aurthorization & Proxy-Authorization
headers is: 

digest-uri       = "uri" "=" digest-uri-value
digest-uri-value = request-uri   ; As specified by HTTP/1.1


RFC2543, updates this definition to be:

        2.   The BNF for digest-uri-value is:

digest-uri-value  =  Request-URI ; a defined in Section 4.3

Yet, all the sample call flows for the uri parameters for the Aurthorization
& Proxy-Authorization headers that I've seen (i.e. SIP Telephony Service
Examples With Call Flows)
enclose the uri parameter in quotes (i.e. a quoted string) just like the
HTTP docs:

uri="sip:xxx.yyyy.com"

Syntacically, I can't see how this is possible.  Am I wrong?  If so, could
someone point out the relevant section that explains how this is possible?

Thank you.

Linden deCarmo
Netspeak Corporation
902 Clint Moore Road
Suite 104
Boca Raton, FL 33487




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To: "'Raja Badri'" <raja.badri@santera.com>,
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Subject: RE: SIP BCP-T question
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The ISUP maintenance messages relate to the physical trunk circuit, which
when not being used on a call, is still physically present and in an IDLE
(or blocked) state. SIP BCT-T sets up calls over an RTP "logical"
connection, which is not a physical circuit. After the end of the call, the
RTP connection ceases to exist. Therefore is does not make sense to block
it, unblock it, reset it etc......
 
Steve Carr 
Unisphere Solutions Inc. 
Boca Raton 
scarr@unispheresolutions.com 
Office (561) 955-8288 
Fax    (561) 955-6477 

www.unispheresolutions.com 

-----Original Message-----
From: Raja Badri [mailto:raja.badri@santera.com]
Sent: Monday, February 14, 2000 1:04 PM
To: sip ML
Subject: SIP BCP-T question


Hi,
 
I have one question regarding the SIP BCP-T.
INFO message outlines the Mid-Call ISUP messages.
What is the reason behind not including the ISUP maintenance 
messages like Block, Unblock and Reset ?
 
Thanks,
Raja Badri
Santera Systems Inc.
 




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From: Matthias.Lang@ericsson.com
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: sip@lists.research.bell-labs.com
Subject: Re: Question about media desinations
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Jonathan Rosenberg writes:
 > Hmmm. Actually, placing the source address in the SDP solves another
 > problem. When multiple 200 OK (or 183 or whatever) arrive with SDP, the
 > caller is going to receive audio all on the same port, with no way to
 > figure out which RTP stream corresponds to which call leg. By adding the
 > source address, the UAC will be able to correlate them.

The easy way to figure out which RTP stream corresponds to which SIP
call is to use different destination port numbers. I don't think 
you can always distinguish calls based on source address, I don't
think there's anything in the RTP spec which says I can't send two
different RTP streams from the same port.

It seems like we have a few choices:

1. Distinguish streams based on destination port #. This might make
   life harder for firewalls and firewall administrators; it's easiest
   to just open port 5004 for RTP.

2. Distinguish streams based on SSRC. This is a bad idea; you can't
   be sure that a client won't change SSRC from packet to packet.
   You also can't be sure that two clients talkign to a gateway
   won't choose the same SSRC, however unlikely that may be in theory.

3. Distinguish streams based on source address and port. (as above)

The first option seems qutie clearly the least of three evils.

Matthias
   



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 21 15:25:53 2000
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Henning Schulzrinne wrote:
> 
> Jonathan Rosenberg wrote:
> >
> 
> >
> > However, you cannot separate the header from the protocol semantics and
> > exchanges it is used in. To date, there are two applications:
> >
> > 1. Returned in response to OPTIONS to indicate capabilities (this is
> > what is described in rfc2543bis)
> 
> Given that, I'm somewhat tempted to explore a different, simpler avenue
> initially: assume that the client declares all beyond-RFC2543
> capabilities, possibly allow short-hand feature names (as Anders is
> suggesting) and wait until we have a better idea as to whether we'll
> have 10 or 100 extensions. Clients that have 100 extensions, can offer
> the
> 
>   btw-I-know-more-than-this-please-ask-again
> 
> extension in Supported (with a somewhat shorter name, like "shy"), which
> is the current model.

That certainly would simplify things. Its all a matter of engineering
goals - the negotiation mechanism is quite powerful and flexible, and
will grow to handle a large extension space, whilst what you are
proposing is far simpler and handles a smaller case. Which is our goal?
Well, I'm inclined to agree with you that KISS probably wins over a
yet-to-be-fully demonstrated need. We can wait on the more extensive
negotiation stuff, so long as there is a way to add it later ("shy", as
you suggest).


> > So, I would argue that (1) is something which might well be included in
> > the next version of the base spec - its completely backwards compatible
> > and easy, and well in line with OPTIONS operation now. The other use,
> > (2), is new and different. So, where does the SYNTAX itself of Supported
> > get defined? Doesn't really matter... may very well be both.
> 
> I'd rather have one place. If timing dictates this, we can have a
> Proposed Standard for the Supported header syntax and use (nothing else)
> pushed out in a matter of weeks, which would then be labeled as
> "obsoleted by" RFC2543bis whenever it reaches Draft.

Thats fine. Basically the server features draft does this now, except I
don't know if we were going to fold it into draft when the time came.



So, the current proposal is to write a draft that:

1. Defines the Supported header syntax
2. specifies its usage in OPTIONS responses
3. specifies its usage in requests - basically, it lists all features
with short names
4. supports no negotiation

This draft will be obsoleted by rfc2543bis when it goes to rfc, but no
way we can wait that long for this.

How do people feel about this? If we get consensus, I'll do another rev
on the document and we can push it out the door.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: Francois Menard List Account <fm-listproc@fmmo.ca>,
        James Kempf <James.Kempf@Eng.Sun.COM>, mat@cisco.com,
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Below

> -----Original Message-----
> From:	Jonathan Rosenberg [SMTP:jdrosen@dynamicsoft.com]
> Sent:	Monday, February 21, 2000 10:44 AM
> To:	Morrow, Glenn [RICH2:C301:EXCH]
> Cc:	Francois Menard List Account; James Kempf; mat@cisco.com;
> sip@lists.research.bell-labs.com; etremblay@mediatrix.com
> Subject:	Re: DHCP option tags for SIP
> 
> 
> 
> > Glenn Morrow wrote:
> > 
> > Why can't the downloaded UAS, by bootp or other means, simply have the
> > configured proxy address? The configuration can be changed later with
> > SIP signaling without reboot for the in-service cases.
> 
> The place I download my client from is generally unrelated to my service
> provider (I downloaded Netscape from Netscapes site... has nothing to do
> with my ISP). 
> 
	TRUE but ISPs can download software too using the same method while
the UAS is being used (inservice). DHCP usually get invoked at boot-up. Some
ISPs use DHCP right to the customers devices, others do not.

> Even presuming you did this, how does one change this later with SIP
> signaling? Changing this is exactly what DHCP or SLP is desired for...
	Are you trying to convince people that SIP can't do this? I believe
there are many methods that could be used to facilitate this.
INVITE/BYE/CANCEL CONTACT header, Registration, Options, etc..

> > If we do go forward with these changes, what do think SG-16 will want
> > to do in another month?
> 
> I'm not sure I understand this... is SG16 planning some kind of activity
> in this area (SIP configuration)?
	That would be something.

	What I mean is: " Perhaps one  might also wish to include H.323
Gatekeeper address discovery extensions for DHCP, too". 

	Would you support this? 

	What would be your reasoning for not supporting this? 

	I think you might find yourself repeating the arguments above and
others of the chain. 

	If not, you might as well go ahead and change the draft to include
H.323 gatekeeper discovery as well. 


	Don't get me wrong. I'm just trying to offer some constructive
criticism. 

	There is no hidden agenda on changing DHCP. There are simply other
ways to accomplish the function desired without it.


> -Jonathan R.
> 
> -- 
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120 
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 

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<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">Jonathan Rosenberg =
[SMTP:jdrosen@dynamicsoft.com]</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Monday, February 21, 2000 10:44 AM</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">Morrow, Glenn [RICH2:C301:EXCH]</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Cc:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">Francois Menard List Account; James Kempf; =
mat@cisco.com; sip@lists.research.bell-labs.com; =
etremblay@mediatrix.com</FONT></P>

<P><B><FONT SIZE=3D1 =
FACE=3D"Arial">Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT>=
</B> <FONT SIZE=3D1 FACE=3D"Arial">Re: DHCP option tags for SIP</FONT>
</P>
<BR>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt; Glenn Morrow wrote:</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Why can't the downloaded UAS, by =
bootp or other means, simply have the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; configured proxy address? The =
configuration can be changed later with</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; SIP signaling without reboot for =
the in-service cases.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">The place I download my client from is =
generally unrelated to my service</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">provider (I downloaded Netscape from =
Netscapes site... has nothing to do</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">with my ISP). </FONT>
</P>

<P><B><I><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">TRUE but ISPs =
can download software too using the same method while the UAS is being =
used (inservice). DHCP usually get invoked at boot-up. Some ISPs use =
DHCP right to the customers devices, others do =
not.</FONT></I></B><I></I></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Even presuming you did this, how does =
one change this later with SIP</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">signaling? Changing this is exactly =
what DHCP or SLP is desired for...</FONT>
<BR><B><I><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Are you =
trying to convince people that SIP can't do this? I believe there are =
many methods that could be used to facilitate this. INVITE/BYE/CANCEL =
CONTACT header, Registration, Options, etc..</FONT></I></B><I></I></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt; If we do go forward with these =
changes, what do think SG-16 will want</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; to do in another month?</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I'm not sure I understand this... is =
SG16 planning some kind of activity</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">in this area (SIP =
configuration)?</FONT>
<BR><B><I><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">That would be =
something.</FONT></I></B>
</P>

<P><B><I><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">What I mean =
is: &quot; Perhaps one&nbsp; might also wish to include H.323 =
Gatekeeper address discovery extensions for DHCP, too&quot;. =
</FONT></I></B></P>

<P><B><I><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Would you =
support this? </FONT></I></B>
</P>

<P><B><I><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">What would be =
your reasoning for not supporting this? </FONT></I></B>
</P>

<P><B><I><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">I think you =
might find yourself repeating the arguments above and others of the =
chain. </FONT></I></B>
</P>

<P><B><I><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">If not, you =
might as well go ahead and change the draft to include H.323 gatekeeper =
discovery as well. </FONT></I></B>
</P>
<BR>

<P><B><I><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Don't get me =
wrong. I'm just trying to offer some constructive criticism. =
</FONT></I></B>
</P>

<P><B><I><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">There is no =
hidden agenda on changing DHCP. There are simply other ways to =
accomplish the function desired without it.</FONT></I></B>
</P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Arial">-Jonathan R.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">-- </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Jonathan D. =
Rosenberg&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
200 Executive Drive</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Chief =
Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Suite 120 </FONT>
<BR><FONT SIZE=3D2 =
FACE=3D"Arial">dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; West Orange, NJ 07052</FONT>
<BR><FONT SIZE=3D2 =
FACE=3D"Arial">jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp; FAX:&nbsp;&nbsp; (732) 741-4778</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://www.cs.columbia.edu/~jdrosen" =
TARGET=3D"_blank">http://www.cs.columbia.edu/~jdrosen</A></FONT></U><FON=
T SIZE=3D2 =
FACE=3D"Arial">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: =
(732) 741-7244</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://www.dynamicsoft.com" =
TARGET=3D"_blank">http://www.dynamicsoft.com</A></FONT></U>
</P>
</UL>
</BODY>
</HTML>
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        SIP <sip@lists.research.bell-labs.com>
Subject: Re: Extension negotiation draft
References: <38A85EA3.6DB9A152@hplb.hpl.hp.com> <38B18BCC.C9028997@dynamicsoft.com> <38B194DE.183CA9DF@cs.columbia.edu> <38B19EB8.68CC1B10@dynamicsoft.com>
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Jonathan Rosenberg wrote:
> 

> 
> That certainly would simplify things. Its all a matter of engineering
> goals - the negotiation mechanism is quite powerful and flexible, and
> will grow to handle a large extension space, whilst what you are
> proposing is far simpler and handles a smaller case. Which is our goal?
> Well, I'm inclined to agree with you that KISS probably wins over a
> yet-to-be-fully demonstrated need. We can wait on the more extensive
> negotiation stuff, so long as there is a way to add it later ("shy", as
> you suggest).

I'd rather get the KISS draft out in a week and then worry about it
again when/if the need arises. At that time, we'll also have a better
idea, for example, if we need parameterized extensions, as has been
mentioned somewhere along the way. 


> So, the current proposal is to write a draft that:
> 
> 1. Defines the Supported header syntax
> 2. specifies its usage in OPTIONS responses
> 3. specifies its usage in requests - basically, it lists all features
> with short names
> 4. supports no negotiation
> 
> This draft will be obsoleted by rfc2543bis when it goes to rfc, but no
> way we can wait that long for this.

Agreed. I've already included the changes in today's version of
rfc2543bis. It might be useful to have the wording in the draft and
rfc2543bis agree, to simplify the merging later. (Obviously, rfc2543bis
spreads things around a bit more, e.g., the redefinition of the
option-tag and the table indicating allowable use.)

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



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 21 15:57:48 2000
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> Glenn Morrow wrote:
> 
\
>      The place I download my client from is generally unrelated to my
>      service
>      provider (I downloaded Netscape from Netscapes site... has
>      nothing to do
>      with my ISP).
> 
>      TRUE but ISPs can download software too using the same method
>      while the UAS is being used (inservice). DHCP usually get invoked
>      at boot-up. Some ISPs use DHCP right to the customers devices,
>      others do not.

I don't follow what you are trying to say here.

> 
>      Even presuming you did this, how does one change this later with
>      SIP
>      signaling? Changing this is exactly what DHCP or SLP is desired
>      for...
>      Are you trying to convince people that SIP can't do this? I
>      believe there are many methods that could be used to facilitate
>      this. INVITE/BYE/CANCEL CONTACT header, Registration, Options,
>      etc..

All of these things assume you have a server to talk to in the first
place. Thats the problem we are trying to solve.

> 
>      > If we do go forward with these changes, what do think SG-16
>      will want
>      > to do in another month?

>      What I mean is: " Perhaps one  might also wish to include H.323
>      Gatekeeper address discovery extensions for DHCP, too".
> 
>      Would you support this?

Its not for me to support or not support. The SIP wg doesn't work on
H.323, ITU SG16 does. If they want to register DHCP options for
gatekeepers, fine with me. 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Hi,
So if my interpretation is correct, what you are saying is put a "*" when
you dont want to specify a uri, else put a uri.
So in each Accept-Contact hdr, either a "*" or (name-addr |addr-spec) is
mandatory.

Is the example in Page 13 also correct then ?

 Accept-Contact: sip:sales@acme.com ;q=0,
     ;media="!video" ;q=0.1,
     ;mobility="fixed"  ;q=0.6,
     ;mobility="!fixed" ;q=0.4

Im still confused with the use of commas - if they are indeed sep.
Acc-Contact hdrs, shouldnt the example be

 Accept-Contact: sip:sales@acme.com ;q=0,
    * ;media="!video" ;q=0.1,
     *;mobility="fixed"  ;q=0.6,
     *;mobility="!fixed" ;q=0.4

And I guess the same holds for reject contact too ?

i.e.

 Reject-Contact  =  "Reject-Contact" ":"   1# ( ( name-addr | addr-spec |
"*" )  [ *( ";" new-params ) ] )

Regds
Arjun

Arjun Roychowdhyry @ Hughes Software Systems





Jonathan Rosenberg <jdrosen@dynamicsoft.com> on 02/21/2000 09:30:52 PM

To:   Preethy Rangarajan/HSSBLR
cc:   sip@lists.research.bell-labs.com

Subject:  Re: Accept-Contact Header




Thanks for pointing this out. The bnf appears to be wrong, as it
mandates a valid name-addr or addr-spec. The document clearly discusses
the capability of not specifying one. I will fix in the next version
(due out before the draft deadline). The BNF should be:

Accept-Contact = "Accept-Contact" ":" 1#rule
rule   =  (name-addr | addr-spec | "*")
....


So, we'll use the * for anything, aligning with current Contact usage.

-Jonathan R.

rpreethy@hss.hns.com wrote:
>
> hi All,
>
> The document SIP Caller Preferences and Callee Capabilities -
> draft-ietf-sipcallerprefs-00.txt
> gives the grammer for Accept-Contact Header as
>
> Accept-Contact   =  "Accept-Contact" ":" 1# rule
>    rule             =  ( name-addr | addr-spec )
>                        [ *( ";" (cp-params | extension-param | q-param) )
]
>    q-param          =  "q" "=" qvalue
>    extension-param  =  extension-name "=" extension-value
>    extension-name   =  token
>    extension-value  =  token | quoted-string
>
> and the example given is
>
> Accept-Contact: sip:sales@acme.com ;q=0,
>      ;media="!video" ;q=0.1,
>      ;mobility="fixed"  ;q=0.6,
>      ;mobility="!fixed" ;q=0.4
>
> According to my understanding, in this example each of the comma
separated
> values is a separate Accept Contact rule,
> And the parameters of a paticular accept contact rule are separated by
";"
> semicolon.
> Here each of the q-params are followed by a comma and then a semi-colon .
>
> I cannot understand whether to consider this as
> 1) a single accept contact rule with all the entries following
> sip:sales@acme.com as parameters in which case I dont see a need for a
> comma and semi-colon as separaters between parameters.
>
> 2) 4 rules where-in the 2nd 3rd and 4th do not have a nameaddr|addrspec
> entry which is also an error, becos the rule mandates the presence of
> nameaddr|addr-spec.
>
> Can someone explain this to me.
>
> Regards,
> Preethy

--
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com








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From: Francois Menard List Account <fm-listproc@fmmo.ca>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: James Kempf <James.Kempf@Eng.Sun.COM>, mat@cisco.com,
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I also think that it ought to be possible to do it with DHCP.

I agree with Jonathan here.

-=Francois=-

On Mon, 21 Feb 2000, Jonathan Rosenberg wrote:

> I'd like to try and keep this discussion focused in order to quickly
> answer the following question:
> 
>   Should we "fast track" a DHCP options for finding a SIP server
> document?
> 
> Neither DHCP nor SLP are capable of providing complicated application
> layer configuration parameters, such as dialing plans or addresses.
> Neither are good for capabilities exchange, which is something different

James, do you have a rebuttal for this ?

-=Francois=-




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Cool beans. I think we're all in agreement now. Comments below anyway.

Jonathan Rosenberg wrote:
> 
> Anders Kristensen wrote:
> >
> > I'll go through these in turns.
> >
> >   1. Define Supported in the base spec.
> >
> > The draft can be seen as having two parts: the definition of the
> > Supported header and a proposal for a feature negotiation mechanism. I
> > think the two should be separated. It makes perfectly good sense for a
> > server to signal supported features in an OPTIONs response without
> > having to implement the negotiation mechanism, and likewise, clients
> > should be allowed to indicate that they support, say, reliable 1xx's
> > without having to implement the negotiation mechanism.
> 
> I agree that Supported has uses outside of the negotiation mechanism. In
> fact,
> rfc2543bis has a definition of Supported.
> 
> However, you cannot separate the header from the protocol semantics and
> exchanges it is used in. To date, there are two applications:
> 
> 1. Returned in response to OPTIONS to indicate capabilities (this is
> what is described in rfc2543bis)
> 2. the negotiation mechanism in the draft

There is a third which is the one I'm concerned about here, which is to
allow clients to tell the server which features it supports without
having to implement the negotiation mechanism.  Like Henning, I believe
that many clients can get away with not implementing the negotiation
mechanism simply by listing all known features.

> 
> So, I would argue that (1) is something which might well be included in
> the next version of the base spec - its completely backwards compatible
> and easy, and well in line with OPTIONS operation now. The other use,
> (2), is new and different. So, where does the SYNTAX itself of Supported
> get defined? Doesn't really matter... may very well be both.
> 
> >   2. Do *not* let presence of the Supported header implicitly
> >      signal support for the extension negotiation mechanism.
> >
> > This is a corollory of the previous argument. If UAs can list supported
> > features independent of the negotiation mechanism then presence of the
> > Supported header cannot be taken to implicitly indicate support for the
> > latter.
> 
> Again, you can't separate the header from the protocol behavior. I would
> agree that adding Supported to an OPTIONS response does not imply
> understanding of the negotiation mechanism. However, placement of
> Supported in a request is strongly coupled with the negotiation
> mechanism, and I would be reluctant to allow it without the full thing.

Well, this is really what I argued against. A great part of the value of
letting clients list features known to them is that it becomes a
mechanism for not having to implement the negotiation. I think this
advantage weighs much more heavily than saving listing the extra tag
when negotiation *is* supported.

If I understand your 4-step plan correctly this is what it is saying
also with the "shy" tag. Basically the Supported header can be used in
requests without it signaling support for negotiation, which is instead
signalled explicitly with the inclusion of the "shy" tag.


> So, in the interests of short fields (which you argue for below), I
> would prefer to allow the presence of Supported alone in a request be
> used to indicate this.
> 
> >
> >   3. Drop the reversed domain name feature naming convention.
> >
> > It makes messages longer and is unnecessary.
> >
> > [...]
> > "x-" prefix for experimental or proprietary extensions or continue to
> > use the domain namespaces for this purpose.
> 
> OK.... I'll buy on this one. However, I'd still like the base spec to
> define usage of the domain-based namespaces for unregistered extensions.

Agreed.

I must say I really don't like the idea of making it optional whether or
not to include the "org.ietf.sip" prefix. It complicates matters for no
extra gain.


-- 
Anders Kristensen <ak@hplb.hpl.hp.com>,
http://www-uk.hpl.hp.com/people/ak/
Hewlett-Packard Labs, Bristol, UK



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 Hi:

     When reading RFC 2543, I have a question regarding path of ACK and BYE.
     In section 4.2.4, there writes "If the INVITE request contained a
contact header, the callee should send a BYE request to that address rather
than the from address ".
And in section 10.5.1, there has " If the response contained a Contact
header field, the ACK may be sent to the address listed in that Contact
header field."  So I can draw a following call flow chart :

user A          Proxy          User B
   INVITE
  ----------->  INVITE
                      --------->
                             200
                      <---------
                200
   <----------
   ACK
   ----------------------------->
      The call is established
                          BYE
   <----------------------------
  In the example ,when do I release the call resource in the Proxy Server ?
  Thanks !

  Shaohua Yin









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yinshaohua wrote:
> 
>  Hi:
> 
>      When reading RFC 2543, I have a question regarding path of ACK and BYE.
>      In section 4.2.4, there writes "If the INVITE request contained a
> contact header, the callee should send a BYE request to that address rather
> than the from address ".
> And in section 10.5.1, there has " If the response contained a Contact
> header field, the ACK may be sent to the address listed in that Contact
> header field."  So I can draw a following call flow chart :
> 
> user A          Proxy          User B
>    INVITE
>   ----------->  INVITE
>                       --------->
>                              200
>                       <---------
>                 200
>    <----------
>    ACK
>    ----------------------------->
>       The call is established
>                           BYE
>    <----------------------------

Correct.

>   In the example ,when do I release the call resource in the Proxy Server ?
>   Thanks !

What call resource? If the proxy doesn't record-route, it can't maintain
call state, and thus there is no call resource to release. The proxy
will probably maintain transaction state in this case (what we call
stateful). That state is released after some short period of time after
the 200 OK has been forwarded; 32 seconds is reasonable based on the
timers defined in the spec. In the case of a stateless proxy, there is
no state at all to maintain or release.

If the proxy record-routes, it will see the ACK and the BYE. If its
maintaining call state, it can release that state upon the BYE. Also,
the usage of the session timer
(http://www.ietf.org/internet-drafts/draft-ietf-sip-session-timer-00.txt)
is recommended in case the BYE never gets sent or is lost. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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archow@hss.hns.com wrote:
> 
> Hi,
> So if my interpretation is correct, what you are saying is put a "*" when
> you dont want to specify a uri, else put a uri.
> So in each Accept-Contact hdr, either a "*" or (name-addr |addr-spec) is
> mandatory.
> 
> Is the example in Page 13 also correct then ?

No; there must be a star there.

> 
>  Accept-Contact: sip:sales@acme.com ;q=0,
>      ;media="!video" ;q=0.1,
>      ;mobility="fixed"  ;q=0.6,
>      ;mobility="!fixed" ;q=0.4
>
> Im still confused with the use of commas - if they are indeed sep.
> Acc-Contact hdrs, shouldnt the example be
> 
>  Accept-Contact: sip:sales@acme.com ;q=0,
>     * ;media="!video" ;q=0.1,
>      *;mobility="fixed"  ;q=0.6,
>      *;mobility="!fixed" ;q=0.4

Yes. This is correct.

> 
> And I guess the same holds for reject contact too ?
> 
> i.e.
> 
>  Reject-Contact  =  "Reject-Contact" ":"   1# ( ( name-addr | addr-spec |
> "*" )  [ *( ";" new-params ) ] )

Yes.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Subject: Cross-domain QoS setup
To: jdrosen@dynamicsoft.com, hgs@cs.columbia.edu, Steven.R.Donovan@wcom.com
Date: Tue, 22 Feb 2000 10:10:52 -0600 (CST)
Cc: sip@lists.research.bell-labs.com
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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Since draft-ietf-mmusic-sdp-qos-00.txt expired in December, and is
functionality we still need to standardize, it will (presumably) be 
reissued soon. Speaking on behalf of the PacketCable group, we have
some suggestions of enhancements to the upcoming release of this
document which would allow it to support a more robust network 
architecture.

The current draft makes an assumption that end-to-end network 
resource reservation is controlled by a single entity. For calls
being placed across adminstrative domains, however, this will
not be the case. Under such circumstances, the reservation of
resources needs a greater degree of coordination than is 
currently provided in the draft.

In particular, the calling user agent needs the ability to
communicate to the called user agent that it has reserved 
the necessary resources on its end, and that alerting may
begin as soon as the called user agent has done the same.

Our proposed method of doing so is as follows:

1) As in the current draft, the INVITE message contains
   a=qos:... lines to indicate the calling party's preferences
   for which streams have mandatory or optional reservation
   couplings associated with them.

2) Similary, the 18x response contains a subset of the INVITE
   preconditions.

3) The PRACK confirming the 18x will echo back the finally
   agreed-on preconditions. This will serve as a "go ahead"
   message to the called party to begin its actual resource
   reservation.

4) After the calling side has procured reservations for the
   required resources, it sends an INFO message to the called
   party. This INFO message contains an SDP with a confirmation
   that the required streams have been reserved. For the purposes
   of this message, we add two values to the "strength" field
   in the current draft: "reserved" and "resvfail". The presence
   of the "resvfail" token allows the called party to terminate
   the call with an error code (for mandatory streams) or to
   remove its local reservation (for optional streams).

5) The 200 response to the INFO message similarly contains
   SDP indicating, on a per-stream basis, the status of
   its local reservations (allowing the calling party to
   remove reservations for optional streams which were not
   reserved on the terminating side).

An example callflow follows (with unnecessary but present details, 
like tags and call-ids, removed). To prevent abiguity, I have inserted 
a "Session" header for all cases where SDP is present in a way that is 
not defined in the base specification and cannot be determined from
context (e.g. the response to a QoS INFO message).

--------------------
UAC->UAS

INVITE sip:adam.roach@ericsson.com SIP/2.0
To: Adam Roach <adam.roach@ericsson.com>
From: John Q. Public <jpublic@ietf.org>
CSeq: 1 INVITE
Content-Type: application/sdp

v=0
m=audio 48327 RTP/AVP 0
a=qos:mandatory sendrecv
m=video 10275 RTP/AVP 31
a=qos:mandatory sendrecv
--------------------
UAS->UAC

SIP/2.0 183 Preconditions
To: Adam Roach <adam.roach@ericsson.com>
From: John Q. Public <jpublic@ietf.org>
CSeq: 1 INVITE
Session: QoS
Content-Type: application/sdp

v=0
m=audio 48327 RTP/AVP 0
a=qos:mandatory sendrecv
m=video 10275 RTP/AVP 31
a=qos:optional sendrecv
--------------------
UAC->UAS

PRACK sip:adam.roach@ericsson.com SIP/2.0
To: Adam Roach <adam.roach@ericsson.com>
From: John Q. Public <jpublic@ietf.org>
CSeq: 2 PRACK
Session: QoS
Content-Type: application/sdp

v=0
m=audio 48327 RTP/AVP 0
a=qos:mandatory sendrecv
m=video 10275 RTP/AVP 31
a=qos:optional sendrecv
--------------------
UAS->UAC

SIP/2.0 200 OK
To: Adam Roach <adam.roach@ericsson.com>
From: John Q. Public <jpublic@ietf.org>
CSeq: 2 PRACK
Content-Length: 0
--------------------
Both parties perform resource reservation
--------------------
UAC->UAS

INFO sip:adam.roach@ericsson.com SIP/2.0
To: Adam Roach <adam.roach@ericsson.com>
From: John Q. Public <jpublic@ietf.org>
CSeq: 3 INFO
Session: QoS
Content-Type: application/sdp

v=0
m=audio 48327 RTP/AVP 0
a=qos:reserved sendrecv
m=video 10275 RTP/AVP 31
a=qos:resvfail sendrecv
--------------------
UAS->UAC

SIP/2.0 200 OK
To: Adam Roach <adam.roach@ericsson.com>
From: John Q. Public <jpublic@ietf.org>
CSeq: 3 INFO
Content-Type: application/sdp

v=0
m=audio 48327 RTP/AVP 0
a=qos:reserved sendrecv
m=video 10275 RTP/AVP 31
a=qos:resvfail sendrecv
--------------------
The called party cancels the videostream reservation
and begins alerting the called party.
--------------------
UAS->UAC

SIP/2.0 180 Ringing
To: Adam Roach <adam.roach@ericsson.com>
From: John Q. Public <jpublic@ietf.org>
CSeq: 1 INVITE
Content-Length: 0
--------------------

-- 
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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Subject: Re: About Timer
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Hi,
if you are talking about a stateful proxy, then it must have inserted
Record-Route header. If it did,  ACK and BYE will pass through it.

Regds
Arjun
--
Arjun Roychowdhury @ Hughes Software Systems





"yinshaohua" <yin@huawei.com.cn> on 02/22/2000 06:21:42 PM

Please respond to "yinshaohua" <yin@huawei.com.cn>

To:   sip@lists.research.bell-labs.com
cc:

Subject:  About Timer




 Hi:

     When reading RFC 2543, I have a question regarding path of ACK and
BYE.
     In section 4.2.4, there writes "If the INVITE request contained a
contact header, the callee should send a BYE request to that address rather
than the from address ".
And in section 10.5.1, there has " If the response contained a Contact
header field, the ACK may be sent to the address listed in that Contact
header field."  So I can draw a following call flow chart :

user A          Proxy          User B
   INVITE
  ----------->  INVITE
                      --------->
                             200
                      <---------
                200
   <----------
   ACK
   ----------------------------->
      The call is established
                          BYE
   <----------------------------
  In the example ,when do I release the call resource in the Proxy Server ?
  Thanks !

  Shaohua Yin
















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Date: Tue, 22 Feb 2000 13:09:36 -0600
To: "Adam B. Roach" <Adam.Roach@Ericsson.com>, jdrosen@dynamicsoft.com,
        hgs@cs.columbia.edu, Steven.R.Donovan@wcom.com
From: "James M. Polk" <jmpolk@cisco.com>
Subject: Re: Cross-domain QoS setup
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Adam

I think I see a flaw in this suggestion, but I'm not sure:

What happens if you have, for example, Level 3's pure IP network between  the
Calling agent's Domain and the Called agents Domain? 

Expanding on this, say I'm within my domain calling from my SIP IP Phone,
Cisco.com,  who has a direct IP connection into Level 3's IP Backbone (I don't
think we do -- but work within me here), but you are on a Cable Modem in the
AT&T Broadband Network in Dallas with VoIP enabled to your home; your example
seems to only allow for interaction between the calling and called domains,
does it matter than there will likely (more so in the future than now) be a
additional IP domain in the middle that could very well muck things up quite a
bit?

Further, if it does matter, what indication will both sides get if this middle
domain is the only one not to comply with the requested QoS settings?

At 10:10 AM 2/22/2000 -0600, Adam B. Roach wrote:
>Since draft-ietf-mmusic-sdp-qos-00.txt expired in December, and is
>functionality we still need to standardize, it will (presumably) be 
>reissued soon. Speaking on behalf of the PacketCable group, we have
>some suggestions of enhancements to the upcoming release of this
>document which would allow it to support a more robust network 
>architecture.
>
>The current draft makes an assumption that end-to-end network 
>resource reservation is controlled by a single entity. For calls
>being placed across adminstrative domains, however, this will
>not be the case. Under such circumstances, the reservation of
>resources needs a greater degree of coordination than is 
>currently provided in the draft.
>
>In particular, the calling user agent needs the ability to
>communicate to the called user agent that it has reserved 
>the necessary resources on its end, and that alerting may
>begin as soon as the called user agent has done the same.
>
>Our proposed method of doing so is as follows:
>
>1) As in the current draft, the INVITE message contains
>   a=qos:... lines to indicate the calling party's preferences
>   for which streams have mandatory or optional reservation
>   couplings associated with them.
>
>2) Similary, the 18x response contains a subset of the INVITE
>   preconditions.
>
>3) The PRACK confirming the 18x will echo back the finally
>   agreed-on preconditions. This will serve as a "go ahead"
>   message to the called party to begin its actual resource
>   reservation.
>
>4) After the calling side has procured reservations for the
>   required resources, it sends an INFO message to the called
>   party. This INFO message contains an SDP with a confirmation
>   that the required streams have been reserved. For the purposes
>   of this message, we add two values to the "strength" field
>   in the current draft: "reserved" and "resvfail". The presence
>   of the "resvfail" token allows the called party to terminate
>   the call with an error code (for mandatory streams) or to
>   remove its local reservation (for optional streams).
>
>5) The 200 response to the INFO message similarly contains
>   SDP indicating, on a per-stream basis, the status of
>   its local reservations (allowing the calling party to
>   remove reservations for optional streams which were not
>   reserved on the terminating side).
>
>An example callflow follows (with unnecessary but present details, 
>like tags and call-ids, removed). To prevent abiguity, I have inserted 
>a "Session" header for all cases where SDP is present in a way that is 
>not defined in the base specification and cannot be determined from
>context (e.g. the response to a QoS INFO message).
>
>--------------------
>UAC->UAS
>
>INVITE sip:adam.roach@ericsson.com SIP/2.0
>To: Adam Roach <adam.roach@ericsson.com>
>From: John Q. Public <jpublic@ietf.org>
>CSeq: 1 INVITE
>Content-Type: application/sdp
>
>v=0
>m=audio 48327 RTP/AVP 0
>a=qos:mandatory sendrecv
>m=video 10275 RTP/AVP 31
>a=qos:mandatory sendrecv
>--------------------
>UAS->UAC
>
>SIP/2.0 183 Preconditions
>To: Adam Roach <adam.roach@ericsson.com>
>From: John Q. Public <jpublic@ietf.org>
>CSeq: 1 INVITE
>Session: QoS
>Content-Type: application/sdp
>
>v=0
>m=audio 48327 RTP/AVP 0
>a=qos:mandatory sendrecv
>m=video 10275 RTP/AVP 31
>a=qos:optional sendrecv
>--------------------
>UAC->UAS
>
>PRACK sip:adam.roach@ericsson.com SIP/2.0
>To: Adam Roach <adam.roach@ericsson.com>
>From: John Q. Public <jpublic@ietf.org>
>CSeq: 2 PRACK
>Session: QoS
>Content-Type: application/sdp
>
>v=0
>m=audio 48327 RTP/AVP 0
>a=qos:mandatory sendrecv
>m=video 10275 RTP/AVP 31
>a=qos:optional sendrecv
>--------------------
>UAS->UAC
>
>SIP/2.0 200 OK
>To: Adam Roach <adam.roach@ericsson.com>
>From: John Q. Public <jpublic@ietf.org>
>CSeq: 2 PRACK
>Content-Length: 0
>--------------------
>Both parties perform resource reservation
>--------------------
>UAC->UAS
>
>INFO sip:adam.roach@ericsson.com SIP/2.0
>To: Adam Roach <adam.roach@ericsson.com>
>From: John Q. Public <jpublic@ietf.org>
>CSeq: 3 INFO
>Session: QoS
>Content-Type: application/sdp
>
>v=0
>m=audio 48327 RTP/AVP 0
>a=qos:reserved sendrecv
>m=video 10275 RTP/AVP 31
>a=qos:resvfail sendrecv
>--------------------
>UAS->UAC
>
>SIP/2.0 200 OK
>To: Adam Roach <adam.roach@ericsson.com>
>From: John Q. Public <jpublic@ietf.org>
>CSeq: 3 INFO
>Content-Type: application/sdp
>
>v=0
>m=audio 48327 RTP/AVP 0
>a=qos:reserved sendrecv
>m=video 10275 RTP/AVP 31
>a=qos:resvfail sendrecv
>--------------------
>The called party cancels the videostream reservation
>and begins alerting the called party.
>--------------------
>UAS->UAC
>
>SIP/2.0 180 Ringing
>To: Adam Roach <adam.roach@ericsson.com>
>From: John Q. Public <jpublic@ietf.org>
>CSeq: 1 INVITE
>Content-Length: 0
>--------------------
>
>-- 
>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
>
>

*************************************
"At the end of the day... the most committed win!"

James M. Polk
Sr. Product Manager, Multiservice Architecture and Standards
Enterprise Voice Business Unit
Cisco Systems
Dallas, Texas
w) 972.813.5208
f)  972.813.5280
www.cisco.com
--=====================_2567198==_.ALT
Content-Type: text/html; charset="us-ascii"

<html><div>Adam</div>
<br>
<div>I think I see a flaw in this suggestion, but I'm not sure:</div>
<br>
<div>What happens if you have, for example, Level 3's pure IP network
between&nbsp; the Calling agent's Domain and the Called agents Domain?
</div>
<br>
<div>Expanding on this, say I'm within my domain calling from my SIP IP
Phone, Cisco.com,&nbsp; who has a direct IP connection into Level 3's IP
Backbone (I don't think we do -- but work within me here), but you are on
a Cable Modem in the AT&amp;T Broadband Network in Dallas with VoIP
enabled to your home; your example seems to only allow for interaction
between the calling and called domains, does it matter than there will
likely (more so in the future than now) be a additional IP domain in the
middle that could very well muck things up quite a bit?</div>
<br>
<div>Further, if it does matter, what indication will both sides get if
this middle domain is the only one not to comply with the requested QoS
settings?</div>
<br>
<div>At 10:10 AM 2/22/2000 -0600, Adam B. Roach wrote:</div>
<div>&gt;Since draft-ietf-mmusic-sdp-qos-00.txt expired in December, and
is</div>
<div>&gt;functionality we still need to standardize, it will (presumably)
be </div>
<div>&gt;reissued soon. Speaking on behalf of the PacketCable group, we
have</div>
<div>&gt;some suggestions of enhancements to the upcoming release of
this</div>
<div>&gt;document which would allow it to support a more robust network
</div>
<div>&gt;architecture.</div>
<div>&gt;</div>
<div>&gt;The current draft makes an assumption that end-to-end network
</div>
<div>&gt;resource reservation is controlled by a single entity. For
calls</div>
<div>&gt;being placed across adminstrative domains, however, this
will</div>
<div>&gt;not be the case. Under such circumstances, the reservation
of</div>
<div>&gt;resources needs a greater degree of coordination than is 
</div>
<div>&gt;currently provided in the draft.</div>
<div>&gt;</div>
<div>&gt;In particular, the calling user agent needs the ability
to</div>
<div>&gt;communicate to the called user agent that it has reserved
</div>
<div>&gt;the necessary resources on its end, and that alerting 
may</div>
<div>&gt;begin as soon as the called user agent has done the 
same.</div>
<div>&gt;</div>
<div>&gt;Our proposed method of doing so is as follows:</div>
<div>&gt;</div>
<div>&gt;1) As in the current draft, the INVITE message contains</div>
<div>&gt;&nbsp;&nbsp; a=qos:... lines to indicate the calling party's
preferences</div>
<div>&gt;&nbsp;&nbsp; for which streams have mandatory or optional
reservation</div>
<div>&gt;&nbsp;&nbsp; couplings associated with them.</div>
<div>&gt;</div>
<div>&gt;2) Similary, the 18x response contains a subset of the
INVITE</div>
<div>&gt;&nbsp;&nbsp; preconditions.</div>
<div>&gt;</div>
<div>&gt;3) The PRACK confirming the 18x will echo back the
finally</div>
<div>&gt;&nbsp;&nbsp; agreed-on preconditions. This will serve as a
&quot;go ahead&quot;</div>
<div>&gt;&nbsp;&nbsp; message to the called party to begin its actual
resource</div>
<div>&gt;&nbsp;&nbsp; reservation.</div>
<div>&gt;</div>
<div>&gt;4) After the calling side has procured reservations for
the</div>
<div>&gt;&nbsp;&nbsp; required resources, it sends an INFO message to the
called</div>
<div>&gt;&nbsp;&nbsp; party. This INFO message contains an SDP with a
confirmation</div>
<div>&gt;&nbsp;&nbsp; that the required streams have been reserved. For
the purposes</div>
<div>&gt;&nbsp;&nbsp; of this message, we add two values to the
&quot;strength&quot; field</div>
<div>&gt;&nbsp;&nbsp; in the current draft: &quot;reserved&quot; and
&quot;resvfail&quot;. The presence</div>
<div>&gt;&nbsp;&nbsp; of the &quot;resvfail&quot; token allows the called
party to terminate</div>
<div>&gt;&nbsp;&nbsp; the call with an error code (for mandatory streams)
or to</div>
<div>&gt;&nbsp;&nbsp; remove its local reservation (for optional
streams).</div>
<div>&gt;</div>
<div>&gt;5) The 200 response to the INFO message similarly
contains</div>
<div>&gt;&nbsp;&nbsp; SDP indicating, on a per-stream basis, the status
of</div>
<div>&gt;&nbsp;&nbsp; its local reservations (allowing the calling party
to</div>
<div>&gt;&nbsp;&nbsp; remove reservations for optional streams which were
not</div>
<div>&gt;&nbsp;&nbsp; reserved on the terminating side).</div>
<div>&gt;</div>
<div>&gt;An example callflow follows (with unnecessary but present
details, </div>
<div>&gt;like tags and call-ids, removed). To prevent abiguity, I have
inserted </div>
<div>&gt;a &quot;Session&quot; header for all cases where SDP is present
in a way that is </div>
<div>&gt;not defined in the base specification and cannot be determined
from</div>
<div>&gt;context (e.g. the response to a QoS INFO message).</div>
<div>&gt;</div>
<div>&gt;--------------------</div>
<div>&gt;UAC-&gt;UAS</div>
<div>&gt;</div>
<div>&gt;INVITE sip:adam.roach@ericsson.com SIP/2.0</div>
<div>&gt;To: Adam Roach &lt;adam.roach@ericsson.com&gt;</div>
<div>&gt;From: John Q. Public &lt;jpublic@ietf.org&gt;</div>
<div>&gt;CSeq: 1 INVITE</div>
<div>&gt;Content-Type: application/sdp</div>
<div>&gt;</div>
<div>&gt;v=0</div>
<div>&gt;m=audio 48327 RTP/AVP 0</div>
<div>&gt;a=qos:mandatory sendrecv</div>
<div>&gt;m=video 10275 RTP/AVP 31</div>
<div>&gt;a=qos:mandatory sendrecv</div>
<div>&gt;--------------------</div>
<div>&gt;UAS-&gt;UAC</div>
<div>&gt;</div>
<div>&gt;SIP/2.0 183 Preconditions</div>
<div>&gt;To: Adam Roach &lt;adam.roach@ericsson.com&gt;</div>
<div>&gt;From: John Q. Public &lt;jpublic@ietf.org&gt;</div>
<div>&gt;CSeq: 1 INVITE</div>
<div>&gt;Session: QoS</div>
<div>&gt;Content-Type: application/sdp</div>
<div>&gt;</div>
<div>&gt;v=0</div>
<div>&gt;m=audio 48327 RTP/AVP 0</div>
<div>&gt;a=qos:mandatory sendrecv</div>
<div>&gt;m=video 10275 RTP/AVP 31</div>
<div>&gt;a=qos:optional sendrecv</div>
<div>&gt;--------------------</div>
<div>&gt;UAC-&gt;UAS</div>
<div>&gt;</div>
<div>&gt;PRACK sip:adam.roach@ericsson.com SIP/2.0</div>
<div>&gt;To: Adam Roach &lt;adam.roach@ericsson.com&gt;</div>
<div>&gt;From: John Q. Public &lt;jpublic@ietf.org&gt;</div>
<div>&gt;CSeq: 2 PRACK</div>
<div>&gt;Session: QoS</div>
<div>&gt;Content-Type: application/sdp</div>
<div>&gt;</div>
<div>&gt;v=0</div>
<div>&gt;m=audio 48327 RTP/AVP 0</div>
<div>&gt;a=qos:mandatory sendrecv</div>
<div>&gt;m=video 10275 RTP/AVP 31</div>
<div>&gt;a=qos:optional sendrecv</div>
<div>&gt;--------------------</div>
<div>&gt;UAS-&gt;UAC</div>
<div>&gt;</div>
<div>&gt;SIP/2.0 200 OK</div>
<div>&gt;To: Adam Roach &lt;adam.roach@ericsson.com&gt;</div>
<div>&gt;From: John Q. Public &lt;jpublic@ietf.org&gt;</div>
<div>&gt;CSeq: 2 PRACK</div>
<div>&gt;Content-Length: 0</div>
<div>&gt;--------------------</div>
<div>&gt;Both parties perform resource reservation</div>
<div>&gt;--------------------</div>
<div>&gt;UAC-&gt;UAS</div>
<div>&gt;</div>
<div>&gt;INFO sip:adam.roach@ericsson.com SIP/2.0</div>
<div>&gt;To: Adam Roach &lt;adam.roach@ericsson.com&gt;</div>
<div>&gt;From: John Q. Public &lt;jpublic@ietf.org&gt;</div>
<div>&gt;CSeq: 3 INFO</div>
<div>&gt;Session: QoS</div>
<div>&gt;Content-Type: application/sdp</div>
<div>&gt;</div>
<div>&gt;v=0</div>
<div>&gt;m=audio 48327 RTP/AVP 0</div>
<div>&gt;a=qos:reserved sendrecv</div>
<div>&gt;m=video 10275 RTP/AVP 31</div>
<div>&gt;a=qos:resvfail sendrecv</div>
<div>&gt;--------------------</div>
<div>&gt;UAS-&gt;UAC</div>
<div>&gt;</div>
<div>&gt;SIP/2.0 200 OK</div>
<div>&gt;To: Adam Roach &lt;adam.roach@ericsson.com&gt;</div>
<div>&gt;From: John Q. Public &lt;jpublic@ietf.org&gt;</div>
<div>&gt;CSeq: 3 INFO</div>
<div>&gt;Content-Type: application/sdp</div>
<div>&gt;</div>
<div>&gt;v=0</div>
<div>&gt;m=audio 48327 RTP/AVP 0</div>
<div>&gt;a=qos:reserved sendrecv</div>
<div>&gt;m=video 10275 RTP/AVP 31</div>
<div>&gt;a=qos:resvfail sendrecv</div>
<div>&gt;--------------------</div>
<div>&gt;The called party cancels the videostream reservation</div>
<div>&gt;and begins alerting the called party.</div>
<div>&gt;--------------------</div>
<div>&gt;UAS-&gt;UAC</div>
<div>&gt;</div>
<div>&gt;SIP/2.0 180 Ringing</div>
<div>&gt;To: Adam Roach &lt;adam.roach@ericsson.com&gt;</div>
<div>&gt;From: John Q. Public &lt;jpublic@ietf.org&gt;</div>
<div>&gt;CSeq: 1 INVITE</div>
<div>&gt;Content-Length: 0</div>
<div>&gt;--------------------</div>
<div>&gt;</div>
<div>&gt;-- </div>
<div>&gt;Adam Roach, Ericsson Inc. |&nbsp; Ph: +1 972 583 7594 | 1010 E.
Arapaho, MS L-04</div>
<div>&gt;adam.roach@ericsson.com&nbsp;&nbsp; | Fax: +1 972 669 0154 |
Richardson, TX 75081 USA</div>
<div>&gt;</div>
<div>&gt;</div>
<br>

<div align="center">
*************************************<br>
&quot;At the end of the day... the most committed win!&quot;<br>
<br>
</div>
James M. Polk<br>
Sr. Product Manager, Multiservice Architecture and Standards<br>
Enterprise Voice Business Unit<br>
Cisco Systems<br>
Dallas, Texas<br>
w) 972.813.5208<br>
f)&nbsp; 972.813.5280<br>
<a href="http://www.cisco.com/" eudora="autourl">www.cisco.com</a></html>

--=====================_2567198==_.ALT--




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From: James Kempf <James.Kempf@eng.sun.com>
Reply-To: James Kempf <James.Kempf@eng.sun.com>
Subject: Re: DHCP option tags for SIP
To: eussean@exu.ericsson.se, adutta@telcordia.com
Cc: gmorrow@nortelnetworks.com, jdrosen@dynamicsoft.com, fm-listproc@fmmo.ca,
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>For a wireless roaming user (e.g. moving between cells, subnets, and domains)
>where fast handoff is an issue during an ongoing call, DHCP option tag for
>finding out the nearest SIP proxy would be also very useful.
>

It is not yet clear whether DHCP will be used during wireless roaming. It
is a proposed option, but there are a few others.

		jak




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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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        sip@lists.research.bell-labs.com
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"Adam B. Roach" wrote:
> The current draft makes an assumption that end-to-end network 
> resource reservation is controlled by a single entity. For calls
> being placed across adminstrative domains, however, this will
> not be the case. Under such circumstances, the reservation of
> resources needs a greater degree of coordination than is 
> currently provided in the draft.

Not really. Resource reservation is under the "control" of all the
network elements along the path that perform some kind of admission
control and resource allocation. These elements will often belong to
different administrative domains. The draft makes no assumptions about
the number of providers along the reservation path, or anything like
that. It only ties resource reservation and signaling together at the
endpoints, dicatating a particular temporal sequence.

> 
> In particular, the calling user agent needs the ability to
> communicate to the called user agent that it has reserved
> the necessary resources on its end, and that alerting may
> begin as soon as the called user agent has done the same.

This is something different from what you are discussing above. It is,
in fact, covered in the draft. If A calls B, the RSVP RESV message from
A to B confirms resources are reserved from A to B, and the RESVCONF
from A to B confirms resource reservation from B to A. This allows B to
know whether resources are reserved in both directions. A mirror case
exists for A. 

The difficulty arises only with the segmented DQoS model in DCS, which
does not provide a way to confirm resource reservation (I think),
whereas RSVP does.

So, the question is this. We know we need a way for both parties to know
when resources are reserved in both directions. Is this a function that
belongs somewhere in SIP (through INFO, as you propose, or some other
means), or in a reservation protocol. The mechanism exists already now
in RSVP, but not in DQoS (DCS folks, please correct me if I am
mistaken). So, do we add it to SIP, in which case its not needed for
RSVP, or should it be added to DQoS, in which case the SIP operation is
independent of the underlying reservation mechanism?

I have argued that confirmation of resource reservation is rightly a job
for a resource reservation protocol. One of the things I like about SIP
is its separation from session level control and network layer issues,
such as resource reservation.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb 22 21:03:58 2000
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Date: Tue, 22 Feb 2000 19:58:51 -0600
From: Henry Sinnreich <henry.sinnreich@wcom.com>
Subject: RE: Cross-domain QoS setup
In-reply-to: <38B31536.71A6A8DD@dynamicsoft.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        "Adam B. Roach" <Adam.Roach@Ericsson.com>
Cc: hgs@cs.columbia.edu, Steven.R.Donovan@wcom.com,
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Detailed call flows addressing this issue are in
http://search.ietf.org/internet-drafts/draft-sinnreich-interdoma
in-sip-qos-osp-00.txt
in general agreement with Jonathan's position.
We hope to have a new version in time for the 47th IETF.

Henry

>-----Original Message-----
>From: owner-sip@lists.research.bell-labs.com
>[mailto:owner-sip@lists.research.bell-labs.com]On
>Behalf Of Jonathan
>Rosenberg
>Sent: Tuesday, February 22, 2000 5:01 PM
>To: Adam B. Roach
>Cc: hgs@cs.columbia.edu; Steven.R.Donovan@wcom.com;
>sip@lists.research.bell-labs.com
>Subject: Re: Cross-domain QoS setup
>
>
>
>
>"Adam B. Roach" wrote:
>> The current draft makes an assumption that
>end-to-end network
>> resource reservation is controlled by a single
>entity. For calls
>> being placed across adminstrative domains, however, this will
>> not be the case. Under such circumstances, the reservation of
>> resources needs a greater degree of coordination than is
>> currently provided in the draft.
>
>Not really. Resource reservation is under the
>"control" of all the
>network elements along the path that perform some kind
>of admission
>control and resource allocation. These elements will
>often belong to
>different administrative domains. The draft makes no
>assumptions about
>the number of providers along the reservation path, or
>anything like
>that. It only ties resource reservation and signaling
>together at the
>endpoints, dicatating a particular temporal sequence.
>
>>
>> In particular, the calling user agent needs the ability to
>> communicate to the called user agent that it has reserved
>> the necessary resources on its end, and that alerting may
>> begin as soon as the called user agent has done the same.
>
>This is something different from what you are
>discussing above. It is,
>in fact, covered in the draft. If A calls B, the RSVP
>RESV message from
>A to B confirms resources are reserved from A to B,
>and the RESVCONF
>from A to B confirms resource reservation from B to A.
>This allows B to
>know whether resources are reserved in both
>directions. A mirror case
>exists for A.
>
>The difficulty arises only with the segmented DQoS
>model in DCS, which
>does not provide a way to confirm resource reservation
>(I think),
>whereas RSVP does.
>
>So, the question is this. We know we need a way for
>both parties to know
>when resources are reserved in both directions. Is
>this a function that
>belongs somewhere in SIP (through INFO, as you
>propose, or some other
>means), or in a reservation protocol. The mechanism
>exists already now
>in RSVP, but not in DQoS (DCS folks, please correct me if I am
>mistaken). So, do we add it to SIP, in which case its
>not needed for
>RSVP, or should it be added to DQoS, in which case the
>SIP operation is
>independent of the underlying reservation mechanism?
>
>I have argued that confirmation of resource
>reservation is rightly a job
>for a resource reservation protocol. One of the things
>I like about SIP
>is its separation from session level control and
>network layer issues,
>such as resource reservation.
>
>-Jonathan R.
>
>--
>Jonathan D. Rosenberg                       200 Executive Drive
>Chief Scientist                             Suite 120
>dynamicsoft                                 West
>Orange, NJ 07052
>jdrosen@dynamicsoft.com                     FAX:
>(732) 741-4778
>http://www.cs.columbia.edu/~jdrosen         PHONE:
>(732) 741-7244
>http://www.dynamicsoft.com
>




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From: William Marshall <wtm@research.att.com>
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Date: Wed, 23 Feb 2000 00:23:52 -0500 (EST)
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Subject: Re: Cross-domain QoS setup
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James M. Polk wrote:
> does it matter than there will likely (more so in the future than now) be a
> additional IP domain in the middle that could very well muck things up quite a
> bit?

This is exactly the point.  In general there will be three (or more) service
providers involved in establishing the QoS, and their service level agreements
may be (and in some cases definitely are) different.  The router at the
edge of each service provider's network needs to convert the resource
reservation request into something that the next service provider is willing
to handle; in general this may not always be RSVP.  Depending on end-to-end
RSVP for a successful SIP connection would be an unfortunate dependency.

Bill Marshall
wtm@research.att.com

-----original message-----
> 
> 
> Adam
> 
> I think I see a flaw in this suggestion, but I'm not sure:
> 
> What happens if you have, for example, Level 3's pure IP network between  the
> Calling agent's Domain and the Called agents Domain? 
> 
> Expanding on this, say I'm within my domain calling from my SIP IP Phone,
> Cisco.com,  who has a direct IP connection into Level 3's IP Backbone (I don't
> think we do -- but work within me here), but you are on a Cable Modem in the
> AT&T Broadband Network in Dallas with VoIP enabled to your home; your example
> seems to only allow for interaction between the calling and called domains,
> does it matter than there will likely (more so in the future than now) be a
> additional IP domain in the middle that could very well muck things up quite a
> bit?
> 
> Further, if it does matter, what indication will both sides get if this middle
> domain is the only one not to comply with the requested QoS settings?
> 
> At 10:10 AM 2/22/2000 -0600, Adam B. Roach wrote:
> >Since draft-ietf-mmusic-sdp-qos-00.txt expired in December, and is
> >functionality we still need to standardize, it will (presumably) be 
> >reissued soon. Speaking on behalf of the PacketCable group, we have
> >some suggestions of enhancements to the upcoming release of this
> >document which would allow it to support a more robust network 
> >architecture.
> >
> >The current draft makes an assumption that end-to-end network 
> >resource reservation is controlled by a single entity. For calls
> >being placed across adminstrative domains, however, this will
> >not be the case. Under such circumstances, the reservation of
> >resources needs a greater degree of coordination than is 
> >currently provided in the draft.
> >
> >In particular, the calling user agent needs the ability to
> >communicate to the called user agent that it has reserved 
> >the necessary resources on its end, and that alerting may
> >begin as soon as the called user agent has done the same.
> >
> >Our proposed method of doing so is as follows:
> >
> >1) As in the current draft, the INVITE message contains
> >   a=qos:... lines to indicate the calling party's preferences
> >   for which streams have mandatory or optional reservation
> >   couplings associated with them.
> >
> >2) Similary, the 18x response contains a subset of the INVITE
> >   preconditions.
> >
> >3) The PRACK confirming the 18x will echo back the finally
> >   agreed-on preconditions. This will serve as a "go ahead"
> >   message to the called party to begin its actual resource
> >   reservation.
> >
> >4) After the calling side has procured reservations for the
> >   required resources, it sends an INFO message to the called
> >   party. This INFO message contains an SDP with a confirmation
> >   that the required streams have been reserved. For the purposes
> >   of this message, we add two values to the "strength" field
> >   in the current draft: "reserved" and "resvfail". The presence
> >   of the "resvfail" token allows the called party to terminate
> >   the call with an error code (for mandatory streams) or to
> >   remove its local reservation (for optional streams).
> >
> >5) The 200 response to the INFO message similarly contains
> >   SDP indicating, on a per-stream basis, the status of
> >   its local reservations (allowing the calling party to
> >   remove reservations for optional streams which were not
> >   reserved on the terminating side).
> >
> >An example callflow follows (with unnecessary but present details, 
> >like tags and call-ids, removed). To prevent abiguity, I have inserted 
> >a "Session" header for all cases where SDP is present in a way that is 
> >not defined in the base specification and cannot be determined from
> >context (e.g. the response to a QoS INFO message).
> >
> >--------------------
> >UAC->UAS
> >
> >INVITE sip:adam.roach@ericsson.com SIP/2.0
> >To: Adam Roach <adam.roach@ericsson.com>
> >From: John Q. Public <jpublic@ietf.org>
> >CSeq: 1 INVITE
> >Content-Type: application/sdp
> >
> >v=0
> >m=audio 48327 RTP/AVP 0
> >a=qos:mandatory sendrecv
> >m=video 10275 RTP/AVP 31
> >a=qos:mandatory sendrecv
> >--------------------
> >UAS->UAC
> >
> >SIP/2.0 183 Preconditions
> >To: Adam Roach <adam.roach@ericsson.com>
> >From: John Q. Public <jpublic@ietf.org>
> >CSeq: 1 INVITE
> >Session: QoS
> >Content-Type: application/sdp
> >
> >v=0
> >m=audio 48327 RTP/AVP 0
> >a=qos:mandatory sendrecv
> >m=video 10275 RTP/AVP 31
> >a=qos:optional sendrecv
> >--------------------
> >UAC->UAS
> >
> >PRACK sip:adam.roach@ericsson.com SIP/2.0
> >To: Adam Roach <adam.roach@ericsson.com>
> >From: John Q. Public <jpublic@ietf.org>
> >CSeq: 2 PRACK
> >Session: QoS
> >Content-Type: application/sdp
> >
> >v=0
> >m=audio 48327 RTP/AVP 0
> >a=qos:mandatory sendrecv
> >m=video 10275 RTP/AVP 31
> >a=qos:optional sendrecv
> >--------------------
> >UAS->UAC
> >
> >SIP/2.0 200 OK
> >To: Adam Roach <adam.roach@ericsson.com>
> >From: John Q. Public <jpublic@ietf.org>
> >CSeq: 2 PRACK
> >Content-Length: 0
> >--------------------
> >Both parties perform resource reservation
> >--------------------
> >UAC->UAS
> >
> >INFO sip:adam.roach@ericsson.com SIP/2.0
> >To: Adam Roach <adam.roach@ericsson.com>
> >From: John Q. Public <jpublic@ietf.org>
> >CSeq: 3 INFO
> >Session: QoS
> >Content-Type: application/sdp
> >
> >v=0
> >m=audio 48327 RTP/AVP 0
> >a=qos:reserved sendrecv
> >m=video 10275 RTP/AVP 31
> >a=qos:resvfail sendrecv
> >--------------------
> >UAS->UAC
> >
> >SIP/2.0 200 OK
> >To: Adam Roach <adam.roach@ericsson.com>
> >From: John Q. Public <jpublic@ietf.org>
> >CSeq: 3 INFO
> >Content-Type: application/sdp
> >
> >v=0
> >m=audio 48327 RTP/AVP 0
> >a=qos:reserved sendrecv
> >m=video 10275 RTP/AVP 31
> >a=qos:resvfail sendrecv
> >--------------------
> >The called party cancels the videostream reservation
> >and begins alerting the called party.
> >--------------------
> >UAS->UAC
> >
> >SIP/2.0 180 Ringing
> >To: Adam Roach <adam.roach@ericsson.com>
> >From: John Q. Public <jpublic@ietf.org>
> >CSeq: 1 INVITE
> >Content-Length: 0
> >--------------------
> >
> >-- 
> >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
> >
> >
> 
> *************************************
> "At the end of the day... the most committed win!"
> 
> James M. Polk
> Sr. Product Manager, Multiservice Architecture and Standards
> Enterprise Voice Business Unit
> Cisco Systems
> Dallas, Texas
> w) 972.813.5208
> f)  972.813.5280
> www.cisco.com
> --=====================_2567198==_.ALT
> Content-Type: text/html; charset="us-ascii"
> 
> <html><div>Adam</div>
> <br>
> <div>I think I see a flaw in this suggestion, but I'm not sure:</div>
> <br>
> <div>What happens if you have, for example, Level 3's pure IP network
> between&nbsp; the Calling agent's Domain and the Called agents Domain?
> </div>
> <br>
> <div>Expanding on this, say I'm within my domain calling from my SIP IP
> Phone, Cisco.com,&nbsp; who has a direct IP connection into Level 3's IP
> Backbone (I don't think we do -- but work within me here), but you are on
> a Cable Modem in the AT&amp;T Broadband Network in Dallas with VoIP
> enabled to your home; your example seems to only allow for interaction
> between the calling and called domains, does it matter than there will
> likely (more so in the future than now) be a additional IP domain in the
> middle that could very well muck things up quite a bit?</div>
> <br>
> <div>Further, if it does matter, what indication will both sides get if
> this middle domain is the only one not to comply with the requested QoS
> settings?</div>
> <br>
> <div>At 10:10 AM 2/22/2000 -0600, Adam B. Roach wrote:</div>
> <div>&gt;Since draft-ietf-mmusic-sdp-qos-00.txt expired in December, and
> is</div>
> <div>&gt;functionality we still need to standardize, it will (presumably)
> be </div>
> <div>&gt;reissued soon. Speaking on behalf of the PacketCable group, we
> have</div>
> <div>&gt;some suggestions of enhancements to the upcoming release of
> this</div>
> <div>&gt;document which would allow it to support a more robust network
> </div>
> <div>&gt;architecture.</div>
> <div>&gt;</div>
> <div>&gt;The current draft makes an assumption that end-to-end network
> </div>
> <div>&gt;resource reservation is controlled by a single entity. For
> calls</div>
> <div>&gt;being placed across adminstrative domains, however, this
> will</div>
> <div>&gt;not be the case. Under such circumstances, the reservation
> of</div>
> <div>&gt;resources needs a greater degree of coordination than is 
> </div>
> <div>&gt;currently provided in the draft.</div>
> <div>&gt;</div>
> <div>&gt;In particular, the calling user agent needs the ability
> to</div>
> <div>&gt;communicate to the called user agent that it has reserved
> </div>
> <div>&gt;the necessary resources on its end, and that alerting 
> may</div>
> <div>&gt;begin as soon as the called user agent has done the 
> same.</div>
> <div>&gt;</div>
> <div>&gt;Our proposed method of doing so is as follows:</div>
> <div>&gt;</div>
> <div>&gt;1) As in the current draft, the INVITE message contains</div>
> <div>&gt;&nbsp;&nbsp; a=qos:... lines to indicate the calling party's
> preferences</div>
> <div>&gt;&nbsp;&nbsp; for which streams have mandatory or optional
> reservation</div>
> <div>&gt;&nbsp;&nbsp; couplings associated with them.</div>
> <div>&gt;</div>
> <div>&gt;2) Similary, the 18x response contains a subset of the
> INVITE</div>
> <div>&gt;&nbsp;&nbsp; preconditions.</div>
> <div>&gt;</div>
> <div>&gt;3) The PRACK confirming the 18x will echo back the
> finally</div>
> <div>&gt;&nbsp;&nbsp; agreed-on preconditions. This will serve as a
> &quot;go ahead&quot;</div>
> <div>&gt;&nbsp;&nbsp; message to the called party to begin its actual
> resource</div>
> <div>&gt;&nbsp;&nbsp; reservation.</div>
> <div>&gt;</div>
> <div>&gt;4) After the calling side has procured reservations for
> the</div>
> <div>&gt;&nbsp;&nbsp; required resources, it sends an INFO message to the
> called</div>
> <div>&gt;&nbsp;&nbsp; party. This INFO message contains an SDP with a
> confirmation</div>
> <div>&gt;&nbsp;&nbsp; that the required streams have been reserved. For
> the purposes</div>
> <div>&gt;&nbsp;&nbsp; of this message, we add two values to the
> &quot;strength&quot; field</div>
> <div>&gt;&nbsp;&nbsp; in the current draft: &quot;reserved&quot; and
> &quot;resvfail&quot;. The presence</div>
> <div>&gt;&nbsp;&nbsp; of the &quot;resvfail&quot; token allows the called
> party to terminate</div>
> <div>&gt;&nbsp;&nbsp; the call with an error code (for mandatory streams)
> or to</div>
> <div>&gt;&nbsp;&nbsp; remove its local reservation (for optional
> streams).</div>
> <div>&gt;</div>
> <div>&gt;5) The 200 response to the INFO message similarly
> contains</div>
> <div>&gt;&nbsp;&nbsp; SDP indicating, on a per-stream basis, the status
> of</div>
> <div>&gt;&nbsp;&nbsp; its local reservations (allowing the calling party
> to</div>
> <div>&gt;&nbsp;&nbsp; remove reservations for optional streams which were
> not</div>
> <div>&gt;&nbsp;&nbsp; reserved on the terminating side).</div>
> <div>&gt;</div>
> <div>&gt;An example callflow follows (with unnecessary but present
> details, </div>
> <div>&gt;like tags and call-ids, removed). To prevent abiguity, I have
> inserted </div>
> <div>&gt;a &quot;Session&quot; header for all cases where SDP is present
> in a way that is </div>
> <div>&gt;not defined in the base specification and cannot be determined
> from</div>
> <div>&gt;context (e.g. the response to a QoS INFO message).</div>
> <div>&gt;</div>
> <div>&gt;--------------------</div>
> <div>&gt;UAC-&gt;UAS</div>
> <div>&gt;</div>
> <div>&gt;INVITE sip:adam.roach@ericsson.com SIP/2.0</div>
> <div>&gt;To: Adam Roach &lt;adam.roach@ericsson.com&gt;</div>
> <div>&gt;From: John Q. Public &lt;jpublic@ietf.org&gt;</div>
> <div>&gt;CSeq: 1 INVITE</div>
> <div>&gt;Content-Type: application/sdp</div>
> <div>&gt;</div>
> <div>&gt;v=0</div>
> <div>&gt;m=audio 48327 RTP/AVP 0</div>
> <div>&gt;a=qos:mandatory sendrecv</div>
> <div>&gt;m=video 10275 RTP/AVP 31</div>
> <div>&gt;a=qos:mandatory sendrecv</div>
> <div>&gt;--------------------</div>
> <div>&gt;UAS-&gt;UAC</div>
> <div>&gt;</div>
> <div>&gt;SIP/2.0 183 Preconditions</div>
> <div>&gt;To: Adam Roach &lt;adam.roach@ericsson.com&gt;</div>
> <div>&gt;From: John Q. Public &lt;jpublic@ietf.org&gt;</div>
> <div>&gt;CSeq: 1 INVITE</div>
> <div>&gt;Session: QoS</div>
> <div>&gt;Content-Type: application/sdp</div>
> <div>&gt;</div>
> <div>&gt;v=0</div>
> <div>&gt;m=audio 48327 RTP/AVP 0</div>
> <div>&gt;a=qos:mandatory sendrecv</div>
> <div>&gt;m=video 10275 RTP/AVP 31</div>
> <div>&gt;a=qos:optional sendrecv</div>
> <div>&gt;--------------------</div>
> <div>&gt;UAC-&gt;UAS</div>
> <div>&gt;</div>
> <div>&gt;PRACK sip:adam.roach@ericsson.com SIP/2.0</div>
> <div>&gt;To: Adam Roach &lt;adam.roach@ericsson.com&gt;</div>
> <div>&gt;From: John Q. Public &lt;jpublic@ietf.org&gt;</div>
> <div>&gt;CSeq: 2 PRACK</div>
> <div>&gt;Session: QoS</div>
> <div>&gt;Content-Type: application/sdp</div>
> <div>&gt;</div>
> <div>&gt;v=0</div>
> <div>&gt;m=audio 48327 RTP/AVP 0</div>
> <div>&gt;a=qos:mandatory sendrecv</div>
> <div>&gt;m=video 10275 RTP/AVP 31</div>
> <div>&gt;a=qos:optional sendrecv</div>
> <div>&gt;--------------------</div>
> <div>&gt;UAS-&gt;UAC</div>
> <div>&gt;</div>
> <div>&gt;SIP/2.0 200 OK</div>
> <div>&gt;To: Adam Roach &lt;adam.roach@ericsson.com&gt;</div>
> <div>&gt;From: John Q. Public &lt;jpublic@ietf.org&gt;</div>
> <div>&gt;CSeq: 2 PRACK</div>
> <div>&gt;Content-Length: 0</div>
> <div>&gt;--------------------</div>
> <div>&gt;Both parties perform resource reservation</div>
> <div>&gt;--------------------</div>
> <div>&gt;UAC-&gt;UAS</div>
> <div>&gt;</div>
> <div>&gt;INFO sip:adam.roach@ericsson.com SIP/2.0</div>
> <div>&gt;To: Adam Roach &lt;adam.roach@ericsson.com&gt;</div>
> <div>&gt;From: John Q. Public &lt;jpublic@ietf.org&gt;</div>
> <div>&gt;CSeq: 3 INFO</div>
> <div>&gt;Session: QoS</div>
> <div>&gt;Content-Type: application/sdp</div>
> <div>&gt;</div>
> <div>&gt;v=0</div>
> <div>&gt;m=audio 48327 RTP/AVP 0</div>
> <div>&gt;a=qos:reserved sendrecv</div>
> <div>&gt;m=video 10275 RTP/AVP 31</div>
> <div>&gt;a=qos:resvfail sendrecv</div>
> <div>&gt;--------------------</div>
> <div>&gt;UAS-&gt;UAC</div>
> <div>&gt;</div>
> <div>&gt;SIP/2.0 200 OK</div>
> <div>&gt;To: Adam Roach &lt;adam.roach@ericsson.com&gt;</div>
> <div>&gt;From: John Q. Public &lt;jpublic@ietf.org&gt;</div>
> <div>&gt;CSeq: 3 INFO</div>
> <div>&gt;Content-Type: application/sdp</div>
> <div>&gt;</div>
> <div>&gt;v=0</div>
> <div>&gt;m=audio 48327 RTP/AVP 0</div>
> <div>&gt;a=qos:reserved sendrecv</div>
> <div>&gt;m=video 10275 RTP/AVP 31</div>
> <div>&gt;a=qos:resvfail sendrecv</div>
> <div>&gt;--------------------</div>
> <div>&gt;The called party cancels the videostream reservation</div>
> <div>&gt;and begins alerting the called party.</div>
> <div>&gt;--------------------</div>
> <div>&gt;UAS-&gt;UAC</div>
> <div>&gt;</div>
> <div>&gt;SIP/2.0 180 Ringing</div>
> <div>&gt;To: Adam Roach &lt;adam.roach@ericsson.com&gt;</div>
> <div>&gt;From: John Q. Public &lt;jpublic@ietf.org&gt;</div>
> <div>&gt;CSeq: 1 INVITE</div>
> <div>&gt;Content-Length: 0</div>
> <div>&gt;--------------------</div>
> <div>&gt;</div>
> <div>&gt;-- </div>
> <div>&gt;Adam Roach, Ericsson Inc. |&nbsp; Ph: +1 972 583 7594 | 1010 E.
> Arapaho, MS L-04</div>
> <div>&gt;adam.roach@ericsson.com&nbsp;&nbsp; | Fax: +1 972 669 0154 |
> Richardson, TX 75081 USA</div>
> <div>&gt;</div>
> <div>&gt;</div>
> <br>
> 
> <div align="center">
> *************************************<br>
> &quot;At the end of the day... the most committed win!&quot;<br>
> <br>
> </div>
> James M. Polk<br>
> Sr. Product Manager, Multiservice Architecture and Standards<br>
> Enterprise Voice Business Unit<br>
> Cisco Systems<br>
> Dallas, Texas<br>
> w) 972.813.5208<br>
> f)&nbsp; 972.813.5280<br>
> <a href="http://www.cisco.com/" eudora="autourl">www.cisco.com</a></html>
> 
> --=====================_2567198==_.ALT--
> 
> 



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Jonathan Rosenberg wrote:
> The difficulty arises only with the segmented DQoS model in DCS, which
> does not provide a way to confirm resource reservation (I think),
> whereas RSVP does.
...          .....      ... The mechanism exists already now
> in RSVP, but not in DQoS (DCS folks, please correct me if I am
> mistaken).  

RSVP only provides this if (1) both endpoints support RSVP, and (2)
the backbone transports RSVP.  Neither is a valid assumption in
DQoS or DCS.

Further, please remember the address privacy requirements.  Whatever
the confirmation message looks like, the anonymizer must be able
to parse and rewrite it (just like it does for SIP messages) to update all
the addresses.  A single mechanism is much preferred.

Bill Marshall
wtm@research.att.com


----- original message -----
> 
> 
> "Adam B. Roach" wrote:
> > The current draft makes an assumption that end-to-end network 
> > resource reservation is controlled by a single entity. For calls
> > being placed across adminstrative domains, however, this will
> > not be the case. Under such circumstances, the reservation of
> > resources needs a greater degree of coordination than is 
> > currently provided in the draft.
> 
> Not really. Resource reservation is under the "control" of all the
> network elements along the path that perform some kind of admission
> control and resource allocation. These elements will often belong to
> different administrative domains. The draft makes no assumptions about
> the number of providers along the reservation path, or anything like
> that. It only ties resource reservation and signaling together at the
> endpoints, dicatating a particular temporal sequence.
> 
> > 
> > In particular, the calling user agent needs the ability to
> > communicate to the called user agent that it has reserved
> > the necessary resources on its end, and that alerting may
> > begin as soon as the called user agent has done the same.
> 
> This is something different from what you are discussing above. It is,
> in fact, covered in the draft. If A calls B, the RSVP RESV message from
> A to B confirms resources are reserved from A to B, and the RESVCONF
> from A to B confirms resource reservation from B to A. This allows B to
> know whether resources are reserved in both directions. A mirror case
> exists for A. 
> 
> The difficulty arises only with the segmented DQoS model in DCS, which
> does not provide a way to confirm resource reservation (I think),
> whereas RSVP does.
> 
> So, the question is this. We know we need a way for both parties to know
> when resources are reserved in both directions. Is this a function that
> belongs somewhere in SIP (through INFO, as you propose, or some other
> means), or in a reservation protocol. The mechanism exists already now
> in RSVP, but not in DQoS (DCS folks, please correct me if I am
> mistaken). So, do we add it to SIP, in which case its not needed for
> RSVP, or should it be added to DQoS, in which case the SIP operation is
> independent of the underlying reservation mechanism?
> 
> I have argued that confirmation of resource reservation is rightly a job
> for a resource reservation protocol. One of the things I like about SIP
> is its separation from session level control and network layer issues,
> such as resource reservation.
> 
> -Jonathan R.
> 
> -- 
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120 
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 



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Subject: Re: Accept-Contact Header
References: <6525688C.0045A19B.00@sampark.hss.hns.com> <38B16134.824A5194@dynamicsoft.com> <38B16A38.CE3F5187@hplb.hpl.hp.com>
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Thanks for catching this. Its been fixed.

-Jonathan R.

Anders Byttner wrote:
> 
> Just wanted to point out another error in this draft:
> 
> The proposed matching algorithm is not correct. Consider for example if
> we have a contact entry that is language="en,ge" and the rule is
> language="en,nl". Certainly we want this to be a match, since both
> parties speak one common language. But the rule is clearly not a subset
> of the entry . In the draft there is an example, which is not accurate
> following the above algorithm:
> 
> ;duplex="full,half"
> matches the contact entry
> ;duplex="full"
> 
> As we see the rule set is clearly not a subset of the entry set (rather
> a superset) and according to the proposed algorithm there should not be
> a match. But we want it to be a match (as stated in the example, and in
> the text above). The algorithm may instead be changed to the following:
> 
> - The parameter name is first checked. If it is not defined in the
> contact entry, the   parameter is ignored.
> - Parameter values are treated as sets, so the intersection of the rule
> set and the entry set   for this parameter is checked.
> - If the intersection is non-empty and the rule set is not negated we
> have a match.
> - If the intersection is empty and the rule set is negated we also have
> a match.
> 
> regards
> //Anders
> 
> Jonathan Rosenberg wrote:
> >
> > Thanks for pointing this out. The bnf appears to be wrong, as it
> > mandates a valid name-addr or addr-spec. The document clearly discusses
> > the capability of not specifying one. I will fix in the next version
> > (due out before the draft deadline). The BNF should be:
> >
> > Accept-Contact = "Accept-Contact" ":" 1#rule
> > rule   =  (name-addr | addr-spec | "*")
> > ....
> >
> > So, we'll use the * for anything, aligning with current Contact usage.
> >
> > -Jonathan R.
> >
> > rpreethy@hss.hns.com wrote:
> > >
> > > hi All,
> > >
> > > The document SIP Caller Preferences and Callee Capabilities -
> > > draft-ietf-sipcallerprefs-00.txt
> > > gives the grammer for Accept-Contact Header as
> > >
> > > Accept-Contact   =  "Accept-Contact" ":" 1# rule
> > >    rule             =  ( name-addr | addr-spec )
> > >                        [ *( ";" (cp-params | extension-param | q-param) ) ]
> > >    q-param          =  "q" "=" qvalue
> > >    extension-param  =  extension-name "=" extension-value
> > >    extension-name   =  token
> > >    extension-value  =  token | quoted-string
> > >
> > > and the example given is
> > >
> > > Accept-Contact: sip:sales@acme.com ;q=0,
> > >      ;media="!video" ;q=0.1,
> > >      ;mobility="fixed"  ;q=0.6,
> > >      ;mobility="!fixed" ;q=0.4
> > >
> > > According to my understanding, in this example each of the comma  separated
> > > values is a separate Accept Contact rule,
> > > And the parameters of a paticular accept contact rule are separated by ";"
> > > semicolon.
> > > Here each of the q-params are followed by a comma and then a semi-colon .
> > >
> > > I cannot understand whether to consider this as
> > > 1) a single accept contact rule with all the entries following
> > > sip:sales@acme.com as parameters in which case I dont see a need for a
> > > comma and semi-colon as separaters between parameters.
> > >
> > > 2) 4 rules where-in the 2nd 3rd and 4th do not have a nameaddr|addrspec
> > > entry which is also an error, becos the rule mandates the presence of
> > > nameaddr|addr-spec.
> > >
> > > Can someone explain this to me.
> > >
> > > Regards,
> > > Preethy
> >
> > --
> > Jonathan D. Rosenberg                       200 Executive Drive
> > Chief Scientist                             Suite 120
> > dynamicsoft                                 West Orange, NJ 07052
> > jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> > http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> > http://www.dynamicsoft.com
> 
> --
> --------------------------------------------------------------------------
> Anders Byttner                          Email:    <andbyt@hplb.hpl.hp.com>
> Hewlett-Packard Laboratories            Cellular: +44-(0)7901-805 479
> Internet Communication                  Work:     +44-(0)117-312 96 03
> 
> Filton Road, Stoke Grifford
> Bristol
> BS34 8QZ
> United Kingdom

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Folks,

Some colleagues and I have written and Internet Draft on getting SIP
through firewalls and NATs. It discusess the various configurations and
problems, and describes in detail the processing required to get things
working. The draft should appear in the archives shortly. Until it does,
you can pick up a copy at:

http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-sip-firewalls-00.txt

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb 23 02:33:51 2000
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Kim Liu wrote:
> 
> Section 6.40.2 states that a proxy should check that the address in
> the last Via: header matches the address that the request was actually
> received from.  If the addresses do not match, the proxy must add the
> received tag with the address that the request was actually received
> from.
> 
> My problem is with what happens if the address/hostname in the Via
> header matches several IP addresses, including the one actually used
> to send it, but not all of the addresses are routable, preferred, or
> public.
> 
> That is, suppose a server receives a Via header like:
> 
>   Via: SIP/2.0/UDP aka.generic.com:5060
> 
> in a UDP packet with an originating address of 10.100.100.100.
> The server looks up the IP addresses for aka.generic.com to see
> whether or not 10.100.100.100 is one of its addresses.  The
> returned addresses for aka.generic.com are:
> 
>   192.168.128.1, 192.168.128.2, 192.168.128.3, 192.168.128.3, and
>   10.100.100.100.
> 
> So, since the 10.100.100.100 address is present, no received tag is
> added to the Via field.  Now, when the response comes back through
> the proxy, and the proxy tries to forward it back along the Via
> lines, which IP address should the proxy choose to send the UDP
> message to?  (Of course, the 192.* addresses might be public
> interfaces, and the 10.100.100.100 address is on a private network
> to keep call information private, etc.)  Could a recieved= tag
> be added to the Via: header anyways to be used to tell the proxy
> exactly which IP address the response should be sent to?

I would not recommend that. The reason for having a hostname in the Via
field as an option is to add a layer of indirection. This would allow a
set of proxy servers to be used for load balancing purposes through DNS
round robining. In many cases, its not actually important for the
response to even go back to the same server the request came from (not
important at all for stateless proxies). Thus, my recommendation would
be:

  If the Via contains a hostname, and the IP address the request came
from is one of the addresses returned   from DNS query for that
hostname, no received tag is added.

If your proxy is such that responses should always be sent to just that
proxy, well, then don't put multiple addresses for it in DNS, or don't
put a hostname in the Via header.

Also, in your example above, DNS queries from outside the network the
server sits in are never going to return a net 10 address. Thus, if a
request has passed through a NAT, and the source IP address is a net 10
address, and the Via header contained a hostname, you can be sure the
received tag will be inserted. This is also what you want.


> 
> Also, I think I may have seen this question earlier, but I cannot
> find it now (is there a mailing list archive somewhere?)

Yes. Its linked from the sip page at IETF. Its:
http://www.bell-labs.com/mailing-lists/sip 

> -- how does
> the Via:/received affect loop detection? 

It doesn't.

> The example given in
> the draft indicates that one use of the received tag was for when
> a SIP request is passing through NAT.  When checking for loops
> and a received tag is present, should only the address in the
> received tag be checked?  

The received tag is not used for loop detection. A proxy looks for Via
fields that contain its own address that it inserted into the sent-by
field. Some recent posts to the list discuss additional checks that must
be made to avoid false classification of loops when calls are forwarded
to other users on the same network.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb 23 04:11:56 2000
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These questions have probably been discussed in the past - I will ask one
more time.

1) Does To header have any role in call routing ( from a proxy's 
standpoint) - I mean in determining the next hop.

2) A stateful proxy receives a retransmitted request for which it 
**proxied** a 200 OK recently. Should it re-proxy the 200 OK ?

Thanks,
Shail



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Shail Bhatnagar wrote:
> 
> These questions have probably been discussed in the past - I will ask one
> more time.
> 
> 1) Does To header have any role in call routing ( from a proxy's
> standpoint) - I mean in determining the next hop.

Normally, no. When using the registration database, it is the request
URI that is used as a key. The request URI is the entity being addressed
at that particular server, so that is what will generally be used.
However, no one can dictate the logic used by a proxy for determining
how a call should be handled. Certainly its possible that the To field
might be used for some kind of policy decision.

> 
> 2) A stateful proxy receives a retransmitted request for which it
> **proxied** a 200 OK recently. Should it re-proxy the 200 OK ?

No. Retransmissions of 200 OK to INVITE are handled end to end.

-Jonathan R.



-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb 23 11:02:15 2000
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William Marshall writes:
 > This is exactly the point.  In general there will be three (or more) service
 > providers involved in establishing the QoS, and their service level agreements
 > may be (and in some cases definitely are) different.  The router at the
 > edge of each service provider's network needs to convert the resource
 > reservation request into something that the next service provider is willing
 > to handle; in general this may not always be RSVP.  Depending on end-to-end
 > RSVP for a successful SIP connection would be an unfortunate dependency.

I think that this is somewhat speculative at this
point because backbone providers up until recently
haven't had a business case to provide
differential treatment of EF traffic, say, through
their network. If there's a business case for RSVP
and a solution that will scale to their network,
I'm sure there will be providers who offer it.

But I'm not even sure that it is the case that the
backbone provider even needs to participate in
RSVP in the "normal" case (other than carrying the
datagram over their network). Suppose we have the
following network:

UAC->[cloud]->Border<==>Border->[BB cloud]->Border<==>Border->[cloud]->UAC

Further, suppose that there is a SLA across the
backbone cloud which gives 20% EF traffic
guarantees. In this case, the backbone provider's
border router (or any for that matter) does not
need to participate in RSVP because the border
router at the access network will do the flow
admission control at its backbone border router.
The BB router's job is to traffic shape the SLA,
but it needs to do that anyway. The access
network's border router needs to make certain that
the admission control operation takes into account
the SLA with the backbone operator, which it needs
to do anyway as well.

It's even better with aggregated RSVP since it
allows multiple levels of aggregation: between the
access router and the border router, flows could
be aggregated to relieve the micro flow load at
the border router. Since aggregated RSVP "tunnels"
the "end to end" RSVP requests to the next
aggregation point, we can use the same technique
to tunnel the RSVP request across the backbone so
that it doesn't matter whether the BB provider
does or does not allow/act on RSVP in their
network -- it will be transparent to them (and in
fact, the RSVP requests should use integrity
objects to disallow unauthorized use).

I believe this extends to the general case where
you have more than one BB provider too: so long as
the intermediate providers honor the SLA that the
initial BB provider sold to the access provider,
all is well. There doesn't need to be any
admission control in center because the admission
control was done at the choke point. I believe
this also extends to non-static SLA's as well if
we want to use RSVP in the core: with the use of
RSVP and COPS at the access provider's border,
large(r) pipes can be created which so long as
they can be billed correctly would provide a
dynamic SLA capability too which doesn't even
exist today that I'm aware of.

All in all, RSVP+Policy+Aggregation gives a lot of
flexibility.

		Mike



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Date: Wed, 23 Feb 2000 10:57:00 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: DHCP option tags for SIP
In-reply-to: <38AEDBC8.E3D77507@cs.columbia.edu>
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In actually doing this stuff, I can say that our experimental deployment
model is that the phone registers with an "identity" value which is
effectively unique to the phone. This "identity" is associated with users
(domain addresses) and phone numbers in a profile schema (directory). We
provide a web interface for managing the profile, or one can manipulate it
with SIP registrations. This allows users to forward their phones, move
them, clone them, sell them and buy new ones, etc.

The phone is not the user, nor is it the phone number. It's just a piece of
gear that may (or may not) be on the contact list for a given request URI as
that request URI evolves from the original request through processing by
proxies, etc.

--
Dean

> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Henning
> Schulzrinne
> Sent: Saturday, February 19, 2000 12:07 PM
> To: Michael Thomas
> Cc: sip@lists.research.bell-labs.com
> Subject: Re: DHCP option tags for SIP
>
>
> Michael Thomas wrote:
> >
> > Henning Schulzrinne writes:
> >  > As far as I can tell from our implementation experience,
> this is pretty
> >  > much (*) the last piece of information needed to configure a
> phone. At
> >  > the DHCP level, we currently use for a basic phone:
> >  >
> >  > - IP address, DNS server address, default gateway (standard
> DHCP fare)
> >  > - time server address
> >  > - outbound SIP proxy server
> >  > - normal boot image information for phones with downloadable code
> >
> >    Dumb question: would a phone need its E.164 address(es)
> >    when registering, etc?
>
> Actually, an interesting question. A phone would need the identity with
> which it should register. From what I can tell, PBX will probably have
> two identities for their phones, one "legacy" (E.164) identity and one
> or more SIP identities (email address). It could be more than one if two
> people share a phone, as our grad students do.
>
> It might even have a third identity reflecting the phone's location,
> e.g., "Room2F524@somewhere.com". The first two types of identities would
> change if a person moved into a different office (assuming that she
> leaves her phone behind), while the latter would stay constant.
> Conversely, if people take their phones with them, only the last
> identity would change.
>
> Since these identities don't generally change all that often, this could
> be hard-coded. (Fancier mechanisms, such as identities conveyed by
> PalmPilots, have been discussed elsewhere.) Alternatively, a few of us
> have discussed that there might well be a need for an additional
> configuration parameter, namely a list of "identities" that the phone
> should register with. Pretty straightforward to define, but I kind of
> wanted to get the server issue out of the way first.
>
>
> >
> >  > There are other SIP-level configurations that are
> inappropriate, in my
> >  > view, for DHCP, things like speed dial configurations or other UI
> >  > assignments, any end system CPL, etc.
> >
> >    This, however, was my point. If you believe in directories
> >    for configless widgets, it would make a lot more sense to
> >    dump all of this information there and have a pointer to
> >    the directory in DHCP. If the sum total of configuration
> >    that a configurationless device is greater than just the
> >    address of the first hop proxy, then all this new
> >    option is doing is solving a small part of a previously
> >    unsolved problem.
>
> Our current intent is to put some of this into the REGISTER response,
> using an appropriate MIME type, since it avoids having to define yet
> another directory protocol, more servers, etc. (Putting LDAP into an
> embedded system strikes me as challenging, to put it mildly.)
>
> However, the outbound proxy can be different from my registration
> server.
>
> Indeed, in order to find my non-multicast registration server, I need to
> know my identity (see discussion above).
>
> Thus, overall, I see DHCP used for things that are needed to bootstrap
> into the SIP infrastructure with minimum cost and without creating
> seventeen new directory servers. Almost all SOHO-style integrated
> devices support DHCP, for obvious reasons, while installing and
> configuring LDAP, say, seems like a stretch for this type of
> application. (You don't mention a specific directory service, so pardon
> me if I'm misinterpreting your remarks.)
>
> Overall, the larger the number of servers, the higher the probability
> that one of the alphabet soup servers is down, misconfigured. In my
> view, it would be nice if I can install a basic Internet PBX with two
> servers, a DHCP server and a SIP server. These seem to be required,
> adding anything else as a basic requirement to make phones ring seems
> like it needs a fair amount of justification for the complexity. This
> does not mean that more complicated scenarios can't make good use of SLP
> or LDAP. (We use LDAP in our SIP server and PC client, for example.)
>
> >
> >                 Mike
>
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
>




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Date: Wed, 23 Feb 2000 10:41:53 -0600
From: Michelle Kitchings <Michelle.Kitchings@exu.ericsson.se>
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Cc: Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
        Anders Kristensen <ak@hplb.hpl.hp.com>,
        SIP <sip@lists.research.bell-labs.com>
Subject: Re: Extension negotiation draft
References: <38A85EA3.6DB9A152@hplb.hpl.hp.com> <38B18BCC.C9028997@dynamicsoft.com> <38B194DE.183CA9DF@cs.columbia.edu> <38B19EB8.68CC1B10@dynamicsoft.com>
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Jonathan Rosenberg wrote:
...
> 
> So, the current proposal is to write a draft that:
> 
> 1. Defines the Supported header syntax
> 2. specifies its usage in OPTIONS responses
> 3. specifies its usage in requests - basically, it lists all features
> with short names
> 4. supports no negotiation
> 
> This draft will be obsoleted by rfc2543bis when it goes to rfc, but no
> way we can wait that long for this.
> 
> How do people feel about this? If we get consensus, I'll do another rev
> on the document and we can push it out the door.
> 
> -Jonathan R.

FWIW, I like it!
(I especially like #3 :)

Michelle :^)

-- 
Michelle Kitchings | Ph:  +1 972 583 7101 | 1010 E. Arapaho, MS L-04
Ericsson, Inc.     | Fax: +1 972 669 0154 | Richardson, TX 75081 USA



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb 23 13:10:02 2000
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Jonathan Rosenberg wrote:
> 

> >
> > 2) A stateful proxy receives a retransmitted request for which it
> > **proxied** a 200 OK recently. Should it re-proxy the 200 OK ?
> 
> No. Retransmissions of 200 OK to INVITE are handled end to end.
> 
> -Jonathan R.
> 

Yes, but what should the stateful proxy do with this re-transmitted INVITE.
Forward it on or simply drop it and wait for UAS re-transmission logic to
retransmit the 200 OK  or re-send the last provisional it sent upstream.

Thanks,
Shail



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb 23 14:03:59 2000
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Shail Bhatnagar wrote:
> 
> Jonathan Rosenberg wrote:
> >
> 
> > >
> > > 2) A stateful proxy receives a retransmitted request for which it
> > > **proxied** a 200 OK recently. Should it re-proxy the 200 OK ?
> >
> > No. Retransmissions of 200 OK to INVITE are handled end to end.
> >
> > -Jonathan R.
> >
> 
> Yes, but what should the stateful proxy do with this re-transmitted INVITE.
> Forward it on or simply drop it and wait for UAS re-transmission logic to
> retransmit the 200 OK  or re-send the last provisional it sent upstream.

The proxy can either drop it or send it. Its not needed to ensure
reliability. The 200 
retransmissions at the UAS will handle this.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb 23 15:21:44 2000
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Date: Tue, 22 Feb 2000 22:56:33 -0800 (PST)
From: Steve Little <steve_sip@yahoo.com>
Subject: Diff. between branch and tag
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Hi,
What is the difference between inserting a branch
param in a Via and a tag in To, From headers ?
Dont both uniquely identify req-resp for forking
proxies ?

Thx

__________________________________________________
Do You Yahoo!?
Talk to your friends online with Yahoo! Messenger.
http://im.yahoo.com



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb 23 15:33:59 2000
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Organization: dynamicsoft
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Branch IDs allow proxies to match responses to forked requests. Without
them, a proxy wouldn't be able to tell which branch a response
corresponds to. Tags are of no help here since they are not known until
responses arrive.

---
Igor Slepchin


Steve Little wrote:
> 
> Hi,
> What is the difference between inserting a branch
> param in a Via and a tag in To, From headers ?
> Dont both uniquely identify req-resp for forking
> proxies ?
> 
> Thx
> 
> __________________________________________________
> Do You Yahoo!?
> Talk to your friends online with Yahoo! Messenger.
> http://im.yahoo.com



From owner-sip-outgoing@lists.research.bell-labs.com  Wed Feb 23 17:40:11 2000
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Date: Wed, 23 Feb 2000 16:33:40 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: SIP BCP-T question
In-reply-to: 
 <153BDA136259D311859900A0C99B5263027750@boca-isbu-nt04.boca.unisphere.com>
To: "Carr, Steve" <sCarr@unispheresolutions.com>,
        "'Raja Badri'" <raja.badri@santera.com>,
        sip ML <sip@lists.research.bell-labs.com>
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It seems that these might really be "no-op" operations rather than
"nonexistent" operations.

There's a difference.

--
Dean

> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Carr, Steve
> Sent: Monday, February 14, 2000 12:25 PM
> To: 'Raja Badri'; sip ML
> Subject: RE: SIP BCP-T question
>
>
> The ISUP maintenance messages relate to the physical trunk circuit, which
> when not being used on a call, is still physically present and in an IDLE
> (or blocked) state. SIP BCT-T sets up calls over an RTP "logical"
> connection, which is not a physical circuit. After the end of the
> call, the
> RTP connection ceases to exist. Therefore is does not make sense to block
> it, unblock it, reset it etc......
>
> Steve Carr
> Unisphere Solutions Inc.
> Boca Raton
> scarr@unispheresolutions.com
> Office (561) 955-8288
> Fax    (561) 955-6477
>
> www.unispheresolutions.com
>
> -----Original Message-----
> From: Raja Badri [mailto:raja.badri@santera.com]
> Sent: Monday, February 14, 2000 1:04 PM
> To: sip ML
> Subject: SIP BCP-T question
>
>
> Hi,
>
> I have one question regarding the SIP BCP-T.
> INFO message outlines the Mid-Call ISUP messages.
> What is the reason behind not including the ISUP maintenance
> messages like Block, Unblock and Reset ?
>
> Thanks,
> Raja Badri
> Santera Systems Inc.
>
>
>




From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 24 00:29:36 2000
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Tags are inserted by UASs, not proxies, and are needed to allow UAC to
distinguish multiple 200 OK responses sent by several UASs in respponse
to a forked request or, equivalently, to distinguish multiple instances
of a user identified by the URL in the request URI of original request.
Tags are _not_ used to match responses to forked requests at the
proxies.

Steve Little wrote:
> Ive have seem implementations insert a tag field
> irrespective of whether it forks or not.

UAS has no reliable way of determining if the request has been forked or
not. Thus, to be safe it almost needs to add a tag. Proxies only insert
tags into the final responses they generate themselves; they never
insert tags into requests or responses they forward.

> Also, if forking proxies insert tags while forwarding
> the request, then isnt that enough ? Then when a
> receiving entity gets the request, it responds and
> copies the tag in it. 

Since a request can be forked several times on its way to UAS, a single
"tag" (or whatever you like to call it) added to the request by one of
the proxies is not sufficient for the next forking proxy along the chain
to match responses on its own branches; every proxy that forked the
request would need to add its own unique IDs to the branches it created.
This is precisely what's being achieved by branch parameter in Via
header.

> why should a reciever try and figure out if
> its forked ? 

It should not and, moreover, cannot (I assume a receiver here means
UAS). Instead, it should always add tags to all responses it generates.

---
Igor Slepchin



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 24 01:44:02 2000
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Message-ID: <38B4D36E.329DE08A@dynamicsoft.com>
Date: Thu, 24 Feb 2000 01:45:02 -0500
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Folks,

There has been some discussion in response to our proposal to fast track
a DHCP option for SIP. This discussion generally indicated that DHCP
would not be sufficient for autoconfiguration. We (the wg chairs) think
a DHCP option for finding a default proxy is a useful thing, even
if it is not a complete solution. We therefore propose to define the
DHCP
option, making it available for implementation, and carry on the dialog
on
managing SIP endpoints as a continuing thread.

As such, this message will serve as a working group last call on the
DHCP options document:
http://search.ietf.org/internet-drafts/draft-nair-sip-dhcp-01.txt

We are trying to coordinate a parallel last call on the dhcp list, which
must review the document as well.

The working group last call will last three weeks. Please send comments
to the list or to the authors during this time.

Thanks,
Jonathan R.




-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 24 02:59:40 2000
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Dean Willis wrote:
> 
> In actually doing this stuff, I can say that our experimental deployment
> model is that the phone registers with an "identity" value which is
> effectively unique to the phone. This "identity" is associated with users
> (domain addresses) and phone numbers in a profile schema (directory). We
> provide a web interface for managing the profile, or one can manipulate it
> with SIP registrations. This allows users to forward their phones, move
> them, clone them, sell them and buy new ones, etc.

If I understand you right, the phone would have an ID, something like
sip:phone3736489@wcom.com. When plugged in, it automatically registers
this address, binding it to its current IP address. In addition, users
can go to a web page (or use a SIP registration), and basically create
another binding, mapping their name (sip:dean.willis@wcom.com) to the ID
of the phone (sip:3736489@wcom.com). Is that right?

Neat idea... solves the problem that a standalone phone is not always
going to have an easy way for the user to enter their name that it
should use in a registration. It is basically adding another layer of
indirection: user->device->address, rather than just user->address.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 24 03:05:59 2000
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William Marshall wrote:
> 
> Jonathan Rosenberg wrote:
> > The difficulty arises only with the segmented DQoS model in DCS, which
> > does not provide a way to confirm resource reservation (I think),
> > whereas RSVP does.
> ...          .....      ... The mechanism exists already now
> > in RSVP, but not in DQoS (DCS folks, please correct me if I am
> > mistaken).
> 
> 
> Further, please remember the address privacy requirements.  Whatever
> the confirmation message looks like, the anonymizer must be able
> to parse and rewrite it (just like it does for SIP messages) to update all
> the addresses.  A single mechanism is much preferred.

Well, an anonymizer can't have just a single protocol mechanism. By its
very definition, the anonymizer must handle both SIP and RTP at least,
which is already two protocols. Since its handling media streams, it
must also initiate and terminate QoS reservations if you want voice
quality through the anonymizer. So, its already doing resource
reservation. If the resource reservation protocol is providing
confirmations, its not any extra protocol work as compared to putting
these confirmations in the signaling protocol.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 24 03:58:05 2000
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Date: Thu, 24 Feb 2000 00:58:03 -0800
To: Dean Willis <dean.willis@wcom.com>
From: Taichi Fu <tfu@cisco.com>
Subject: RE: SIP BCP-T question
Cc: "Carr, Steve" <sCarr@unispheresolutions.com>,
        "'Raja Badri'" <raja.badri@santera.com>,
        sip ML <sip@lists.research.bell-labs.com>
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--=====================_3832580==_.ALT
Content-Type: text/plain; charset="us-ascii"; format=flowed

 From BCP-T:
    "This document only takes into account the call control
    functionality of ISUP. Maintenance messages are outside
    the scope of this document."

Suppose that messages such as Circuit Reset may also be received during a 
call setup
phase (to abort the setup), then should they also be mapped to such as BYE 
or CANCEL
(by PSTN GW)?

Thanks,
-Taichi


At 04:33 PM 2/23/00 -0600, Dean Willis wrote:
>It seems that these might really be "no-op" operations rather than
>"nonexistent" operations.
>
>There's a difference.
>
>--
>Dean
>
> > -----Original Message-----
> > From: owner-sip@lists.research.bell-labs.com
> > [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Carr, Steve
> > Sent: Monday, February 14, 2000 12:25 PM
> > To: 'Raja Badri'; sip ML
> > Subject: RE: SIP BCP-T question
> >
> >
> > The ISUP maintenance messages relate to the physical trunk circuit, which
> > when not being used on a call, is still physically present and in an IDLE
> > (or blocked) state. SIP BCT-T sets up calls over an RTP "logical"
> > connection, which is not a physical circuit. After the end of the
> > call, the
> > RTP connection ceases to exist. Therefore is does not make sense to block
> > it, unblock it, reset it etc......
> >
> > Steve Carr
> > Unisphere Solutions Inc.
> > Boca Raton
> > scarr@unispheresolutions.com
> > Office (561) 955-8288
> > Fax    (561) 955-6477
> >
> > www.unispheresolutions.com
> >
> > -----Original Message-----
> > From: Raja Badri [mailto:raja.badri@santera.com]
> > Sent: Monday, February 14, 2000 1:04 PM
> > To: sip ML
> > Subject: SIP BCP-T question
> >
> >
> > Hi,
> >
> > I have one question regarding the SIP BCP-T.
> > INFO message outlines the Mid-Call ISUP messages.
> > What is the reason behind not including the ISUP maintenance
> > messages like Block, Unblock and Reset ?
> >
> > Thanks,
> > Raja Badri
> > Santera Systems Inc.
> >
> >
> >
>
>

--=====================_3832580==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
 From BCP-T:<br>
<font face="Courier New, Courier" size=4>&nbsp;&nbsp; &quot;This document
only takes into account the call control<br>
&nbsp;&nbsp; functionality of ISUP. Maintenance messages are 
outside<br>
&nbsp;&nbsp; the scope of this document.&quot;<br>
<br>
</font>Suppose that messages such as Circuit Reset may also be received
during a call setup<br>
phase (to abort the setup), then should they also be mapped to such as
BYE or CANCEL<br>
(by PSTN GW)?<br>
<br>
Thanks,<br>
-Taichi<br>
<br>
<br>
At 04:33 PM 2/23/00 -0600, Dean Willis wrote:<br>
<blockquote type=cite cite>It seems that these might really be
&quot;no-op&quot; operations rather than<br>
&quot;nonexistent&quot; operations.<br>
<br>
There's a difference.<br>
<br>
--<br>
Dean<br>
<br>
&gt; -----Original Message-----<br>
&gt; From: owner-sip@lists.research.bell-labs.com<br>
&gt;
[<a href="mailto:owner-sip@lists.research.bell-labs.com%5DOn" eudora="autourl">mailto:owner-sip@lists.research.bell-labs.com]On</a>
Behalf Of Carr, Steve<br>
&gt; Sent: Monday, February 14, 2000 12:25 PM<br>
&gt; To: 'Raja Badri'; sip ML<br>
&gt; Subject: RE: SIP BCP-T question<br>
&gt;<br>
&gt;<br>
&gt; The ISUP maintenance messages relate to the physical trunk circuit,
which<br>
&gt; when not being used on a call, is still physically present and in an
IDLE<br>
&gt; (or blocked) state. SIP BCT-T sets up calls over an RTP
&quot;logical&quot;<br>
&gt; connection, which is not a physical circuit. After the end of
the<br>
&gt; call, the<br>
&gt; RTP connection ceases to exist. Therefore is does not make sense to
block<br>
&gt; it, unblock it, reset it etc......<br>
&gt;<br>
&gt; Steve Carr<br>
&gt; Unisphere Solutions Inc.<br>
&gt; Boca Raton<br>
&gt; scarr@unispheresolutions.com<br>
&gt; Office (561) 955-8288<br>
&gt; Fax&nbsp;&nbsp;&nbsp; (561) 955-6477<br>
&gt;<br>
&gt;
<a href="http://www.unispheresolutions.com/" eudora="autourl">www.unispheresolutions.com</a><br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: Raja Badri
[<a href="mailto:raja.badri@santera.com" eudora="autourl">mailto:raja.badri@santera.com</a>]<br>
&gt; Sent: Monday, February 14, 2000 1:04 PM<br>
&gt; To: sip ML<br>
&gt; Subject: SIP BCP-T question<br>
&gt;<br>
&gt;<br>
&gt; Hi,<br>
&gt;<br>
&gt; I have one question regarding the SIP BCP-T.<br>
&gt; INFO message outlines the Mid-Call ISUP messages.<br>
&gt; What is the reason behind not including the ISUP maintenance<br>
&gt; messages like Block, Unblock and Reset ?<br>
&gt;<br>
&gt; Thanks,<br>
&gt; Raja Badri<br>
&gt; Santera Systems Inc.<br>
&gt;<br>
&gt;<br>
&gt;<br>
<br>
<br>
</blockquote></html>

--=====================_3832580==_.ALT--




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Message-ID: <38B4E85B.5C43DA7C@dynamicsoft.com>
Date: Thu, 24 Feb 2000 03:14:19 -0500
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Folks,

Note that an updated INFO draft has been submitted:
http://www.ietf.org/internet-drafts/draft-ietf-sip-info-method-02.txt

This incorporates comments received during working group last call. I
believe this document is ready to be sent to IESG for proposed. I will
do so Friday unless someone objects. 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com




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Date: Thu, 24 Feb 2000 09:33:55 -0500
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: Dean Willis <dean.willis@wcom.com>, Michael Thomas <mat@cisco.com>,
        sip@lists.research.bell-labs.com
Subject: Re: DHCP option tags for SIP
References: <003101bf7e1f$007ad880$a6e123a6@mcit.com> <38B4E482.AA6F472F@dynamicsoft.com>
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Jonathan Rosenberg wrote:
> 
> Dean Willis wrote:
> >
> > In actually doing this stuff, I can say that our experimental deployment
> > model is that the phone registers with an "identity" value which is
> > effectively unique to the phone. This "identity" is associated with users
> > (domain addresses) and phone numbers in a profile schema (directory). We
> > provide a web interface for managing the profile, or one can manipulate it
> > with SIP registrations. This allows users to forward their phones, move
> > them, clone them, sell them and buy new ones, etc.
> 
> If I understand you right, the phone would have an ID, something like
> sip:phone3736489@wcom.com. When plugged in, it automatically registers
> this address, binding it to its current IP address. In addition, users
> can go to a web page (or use a SIP registration), and basically create
> another binding, mapping their name (sip:dean.willis@wcom.com) to the ID
> of the phone (sip:3736489@wcom.com). Is that right?
> 
> Neat idea... solves the problem that a standalone phone is not always
> going to have an easy way for the user to enter their name that it
> should use in a registration. It is basically adding another layer of
> indirection: user->device->address, rather than just user->address.

There's an equivalent way of doing this, namely using the hardware ID
(aka MAC address). This also provides some level of authentication. One
possible problem of having a "back-channel" registration mechanism is to
make sure that the lifetimes of the registrations are lined up, so that
my user registration goes away if the low-level device registration goes
away. For that reason, I had suggested a configuration mapping where the
phone obtains a list of identities based on its hardware ID and
REGISTERs all of them. This also allows a web-based configuration
mechanism, but doesn't require anything special on the SIP server.

Either, I believe, can be made to work, depending on the assumptions as
to what's changeable, what's fixed and what's easier to integrate.

Beyond these details, I agree that we need to look at registration and
configuration more closely. We had separately discussed the "hotel
problem", for example.

A related problem, discussed earlier, is the current problem of
registrar crashes. In our local IP phone configuration, we have had the
problem of registrars crashing and then having to wait an hour until all
the registrations are back.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



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From: "Joshua Moloney" <jmoloney@nortelnetworks.com>
To: "'sip@lists.research.bell-labs.com'" <sip@lists.research.bell-labs.com>
Subject: Content-Encoding vs. Content-Transfer-Encoding
Date: Thu, 24 Feb 2000 15:15:18 -0000
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Hi all,

I have a question about the "Content-Encoding" entity header.
Why is the MIME "Content-Transfer-Encoding" header defined in RFC 2045 not
used instead?
Am I correct in assuming that they do the same job?


For example, what headers should I include on a message with an
"application/ISUP" message body? :-

Content-Type: application/ISUP
Content-Transfer-Encoding: binary
.....

OR

Content-Type: application/ISUP
Content-Encoding: binary
.....

Thanks in advance,

Josh Moloney

Nortel Networks
jmoloney@nortelnetworks.com



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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2651.65">
<TITLE>Content-Encoding vs. Content-Transfer-Encoding</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2 FACE=3D"Arial">Hi all,</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I have a question about the =
&quot;Content-Encoding&quot; entity header.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Why is the MIME =
&quot;Content-Transfer-Encoding&quot; header defined in RFC 2045 not =
used instead?</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Am I correct in assuming that they do =
the same job?</FONT>
</P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Arial">For example, what headers should I =
include on a message with an &quot;application/ISUP&quot; message body? =
:-</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Content-Type: application/ISUP</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Content-Transfer-Encoding: =
binary</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">.....</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">OR</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Content-Type: application/ISUP</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Content-Encoding: binary</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">.....</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Thanks in advance,</FONT>
</P>

<P><FONT COLOR=3D"#000000" SIZE=3D2 FACE=3D"Arial">Josh Moloney</FONT>
</P>

<P><B><FONT COLOR=3D"#808080" SIZE=3D2 FACE=3D"Tahoma">Nortel =
Networks</FONT></B>
<BR><B><FONT COLOR=3D"#C0C0C0" SIZE=3D2 =
FACE=3D"Tahoma">jmoloney@nortelnetworks.com</FONT></B>
</P>
<BR>

</BODY>
</HTML>
------_=_NextPart_001_01BF7ED9.FAEFFF7E--



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 24 10:46:12 2000
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Date: Thu, 24 Feb 2000 09:41:02 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: DHCP option tags for SIP
In-reply-to: <38B54153.3A345711@cs.columbia.edu>
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: Michael Thomas <mat@cisco.com>, sip@lists.research.bell-labs.com
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We thought about this, but didn't like it much given the way that our
experimental provisioning systems and UASes work. For example, my PC sip
client has nine different MAC addresses depending on where I'm sitting. Much
of the time it has two simultaneously, and sometimes it has three.  Also,
sometimes it is not the same physical device at all, but I want it to appear
to be (identity sharing), so there might be multiple sets of nine different
MAC addresses which are all "the same phone".

Given that the device identity (in the proposal, a MAC address) really has
to be distinctively inserted into the routing system, use of the MAC address
in this environment could get really messy.

We decided it was just easier to assign distinguished names to the devices,
and put only the distniguished name into the routing system.

Now, this might be very different in an environment where the UAS-of-choice
is a hardware phone that is persistent in its association with a singular
MAC address. I think Henning's approach has the potential to work quite well
here.

--
Dean

> -----Original Message-----
> From: hgs@cs.columbia.edu [mailto:hgs@cs.columbia.edu]On Behalf Of
> Henning Schulzrinne
> Sent: Thursday, February 24, 2000 8:34 AM
> To: Jonathan Rosenberg
> Cc: Dean Willis; Michael Thomas; sip@lists.research.bell-labs.com
> Subject: Re: DHCP option tags for SIP
>
>
> Jonathan Rosenberg wrote:
> >
> > Dean Willis wrote:
> > >
> > > In actually doing this stuff, I can say that our experimental
> deployment
> > > model is that the phone registers with an "identity" value which is
> > > effectively unique to the phone. This "identity" is
> associated with users
> > > (domain addresses) and phone numbers in a profile schema
> (directory). We
> > > provide a web interface for managing the profile, or one can
> manipulate it
> > > with SIP registrations. This allows users to forward their
> phones, move
> > > them, clone them, sell them and buy new ones, etc.
> >
> > If I understand you right, the phone would have an ID, something like
> > sip:phone3736489@wcom.com. When plugged in, it automatically registers
> > this address, binding it to its current IP address. In addition, users
> > can go to a web page (or use a SIP registration), and basically create
> > another binding, mapping their name (sip:dean.willis@wcom.com) to the ID
> > of the phone (sip:3736489@wcom.com). Is that right?
> >
> > Neat idea... solves the problem that a standalone phone is not always
> > going to have an easy way for the user to enter their name that it
> > should use in a registration. It is basically adding another layer of
> > indirection: user->device->address, rather than just user->address.
>
> There's an equivalent way of doing this, namely using the hardware ID
> (aka MAC address). This also provides some level of authentication. One
> possible problem of having a "back-channel" registration mechanism is to
> make sure that the lifetimes of the registrations are lined up, so that
> my user registration goes away if the low-level device registration goes
> away. For that reason, I had suggested a configuration mapping where the
> phone obtains a list of identities based on its hardware ID and
> REGISTERs all of them. This also allows a web-based configuration
> mechanism, but doesn't require anything special on the SIP server.
>
> Either, I believe, can be made to work, depending on the assumptions as
> to what's changeable, what's fixed and what's easier to integrate.
>
> Beyond these details, I agree that we need to look at registration and
> configuration more closely. We had separately discussed the "hotel
> problem", for example.
>
> A related problem, discussed earlier, is the current problem of
> registrar crashes. In our local IP phone configuration, we have had the
> problem of registrars crashing and then having to wait an hour until all
> the registrations are back.
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
>




From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 24 10:57:51 2000
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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To: Dean Willis <dean.willis@wcom.com>
Cc: sip@lists.research.bell-labs.com
Subject: Re: DHCP option tags for SIP
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Dean Willis wrote:
> 
> We thought about this, but didn't like it much given the way that our
> experimental provisioning systems and UASes work. For example, my PC sip
> client has nine different MAC addresses depending on where I'm sitting. Much
> of the time it has two simultaneously, and sometimes it has three.  Also,
> sometimes it is not the same physical device at all, but I want it to appear
> to be (identity sharing), so there might be multiple sets of nine different
> MAC addresses which are all "the same phone".
> 
> Given that the device identity (in the proposal, a MAC address) really has
> to be distinctively inserted into the routing system, use of the MAC address
> in this environment could get really messy.
> 
> We decided it was just easier to assign distinguished names to the devices,
> and put only the distniguished name into the routing system.
> 
> Now, this might be very different in an environment where the UAS-of-choice
> is a hardware phone that is persistent in its association with a singular
> MAC address. I think Henning's approach has the potential to work quite well
> here.
> 

I agree that PCs and hardware phones have very different configuration
requirements. PCs typically already have all kinds of identifiers
available to them (CPU identifiers, GUIDs, OS license numbers, software
license numbers, user names, host names, etc.) and users can easily
create new ones via simple GUIs. On a hardware phone, installing a few
hundred phones and having the installer (or, worse, the phone's user)
key in some unique identifier for each is not very attractive. Having a
bar code on the bottom of the phone, reflecting its MAC address or, if
such a thing were available, some other globally unique identifier, is a
lot easier to manage. Columbia hands out hundreds of phones each year to
incoming freshmen, for example.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 24 11:17:56 2000
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Date: Thu, 24 Feb 2000 18:14:01 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
Organization: Oy L M Ericsson Ab
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To: tfu@cisco.com
Cc: sip <sip@lists.research.bell-labs.com>
Subject: Re: SIP BCP-T question
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Hi,

I am afraid you are talking about another draft.

Your quote is from:
http://www.ietf.org/internet-drafts/draft-camarillo-mmusic-sip-isup-bcp-00.txt

This thread deals with another draft:
http://www.ietf.org/internet-drafts/draft-zimmerer-sip-bcp-t-00.txt

If you have any comments on the first draft just let me know because a
new version is about to be released.

Best regards,

Gonzalo

tfu@cisco.com wrote:
> 
>  From BCP-T:
>     "This document only takes into account the call control
>     functionality of ISUP. Maintenance messages are outside
>     the scope of this document."
> 
> Suppose that messages such as Circuit Reset may also be received
> during a
> call setup
> phase (to abort the setup), then should they also be mapped to such as
> BYE
> or CANCEL
> (by PSTN GW)?
> 
> Thanks,
> -Taichi
> 
> At 04:33 PM 2/23/00 -0600, Dean Willis wrote:
> >It seems that these might really be "no-op" operations rather than
> >"nonexistent" operations.
> >
> >There's a difference.
> >
> >--
> >Dean
> >
> > > -----Original Message-----
> > > From: owner-sip@lists.research.bell-labs.com
> > > [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Carr,
> Steve
> > > Sent: Monday, February 14, 2000 12:25 PM
> > > To: 'Raja Badri'; sip ML
> > > Subject: RE: SIP BCP-T question
> > >
> > >
> > > The ISUP maintenance messages relate to the physical trunk
> circuit, which
> > > when not being used on a call, is still physically present and in
> an IDLE
> > > (or blocked) state. SIP BCT-T sets up calls over an RTP "logical"
> > > connection, which is not a physical circuit. After the end of the
> > > call, the
> > > RTP connection ceases to exist. Therefore is does not make sense
> to block
> > > it, unblock it, reset it etc......
> > >
> > > Steve Carr
> > > Unisphere Solutions Inc.
> > > Boca Raton
> > > scarr@unispheresolutions.com
> > > Office (561) 955-8288
> > > Fax    (561) 955-6477
> > >
> > > www.unispheresolutions.com
> > >
> > > -----Original Message-----
> > > From: Raja Badri [mailto:raja.badri@santera.com]
> > > Sent: Monday, February 14, 2000 1:04 PM
> > > To: sip ML
> > > Subject: SIP BCP-T question
> > >
> > >
> > > Hi,
> > >
> > > I have one question regarding the SIP BCP-T.
> > > INFO message outlines the Mid-Call ISUP messages.
> > > What is the reason behind not including the ISUP maintenance
> > > messages like Block, Unblock and Reset ?
> > >
> > > Thanks,
> > > Raja Badri
> > > Santera Systems Inc.
> > >
> > >
> > >
> >
> >
> 
>     ---------------------------------------------------------------
> From BCP-T:
>    "This document only takes into account the call control
>    functionality of ISUP. Maintenance messages are outside
>    the scope of this document."
> 
> Suppose that messages such as Circuit Reset may also be received
> during a call setup
> phase (to abort the setup), then should they also be mapped to such as
> BYE or CANCEL
> (by PSTN GW)?
> 
> Thanks,
> -Taichi
> 
> At 04:33 PM 2/23/00 -0600, Dean Willis wrote:
> 
> > It seems that these might really be "no-op" operations rather than
> > "nonexistent" operations.
> >
> > There's a difference.
> >
> > --
> > Dean
> >
> > > -----Original Message-----
> > > From: owner-sip@lists.research.bell-labs.com
> > > [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Carr,
> > Steve
> > > Sent: Monday, February 14, 2000 12:25 PM
> > > To: 'Raja Badri'; sip ML
> > > Subject: RE: SIP BCP-T question
> > >
> > >
> > > The ISUP maintenance messages relate to the physical trunk
> > circuit, which
> > > when not being used on a call, is still physically present and in
> > an IDLE
> > > (or blocked) state. SIP BCT-T sets up calls over an RTP "logical"
> > > connection, which is not a physical circuit. After the end of the
> > > call, the
> > > RTP connection ceases to exist. Therefore is does not make sense
> > to block
> > > it, unblock it, reset it etc......
> > >
> > > Steve Carr
> > > Unisphere Solutions Inc.
> > > Boca Raton
> > > scarr@unispheresolutions.com
> > > Office (561) 955-8288
> > > Fax    (561) 955-6477
> > >
> > > www.unispheresolutions.com
> > >
> > > -----Original Message-----
> > > From: Raja Badri [mailto:raja.badri@santera.com]
> > > Sent: Monday, February 14, 2000 1:04 PM
> > > To: sip ML
> > > Subject: SIP BCP-T question
> > >
> > >
> > > Hi,
> > >
> > > I have one question regarding the SIP BCP-T.
> > > INFO message outlines the Mid-Call ISUP messages.
> > > What is the reason behind not including the ISUP maintenance
> > > messages like Block, Unblock and Reset ?
> > >
> > > Thanks,
> > > Raja Badri
> > > Santera Systems Inc.
> > >
> > >
> > >
> >
> >

-- 
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



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From: James Kempf <James.Kempf@Eng.Sun.COM>
Reply-To: James Kempf <James.Kempf@Eng.Sun.COM>
Subject: Re: DHCP option tags for SIP
To: dean.willis@wcom.com, jdrosen@dynamicsoft.com
Cc: schulzrinne@cs.columbia.edu, mat@cisco.com,
        sip@lists.research.bell-labs.com
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>If I understand you right, the phone would have an ID, something like
>sip:phone3736489@wcom.com. When plugged in, it automatically registers
>this address, binding it to its current IP address. In addition, users
>can go to a web page (or use a SIP registration), and basically create
>another binding, mapping their name (sip:dean.willis@wcom.com) to the ID
>of the phone (sip:3736489@wcom.com). Is that right?
>
>Neat idea... solves the problem that a standalone phone is not always
>going to have an easy way for the user to enter their name that it
>should use in a registration. It is basically adding another layer of
>indirection: user->device->address, rather than just user->address.


Cellular phones already have such an address (they are required by
the FCC to do so). It's called the Electronic Serial Number(ESN) and
is used in the authentication algorithms.

		jak




From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 24 11:59:59 2000
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Date: Thu, 24 Feb 2000 08:56:14 -0800
To: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
From: Taichi Fu <tfu@cisco.com>
Subject: SIP ISUP BCP-00 question(was: Re: SIP BCP-T question)
Cc: sip <sip@lists.research.bell-labs.com>
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Gonzalo,

Thanks for your clarification. My question is more on your draft.
I think those maintenance messages need also  to be covered.
Execuse me if you already have them in the coming version.

Best regards,
-Taichi

At 06:14 PM 2/24/00 +0200, Gonzalo Camarillo wrote:
>Hi,
>
>I am afraid you are talking about another draft.
>
>Your quote is from:
>http://www.ietf.org/internet-drafts/draft-camarillo-mmusic-sip-isup-bcp-00.txt
>
>This thread deals with another draft:
>http://www.ietf.org/internet-drafts/draft-zimmerer-sip-bcp-t-00.txt
>
>If you have any comments on the first draft just let me know because a
>new version is about to be released.
>
>Best regards,
>
>Gonzalo
>
>tfu@cisco.com wrote:
> >
> >  From BCP-T:
> >     "This document only takes into account the call control
> >     functionality of ISUP. Maintenance messages are outside
> >     the scope of this document."
> >
> > Suppose that messages such as Circuit Reset may also be received
> > during a
> > call setup
> > phase (to abort the setup), then should they also be mapped to such as
> > BYE
> > or CANCEL
> > (by PSTN GW)?
> >
> > Thanks,
> > -Taichi
> >
> > At 04:33 PM 2/23/00 -0600, Dean Willis wrote:
> > >It seems that these might really be "no-op" operations rather than
> > >"nonexistent" operations.
> > >
> > >There's a difference.
> > >
> > >--
> > >Dean
> > >
> > > > -----Original Message-----
> > > > From: owner-sip@lists.research.bell-labs.com
> > > > [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Carr,
> > Steve
> > > > Sent: Monday, February 14, 2000 12:25 PM
> > > > To: 'Raja Badri'; sip ML
> > > > Subject: RE: SIP BCP-T question
> > > >
> > > >
> > > > The ISUP maintenance messages relate to the physical trunk
> > circuit, which
> > > > when not being used on a call, is still physically present and in
> > an IDLE
> > > > (or blocked) state. SIP BCT-T sets up calls over an RTP "logical"
> > > > connection, which is not a physical circuit. After the end of the
> > > > call, the
> > > > RTP connection ceases to exist. Therefore is does not make sense
> > to block
> > > > it, unblock it, reset it etc......
> > > >
> > > > Steve Carr
> > > > Unisphere Solutions Inc.
> > > > Boca Raton
> > > > scarr@unispheresolutions.com
> > > > Office (561) 955-8288
> > > > Fax    (561) 955-6477
> > > >
> > > > www.unispheresolutions.com
> > > >
> > > > -----Original Message-----
> > > > From: Raja Badri [mailto:raja.badri@santera.com]
> > > > Sent: Monday, February 14, 2000 1:04 PM
> > > > To: sip ML
> > > > Subject: SIP BCP-T question
> > > >
> > > >
> > > > Hi,
> > > >
> > > > I have one question regarding the SIP BCP-T.
> > > > INFO message outlines the Mid-Call ISUP messages.
> > > > What is the reason behind not including the ISUP maintenance
> > > > messages like Block, Unblock and Reset ?
> > > >
> > > > Thanks,
> > > > Raja Badri
> > > > Santera Systems Inc.
> > > >
> > > >
> > > >
> > >
> > >
> >
> >     ---------------------------------------------------------------
> > From BCP-T:
> >    "This document only takes into account the call control
> >    functionality of ISUP. Maintenance messages are outside
> >    the scope of this document."
> >
> > Suppose that messages such as Circuit Reset may also be received
> > during a call setup
> > phase (to abort the setup), then should they also be mapped to such as
> > BYE or CANCEL
> > (by PSTN GW)?
> >
> > Thanks,
> > -Taichi
> >
> > At 04:33 PM 2/23/00 -0600, Dean Willis wrote:
> >
> > > It seems that these might really be "no-op" operations rather than
> > > "nonexistent" operations.
> > >
> > > There's a difference.
> > >
> > > --
> > > Dean
> > >
> > > > -----Original Message-----
> > > > From: owner-sip@lists.research.bell-labs.com
> > > > [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Carr,
> > > Steve
> > > > Sent: Monday, February 14, 2000 12:25 PM
> > > > To: 'Raja Badri'; sip ML
> > > > Subject: RE: SIP BCP-T question
> > > >
> > > >
> > > > The ISUP maintenance messages relate to the physical trunk
> > > circuit, which
> > > > when not being used on a call, is still physically present and in
> > > an IDLE
> > > > (or blocked) state. SIP BCT-T sets up calls over an RTP "logical"
> > > > connection, which is not a physical circuit. After the end of the
> > > > call, the
> > > > RTP connection ceases to exist. Therefore is does not make sense
> > > to block
> > > > it, unblock it, reset it etc......
> > > >
> > > > Steve Carr
> > > > Unisphere Solutions Inc.
> > > > Boca Raton
> > > > scarr@unispheresolutions.com
> > > > Office (561) 955-8288
> > > > Fax    (561) 955-6477
> > > >
> > > > www.unispheresolutions.com
> > > >
> > > > -----Original Message-----
> > > > From: Raja Badri [mailto:raja.badri@santera.com]
> > > > Sent: Monday, February 14, 2000 1:04 PM
> > > > To: sip ML
> > > > Subject: SIP BCP-T question
> > > >
> > > >
> > > > Hi,
> > > >
> > > > I have one question regarding the SIP BCP-T.
> > > > INFO message outlines the Mid-Call ISUP messages.
> > > > What is the reason behind not including the ISUP maintenance
> > > > messages like Block, Unblock and Reset ?
> > > >
> > > > Thanks,
> > > > Raja Badri
> > > > Santera Systems Inc.
> > > >
> > > >
> > > >
> > >
> > >
>
>--
>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
>




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Date: Thu, 24 Feb 2000 10:04:39 -0600
To: Dean Willis <dean.willis@wcom.com>
From: Richard Shockey <rshockey@ix.netcom.com>
Subject: RE: DHCP option tags for SIP
Cc: Michael Thomas <mat@cisco.com>, sip@lists.research.bell-labs.com
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At 09:41 AM 2/24/2000 -0600, Dean Willis wrote:

>We thought about this, but didn't like it much given the way that our
>experimental provisioning systems and UASes work. For example, my PC sip
>client has nine different MAC addresses depending on where I'm sitting. Much
>of the time it has two simultaneously, and sometimes it has three.  Also,
>sometimes it is not the same physical device at all, but I want it to appear
>to be (identity sharing), so there might be multiple sets of nine different
>MAC addresses which are all "the same phone".
>
>Given that the device identity (in the proposal, a MAC address) really has
>to be distinctively inserted into the routing system, use of the MAC address
>in this environment could get really messy.

I agree.



>We decided it was just easier to assign distinguished names to the devices,
>and put only the distniguished name into the routing system.


I certainly think the distinguished name makes more sense in this 
environment. It also may be more useful in mobile environments.

Actually some of this looks like a "cell phone" network at a sufficient 
level of abstraction.




 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
Richard Shockey
Shockey Consulting LLC
8045 Big Bend Blvd. Suite 110
St. Louis, MO 63119
Voice 314.918.9020
eFAX Fax to EMail 815.333.1237 (Preferred for Fax)
INTERNET Mail & IFAX : rshockey@ix.netcom.com
GSTN Fax 314.918.9015
MediaGate iPost VoiceMail and Fax 800.260.4464
<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<



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Jonathan Rosenberg wrote:
> 
> I would further clarify this:
> 
>      A UAC copies the tag from the response into the ACK, but it MUST
> NOT
>      copy the tag into any subsequent requests unless the response was
>      a 200 class response to an INVITE.

I'm not sure it's a problem, but this can give rise to curious
situations when the the non-2xx final response was generated by a UAS. 
In this case the UAS has inserted a To tag and hence believes the call
consists of a single leg, whose UAS identifier (the To of the original
request) has a tag.

The UAC ACKs the response (ACK includes To tag) and resubmits the
request *without* To tag. This basically constitutes initiating a new
transaction in a new leg.

Now I'm not sure if it's well-specified whether the UAS will associate
that new request with the existing leg (which has a To tag). I think
this would (or should) be the correct behaviour, as it really is the
same leg, but I would imagine that different implementations might
differ on this point. The consequence is that the second response (say a
2xx) may be returned with the *original* To tag.

So now we have a UAC which basically knows of two legs: the original
*with* a To tag and a new one without.  It then receives a response
whose To, From pair would seem to indicate that it belongs to the
original leg, but which really belongs to the second - the one that
knows of the transaction...

This all seems a bit fishy, no?

Maybe it should be clarified whether the UAS should respond to the
second request with the old or a new To tag, and whether the UAC is
supposed to think of the two requests as belonging to different legs?

Anders

-- 
Anders Kristensen <ak@hplb.hpl.hp.com>,
http://www-uk.hpl.hp.com/people/ak/
Hewlett-Packard Labs, Bristol, UK



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From: "Lalwaney, Poornima (SD-EX)" <PLalwaney@gi.com>
Subject: RE: Cross-domain QoS setup
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        "Adam B. Roach" <Adam.Roach@Ericsson.com>,
        "'mat@cisco.com'" <mat@cisco.com>
Cc: hgs@cs.columbia.edu, Steven.R.Donovan@wcom.com,
        sip@lists.research.bell-labs.com
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If the SDP in a SIP INVITE can carry "pre-conditions" to call setup, it
seems very reasonable to me that a subsequent provisional response and/or
its acknowledgement  be allowed to carry an indication in the SDP that the
requested "pre-condition" has been satisfied/not satisfied. 

Cross domain QOS  is an example where such functionality MAY be needed if
RSVP is not used for end to end reservation. 
Rather than talk about the best cross-domain QOS mechanism (to which there
is no proven correct answer), lets focus on the SIP signaling issue:  Can
the SDP in a SIP provisional response/PRACK/info message sent prior to the
final 200OK indicate that the precondition specified in an INVITE has been
satisfied? 

If the answer is yes, one of  the ways to indicate this is through
additional attributes to the pre-condition strength field in the sdp
pre-conditions draft. 



	Thanks..
	Poornima



	---------------------------------------
	Poornima Lalwaney
	Motorola
	858-404-3494
	---------------------------------------
> -----Original Message-----
> From:	Jonathan Rosenberg [SMTP:jdrosen@dynamicsoft.com]
> Sent:	Tuesday, February 22, 2000 3:01 PM
> To:	Adam B. Roach
> Cc:	hgs@cs.columbia.edu; Steven.R.Donovan@wcom.com;
> sip@lists.research.bell-labs.com
> Subject:	Re: Cross-domain QoS setup
> 
> 
> 
> "Adam B. Roach" wrote:
> > The current draft makes an assumption that end-to-end network 
> > resource reservation is controlled by a single entity. For calls
> > being placed across adminstrative domains, however, this will
> > not be the case. Under such circumstances, the reservation of
> > resources needs a greater degree of coordination than is 
> > currently provided in the draft.
> 
> Not really. Resource reservation is under the "control" of all the
> network elements along the path that perform some kind of admission
> control and resource allocation. These elements will often belong to
> different administrative domains. The draft makes no assumptions about
> the number of providers along the reservation path, or anything like
> that. It only ties resource reservation and signaling together at the
> endpoints, dicatating a particular temporal sequence.
> 
> > 
> > In particular, the calling user agent needs the ability to
> > communicate to the called user agent that it has reserved
> > the necessary resources on its end, and that alerting may
> > begin as soon as the called user agent has done the same.
> 
> This is something different from what you are discussing above. It is,
> in fact, covered in the draft. If A calls B, the RSVP RESV message from
> A to B confirms resources are reserved from A to B, and the RESVCONF
> from A to B confirms resource reservation from B to A. This allows B to
> know whether resources are reserved in both directions. A mirror case
> exists for A. 
> 
> The difficulty arises only with the segmented DQoS model in DCS, which
> does not provide a way to confirm resource reservation (I think),
> whereas RSVP does.
> 
> So, the question is this. We know we need a way for both parties to know
> when resources are reserved in both directions. Is this a function that
> belongs somewhere in SIP (through INFO, as you propose, or some other
> means), or in a reservation protocol. The mechanism exists already now
> in RSVP, but not in DQoS (DCS folks, please correct me if I am
> mistaken). So, do we add it to SIP, in which case its not needed for
> RSVP, or should it be added to DQoS, in which case the SIP operation is
> independent of the underlying reservation mechanism?
> 
> I have argued that confirmation of resource reservation is rightly a job
> for a resource reservation protocol. One of the things I like about SIP
> is its separation from session level control and network layer issues,
> such as resource reservation.
> 
> -Jonathan R.
> 
> -- 
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120 
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com

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charset=3DUS-ASCII">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2650.12">
<TITLE>RE: Cross-domain QoS setup</TITLE>
</HEAD>
<BODY>
<BR>
<BR>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">If the SDP in a SIP =
INVITE can carry &quot;pre-conditions&quot; to call setup, it seems =
very reasonable to me that a subsequent provisional response and/or its =
acknowledgement&nbsp; be allowed to carry an indication in the SDP that =
the requested &quot;pre-condition&quot; has been satisfied/not =
satisfied. </FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Cross domain =
QOS&nbsp; is an example where such functionality MAY be needed if RSVP =
is not used for end to end reservation. </FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Rather than talk =
about the best cross-domain QOS mechanism (to which there is no proven =
correct answer), lets focus on the SIP signaling issue:&nbsp; Can the =
SDP in a SIP provisional response/PRACK/info message sent prior to the =
final 200OK indicate that the precondition specified in an INVITE has =
been satisfied? </FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">If the answer is =
yes, one of&nbsp; the ways to indicate this is through additional =
attributes to the pre-condition strength field in the sdp =
pre-conditions draft. </FONT></P>
<BR>
<BR>
<UL>
<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Thanks..</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Poornima</FONT>
</P>
<BR>
<BR>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">---------------------------------------</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Poornima =
Lalwaney</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Motorola</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">858-404-3494</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">---------------------------------------</FONT>
<BR><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">Jonathan Rosenberg =
[SMTP:jdrosen@dynamicsoft.com]</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Tuesday, February 22, 2000 3:01 PM</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">Adam B. Roach</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Cc:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">hgs@cs.columbia.edu; Steven.R.Donovan@wcom.com; =
sip@lists.research.bell-labs.com</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">Re: Cross-domain QoS setup</FONT>
</P>
<BR>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Arial">&quot;Adam B. Roach&quot; =
wrote:</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; The current draft makes an =
assumption that end-to-end network </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; resource reservation is =
controlled by a single entity. For calls</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; being placed across =
adminstrative domains, however, this will</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; not be the case. Under such =
circumstances, the reservation of</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; resources needs a greater degree =
of coordination than is </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; currently provided in the =
draft.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Not really. Resource reservation is =
under the &quot;control&quot; of all the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">network elements along the path that =
perform some kind of admission</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">control and resource allocation. =
These elements will often belong to</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">different administrative domains. The =
draft makes no assumptions about</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">the number of providers along the =
reservation path, or anything like</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">that. It only ties resource =
reservation and signaling together at the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">endpoints, dicatating a particular =
temporal sequence.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; In particular, the calling user =
agent needs the ability to</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; communicate to the called user =
agent that it has reserved</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; the necessary resources on its =
end, and that alerting may</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; begin as soon as the called user =
agent has done the same.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">This is something different from what =
you are discussing above. It is,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">in fact, covered in the draft. If A =
calls B, the RSVP RESV message from</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">A to B confirms resources are =
reserved from A to B, and the RESVCONF</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">from A to B confirms resource =
reservation from B to A. This allows B to</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">know whether resources are reserved =
in both directions. A mirror case</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">exists for A. </FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">The difficulty arises only with the =
segmented DQoS model in DCS, which</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">does not provide a way to confirm =
resource reservation (I think),</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">whereas RSVP does.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">So, the question is this. We know we =
need a way for both parties to know</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">when resources are reserved in both =
directions. Is this a function that</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">belongs somewhere in SIP (through =
INFO, as you propose, or some other</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">means), or in a reservation protocol. =
The mechanism exists already now</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">in RSVP, but not in DQoS (DCS folks, =
please correct me if I am</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">mistaken). So, do we add it to SIP, =
in which case its not needed for</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">RSVP, or should it be added to DQoS, =
in which case the SIP operation is</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">independent of the underlying =
reservation mechanism?</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I have argued that confirmation of =
resource reservation is rightly a job</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">for a resource reservation protocol. =
One of the things I like about SIP</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">is its separation from session level =
control and network layer issues,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">such as resource reservation.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">-Jonathan R.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">-- </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Jonathan D. =
Rosenberg&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
200 Executive Drive</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Chief =
Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Suite 120 </FONT>
<BR><FONT SIZE=3D2 =
FACE=3D"Arial">dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; West Orange, NJ 07052</FONT>
<BR><FONT SIZE=3D2 =
FACE=3D"Arial">jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp; FAX:&nbsp;&nbsp; (732) 741-4778</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://www.cs.columbia.edu/~jdrosen" =
TARGET=3D"_blank">http://www.cs.columbia.edu/~jdrosen</A></FONT></U><FON=
T SIZE=3D2 =
FACE=3D"Arial">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: =
(732) 741-7244</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://www.dynamicsoft.com" =
TARGET=3D"_blank">http://www.dynamicsoft.com</A></FONT></U>
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Date: Thu, 24 Feb 2000 14:20:21 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: SIP BCP-T Current Drafts (was RE: SIP BCP-T question)
In-reply-to: <38B558C9.373770DE@lmf.ericsson.se>
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I've heard many implementors continuing to reference the old BCP draft and
claiming to "Support SIP-T".

I'd like to make two points:

1) As Gonzalo points out, the draft draft-zimmerer-mmusic-sip-bcp-t-00.txt
is quite obsolete. I believe draft-camarillo-mmusic-sip-isup-bcp-00.txt is
too, as is draft-ietf-mmusic-sip-isup-mime-00.txt

The current drafts are draft-zimmerer-sip-bcp-t-00.txt and
draft-ietf-sip-isup-mime-00.txt. I think these combine most of the previous
work in this area.

2) This is a "work in progress", NOT a standard. It's not even a proposed
standard. I bet its going to change a little more before it makes it to
proposed standard (don't they all?). Implementors should plan on making
changes as the work matures.

--
Dean Willis
SIP WG co-chair


> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Gonzalo
> Camarillo
> Sent: Thursday, February 24, 2000 10:14 AM
> To: tfu@cisco.com
> Cc: sip
> Subject: Re: SIP BCP-T question
>
>
> Hi,
>
> I am afraid you are talking about another draft.
>
> Your quote is from:
> http://www.ietf.org/internet-drafts/draft-camarillo-mmusic-sip-isu
> p-bcp-00.txt
>
> This thread deals with another draft:
> http://www.ietf.org/internet-drafts/draft-zimmerer-sip-bcp-t-00.txt
>
> If you have any comments on the first draft just let me know because a
> new version is about to be released.
>
> Best regards,
>
> Gonzalo
>
> tfu@cisco.com wrote:
> >
> >  From BCP-T:
> >     "This document only takes into account the call control
> >     functionality of ISUP. Maintenance messages are outside
> >     the scope of this document."
> >
> > Suppose that messages such as Circuit Reset may also be received
> > during a
> > call setup
> > phase (to abort the setup), then should they also be mapped to such as
> > BYE
> > or CANCEL
> > (by PSTN GW)?
> >
> > Thanks,
> > -Taichi
> >
> > At 04:33 PM 2/23/00 -0600, Dean Willis wrote:
> > >It seems that these might really be "no-op" operations rather than
> > >"nonexistent" operations.
> > >
> > >There's a difference.
> > >
> > >--
> > >Dean
> > >
> > > > -----Original Message-----
> > > > From: owner-sip@lists.research.bell-labs.com
> > > > [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Carr,
> > Steve
> > > > Sent: Monday, February 14, 2000 12:25 PM
> > > > To: 'Raja Badri'; sip ML
> > > > Subject: RE: SIP BCP-T question
> > > >
> > > >
> > > > The ISUP maintenance messages relate to the physical trunk
> > circuit, which
> > > > when not being used on a call, is still physically present and in
> > an IDLE
> > > > (or blocked) state. SIP BCT-T sets up calls over an RTP "logical"
> > > > connection, which is not a physical circuit. After the end of the
> > > > call, the
> > > > RTP connection ceases to exist. Therefore is does not make sense
> > to block
> > > > it, unblock it, reset it etc......
> > > >
> > > > Steve Carr
> > > > Unisphere Solutions Inc.
> > > > Boca Raton
> > > > scarr@unispheresolutions.com
> > > > Office (561) 955-8288
> > > > Fax    (561) 955-6477
> > > >
> > > > www.unispheresolutions.com
> > > >
> > > > -----Original Message-----
> > > > From: Raja Badri [mailto:raja.badri@santera.com]
> > > > Sent: Monday, February 14, 2000 1:04 PM
> > > > To: sip ML
> > > > Subject: SIP BCP-T question
> > > >
> > > >
> > > > Hi,
> > > >
> > > > I have one question regarding the SIP BCP-T.
> > > > INFO message outlines the Mid-Call ISUP messages.
> > > > What is the reason behind not including the ISUP maintenance
> > > > messages like Block, Unblock and Reset ?
> > > >
> > > > Thanks,
> > > > Raja Badri
> > > > Santera Systems Inc.
> > > >
> > > >
> > > >
> > >
> > >
> >
> >     ---------------------------------------------------------------
> > From BCP-T:
> >    "This document only takes into account the call control
> >    functionality of ISUP. Maintenance messages are outside
> >    the scope of this document."
> >
> > Suppose that messages such as Circuit Reset may also be received
> > during a call setup
> > phase (to abort the setup), then should they also be mapped to such as
> > BYE or CANCEL
> > (by PSTN GW)?
> >
> > Thanks,
> > -Taichi
> >
> > At 04:33 PM 2/23/00 -0600, Dean Willis wrote:
> >
> > > It seems that these might really be "no-op" operations rather than
> > > "nonexistent" operations.
> > >
> > > There's a difference.
> > >
> > > --
> > > Dean
> > >
> > > > -----Original Message-----
> > > > From: owner-sip@lists.research.bell-labs.com
> > > > [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Carr,
> > > Steve
> > > > Sent: Monday, February 14, 2000 12:25 PM
> > > > To: 'Raja Badri'; sip ML
> > > > Subject: RE: SIP BCP-T question
> > > >
> > > >
> > > > The ISUP maintenance messages relate to the physical trunk
> > > circuit, which
> > > > when not being used on a call, is still physically present and in
> > > an IDLE
> > > > (or blocked) state. SIP BCT-T sets up calls over an RTP "logical"
> > > > connection, which is not a physical circuit. After the end of the
> > > > call, the
> > > > RTP connection ceases to exist. Therefore is does not make sense
> > > to block
> > > > it, unblock it, reset it etc......
> > > >
> > > > Steve Carr
> > > > Unisphere Solutions Inc.
> > > > Boca Raton
> > > > scarr@unispheresolutions.com
> > > > Office (561) 955-8288
> > > > Fax    (561) 955-6477
> > > >
> > > > www.unispheresolutions.com
> > > >
> > > > -----Original Message-----
> > > > From: Raja Badri [mailto:raja.badri@santera.com]
> > > > Sent: Monday, February 14, 2000 1:04 PM
> > > > To: sip ML
> > > > Subject: SIP BCP-T question
> > > >
> > > >
> > > > Hi,
> > > >
> > > > I have one question regarding the SIP BCP-T.
> > > > INFO message outlines the Mid-Call ISUP messages.
> > > > What is the reason behind not including the ISUP maintenance
> > > > messages like Block, Unblock and Reset ?
> > > >
> > > > Thanks,
> > > > Raja Badri
> > > > Santera Systems Inc.
> > > >
> > > >
> > > >
> > >
> > >
>
> --
> 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
>




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From: Aparna.Vemuri@Level3.com
To: dean.willis@wcom.com, sip@lists.research.bell-labs.com
Subject: RE: SIP BCP-T Current Drafts (was RE: SIP BCP-T question)
Date: Thu, 24 Feb 2000 13:57:19 -0700
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The confusion probably stems from the fact there are around 5 drafts
pertaining to the ISUP MIME and SIP-T efforts at the IETF web-site. 
It would be great if the obsolete drafts could be taken out, although they
haven't expired yet.

Thanks,
Aparna

Aparna V.
Level (3) Communications.

-----Original Message-----
From: Dean Willis [mailto:dean.willis@wcom.com]
Sent: Thursday, February 24, 2000 1:20 PM
To: sip
Subject: SIP BCP-T Current Drafts (was RE: SIP BCP-T question)



I've heard many implementors continuing to reference the old BCP draft and
claiming to "Support SIP-T".

I'd like to make two points:

1) As Gonzalo points out, the draft draft-zimmerer-mmusic-sip-bcp-t-00.txt
is quite obsolete. I believe draft-camarillo-mmusic-sip-isup-bcp-00.txt is
too, as is draft-ietf-mmusic-sip-isup-mime-00.txt

The current drafts are draft-zimmerer-sip-bcp-t-00.txt and
draft-ietf-sip-isup-mime-00.txt. I think these combine most of the previous
work in this area.

2) This is a "work in progress", NOT a standard. It's not even a proposed
standard. I bet its going to change a little more before it makes it to
proposed standard (don't they all?). Implementors should plan on making
changes as the work matures.

--
Dean Willis
SIP WG co-chair


> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Gonzalo
> Camarillo
> Sent: Thursday, February 24, 2000 10:14 AM
> To: tfu@cisco.com
> Cc: sip
> Subject: Re: SIP BCP-T question
>
>
> Hi,
>
> I am afraid you are talking about another draft.
>
> Your quote is from:
> http://www.ietf.org/internet-drafts/draft-camarillo-mmusic-sip-isu
> p-bcp-00.txt
>
> This thread deals with another draft:
> http://www.ietf.org/internet-drafts/draft-zimmerer-sip-bcp-t-00.txt
>
> If you have any comments on the first draft just let me know because a
> new version is about to be released.
>
> Best regards,
>
> Gonzalo
>
> tfu@cisco.com wrote:
> >
> >  From BCP-T:
> >     "This document only takes into account the call control
> >     functionality of ISUP. Maintenance messages are outside
> >     the scope of this document."
> >
> > Suppose that messages such as Circuit Reset may also be received
> > during a
> > call setup
> > phase (to abort the setup), then should they also be mapped to such as
> > BYE
> > or CANCEL
> > (by PSTN GW)?
> >
> > Thanks,
> > -Taichi
> >
> > At 04:33 PM 2/23/00 -0600, Dean Willis wrote:
> > >It seems that these might really be "no-op" operations rather than
> > >"nonexistent" operations.
> > >
> > >There's a difference.
> > >
> > >--
> > >Dean
> > >
> > > > -----Original Message-----
> > > > From: owner-sip@lists.research.bell-labs.com
> > > > [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Carr,
> > Steve
> > > > Sent: Monday, February 14, 2000 12:25 PM
> > > > To: 'Raja Badri'; sip ML
> > > > Subject: RE: SIP BCP-T question
> > > >
> > > >
> > > > The ISUP maintenance messages relate to the physical trunk
> > circuit, which
> > > > when not being used on a call, is still physically present and in
> > an IDLE
> > > > (or blocked) state. SIP BCT-T sets up calls over an RTP "logical"
> > > > connection, which is not a physical circuit. After the end of the
> > > > call, the
> > > > RTP connection ceases to exist. Therefore is does not make sense
> > to block
> > > > it, unblock it, reset it etc......
> > > >
> > > > Steve Carr
> > > > Unisphere Solutions Inc.
> > > > Boca Raton
> > > > scarr@unispheresolutions.com
> > > > Office (561) 955-8288
> > > > Fax    (561) 955-6477
> > > >
> > > > www.unispheresolutions.com
> > > >
> > > > -----Original Message-----
> > > > From: Raja Badri [mailto:raja.badri@santera.com]
> > > > Sent: Monday, February 14, 2000 1:04 PM
> > > > To: sip ML
> > > > Subject: SIP BCP-T question
> > > >
> > > >
> > > > Hi,
> > > >
> > > > I have one question regarding the SIP BCP-T.
> > > > INFO message outlines the Mid-Call ISUP messages.
> > > > What is the reason behind not including the ISUP maintenance
> > > > messages like Block, Unblock and Reset ?
> > > >
> > > > Thanks,
> > > > Raja Badri
> > > > Santera Systems Inc.
> > > >
> > > >
> > > >
> > >
> > >
> >
> >     ---------------------------------------------------------------
> > From BCP-T:
> >    "This document only takes into account the call control
> >    functionality of ISUP. Maintenance messages are outside
> >    the scope of this document."
> >
> > Suppose that messages such as Circuit Reset may also be received
> > during a call setup
> > phase (to abort the setup), then should they also be mapped to such as
> > BYE or CANCEL
> > (by PSTN GW)?
> >
> > Thanks,
> > -Taichi
> >
> > At 04:33 PM 2/23/00 -0600, Dean Willis wrote:
> >
> > > It seems that these might really be "no-op" operations rather than
> > > "nonexistent" operations.
> > >
> > > There's a difference.
> > >
> > > --
> > > Dean
> > >
> > > > -----Original Message-----
> > > > From: owner-sip@lists.research.bell-labs.com
> > > > [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Carr,
> > > Steve
> > > > Sent: Monday, February 14, 2000 12:25 PM
> > > > To: 'Raja Badri'; sip ML
> > > > Subject: RE: SIP BCP-T question
> > > >
> > > >
> > > > The ISUP maintenance messages relate to the physical trunk
> > > circuit, which
> > > > when not being used on a call, is still physically present and in
> > > an IDLE
> > > > (or blocked) state. SIP BCT-T sets up calls over an RTP "logical"
> > > > connection, which is not a physical circuit. After the end of the
> > > > call, the
> > > > RTP connection ceases to exist. Therefore is does not make sense
> > > to block
> > > > it, unblock it, reset it etc......
> > > >
> > > > Steve Carr
> > > > Unisphere Solutions Inc.
> > > > Boca Raton
> > > > scarr@unispheresolutions.com
> > > > Office (561) 955-8288
> > > > Fax    (561) 955-6477
> > > >
> > > > www.unispheresolutions.com
> > > >
> > > > -----Original Message-----
> > > > From: Raja Badri [mailto:raja.badri@santera.com]
> > > > Sent: Monday, February 14, 2000 1:04 PM
> > > > To: sip ML
> > > > Subject: SIP BCP-T question
> > > >
> > > >
> > > > Hi,
> > > >
> > > > I have one question regarding the SIP BCP-T.
> > > > INFO message outlines the Mid-Call ISUP messages.
> > > > What is the reason behind not including the ISUP maintenance
> > > > messages like Block, Unblock and Reset ?
> > > >
> > > > Thanks,
> > > > Raja Badri
> > > > Santera Systems Inc.
> > > >
> > > >
> > > >
> > >
> > >
>
> --
> 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
>




From owner-sip-outgoing@lists.research.bell-labs.com  Thu Feb 24 23:16:16 2000
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From: "David Israel" <david@ipunity.com>
To: <sip@lists.research.bell-labs.com>
Subject: On behalf of Jimmy Zhang
Date: Thu, 24 Feb 2000 20:13:10 -0800
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Hello,

   My job is to implement a subset  of SIP. I would like to obtain, as many
as possible, real SIP messages that  covers most(if not all) tags, keywords,
and etc. Could someone help me on this?

  Also, I would like to report the bugs I have encountered while I was
reading RFC2543. First the "toplabel" on page 21 is not correct, to make it
work, prepend * to alphanum. Another typo takes place on page 40, the first
vertical bar after Global-Failure should have been "=" instead.

thanks

-- Jimmy Zhang (jz@ipunity.com)

   Software Engineer

   IP Unity




From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 25 02:55:55 2000
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From: "Stanislav S. Timinsky" <timinsky@mera.ru>
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Hello David,

Friday, February 25, 2000, 7:13:10 AM, you wrote:

David Israel> Hello,

David Israel>    My job is to implement a subset  of SIP. I would like to obtain, as many
David Israel> as possible, real SIP messages that  covers most(if not all) tags, keywords,
David Israel> and etc. Could someone help me on this?

David Israel>   Also, I would like to report the bugs I have encountered while I was
David Israel> reading RFC2543. First the "toplabel" on page 21 is not correct, to make it
David Israel> work, prepend * to alphanum. Another typo takes place on page 40, the first
David Israel> vertical bar after Global-Failure should have been "=" instead.

David Israel> thanks

David Israel> -- Jimmy Zhang (jz@ipunity.com)

David Israel>    Software Engineer

David Israel>    IP Unity

1.

Some test messages are at:
http://www.cs.columbia.edu/~hgs/sip/bakeoff2/testmsg.html
and
http://www.cs.columbia.edu/~hgs/sip/drafts/draft-sparks-sip-service-examples-00.txt

2.

As to inexactnesses in RFC-2543, they were also noted by me at realization of the SIP stack.
I want to notice, that RFC-2327 (SDP) also has some bugs in grammar description.

-- 
Best regards,
 Stanislav S. Timinsky                      mailto:timinsky@mera.ru
 Mera Labs.





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Date: Fri, 25 Feb 2000 15:17:30 +0200
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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Hi,

Yes, there is a section about it. The draft will be available pretty
soon.

Best regards,

Gonzalo

tfu@cisco.com wrote:
> 
> Gonzalo,
> 
> Thanks for your clarification. My question is more on your draft.
> I think those maintenance messages need also  to be covered.
> Execuse me if you already have them in the coming version.
> 
> Best regards,
> -Taichi
> 
> At 06:14 PM 2/24/00 +0200, Gonzalo Camarillo wrote:
> >Hi,
> >
> >I am afraid you are talking about another draft.
> >
> >Your quote is from:
> >http://www.ietf.org/internet-drafts/draft-camarillo-mmusic-sip-isup-bcp-00.txt
> >
> >This thread deals with another draft:
> >http://www.ietf.org/internet-drafts/draft-zimmerer-sip-bcp-t-00.txt
> >
> >If you have any comments on the first draft just let me know because a
> >new version is about to be released.
> >
> >Best regards,
> >
> >Gonzalo
> >
> >tfu@cisco.com wrote:
> > >
> > >  From BCP-T:
> > >     "This document only takes into account the call control
> > >     functionality of ISUP. Maintenance messages are outside
> > >     the scope of this document."
> > >
> > > Suppose that messages such as Circuit Reset may also be received
> > > during a
> > > call setup
> > > phase (to abort the setup), then should they also be mapped to such as
> > > BYE
> > > or CANCEL
> > > (by PSTN GW)?
> > >
> > > Thanks,
> > > -Taichi
> > >
> > > At 04:33 PM 2/23/00 -0600, Dean Willis wrote:
> > > >It seems that these might really be "no-op" operations rather than
> > > >"nonexistent" operations.
> > > >
> > > >There's a difference.
> > > >
> > > >--
> > > >Dean
> > > >
> > > > > -----Original Message-----
> > > > > From: owner-sip@lists.research.bell-labs.com
> > > > > [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Carr,
> > > Steve
> > > > > Sent: Monday, February 14, 2000 12:25 PM
> > > > > To: 'Raja Badri'; sip ML
> > > > > Subject: RE: SIP BCP-T question
> > > > >
> > > > >
> > > > > The ISUP maintenance messages relate to the physical trunk
> > > circuit, which
> > > > > when not being used on a call, is still physically present and in
> > > an IDLE
> > > > > (or blocked) state. SIP BCT-T sets up calls over an RTP "logical"
> > > > > connection, which is not a physical circuit. After the end of the
> > > > > call, the
> > > > > RTP connection ceases to exist. Therefore is does not make sense
> > > to block
> > > > > it, unblock it, reset it etc......
> > > > >
> > > > > Steve Carr
> > > > > Unisphere Solutions Inc.
> > > > > Boca Raton
> > > > > scarr@unispheresolutions.com
> > > > > Office (561) 955-8288
> > > > > Fax    (561) 955-6477
> > > > >
> > > > > www.unispheresolutions.com
> > > > >
> > > > > -----Original Message-----
> > > > > From: Raja Badri [mailto:raja.badri@santera.com]
> > > > > Sent: Monday, February 14, 2000 1:04 PM
> > > > > To: sip ML
> > > > > Subject: SIP BCP-T question
> > > > >
> > > > >
> > > > > Hi,
> > > > >
> > > > > I have one question regarding the SIP BCP-T.
> > > > > INFO message outlines the Mid-Call ISUP messages.
> > > > > What is the reason behind not including the ISUP maintenance
> > > > > messages like Block, Unblock and Reset ?
> > > > >
> > > > > Thanks,
> > > > > Raja Badri
> > > > > Santera Systems Inc.
> > > > >
> > > > >
> > > > >
> > > >
> > > >
> > >
> > >     ---------------------------------------------------------------
> > > From BCP-T:
> > >    "This document only takes into account the call control
> > >    functionality of ISUP. Maintenance messages are outside
> > >    the scope of this document."
> > >
> > > Suppose that messages such as Circuit Reset may also be received
> > > during a call setup
> > > phase (to abort the setup), then should they also be mapped to such as
> > > BYE or CANCEL
> > > (by PSTN GW)?
> > >
> > > Thanks,
> > > -Taichi
> > >
> > > At 04:33 PM 2/23/00 -0600, Dean Willis wrote:
> > >
> > > > It seems that these might really be "no-op" operations rather than
> > > > "nonexistent" operations.
> > > >
> > > > There's a difference.
> > > >
> > > > --
> > > > Dean
> > > >
> > > > > -----Original Message-----
> > > > > From: owner-sip@lists.research.bell-labs.com
> > > > > [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Carr,
> > > > Steve
> > > > > Sent: Monday, February 14, 2000 12:25 PM
> > > > > To: 'Raja Badri'; sip ML
> > > > > Subject: RE: SIP BCP-T question
> > > > >
> > > > >
> > > > > The ISUP maintenance messages relate to the physical trunk
> > > > circuit, which
> > > > > when not being used on a call, is still physically present and in
> > > > an IDLE
> > > > > (or blocked) state. SIP BCT-T sets up calls over an RTP "logical"
> > > > > connection, which is not a physical circuit. After the end of the
> > > > > call, the
> > > > > RTP connection ceases to exist. Therefore is does not make sense
> > > > to block
> > > > > it, unblock it, reset it etc......
> > > > >
> > > > > Steve Carr
> > > > > Unisphere Solutions Inc.
> > > > > Boca Raton
> > > > > scarr@unispheresolutions.com
> > > > > Office (561) 955-8288
> > > > > Fax    (561) 955-6477
> > > > >
> > > > > www.unispheresolutions.com
> > > > >
> > > > > -----Original Message-----
> > > > > From: Raja Badri [mailto:raja.badri@santera.com]
> > > > > Sent: Monday, February 14, 2000 1:04 PM
> > > > > To: sip ML
> > > > > Subject: SIP BCP-T question
> > > > >
> > > > >
> > > > > Hi,
> > > > >
> > > > > I have one question regarding the SIP BCP-T.
> > > > > INFO message outlines the Mid-Call ISUP messages.
> > > > > What is the reason behind not including the ISUP maintenance
> > > > > messages like Block, Unblock and Reset ?
> > > > >
> > > > > Thanks,
> > > > > Raja Badri
> > > > > Santera Systems Inc.
> > > > >
> > > > >
> > > > >
> > > >
> > > >
> >
> >--
> >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
> >

-- 
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



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 25 09:27:59 2000
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David Israel wrote:
> 
>   Also, I would like to report the bugs I have encountered while I was
> reading RFC2543. First the "toplabel" on page 21 is not correct, to make it
> work, prepend * to alphanum. 

Why? This grammar seems to work. The * appears outside the parens, so
there can be an arbitrary number of alphanums in the middle. Putting a *
in front of either of the alphanums is redundant. I'll also note that it
is identical to the grammar of rfc2396:

> toplabel      = alpha | alpha *( alphanum | "-" ) alphanum




Another typo takes place on page 40, the first
> vertical bar after Global-Failure should have been "=" instead.

Thanks, we'll fix that.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 25 09:36:31 2000
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"Stanislav S. Timinsky" wrote:
> 
> 
> As to inexactnesses in RFC-2543, they were also noted by me at realization of the SIP stack.
> I want to notice, that RFC-2327 (SDP) also has some bugs in grammar description.

Folks, if you find bugs or problems, please be sure to report them to
the list or to the authors privately. We will incorporate them into the
revised spec.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 25 10:42:03 2000
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Date: Fri, 25 Feb 2000 10:38:21 -0500
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Hi guys,

Assuming a goal of achieving "assured QoS" using generic SIP:
(In assured QoS,  the phone doesn't ring until resource allocation has
been confirmed
vs. enabled QoS, when the phone rings and can be answered before QoS is
established.
In fact, if resources are not available, the enabled QoS conversation
would continue
(without QoS)).

I understand that assured QoS can be achieved using the provisioned
(push) model of COPS.
(REQ, ...  DEC, RPT, ... DEC, RTP, ... DEC, RPT ...)
Some examples of proposed provisioned models (push model) which (can)
achieve assured QoS are:
a) DQoS (using Gate objects)
b) COPS-PR (using ASN.1 encoded objects),
... which ones am I leaving out?

However, I have not seen assured QoS achieved using the
outsourcing model (pull model) of COPS.
(REQ, DEC, RPT, .... REQ, DEC, RPT,  ... REQ, DEC, RPT, ...)

Is it even possible to achieve assured QoS using the outsourcing model
(pull model) of COPS?

If so, could someone please sketch a brief call flow for me of
how to achieve assured QoS using the outsourcing model of COPS?

Thanks for your help!

Bryan

Bryan J. Byerly
byerly@cisco.com




From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 25 12:58:00 2000
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I need a clarification on the handling of Contact and Record-Route
headers in SIP UAs using TCP as the transport. 

Is it the same as UDP ?

The reopening of new TCP connections to handle Contacts and RRs
has overheads and complicates the implementation of the UA.

Thanks

-- Sudipto



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 25 14:15:53 2000
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Cc: "'sip@lists.research.bell-labs.com'" <sip@lists.research.bell-labs.com>
Subject: Re: Content-Encoding vs. Content-Transfer-Encoding
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The Content-Transfer-Encoding header was primarily meant to allow
message bodies to be transformed into formats that could be transferred
on channels that were not 8 bit clean. HTTP, which makes use of many of
the MIME headers, is 8 bit clean, and thus did not need
Content-Transfer-Encoding. SIP followed suit, and so does not use it
either. Content-Encoding is used for things like compression, which is
different (in fact, the opposite, since conversion of a binary file to a
7 bit transfer channel INCREASES the size of the body). 

-Jonathan R.

> Joshua Moloney wrote:
> 
> Hi all,
> 
> I have a question about the "Content-Encoding" entity header.
> Why is the MIME "Content-Transfer-Encoding" header defined in RFC 2045
> not used instead?
> Am I correct in assuming that they do the same job?
> 
> For example, what headers should I include on a message with an
> "application/ISUP" message body? :-
> 
> Content-Type: application/ISUP
> Content-Transfer-Encoding: binary
> .....
> 
> OR
> 
> Content-Type: application/ISUP
> Content-Encoding: binary
> .....
> 
> Thanks in advance,
> 
> Josh Moloney
> 
> Nortel Networks
> jmoloney@nortelnetworks.com

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 25 14:39:53 2000
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        "'SIP List' (E-mail)" <sip@lists.research.bell-labs.com>,
        "Henning Schulzrinne (E-mail)" <schulzrinne@cs.columbia.edu>
Subject: Re: UAC use of tags received in responses
References: <003201bf73dd$40e24300$8a9223a6@mcit.com> <38A33790.5A052AC6@dynamicsoft.com> <38B574B5.50448657@hplb.hpl.hp.com>
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Anders Kristensen wrote:
> 
> I'm not sure it's a problem, but this can give rise to curious
> situations when the the non-2xx final response was generated by a UAS.
> In this case the UAS has inserted a To tag and hence believes the call
> consists of a single leg, whose UAS identifier (the To of the original
> request) has a tag.
> 
> The UAC ACKs the response (ACK includes To tag) and resubmits the
> request *without* To tag. This basically constitutes initiating a new
> transaction in a new leg.
> 
> Now I'm not sure if it's well-specified whether the UAS will associate
> that new request with the existing leg (which has a To tag). I think
> this would (or should) be the correct behaviour, as it really is the
> same leg, but I would imagine that different implementations might
> differ on this point. The consequence is that the second response (say a
> 2xx) may be returned with the *original* To tag.

Hmm. I don't think this would happen. If the UAS receives a request and
sends a non-200 response with a tag, the call leg is over. It doesn't
know whether it will ever receieve another request, so I imagine it will
flush the state, and thats it. When the new request arrives, it creates
a new call leg. The tag in the response will probably be different than
the previous, or it might be the same. 

On the UAC, this basically means it really can't associate a tag with
the call leg until the 200 OK. Before that, any tags it sees in
provisionals or non-200s might not be the final tag. 


> Maybe it should be clarified whether the UAS should respond to the
> second request with the old or a new To tag, and whether the UAC is
> supposed to think of the two requests as belonging to different legs?

I don't think we can impose requirements on maintenance of state at a UA
after rejecting a new INVITE. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 25 16:18:01 2000
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To: Sudipto Mukherjee <sudiptom@cisco.com>
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Sudipto Mukherjee wrote:
> 
> I need a clarification on the handling of Contact and Record-Route
> headers in SIP UAs using TCP as the transport.
> 
> Is it the same as UDP ?

Yes. If the client or proxy needs to send a request or response
somewhere, and a TCP connection isn't open, and that request must be
sent TCP, a connection is opened. 

> 
> The reopening of new TCP connections to handle Contacts and RRs
> has overheads and complicates the implementation of the UA.

The opening is only if a connection to that destination doesn't already
exist. SIP allows TCP connections to be reused for other requests. In
many cases, the connection will already be there.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 25 19:12:04 2000
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To: Bryan Byerly <byerly@cisco.com>, sip@lists.research.bell-labs.com,
        rap@iphighway.com
From: "Lyndon Ong" <long@nortelnetworks.com>
Subject: Re: Question on achieveing Assured QoS using SIP/COPS
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Hi,

Isn't providing your "assured QoS" a matter of having the hooks in call
setup signaling (i.e., SIP and SDP) to indicate that resource allocation is
required prior to ringing, not strictly a COPS issue?  Jonathan, et al's
draft-ietf-mmusic-sdp-qos-00.txt proposes a way to indicate this in SDP and
some SIP extension is also probably involved.

COPS-PR vs. COPS-RSVP, on the other hand, depends on how you are doing
resource allocation decisions:
-- COPS-PR could be used where you are pushing resource allocation
decisions from a central point to edge devices, while
-- COPS-RSVP or outsource would be used where you rely on RSVP signaling
end-to-end to initiate resource allocation, and edge devices detecting the
RSVP PATH/RESV use COPS-RSVP to outsource the decision to a PDP.

By the way, "assured" might be confused with "assured forwarding" in
Diffserv, a completely different concept...

L. Ong

At 10:38 AM 2/25/00 -0500, Bryan Byerly wrote:
>Hi guys,
>
>Assuming a goal of achieving "assured QoS" using generic SIP:
>(In assured QoS,  the phone doesn't ring until resource allocation has
>been confirmed
>vs. enabled QoS, when the phone rings and can be answered before QoS is
>established.
>In fact, if resources are not available, the enabled QoS conversation
>would continue
>(without QoS)).
>
>I understand that assured QoS can be achieved using the provisioned
>(push) model of COPS.
>(REQ, ...  DEC, RPT, ... DEC, RTP, ... DEC, RPT ...)
>Some examples of proposed provisioned models (push model) which (can)
>achieve assured QoS are:
>a) DQoS (using Gate objects)
>b) COPS-PR (using ASN.1 encoded objects),
>... which ones am I leaving out?
>
>However, I have not seen assured QoS achieved using the
>outsourcing model (pull model) of COPS.
>(REQ, DEC, RPT, .... REQ, DEC, RPT,  ... REQ, DEC, RPT, ...)
>
>Is it even possible to achieve assured QoS using the outsourcing model
>(pull model) of COPS?
>
>If so, could someone please sketch a brief call flow for me of
>how to achieve assured QoS using the outsourcing model of COPS?
>
>Thanks for your help!
>
>Bryan
>
>Bryan J. Byerly
>byerly@cisco.com
>
>
>



From owner-sip-outgoing@lists.research.bell-labs.com  Fri Feb 25 19:51:49 2000
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Subject: Re: [Fwd: Aurthorization & Proxy-Authorization URIs]
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Jonathan Rosenberg wrote:
> 
> ?? I guess the quotes shouldn't be there according to the current BNF,
> but it might make parsing harder.?

Since the digest authentication mechanism attempts to follow RFC 2617
(obviously, 2617 postdates 2543, but it will be the normative reference
in the next round), request-uri is indeed defined without being a quoted
string, even though the uri is not a token. (Unless you put an unescaped
comma into the URI, there shouldn't be delineation problems. But we had
this discussion in another context, where it became clear that URIs can
indeed contain commas.) Some parsers allow all parameters to be either
tokens or quoted strings, since that's easiest and never wrong.

Unfortunately, RFC 2617 itself is not consistent. In the BNF (3.2.2),
it's without quotes, in the example (Section 3.5), it shows up *with*
quotes. I guess we need to contact the RFC 2617 authors to find out what
people really do and what they intended. Another instance of URIs (in
WWW-Authenticate) is always with quotes.

Thus, the confusion is more fundamental.

> 
> Subject: Aurthorization & Proxy-Authorization URIs
> Date: Mon, 21 Feb 2000 14:53:22 -0500
> From: Linden deCarmo <ldeCarmo@netspeak.com>
> To: sip@lists.research.bell-labs.com
> 
> The BNF for digest-uri defined by "HTTP Authentication: Basic and Digest
> Access Authentication" for use in Aurthorization & Proxy-Authorization
> headers is:
> 
> digest-uri       = "uri" "=" digest-uri-value
> digest-uri-value = request-uri   ; As specified by HTTP/1.1
> 
> RFC2543, updates this definition to be:
> 
>         2.   The BNF for digest-uri-value is:
> 
> digest-uri-value  =  Request-URI ; a defined in Section 4.3
> 
> Yet, all the sample call flows for the uri parameters for the Aurthorization
> & Proxy-Authorization headers that I've seen (i.e. SIP Telephony Service
> Examples With Call Flows)
> enclose the uri parameter in quotes (i.e. a quoted string) just like the
> HTTP docs:
> 
> uri="sip:xxx.yyyy.com"
> 
> Syntacically, I can't see how this is possible.  Am I wrong?  If so, could
> someone point out the relevant section that explains how this is possible?
> 
> Thank you.
> 
> Linden deCarmo
> Netspeak Corporation
> 902 Clint Moore Road
> Suite 104
> Boca Raton, FL 33487



From owner-sip-outgoing@lists.research.bell-labs.com  Sat Feb 26 14:45:38 2000
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From: "Thomas Magedanz" <magedanz@fokus.gmd.de>
To: <sob@harvard.edu>, <vern@aciri.org>, <spirits@lists.research.bell-lab>,
        <sigtran@standards.nortelnetworks.com>,
        <megaco@standards.nortelnetworks.com>,
        <sip@lists.research.bell-labs.com>
Subject: CFP: Computer Networks Special Issue on IN and IP Convergence
Date: Fri, 25 Feb 2000 14:17:24 +0100
Message-ID: <008101bf7f92$afc661f0$ec4e25c3@ikvintern.de>
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Ladies and Gentlemen!

Computer Networks Journal plans to publish in February 2001 a special
issue on "Voice / Data Integration - A Snapshot of Intelligent Network
and Internet Convergence".

Guest editors: T. Magedanz (IKV++) and M. Smirnov (GMD FOKUS)

Deadline for submission is June 2000!
Maybe this is a good candidate for disseminating your results!

For more infos look at: http://www.fokus.gmd.de/events/IN-Internet/

Best regards,
Thomas Magedanz (magedanz@ikv.de)





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Date: Fri, 25 Feb 2000 10:08:05 -0600
From: Henry Sinnreich <henry.sinnreich@wcom.com>
Subject: RE: Question on achieveing Assured QoS using SIP/COPS
In-reply-to: <38B6A1ED.C9396BFA@cisco.com>
To: Bryan Byerly <byerly@cisco.com>, sip@lists.research.bell-labs.com,
        rap@iphighway.com
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>If so, could someone please sketch a brief call flow for me of
>how to achieve assured QoS using the outsourcing model of COPS?

Please see the attached PowerPoint charts from the last SIP WG
meeting.
Sorry, the ASCII format is just not too legible, but is in
http://www.ietf.org/internet-drafts/draft-sinnreich-interdomain-
sip-qos-osp-00.txt

Henry


>-----Original Message-----
>From: owner-sip@lists.research.bell-labs.com
>[mailto:owner-sip@lists.research.bell-labs.com]On
>Behalf Of Bryan Byerly
>Sent: Friday, February 25, 2000 9:38 AM
>To: sip@lists.research.bell-labs.com; rap@iphighway.com
>Cc: byerly@cisco.com
>Subject: Question on achieveing Assured QoS using SIP/COPS
>
>
>Hi guys,
>
>Assuming a goal of achieving "assured QoS" using generic SIP:
>(In assured QoS,  the phone doesn't ring until
>resource allocation has
>been confirmed
>vs. enabled QoS, when the phone rings and can be
>answered before QoS is
>established.
>In fact, if resources are not available, the enabled
>QoS conversation
>would continue
>(without QoS)).
>
>I understand that assured QoS can be achieved using
>the provisioned
>(push) model of COPS.
>(REQ, ...  DEC, RPT, ... DEC, RTP, ... DEC, RPT ...)
>Some examples of proposed provisioned models (push
>model) which (can)
>achieve assured QoS are:
>a) DQoS (using Gate objects)
>b) COPS-PR (using ASN.1 encoded objects),
>... which ones am I leaving out?
>
>However, I have not seen assured QoS achieved using the
>outsourcing model (pull model) of COPS.
>(REQ, DEC, RPT, .... REQ, DEC, RPT,  ... REQ, DEC, RPT, ...)
>
>Is it even possible to achieve assured QoS using the
>outsourcing model
>(pull model) of COPS?
>
>If so, could someone please sketch a brief call flow for me of
>how to achieve assured QoS using the outsourcing model of COPS?
>
>Thanks for your help!
>
>Bryan
>
>Bryan J. Byerly
>byerly@cisco.com
>
>

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ug0ADwAAAAAAAAAAACAAtoEAAAAASW50ZXJkb21haW4ucHB0UEsFBgAAAAABAAEAPQAAAP8TAQAA
AA==

------=_NextPart_000_0006_01BF7F78.354E7300--




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It is my understanding is that 'assured QoS' has nothing to do with the
authorisation for which COPS is used. Can you elaborate your question?

Burcak Beser
3Com Carrier Systems Group





Bryan Byerly <byerly@cisco.com> on 02/25/2000 09:38:21 AM

To:   sip@lists.research.bell-labs.com, rap@iphighway.com
cc:   byerly@cisco.com (bcc: Burcak Beser/MW/US/3Com)

Subject:  Question on achieveing Assured QoS using SIP/COPS




Hi guys,

Assuming a goal of achieving "assured QoS" using generic SIP:
(In assured QoS,  the phone doesn't ring until resource allocation has
been confirmed
vs. enabled QoS, when the phone rings and can be answered before QoS is
established.
In fact, if resources are not available, the enabled QoS conversation
would continue
(without QoS)).

I understand that assured QoS can be achieved using the provisioned
(push) model of COPS.
(REQ, ...  DEC, RPT, ... DEC, RTP, ... DEC, RPT ...)
Some examples of proposed provisioned models (push model) which (can)
achieve assured QoS are:
a) DQoS (using Gate objects)
b) COPS-PR (using ASN.1 encoded objects),
... which ones am I leaving out?

However, I have not seen assured QoS achieved using the
outsourcing model (pull model) of COPS.
(REQ, DEC, RPT, .... REQ, DEC, RPT,  ... REQ, DEC, RPT, ...)

Is it even possible to achieve assured QoS using the outsourcing model
(pull model) of COPS?

If so, could someone please sketch a brief call flow for me of
how to achieve assured QoS using the outsourcing model of COPS?

Thanks for your help!

Bryan

Bryan J. Byerly
byerly@cisco.com







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From: Francois Menard List Account <fm-listproc@fmmo.ca>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: Dean Willis <dean.willis@wcom.com>,
        Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
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Subject: Re: DHCP option tags for SIP
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> another binding, mapping their name (sip:dean.willis@wcom.com) to the ID
> of the phone (sip:3736489@wcom.com). Is that right?
> 
> Neat idea... solves the problem that a standalone phone is not always
> going to have an easy way for the user to enter their name that it
> should use in a registration. It is basically adding another layer of
> indirection: user->device->address, rather than just user->address.
> 

I've always maintained that there should be an easy way for a user
to log into a SIP phone before doing anything.  Ever since I heard that 
such a think has been tried out at 3COM, the thought of zapping my profile
using my Palm Pilot into a SIP phone with a SIR serial port remained
anchored in my brain.  I've always thought since that every
SIP phone should have an IR port for fast logons.

What comes with the profile is sometimes quite painful to enter manually,
when you think about it, a 2048bit PGP key isin't so easy to key in
either.

What I mean by this is that relying on MAC addresses as UUIDs and phone
numbers as authentification certificates has to stop at a point.  I do not
like the idea of resorting to multiple levels of indirection to circumvent
the problem of lack of enough information to begin with.  

-=Francois=-






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Folks,

I've submitted a revised version of the Supported header draft, removing
the feature negotiation, as discussed on the list. The plan of attack is
to submit this for proposed standard asap, and then fold it into the
next draft of the sip spec. The revised draft defines the SUpported
header, and specifies its usage in requests and in response to OPTIONS. 

Until the draft appears in the archives, you can pick up a copy at:
http://www.cs.columbia.edu/~jdrosen/papers/draft-ietf-sip-serverfeatures-01.txt

I will send this to IESG for proposed next Monday (March 6), unless
there are still comments/concerns.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 28 01:17:50 2000
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From: Shail Bhatnagar <shbhatna@cisco.com>
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Subject: URI comparison usage
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The question about URL comparison rules has come up a number of times in the
past and I think rules have been defined. My question is from a proxy
server standpoint. Why can't a 
SIP proxy server simply do a strcmp() for the From header and probably
a strcmp() for the addr-spec portion of the To header ?? Why would any 
proxy or user agent do anything with the From or To header except appending
a tag to the To ? IS it really necessary for the proxy to parse the input 
From/To and compare against parsed form of the From/To stored in the fast 
lookup data structure ??

Also, my guess is that the following 
    sip:abc@domain.com;tag=1234-ABC

is not a valid To header since tag could be a user-defined parameter in
the SIP-URL rather than a addr-param. Is it said somewhere in the spec or 
I am missing something ? 

Thanks,
Shail



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Shail Bhatnagar wrote:
> 
> The question about URL comparison rules has come up a number of times in the
> past and I think rules have been defined. My question is from a proxy
> server standpoint. Why can't a
> SIP proxy server simply do a strcmp() for the From header and probably
> a strcmp() for the addr-spec portion of the To header ?? Why would any
> proxy or user agent do anything with the From or To header except appending
> a tag to the To ? IS it really necessary for the proxy to parse the input
> From/To and compare against parsed form of the From/To stored in the fast
> lookup data structure ??

I think so. User agents might do things like re-order parameters, for
example. Since they must parse these headers completely, its not
desirable for them to have to remember the details of the LWS and
parameter ordering present in the field before it was parsed.

> 
> Also, my guess is that the following
>     sip:abc@domain.com;tag=1234-ABC
> 
> is not a valid To header since tag could be a user-defined parameter in
> the SIP-URL rather than a addr-param. Is it said somewhere in the spec or
> I am missing something ?

No, this is valid. If, for some reason, someone defines a SIP URL
parameter called tag, the To field would have been:

To: <sip:abc@domain.com;tag=1234-ABC>;tag=1122

Where the real To tag is 1122.

Remember, if the URL contains a semicolon (which means it has a
parameter), it must be enclosed in angle brackets. Thus, if there are
angle brackets, tag is outside of it. If there are no angle brackets,
there are no URL parameters.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 28 06:10:06 2000
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Reply-To: "yinshaohua" <yin@huawei.com.cn>
From: "yinshaohua" <yin@huawei.com.cn>
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Hi:

   In RFC2543, The re-INVITE is used to change the session descrioption of a
existing call.
   If a UA receives a INVITE request for an existing call leg with a higher
CSeq sequence number when processing a previous INVITE , I think the UA
SHOULD stop the transaction of previous INVITE and then process the
re-INVITE request, otherwise, there have two transactions in a call leg and
may lead to confusion of call state.
   Would you tell me whether I am correct ? If not, how to process the
re-INVITE?
   Thanks!

ShaoHua Yin
2000/2/28









From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 28 07:00:03 2000
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To: "yinshaohua" <yin@huawei.com.cn>
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Subject: Re:
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Hello yinshaohua,

Monday, February 28, 2000, 2:04:10 PM, you wrote:

yinshaohua> Hi:

yinshaohua>    In RFC2543, The re-INVITE is used to change the session descrioption of a
yinshaohua> existing call.
yinshaohua>    If a UA receives a INVITE request for an existing call leg with a higher
yinshaohua> CSeq sequence number when processing a previous INVITE , I think the UA
yinshaohua> SHOULD stop the transaction of previous INVITE and then process the
yinshaohua> re-INVITE request, otherwise, there have two transactions in a call leg and
yinshaohua> may lead to confusion of call state.
yinshaohua>    Would you tell me whether I am correct ? If not, how to process the
yinshaohua> re-INVITE?
yinshaohua>    Thanks!

yinshaohua> ShaoHua Yin
yinshaohua> 2000/2/28


I think that You are mistaken in the moment, when UA should stop the
previous INVITE-transaction. If re-INVITE has incompatible session
description (and You stopped the current INVITE-transaction and RTP
media flows), You can't return to the previous call
(INVITE-transaction). My opinion is that the UA should stop the previous
INVITE-transaction, if re-INVITE transaction has been finished (200-response has been received).


-- 
Best regards,
 Stanislav S. Timinsky                      mailto:timinsky@mera.ru
 Mera Labs.





From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 28 07:18:08 2000
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        "Henning Schulzrinne (E-mail)" <schulzrinne@cs.columbia.edu>
Subject: Re: UAC use of tags received in responses
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Jonathan Rosenberg wrote:
> 
> Anders Kristensen wrote:
> >
> > I'm not sure it's a problem, but this can give rise to curious
> > situations when the the non-2xx final response was generated by a UAS.
> > In this case the UAS has inserted a To tag and hence believes the call
> > consists of a single leg, whose UAS identifier (the To of the original
> > request) has a tag.
> >
> > The UAC ACKs the response (ACK includes To tag) and resubmits the
> > request *without* To tag. This basically constitutes initiating a new
> > transaction in a new leg.
> >
> > Now I'm not sure if it's well-specified whether the UAS will associate
> > that new request with the existing leg (which has a To tag). I think
> > this would (or should) be the correct behaviour, as it really is the
> > same leg, but I would imagine that different implementations might
> > differ on this point. The consequence is that the second response (say a
> > 2xx) may be returned with the *original* To tag.
> 
> Hmm. I don't think this would happen. If the UAS receives a request and
> sends a non-200 response with a tag, the call leg is over. It doesn't
> know whether it will ever receieve another request, so I imagine it will
> flush the state, and thats it.

This would be true if it wasn't because the UAS has to keep transaction
state hanging around for retransmissions - for a period of approximately
30 seconds. This is easily enough time for someone to manually type in a
password and resubmit the request.

> When the new request arrives, it creates
> a new call leg. The tag in the response will probably be different than
> the previous, or it might be the same.
> 
> On the UAC, this basically means it really can't associate a tag with
> the call leg until the 200 OK. Before that, any tags it sees in
> provisionals or non-200s might not be the final tag.

That might work, although it's a bit awkward. It's basically saying that
the UAC shouldn't represent the initial leg as a real leg.

> 
> > Maybe it should be clarified whether the UAS should respond to the
> > second request with the old or a new To tag, and whether the UAC is
> > supposed to think of the two requests as belonging to different legs?
> 
> I don't think we can impose requirements on maintenance of state at a UA
> after rejecting a new INVITE.

So you're saying that both behaviours are acceptable: if the UAS knows
of the old leg it may or may not respond with the old tag, and it's up
to the UAC to recognize that the response to the resubmitted request
belongs to the new leg.  This seems like a reasonable solution to me,
although it basically modifies the rule that a leg is identified by
Call-ID, From, and To headers.

It is maybe worth noting that the UAC can address this problem
differently by forcing the reissued request to belong to a different leg
(by using a different From tag) or a different call (by using a
different Call-ID). This denies the UAS (and proxies) the opportunity to
recognize that the second request is related to the first, and so should
be discouraged.

Anders

-- 
Anders Kristensen <ak@hplb.hpl.hp.com>,
http://www-uk.hpl.hp.com/people/ak/
Hewlett-Packard Labs, Bristol, UK



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My understanding is that CANCELs are hop-by-hop requests only, i.e.
proxies respond to CANCELs and then may or may not themselves CANCEL
branches of the corresponding proxied request. These CANCELs then have a
single Via header identifying the proxy.  However, I don't think it
states this conclusively in the spec.  Also this seems to me to be an
(insignificant maybe) security hole as CANCELs cannot be authenticated
by the caller?

Anders

-- 
Anders Kristensen <ak@hplb.hpl.hp.com>,
http://www-uk.hpl.hp.com/people/ak/
Hewlett-Packard Labs, Bristol, UK



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 28 09:36:05 2000
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"Stanislav S. Timinsky" wrote:
> 
> Hello yinshaohua,
> 
> Monday, February 28, 2000, 2:04:10 PM, you wrote:
> 
> yinshaohua> Hi:
> 
> yinshaohua>    In RFC2543, The re-INVITE is used to change the session descrioption of a
> yinshaohua> existing call.
> yinshaohua>    If a UA receives a INVITE request for an existing call leg with a higher
> yinshaohua> CSeq sequence number when processing a previous INVITE , I think the UA
> yinshaohua> SHOULD stop the transaction of previous INVITE and then process the
> yinshaohua> re-INVITE request, otherwise, there have two transactions in a call leg and
> yinshaohua> may lead to confusion of call state.
> yinshaohua>    Would you tell me whether I am correct ? If not, how to process the
> yinshaohua> re-INVITE?
> yinshaohua>    Thanks!
> 
> yinshaohua> ShaoHua Yin
> yinshaohua> 2000/2/28
> 
> I think that You are mistaken in the moment, when UA should stop the
> previous INVITE-transaction. If re-INVITE has incompatible session
> description (and You stopped the current INVITE-transaction and RTP
> media flows), You can't return to the previous call
> (INVITE-transaction). My opinion is that the UA should stop the previous
> INVITE-transaction, if re-INVITE transaction has been finished (200-response has been received).

No. You should always complete the transaction, otherwise you end up
with state hanging around in proxies longer than it needs to be. As a
general rule, transactions are independent and should each individually
be completed.

There is never confusion with call state or session state. Regarding
session state, the most recent SDP (as indicated by CSeq) is the one to
use. Regarding call state, non-200 class responses to reinvites do not
terminate the call, so the call remains active no matter what the
response to re-INVITE.

For example, lets assume that both session descriptions are OK. In this
case, the right approach is to send a 200 OK to the first and then the
second re-invite. If the first SDP is not acceptable, but the second is,
respond to the first with 400 and the second with 200. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 28 09:44:07 2000
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The IESG has received a request from the Session Initiation Protocol
Working Group to consider The SIP INFO Method
<draft-ietf-sip-info-method-02.txt> as a Proposed Standard.

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by March 13, 2000.

Files can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-sip-info-method-02.txt





From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 28 11:56:08 2000
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Anders Kristensen wrote:
> 
> My understanding is that CANCELs are hop-by-hop requests only, i.e.
> proxies respond to CANCELs and then may or may not themselves CANCEL
> branches of the corresponding proxied request. These CANCELs then have a
> single Via header identifying the proxy. 

True, excepting that stateless proxies will forward CANCEL, so there may
be more than one Via if there are stateless proxies between.

> However, I don't think it
> states this conclusively in the spec. 

Well, there is this text:

> The Call-ID, To, the numeric part of CSeq and From headers in the
>    CANCEL request are identical to those in the original request. This
>    allows a CANCEL request to be matched with the request it cancels.
>    However, to allow the client to distinguish responses to the CANCEL
>    from those to the original request, the CSeq Method component is set
>    to CANCEL. The Via header field is initialized to the proxy issuing
>    the CANCEL request. (Thus, responses to this CANCEL request only
>    reach the issuing proxy.)
> 

Particularly, the last sentence. It could be clearer. 

 Also this seems to me to be an
> (insignificant maybe) security hole as CANCELs cannot be authenticated
> by the caller?

I think you mean by the called party. Yes, they cannot be authenticated.
The problem is that the CANCEL can be generated by any proxy along the
path. So, the right way to handle CANCEL security is via hop by hop
stuff. If, for example, there is an IPSEC connection between proxies,
some request initiated through that connection can only be cancelled
through that connection also. In the absence of IPSEC, you should only
accept a CANCEL from the same IP address that sent the request. THe spec
is completely silent on these issues. They are being worked on.

Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Date: Mon, 28 Feb 2000 12:42:39 -0500
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Now that SRV records are a Proposed Standard (RFC 2782), I would suggest
simplifying the lookup procedure to include SRV as SHOULD rather than as
an optional procedure. This does not force modifying DNS to include SRV
records, but does encourage implementation of SRV in SIP clients.

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



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Date: Wed, 23 Feb 2000 20:29:50 -0800 (PST)
From: Steve Little <steve_sip@yahoo.com>
Subject: Re: Diff. between branch and tag
To: Igor Slepchin <islepchin@dynamicsoft.com>
Cc: SIP Mailing List <sip@lists.research.bell-labs.com>
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Hi,
Ive have seem implementations insert a tag field
irrespective of whether it forks or not.
Also, if forking proxies insert tags while forwarding
the request, then isnt that enough ? Then when a
receiving entity gets the request, it responds and
copies the tag in it. That way requests can be matched
wth responses, and the generation of tags can be the
rsponsibility of the entity that actually forks a
request - why should a reciever try and figure out if
its forked ? Isnt it simpler for the sender to do it ?

Thx




--- Igor Slepchin <islepchin@dynamicsoft.com> wrote:
> Branch IDs allow proxies to match responses to
> forked requests. Without
> them, a proxy wouldn't be able to tell which branch
> a response
> corresponds to. Tags are of no help here since they
> are not known until
> responses arrive.
> 
> ---
> Igor Slepchin
> 
> 
> Steve Little wrote:
> > 
> > Hi,
> > What is the difference between inserting a branch
> > param in a Via and a tag in To, From headers ?
> > Dont both uniquely identify req-resp for forking
> > proxies ?
> > 
> > Thx
> > 
> > __________________________________________________
> > Do You Yahoo!?
> > Talk to your friends online with Yahoo! Messenger.
> > http://im.yahoo.com
> 
__________________________________________________
Do You Yahoo!?
Talk to your friends online with Yahoo! Messenger.
http://im.yahoo.com



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Date: Sat, 26 Feb 2000 21:36:42 -0500
To: sip@lists.research.bell-labs.com, rap@iphighway.com, byerly@cisco.com
From: Shai Herzog <herzog@iphighway.com>
Subject: Re: FW: Question on achieveing Assured QoS using SIP/COPS
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--=====================_171463521==_.ALT
Content-Type: text/plain; charset="us-ascii"; format=flowed

Bryan, Burcak

Let me make a few assumptions:

1. We are talking about Cable modems and PacketCable DQoS (DOCSIS 1.1)
2. We are looking per-call QoS
3. "Assured" means Resources are allocated before a ring (a ring requires
    QoS), and resources are committed after the other side picked up
    (call commences).

The DQoS is a bit of an odd-ball standard, since it mixes RSVP
and COPS-RSVP Push model (gate operations). But, we all know that
COPS-RSVP wasn't designed to be in a push model.

I think there is a "better" solution just by using RSVP to allocate,
open and close gates:

Stage 1 (in parallel)

MTA-->(NCD/DCS)-->CMS/PDP (CMS makes a gate decision, resources are allocated)
  |                /|\
   \                | COPS-RSVP (outsourcing)
    \               |
     \(RSVP 1)---->CMTS

Stage 2:

Phone ring, other side answer

MTA-->(RSVP 2)---->CMTS    (confirms the reservation)

-----
RSVP(1) only ask for authorization and allocation, no actual commitment
         of resources. Resources are still in solicited mode.
RSVP(2) Commits resources (goes into unsolicited mode).
The difference may be a flag (allocate only, or commit).

This way, the Outsourcing model is maintained in COPS-RSVP (and
we all breath easier ;-), and basically only one protocol is
used to authorize allocate and reserve resources, i.e., RSVP/COPS.

I believe this is the closest DQoS can get to be compliant with
the COPS-RSVP architecture.

Shai

>-----Original Message-----
>From: Burcak_Beser@3com.com
>To: Bryan Byerly
>Cc: sip@lists.research.bell-labs.com; rap@iphighway.com;
>byerly@cisco.com
>Sent: 02/25/2000 11:49 AM
>Subject: Re: Question on achieveing Assured QoS using SIP/COPS
>
>
>
>It is my understanding is that 'assured QoS' has nothing to do with the
>authorisation for which COPS is used. Can you elaborate your question?
>
>Burcak Beser
>3Com Carrier Systems Group
>
>
>
>
>
>Bryan Byerly <byerly@cisco.com> on 02/25/2000 09:38:21 AM
>
>To:   sip@lists.research.bell-labs.com, rap@iphighway.com
>cc:   byerly@cisco.com (bcc: Burcak Beser/MW/US/3Com)
>
>Subject:  Question on achieveing Assured QoS using SIP/COPS
>
>
>
>
>Hi guys,
>
>Assuming a goal of achieving "assured QoS" using generic SIP:
>(In assured QoS,  the phone doesn't ring until resource allocation has
>been confirmed
>vs. enabled QoS, when the phone rings and can be answered before QoS is
>established.
>In fact, if resources are not available, the enabled QoS conversation
>would continue
>(without QoS)).
>
>I understand that assured QoS can be achieved using the provisioned
>(push) model of COPS.
>(REQ, ...  DEC, RPT, ... DEC, RTP, ... DEC, RPT ...)
>Some examples of proposed provisioned models (push model) which (can)
>achieve assured QoS are:
>a) DQoS (using Gate objects)
>b) COPS-PR (using ASN.1 encoded objects),
>... which ones am I leaving out?
>
>However, I have not seen assured QoS achieved using the
>outsourcing model (pull model) of COPS.
>(REQ, DEC, RPT, .... REQ, DEC, RPT,  ... REQ, DEC, RPT, ...)
>
>Is it even possible to achieve assured QoS using the outsourcing model
>(pull model) of COPS?
>
>If so, could someone please sketch a brief call flow for me of
>how to achieve assured QoS using the outsourcing model of COPS?
>
>Thanks for your help!
>
>Bryan
>
>Bryan J. Byerly
>byerly@cisco.com


__________________________________________________________________
Shai Herzog, Founder & CTO   IPHighway Inc.   Tel : (914) 654-4810
55 New York Avenue                            Main: (508) 620-1141
Framingham, MA 01701                          Fax : (212) 656-1006




                              
--=====================_171463521==_.ALT
Content-Type: text/html; charset="us-ascii"

<html>
Bryan, Burcak<br>
<br>
Let me make a few assumptions:<br>
<br>
1. We are talking about Cable modems and PacketCable DQoS (DOCSIS
1.1)<br>
2. We are looking per-call QoS<br>
3. &quot;Assured&quot; means Resources are allocated before a ring (a
ring requires<br>
&nbsp;&nbsp; QoS), and resources are committed after the other side
picked up <br>
&nbsp;&nbsp; (call commences).<br>
<br>
The DQoS is a bit of an odd-ball standard, since it mixes RSVP<br>
and COPS-RSVP Push model (gate operations). But, we all know that <br>
COPS-RSVP wasn't designed to be in a push model.<br>
<br>
I think there is a &quot;better&quot; solution just by using RSVP to
allocate,<br>
open and close gates:<br>
<br>
Stage 1 (in parallel)<br>
<br>
MTA--&gt;(NCD/DCS)--&gt;CMS/PDP (CMS makes a gate decision, resources are
allocated)<br>
&nbsp;|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
/|\<br>
&nbsp;
\&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
| COPS-RSVP (outsourcing)<br>
&nbsp;&nbsp;
\&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
| <br>
&nbsp;&nbsp;&nbsp; \(RSVP 1)----&gt;CMTS<br>
<br>
Stage 2:<br>
<br>
Phone ring, other side answer<br>
<br>
MTA--&gt;(RSVP 2)----&gt;CMTS&nbsp;&nbsp;&nbsp; (confirms the
reservation)<br>
<br>
-----<br>
RSVP(1) only ask for authorization and allocation, no actual
commitment<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; of resources. Resources are
still in solicited mode.<br>
RSVP(2) Commits resources (goes into unsolicited mode).<br>
The difference may be a flag (allocate only, or commit).<br>
<br>
This way, the Outsourcing model is maintained in COPS-RSVP (and <br>
we all breath easier ;-), and basically only one protocol is <br>
used to authorize allocate and reserve resources, i.e., RSVP/COPS.<br>
<br>
I believe this is the closest DQoS can get to be compliant with <br>
the COPS-RSVP architecture.<br>
<br>
Shai<br>
<br>
<blockquote type=cite cite>-----Original Message-----<br>
From: Burcak_Beser@3com.com<br>
To: Bryan Byerly<br>
Cc: sip@lists.research.bell-labs.com; rap@iphighway.com;<br>
byerly@cisco.com<br>
Sent: 02/25/2000 11:49 AM<br>
Subject: Re: Question on achieveing Assured QoS using SIP/COPS<br>
<br>
<br>
<br>
It is my understanding is that 'assured QoS' has nothing to do with
the<br>
authorisation for which COPS is used. Can you elaborate your
question?<br>
<br>
Burcak Beser<br>
3Com Carrier Systems Group<br>
<br>
<br>
<br>
<br>
<br>
Bryan Byerly &lt;byerly@cisco.com&gt; on 02/25/2000 09:38:21 AM<br>
<br>
To:&nbsp;&nbsp; sip@lists.research.bell-labs.com, rap@iphighway.com<br>
cc:&nbsp;&nbsp; byerly@cisco.com (bcc: Burcak Beser/MW/US/3Com)<br>
<br>
Subject:&nbsp; Question on achieveing Assured QoS using SIP/COPS<br>
<br>
<br>
<br>
<br>
Hi guys,<br>
<br>
Assuming a goal of achieving &quot;assured QoS&quot; using generic
SIP:<br>
(In assured QoS,&nbsp; the phone doesn't ring until resource allocation
has<br>
been confirmed<br>
vs. enabled QoS, when the phone rings and can be answered before QoS
is<br>
established.<br>
In fact, if resources are not available, the enabled QoS
conversation<br>
would continue<br>
(without QoS)).<br>
<br>
I understand that assured QoS can be achieved using the provisioned<br>
(push) model of COPS.<br>
(REQ, ...&nbsp; DEC, RPT, ... DEC, RTP, ... DEC, RPT ...)<br>
Some examples of proposed provisioned models (push model) which
(can)<br>
achieve assured QoS are:<br>
a) DQoS (using Gate objects)<br>
b) COPS-PR (using ASN.1 encoded objects),<br>
... which ones am I leaving out?<br>
<br>
However, I have not seen assured QoS achieved using the<br>
outsourcing model (pull model) of COPS.<br>
(REQ, DEC, RPT, .... REQ, DEC, RPT,&nbsp; ... REQ, DEC, RPT, ...)<br>
<br>
Is it even possible to achieve assured QoS using the outsourcing
model<br>
(pull model) of COPS?<br>
<br>
If so, could someone please sketch a brief call flow for me of<br>
how to achieve assured QoS using the outsourcing model of COPS?<br>
<br>
Thanks for your help!<br>
<br>
Bryan<br>
<br>
Bryan J. Byerly<br>
byerly@cisco.com<br>
</blockquote><br>
<br>
<div>__________________________________________________________________</div>
<div>Shai Herzog, Founder &amp; CTO&nbsp;&nbsp; IPHighway
Inc.&nbsp;&nbsp; Tel : (914) 654-4810</div>
<div>55 New York
Avenue&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;
Main: (508) 620-1141</div>
<div>Framingham, MA
01701&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;
Fax : (212) 656-1006</div>
<div>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</div>
<br>
<div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </div>
<div>&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;
</div>
&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; 
</html>

--=====================_171463521==_.ALT--




From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 28 14:38:31 2000
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Date: Mon, 28 Feb 2000 13:35:11 -0600
From: Alan Johnston <alan.johnston@wcom.com>
Subject: Re: CANCEL: Hop-by-hop or end-to-end?
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: Anders Kristensen <ak@hplb.hpl.hp.com>,
        SIP <sip@lists.research.bell-labs.com>
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The current version of the SIP Telephony Call Flow Examples document
(draft-johnston-sip-call-flows-00.txt) incorrectly shows end-to-end behavior for
CANCELs.  This is corrected in the new draft which will be published this week.

Alan Johnston
MCI WorldCom

Jonathan Rosenberg wrote:
> 
> Anders Kristensen wrote:
> >
> > My understanding is that CANCELs are hop-by-hop requests only, i.e.
> > proxies respond to CANCELs and then may or may not themselves CANCEL
> > branches of the corresponding proxied request. These CANCELs then have a
> > single Via header identifying the proxy.
> 
> True, excepting that stateless proxies will forward CANCEL, so there may
> be more than one Via if there are stateless proxies between.
> 
> > However, I don't think it
> > states this conclusively in the spec.
> 
> Well, there is this text:
> 
> > The Call-ID, To, the numeric part of CSeq and From headers in the
> >    CANCEL request are identical to those in the original request. This
> >    allows a CANCEL request to be matched with the request it cancels.
> >    However, to allow the client to distinguish responses to the CANCEL
> >    from those to the original request, the CSeq Method component is set
> >    to CANCEL. The Via header field is initialized to the proxy issuing
> >    the CANCEL request. (Thus, responses to this CANCEL request only
> >    reach the issuing proxy.)
> >
> 
> Particularly, the last sentence. It could be clearer.
> 
>  Also this seems to me to be an
> > (insignificant maybe) security hole as CANCELs cannot be authenticated
> > by the caller?
> 
> I think you mean by the called party. Yes, they cannot be authenticated.
> The problem is that the CANCEL can be generated by any proxy along the
> path. So, the right way to handle CANCEL security is via hop by hop
> stuff. If, for example, there is an IPSEC connection between proxies,
> some request initiated through that connection can only be cancelled
> through that connection also. In the absence of IPSEC, you should only
> accept a CANCEL from the same IP address that sent the request. THe spec
> is completely silent on these issues. They are being worked on.
> 
> Thanks,
> Jonathan R.
> 
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com



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From: Jean-Francois Mule <jfmule@clarent.com>
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Cc: sip@lists.research.bell-labs.com
Subject: SRV records
Date: Mon, 28 Feb 2000 11:56:14 -0800
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I second that.
On the server side, SRV RR (experimental RFC2052) has been supported in BIND
since version 4.9.5 (the latest being BIND 8.2.2. includes an update on SRV
RR).
Jean-Francois.
--------------------------------------------------------------
Jean-François Mulé
Clarent Corporation - Standards & Architecture Group
T: +1 650 481-2835
jfm@clarent.com

> -----Original Message-----
> From: Henning Schulzrinne [mailto:schulzrinne@cs.columbia.edu]
> Sent: Monday, February 28, 2000 9:43 AM
> To: sip@lists.research.bell-labs.com
> Subject: SRV records
> 
> 
> Now that SRV records are a Proposed Standard (RFC 2782), I 
> would suggest
> simplifying the lookup procedure to include SRV as SHOULD 
> rather than as
> an optional procedure. This does not force modifying DNS to 
> include SRV
> records, but does encourage implementation of SRV in SIP clients.
> 
> Comments?
> -- 
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> 



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Date: Mon, 28 Feb 2000 14:23:49 -0600
From: Mart Nurmet <mart.nurmet@inet.com>
Organization: Inet - Product Line Management
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To: sip@lists.research.bell-labs.com
Subject: Information request!  (Billing Package?)
References: <200002151701.LAA11063@b04a24.exu.ericsson.se>
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I am looking for information if there has been any work done to generate a
billing package for SIP.  From my look at the available information there does
not seem to be any.  Is that the case or am I missing something?

Thank you,
-Mart




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To: Shai Herzog <herzog@iphighway.com>
Cc: sip@lists.research.bell-labs.com, rap@iphighway.com, byerly@cisco.com
Message-ID: <88256893.00708D4F.00@hqoutbound.ops.3com.com>
Date: Mon, 28 Feb 2000 14:31:17 -0600
Subject: Re: FW: Question on achieveing Assured QoS using SIP/COPS
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Shai,

> 1. We are talking about Cable modems and PacketCable DQoS (DOCSIS 1.1)

I do not think that this is a PacketCable specific requirement. I believe that
Bryan set the objective very good:

'In assured QoS,  the phone doesn't ring until resource allocation has
been confirmed vs. enabled QoS, when the phone rings and can be answered before
QoS is established.'

I also believe that this will be vald for other applications as well. One that
comes to my mind immediately is Gaming applications.

Although DQoS uses COPS-push model for authorisation it is not a requirement of
the 'assured QoS'.

As you mention in your mail the 'assured QoS' is related with the QoS signalling
which can be achieved without any Authorisation (i.e. need for COPS protocol).

Related with the COPS-push or COPS-pull models it is very possible to model the
below architecture (Which was depicted in "draft-dcsgroup-sip-call-auth").

Client         CMTS      Call Server
|              |              |
| Invite       |              |    Invite
|---------------------------------->|------------------------>
|              |              | Client Authentication
|              |  Gate Setup  | and Call Authorisation
|              |<----------------|
|              |
|              | Gate-Setup includes an Authorised IP Flow
|              | profile and a GateID as a pointer
|              |
|              |
|              | 200OK [+GateID] | 200 OK
|<----------------------------------|<------------------------
|              |              |
| PATH [+GateID]    |
|---------------->|
|              | The PATH Message includes GateID which identifies
|              | Authorised IP flow profile. CMTS checks the
|              | committed profile against


Instead one can re-draw the above architecture as:

Client    CMTS      PDP       Call Server
|         |         |              |
| Invite  |         |              |     Invite
|---------------------------------------->|------------------->
|         |         |              | Client Authentication
|         |         |  Gate Setup  | and Call Authorisation
|         |         |<----------------|
|         |         |              |
|         |         | Auth. Token  |
|         |         |---------------->|
|         |         |
|         |         | Auth Token is used in PDP for Authorised Profile
|         |         | identification
|         |         |
|         |     200OK [+AuthToken]       |    200 OK
|<----------------------------------------|<-----------------------
|         |         |
| PATH [+AT]|       |
|---------->| REQ [+AT]  |
|         |---------->|
|         |         | PDP makes the decision using the RSVP profile and
|         | DEC          | Authorised profile
|         |<----------|
|         |         |
|         | PATH    |
|         |----------------------------------------------------->

The same messaging also applies for the RESV message.

As can be seen from above architectures the change from PUSH to PULL model can
be achieved without any change in the signalling model.

Right now I am working on the changes to "draft-dcsgroup-sip-call-auth" and
making the draft more oriented towards SIP than COPS. Until now I was thinking
that a COPS usage draft is not necessary but maybe I was mistaken.

If there is a need I can pull my notes regarding the scrapped parts of the "
draft-dcsgroup-sip-call-auth" draft and make an 'Application Authorisation per
Use' Architecture document, which would specify the means of application servers
giving the Authorised profile and pulling Authorisation Token from PDP in a
standard manner (which would be a PUSH model).

Burcak Beser
3Com Carrier Access Group





Shai Herzog <herzog@iphighway.com> on 02/26/2000 08:36:42 PM

To:   sip@lists.research.bell-labs.com, rap@iphighway.com, byerly@cisco.com
cc:    (bcc: Burcak Beser/MW/US/3Com)

Subject:  Re: FW: Question on achieveing Assured QoS using SIP/COPS




Bryan, Burcak

Let me make a few assumptions:

1. We are talking about Cable modems and PacketCable DQoS (DOCSIS 1.1)
2. We are looking per-call QoS
3. "Assured" means Resources are allocated before a ring (a ring requires
    QoS), and resources are committed after the other side picked up
    (call commences).

The DQoS is a bit of an odd-ball standard, since it mixes RSVP
and COPS-RSVP Push model (gate operations). But, we all know that
COPS-RSVP wasn't designed to be in a push model.

I think there is a "better" solution just by using RSVP to allocate,
open and close gates:

Stage 1 (in parallel)

MTA-->(NCD/DCS)-->CMS/PDP (CMS makes a gate decision, resources are allocated)
  |                /|\
   \                | COPS-RSVP (outsourcing)
    \               |
     \(RSVP 1)---->CMTS

Stage 2:

Phone ring, other side answer

MTA-->(RSVP 2)---->CMTS    (confirms the reservation)

-----
RSVP(1) only ask for authorization and allocation, no actual commitment
         of resources. Resources are still in solicited mode.
RSVP(2) Commits resources (goes into unsolicited mode).
The difference may be a flag (allocate only, or commit).

This way, the Outsourcing model is maintained in COPS-RSVP (and
we all breath easier ;-), and basically only one protocol is
used to authorize allocate and reserve resources, i.e., RSVP/COPS.

I believe this is the closest DQoS can get to be compliant with
the COPS-RSVP architecture.

Shai

>-----Original Message-----
>From: Burcak_Beser@3com.com
>To: Bryan Byerly
>Cc: sip@lists.research.bell-labs.com; rap@iphighway.com;
>byerly@cisco.com
>Sent: 02/25/2000 11:49 AM
>Subject: Re: Question on achieveing Assured QoS using SIP/COPS
>
>
>
>It is my understanding is that 'assured QoS' has nothing to do with the
>authorisation for which COPS is used. Can you elaborate your question?
>
>Burcak Beser
>3Com Carrier Systems Group
>
>
>
>
>
>Bryan Byerly <byerly@cisco.com> on 02/25/2000 09:38:21 AM
>
>To:   sip@lists.research.bell-labs.com, rap@iphighway.com
>cc:   byerly@cisco.com (bcc: Burcak Beser/MW/US/3Com)
>
>Subject:  Question on achieveing Assured QoS using SIP/COPS
>
>
>
>
>Hi guys,
>
>Assuming a goal of achieving "assured QoS" using generic SIP:
>(In assured QoS,  the phone doesn't ring until resource allocation has
>been confirmed
>vs. enabled QoS, when the phone rings and can be answered before QoS is
>established.
>In fact, if resources are not available, the enabled QoS conversation
>would continue
>(without QoS)).
>
>I understand that assured QoS can be achieved using the provisioned
>(push) model of COPS.
>(REQ, ...  DEC, RPT, ... DEC, RTP, ... DEC, RPT ...)
>Some examples of proposed provisioned models (push model) which (can)
>achieve assured QoS are:
>a) DQoS (using Gate objects)
>b) COPS-PR (using ASN.1 encoded objects),
>... which ones am I leaving out?
>
>However, I have not seen assured QoS achieved using the
>outsourcing model (pull model) of COPS.
>(REQ, DEC, RPT, .... REQ, DEC, RPT,  ... REQ, DEC, RPT, ...)
>
>Is it even possible to achieve assured QoS using the outsourcing model
>(pull model) of COPS?
>
>If so, could someone please sketch a brief call flow for me of
>how to achieve assured QoS using the outsourcing model of COPS?
>
>Thanks for your help!
>
>Bryan
>
>Bryan J. Byerly
>byerly@cisco.com


__________________________________________________________________
Shai Herzog, Founder & CTO   IPHighway Inc.   Tel : (914) 654-4810
55 New York Avenue                            Main: (508) 620-1141
Framingham, MA 01701                          Fax : (212) 656-1006






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--0__=htYn2KtOuwCcENmBujO0t5Xwl24BzyPqzO4P5hskcTH6ZhelmPwzDcQT--




From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 28 16:50:05 2000
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To: Mart Nurmet <mart.nurmet@inet.com>
Cc: sip@lists.research.bell-labs.com
Subject: Re: Information request!  (Billing Package?)
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There is no standardized "billing package" for SIP. There will be a MIB,
and there are local log and transaction records generated by servers and
gateways. This information can be collected to bill in any way you like.

-Jonathan R.

Mart Nurmet wrote:
> 
> I am looking for information if there has been any work done to generate a
> billing package for SIP.  From my look at the available information there does
> not seem to be any.  Is that the case or am I missing something?
> 
> Thank you,
> -Mart

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 28 16:56:03 2000
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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Subject: Re: SRV records
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My only concern is regarding latency. Every proxy will do an SRV query,
and in most cases, will get (and have to wait for) a negative response.
Then, it needs to do another query for A records. Negative responses can
take a long time... not sure if thats just if there is no server for
that domain, or also if there is a server for that domain, but the
requested record doesn't exist.

-Jonathan R.

Henning Schulzrinne wrote:
> 
> Now that SRV records are a Proposed Standard (RFC 2782), I would suggest
> simplifying the lookup procedure to include SRV as SHOULD rather than as
> an optional procedure. This does not force modifying DNS to include SRV
> records, but does encourage implementation of SRV in SIP clients.
> 
> Comments?
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Mon Feb 28 17:03:42 2000
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Jonathan Rosenberg wrote:
> 
> My only concern is regarding latency. Every proxy will do an SRV query,
> and in most cases, will get (and have to wait for) a negative response.
> Then, it needs to do another query for A records. Negative responses can
> take a long time... not sure if thats just if there is no server for
> that domain, or also if there is a server for that domain, but the
> requested record doesn't exist.
> 

We might have had this discussion before, but my experience is that if a
domain exists, but a record doesn't, the response is very fast (110 ms
without caching and 20 ms with caching, including the "host" application
overhead). If the domain doesn't exist, it doesn't matter one way or the
other. I just tried

host -t srv sip.udp.dynamicsoft.com
sip.udp.dynamicsoft.com does not exist (Authoritative answer)

and it returns immediately.

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



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb 29 00:43:43 2000
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Date: Tue, 29 Feb 2000 00:42:41 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Folks,

In order to move the SIP-T work towards completion, we've formed a SIP-T
design team. The goal of the design team is to rapidly reach closure on
the three remaining documents in the SIP-T umbrella - (1) the ISUP MIME
type, (2) the SIP/ISUP interworking document, and (3) the umbrella SIP-T
spec. The fourth, INFO, is now in IESG last call. The first task is to
reach closure on the MIME draft, which I think is nearly ready.

If you're interested in joining this team, please let me know and I'll
add you.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb 29 01:09:51 2000
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Steve Little wrote:
> 
> Hi,
> Ive have seem implementations insert a tag field
> irrespective of whether it forks or not.

Only a UAS inserts a tag. It has no way of knowing whether the request
forked or not. The only thing you know is that if there is a single Via
in the request when it arrives at the UAS, it didn't fork. Thats why the
spec doesn't mandate tags in this (mostly useless) case. I think its a
good idea (and probably should be a MUST) to insert a tag always.

> Also, if forking proxies insert tags while forwarding
> the request, then isnt that enough ? Then when a
> receiving entity gets the request, it responds and
> copies the tag in it.

No, proxies can't insert tags. Tags go in the To field, and there is
only one that can be placed there. If the first forking proxy inserts a
tag in the To field, as you suggest, and then it forks downstream, the
second proxy can't insert a tag, since there already is one there. So,
it leaves it alone. All user agents downstream of the second forking
proxy that respond will do so with the same tag. That defeats the whole
purpose.

> That way requests can be matched
> wth responses,

This doesn't allow requests to be matched with responses in proxies.
*EACH* forking proxy needs to insert some kind of data. There is only
room for one tag in the To field. Thats why we use the Via header for
the branch parameter - each proxy can get its own.

> and the generation of tags can be the
> rsponsibility of the entity that actually forks a
> request - why should a reciever try and figure out if
> its forked ? 

As I said, it can't, shouldn't, and doesn't have to.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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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		: The SIP Supported Header
	Author(s)	: J. Rosenberg, H. Schulzrinne
	Filename	: draft-ietf-sip-serverfeatures-01.txt
	Pages		: 7
	Date		: 28-Feb-00
	
The Session Initiation Protocol (SIP) defines mechanims that allow a
client to mandate that a server support specific extensions in order
to process a request. However, there is currently no way for a server
to determine what features are supported in a client, in order to use
those features to process the request. This capability is needed for
numerous extensions currently under development, such as provisional
response reliability and the session progress message. We also note
that there is currently no way for a client to query a server about
the features it supports. This document defines a SIP extension that
allows clients to indicate, in a request, the set of features
supported. We also define a mechanism that allows clients, through an
OPTIONS request, to determine the features supported by a server.

A URL for this Internet-Draft is:
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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		: DHCP Option for SIP Servers
	Author(s)	: G. Nair, H. Schulzrinne
	Filename	: draft-ietf-sip-dhcp-00.txt
	Pages		: 6
	Date		: 28-Feb-00
	
This document defines DHCP options that contain one or more pointers
to one or more SIP servers. This is one of the many methods that a
SIP client can use to obtain the addresses of the SIP servers.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-dhcp-00.txt

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From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb 29 06:49:54 2000
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From: "Buchanan, Paul" <Paul.Buchanan@cwcom.co.uk>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        Mart Nurmet <mart.nurmet@inet.com>
Cc: sip@lists.research.bell-labs.com
Subject: RE: Information request!  (Billing Package?)
Date: Tue, 29 Feb 2000 10:35:43 -0000
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CDR collection should be defacto and easily accessible right down to the SIP
client. 

I believe that SIP should interwork with the current VOIP suppliers like
XACCT, Mind CTI , Clarent, Sollect, Portal, Kenan and their contemporaries.
I can see a major problem with the FORKING capabilities of SIP as the
provider will be responsible for originating calls from the Gateway. In
effect we will have to make the receiver pay for calls which turns the
billing world on its head.

My concern is the security issues and, in addition, how service companies
can make an honest profit. As IP transmission costs plummet - so does the
cost of telephony generally and so does the service providers ability to
reinvest in the web infrastructure. I have several ideas on how this should
happen as SIP becomes a global product.

Paul Buchanan
Darkness Removal a Speciality 
VOIP/DSL/ATM Technical Architect IP Net Technology Planning
Waterside House, Longshot Lane, Bracknell Berks RG12 1XL
TEL:			01344 704064
FAX:			01344 726304
MOBILE (BEST BET)	0780 833 8143
PA247.com		paulbuchanan@mail.com
Home:			01227 450900
Home Fax:		01227 478929
Work Mail			paul.buchanan@cwcom.co.uk
"Those who bring sunshine into the lives of others, cannot keep it from
themselves."



> -----Original Message-----
> From:	Jonathan Rosenberg [SMTP:jdrosen@dynamicsoft.com]
> Sent:	Monday, February 28, 2000 9:50 PM
> To:	Mart Nurmet
> Cc:	sip@lists.research.bell-labs.com
> Subject:	Re: Information request!  (Billing Package?)
> 
> There is no standardized "billing package" for SIP. There will be a MIB,
> and there are local log and transaction records generated by servers and
> gateways. This information can be collected to bill in any way you like.
> 
> -Jonathan R.
> 
> Mart Nurmet wrote:
> > 
> > I am looking for information if there has been any work done to generate
> a
> > billing package for SIP.  From my look at the available information
> there does
> > not seem to be any.  Is that the case or am I missing something?
> > 
> > Thank you,
> > -Mart
> 
> -- 
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120 
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb 29 09:32:09 2000
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From: "Joshua Moloney" <jmoloney@nortelnetworks.com>
To: "'sip@lists.research.bell-labs.com'" <sip@lists.research.bell-labs.com>
Subject: Content-Encoding header (again)
Date: Tue, 29 Feb 2000 14:26:55 -0000
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This message is in MIME format. Since your mail reader does not understand
this format, some or all of this message may not be legible.

------_=_NextPart_001_01BF82C1.0B83077E
Content-Type: text/plain

Hi,

Thanks for the reply Jonathan. It helped me (I think!) with a few questions
I had about the headers, but I guess I really asked the wrong question. So
here goes...

In a SIP message with multiple payloads (i.e. multipart/mixed content-type),
the following example is often cited:

     	INVITE sip:13035553142@Den1.level3.com SIP/2.0
	From: sip:13035553355@den3.level3.com
	To: sip:13035553142@Den1.level3.com
	Call-ID: DEN1231999021712095500999@Den1.level3.com
	Content-Length: 377
	Content-Type: multipart/mixed; boundary=unique-boundary-1
	MIME-Version: 1.0          

	--unique-boundary-1
	Content-Type: application/SDP; charset=ISO-10646
	v=0
	o=ezimmerer 2890844526 2890842807 IN IP4 126.16.64.4
	s=SDP seminar
   	c=IN IP4 MG122.level3.com
	t= 2873397496	2873404696
	m=audio 9092 RTP/AVP 0 3 4

	--unique-boundary-1
	Content-type:application/ISUP; version=ETSI1
	Content-Transfer-Encoding: binary

	89 8b 0e 95 1e 1e 1e 06 26 05 0d f5 01 06 10 04 00 

	--unique-boundary-1--

The ISUP payload has a Content-Transfer-Encoding header. Is this correct? Is
it required at all (for the reasons stated in your original response), or
should the Content-Encoding header be used instead? Or, should a parser be
able to understand both types of header (thus suggesting that a SIP parser
should be fully MIME compliant)?

Cheers,

Josh Moloney
Nortel Networks

> -----Original Message-----
> From:	Jonathan Rosenberg [SMTP:jdrosen@dynamicsoft.com]
> Sent:	25 February 2000 19:17
> To:	Moloney, Joshua [MDN05:7U46:EXCH]
> Cc:	'sip@lists.research.bell-labs.com'
> Subject:	Re: Content-Encoding vs. Content-Transfer-Encoding
> 
> The Content-Transfer-Encoding header was primarily meant to allow
> message bodies to be transformed into formats that could be transferred
> on channels that were not 8 bit clean. HTTP, which makes use of many of
> the MIME headers, is 8 bit clean, and thus did not need
> Content-Transfer-Encoding. SIP followed suit, and so does not use it
> either. Content-Encoding is used for things like compression, which is
> different (in fact, the opposite, since conversion of a binary file to a
> 7 bit transfer channel INCREASES the size of the body). 
> 
> -Jonathan R.
> 
> > Joshua Moloney wrote:
> > 
> > Hi all,
> > 
> > I have a question about the "Content-Encoding" entity header.
> > Why is the MIME "Content-Transfer-Encoding" header defined in RFC 2045
> > not used instead?
> > Am I correct in assuming that they do the same job?
> > 
> > For example, what headers should I include on a message with an
> > "application/ISUP" message body? :-
> > 
> > Content-Type: application/ISUP
> > Content-Transfer-Encoding: binary
> > .....
> > 
> > OR
> > 
> > Content-Type: application/ISUP
> > Content-Encoding: binary
> > .....
> > 
> > Thanks in advance,
> > 
> > Josh Moloney
> > 
> > Nortel Networks
> > jmoloney@nortelnetworks.com
> 
> -- 
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120 
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 

------_=_NextPart_001_01BF82C1.0B83077E
Content-Type: text/html
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2651.65">
<TITLE>Content-Encoding header (again)</TITLE>
</HEAD>
<BODY>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Hi,</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Thanks for the reply =
Jonathan. It helped me (I think!) with a few questions I had about the =
headers, but I guess I really asked the wrong question. So here =
goes...</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">In a SIP message =
with multiple payloads (i.e. multipart/mixed content-type), the =
following example is often cited:</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;&nbsp; INVITE =
sip:13035553142@Den1.level3.com SIP/2.0</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Courier New">From: =
sip:13035553355@den3.level3.com</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Courier New">To: =
sip:13035553142@Den1.level3.com</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Courier New">Call-ID: =
DEN1231999021712095500999@Den1.level3.com</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Courier New">Content-Length: 377</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Courier New">Content-Type: multipart/mixed; =
boundary=3Dunique-boundary-1</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Courier New">MIME-Version: =
1.0&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </FONT>
</P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Courier New">--unique-boundary-1</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Courier New">Content-Type: application/SDP; =
charset=3DISO-10646</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Courier New">v=3D0</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Courier New">o=3Dezimmerer 2890844526 2890842807 IN =
IP4 126.16.64.4</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Courier New">s=3DSDP seminar</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Courier New">&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;&nbsp; c=3DIN IP4 MG122.level3.com</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Courier New">t=3D 2873397496&nbsp;&nbsp; =
2873404696</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Courier New">m=3Daudio 9092 RTP/AVP 0 3 4</FONT>
</P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Courier New">--unique-boundary-1</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Courier New">Content-type:application/ISUP; =
version=3DETSI1</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<B> <FONT =
COLOR=3D"#FF0000" SIZE=3D2 FACE=3D"Courier =
New">Content-Transfer-Encoding: binary</FONT></B>
</P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Courier New">89 8b 0e 95 1e 1e 1e 06 26 05 0d f5 01 06 =
10 04 00 </FONT>
</P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT COLOR=3D"#0000FF" =
SIZE=3D2 FACE=3D"Courier New">--unique-boundary-1--</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">The ISUP payload has =
a Content-Transfer-Encoding header. Is this correct? Is it required at =
all (for the reasons stated in your original response), or should the =
Content-Encoding header be used instead? Or, should a parser be able to =
understand both types of header (thus suggesting that a SIP parser =
should be fully MIME compliant)?</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Cheers,</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Josh Moloney</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Nortel =
Networks</FONT>
</P>
<UL>
<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">Jonathan Rosenberg =
[SMTP:jdrosen@dynamicsoft.com]</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">25 February 2000 19:17</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">Moloney, Joshua [MDN05:7U46:EXCH]</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Cc:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">'sip@lists.research.bell-labs.com'</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">Re: Content-Encoding vs. =
Content-Transfer-Encoding</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">The Content-Transfer-Encoding header =
was primarily meant to allow</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">message bodies to be transformed into =
formats that could be transferred</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">on channels that were not 8 bit =
clean. HTTP, which makes use of many of</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">the MIME headers, is 8 bit clean, and =
thus did not need</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Content-Transfer-Encoding. SIP =
followed suit, and so does not use it</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">either. Content-Encoding is used for =
things like compression, which is</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">different (in fact, the opposite, =
since conversion of a binary file to a</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">7 bit transfer channel INCREASES the =
size of the body). </FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">-Jonathan R.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt; Joshua Moloney wrote:</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Hi all,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; I have a question about the =
&quot;Content-Encoding&quot; entity header.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Why is the MIME =
&quot;Content-Transfer-Encoding&quot; header defined in RFC 2045</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; not used instead?</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Am I correct in assuming that =
they do the same job?</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; For example, what headers should =
I include on a message with an</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &quot;application/ISUP&quot; =
message body? :-</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Content-Type: =
application/ISUP</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Content-Transfer-Encoding: =
binary</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; .....</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; OR</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Content-Type: =
application/ISUP</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Content-Encoding: binary</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; .....</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Thanks in advance,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Josh Moloney</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Nortel Networks</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; =
jmoloney@nortelnetworks.com</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">-- </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Jonathan D. =
Rosenberg&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
200 Executive Drive</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Chief =
Scientist&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Suite 120 </FONT>
<BR><FONT SIZE=3D2 =
FACE=3D"Arial">dynamicsoft&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; West Orange, NJ 07052</FONT>
<BR><FONT SIZE=3D2 =
FACE=3D"Arial">jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp; FAX:&nbsp;&nbsp; (732) 741-4778</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://www.cs.columbia.edu/~jdrosen" =
TARGET=3D"_blank">http://www.cs.columbia.edu/~jdrosen</A></FONT></U><FON=
T SIZE=3D2 =
FACE=3D"Arial">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: =
(732) 741-7244</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://www.dynamicsoft.com" =
TARGET=3D"_blank">http://www.dynamicsoft.com</A></FONT></U>
</P>
</UL>
</BODY>
</HTML>
------_=_NextPart_001_01BF82C1.0B83077E--



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb 29 11:20:06 2000
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From: Wing Man Kwok <wkwok@telogy.com>
To: "'sip@lists.research.bell-labs.com'" <sip@lists.research.bell-labs.com>
Subject: SIP support of caller ID
Date: Tue, 29 Feb 2000 11:16:51 -0500
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All,

I read from the SIP FAQ that SIP supports caller ID.  But I cannot find any
description of this in rfc2543.  How does SIP support caller ID?

Thanks in advance

Wing Man Kwok
Telogy Networks



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb 29 11:37:54 2000
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Date: Tue, 29 Feb 2000 16:34:41 +0000
From: Benny Prijono <seventhson@theseventhson.freeserve.co.uk>
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Hi all,

Section 12.3.1 of the draft states that:
  "To prevent loops, a server must check if its own address is
   already contained in the Via header of the incoming request"

Section 6.44.1 states that a 482 (Loop Detected) response MUST be
returned when a loop is detected.

And finally, section 12.3 (last statement) states that a stateless proxy
forwards every request and response it received.

Based on these statements, I think this is the correct behaviour for
stateless proxy:
1. When the loop is detected, the stateless proxy will reply 
   with 482 (Loop Detected). 
2. The UAC then will send ACK.
3. Proxy will forward ACK to the UAS.
4. UAS will simply drop the ACK, maybe after complains loudly
   about the invalid message.

If this is the correct behaviour, then I'd like to know if there are any
ways to prevent the ACK to be received by the UAS.

Thanks in advance.

-- 
cheers,
Bennylp




From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb 29 12:34:00 2000
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Date: Tue, 29 Feb 2000 12:30:50 -0500
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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To: Wing Man Kwok <wkwok@telogy.com>
Cc: "'sip@lists.research.bell-labs.com'" <sip@lists.research.bell-labs.com>
Subject: Re: SIP support of caller ID
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Wing Man Kwok wrote:
> 
> All,
> 
> I read from the SIP FAQ that SIP supports caller ID.  But I cannot find any
> description of this in rfc2543.  How does SIP support caller ID?

Depends on the details, but basically, simply stick the caller name in
the display name field, and the number in the user name. Done.

> 
> Thanks in advance
> 
> Wing Man Kwok
> Telogy Networks

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



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Date: Tue, 29 Feb 2000 13:40:54 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: SRV records
In-reply-to: <38BAB38F.C3EC9A26@cs.columbia.edu>
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I concur.

--
Dean

> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Henning
> Schulzrinne
> Sent: Monday, February 28, 2000 11:43 AM
> To: sip@lists.research.bell-labs.com
> Subject: SRV records
> 
> 
> Now that SRV records are a Proposed Standard (RFC 2782), I would suggest
> simplifying the lookup procedure to include SRV as SHOULD rather than as
> an optional procedure. This does not force modifying DNS to include SRV
> records, but does encourage implementation of SRV in SIP clients.
> 
> Comments?
> -- 
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> 



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb 29 14:49:06 2000
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Date: Tue, 29 Feb 2000 13:41:02 -0600
From: Dean Willis <dean.willis@wcom.com>
Subject: RE: Information request!  (Billing Package?)
In-reply-to: <38BAD954.58C89727@inet.com>
To: Mart Nurmet <mart.nurmet@inet.com>, sip@lists.research.bell-labs.com
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What would a billing package look like? I've never really understood the
concept, and it might be useful to talk about it some more.

Here's how some implementors are doing it:

1) Gateways throw CDRs using RADIUS or something similar. This feeds into
rating, reporting, and billing systems,

2) SIP servers record transaction logs. These may feed into rating and
billing systems, and probably feed into reporting systems.

3) QoS PEP and PDPs log resource usages. These feed upstream as well.

4) Other systems, like profile managers, report usage.

5) Rating, reporting, and billing systems sift thru the vast amounts of
information coming at them and make useful things happen.

All of these things use other protocols and systems -- they aren't SIP
specific. They do rely on SIP functions like authentication -- that is, the
goal of achieving billing functions has driven some requirements into the
design of SIP. However, other than driving some requirements to SIP, such
efforts are generally considered to be outside the scope of the SIP WG.

Are you perhaps proposing to add functionality to SIP to simplify or enhance
reporting capabilities or to simplify the task of driving downstream billing
systems?

--
Dena
> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Mart Nurmet
> Sent: Monday, February 28, 2000 2:24 PM
> To: sip@lists.research.bell-labs.com
> Subject: Information request! (Billing Package?)
>
>
> I am looking for information if there has been any work done to generate a
> billing package for SIP.  From my look at the available
> information there does
> not seem to be any.  Is that the case or am I missing something?
>
> Thank you,
> -Mart
>
>




From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb 29 16:17:50 2000
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From: Shail Bhatnagar <shbhatna@cisco.com>
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Jonathan, Thanks for your response. I will not argue for strcmp of name-addr
but I will say that why would a SIP ua or proxy alter the name-addr portion of 
From/To. The name-addr in From/To is supposed to remain unchanged for the call.
It makes sense for any implementation of SIP to simply keep the name-addr of
From/To as it came in the first time instead of keeping it in parsed form
and generating the unparsed form from it.

Shail.


Jonathan Rosenberg wrote:
> 
>
> 
> I think so. User agents might do things like re-order parameters, for
> example. Since they must parse these headers completely, its not
> desirable for them to have to remember the details of the LWS and
> parameter ordering present in the field before it was parsed.
> 
> >
> > Also, my guess is that the following
> >     sip:abc@domain.com;tag=1234-ABC
> >
> > is not a valid To header since tag could be a user-defined parameter in
> > the SIP-URL rather than a addr-param. Is it said somewhere in the spec or
> > I am missing something ?
> 
> No, this is valid. If, for some reason, someone defines a SIP URL
> parameter called tag, the To field would have been:
> 
> To: <sip:abc@domain.com;tag=1234-ABC>;tag=1122
> 
> Where the real To tag is 1122.
> 
> Remember, if the URL contains a semicolon (which means it has a
> parameter), it must be enclosed in angle brackets. Thus, if there are
> angle brackets, tag is outside of it. If there are no angle brackets,
> there are no URL parameters.
> 
> -Jonathan R.
> --
> Jonathan D. Rosenberg                       200 Executive Drive
> Chief Scientist                             Suite 120
> dynamicsoft                                 West Orange, NJ 07052
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com

-- 
Best regards,
Shail



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb 29 17:32:17 2000
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From: "Dave Walker" <drwalker@ss8networks.com>
To: "Henning Schulzrinne" <schulzrinne@cs.columbia.edu>,
        "Wing Man Kwok" <wkwok@telogy.com>
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Subject: RE: SIP support of caller ID
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It sounds appealing.  But what can we do about a user who feels like 
lying?  For example, I'll always answer a call from "Bill Gates" (or 
will I??).  If the user is known to a proxy in the signalling path, it's
possible for that proxy to insert or validate the display name and
then, more importantly, to signal that it has done so.  This could,
for example, use a new addr-param (;screenedby=servername) in the From,
or a new header (Screened-Display: displayname servername).  Of course, 
this assumes that servers are more trustworthy than users.  If someone's 
server sends me too many fraudulent displaynames, maybe I'll just stop 
accepting calls from it.

Dave.

> -----Original Message-----
> From: owner-sip@lists.research.bell-labs.com
> [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Henning
> Schulzrinne
> Sent: Tuesday, February 29, 2000 12:31 PM
> To: Wing Man Kwok
> Cc: 'sip@lists.research.bell-labs.com'
> Subject: Re: SIP support of caller ID
> 
> 
> Wing Man Kwok wrote:
> > 
> > All,
> > 
> > I read from the SIP FAQ that SIP supports caller ID.  But I 
> cannot find any
> > description of this in rfc2543.  How does SIP support caller ID?
> 
> Depends on the details, but basically, simply stick the caller name in
> the display name field, and the number in the user name. Done.
> 
> > 
> > Thanks in advance
> > 
> > Wing Man Kwok
> > Telogy Networks
> 
> -- 
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> 
> 



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb 29 18:45:51 2000
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Forwarded from Larry Amiot; please contact him directly if interested.
Unfortunately, this is during IETF/VON/Infocom week.

The Spring Internet2 meeting is being held 
in Washington DC on March 28 and 29. One of the tracks is a digital
video 
track, and one of the sessions that I chair on Tuesday, March 28 is the 
Digital Video Conferencing Subcommittee. We are very interested in
learning 
more about SIP and more importantly the possible potential future impact
of 
SIP on H.323 video conferencing. If you know of anyone who might be 
interested in making a presentation on this subject, please have them 
contact me (630 252 5432 or amiot@anl.gov). Thanks.....Larry

*****************************************
Larry Amiot                             *
Committee on Institutional Cooperation  *
Visiting Senior Research Scientist      *
With an Office at:                      *
Argonne National Laboratory             *
Office Telephone- 630 252 5432          *
Email- amiot@anl.gov                    *
*****************************************


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



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb 29 19:27:49 2000
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Since this question seems to come up a lot, I've added it to the SIP FAQ
at http://www.cs.columbia.edu/sip, with additional details.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



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Hi Shai/Burcak,

> Bryan, Burcak
>
> Let me make a few assumptions:
>
> 1. We are talking about Cable modems and PacketCable DQoS (DOCSIS 1.1)

> 2. We are looking per-call QoS
> 3. "Assured" means Resources are allocated before a ring (a ring
requires
>   QoS), and resources are committed after the other side picked up
>   (call commences).

Shai, assumptions (2) and (3) are good.  However, (1) is off the mark.
I'm actually tryng to understand QoS with vanilla SIP.
Thanks for the responses guys!

Along those lines, here's a follow up question:

Assuming an objective of assured QoS between two SIP gateways,
I can think of 2 different ways to trigger the messages.
(If anyone can anyone think of more, I'd appreciate a call flow sketch)

I'm trying to understand the tradeoffs between the two
triggering methods and would be interested on preferences
anyone has.  I'm especially interested in if the 183 implies
one flow over the other and is the ResvConf reliable.

-------------------
SIP/RSVP message flow 1:
-------------------

      SIP               SIP               SIP
      GW                Proxy             GW
       |                 |                 |
       |--[1]INVITE----->|                 |
       |                 |                 |
       |<-[2]100---------|                 |
       |                 |--[3]INVITE----->|  1 ott
       |                 |                 |
       |                 |<-[4]100---------|
       |                 |                 |
 2 ott |<------[5]Path---------------------|
       |                 |                 |
       |-------[6]Resv-------------------->|  3 ott
       |                 |                 |
 4 ott |<-[8]183---------|<-[7]183---------|
       |                 |                 |
       |-------[9]Path-------------------->|  5 ott
       |                 |                 |
 6 ott |<------[10]Resv--------------------|
       |                 |                 |
       |--[11]PRACK(183)>|--[12]PRACK(183)>|  7 ott
       |                 |              ringing
       |                 |                 |
 8 ott |<---[14]180------|<-[13]180--------|
    ringback             |                 |
       |                 |                 |
       |                 |              off hook
       |<---[16]200------|<-[15]200--------|
       |                 |                 |
       |----[17]ACK----->|--[18]ACK------->|
       |                 |                 |
       |<===============RTP===============>|
       |                 |                 |


Event   Triggered by:
[1]     dialed digits
[2]     [1]  optional
[3]     [1]
[4]     [3]  optional
[5]     [3]
[6]     [5]
[7]     [6]
[8]     [7]
[9]     [8]
[10]    [9]
[11]    [10]
[12]    [11]
[13]    [12]
[14]    [13]
[15]    off hook
[16]    [15]
[17]    [16]
[18]    [17]

This triggering method incurs 8 one-way trip times (ott's) until
ringback.



-------------------
SIP/RSVP message flow 2:
-------------------

      SIP               SIP               SIP
      GW                Proxy             GW
       |                 |                 |
       |--[1]INVITE----->|                 |
       |                 |                 |
       |<-[2]100---------|                 |
       |                 |--[3]INVITE----->|  1 ott  --+
       |                 |                 |           |
       |                 |<-[4]100---------|  <--------+
       |                 |                 |           |
 2 ott |<-[6]183---------|<-[5]183---------|  <--------+
       |                 |                 |           |
       |-------[7]Path-------------------->|  3 ott    |
       |                 |                 |           |
 4 ott |<------[8]Resv---------------------|           |
       |                 |                 |           |
       |-------[9]ResvConfirm------------->|  5 ott    |
       |                 |                 |           |
 2 ott |<------[10]Path--------------------|   <-------+
       |                 |                 |
       |-------[11]Resv------------------->|  3 ott
       |                 |              ringing
 6 ott |<-[13]180--------|<-[12]180--------|  triggered by [9] & [11]
     ringback            |                 |
       |                 |                 |
       |                 |              offhook
       |<-[15]200--------|<-[14]200--------|
       |                 |                 |
       |--[16]ACK------->|--[17]ACK------->|
       |                 |                 |
       |<===============RTP===============>|
       |                 |                 |


Event   Triggered by:
[1]     dialed digits
[2]     [1]  optional
[3]     [1]
[4]     [3]  optional
[5]     [3]
[6]     [5]
[7]     [6]
[8]     [7]
[9]     [8]
[10]    [3]
[11]    [10]
[12]    [9] & [11]
[13]    [12]
[14]    offhook
[15]    [14]
[16]    [15]
[17]    [16]

This triggering method incurs 6 one-way trip times (ott's) until
ringback.


Discussion & Questions:

So, flow 1 is sequential; whereas flow 2 has some parallelism.
Both flows are providing "assured QoS".

BTW, my guess is someone will complain about the added complexity
of having a SIP proxy in the middle.  Feel free to mentally edit
it out.  (For now, my (rather simplistic) assumption is to ignore its
affect on ott times).

I recall reading in the RSVP RFC (RFC 2205, section 2.6, bullet 2) that:

      o    The receipt of a ResvConf gives no guarantees.  Assume the
           first two reservation requests from receivers R1 and R2
           arrive at the node where they are merged.  R2, whose
           reservation was the second to arrive at that node, may
           receive a ResvConf from that node while R1's request has not
           yet propagated all the way to a matching sender and may still

           fail.  Thus, R2 may receive a ResvConf although there is no
           end-to-end reservation in place; furthermore, R2 may receive
           a ResvConf followed by a ResvErr.

1) Does this mean that the ResvConf is not reliable and should not be
   depended upon (as shown in SIP/RSVP message flow 2)?

2) It seems like ResvConf may only be unreliable if merging of RSVP
    messages occurs.
    a) In reality, does anyone actually merge RSVP messages?
    b) Are there any other cases to consider when ResvConf is
unreliable?

2) Can anyone do better than 6 ott until ringback?

3) Does the 183 have any inherent meaning such as:
    Flow of the 183 message implies the sender has already
    established one-way QoS? (as shown in SIP/RSVP message flow 1)?
    or conversely, 183 implies that the sender has not yet
    establised one-way QoS?


Thanks for your help!

Bryan


Bryan J. Byerly
byerly@cisco.com
(919) 392-7091









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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Hi Shai/Burcak,
<p>> Bryan, Burcak
<br>>
<br>>&nbsp;Let me make a few assumptions:
<br>>
<br>> 1. We are talking about Cable modems and PacketCable DQoS (DOCSIS
1.1)
<br>> 2. We are looking per-call QoS
<br>> 3. "Assured" means Resources are allocated before a ring (a ring
requires
<br>>&nbsp;&nbsp; QoS), and resources are committed after the other side
picked up
<br>>&nbsp;&nbsp; (call commences).
<p>Shai, assumptions (2) and (3) are good.&nbsp; However, (1) is off the
mark.
<br>I'm actually tryng to understand QoS&nbsp;with vanilla SIP.
<br>Thanks for the responses guys!
<p>Along those lines, here's a follow up question:
<p>Assuming an objective of assured QoS&nbsp;between two SIP&nbsp;gateways,
<br>I can think of 2 different ways to trigger the messages.
<br>(If anyone can anyone think of more, I'd appreciate a call flow sketch)
<p>I'm trying to understand the tradeoffs between the two
<br>triggering methods and would be interested on preferences
<br>anyone has.&nbsp; I'm especially interested in if the 183 implies
<br>one flow over the other and is the ResvConf reliable.
<p>-------------------
<br>SIP/RSVP&nbsp;message flow 1:
<br>-------------------
<p><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; SIP&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
SIP&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
SIP</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; GW&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Proxy&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
GW</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |--[1]INVITE----->|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&lt;-[2]100---------|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|--[3]INVITE----->|&nbsp; 1 ott</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|&lt;-[4]100---------|</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;2 ott |&lt;------[5]Path---------------------|</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |-------[6]Resv-------------------->|&nbsp;
3 ott</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;4 ott |&lt;-[8]183---------|&lt;-[7]183---------|</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |-------[9]Path-------------------->|&nbsp;
5 ott</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;6 ott |&lt;------[10]Resv--------------------|</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |--[11]PRACK(183)>|--[12]PRACK(183)>|&nbsp;
7 ott</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;
ringing</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;8 ott |&lt;---[14]180------|&lt;-[13]180--------|</tt>
<br><tt>&nbsp;&nbsp;&nbsp; ringback&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;
|</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;
off hook</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&lt;---[16]200------|&lt;-[15]200--------|</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |----[17]ACK----->|--[18]ACK------->|</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&lt;===============RTP===============>|</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt></tt>&nbsp;<tt></tt>
<p><tt>Event&nbsp;&nbsp; Triggered by:</tt>
<br><tt>[1]&nbsp;&nbsp;&nbsp;&nbsp; dialed digits</tt>
<br><tt>[2]&nbsp;&nbsp;&nbsp;&nbsp; [1]&nbsp; optional</tt>
<br><tt>[3]&nbsp;&nbsp;&nbsp;&nbsp; [1]</tt>
<br><tt>[4]&nbsp;&nbsp;&nbsp;&nbsp; [3]&nbsp; optional</tt>
<br><tt>[5]&nbsp;&nbsp;&nbsp;&nbsp; [3]</tt>
<br><tt>[6]&nbsp;&nbsp;&nbsp;&nbsp; [5]</tt>
<br><tt>[7]&nbsp;&nbsp;&nbsp;&nbsp; [6]</tt>
<br><tt>[8]&nbsp;&nbsp;&nbsp;&nbsp; [7]</tt>
<br><tt>[9]&nbsp;&nbsp;&nbsp;&nbsp; [8]</tt>
<br><tt>[10]&nbsp;&nbsp;&nbsp; [9]</tt>
<br><tt>[11]&nbsp;&nbsp;&nbsp; [10]</tt>
<br><tt>[12]&nbsp;&nbsp;&nbsp; [11]</tt>
<br><tt>[13]&nbsp;&nbsp;&nbsp; [12]</tt>
<br><tt>[14]&nbsp;&nbsp;&nbsp; [13]</tt>
<br><tt>[15]&nbsp;&nbsp;&nbsp; off hook</tt>
<br><tt>[16]&nbsp;&nbsp;&nbsp; [15]</tt>
<br><tt>[17]&nbsp;&nbsp;&nbsp; [16]</tt>
<br><tt>[18]&nbsp;&nbsp;&nbsp; [17]</tt>
<p>This triggering method incurs 8 one-way trip times (ott's) until ringback.
<br>&nbsp;
<br>&nbsp;
<p>-------------------
<br>SIP/RSVP&nbsp;message flow 2:
<br>-------------------
<p><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; SIP&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
SIP&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
SIP</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; GW&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Proxy&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
GW</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |--[1]INVITE----->|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&lt;-[2]100---------|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|--[3]INVITE----->|&nbsp; 1 ott&nbsp; --+</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|&lt;-[4]100---------|&nbsp; &lt;--------+</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |</tt>
<br><tt>&nbsp;2 ott |&lt;-[6]183---------|&lt;-[5]183---------|&nbsp; &lt;--------+</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |-------[7]Path-------------------->|&nbsp;
3 ott&nbsp;&nbsp;&nbsp; |</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |</tt>
<br><tt>&nbsp;4 ott |&lt;------[8]Resv---------------------|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |-------[9]ResvConfirm------------->|&nbsp;
5 ott&nbsp;&nbsp;&nbsp; |</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |</tt>
<br><tt>&nbsp;2 ott |&lt;------[10]Path--------------------|&nbsp;&nbsp;
&lt;-------+</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |-------[11]Resv------------------->|&nbsp;
3 ott</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;
ringing</tt>
<br><tt>&nbsp;6 ott |&lt;-[13]180--------|&lt;-[12]180--------|&nbsp; triggered
by [9] &amp; [11]</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp; ringback&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;
|</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;
offhook</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&lt;-[15]200--------|&lt;-[14]200--------|</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |--[16]ACK------->|--[17]ACK------->|</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&lt;===============RTP===============>|</tt>
<br><tt>&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;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|</tt>
<br><tt></tt>&nbsp;<tt></tt>
<p><tt>Event&nbsp;&nbsp; Triggered by:</tt>
<br><tt>[1]&nbsp;&nbsp;&nbsp;&nbsp; dialed digits</tt>
<br><tt>[2]&nbsp;&nbsp;&nbsp;&nbsp; [1]&nbsp; optional</tt>
<br><tt>[3]&nbsp;&nbsp;&nbsp;&nbsp; [1]</tt>
<br><tt>[4]&nbsp;&nbsp;&nbsp;&nbsp; [3]&nbsp; optional</tt>
<br><tt>[5]&nbsp;&nbsp;&nbsp;&nbsp; [3]</tt>
<br><tt>[6]&nbsp;&nbsp;&nbsp;&nbsp; [5]</tt>
<br><tt>[7]&nbsp;&nbsp;&nbsp;&nbsp; [6]</tt>
<br><tt>[8]&nbsp;&nbsp;&nbsp;&nbsp; [7]</tt>
<br><tt>[9]&nbsp;&nbsp;&nbsp;&nbsp; [8]</tt>
<br><tt>[10]&nbsp;&nbsp;&nbsp; [3]</tt>
<br><tt>[11]&nbsp;&nbsp;&nbsp; [10]</tt>
<br><tt>[12]&nbsp;&nbsp;&nbsp; [9] &amp; [11]</tt>
<br><tt>[13]&nbsp;&nbsp;&nbsp; [12]</tt>
<br><tt>[14]&nbsp;&nbsp;&nbsp; offhook</tt>
<br><tt>[15]&nbsp;&nbsp;&nbsp; [14]</tt>
<br><tt>[16]&nbsp;&nbsp;&nbsp; [15]</tt>
<br><tt>[17]&nbsp;&nbsp;&nbsp; [16]</tt>
<p>This triggering method incurs 6 one-way trip times (ott's) until ringback.
<br>&nbsp;
<p>Discussion &amp; Questions:
<p>So, flow 1 is sequential; whereas flow 2 has some parallelism.
<br>Both flows are providing "assured QoS".
<p>BTW, my guess is someone will complain about the added complexity
<br>of having a SIP&nbsp;proxy in the middle.&nbsp; Feel free to mentally
edit
<br>it out.&nbsp; (For now, my (rather simplistic)&nbsp;assumption is to
ignore its affect on ott times).
<p>I recall reading in the RSVP&nbsp;RFC&nbsp;(RFC 2205, section 2.6, bullet
2) that:
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; o&nbsp;&nbsp;&nbsp; The receipt of a
ResvConf gives no guarantees.&nbsp; Assume the
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; first
two reservation requests from receivers R1 and R2
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; arrive
at the node where they are merged.&nbsp; R2, whose
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; reservation
was the second to arrive at that node, may
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; receive
a ResvConf from that node while R1's request has not
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; yet propagated
all the way to a matching sender and may still
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; fail.&nbsp;
Thus, R2 may receive a ResvConf although there is no
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; end-to-end
reservation in place; furthermore, R2 may receive
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a ResvConf
followed by a ResvErr.
<p>1)&nbsp;Does this mean that the ResvConf is not reliable and should
not be
<br>&nbsp;&nbsp; depended upon (as shown in SIP/RSVP message flow 2)?
<p>2)&nbsp;It seems like ResvConf may only be unreliable if merging of
RSVP
<br>&nbsp;&nbsp;&nbsp; messages occurs.
<br>&nbsp;&nbsp;&nbsp; a)&nbsp;In reality, does anyone actually merge RSVP&nbsp;messages?
<br>&nbsp;&nbsp;&nbsp; b) Are there any other cases to consider when ResvConf
is unreliable?
<p>2)&nbsp;Can anyone do better than 6 ott until ringback?
<p>3)&nbsp;Does the 183 have any inherent meaning such as:
<br>&nbsp;&nbsp;&nbsp; Flow of the 183 message implies the sender has already
<br>&nbsp;&nbsp;&nbsp; established one-way QoS? (as shown in SIP/RSVP&nbsp;message
flow 1)?
<br>&nbsp;&nbsp;&nbsp; or conversely, 183 implies that the sender has not
yet
<br>&nbsp;&nbsp;&nbsp; establised one-way QoS?
<br>&nbsp;
<p>Thanks for your help!
<p>Bryan
<br>&nbsp;
<p>Bryan J. Byerly
<br>byerly@cisco.com
<br>(919) 392-7091
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;
<br>&nbsp;</html>

--------------9E29D836C60967107F9EE5B9--




From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb 29 20:23:45 2000
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From: William Marshall <wtm@research.att.com>
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Date: Tue, 29 Feb 2000 20:17:39 -0500 (EST)
Message-Id: <200003010117.UAA95800@fish-ha.research.att.com>
To: drwalker@ss8networks.com
Cc: schulzrinne@cs.columbia.edu, wkwok@telogy.com,
        sip@lists.research.bell-labs.com
Subject: Re: SIP support of caller ID
Sender: owner-sip@lists.research.bell-labs.com
Precedence: bulk

This is one of the problems that was identified by the DCS group.

See draft-dcsgroup-sip-privacy-00 for a discussion of this particular
issue, and draft-dcsgroup-sip-arch-00 for the overall system architecture.

This draft is currently being revised based on the discussions held
in Washington at the last IETF meeting, and will be re-issued within
the next two weeks for discussion at Adelaide.

Bill Marshall
wtm@research.att.com

drwalker@ss8networks.com wrote:
> 
> It sounds appealing.  But what can we do about a user who feels like 
> lying?  For example, I'll always answer a call from "Bill Gates" (or 
> will I??).  If the user is known to a proxy in the signalling path, it's
> possible for that proxy to insert or validate the display name and
> then, more importantly, to signal that it has done so.  This could,
> for example, use a new addr-param (;screenedby=servername) in the From,
> or a new header (Screened-Display: displayname servername).  Of course, 
> this assumes that servers are more trustworthy than users.  If someone's 
> server sends me too many fraudulent displaynames, maybe I'll just stop 
> accepting calls from it.
> 
> Dave.
> 
> > -----Original Message-----
> > From: owner-sip@lists.research.bell-labs.com
> > [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of Henning
> > Schulzrinne
> > Sent: Tuesday, February 29, 2000 12:31 PM
> > To: Wing Man Kwok
> > Cc: 'sip@lists.research.bell-labs.com'
> > Subject: Re: SIP support of caller ID
> > 
> > 
> > Wing Man Kwok wrote:
> > > 
> > > All,
> > > 
> > > I read from the SIP FAQ that SIP supports caller ID.  But I 
> > cannot find any
> > > description of this in rfc2543.  How does SIP support caller ID?
> > 
> > Depends on the details, but basically, simply stick the caller name in
> > the display name field, and the number in the user name. Done.
> > 
> > > 
> > > Thanks in advance
> > > 
> > > Wing Man Kwok
> > > Telogy Networks
> > 
> > -- 
> > Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> > 
> > 
> 



From owner-sip-outgoing@lists.research.bell-labs.com  Tue Feb 29 22:11:59 2000
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To: Dean Willis <dean.willis@wcom.com>, Mart Nurmet <mart.nurmet@inet.com>,
        sip@lists.research.bell-labs.com
Subject: RE: Information request!  (Billing Package?)
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Interesting, I assumed everything EXCEPT a gateway throws CDRs.
Assuming a decomposed gateway, the GW feeds statistics to a Call Agent
(which would be a SIP UA among other things) that throws CDRs.

The billing system collates all the CDRs.

The spec required is, of course, the format of the CDR and the way they
are transported.  You could claim that a "transaction log" is a form of
CDR I suppose.

I'd certainly agree that SIP itself doesn't have any "billing package".
A UA and a server might, but that is independent of the protocol.  Billing
might very well impact some things in SIP.  

Brian

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@wcom.com]
> Sent: Tuesday, February 29, 2000 2:41 PM
> To: Mart Nurmet; sip@lists.research.bell-labs.com
> Subject: RE: Information request! (Billing Package?)
> 
> 
> What would a billing package look like? I've never really 
> understood the
> concept, and it might be useful to talk about it some more.
> 
> Here's how some implementors are doing it:
> 
> 1) Gateways throw CDRs using RADIUS or something similar. 
> This feeds into
> rating, reporting, and billing systems,
> 
> 2) SIP servers record transaction logs. These may feed into rating and
> billing systems, and probably feed into reporting systems.
> 
> 3) QoS PEP and PDPs log resource usages. These feed upstream as well.
> 
> 4) Other systems, like profile managers, report usage.
> 
> 5) Rating, reporting, and billing systems sift thru the vast 
> amounts of
> information coming at them and make useful things happen.
> 
> All of these things use other protocols and systems -- they aren't SIP
> specific. They do rely on SIP functions like authentication 
> -- that is, the
> goal of achieving billing functions has driven some 
> requirements into the
> design of SIP. However, other than driving some requirements 
> to SIP, such
> efforts are generally considered to be outside the scope of 
> the SIP WG.
> 
> Are you perhaps proposing to add functionality to SIP to 
> simplify or enhance
> reporting capabilities or to simplify the task of driving 
> downstream billing
> systems?
> 
> --
> Dena
> > -----Original Message-----
> > From: owner-sip@lists.research.bell-labs.com
> > [mailto:owner-sip@lists.research.bell-labs.com]On Behalf Of 
> Mart Nurmet
> > Sent: Monday, February 28, 2000 2:24 PM
> > To: sip@lists.research.bell-labs.com
> > Subject: Information request! (Billing Package?)
> >
> >
> > I am looking for information if there has been any work 
> done to generate a
> > billing package for SIP.  From my look at the available
> > information there does
> > not seem to be any.  Is that the case or am I missing something?
> >
> > Thank you,
> > -Mart
> >
> >
> 
> 



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Shail Bhatnagar wrote:
> 
> Jonathan, Thanks for your response. I will not argue for strcmp of name-addr
> but I will say that why would a SIP ua or proxy alter the name-addr portion of
> From/To. The name-addr in From/To is supposed to remain unchanged for the call.
> It makes sense for any implementation of SIP to simply keep the name-addr of
> From/To as it came in the first time instead of keeping it in parsed form
> and generating the unparsed form from it.

Proxies and uas will likely need to parse these in order to provide
features. For example, call screening will require a server to parse the
>From field. I suspect most proxies will parse, at least, To, From,
Call-ID, CSeq and Via. Once parsed, you don't want to also have to
remember the original unparsed version. When you reconstruct the URI,
case or something like this might not be the same.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com





