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        dwillis@dynamicsoft.com, sip@lists.bell-labs.com
Subject: Re: [SIP] Basic questions
References: <000b01bfca41$99f4b760$6701a8c0@plong> <39343C9C.F79563C0@dynamicsoft.com> <019501bfcac3$f8e00e50$0401a8c0@mds.local> <3934A96D.C353848A@dynamicsoft.com> <3934CA1E.5F3F4B3A@ubiquity.net>
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This is just my error; I made the statement without checking. So zero
lines is fine as far as SDP is concerned.

Even within SIP, I had forgotten about this tidbit in rfc2543:

> B.4 Delayed Media Streams
> 
>    In some cases, a caller may not know the set of media formats which
>    it can support at the time it would like to issue an invitation. This
>    is the case when the caller is actually a gateway to another protocol
>    which performs media format negotiation after call setup. When this
>    occurs, a caller MAY issue an INVITE with a session description that
>    contains no media lines. 

See, thats what happens when you always post at 3 in the morning ;)

-Jonathan R.

James Undery wrote:
> 
> Jonathan Rosenberg wrote:
> 
> > Sorry; I meant at least a single m line (i.e., zero m lines not
> > allowed).
> >
> 
> 4th paragraph of section 6 of RFC 2327 starts "An announcement consists of a
> session-level section followed by zero or more media-level sections". While this
> may be useful in a generic case the lack of media means the it is of no use to
> SIP. I would suggest adding a line along the lines of, SDP used in SIP must
> contain one or more media-level descriptions, to the bis draft. (Obviously
> reworded to avoid my awkward prose ;^) )
> 
> James Undery

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Thu Jun  1 00:32:42 2000
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Subject: Re: [SIP] SIP Application Server
References: <NDBBJGPOALAMPHONHENFEEMODBAA.scott@broadsoft.com> <3933EEB2.893C06AB@ubiquity.net> <39343EDC.2B4005FD@dynamicsoft.com> <3934CA2C.F4538AD5@ubiquity.net>
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Neil Deason wrote:
> 
> Jonathan I agree that a service logic enhanced Proxy Server is an application server.
> 
> However, I am not yet convinced this sits in complete harmony with the ISC App WG's
> intention to separate the call signalling (Proxy/Softswitch) and service control (App
> Server) and use standard RFC2543 SIP as the interface between them. In the case of
> the service logic enhanced proxy isn't the interface between vanilla proxy and the
> service control defined by CPL/SIP-CGI/Servlet API etc not RFC2543 SIP.

No; the "vanilla proxy" speaks SIP to the application server, which is
just a smarter proxy (or redirect server or whatever). This app server,
internally, may use any number of APIs, including the ones you mention.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
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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"Schroeder, Tim" wrote:
> 
> James Undery writes:
> I think what is being hinted at albeit subtlety is, is the transaction ever
> going to end? (The Halting Problem.) Although I can't think of a case off
> hand where you'd want to provisionally respond to a PRACK, specifically
> allowing it is just asking for trouble between legitimate implementations.
> [...] Except it may spawn another PRACK that spawns another PRACK ad
> infinitum.

I'd still argue that this just means you have a buggy server. Reliable
provisionals are not sent "just for fun" - it is usually for a reason.
So, the case I was considering was where the first PRACK had some
special data in a provisional response that needs to be sent reliably.
This data would itself be acknowledged with a PRACK, but there is no
reason for the second PRACK to have a reliable provisional response.

That said, if no one has any applications for this, and people think
that folks will just get into trouble with it, I'll remove that.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
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: [SIP] comments on draft-ietf-sip-100rel-01.txt
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Thanks for your comments. Responses below:


"Schroeder, Tim" wrote:
> 
> Section 3, Paragraph 1: "Note that a UAS MUST NOT send a response reliably
> unless there was a Supported header in the request indicating support for
> this extension."
> 
> This should be "... unless there was a Supported or Require header in the
> request indicating support for this extension."

Fixed.

> 
> -----
> 
> Section 3, Paragraph 3: "The UAC can determine that the response is to be
> transmitted reliably by the presence of the RSeq header."
> 
> I believe this should read "... by the presence of a Require header
> containing the option tag 100rel."  The use of the Require header in the
> response is mandated (correctly, in my opinion) by Section 5.2, Paragraph 6
> (UAS behavior).  This also follows draft-ietf-sip-serverfeatures-02.txt
> which "allows for the Require header to appear in responses. It is used to
> indicate what options must be understood by the UAC in order to process the
> response."
> In general, I think it's best to explicitly state what extensions/features
> are being used, rather than having the user agent deduce them from the
> presence/absence of headers.  This also will allow the Rseq header to be
> reused in other extensions where it might make sense, rather than having to
> create another extension-specific header.

Good point. Fixed.

> 
> -----
> 
> Section 5.1, Paragraph 2: "The rest of this discussion assumes this header
> has been inserted into a request."
> 
> This should read "... one of these headers has been inserted ...".  The
> previous paragraph was improved to allow for the UAC to insert either the
> Supported header or the Require header, and this sentence needs to follow
> suit.

Fixed.

> 
> -----
> 
> Section 5.1, Paragraph 11: "It is not necessary to include the Supported
> header listing the option tag 100rel in the PRACK request."
> 
> I disagree.  Given that the UAS may try to send reliable provisional
> responses to a PRACK (it's mentioned, and seems reasonable for completeness,
> though I don't see an obvious use of this), I think the UAC ought to say
> whether it supports or requires the 100rel feature.  Yes, the UAS can deduce
> that the feature is supported by the presence of the PRACK method, but (1)
> the UAS cannot tell from the existence of the PRACK method whether the UAC
> *requires* or merely *supports* the 100rel feature, and (2) the PRACK method
> could never be used in support of any other feature.  (2) might not be much
> of a problem, but I think either a Require or Supported header is needed to
> address (1).  Mandating the header would also make PRACK more consistent
> with the other methods, all of which mandate either the Require or Supported
> header.

Given the change that you can't send a reliable provisional response to
PRACK, it seems this text is actually fine - its not needed to include
this Supported header in PRACK.

> 
> -----
> 
> Section 5.2, Paragraph 1: "The UAS may send any provisional response
> reliably, so long as the initial request contained a Supported header
> indicating that this feature is understood."
> 
> This should read "... the initial request contained either a Supported or
> Require header indicating that this feature is understood."

Actually, not quite, since the strengths are different. It now reads:

The UAS \MAY send any provisional response reliably, so long as the
initial request contained a \header{Supported} header indicating
that this feature is understood. The UAS \MUST send any non-100
response reliably if the initial request contained a \header{Require}
header indicating that the feature is mandatory for all provisional
responses. If the UAS is unwilling to do so, it \MUST reject the
initial request with a 420 (Bad Extension) and include a
\header{Unsupported} header containing the option tag {\sf 100rel}. 


> 
> -----
> 
> Lastly (and maybe I've missed my chance on this one, since the "open issue"
> section went away), did we ever come up with an application where the
> cumulative response mechanism was needed?  I thought we had come to
> consensus to drop it unless there was a demonstrated need (which I haven't
> seen, though I may have just missed it).

At the meeting, I thought we had decided to keep it, but there wasn't a
whole lot of comment, and I am willing to entertain arguments for
removing it. I can think of applications where its useful (using
provisionals for call queue status, and having several changes all sent
in rapid succession). 

> 
> The UAC must PRACK a provisional response, which seems to disallow
> accumulating responses in the hopes of sending one PRACK for the group.

That is a good point, and we would need to change that to SHOULD if this
mechanism stays.

> Only if a provisional response arrives which is known to be out-of-sequence,
> and only if it is then cached by the UAC, and only if the previous missing
> response then arrives, does it seem that the cumulative response mechanism
> could come into play.  The examples section uses "two provisional responses
> in rapid succession" to address this, but I'm not sure that's sufficient.
> When the first response arrives, I read section 5.1 (UAC behavior) to
> require that the UAC construct and send a PRACK; I'm not sure waiting around
> to see if another response arrives before doing so is reasonable -- it would
> delay the acknowledgement of the first response in the usually vain hope
> that another response would quickly arrive. 

THis is no different than data buffering in any stream oriented protocol
- you wait for so long and then send.

 Maybe if the two responses were
> sent together in a single UDP packet, the cumulative response would be
> justified?
> 
> Also, the last paragraph of section 3 and the penultimate paragraph of
> section 5.2 both recommend that a UAS wait for acknowledgement of a
> provisional response before sending the next one.  Any UAS following this
> recommended procedure would certainly never need the cumulative response
> mechanism.

The mechanism is for the benefit of the UAC, in the case where this
recommendation is not followed.

> 
> I'd take out all the verbiage dealing with the cumulative responses, in the
> interests of simplicity and clarity. 

Any other opinions on this?

> There'd no longer need to be any
> difference in handling between the first reliable response (which cannot be
> part of a cumulative group because of the need to agree on the first Rseq)
> and any further reliable responses.  

I think that is a different issue.


-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Jonathan Rosenberg wrote:

> "Schroeder, Tim" wrote:
> >
> > James Undery writes:
> > I think what is being hinted at albeit subtlety is, is the transaction ever
> > going to end? (The Halting Problem.) Although I can't think of a case off
> > hand where you'd want to provisionally respond to a PRACK, specifically
> > allowing it is just asking for trouble between legitimate implementations.
> > [...] Except it may spawn another PRACK that spawns another PRACK ad
> > infinitum.
>
> I'd still argue that this just means you have a buggy server. Reliable
> provisionals are not sent "just for fun" - it is usually for a reason.
> So, the case I was considering was where the first PRACK had some
> special data in a provisional response that needs to be sent reliably.
> This data would itself be acknowledged with a PRACK, but there is no
> reason for the second PRACK to have a reliable provisional response.

My argument is the two entities aren't buggy because, you may decide the second
PRACK can't contain data that requires a reliable provisional response, but how
do you know some application won't. Personally I can't see the need for
provisional responses to a PRACK except 100 (which can't be sent reliably anyway)
and you know the final response will get there (or else you've bigger problems
anyway).

>
>
> That said, if no one has any applications for this, and people think
> that folks will just get into trouble with it, I'll remove that.
>

James Undery



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I agree with Rick here :) 

The quotation marks (") have been removed per the discussion, the 
use of the term "reserved" has been changed to un-reserved and 
the use of escaped characters has been clarified because of this.

However, i also thought the "," should have been removed and the 
"[" and "]" added.  I think there was just an error in transcribing all
the information into the new bis.

can we get final clarity on this matter?

thanx 

gethin

On Wed, 31 May 2000, Rick Workman wrote:
> 
> Further to Gethin's point, an earlier discussion on the unreserved characters
> for paramaters and headers doesn't seem to have been accuarately captured in the
> May bis. I think the two key requirements for these character sets are 1) they
> should be subsets of the URL reserved character set (ref RFC 2396/2732), and 2)
> they should avoid any characters which would cause ambiguities in the SIP
> grammar.
> 
> 1. It was discussed that "," should not be included in param-unreserved (and by
> extension with hnv-unreserved) "because URLs are allowed in certain headers and
> allowing commas in SIP URLs will make parsing those headers sometime
> impossible.". I thought this point was accepted, but maybe I missed something.
> 
> 2. RFC 2732 adds "[" and "]" to the URL reserved set to support IPv6 address
> syntax. As a consequence, I think these should also be added to param-unreserved
> and hnv-unreserved, since I don't think they cause any ambiguity issues.
> 
> 

-- 
Gethin Liddell
Ubiquity Software Corporation

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


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From: Gethin Liddell <gethin@ubiquity.net>
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@lists.bell-labs.com
Subject: Re: [SIP] Case sensitivity
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Hi people.

I hate to go over this topic time after time and appologize in advance for
bringing it up again but I have to query the case sensitivity issue of the
"user" element in the Sip URL.

This message is in fact in response to Henning's posting one month ago 
(to the day!!) and again i appologise, this time for the lateness of this posting.

My query arrises because of the examples given in the bis Section 2 after table 2.

we have the example 

sip:alice%40example.com@gateway.com

now this will be used, i suspect, by the gateway to convert to alice@example.com within
the gateway.com domain.  So the example.com has now become a host, which is 
case-insensitive (CI).  however the user is compared case-sensitive (CS).

so:

sip:alice%40EXAMPLE.com@gateway.com 

will not equate to the former even though the gateway.com considers both to be the same.

is this a problem?
should the user not be CI as well as the host? 
then what if alice has a password?  

sip:alice%3Apassword%40example.com@gateway.com

This would have to be CI as well.  But then passwords
are more often than not CS!!!

i fear i may be over-concerned about this and worrying about nothing. 
i.e. what the gateway.com does is out of the scope of SIP and not our concern

any thoughts?

On Mon, 01 May 2000, Henning Schulzrinne wrote:
> I believe we need to take a closer look at the annoying topic of
> case-sensitivity (or not).
> 
> A rough summary is:
> 
> Method              CS
> Header field name   CI
> Any Accept-* token  CI  (includes compression, language, media type)
> SDP encoding name   CI  (such as PCMU, L16, etc.)
> URL                 mostly CS (except host)
> 
> Some additional issues:
> (1) Parameters like ;level or ;q in Accept. Are ;LEVEL and ;Q the same
> thing? RFC 2616 seems to indicate this:
> 
> 3.7:
> 
> The type, subtype, and parameter attribute names are case-
>    insensitive. Parameter values might or might not be case-sensitive,
>    depending on the semantics of the parameter name.
> 
> 2.1 Augmented BNF
> 
> "literal"
>       Quotation marks surround literal text. Unless stated otherwise,
>       the text is case-insensitive.
> 
> [This would apply to most enumerated parameters, like "hop" or "hide"
> tokens in our case, or the "action" tokens in Contact or
> Content-Disposition or the Priority values.]
> 
> My general impression is that HTTP/1.x seems to treat tokens not
> enclosed in quotation marks as CI and anything in quotation marks as CS,
> with URLs as a separate matter.
> 
> (2) HTTP-date (and, thus, rfc1123-date) is case sensitive.
> 
> Any other cases?
> -- 
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
-- 
Gethin Liddell
Ubiquity Software Corporation

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


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From sip-admin@lists.bell-labs.com  Thu Jun  1 06:26:17 2000
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Date: Thu, 01 Jun 2000 11:23:13 +0100
From: Neil Deason <ndeason@ubiquity.net>
Organization: Ubiquity Software Corporation Limited
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Subject: Re: [SIP] SIP Application Server
References: <NDBBJGPOALAMPHONHENFEEMODBAA.scott@broadsoft.com> <3933EEB2.893C06AB@ubiquity.net> <39343EDC.2B4005FD@dynamicsoft.com> <3934CA2C.F4538AD5@ubiquity.net> <3935E680.B7598DA4@dynamicsoft.com>
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Jonathan Rosenberg wrote:

> Neil Deason wrote:
> >
> > Jonathan I agree that a service logic enhanced Proxy Server is an application server.
> >
> > However, I am not yet convinced this sits in complete harmony with the ISC App WG's
> > intention to separate the call signalling (Proxy/Softswitch) and service control (App
> > Server) and use standard RFC2543 SIP as the interface between them. In the case of
> > the service logic enhanced proxy isn't the interface between vanilla proxy and the
> > service control defined by CPL/SIP-CGI/Servlet API etc not RFC2543 SIP.
>
> No; the "vanilla proxy" speaks SIP to the application server, which is
> just a smarter proxy (or redirect server or whatever). This app server,
> internally, may use any number of APIs, including the ones you mention.

Yes of course a separate SIP network element can talk SIP to a smart proxy/application
server. But the point I was trying to make is that the application server encapsulates a
vanilla proxy server, i.e. the proxy functionality as described by rfc2543, plus service
logic accessed through CPL/SIP-CGI/Servlet API/whatever. My difficulty is in seeing how
this fits with the ISC application working group premise of separating call signalling from
service control via a SIP messaging interface. This seems like an unnecessary IN model of
separation. In the App Server the proxy part does signalling but it is clearly not
separated from controlling service logic by a SIP messaging interface.

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




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From sip-admin@lists.bell-labs.com  Thu Jun  1 07:02:02 2000
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        Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@lists.bell-labs.com
Subject: Re: [SIP] Case sensitivity
References: <3908D34E.58792A22@dynamicsoft.com> <390D8F1F.EE1C1F4E@dynamicsoft.com> <390D9531.B81CA74D@cs.columbia.edu> <00060111060102.32012@gethin>
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I agree that we're now getting into one of the more arcane areas. The
reason that user names are CS is that they are CS in a number of
operating systems and that RFC 822 defines local-part (the local name)
to be CS. It is then up to the gateway to treat a user name like foo@bar
appropriately. The CI/CS only matters for comparing registrations, for
example, where this form is not likely to occur in any event.

Gethin Liddell wrote:
> 
> Hi people.
> 
> I hate to go over this topic time after time and appologize in advance for
> bringing it up again but I have to query the case sensitivity issue of the
> "user" element in the Sip URL.
> 
> This message is in fact in response to Henning's posting one month ago
> (to the day!!) and again i appologise, this time for the lateness of this posting.
> 
> My query arrises because of the examples given in the bis Section 2 after table 2.
> 
> we have the example
> 
> sip:alice%40example.com@gateway.com
> 
> now this will be used, i suspect, by the gateway to convert to alice@example.com within
> the gateway.com domain.  So the example.com has now become a host, which is
> case-insensitive (CI).  however the user is compared case-sensitive (CS).
> 
> so:
> 
> sip:alice%40EXAMPLE.com@gateway.com
> 
> will not equate to the former even though the gateway.com considers both to be the same.
> 
> is this a problem?
> should the user not be CI as well as the host?
> then what if alice has a password?
> 
> sip:alice%3Apassword%40example.com@gateway.com
> 
> This would have to be CI as well.  But then passwords
> are more often than not CS!!!
> 
> i fear i may be over-concerned about this and worrying about nothing.
> i.e. what the gateway.com does is out of the scope of SIP and not our concern
> 
> any thoughts?
> 
> On Mon, 01 May 2000, Henning Schulzrinne wrote:
> > I believe we need to take a closer look at the annoying topic of
> > case-sensitivity (or not).
> >
> > A rough summary is:
> >
> > Method              CS
> > Header field name   CI
> > Any Accept-* token  CI  (includes compression, language, media type)
> > SDP encoding name   CI  (such as PCMU, L16, etc.)
> > URL                 mostly CS (except host)
> >
> > Some additional issues:
> > (1) Parameters like ;level or ;q in Accept. Are ;LEVEL and ;Q the same
> > thing? RFC 2616 seems to indicate this:
> >
> > 3.7:
> >
> > The type, subtype, and parameter attribute names are case-
> >    insensitive. Parameter values might or might not be case-sensitive,
> >    depending on the semantics of the parameter name.
> >
> > 2.1 Augmented BNF
> >
> > "literal"
> >       Quotation marks surround literal text. Unless stated otherwise,
> >       the text is case-insensitive.
> >
> > [This would apply to most enumerated parameters, like "hop" or "hide"
> > tokens in our case, or the "action" tokens in Contact or
> > Content-Disposition or the Priority values.]
> >
> > My general impression is that HTTP/1.x seems to treat tokens not
> > enclosed in quotation marks as CI and anything in quotation marks as CS,
> > with URLs as a separate matter.
> >
> > (2) HTTP-date (and, thus, rfc1123-date) is case sensitive.
> >
> > Any other cases?
> > --
> > Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> >
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> --
> Gethin Liddell
> Ubiquity Software Corporation
> 
> http://www.ubiquity.net
> mailto:gethin@ubiquity.net
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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From sip-admin@lists.bell-labs.com  Thu Jun  1 09:12:25 2000
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Hi!

Section 10.4.1 states of the SIP draft states that a client using UDP
should retransmit  BYE CANCEL OPTION and REGISTER requests. Does this mean
proxy servers dont retransmit them. If so what happens in a scenario like
this:
     Client A connects to proxy P on TCP and sends a BYE. P connects to
client B on UDP. Since A does not retransmit the BYE how can reliablity be
achieved unless proxy P retransmits.

thanks
bhaskar




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From sip-admin@lists.bell-labs.com  Thu Jun  1 09:14:49 2000
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Date: Thu, 01 Jun 2000 09:14:43 -0400
From: Mahesh Shankar <mshanka@unity.ncsu.edu>
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Hi,

I was wondering if the CPL allows dynamic creation of services, or is
that capability not part of the CPL's functionality ?

Thanks.

-M

--
_________________________________________________________________________

              If opportunity doesn't knock, build a door
_________________________________________________________________________





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Subject: RE: [SIP] retransmission of BYE OPTIONS REGISTER request by proxies
Date: Thu, 1 Jun 2000 14:17:32 +0100
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> Section 10.4.1 states of the SIP draft states that a client using
> UDP should retransmit  BYE CANCEL OPTION and REGISTER requests. Does
> this mean proxy servers dont retransmit them.

No.  "client" is in the general sense; i.e., it is a SIP entity
that transmits requests.  (Have a look at the definitions under
section 1.3).


 - Jo.



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To: mshanka@unity.ncsu.edu
Subject: Re: [SIP] CPL
Cc: sip@lists.bell-labs.com
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> 
> Hi,
> 
> I was wondering if the CPL allows dynamic creation of services, or is
> that capability not part of the CPL's functionality ?
> 
> Thanks.
> 
> -M
> 

CPL is best thought of as a wire representation of service logic. It is probable
that most servers will "compile" CPL into a more appropriate internal format for
executing and storing. So to answer your question, I expect that there will be
demand for service creation environments that you can use to build services 
(even dynamically) and one possible output format will be CPL. For example, it
would be trivial to use CGI/ASP to dynamically generate CPL in response to an
HTTP request to retrieve the service logic for a particular user. The utility of
doing so is another question altogether ;)

my two cents
sean

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


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Sean Olson writes:
 > CPL is best thought of as a wire representation of service logic. It is probable
 > that most servers will "compile" CPL into a more appropriate internal format for
 > executing and storing. So to answer your question, I expect that there will be
 > demand for service creation environments that you can use to build services 
 > (even dynamically) and one possible output format will be CPL. For example, it
 > would be trivial to use CGI/ASP to dynamically generate CPL in response to an
 > HTTP request to retrieve the service logic for a particular user. The utility of
 > doing so is another question altogether ;)

   I'm going to show my complete ignorance here,
   but doesn't this beg the "sandbox" question?

   Related: how is authorization accomplished?

	    Mike


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From sip-admin@lists.bell-labs.com  Thu Jun  1 10:43:27 2000
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From: Linden deCarmo <ldeCarmo@netspeak.com>
To: "'Dean Willis'" <dwillis@dynamicsoft.com>,
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Subject: RE: [SIP] Suspend a session??
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Yes, we've considered this, BUT, in your scenario, how does A (or in
reality, the proxy) distinguish between a conventional hold and an on-hook
hold?

The reason the distinction is important is this:

If B goes onhook during the prescribed time period, A should be billed.
If B doesn't go onhook, A should be billed up to the onhook, but not during
the "suspend" period.
If B places A on hold, A should be billed for the time on hold.

Will your proposal satisify these requirements?  If so, could you explain
how?

Thanks.

Linden deCarmo
Netspeak Corporation
902 Clint Moore Road
Suite 104
Boca Raton, FL 33487

> -----Original Message-----
> From:	Dean Willis [SMTP:dwillis@dynamicsoft.com]
> Sent:	Wednesday, May 31, 2000 7:07 PM
> To:	Linden deCarmo
> Cc:	sip@lists.bell-labs.com
> Subject:	Re: [SIP] Suspend a session??
> 
> 
> Have you considered building an auto-hold on switch-hook with a timer
> that initaies a BYE into your UA? My first thought is that the suggested
> behavior is a UA problem, not a proxy problem. Unless it is a flaw in
> the billing model . . .
> 
> B calls A.
> B places receiver on-hook.
> B sends reINVITE-to-HOLD to A.
> B starts timer.
> Timer expires.
> B sends BYE to A.
> 
> The only issue I see here is that B might vanish during the hold-period
> and fail to send the BYE. This is a risk with call-hold in general, and
> I believe there has been a proposal to use a configurable hold-timer in
> end devices to recover from it.
> 
> --
> Dean
> 
> Linden deCarmo wrote:
> > 
> > Right now, SIP smoothly handles one party putting another party on hold
> via
> > a re-INVITE with modified SDP.  However, we have a scenario where the
> > session needs to be suspended (not just placed on hold), and I'm not
> sure
> > how this can be done without using proprietary headers.
> > 
> > Here are ther details.  If party A calls party B , party B should be
> able to
> > hang up without immediately terminating the session.  If party B picks
> up
> > the phone within some prescribed time window, the session continues.
> > However, if they don't pick it up within the time window, the session
> > terminates (via a BYE).
> > 
> > We're looking for a mechanism for party B to communicate with either
> party A
> > or a proxy server, to inform the distant party that the session has been
> > suspended, and not just placed on hold.  The primary motivator for this
> > notification is billing.  Without notification of suspension, its
> impossible
> > to determine when to safely stop billing.
> > 
> > Thanks.
> > 
> > Linden deCarmo
> > Netspeak Corporation
> > 902 Clint Moore Road
> > Suite 104
> > Boca Raton, FL 33487
> > 
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip


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Subject: Re: [SIP] CPL
Cc: mshanka@unity.ncsu.edu, sip@lists.bell-labs.com
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> 
> Sean Olson writes:
>  > CPL is best thought of as a wire representation of service logic. It is probable
>  > that most servers will "compile" CPL into a more appropriate internal format for
>  > executing and storing. So to answer your question, I expect that there will be
>  > demand for service creation environments that you can use to build services 
>  > (even dynamically) and one possible output format will be CPL. For example, it
>  > would be trivial to use CGI/ASP to dynamically generate CPL in response to an
>  > HTTP request to retrieve the service logic for a particular user. The utility of
>  > doing so is another question altogether ;)
> 
>    I'm going to show my complete ignorance here,
>    but doesn't this beg the "sandbox" question?
> 
>    Related: how is authorization accomplished?
> 
> 	    Mike
> 

If you really want to take a dive here, I would argue that you re-use everything
that HTTP/HTTPS provides in terms of security and authorization. Of course you
can always sign the CPL script as you would any other XML document for another
layer of security. 

Regarding the sandbox issue, yes, CPL was designed to avoid this by providing a
limited set of capabilities and a simple DAG, etc. etc. You don't lose this by
what I'm suggesting as long as you continue to use CPL as your on the wire format.
(the sandbox on the CPL service execution platform is preserved even if the
 CGI/ASP server is prone to infinite loops or whatever.) Personally, I see the
value in providing a sandbox, but I don't think CPL is the only way to do this.

sean



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From sip-admin@lists.bell-labs.com  Thu Jun  1 11:20:56 2000
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To: Neil Deason <ndeason@ubiquity.net>
Cc: Scott Hoffpauir <scott@broadsoft.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@lists.bell-labs.com
Subject: RE: [SIP] SIP Application Server
Date: Thu, 1 Jun 2000 11:20:40 -0400 
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Neil,

In the ISC model, the role of the SoftSwitch is to present a single
protocol/interface by which an AppServer can participate in servicing the
call and controlling various legs of a call.  While the communication
between the AppServer and the SoftSwitch is proposed to be SIP, the actual
endpoints can use various protocols including PSTN and IP based (and
possibly none of them are SIP endpoints). This separates the task of the
service logic from details of network and protocols, presenting a single SIP
interface between the two.

Considering just the interface between the AppServer and the SoftSwitch, the
AppServer can act as a redirect server for applications such as LNP and 800
translation.  It can act as a proxy server for applications such as calling
card. And it can participate as a UAS in applications such as voicemail.  As
Jonathan mentioned,  it is the complexity and functionality of the actual
application which determines the role of the AppServer in its communication
with the SoftSwitch.

Fardad Farahmand
ffarahmand@sonusnet.com


> -----Original Message-----
> From: Neil Deason [mailto:ndeason@ubiquity.net]
> Sent: Thursday, June 01, 2000 6:23 AM
> To: Jonathan Rosenberg
> Cc: Scott Hoffpauir; sip@lists.bell-labs.com
> Subject: Re: [SIP] SIP Application Server
> 
> 
> Jonathan Rosenberg wrote:
> 
> > Neil Deason wrote:
> > >
> > > Jonathan I agree that a service logic enhanced Proxy 
> Server is an application server.
> > >
> > > However, I am not yet convinced this sits in complete 
> harmony with the ISC App WG's
> > > intention to separate the call signalling 
> (Proxy/Softswitch) and service control (App
> > > Server) and use standard RFC2543 SIP as the interface 
> between them. In the case of
> > > the service logic enhanced proxy isn't the interface 
> between vanilla proxy and the
> > > service control defined by CPL/SIP-CGI/Servlet API etc 
> not RFC2543 SIP.
> >
> > No; the "vanilla proxy" speaks SIP to the application 
> server, which is
> > just a smarter proxy (or redirect server or whatever). This 
> app server,
> > internally, may use any number of APIs, including the ones 
> you mention.
> 
> Yes of course a separate SIP network element can talk SIP to 
> a smart proxy/application
> server. But the point I was trying to make is that the 
> application server encapsulates a
> vanilla proxy server, i.e. the proxy functionality as 
> described by rfc2543, plus service
> logic accessed through CPL/SIP-CGI/Servlet API/whatever. My 
> difficulty is in seeing how
> this fits with the ISC application working group premise of 
> separating call signalling from
> service control via a SIP messaging interface. This seems 
> like an unnecessary IN model of
> separation. In the App Server the proxy part does signalling 
> but it is clearly not
> separated from controlling service logic by a SIP messaging interface.
> 
> Neil.
> --
> Ubiquity Software Corporation, UK           http://www.ubiquity.net
> 
> 
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 


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It seems to me that this should be handled by the UA itself. Once the UA
determines the correct functionilty that the real end-device/user wishes to
achieve, it should signal proxies and far-end stations. For instance a
legacy device may which to indicate flash and collect more digits for a
3-way call by the user using the hook as you say. 

This would result in a separate transaction but I believe it doesn't require
something like a  "suspend" method or indication at the SIP level.  

I suggest that the UA itself should handle this with timers, etc.. based
upon the physical device and human interface it is trying to achieve.

For the billing issue the UA could simply use a timer started based on
physical off-hook stimulus to determine when to actually send a BYE or not.

If this doesn't address your concerns please let me know. 

> -----Original Message-----
> From:	Linden deCarmo [SMTP:ldeCarmo@netspeak.com]
> Sent:	Wednesday, May 31, 2000 1:54 PM
> To:	sip@lists.bell-labs.com
> Subject:	[SIP] Suspend a session??
> 
> Right now, SIP smoothly handles one party putting another party on hold
> via
> a re-INVITE with modified SDP.  However, we have a scenario where the
> session needs to be suspended (not just placed on hold), and I'm not sure
> how this can be done without using proprietary headers.
> 
> Here are ther details.  If party A calls party B , party B should be able
> to
> hang up without immediately terminating the session.  If party B picks up
> the phone within some prescribed time window, the session continues.
> However, if they don't pick it up within the time window, the session
> terminates (via a BYE).
> 
> We're looking for a mechanism for party B to communicate with either party
> A
> or a proxy server, to inform the distant party that the session has been
> suspended, and not just placed on hold.  The primary motivator for this
> notification is billing.  Without notification of suspension, its
> impossible
> to determine when to safely stop billing.
> 
> Thanks.
> 
> Linden deCarmo
> Netspeak Corporation
> 902 Clint Moore Road
> Suite 104
> Boca Raton, FL 33487
> 
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

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<TITLE>RE: [SIP] Suspend a session??</TITLE>
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<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">It seems to me that =
this should be handled by the UA itself. Once the UA determines the =
correct functionilty that the real end-device/user wishes to achieve, =
it should signal proxies and far-end stations. For instance a legacy =
device may which to indicate flash and collect more digits for a 3-way =
call by the user using the hook as you say. </FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">This would result in =
a separate transaction but I believe it doesn't require something like =
a&nbsp; &quot;suspend&quot; method or indication at the SIP =
level.&nbsp; </FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">I suggest that the =
UA itself should handle this with timers, etc.. based upon the physical =
device and human interface it is trying to achieve.</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">For the billing =
issue the UA could simply use a timer started based on physical =
off-hook stimulus to determine when to actually send a BYE or =
not.</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">If this doesn't =
address your concerns please let me know. </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">Linden deCarmo =
[SMTP:ldeCarmo@netspeak.com]</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Wednesday, May 31, 2000 1:54 PM</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">sip@lists.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">[SIP] Suspend a session??</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Right now, SIP smoothly handles one =
party putting another party on hold via</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">a re-INVITE with modified SDP.&nbsp; =
However, we have a scenario where the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">session needs to be suspended (not =
just placed on hold), and I'm not sure</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">how this can be done without using =
proprietary headers.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Here are ther details.&nbsp; If party =
A calls party B , party B should be able to</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">hang up without immediately =
terminating the session.&nbsp; If party B picks up</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">the phone within some prescribed time =
window, the session continues.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">However, if they don't pick it up =
within the time window, the session</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">terminates (via a BYE).</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">We're looking for a mechanism for =
party B to communicate with either party A</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">or a proxy server, to inform the =
distant party that the session has been</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">suspended, and not just placed on =
hold.&nbsp; The primary motivator for this</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">notification is billing.&nbsp; =
Without notification of suspension, its impossible</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">to determine when to safely stop =
billing.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Thanks.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Linden deCarmo</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Netspeak Corporation</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">902 Clint Moore Road</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Suite 104</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Boca Raton, FL 33487</FONT>
</P>
<BR>
<BR>

<P><FONT SIZE=3D2 =
FACE=3D"Arial">_______________________________________________</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">SIP mailing list</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">SIP@lists.bell-labs.com</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://lists.bell-labs.com/mailman/listinfo/sip" =
TARGET=3D"_blank">http://lists.bell-labs.com/mailman/listinfo/sip</A></F=
ONT></U>
</P>
</UL>
</BODY>
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From sip-admin@lists.bell-labs.com  Thu Jun  1 13:10:08 2000
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From: Linden deCarmo <ldeCarmo@netspeak.com>
To: "'Glenn Morrow'" <gmorrow@nortelnetworks.com>, sip@lists.bell-labs.com
Subject: RE: [SIP] Suspend a session??
Date: Thu, 1 Jun 2000 13:08:51 -0400 
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Here's the problem with the UA doing it:

User goes on hook (time 0 ).  UA starts timer.
Timer expires (time 30 seconds). UA realizes session expires so it issues a
BYE.

Since the proxy sees the BYE 30 seconds later, thats when it stops billing.
I do not believe your solution addresses how the UA signals to the proxy
that billing should stop at time 0 rather than time 30.  

I realize I can do this with a proprietary method or header, but I'd rather
not.

Linden deCarmo
Netspeak Corporation
902 Clint Moore Road
Suite 104
Boca Raton, FL 33487

> -----Original Message-----
> From:	Glenn Morrow [SMTP:gmorrow@nortelnetworks.com]
> Sent:	Thursday, June 01, 2000 11:38 AM
> To:	sip@lists.bell-labs.com
> Subject:	RE: [SIP] Suspend a session??
> 
> It seems to me that this should be handled by the UA itself. Once the UA
> determines the correct functionilty that the real end-device/user wishes
> to achieve, it should signal proxies and far-end stations. For instance a
> legacy device may which to indicate flash and collect more digits for a
> 3-way call by the user using the hook as you say. 
> 
> This would result in a separate transaction but I believe it doesn't
> require something like a  "suspend" method or indication at the SIP level.
> 
> 
> I suggest that the UA itself should handle this with timers, etc.. based
> upon the physical device and human interface it is trying to achieve.
> 
> For the billing issue the UA could simply use a timer started based on
> physical off-hook stimulus to determine when to actually send a BYE or
> not.
> 
> If this doesn't address your concerns please let me know. 
> 
> 	-----Original Message----- 
> From:   Linden deCarmo [SMTP:ldeCarmo@netspeak.com] 
> Sent:   Wednesday, May 31, 2000 1:54 PM 
> To:     sip@lists.bell-labs.com 
> Subject:        [SIP] Suspend a session?? 
> 
> 	Right now, SIP smoothly handles one party putting another party on
> hold via 
> a re-INVITE with modified SDP.  However, we have a scenario where the 
> session needs to be suspended (not just placed on hold), and I'm not sure 
> how this can be done without using proprietary headers. 
> 
> 	Here are ther details.  If party A calls party B , party B should be
> able to 
> hang up without immediately terminating the session.  If party B picks up 
> the phone within some prescribed time window, the session continues. 
> However, if they don't pick it up within the time window, the session 
> terminates (via a BYE). 
> 
> 	We're looking for a mechanism for party B to communicate with either
> party A 
> or a proxy server, to inform the distant party that the session has been 
> suspended, and not just placed on hold.  The primary motivator for this 
> notification is billing.  Without notification of suspension, its
> impossible 
> to determine when to safely stop billing. 
> 
> 	Thanks. 
> 
> 	Linden deCarmo 
> Netspeak Corporation 
> 902 Clint Moore Road 
> Suite 104 
> Boca Raton, FL 33487 
> 
> 
> 
> 	_______________________________________________ 
> SIP mailing list 
> SIP@lists.bell-labs.com 
> <http://lists.bell-labs.com/mailman/listinfo/sip> 
> 


_______________________________________________
SIP mailing list
SIP@lists.bell-labs.com
http://lists.bell-labs.com/mailman/listinfo/sip


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From: "Glenn Morrow" <gmorrow@nortelnetworks.com>
To: "'Linden deCarmo'" <ldeCarmo@netspeak.com>, sip@lists.bell-labs.com
Subject: RE: [SIP] Suspend a session??
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Perhaps we could timestamp the BYE?

> -----Original Message-----
> From:	Linden deCarmo [SMTP:ldeCarmo@netspeak.com]
> Sent:	Thursday, June 01, 2000 12:09 PM
> To:	Morrow, Glenn [RICH2:C301:EXCH]; sip@lists.bell-labs.com
> Subject:	RE: [SIP] Suspend a session??
> 
> Here's the problem with the UA doing it:
> 
> User goes on hook (time 0 ).  UA starts timer.
> Timer expires (time 30 seconds). UA realizes session expires so it issues
> a
> BYE.
> 
> Since the proxy sees the BYE 30 seconds later, thats when it stops
> billing.
> I do not believe your solution addresses how the UA signals to the proxy
> that billing should stop at time 0 rather than time 30.  
> 
> I realize I can do this with a proprietary method or header, but I'd
> rather
> not.
> 
> Linden deCarmo
> Netspeak Corporation
> 902 Clint Moore Road
> Suite 104
> Boca Raton, FL 33487
> 
> > -----Original Message-----
> > From:	Glenn Morrow [SMTP:gmorrow@nortelnetworks.com]
> > Sent:	Thursday, June 01, 2000 11:38 AM
> > To:	sip@lists.bell-labs.com
> > Subject:	RE: [SIP] Suspend a session??
> > 
> > It seems to me that this should be handled by the UA itself. Once the UA
> > determines the correct functionilty that the real end-device/user wishes
> > to achieve, it should signal proxies and far-end stations. For instance
> a
> > legacy device may which to indicate flash and collect more digits for a
> > 3-way call by the user using the hook as you say. 
> > 
> > This would result in a separate transaction but I believe it doesn't
> > require something like a  "suspend" method or indication at the SIP
> level.
> > 
> > 
> > I suggest that the UA itself should handle this with timers, etc.. based
> > upon the physical device and human interface it is trying to achieve.
> > 
> > For the billing issue the UA could simply use a timer started based on
> > physical off-hook stimulus to determine when to actually send a BYE or
> > not.
> > 
> > If this doesn't address your concerns please let me know. 
> > 
> > 	-----Original Message----- 
> > From:   Linden deCarmo [SMTP:ldeCarmo@netspeak.com] 
> > Sent:   Wednesday, May 31, 2000 1:54 PM 
> > To:     sip@lists.bell-labs.com 
> > Subject:        [SIP] Suspend a session?? 
> > 
> > 	Right now, SIP smoothly handles one party putting another party on
> > hold via 
> > a re-INVITE with modified SDP.  However, we have a scenario where the 
> > session needs to be suspended (not just placed on hold), and I'm not
> sure 
> > how this can be done without using proprietary headers. 
> > 
> > 	Here are ther details.  If party A calls party B , party B should be
> > able to 
> > hang up without immediately terminating the session.  If party B picks
> up 
> > the phone within some prescribed time window, the session continues. 
> > However, if they don't pick it up within the time window, the session 
> > terminates (via a BYE). 
> > 
> > 	We're looking for a mechanism for party B to communicate with either
> > party A 
> > or a proxy server, to inform the distant party that the session has been
> 
> > suspended, and not just placed on hold.  The primary motivator for this 
> > notification is billing.  Without notification of suspension, its
> > impossible 
> > to determine when to safely stop billing. 
> > 
> > 	Thanks. 
> > 
> > 	Linden deCarmo 
> > Netspeak Corporation 
> > 902 Clint Moore Road 
> > Suite 104 
> > Boca Raton, FL 33487 
> > 
> > 
> > 
> > 	_______________________________________________ 
> > SIP mailing list 
> > SIP@lists.bell-labs.com 
> > <http://lists.bell-labs.com/mailman/listinfo/sip> 
> > 
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

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<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Perhaps we could =
timestamp the BYE?</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">Linden deCarmo =
[SMTP:ldeCarmo@netspeak.com]</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Thursday, June 01, 2000 12:09 PM</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]; =
sip@lists.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: [SIP] Suspend a session??</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Here's the problem with the UA doing =
it:</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">User goes on hook (time 0 ).&nbsp; UA =
starts timer.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Timer expires (time 30 seconds). UA =
realizes session expires so it issues a</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">BYE.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Since the proxy sees the BYE 30 =
seconds later, thats when it stops billing.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">I do not believe your solution =
addresses how the UA signals to the proxy</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">that billing should stop at time 0 =
rather than time 30.&nbsp; </FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I realize I can do this with a =
proprietary method or header, but I'd rather</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">not.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Linden deCarmo</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Netspeak Corporation</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">902 Clint Moore Road</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Suite 104</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Boca Raton, FL 33487</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; From: Glenn Morrow =
[SMTP:gmorrow@nortelnetworks.com]</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Sent: Thursday, June 01, 2000 =
11:38 AM</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; To:&nbsp;&nbsp; =
sip@lists.bell-labs.com</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; =
Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RE: [SIP] Suspend a =
session??</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; It seems to me that this should =
be handled by the UA itself. Once the UA</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; determines the correct =
functionilty that the real end-device/user wishes</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; to achieve, it should signal =
proxies and far-end stations. For instance a</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; legacy device may which to =
indicate flash and collect more digits for a</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; 3-way call by the user using the =
hook as you say. </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; This would result in a separate =
transaction but I believe it doesn't</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; require something like a&nbsp; =
&quot;suspend&quot; method or indication at the SIP level.</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; I suggest that the UA itself =
should handle this with timers, etc.. based</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; upon the physical device and =
human interface it is trying to achieve.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; For the billing issue the UA =
could simply use a timer started based on</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; physical off-hook stimulus to =
determine when to actually send a BYE or</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; not.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; If this doesn't address your =
concerns please let me know. </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
-----Original Message----- </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; From:&nbsp;&nbsp; Linden deCarmo =
[SMTP:ldeCarmo@netspeak.com] </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Sent:&nbsp;&nbsp; Wednesday, May =
31, 2000 1:54 PM </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; To:&nbsp;&nbsp;&nbsp;&nbsp; =
sip@lists.bell-labs.com </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; =
Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [SIP] Suspend a =
session?? </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Right now, SIP smoothly handles one party putting another party =
on</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; hold via </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; a re-INVITE with modified =
SDP.&nbsp; However, we have a scenario where the </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; session needs to be suspended =
(not just placed on hold), and I'm not sure </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; how this can be done without =
using proprietary headers. </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Here are ther details.&nbsp; If party A calls party B , party B should =
be</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; able to </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; hang up without immediately =
terminating the session.&nbsp; If party B picks up </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; the phone within some prescribed =
time window, the session continues. </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; However, if they don't pick it =
up within the time window, the session </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; terminates (via a BYE). </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
We're looking for a mechanism for party B to communicate with =
either</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; party A </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; or a proxy server, to inform the =
distant party that the session has been </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; suspended, and not just placed =
on hold.&nbsp; The primary motivator for this </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; notification is billing.&nbsp; =
Without notification of suspension, its</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; impossible </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; to determine when to safely stop =
billing. </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Thanks. </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Linden deCarmo </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Netspeak Corporation </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; 902 Clint Moore Road </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Suite 104 </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Boca Raton, FL 33487 </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; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
_______________________________________________ </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; SIP mailing list </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; SIP@lists.bell-labs.com </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &lt;<U></U></FONT><U><FONT =
COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://lists.bell-labs.com/mailman/listinfo/sip" =
TARGET=3D"_blank">http://lists.bell-labs.com/mailman/listinfo/sip</A></F=
ONT></U><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
</P>
<BR>

<P><FONT SIZE=3D2 =
FACE=3D"Arial">_______________________________________________</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">SIP mailing list</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">SIP@lists.bell-labs.com</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
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TARGET=3D"_blank">http://lists.bell-labs.com/mailman/listinfo/sip</A></F=
ONT></U>
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To: Sean Olson <eussean@exu.ericsson.se>
Cc: mat@cisco.com, mshanka@unity.ncsu.edu, sip@lists.bell-labs.com
Subject: Re: [SIP] CPL
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Sean Olson writes:
 > >    I'm going to show my complete ignorance here,
 > >    but doesn't this beg the "sandbox" question?
 > > 
 > >    Related: how is authorization accomplished?
 > > 
 > > 	    Mike
 > > 
 > 
 > If you really want to take a dive here, I would argue that you re-use everything
 > that HTTP/HTTPS provides in terms of security and authorization. Of course you
 > can always sign the CPL script as you would any other XML document for another
 > layer of security. 

   What I had in mind here is how would a proxy
   know what kinds of actions a given user with
   a given CPL script is allowed to do? It could
   be as simple as a binary decision of if you
   allow CPL at all, you allow any arbitrary 
   script to be executed. 

   My suspicion is that the binary decision is
   too course though, which make the problem
   more difficult.

		Mike


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From sip-admin@lists.bell-labs.com  Thu Jun  1 15:48:04 2000
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To: Linden deCarmo <ldeCarmo@netspeak.com>
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Subject: Re: [SIP] Suspend a session??
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I think that the solution was pointed out earlier in this thread.
The problem seems to be that you want to charge for the existence of
a signalling connection.  

If billing was based on packets, you would simply put the remote UA 
on hold.  You want to put the remote UA on hold anyway because that 
may lessen the chances of additional billing.  

I feel what you want to do is to issue a re-INVITE to go on hold, 
then somehow issue some "STOP BILLING" request to entities in the 
signalling path that may be accumulating billing data.  That kind of 
request would be subject to abuse of course.

But couldn't that billing entity simply look at the successful hold
to decide to suspend billing (or bill at some reduced rate)?  The 
next signalling that would normally come through is a BYE or another
INVITE which stops or will resume normal billing.

Then I'll argue that if your UA is getting excessively charged for
sessions that are on hold (i.e. generating zero media), then it's
time to find a new service provider.

Regards,

Dave Walker
SS8 Networks
Ottawa, Canada


Linden deCarmo wrote:
> 
> Here's the problem with the UA doing it:
> 
> User goes on hook (time 0 ).  UA starts timer.
> Timer expires (time 30 seconds). UA realizes session expires so it issues a
> BYE.
> 
> Since the proxy sees the BYE 30 seconds later, thats when it stops billing.
> I do not believe your solution addresses how the UA signals to the proxy
> that billing should stop at time 0 rather than time 30.
> 
> I realize I can do this with a proprietary method or header, but I'd rather
> not.
> 
> Linden deCarmo
> Netspeak Corporation
> 902 Clint Moore Road
> Suite 104
> Boca Raton, FL 33487
> 
> > -----Original Message-----
> > From: Glenn Morrow [SMTP:gmorrow@nortelnetworks.com]
> > Sent: Thursday, June 01, 2000 11:38 AM
> > To:   sip@lists.bell-labs.com
> > Subject:      RE: [SIP] Suspend a session??
> >
> > It seems to me that this should be handled by the UA itself. Once the UA
> > determines the correct functionilty that the real end-device/user wishes
> > to achieve, it should signal proxies and far-end stations. For instance a
> > legacy device may which to indicate flash and collect more digits for a
> > 3-way call by the user using the hook as you say.
> >
> > This would result in a separate transaction but I believe it doesn't
> > require something like a  "suspend" method or indication at the SIP level.
> >
> >
> > I suggest that the UA itself should handle this with timers, etc.. based
> > upon the physical device and human interface it is trying to achieve.
> >
> > For the billing issue the UA could simply use a timer started based on
> > physical off-hook stimulus to determine when to actually send a BYE or
> > not.
> >
> > If this doesn't address your concerns please let me know.
> >
> >       -----Original Message-----
> > From:   Linden deCarmo [SMTP:ldeCarmo@netspeak.com]
> > Sent:   Wednesday, May 31, 2000 1:54 PM
> > To:     sip@lists.bell-labs.com
> > Subject:        [SIP] Suspend a session??
> >
> >       Right now, SIP smoothly handles one party putting another party on
> > hold via
> > a re-INVITE with modified SDP.  However, we have a scenario where the
> > session needs to be suspended (not just placed on hold), and I'm not sure
> > how this can be done without using proprietary headers.
> >
> >       Here are ther details.  If party A calls party B , party B should be
> > able to
> > hang up without immediately terminating the session.  If party B picks up
> > the phone within some prescribed time window, the session continues.
> > However, if they don't pick it up within the time window, the session
> > terminates (via a BYE).
> >
> >       We're looking for a mechanism for party B to communicate with either
> > party A
> > or a proxy server, to inform the distant party that the session has been
> > suspended, and not just placed on hold.  The primary motivator for this
> > notification is billing.  Without notification of suspension, its
> > impossible
> > to determine when to safely stop billing.
> >
> >       Thanks.
> >
> >       Linden deCarmo
> > Netspeak Corporation
> > 902 Clint Moore Road
> > Suite 104
> > Boca Raton, FL 33487
> >
> >
> >
> >       _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > <http://lists.bell-labs.com/mailman/listinfo/sip>
> >
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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From sip-admin@lists.bell-labs.com  Thu Jun  1 16:08:47 2000
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Date: Thu, 01 Jun 2000 16:08:16 -0400
From: Hyong Sop Shim <hyongsop@research.telcordia.com>
Organization: Telcordia Technologies
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Subject: [SIP] Could you provide a scenario?
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Hi,

RFC2543bis provides the following as the definition of "Initiator,
calling party, caller":
"The party initiating a conference invitation.  Note that the calling
party does not have to be the same as the one creating the conference."

Could anyone provide a scenario, in which demonstrates this noted case,
i.e., a UA creates a conference and a different UA initiates a
conference invitation to other UAs?

Thanks,

--Hyong Shim
(hyongsop@research.telcordia.com)



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Hi All,
  I have a few basic questions about SIP Registrar functionality.

 - Can a User Agent also act as a Registrar ? Assuming that a UA 
   can act as a Registrar as well, is it expected to NOT respond 
   to any REGISTER requests ?
 
 - Need some clarity on the following para in Section 4.2.6 [2543bis]. 
   "It is recommended that the UA registers via multicast and send a 
   registration to it's home address"

   Does this mean that registry database is maintained at two places,
   one at the registrar listening on the well known multicast address
   and another at the server at the home address ?

   To my understanding, there should be a local SIP Registrar server
   for any network. All UA's should register with this server when they
   come up. However, is it possible for UA's to also cache the registrations
   and use the cached entries for subsequent calls with out going to the
   location services or the registrar in the network ? 

thanks,
aseem


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Cc: sip@lists.bell-labs.com
Subject: Re: [SIP] Could you provide a scenario?
References: <3936C2AF.1953A082@research.telcordia.com>
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Easy: Conference initiated by SAP by some entity A, with the SIP
invitation issued by anybody who happens to be listening to the
announcement.

Hyong Sop Shim wrote:
> 
> Hi,
> 
> RFC2543bis provides the following as the definition of "Initiator,
> calling party, caller":
> "The party initiating a conference invitation.  Note that the calling
> party does not have to be the same as the one creating the conference."
> 
> Could anyone provide a scenario, in which demonstrates this noted case,
> i.e., a UA creates a conference and a different UA initiates a
> conference invitation to other UAs?
> 
> Thanks,
> 
> --Hyong Shim
> (hyongsop@research.telcordia.com)
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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Aseem Agarwal wrote:
> 
> Hi All,
>   I have a few basic questions about SIP Registrar functionality.
> 
>  - Can a User Agent also act as a Registrar ? Assuming that a UA
>    can act as a Registrar as well, is it expected to NOT respond
>    to any REGISTER requests ?

Being a UA is a logical role, as is a registrar. If an entity accepts
REGISTERs and stores location information, its a registrar. So, you can
write a piece of code that is both a UA and a registrar if you want.

> 
>  - Need some clarity on the following para in Section 4.2.6 [2543bis].
>    "It is recommended that the UA registers via multicast and send a
>    registration to it's home address"
> 
>    Does this mean that registry database is maintained at two places,
>    one at the registrar listening on the well known multicast address
>    and another at the server at the home address ?

Maybe, maybe not. You can have a single server listening to both
multicast and unicast.

> 
>    To my understanding, there should be a local SIP Registrar server
>    for any network. All UA's should register with this server when they
>    come up. However, is it possible for UA's to also cache the registrations
>    and use the cached entries for subsequent calls with out going to the
>    location services or the registrar in the network ?

Sure, its possible. Seems a bit more work than its worth, since the UA
basically needs to be a registrar. Also, this only works for multicast,
as that is the only case where other UAs can see the registrations. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Fri Jun  2 01:10:32 2000
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        sip@lists.bell-labs.com
Subject: Re: [SIP] CPL
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Michael Thomas wrote:
> 
>    What I had in mind here is how would a proxy
>    know what kinds of actions a given user with
>    a given CPL script is allowed to do? It could
>    be as simple as a binary decision of if you
>    allow CPL at all, you allow any arbitrary
>    script to be executed.

The point of CPL is that it can't do too much to begin with. Its a
scripting language - there are no variables, no looping constructs,
nothing. There are six or so basic primitives defined - decision making
primitives that check the caller, callee, time of day, or priority, and
action primitives that can forward, reject, redirect. Thats mostly it.
As a result, CPL is safe to execute, since it simply can't do any damage
to a server. Thus, I believe the binary decision about authorization is
perfectly legitimate. 

That said, you *could* specify that a proxy should not accept CPLs from
user Joe that use the time-of-day switch statement, but why?

For more info on CPL, check out the framework document which just issued
today as RFC2824. It compares CPL to other programming interfaces in
which the program executes in a sandbox, such as servlets.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Subject: Re: [SIP] SIP Application Server
References: <NDBBJGPOALAMPHONHENFEEMODBAA.scott@broadsoft.com> <3933EEB2.893C06AB@ubiquity.net> <39343EDC.2B4005FD@dynamicsoft.com> <3934CA2C.F4538AD5@ubiquity.net> <3935E680.B7598DA4@dynamicsoft.com> <39363991.882283F9@ubiquity.net>
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Neil Deason wrote:
> 
> Yes of course a separate SIP network element can talk SIP to a smart proxy/application
> server. But the point I was trying to make is that the application server encapsulates a
> vanilla proxy server, i.e. the proxy functionality as described by rfc2543, plus service
> logic accessed through CPL/SIP-CGI/Servlet API/whatever. My difficulty is in seeing how
> this fits with the ISC application working group premise of separating call signalling from
> service control via a SIP messaging interface. This seems like an unnecessary IN model of
> separation. In the App Server the proxy part does signalling but it is clearly not
> separated from controlling service logic by a SIP messaging interface.

I think you are over-analyzing the meaning of separation of
call-signaling from service control. In reality, all services are
delivered eventually through signaling, and are based on information
provided by signaling. Thus, service logic is quite connected to call
signaling, and embedding a vanilla proxy in an app server is quite a
good way to access it. I would argue this is quite contrary to the IN
model, in which there is an attempt to use a separate protocol for
service control to a separate box. Note however that these protocols
carry much the same information as the signaling protocols themselves.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Fri Jun  2 01:33:31 2000
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Subject: Re: [SIP] Suspend a session??
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Dave Walker wrote:
> 
> I think that the solution was pointed out earlier in this thread.
> The problem seems to be that you want to charge for the existence of
> a signalling connection.

Actually, the problem is that you are trying to bill for services you
are not actually providing. Unless there is either transport level
services, or gateway services, billing anything by the minute is going
to be a tough proposition. Providing a mechanism that allows the user to
stop billing actually makes the job of getting around billing really
easy.

Now, if you were providing QoS, I would argue that "hold" would not also
be accompanied by a release of QoS resources, where perhaps "suspend"
would. In this case, releasing of QoS resources would signal to stop
billing by the minute. But, thats all within the transport
infrastructure, not SIP.

> 
> If billing was based on packets, you would simply put the remote UA
> on hold.  You want to put the remote UA on hold anyway because that
> may lessen the chances of additional billing.
> 
> I feel what you want to do is to issue a re-INVITE to go on hold,
> then somehow issue some "STOP BILLING" request to entities in the
> signalling path that may be accumulating billing data.  That kind of
> request would be subject to abuse of course.

Exactly.

> 
> But couldn't that billing entity simply look at the successful hold
> to decide to suspend billing (or bill at some reduced rate)? 

Not really; this is also subject to abuse. I'll just put you on hold
within SIP, but both near and far end have "updated" SIP terminals which
actually ignore the hold message anyway, and continue to send media.
Free phone calls. Cool.


-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Fri Jun  2 04:22:04 2000
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Jonathan Rosenberg wrote:

> Neil Deason wrote:
> >
> > Yes of course a separate SIP network element can talk SIP to a smart proxy/application
> > server. But the point I was trying to make is that the application server encapsulates a
> > vanilla proxy server, i.e. the proxy functionality as described by rfc2543, plus service
> > logic accessed through CPL/SIP-CGI/Servlet API/whatever. My difficulty is in seeing how
> > this fits with the ISC application working group premise of separating call signalling from
> > service control via a SIP messaging interface. This seems like an unnecessary IN model of
> > separation. In the App Server the proxy part does signalling but it is clearly not
> > separated from controlling service logic by a SIP messaging interface.
>
> I think you are over-analyzing the meaning of separation of
> call-signaling from service control. In reality, all services are
> delivered eventually through signaling, and are based on information
> provided by signaling. Thus, service logic is quite connected to call
> signaling, and embedding a vanilla proxy in an app server is quite a
> good way to access it. I would argue this is quite contrary to the IN
> model, in which there is an attempt to use a separate protocol for
> service control to a separate box. Note however that these protocols
> carry much the same information as the signaling protocols themselves.

Agreed. It was not my analysis that SIP should be used to try and separate out service control
from call control. However, as was pointed out earlier in this thread this could be achieved if
SIP is used to separate a non-SIP signalling protocol from service logic in a 'Softswitch'
network. But then it appears that you would have a separate protocol (SIP) for service control to
a separate box (App Server). Such separation doesn't really apply in the same way to the SIP part
of the network and it sounds a lot like the Softswitch network is a SIP network at the core.
There you find SIP conversant App Servers and around them are SIP Proxy Servers, SIP Redirect
Servers and SIP User Agents. Some of these UAs may in fact represent interfaces to PSTN or even
H.323 gateways. Though how do protocols that don't use SIP for signalling deliver all the rich
new services that SIP enables when, for example, a text/html payload gets discarded at the
SIP/foo gateway. The point of SIP is surely not to create LNP, 800 translation services in an App
Server for protocols such as SS7, H.323 etc Yes, you could recreate the IN this way but why not
start out intending to build something better. You achieve this by building Application Servers
that empower the simple creation of innovative services not Application Servers that merely
recreate old services.

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



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Neil Deason wrote:
> 
> > I think you are over-analyzing the meaning of separation of
> > call-signaling from service control. In reality, all services are
> > delivered eventually through signaling, and are based on information
> > provided by signaling. Thus, service logic is quite connected to call
> > signaling, and embedding a vanilla proxy in an app server is quite a
> > good way to access it. I would argue this is quite contrary to the IN
> > model, in which there is an attempt to use a separate protocol for
> > service control to a separate box. Note however that these protocols
> > carry much the same information as the signaling protocols themselves.
> 
> Agreed. It was not my analysis that SIP should be used to try and separate out service control
> from call control. However, as was pointed out earlier in this thread this could be achieved if
> SIP is used to separate a non-SIP signalling protocol from service logic in a 'Softswitch'
> network. But then it appears that you would have a separate protocol (SIP) for service control to
> a separate box (App Server). Such separation doesn't really apply in the same way to the SIP part
> of the network and it sounds a lot like the Softswitch network is a SIP network at the core.
> There you find SIP conversant App Servers and around them are SIP Proxy Servers, SIP Redirect
> Servers and SIP User Agents. Some of these UAs may in fact represent interfaces to PSTN or even
> H.323 gateways. Though how do protocols that don't use SIP for signalling deliver all the rich
> new services that SIP enables when, for example, a text/html payload gets discarded at the
> SIP/foo gateway. The point of SIP is surely not to create LNP, 800 translation services in an App
> Server for protocols such as SS7, H.323 etc Yes, you could recreate the IN this way but why not
> start out intending to build something better. You achieve this by building Application Servers
> that empower the simple creation of innovative services not Application Servers that merely
> recreate old services.

I couldn't agree more with this sentiment.

That said, I have a sneaking suspicion that people will still want to
build these older services. Fortunately, most are supported by SIP, and
can easily be built ontop of an applications server as well. These
services (things like LNP and freephone) will work for H.323 devices,
PSTN phones or ISDN phones accessing the SIP network through a gateway
or softswitch. Even better, SIPs capabilities for negotiating
extensions, content, methods, and so on, will often allow an app server
to know if the accessing device can support a more innovative service.
In the example you give, where the app server returns HTML to a
softswitch, the app server can know apriori that HTML is or isn't
supported (Accept: text/html in the INVITE). It could then do something
different with the request for non-html access devices, like forward it
to a media server running TTS or something, and read out the web page.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Gethin Liddell wrote:
> 
> hi,
> 
> I'm querying the use of the hidden parameter and received parameter in the same via.
> 
> It is the purpose of the hidden parameter to hide the previous hop from subsequent hopes and this
> is done by encrypting the 'sent-by' part of the Via header.
> 
> However, it is possible that the Proxy has added a received parameter to the encrypted via in order
> to reply to the correct machine.  This received parameter provides open knowledge of the previous
> hop's address and compromises the security.

Good observation.

> 
> Now there are two possibilities here (as far as i can see):
> 
> 1) The received parameter should not be added to the hidden via but cached by the proxy in a lookup
>     table whilst it waits for the response. (or the received parameter can be encrypted as well).
> 
> 2) For a received parameter to be inserted, then the message has probably come through a NAT or firewall.
>     this is sufficent enough for security matters and aslong as the original host is encrypted within the via
>    i.e. the sent-by remains encrypted, all is OK.

There is another possibility, which seems the best choice:

3) Both the sent-by and received parameter are encrypted and placed in
to the Via header.


I don't like (1), since you lose the property of being able to run
statelessly. 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Fri Jun  2 10:17:57 2000
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On Fri, 02 Jun 2000, Jonathan Rosenberg wrote:
> Gethin Liddell wrote:
> > 
> > However, it is possible that the Proxy has added a received parameter to the encrypted via in order
> > to reply to the correct machine.  This received parameter provides open knowledge of the previous
> > hop's address and compromises the security.
> 
> Good observation.
> 
> > 1) The received parameter should not be added to the hidden via but cached by the proxy in a lookup
> >     table whilst it waits for the response. (or the received parameter can be encrypted as well).
> 
> There is another possibility, which seems the best choice:
> 
> 3) Both the sent-by and received parameter are encrypted and placed in
> to the Via header.

actually that's what i meant in the brackets for (1)  :)

-- 
Gethin Liddell
Ubiquity Software Corporation

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


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From sip-admin@lists.bell-labs.com  Fri Jun  2 10:39:06 2000
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Sean Olson wrote:
> 
> Comments inline
> 
> > I propose the following changes to draft-ietf-sip-2543bis-00:
> >
> > Page 59, BNF of sent-by for Via header should read:
> > sent-by = ([user@]host[:port])|(concealed-host)
> > where user is as defined in Section 2, on page 18.
> >
> > Also, in section 4.2.6, page 28, description of Request-URI, delete the
> > sentence "The user name MUST be empty."
> >
> > The reason for both changes is that we have a model in which a SIP Server
> > is providing proxy services to a large collection of users.  It often
> > happens that a client belonging to user1 will send a SIP request such as
> > an INVITE to this server.  The server will receive this request and make
> > various decisions according to logic installed on behalf of user1.  This
> > logic frequently requires stateful interactions.  One decision could be to
> > proxy the INVITE to the server for the invited party (user2).  The proxy
> > server for user2 is located on the same host (and same port) as the server
> > for user1.  The server for user2 may create the response to the INVITE or
> > it may proxy it onwards to one or more clients for user2, or perhaps to
> > other users that have been designated by user2.  This is all according to
> > logic installed on behalf of user2. When the response comes back to the
> > server we want the server for user2 to see the response (and perhaps
> > decide to proxy the request to SIP entities) before the response is seen
> > by the server for user1.  The addition of the user part in the sent-by of
> > the via allows such a routing.  If it is ever used in a situation where
> > there is no concept of different servers for different users it is
> > ignored.
> >
> > Note that the branch parameter does not meet our needs since there may, in
> > fact, be no branching.  Also, note that a common port on the SIP Server
> > services all user agents, so the port parameter could not be used to
> > differentiate servers for different users.  And, there may be more users
> > serviced on this host than could be separated with a typical 16 bit port
> > number.
> >
> 
> You would only need to play these games if the proxy you are referring to is
> transaction-stateless. If it is NOT transaction-stateless, then it is a trivial
> matter to add the user information from the Request-URI to the transaction state
> you are already maintaining.

No. This problem is addressed directly by the new mechanisms for loop
detection and computation of the branch parameter. The spec details
these, but is incorrect in its statement that it need only be done for
forking proxies. All proxies are going to need compute and insert the
first piece of the two described pieces of information. We'll correct
this.

Let me explain how this will work for Rays case with the following
picture:




                               +-------------+
                               |             |
                               V             |
                         +-----------+       |
                         |           |     2 |
                         |           |       |
                         |   Proxy   | ------+
                         |           |
                         |           |
                         +-----------+
                                   \
                           ^        \
                         //          \ 3
                        /             \
                       /               \
                     //  1              \
                    /                    V
           /-----\ /                        /-----\
          /       \                        /       \
         |         |                      |         |
         |    U1   |                      |   U2    |
          \       /                        \       /
           \-----/                          \-----/

(PS - figure created by email effects, http://www.sigsoftware.com, which
generates ascii art - this picture took me one minute to do....)

U1 sends message 1 to its proxy, P. The proxy forwards to itself, and
inserts into message 2 a via header with a branch parameter containing
the hash of the request URI from U1. Also note that because the request
URI is different, as is the Via header, when this request is received by
the proxy over again, it is viewed as a different request, and thus
creates a new transaction. It is also NOT viewed as a loop, since the
branch parameters in the Via indicate to the proxy that the previous
time this request was received, the request URI was different. Now, in
step 3, its forwarded to U2, and again the Via header is inserted, with
a different branch paramter again, reflecting the hash of the request
URI in message 2. When the response from U2 comes back, its forwarded
from the proxy to itself, matched with the correct request because of
the branch ID once more. It is then forwared to U1, and we're done.

This approach effectively encodes the user part of the request URI into
the Via header, as Ray was suggesting. However, by using the branch
parameter, we remain backwards compatible and have greater flexibility.

> > The suggestion for removing the sentence about the user part in the
> > Request-URI of REGISTER requests is similarly motivated. There is no
> > reason given in the rfc why the user part MUST be empty in a Request-URI
> > for REGISTER.  In our model, a client will want to REGISTER itself with
> > the server representing it within the SIP Server.  In practice, it would
> > be sufficient to use the From or To field of the REGISTER to determine
> > which server should be used, but in a large system, where the servers
> > might be distributed among many hosts (even though they are seen as a
> > single SIP server host to the outside world) it is more efficient to use
> > the Request-URI for routing purposes so we want the user name to be part
> > of the Request-URI.

In this case, you should do your routing in a specialized way by looking
at the To field, which truly tells you which registrar to go to. This is
purely a local policy decision.

Adding the user name to the request URI in REGISTER is also not
backwards compatible.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Fri Jun  2 11:21:38 2000
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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On Fri, June 2 2000, "Jonathan Rosenberg" wrote to "Michael Thomas, Sean Olson, mshanka@unity.ncsu.edu, sip@lists.bell-labs.com" saying:

> Michael Thomas wrote:
> > 
> >    What I had in mind here is how would a proxy
> >    know what kinds of actions a given user with
> >    a given CPL script is allowed to do? It could
> >    be as simple as a binary decision of if you
> >    allow CPL at all, you allow any arbitrary
> >    script to be executed.
> 
> The point of CPL is that it can't do too much to begin with. Its a
> scripting language - there are no variables, no looping constructs,
> nothing. There are six or so basic primitives defined - decision making
> primitives that check the caller, callee, time of day, or priority, and
> action primitives that can forward, reject, redirect. Thats mostly it.
> As a result, CPL is safe to execute, since it simply can't do any damage
> to a server. Thus, I believe the binary decision about authorization is
> perfectly legitimate. 
> 
> That said, you *could* specify that a proxy should not accept CPLs from
> user Joe that use the time-of-day switch statement, but why?

It *is* possible that a CPL server could not support or be willing to
support certain services for a partiuclar user.  I think the most sensible
example of this would be a server which will only redirect, not proxy
(either in general or for a particular user).  Another example would be a
server which can't or won't proxy to certain URI schemes -- h323 (once
that's defined), tel, etc.

A CPL server needs to validate a script when it is inserted into the server
-- for XML well-formedness, absence of dangling sub references, non-looping
status, etc.  It can (and should) check the script at the same time to make
sure all the actions it specifies are allowed for the associated user, and
reject it if it isn't.  (This rejection mechanism depends on the submission
mechanism; in draft-lennox-sip-reg it would be a 4xx response, for example.)

That said, how the server *knows* which actions are legal for each user is
entirely a local configuration/policy issue for the CPL server.  Yes, the
administrator needs some way of specifying it, but this isn't any different
from all the other policy issues the server administrator needs to be able
to control.

-- 
Jonathan Lennox
lennox@cs.columbia.edu


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From sip-admin@lists.bell-labs.com  Fri Jun  2 13:15:48 2000
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Date: Fri, 02 Jun 2000 10:14:33 -0700
To: Alberto.Conte@ms.alcatel.fr
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [SIP] INVITE with incomplete SDP description
Cc: Lars Berggren <lars@intertex.se>,
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        Neil Deason <ndeason@ubiquity.net>, IETF SIP <sip@lists.bell-labs.com>
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Alberto,

I also agree with Jonathan, et al, that what you want is a minor 
optimization and is better served with a reINVITE, or INVITE+ACK.  However, 
you may be interested in the discussions of SDPng in the mmusic WG.  The 
goal is for SDPng to have much finer control over the details of 
capabilities negotiation, etc.

thanks,
-rohan


At 06:15 AM 5/26/00 , Alberto Conte wrote:
>Sorry, but I still disagree.
>
>I don't think that specifying what does it mean a port number = 0 in the 
>INVITE
>message (which is at the moment not defined in the RFC 2543) is a way to 
>make the
>protocol more complex.
>
>I think it is just a way to ensure implementation compatibility by 
>specifying how
>this kind of message must be handled. That will never prevent to use the 
>solution
>proposed by Jonathan.
>
>Alberto
>
>Lars Berggren a écrit :
> >
> > On Thu, 25 May 2000, Alberto Conte wrote:
> >
> > > Putting in hold the calle is a solution, but that increases the 
> number of messages to
> > > be exchanged, requiring a re-INVITE transaction!
> > >
> > > The power of SIP is to reduce the message exchanged in case of no 
> error. And this is
> > > actually a case of no-error. It should be better allowing the caller 
> to propose its
> > > media request delaying only the RTP port information in the ACK.
> > >
> > > Alberto
> > >
> >
> > I would rather think the power of SIP is its generality, flexibility and
> > simplicity. A re-INVITE is a very general and powerful mechanism to
> > accomplish many different tasks, including yours, so why not keep it that
> > way instead of specifying special cases which tend to make the protocol
> > more complex.
> >
> > /Lars
> >
> > Lars Berggren       <mailto:lars.berggren@intertex.se>
> > Intertex Data AB    tel: +46-8-6282828
> > Sundbyberg, Sweden  fax: +46-8-6286414
>
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip



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From sip-admin@lists.bell-labs.com  Fri Jun  2 15:15:31 2000
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From: "Schroeder, Tim" <schroeder@tri.sbc.com>
To: "'sip@lists.bell-labs.com'" <sip@lists.bell-labs.com>
Subject: RE: [SIP] comments on draft-ietf-sip-100rel-01.txt
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> > Section 5.1, Paragraph 11: "It is not necessary to include the Supported
> > header listing the option tag 100rel in the PRACK request."

> Given the change that you can't send a reliable provisional response to
> PRACK, it seems this text is actually fine - its not needed to include
> this Supported header in PRACK.

Given the change that you can't send a reliable provisional response to
PRACK, perhaps the current text could remain, and an additional sentence
prohibiting the use of "Require: 100rel" in a PRACK request should be added.
Otherwise, somewhere we'd probably have to clarify what to do if a UAC
requires reliable provisional responses, but the UAS is prohibited from
sending them reliably.  

I'm still in the "no reliable provisional responses to a PRACK" camp.  If
they were allowed, then presumably the UAC would have the right to require
them (unlike the proposed change above).  This seems like it could lead to
trouble in an environment where the UAC and UAS are implemented by different
parties, who don't necessarily trust each other.  Suppose a UAC requires all
provisional responses to be sent reliably for all requests, and habitually
includes "Require: 100rel" in its messages.  That doesn't seem to be too
much of a stretch, or to unequivocally imply a badly written UAC.  And
suppose a UAS desires to send a provisional response to a PRACK (I can't
think of an obvious example here, but if *reliable* provisional responses
are useful, certainly provisional responses must be useful to begin with!).
This would lead to the never-ending series of PRACKs arriving at the UAS,
and the UAS would have to decide not to send provisional responses at all to
defend itself; if it sent a response, it'd be required to do so reliably,
and that would result in yet another PRACK. 

Tim Schroeder
tschroeder@tri.sbc.com
SBC Technology Resources, 9505 Arboretum Blvd., Austin, TX  78759


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From sip-admin@lists.bell-labs.com  Fri Jun  2 15:49:05 2000
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Message-ID: <39380F9F.B2A0ADE1@dynamicsoft.com>
Date: Fri, 02 Jun 2000 15:48:47 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Folks,

At the last IETF, James Kempf presented a draft he and I co-wrote on
using SLP to discover SIP servers. At the time, James was unsure of the
process for registration of these templates. Later on, we discovered
that a standards track RFC does NOT need to be generated. As a result,
the templates from that draft were cleaned up, and are now officially
registered with IANA. You can look them over at:

ftp://ftp.isi.edu/in-notes/iana/assignments/svrloc-templates

Please give them a glance and let the authors or the list know if there
are any comments or issues.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Fri Jun  2 17:22:39 2000
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From: Shail Bhatnagar <shbhatna@cisco.com>
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Subject: [SIP] Forking proxy and Expires timer
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I have a question about the behavior of a forking proxy in a specific
scenario. Here is the scenario :

Proxy receives an INVITE with an Expires header (say 60 seconds) and
forwards
the INVITE to 2 locations. One location immediately returns 302 and the
other
returns 180 ringing (phone keeps ringing). After 60 seconds have
elapsed,
should the proxy return 408 Request Timedout or 302 Moved temporarily (
assuming that proxy is not recursive forking). 
Also, I am assuming that user agent will start the "Expires" timer on
receipt
of the first provisional and the proxy will start the "Expires" timer 
immediately after receiving the INVITE ( or the client's timer will
expire
before the proxy's). 


-- 
Best regards,
Shail


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From sip-admin@lists.bell-labs.com  Fri Jun  2 20:43:20 2000
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Date: Fri, 02 Jun 2000 19:42:58 -0500
From: Henry Sinnreich <Henry.Sinnreich@WCom.com>
Subject: RE: [SIP] SIP Application Server
In-reply-to: <3937B7FB.BC71F9F7@dynamicsoft.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Neil Deason <ndeason@ubiquity.net>
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I fully agree with Jonathan. The whole point is not just to
replicate what can be already done with AIN/PSTN, but for new
services that produce new revenue.

Henry

Henry Sinnreich
MCI WorldCom
400 International Parkway
Richardson, Texas 75081
USA

>-----Original Message-----
>From: sip-admin@lists.bell-labs.com
>[mailto:sip-admin@lists.bell-labs.com]On Behalf Of
>Jonathan Rosenberg
>Sent: Friday, June 02, 2000 8:35 AM
>To: Neil Deason
>Cc: sip@lists.bell-labs.com
>Subject: Re: [SIP] SIP Application Server
>
>
>
>
>Neil Deason wrote:
>>
>> > I think you are over-analyzing the meaning of separation of
>> > call-signaling from service control. In reality,
>all services are
>> > delivered eventually through signaling, and are
>based on information
>> > provided by signaling. Thus, service logic is
>quite connected to call
>> > signaling, and embedding a vanilla proxy in an app
>server is quite a
>> > good way to access it. I would argue this is quite
>contrary to the IN
>> > model, in which there is an attempt to use a
>separate protocol for
>> > service control to a separate box. Note however
>that these protocols
>> > carry much the same information as the signaling
>protocols themselves.
>>
>> Agreed. It was not my analysis that SIP should be
>used to try and separate out service control
>> from call control. However, as was pointed out
>earlier in this thread this could be achieved if
>> SIP is used to separate a non-SIP signalling
>protocol from service logic in a 'Softswitch'
>> network. But then it appears that you would have a
>separate protocol (SIP) for service control to
>> a separate box (App Server). Such separation doesn't
>really apply in the same way to the SIP part
>> of the network and it sounds a lot like the
>Softswitch network is a SIP network at the core.
>> There you find SIP conversant App Servers and around
>them are SIP Proxy Servers, SIP Redirect
>> Servers and SIP User Agents. Some of these UAs may
>in fact represent interfaces to PSTN or even
>> H.323 gateways. Though how do protocols that don't
>use SIP for signalling deliver all the rich
>> new services that SIP enables when, for example, a
>text/html payload gets discarded at the
>> SIP/foo gateway. The point of SIP is surely not to
>create LNP, 800 translation services in an App
>> Server for protocols such as SS7, H.323 etc Yes, you
>could recreate the IN this way but why not
>> start out intending to build something better. You
>achieve this by building Application Servers
>> that empower the simple creation of innovative
>services not Application Servers that merely
>> recreate old services.
>
>I couldn't agree more with this sentiment.
>
>That said, I have a sneaking suspicion that people
>will still want to
>build these older services. Fortunately, most are
>supported by SIP, and
>can easily be built ontop of an applications server as
>well. These
>services (things like LNP and freephone) will work for
>H.323 devices,
>PSTN phones or ISDN phones accessing the SIP network
>through a gateway
>or softswitch. Even better, SIPs capabilities for negotiating
>extensions, content, methods, and so on, will often
>allow an app server
>to know if the accessing device can support a more
>innovative service.
>In the example you give, where the app server returns HTML to a
>softswitch, the app server can know apriori that HTML
>is or isn't
>supported (Accept: text/html in the INVITE). It could
>then do something
>different with the request for non-html access
>devices, like forward it
>to a media server running TTS or something, and read
>out the web page.
>
>-Jonathan R.
>--
>Jonathan D. Rosenberg                       72 Eagle Rock Ave.
>Chief Scientist                             First Floor
>dynamicsoft                                 East
>Hanover, NJ 07936
>jdrosen@dynamicsoft.com                     FAX:
>(973) 952-5050
>http://www.cs.columbia.edu/~jdrosen         PHONE:
>(732) 741-7244
>http://www.dynamicsoft.com
>
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip



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From sip-admin@lists.bell-labs.com  Sat Jun  3 00:08:21 2000
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Subject: Re: [SIP] Forking proxy and Expires timer
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Shail Bhatnagar wrote:
> 
> I have a question about the behavior of a forking proxy in a specific
> scenario. Here is the scenario :
> 
> Proxy receives an INVITE with an Expires header (say 60 seconds) and
> forwards
> the INVITE to 2 locations. One location immediately returns 302 and the
> other
> returns 180 ringing (phone keeps ringing). After 60 seconds have
> elapsed,
> should the proxy return 408 Request Timedout or 302 Moved temporarily (
> assuming that proxy is not recursive forking).

302, as the best response received. Consider timeout is the same as if
408 was actually received.

> Also, I am assuming that user agent will start the "Expires" timer on
> receipt
> of the first provisional and the proxy will start the "Expires" timer
> immediately after receiving the INVITE ( or the client's timer will
> expire
> before the proxy's)

If Expires contains an exact date, then everyone has the same time. Even
with delta-seconds, I would say its the time of sending the INVITE. If a
UA places an Expires in the INVITE, it really doesn't need to *also*
CANCEL the request itself when the timer expires. It could just as well
do that without having the Expires in the INVITE. You can consider
Expires as sort of asking the proxies to do the CANCEL for you. We
should probably clarify that. 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Sat Jun  3 11:26:28 2000
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I'll confirm the suspicion that (some) people do want to deploy "older"
services in a Softswitch/SIP network.  As I've seen and heard many times,
the IP network provides a very attractive (cost-wise) alternative to the
circuit-switched/IN network for carrying telecommunications traffic.  I like
the ISC's goal hosting applications on a separate server from the
Softswitch.  And using SIP as the protocol between these two devices makes
creating and deploying new services easy compared to doing the same thing in
the PSTN.

However, the standards work needed to extend SIP to support deploying
existing PSTN-like services in an IP network (and to "seamlessly" integrate
the PSTN & IP networks) still has a way to go.  Specifically, carrying DTMF
out-of-band and within SIP.  I'm aware of two specific drafts proposing
methods but have not seen nor found any discussion or consensus within the
SIP WG.  Hopefully, interest in this issue will occur at some point.

Regards,
Bert

Bert Culpepper
InterVoice-Brite, Inc.

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Friday, June 02, 2000 9:35 AM
> To: Neil Deason
> Cc: sip@lists.bell-labs.com
> Subject: Re: [SIP] SIP Application Server
> 
> 
> 
> 
> Neil Deason wrote:
> > 
> > > I think you are over-analyzing the meaning of separation of
> > > call-signaling from service control. In reality, all services are
> > > delivered eventually through signaling, and are based on information
> > > provided by signaling. Thus, service logic is quite connected to call
> > > signaling, and embedding a vanilla proxy in an app server is quite a
> > > good way to access it. I would argue this is quite contrary to the IN
> > > model, in which there is an attempt to use a separate protocol for
> > > service control to a separate box. Note however that these protocols
> > > carry much the same information as the signaling protocols themselves.
> > 
> > Agreed. It was not my analysis that SIP should be used to try and
> > separate out service control
> > from call control. However, as was pointed out earlier in this
> > thread this could be achieved if
> > SIP is used to separate a non-SIP signalling protocol from service
> > logic in a 'Softswitch'
> > network. But then it appears that you would have a separate protocol
> > (SIP) for service control to
> > a separate box (App Server). Such separation doesn't really apply in
> > the same way to the SIP part
> > of the network and it sounds a lot like the Softswitch network is a
> > SIP network at the core.
> > There you find SIP conversant App Servers and around them are SIP
> > Proxy Servers, SIP Redirect
> > Servers and SIP User Agents. Some of these UAs may in fact represent
> > interfaces to PSTN or even
> > H.323 gateways. Though how do protocols that don't use SIP for
> > signalling deliver all the rich
> > new services that SIP enables when, for example, a text/html payload
> > gets discarded at the
> > SIP/foo gateway. The point of SIP is surely not to create LNP, 800
> > translation services in an App
> > Server for protocols such as SS7, H.323 etc Yes, you could recreate
> > the IN this way but why not
> > start out intending to build something better. You achieve this by
> > building Application Servers
> > that empower the simple creation of innovative services not
> > Application Servers that merely
> > recreate old services.
> 
> I couldn't agree more with this sentiment.
> 
> That said, I have a sneaking suspicion that people will still want to
> build these older services. Fortunately, most are supported by SIP,
> and
> can easily be built ontop of an applications server as well. These
> services (things like LNP and freephone) will work for H.323 devices,
> PSTN phones or ISDN phones accessing the SIP network through a gateway
> or softswitch. Even better, SIPs capabilities for negotiating
> extensions, content, methods, and so on, will often allow an app
> server
> to know if the accessing device can support a more innovative service.
> In the example you give, where the app server returns HTML to a
> softswitch, the app server can know apriori that HTML is or isn't
> supported (Accept: text/html in the INVITE). It could then do
> something
> different with the request for non-html access devices, like forward
> it
> to a media server running TTS or something, and read out the web page.
> 
> -Jonathan R.
> -- 
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 


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From sip-admin@lists.bell-labs.com  Sun Jun  4 02:50:12 2000
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I suspect I don't like what you're attempting to achieve, but methinks
the friendly UA could always submit an appropriate INFO method which the
"billing proxy" could use to determine an appropriate action.

AS has been pointed out here, it is that UA that is responsible for
translating user action into appropriate signaling.

One might also ask WHY and end device would choose to use a "time
billing proxy" rather than routing around said proxy. There are probably
some cases involving proxy-moderated firewalls and QoS sessions, but I
still suspect the fundamental approach is somehow flawed.

--
Dean

Linden deCarmo wrote:
> 
> Here's the problem with the UA doing it:
> 
> User goes on hook (time 0 ).  UA starts timer.
> Timer expires (time 30 seconds). UA realizes session expires so it issues a
> BYE.
> 
> Since the proxy sees the BYE 30 seconds later, thats when it stops billing.
> I do not believe your solution addresses how the UA signals to the proxy
> that billing should stop at time 0 rather than time 30.
> 
> I realize I can do this with a proprietary method or header, but I'd rather
> not.
> 
> Linden deCarmo
> Netspeak Corporation
> 902 Clint Moore Road
> Suite 104
> Boca Raton, FL 33487
> 
> > -----Original Message-----
> > From: Glenn Morrow [SMTP:gmorrow@nortelnetworks.com]
> > Sent: Thursday, June 01, 2000 11:38 AM
> > To:   sip@lists.bell-labs.com
> > Subject:      RE: [SIP] Suspend a session??
> >
> > It seems to me that this should be handled by the UA itself. Once the UA
> > determines the correct functionilty that the real end-device/user wishes
> > to achieve, it should signal proxies and far-end stations. For instance a
> > legacy device may which to indicate flash and collect more digits for a
> > 3-way call by the user using the hook as you say.
> >
> > This would result in a separate transaction but I believe it doesn't
> > require something like a  "suspend" method or indication at the SIP level.
> >
> >
> > I suggest that the UA itself should handle this with timers, etc.. based
> > upon the physical device and human interface it is trying to achieve.
> >
> > For the billing issue the UA could simply use a timer started based on
> > physical off-hook stimulus to determine when to actually send a BYE or
> > not.
> >
> > If this doesn't address your concerns please let me know.
> >
> >       -----Original Message-----
> > From:   Linden deCarmo [SMTP:ldeCarmo@netspeak.com]
> > Sent:   Wednesday, May 31, 2000 1:54 PM
> > To:     sip@lists.bell-labs.com
> > Subject:        [SIP] Suspend a session??
> >
> >       Right now, SIP smoothly handles one party putting another party on
> > hold via
> > a re-INVITE with modified SDP.  However, we have a scenario where the
> > session needs to be suspended (not just placed on hold), and I'm not sure
> > how this can be done without using proprietary headers.
> >
> >       Here are ther details.  If party A calls party B , party B should be
> > able to
> > hang up without immediately terminating the session.  If party B picks up
> > the phone within some prescribed time window, the session continues.
> > However, if they don't pick it up within the time window, the session
> > terminates (via a BYE).
> >
> >       We're looking for a mechanism for party B to communicate with either
> > party A
> > or a proxy server, to inform the distant party that the session has been
> > suspended, and not just placed on hold.  The primary motivator for this
> > notification is billing.  Without notification of suspension, its
> > impossible
> > to determine when to safely stop billing.
> >
> >       Thanks.
> >
> >       Linden deCarmo
> > Netspeak Corporation
> > 902 Clint Moore Road
> > Suite 104
> > Boca Raton, FL 33487
> >
> >
> >
> >       _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > <http://lists.bell-labs.com/mailman/listinfo/sip>
> >
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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From sip-admin@lists.bell-labs.com  Sun Jun  4 09:28:18 2000
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* There have been a number of comments that SDP carried with SIP 
needs at least one RTP session:-
To which I reply...
Nooooooo....

For VoIP/Games/... usage of SIP I like the delayed media sessions 
section of the bis draft.
Please don't dump it, even though this has no media sessions at all in the SDP.

Even for not-transient situations, we MAY have this situation...for 
example, PINT doesn't use RTP sessions, but it still has SDP.

I suspect that the ATM stuff may NOT use RTP either - one may find 
VoATM distasteful, but there are customers for it; SIP/SDP may be 
used as a hybrid.

Whilst for VoIP sessions, lack of an RTP media session is strange, 
this is not always true, even when SIP is being used.
-- 
All the best, Lawrence
-----------------------------------------------------------------------
| Lawrence Conroy,    | "These Opinions must be mine, 'cos if they    |
| Roke Manor Research |  were my Company's they'd pay me for them"    |
|- lwc@roke.co.uk  ---+- Tel: +44 1794 833666  Fax: +44 1794 833434 --|


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*	UDP support changed from SHOULD to MUST in SIP 2453 bis draft.
It looks like support for UDP is gradually moving from a SHOULD to a 
MUST in SIP.

In PINT we may not have that assumption. We can have a good reason 
for using TCP only (inline content, security using TLS, ...), so *for 
PINT* I'd prefer to leave this as a SHOULD.

I wonder if the move to a MUST in the core spec will cause grief for 
any other extension or application.

The key point here is ->in the core spec<-

I had the same feeling at the Interim meeting in D.C. when the 
subject of "Busy Line Verification" was raised - there's a lot of 
focus on VoIP, when I can easily imagine SIP being used for a much 
wider set of applications even for "straight" conferences; for 
example, Quake, as Dean suggests.
Let's imagine an Operator listening in on the sampled Quake audio 
track that I'm conferencing. How long before the Police arrive? An 
over-emphasis on VoIP can lead to bad assumptions, IMHO - SIP is much 
wider than that.
-- 
lwc@roke.co.uk -- +44 1794 833666
-------------------------------------------------------------
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and that they would very much like people to sue me rather than
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Also, I can't do anything; only our lawyers can, so speak to them.
Oh, and by the way, if our I.T. department has so mangled our
email system that this has been misdirected, beware that having
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this highly sensitive information, unless you immediately
forget that this ever darkened your mailbox. Do it now.
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From sip-admin@lists.bell-labs.com  Sun Jun  4 10:27:09 2000
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Jonathan Rosenberg wrote:
> 

> >
> > - "...the H.245 [8] gateway and user address and then use H.225.0 [9] to
> > establish the call." - What is meant by an "H.245 gateway address" and an
> > "H.245 user address?" There is no such thing. I assume that the author is
> > referring to the H.323 call-signaling addresses of a gateway and endpoint.
> > There is an H.245 control-protocol address, but if a separate H.245 channel
> > is even used, the address is conveyed during call establishment and not
> > before.
> 
> You are of course correct; this should be H.323 call signaling address.

Actually, this was intentional. In the days of H.323v1, pre-Fast
Connect, one could imagine a scenario where you do a normal SIP INVITE
and then obtain the address of the peer. H.245 would then be used to
negotiate codecs and open logical channels. Clearly, this would not work
with any existing H.323 implementation, but I don't see why it couldn't
be made to work. However, the example may be too contrived to be useful.
(However, the straight combination with H.323 doesn't make a whole lot
of sense. Why would you have the mailman ring twice?)

> 
> >
> > - The word, "program,: is used throughout the RFC. Is this any functional
> > entity regardless of implementation and not exclusively a "software
> > program?"
> 
> Of course.

It is somewhat unlikely that you would implement SIP as hard-wired logic
gates or via air valves (yes, there were air valve logic gates at some
point...), so I'm confused as to why program is restricting your
choices.


> 
> The role of UAC and UAS is on a request by request basis, NOT a call by
> call basis. This is probably confusing from the spec, as much of it was
> written considering the initial INVITE request, for which caller and UAC
> are the same.

I tried to clarify the per-request nature of the UA* notion.


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


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Jonathan Rosenberg wrote:
> 

> In addition to this, we can define an explicit "remove me" record-route
> entry. Something like:
> 
> Remove-Route: <sip:user@host;maddr=1.2.3.4>;remove=true
> 
> Proxies would add this when they want out. A UA not understanding this
> will ignore out continue to use the old route, but a new UA would know
> that this means to remove it from the Route in the next request. I think
> this is also backwards compatible.
> 
> Both of these mechanisms combined (remove-route and adding record-route
> for mid-call messages) has the desired result of being backwards
> compatible, providing for restarts after crashes, and allowing proxies
> to remove themselves during a call.

Since this is something that can be added later if there's a real,
demonstrated example, my inclination would be to leave it out of the
base spec at this point as it does impose non-trivial implementation
effort. (In order to stay in the spec for Draft, we have to show two
implementations in any event.)

Another wrinkle in this Record-Route business is that we could
presumably define a hiding mechanism similar to Via hiding mechanism.
(The details would have to differ, since it's the upstream proxy that
would have to do the encryption). Is this useful? Personally, I'd rather
not add this complexity. Via hiding itself doesn't seem to be the
most-implemented feature at this stage...

Currently, there's a silent (and reasonable) assumption that the UAC
issuing the first request inserts a RR. However, would there be a use
for the UAS receiving that request doing this? Clearly, this would only
make sense if that request isn't routed directly, i.e., typically, if
the first INVITE didn't contain a Contact header. Is this useful enough
to be described explicitly? Any existing implementations deal with this
gracefully?

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


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From sip-admin@lists.bell-labs.com  Sun Jun  4 22:20:02 2000
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From: Paul Long <Plong@smithmicro.com>
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Henning,

Re: H.245 addresses
Intentional? I don't understand. There is no such thing, per se, as an H.245
gateway address or an H.245 user address. A gateway and endpoint (I assume
this is what is meant by "user") are canonically identified through their
respective call-signaling addresses. The address of the H.245 control
channel may then be conveyed via call signaling. One cannot first get the
H.245 gateway and user address and _then_ use H.225.0 (I assume this refers
to call signaling) to establish the call. It's the other way around.

Re: "program"
Yeah, I was being pedantic. :-) It just gives me the willies to see
implementation-specific terminology in a functional specification, even
though, yes, it would be very silly to use mechanical relays, air valves, or
uh... sticks and stones :-) ... to implement a SIP entity.

Re: UAC/UAS
I was just reading rfc2543bis-00 and don't remember seeing a clarification
of this. It would have been helpful because one usually thinks of clients
and servers as types or roles of entities rather than types of transactions.
Not a big deal, though.

Paul Long
Smith Micro Software, Inc.

-----Original Message-----
From: hgs@cs.columbia.edu [mailto:hgs@cs.columbia.edu]On Behalf Of
Henning Schulzrinne
Sent: Sunday, June 04, 2000 9:27 AM
To: Jonathan Rosenberg
Cc: Plong@smithmicro.com; sip@lists.bell-labs.com
Subject: Re: [SIP] Basic questions


Jonathan Rosenberg wrote:
>

> >
> > - "...the H.245 [8] gateway and user address and then use H.225.0 [9] to
> > establish the call." - What is meant by an "H.245 gateway address" and
an
> > "H.245 user address?" There is no such thing. I assume that the author
is
> > referring to the H.323 call-signaling addresses of a gateway and
endpoint.
> > There is an H.245 control-protocol address, but if a separate H.245
channel
> > is even used, the address is conveyed during call establishment and not
> > before.
>
> You are of course correct; this should be H.323 call signaling address.

Actually, this was intentional. In the days of H.323v1, pre-Fast
Connect, one could imagine a scenario where you do a normal SIP INVITE
and then obtain the address of the peer. H.245 would then be used to
negotiate codecs and open logical channels. Clearly, this would not work
with any existing H.323 implementation, but I don't see why it couldn't
be made to work. However, the example may be too contrived to be useful.
(However, the straight combination with H.323 doesn't make a whole lot
of sense. Why would you have the mailman ring twice?)

>
> >
> > - The word, "program,: is used throughout the RFC. Is this any
functional
> > entity regardless of implementation and not exclusively a "software
> > program?"
>
> Of course.

It is somewhat unlikely that you would implement SIP as hard-wired logic
gates or via air valves (yes, there were air valve logic gates at some
point...), so I'm confused as to why program is restricting your
choices.


>
> The role of UAC and UAS is on a request by request basis, NOT a call by
> call basis. This is probably confusing from the spec, as much of it was
> written considering the initial INVITE request, for which caller and UAC
> are the same.

I tried to clarify the per-request nature of the UA* notion.


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


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Paul Long wrote:
> 
> Henning,
> 
> Re: H.245 addresses
> Intentional? I don't understand. There is no such thing, per se, as an H.245
> gateway address or an H.245 user address. A gateway and endpoint (I assume
> this is what is meant by "user") are canonically identified through their
> respective call-signaling addresses. The address of the H.245 control
> channel may then be conveyed via call signaling. One cannot first get the
> H.245 gateway and user address and _then_ use H.225.0 (I assume this refers
> to call signaling) to establish the call. It's the other way around.

That's exactly my point. SIP in this fictitious scenario replaces
H.225.0. You then use the IP address and port obtained by SIP for the
H.245 control channel. Yes, it's contrived.

> 
> Re: "program"
> Yeah, I was being pedantic. :-) It just gives me the willies to see
> implementation-specific terminology in a functional specification, even
> though, yes, it would be very silly to use mechanical relays, air valves, or
> uh... sticks and stones :-) ... to implement a SIP entity.

I take concreteness over vagueness any day :-)

> 
> Re: UAC/UAS
> I was just reading rfc2543bis-00 and don't remember seeing a clarification
> of this. It would have been helpful because one usually thinks of clients
> and servers as types or roles of entities rather than types of transactions.
> Not a big deal, though.

It will appear whenever I get a chance to LaTeX the sources again.


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Subject: Re: [SIP] Record-Route (Was: Contact in OPTIONS)
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Henning Schulzrinne wrote:
> 
> Jonathan Rosenberg wrote:
> >
> 
> > In addition to this, we can define an explicit "remove me" record-route
> > entry. Something like:
> >
> > Remove-Route: <sip:user@host;maddr=1.2.3.4>;remove=true
> >
> > Proxies would add this when they want out. A UA not understanding this
> > will ignore out continue to use the old route, but a new UA would know
> > that this means to remove it from the Route in the next request. I think
> > this is also backwards compatible.
> >
> > Both of these mechanisms combined (remove-route and adding record-route
> > for mid-call messages) has the desired result of being backwards
> > compatible, providing for restarts after crashes, and allowing proxies
> > to remove themselves during a call.
> 
> Since this is something that can be added later if there's a real,
> demonstrated example, my inclination would be to leave it out of the
> base spec at this point as it does impose non-trivial implementation
> effort. (In order to stay in the spec for Draft, we have to show two
> implementations in any event.)

I agree.

> 
> Another wrinkle in this Record-Route business is that we could
> presumably define a hiding mechanism similar to Via hiding mechanism.
> (The details would have to differ, since it's the upstream proxy that
> would have to do the encryption). Is this useful? Personally, I'd rather
> not add this complexity. Via hiding itself doesn't seem to be the
> most-implemented feature at this stage...

Via hiding is not stricly needed; only if you want to be stateless AND
hide the route. The far easier approach is simply strip off the via
list, store it, and insert a fresh Via header with some public address
you are not afraid to show to the world, in order to get the response.
When the response comes, slap back on the old Via list, and off you go. 

I'll also note that Via hiding always worried me because it relied on
the *next hop* to encrypt for you; what if the trust relationship is
such that they might not do this for you? Better off protect yourself
than trust the next hop to do so.

So, I would agree that adding this for Record-Route is probably not
needed.

> 
> Currently, there's a silent (and reasonable) assumption that the UAC
> issuing the first request inserts a RR. 

You mean insert or doesn't insert? UAC doesn't insert an RR....

> However, would there be a use
> for the UAS receiving that request doing this? Clearly, this would only
> make sense if that request isn't routed directly, i.e., typically, if
> the first INVITE didn't contain a Contact header. Is this useful enough
> to be described explicitly? Any existing implementations deal with this
> gracefully?

Assuming you meant "doesn't" above, I see no need to explicitly indicate
this, since Contact more or less plays the role of the UA record-route. 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Sun Jun  4 22:57:41 2000
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Lawrence Conroy wrote:
> 
> *       UDP support changed from SHOULD to MUST in SIP 2453 bis draft.
> It looks like support for UDP is gradually moving from a SHOULD to a
> MUST in SIP.

It was to solve the sticky situation that without at least a common
transport, UA to UA communication would not be possible. It would
require proxies that do conversions to record-route, and all kinds of
other stuff. Simply stating that everyone needs to use UDP made things
much easier, and in any case, I suspect nearly everyone does UDP anyway.
However, please shout if this change is not a happy one for you....

In the case of pint, I would simply suggest that all pint
implementations MUST support TCP as well, in which case you won't have a
problem. Yes, you'll have to do UDP also to be compliant to the baseline
spec, but it turns out that supporting both is really not that hard if
done properly.

> I had the same feeling at the Interim meeting in D.C. when the
> subject of "Busy Line Verification" was raised - there's a lot of
> focus on VoIP, when I can easily imagine SIP being used for a much
> wider set of applications even for "straight" conferences; for
> example, Quake, as Dean suggests.


This change has exactly zero to do with telephony. Its as simple as UA
to UA communication not being always possible without a common baseline
transport.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Dean Willis wrote:
> 
> I suspect I don't like what you're attempting to achieve, but methinks
> the friendly UA could always submit an appropriate INFO method which the
> "billing proxy" could use to determine an appropriate action.

May as well use a new method, and call it FREECALL ;)

Seriously, though, developing extensions with known monstorously large
security holes doesn't help anyone. If a new method were defined, people
will think that it is generally useful and apply it in cases where a
trust relationship between the UA and proxy does not exist. Counting on
friednly UAs is like counting on people not to send spam or generate
viruses....

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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"Schroeder, Tim" wrote:
> 
> > > Section 5.1, Paragraph 11: "It is not necessary to include the Supported
> > > header listing the option tag 100rel in the PRACK request."
> 
> > Given the change that you can't send a reliable provisional response to
> > PRACK, it seems this text is actually fine - its not needed to include
> > this Supported header in PRACK.
> 
> Given the change that you can't send a reliable provisional response to
> PRACK, perhaps the current text could remain, and an additional sentence
> prohibiting the use of "Require: 100rel" in a PRACK request should be added.
> Otherwise, somewhere we'd probably have to clarify what to do if a UAC
> requires reliable provisional responses, but the UAS is prohibited from
> sending them reliably.

I added this wording.

> 
> I'm still in the "no reliable provisional responses to a PRACK" camp.

Well, since there were two folks speaking against it, and I spoke for it
but was on the fence, consensus seemed to be to remove it, so its gone -
issue resolved.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Mon Jun  5 06:56:01 2000
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Hi,

I believe that the first point in "Appendix G: Changes to be made" is
invalid and should be removed. Section 6.45.3 and 6.45.4 [Handling of Via's
in requests] are in contradiction to this statement:

* Bug: UDP spec not clear on whether responses sent to source port or port
in Via. Should use port in Via. 

I am guessing that this is merely an oversight as Section 10.2.1 had a
similar statement which was changed in the last draft version to agree with
Section 6.45.3 & 4.

Robert.


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Date: Mon, 5 Jun 2000 12:56:22 +0200
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This is a multi-part message in MIME format.

------=_NextPart_000_0050_01BFCEED.7379DB00
Content-Type: text/plain;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

Hi, I need of a clarification about SIP. I don`t know if these addresses =
are the right places!!!
I am developing a NAT module properly for SIP. I have two scenarios: the =
first with a SIP Client
behind the NAT router (In a private realm), the second with a SIP Client =
and a SIP SERVER behind
a NAT router both. My NAT Module is listening on port 5060 UDP and it =
works when a UDP packet port=20
5060 flows throgh the NAT router. This module works well when it is =
crossed by an Invite directly produced by
a SIP Client. Viceversa it doesn`t work when it is crossed by an Invite =
produced by a SIP Server. The two Invites are different
in the context: In SDP and in some fileds like VIA!!!=20
Are there other differences ????=20
My module works only on the UDP header
and in the case of port 5060 processing SDP and change prvate =
addresses!!!
Thans a lot,
    Raffaele Pellicciotta

------=_NextPart_000_0050_01BFCEED.7379DB00
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2314.1000" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>
<DIV><FONT face=3DArial size=3D2>Hi, I need of a clarification about =
SIP. I don`t=20
know if these addresses are the right places!!!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I am developing&nbsp;a NAT module =
properly for SIP.=20
I have two scenarios: the first with&nbsp;a SIP Client</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>behind the NAT router (In a private =
realm), the=20
second with a SIP Client and a SIP SERVER behind</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>a NAT router both. My NAT Module is =
listening on=20
port 5060 UDP and it works when a UDP packet port </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>5060 flows throgh the NAT router. This =
module works=20
well when it is crossed by an Invite directly produced by</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>a SIP Client. Viceversa it doesn`t work =
when it is=20
crossed by an Invite produced by a SIP Server. The two Invites are=20
different</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>in the context: In SDP and in some =
fileds like=20
VIA!!! </FONT></DIV>
<DIV><FONT face=3DArial size=3D2><STRONG><U>Are there other differences =
????=20
</U></STRONG></FONT></DIV>
<DIV><FONT face=3DArial size=3D2>My module works only on the UDP =
header</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>and in the case of port 5060 processing =
SDP and=20
change prvate addresses!!!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Thans&nbsp;a lot,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; Raffaele=20
Pellicciotta</FONT></DIV></FONT></DIV></BODY></HTML>

------=_NextPart_000_0050_01BFCEED.7379DB00--


__________________________________________________
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From sip-admin@lists.bell-labs.com  Mon Jun  5 07:50:44 2000
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Subject: RE: [SIP] Via discrepancy in latest 2543bis draft.
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Wait up ... quick back pedal ... now I'm not so sure. As written in Section
6.45.4 it does suggest that you would use the source port over the Via port
(as the "sent-by" includes the port part) but is this the best thing to do ?
We know this Via scheme won't work for a NAPT (without a SIP proxy sitting
on the firewall as well) so perhaps we should be using the port in the Via.
This at least allows SIP user-agent spawning to function normally across a
NAT.

Anyway, there is definitely a need for clarification here.

Cheers,

Robert. 


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From sip-admin@lists.bell-labs.com  Mon Jun  5 09:54:54 2000
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From: Gethin Liddell <gethin@ubiquity.net>
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Hi,

I have noticed an inconsistency between the rfc you have authored 
(RFC 2806 URL's for Telephone Calls) and the SIP RFC.

In Section 2.2, you reference the SIP RFC within the BNF for 
future-extension as follows:

future-extension      = ";" 1*(token-char) ["=" ((1*(token-char)
                        ["?" 1*(token-char)]) / quoted-string )]
                        ; See section 2.5.11 and [RFC2543]

My concern is over the quoted-string part of this.

We recently had a long discussion on the Sip Mailing list and 
concluded (among other things :) )that quoted strings should 
not be a part of the Sip-URL as it contravines the URI syntax 
outlined in RFC 2396 (See section 2.4.3 of RFC 2396
where it declares the <"> character as excluded).

In fact, the new SIP BNF currently stands at:

other-param = pname  | pname  =  pvalue
pname = 1*paramchar
pvalue = 1*paramchar
paramchar = param-unreserved | unreserved | escaped
param-unreserved =  /  |  :  |  @  |  &  |  +  |  $  |  , 

could you please let me know if you agree with this conclusion.

thanx for your time

gethin

-- 
Gethin Liddell
Ubiquity Software Corporation

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


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From sip-admin@lists.bell-labs.com  Mon Jun  5 10:36:09 2000
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From: "Farahmand, Fardad" <FFarahmand@sonusnet.com>
To: Henry Sinnreich <Henry.Sinnreich@WCom.com>
Cc: sip@lists.bell-labs.com, Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        Neil Deason <ndeason@ubiquity.net>
Subject: RE: [SIP] SIP Application Server
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I agree. There is no question that it will be the potential and ease of
creating and deploying new services that will be needed to initiate the mass
migration to the next generation network.  I find it hard to believe that
there would be folks that don't think this way.

However, I do believe that for a long time to come, the new network must
support integration with existing device, service functionality and network
infrastructure. You can think of this as the bridge that will bring about
the migration. As Jonathan mentioned, the strategy in application
development can be to provide as many features as a particular device is
capable of handling.  The SoftSwitch with its SIP interface can provide
network transparency where it is needed.  

Fardad Farahmand
ffarahmand@sonusnet.com

> -----Original Message-----
> From: Henry Sinnreich [mailto:Henry.Sinnreich@WCom.com]
> Sent: Friday, June 02, 2000 8:43 PM
> To: Jonathan Rosenberg; Neil Deason
> Cc: sip@lists.bell-labs.com
> Subject: RE: [SIP] SIP Application Server
> 
> 
> I fully agree with Jonathan. The whole point is not just to
> replicate what can be already done with AIN/PSTN, but for new
> services that produce new revenue.
> 
> Henry
> 
> Henry Sinnreich
> MCI WorldCom
> 400 International Parkway
> Richardson, Texas 75081
> USA
> 
> >-----Original Message-----
> >From: sip-admin@lists.bell-labs.com
> >[mailto:sip-admin@lists.bell-labs.com]On Behalf Of
> >Jonathan Rosenberg
> >Sent: Friday, June 02, 2000 8:35 AM
> >To: Neil Deason
> >Cc: sip@lists.bell-labs.com
> >Subject: Re: [SIP] SIP Application Server
> >
> >
> >
> >
> >Neil Deason wrote:
> >>
> >> > I think you are over-analyzing the meaning of separation of
> >> > call-signaling from service control. In reality,
> >all services are
> >> > delivered eventually through signaling, and are
> >based on information
> >> > provided by signaling. Thus, service logic is
> >quite connected to call
> >> > signaling, and embedding a vanilla proxy in an app
> >server is quite a
> >> > good way to access it. I would argue this is quite
> >contrary to the IN
> >> > model, in which there is an attempt to use a
> >separate protocol for
> >> > service control to a separate box. Note however
> >that these protocols
> >> > carry much the same information as the signaling
> >protocols themselves.
> >>
> >> Agreed. It was not my analysis that SIP should be
> >used to try and separate out service control
> >> from call control. However, as was pointed out
> >earlier in this thread this could be achieved if
> >> SIP is used to separate a non-SIP signalling
> >protocol from service logic in a 'Softswitch'
> >> network. But then it appears that you would have a
> >separate protocol (SIP) for service control to
> >> a separate box (App Server). Such separation doesn't
> >really apply in the same way to the SIP part
> >> of the network and it sounds a lot like the
> >Softswitch network is a SIP network at the core.
> >> There you find SIP conversant App Servers and around
> >them are SIP Proxy Servers, SIP Redirect
> >> Servers and SIP User Agents. Some of these UAs may
> >in fact represent interfaces to PSTN or even
> >> H.323 gateways. Though how do protocols that don't
> >use SIP for signalling deliver all the rich
> >> new services that SIP enables when, for example, a
> >text/html payload gets discarded at the
> >> SIP/foo gateway. The point of SIP is surely not to
> >create LNP, 800 translation services in an App
> >> Server for protocols such as SS7, H.323 etc Yes, you
> >could recreate the IN this way but why not
> >> start out intending to build something better. You
> >achieve this by building Application Servers
> >> that empower the simple creation of innovative
> >services not Application Servers that merely
> >> recreate old services.
> >
> >I couldn't agree more with this sentiment.
> >
> >That said, I have a sneaking suspicion that people
> >will still want to
> >build these older services. Fortunately, most are
> >supported by SIP, and
> >can easily be built ontop of an applications server as
> >well. These
> >services (things like LNP and freephone) will work for
> >H.323 devices,
> >PSTN phones or ISDN phones accessing the SIP network
> >through a gateway
> >or softswitch. Even better, SIPs capabilities for negotiating
> >extensions, content, methods, and so on, will often
> >allow an app server
> >to know if the accessing device can support a more
> >innovative service.
> >In the example you give, where the app server returns HTML to a
> >softswitch, the app server can know apriori that HTML
> >is or isn't
> >supported (Accept: text/html in the INVITE). It could
> >then do something
> >different with the request for non-html access
> >devices, like forward it
> >to a media server running TTS or something, and read
> >out the web page.
> >
> >-Jonathan R.
> >--
> >Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> >Chief Scientist                             First Floor
> >dynamicsoft                                 East
> >Hanover, NJ 07936
> >jdrosen@dynamicsoft.com                     FAX:
> >(973) 952-5050
> >http://www.cs.columbia.edu/~jdrosen         PHONE:
> >(732) 741-7244
> >http://www.dynamicsoft.com
> >
> >
> >_______________________________________________
> >SIP mailing list
> >SIP@lists.bell-labs.com
> >http://lists.bell-labs.com/mailman/listinfo/sip
> 
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
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> 


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From sip-admin@lists.bell-labs.com  Mon Jun  5 11:34:14 2000
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From: "Oughton, Andre" <aoughton@comversens.com>
To: sip@lists.bell-labs.com
Subject: [SIP] Overview Text
Date: Mon, 5 Jun 2000 11:33:32 -0400 
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I am new to the SIP mailing list. I am looking for a good
overview text, one that I might recommend as a reference in
a college course on IP based telephony. The text need not
be limited SIP but it must describe SIP in the context of
general networking. An ideal text would start with 
fundamental IP concepts and delve into the specifics associated
with IP based streaming etc ....
Thanks in advance. 
 
---------------------------
Andre R. Oughton
Principal Engineer
Comverse Network Systems
Andover MA
01810
http://www.Comversens.com <http://www.Comversens.com >   
 
---------------------------
 


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"Oughton, Andre" wrote:
> 
> I am new to the SIP mailing list. I am looking for a good
> overview text, one that I might recommend as a reference in
> a college course on IP based telephony. The text need not
> be limited SIP but it must describe SIP in the context of
> general networking. An ideal text would start with
> fundamental IP concepts and delve into the specifics associated
> with IP based streaming etc ....
> Thanks in advance.

"Internetworking Multimedia" by Jon Crowcroft, Mark Handley, and
Ian Wakeman is pretty good.  It contains chapters detailing multi-
casting, coding, compression, SAP/SDP, conference control and of course,
security.  It also covers SIP, although the text identifies it as the
"Section Initiation Protocol" (most definitely a typo) and in at least
one of the call flows I have glanced at, an ACK is sent to a BYE
request (this is wrong; the RFC mandates using ACKs for INVITEs only).

Regards,

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
IN Architecture and Internet Software Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184


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> raf wrote:
> 
> Hi, I need of a clarification about SIP. I don`t know if these
> addresses are the right places!!!
> I am developing a NAT module properly for SIP. I have two scenarios:
> the first with a SIP Client
> behind the NAT router (In a private realm), the second with a SIP
> Client and a SIP SERVER behind
> a NAT router both. My NAT Module is listening on port 5060 UDP and it
> works when a UDP packet port
> 5060 flows throgh the NAT router. This module works well when it is
> crossed by an Invite directly produced by
> a SIP Client. Viceversa it doesn`t work when it is crossed by an
> Invite produced by a SIP Server. The two Invites are different
> in the context: In SDP and in some fileds like VIA!!!

The SDP should be the same unless the server has rewritten it for NAT
traversal. 

For info on how to do NAT with SIP, see:
http://search.ietf.org/internet-drafts/draft-rosenberg-sip-firewalls-00.txt
http://search.ietf.org/internet-drafts/draft-biggs-sip-nat-00.txt

-JOnathan R.


-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Mon Jun  5 12:36:05 2000
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http://www.cs.columbia.edu/~hgs/sip/papers.html has a listing of books
and papers. Also note:

IP Telephony 
     Olivier Hersent, David Gurle, Jean-Pierre Petit. Addison-Wesley,
2000. ISBN 0-201-61910-5. 

which is a book on the subject.

-Jonathan R.

"Oughton, Andre" wrote:
> 
> I am new to the SIP mailing list. I am looking for a good
> overview text, one that I might recommend as a reference in
> a college course on IP based telephony. The text need not
> be limited SIP but it must describe SIP in the context of
> general networking. An ideal text would start with
> fundamental IP concepts and delve into the specifics associated
> with IP based streaming etc ....
> Thanks in advance.
> 
> ---------------------------
> Andre R. Oughton
> Principal Engineer
> Comverse Network Systems
> Andover MA
> 01810
> http://www.Comversens.com <http://www.Comversens.com >
> 
> ---------------------------
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Mon Jun  5 14:01:46 2000
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From: Linden deCarmo <ldeCarmo@netspeak.com>
To: "'Dean Willis'" <dwillis@dynamicsoft.com>
Cc: "'Glenn Morrow'" <gmorrow@nortelnetworks.com>, sip@lists.bell-labs.com
Subject: RE: [SIP] Suspend a session??
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Ok, you've forced to to evaluate what I'm really requesting.  If you throw
out my billing/on hook/off hook application, and examine the root
functionality I'd like to see added, it boils down to extended transactional
roll backs.

Basically, for certain methods, it would be convenient if the requesting
entity could request the amount of time a CANCEL could roll back the
transaction.  The recipient of the request would determine if it supported
extended roll backs for the method in question.

If it supported these extended roll backs, it would send a provisional
(18x-type) response to acknowlege the receipt of the request.  After the
specified time period passed, a final response (2xx) could be issued.
Should a CANCEL be received during this intermediate time period, the
request would be cancelled.

If the UA didn't support extended roll backs, it could immediately return a
final response (2xx) so the solution is completely optional and backwards
compatible.

If you apply this solution to my scenario, here's how it would map out:

UA A calls UA B.
B goes on hook. UA B issues BYE (with rollback: 30 seconds) to UA A.
UA A sends provisional response 18x to UA B.
B goes off hook within 30 seconds.  UA B sends CANCEL to UA A and session
continues.

scenario two:

UA A calls UA B.
B goes on hook. UA B issues BYE (with rollback: 30 seconds) to UA A.
UA A sends provisional response 18x to UA B.
B remains on hook for 30 seconds.  UA A sends final 200 OK response to BYE
and session terminates.

I don't think this solution is useful (or relevant) for an INVITE, but I
believe there are applications for it with OPTIONS, INFO, and BYE.

Linden deCarmo
Netspeak Corporation
902 Clint Moore Road
Suite 104
Boca Raton, FL 33487

> -----Original Message-----
> From:	Dean Willis [SMTP:dwillis@dynamicsoft.com]
> Sent:	Sunday, June 04, 2000 2:42 AM
> To:	Linden deCarmo
> Cc:	'Glenn Morrow'; sip@lists.bell-labs.com
> Subject:	Re: [SIP] Suspend a session??
> 
> 
> I suspect I don't like what you're attempting to achieve, but methinks
> the friendly UA could always submit an appropriate INFO method which the
> "billing proxy" could use to determine an appropriate action.
> 
> AS has been pointed out here, it is that UA that is responsible for
> translating user action into appropriate signaling.
> 
> One might also ask WHY and end device would choose to use a "time
> billing proxy" rather than routing around said proxy. There are probably
> some cases involving proxy-moderated firewalls and QoS sessions, but I
> still suspect the fundamental approach is somehow flawed.
> 
> --
> Dean
> 
> Linden deCarmo wrote:
> > 
> > Here's the problem with the UA doing it:
> > 
> > User goes on hook (time 0 ).  UA starts timer.
> > Timer expires (time 30 seconds). UA realizes session expires so it
> issues a
> > BYE.
> > 
> > Since the proxy sees the BYE 30 seconds later, thats when it stops
> billing.
> > I do not believe your solution addresses how the UA signals to the proxy
> > that billing should stop at time 0 rather than time 30.
> > 
> > I realize I can do this with a proprietary method or header, but I'd
> rather
> > not.
> > 
> > Linden deCarmo
> > Netspeak Corporation
> > 902 Clint Moore Road
> > Suite 104
> > Boca Raton, FL 33487
> > 
> > > -----Original Message-----
> > > From: Glenn Morrow [SMTP:gmorrow@nortelnetworks.com]
> > > Sent: Thursday, June 01, 2000 11:38 AM
> > > To:   sip@lists.bell-labs.com
> > > Subject:      RE: [SIP] Suspend a session??
> > >
> > > It seems to me that this should be handled by the UA itself. Once the
> UA
> > > determines the correct functionilty that the real end-device/user
> wishes
> > > to achieve, it should signal proxies and far-end stations. For
> instance a
> > > legacy device may which to indicate flash and collect more digits for
> a
> > > 3-way call by the user using the hook as you say.
> > >
> > > This would result in a separate transaction but I believe it doesn't
> > > require something like a  "suspend" method or indication at the SIP
> level.
> > >
> > >
> > > I suggest that the UA itself should handle this with timers, etc..
> based
> > > upon the physical device and human interface it is trying to achieve.
> > >
> > > For the billing issue the UA could simply use a timer started based on
> > > physical off-hook stimulus to determine when to actually send a BYE or
> > > not.
> > >
> > > If this doesn't address your concerns please let me know.
> > >
> > >       -----Original Message-----
> > > From:   Linden deCarmo [SMTP:ldeCarmo@netspeak.com]
> > > Sent:   Wednesday, May 31, 2000 1:54 PM
> > > To:     sip@lists.bell-labs.com
> > > Subject:        [SIP] Suspend a session??
> > >
> > >       Right now, SIP smoothly handles one party putting another party
> on
> > > hold via
> > > a re-INVITE with modified SDP.  However, we have a scenario where the
> > > session needs to be suspended (not just placed on hold), and I'm not
> sure
> > > how this can be done without using proprietary headers.
> > >
> > >       Here are ther details.  If party A calls party B , party B
> should be
> > > able to
> > > hang up without immediately terminating the session.  If party B picks
> up
> > > the phone within some prescribed time window, the session continues.
> > > However, if they don't pick it up within the time window, the session
> > > terminates (via a BYE).
> > >
> > >       We're looking for a mechanism for party B to communicate with
> either
> > > party A
> > > or a proxy server, to inform the distant party that the session has
> been
> > > suspended, and not just placed on hold.  The primary motivator for
> this
> > > notification is billing.  Without notification of suspension, its
> > > impossible
> > > to determine when to safely stop billing.
> > >
> > >       Thanks.
> > >
> > >       Linden deCarmo
> > > Netspeak Corporation
> > > 902 Clint Moore Road
> > > Suite 104
> > > Boca Raton, FL 33487
> > >
> > >
> > >
> > >       _______________________________________________
> > > SIP mailing list
> > > SIP@lists.bell-labs.com
> > > <http://lists.bell-labs.com/mailman/listinfo/sip>
> > >
> > 
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 
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What is the proper behavior of a Proxy in the following scenarios:

Scenario1:

OUA-----INVITE----->PS------INVITE----->TUA
OUA2-----CANCEL---->PS

In other words - the CANCEL comes from a different address than the original
INVITE came from.  The same question applies to a "retransmitted" INVITE
request prior to the final response.


Scenario2:

OUA---INVITE---->PS-----INVITE---->TUA
TUA2--->200--->PS

In other words - the final response (or any provisional response) comes from
a different address than the one that the INVITE was sent to.



I know that AFTER the final response, any request can legitimately come from
a different source than the original request, but what is the proper
behavior when this situation occurs PRIOR to the final response?


Thanks,
Kathleen McMurry



--Boundary_(ID_jDlYEneD8akt3t9Xj1VDxQ)
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Content-Transfer-Encoding: quoted-printable

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<HTML>
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<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3DISO-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2652.35">
<TITLE>Request/Responses from other addresses</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2 FACE=3D"Arial">What is the proper behavior of a Proxy =
in the following scenarios:</FONT>
</P>

<P><U><B><FONT SIZE=3D2 =
FACE=3D"Arial">Scenario1</FONT></B></U><B></B><FONT SIZE=3D2 =
FACE=3D"Arial">:</FONT>
</P>

<P><FONT SIZE=3D2 =
FACE=3D"Arial">OUA-----INVITE-----&gt;PS------INVITE-----&gt;TUA</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">OUA2-----CANCEL----&gt;PS</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">In other words - the CANCEL comes from =
a different address than the original INVITE came from.&nbsp; The same =
question applies to a &quot;retransmitted&quot; INVITE request prior to =
the final response.</FONT></P>
<BR>

<P><U><B><FONT SIZE=3D2 FACE=3D"Arial">Scenario2:</FONT></B></U><B></B>
</P>

<P><FONT SIZE=3D2 =
FACE=3D"Arial">OUA---INVITE----&gt;PS-----INVITE----&gt;TUA</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">TUA2---&gt;200---&gt;PS</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">In other words - the final response =
(or any provisional response) comes from a different address than the =
one that the INVITE was sent to.</FONT></P>
<BR>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Arial">I know that AFTER the final response, =
any request can legitimately come from a different source than the =
original request, but what is the proper behavior when this situation =
occurs PRIOR to the final response?</FONT></P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Arial">Thanks,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Kathleen McMurry</FONT>
</P>
<BR>

</BODY>
</HTML>

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> 
> Wait up ... quick back pedal ... now I'm not so sure. As written in Section
> 6.45.4 it does suggest that you would use the source port over the Via port
> (as the "sent-by" includes the port part) but is this the best thing to do ?
> We know this Via scheme won't work for a NAPT (without a SIP proxy sitting
> on the firewall as well) so perhaps we should be using the port in the Via.
> This at least allows SIP user-agent spawning to function normally across a
> NAT.
> 
> Anyway, there is definitely a need for clarification here.
> Cheers,
> 
> Robert. 
>

Definitely confusing. Since 6.45.4 references 6.45.2 which describes the receiver-tagged
Via:, I would assume that you would use the source IP address, BUT, use the port
specified in the sent-by, since this is basically the behaviour specified for proxies
that receive a request from a different address than the top-most Via: specifies.

sean
--
Sean Olson <sean.olson@ericsson.com>
 
> 
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From sip-admin@lists.bell-labs.com  Mon Jun  5 15:32:54 2000
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To: sip@lists.bell-labs.com, Kathleen.McMurry@wcom.com
Subject: Re: [SIP] Request/Responses from other addresses
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> What is the proper behavior of a Proxy in the following scenarios:
> 
> Scenario1:
> 
> OUA-----INVITE----->PS------INVITE----->TUA
> OUA2-----CANCEL---->PS
> 
> In other words - the CANCEL comes from a different address than the original
> INVITE came from.  The same question applies to a "retransmitted" INVITE
> request prior to the final response.
> 
> 
> Scenario2:
> 
> OUA---INVITE---->PS-----INVITE---->TUA
> TUA2--->200--->PS
> 
> In other words - the final response (or any provisional response) comes from
> a different address than the one that the INVITE was sent to.
> 
> I know that AFTER the final response, any request can legitimately come from
> a different source than the original request, but what is the proper
> behavior when this situation occurs PRIOR to the final response?
> 
> 
> Thanks,
> Kathleen McMurry

By different address you mean different source IP address, I'm assuming(?)
(A different From: of course would not cause a problem -- the proxy could return a 481)

I see no reason to state absolutely that the CANCEL or final response MUST come from 
the same IP address. There are obvious applications for exactly this scenario. 

What you want (I assume) is a way to verify/authenticate/authorize the request or the 
response. If your proxy cares about these things, you can use a variety of schemes 
to sign the requests/responses, use transport level security, etc. From a SIP 
point of view, it shouldn't really matter. Consider an analogous situation where
the request/response comes from a different source port. Would you outright disallow
such a thing? If you are only dealing with very simple UAs, then this might make sense.
I have no doubt, however, that you will find some UAs out there that occupy multiple
IP address/ports/subnets. This is really, then, a policy decision, and not a protocol
issue.

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



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From: "Glenn Morrow" <gmorrow@nortelnetworks.com>
To: "'Linden deCarmo'" <ldeCarmo@netspeak.com>,
        "'Dean Willis'" <dwillis@dynamicsoft.com>
Cc: sip@lists.bell-labs.com
Subject: RE: [SIP] Suspend a session??
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From earlier posts it appears that these topics are still a bit too "taboo"
for discussion on the list.  There are many ways the functionality in
question can be accomplished. Perhaps we should leave it at that?

> -----Original Message-----
> From:	Linden deCarmo [SMTP:ldeCarmo@netspeak.com]
> Sent:	Monday, June 05, 2000 1:02 PM
> To:	'Dean Willis'
> Cc:	Morrow, Glenn [RICH2:C301:EXCH]; sip@lists.bell-labs.com
> Subject:	RE: [SIP] Suspend a session??
> 
> Ok, you've forced to to evaluate what I'm really requesting.  If you throw
> out my billing/on hook/off hook application, and examine the root
> functionality I'd like to see added, it boils down to extended
> transactional
> roll backs.
> 
> Basically, for certain methods, it would be convenient if the requesting
> entity could request the amount of time a CANCEL could roll back the
> transaction.  The recipient of the request would determine if it supported
> extended roll backs for the method in question.
> 
> If it supported these extended roll backs, it would send a provisional
> (18x-type) response to acknowlege the receipt of the request.  After the
> specified time period passed, a final response (2xx) could be issued.
> Should a CANCEL be received during this intermediate time period, the
> request would be cancelled.
> 
> If the UA didn't support extended roll backs, it could immediately return
> a
> final response (2xx) so the solution is completely optional and backwards
> compatible.
> 
> If you apply this solution to my scenario, here's how it would map out:
> 
> UA A calls UA B.
> B goes on hook. UA B issues BYE (with rollback: 30 seconds) to UA A.
> UA A sends provisional response 18x to UA B.
> B goes off hook within 30 seconds.  UA B sends CANCEL to UA A and session
> continues.
> 
> scenario two:
> 
> UA A calls UA B.
> B goes on hook. UA B issues BYE (with rollback: 30 seconds) to UA A.
> UA A sends provisional response 18x to UA B.
> B remains on hook for 30 seconds.  UA A sends final 200 OK response to BYE
> and session terminates.
> 
> I don't think this solution is useful (or relevant) for an INVITE, but I
> believe there are applications for it with OPTIONS, INFO, and BYE.
> 
> Linden deCarmo
> Netspeak Corporation
> 902 Clint Moore Road
> Suite 104
> Boca Raton, FL 33487
> 
> > -----Original Message-----
> > From:	Dean Willis [SMTP:dwillis@dynamicsoft.com]
> > Sent:	Sunday, June 04, 2000 2:42 AM
> > To:	Linden deCarmo
> > Cc:	'Glenn Morrow'; sip@lists.bell-labs.com
> > Subject:	Re: [SIP] Suspend a session??
> > 
> > 
> > I suspect I don't like what you're attempting to achieve, but methinks
> > the friendly UA could always submit an appropriate INFO method which the
> > "billing proxy" could use to determine an appropriate action.
> > 
> > AS has been pointed out here, it is that UA that is responsible for
> > translating user action into appropriate signaling.
> > 
> > One might also ask WHY and end device would choose to use a "time
> > billing proxy" rather than routing around said proxy. There are probably
> > some cases involving proxy-moderated firewalls and QoS sessions, but I
> > still suspect the fundamental approach is somehow flawed.
> > 
> > --
> > Dean
> > 
> > Linden deCarmo wrote:
> > > 
> > > Here's the problem with the UA doing it:
> > > 
> > > User goes on hook (time 0 ).  UA starts timer.
> > > Timer expires (time 30 seconds). UA realizes session expires so it
> > issues a
> > > BYE.
> > > 
> > > Since the proxy sees the BYE 30 seconds later, thats when it stops
> > billing.
> > > I do not believe your solution addresses how the UA signals to the
> proxy
> > > that billing should stop at time 0 rather than time 30.
> > > 
> > > I realize I can do this with a proprietary method or header, but I'd
> > rather
> > > not.
> > > 
> > > Linden deCarmo
> > > Netspeak Corporation
> > > 902 Clint Moore Road
> > > Suite 104
> > > Boca Raton, FL 33487
> > > 
> > > > -----Original Message-----
> > > > From: Glenn Morrow [SMTP:gmorrow@nortelnetworks.com]
> > > > Sent: Thursday, June 01, 2000 11:38 AM
> > > > To:   sip@lists.bell-labs.com
> > > > Subject:      RE: [SIP] Suspend a session??
> > > >
> > > > It seems to me that this should be handled by the UA itself. Once
> the
> > UA
> > > > determines the correct functionilty that the real end-device/user
> > wishes
> > > > to achieve, it should signal proxies and far-end stations. For
> > instance a
> > > > legacy device may which to indicate flash and collect more digits
> for
> > a
> > > > 3-way call by the user using the hook as you say.
> > > >
> > > > This would result in a separate transaction but I believe it doesn't
> > > > require something like a  "suspend" method or indication at the SIP
> > level.
> > > >
> > > >
> > > > I suggest that the UA itself should handle this with timers, etc..
> > based
> > > > upon the physical device and human interface it is trying to
> achieve.
> > > >
> > > > For the billing issue the UA could simply use a timer started based
> on
> > > > physical off-hook stimulus to determine when to actually send a BYE
> or
> > > > not.
> > > >
> > > > If this doesn't address your concerns please let me know.
> > > >
> > > >       -----Original Message-----
> > > > From:   Linden deCarmo [SMTP:ldeCarmo@netspeak.com]
> > > > Sent:   Wednesday, May 31, 2000 1:54 PM
> > > > To:     sip@lists.bell-labs.com
> > > > Subject:        [SIP] Suspend a session??
> > > >
> > > >       Right now, SIP smoothly handles one party putting another
> party
> > on
> > > > hold via
> > > > a re-INVITE with modified SDP.  However, we have a scenario where
> the
> > > > session needs to be suspended (not just placed on hold), and I'm not
> > sure
> > > > how this can be done without using proprietary headers.
> > > >
> > > >       Here are ther details.  If party A calls party B , party B
> > should be
> > > > able to
> > > > hang up without immediately terminating the session.  If party B
> picks
> > up
> > > > the phone within some prescribed time window, the session continues.
> > > > However, if they don't pick it up within the time window, the
> session
> > > > terminates (via a BYE).
> > > >
> > > >       We're looking for a mechanism for party B to communicate with
> > either
> > > > party A
> > > > or a proxy server, to inform the distant party that the session has
> > been
> > > > suspended, and not just placed on hold.  The primary motivator for
> > this
> > > > notification is billing.  Without notification of suspension, its
> > > > impossible
> > > > to determine when to safely stop billing.
> > > >
> > > >       Thanks.
> > > >
> > > >       Linden deCarmo
> > > > Netspeak Corporation
> > > > 902 Clint Moore Road
> > > > Suite 104
> > > > Boca Raton, FL 33487
> > > >
> > > >
> > > >
> > > >       _______________________________________________
> > > > SIP mailing list
> > > > SIP@lists.bell-labs.com
> > > > <http://lists.bell-labs.com/mailman/listinfo/sip>
> > > >
> > > 
> > > _______________________________________________
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> > 
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<TITLE>RE: [SIP] Suspend a session??</TITLE>
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<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">From earlier posts =
it appears that these topics are still a bit too &quot;taboo&quot; for =
discussion on the list.&nbsp; There are many ways the functionality in =
question can be accomplished. Perhaps we should leave it at =
that?</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">Linden deCarmo =
[SMTP:ldeCarmo@netspeak.com]</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Monday, June 05, 2000 1:02 PM</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">'Dean Willis'</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Cc:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">Morrow, Glenn [RICH2:C301:EXCH]; =
sip@lists.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: [SIP] Suspend a session??</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Ok, you've forced to to evaluate what =
I'm really requesting.&nbsp; If you throw</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">out my billing/on hook/off hook =
application, and examine the root</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">functionality I'd like to see added, =
it boils down to extended transactional</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">roll backs.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Basically, for certain methods, it =
would be convenient if the requesting</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">entity could request the amount of =
time a CANCEL could roll back the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">transaction.&nbsp; The recipient of =
the request would determine if it supported</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">extended roll backs for the method in =
question.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">If it supported these extended roll =
backs, it would send a provisional</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">(18x-type) response to acknowlege the =
receipt of the request.&nbsp; After the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">specified time period passed, a final =
response (2xx) could be issued.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Should a CANCEL be received during =
this intermediate time period, the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">request would be cancelled.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">If the UA didn't support extended roll =
backs, it could immediately return a</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">final response (2xx) so the solution =
is completely optional and backwards</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">compatible.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">If you apply this solution to my =
scenario, here's how it would map out:</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">UA A calls UA B.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">B goes on hook. UA B issues BYE (with =
rollback: 30 seconds) to UA A.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">UA A sends provisional response 18x =
to UA B.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">B goes off hook within 30 =
seconds.&nbsp; UA B sends CANCEL to UA A and session</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">continues.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">scenario two:</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">UA A calls UA B.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">B goes on hook. UA B issues BYE (with =
rollback: 30 seconds) to UA A.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">UA A sends provisional response 18x =
to UA B.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">B remains on hook for 30 =
seconds.&nbsp; UA A sends final 200 OK response to BYE</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">and session terminates.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I don't think this solution is useful =
(or relevant) for an INVITE, but I</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">believe there are applications for it =
with OPTIONS, INFO, and BYE.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Linden deCarmo</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Netspeak Corporation</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">902 Clint Moore Road</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Suite 104</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Boca Raton, FL 33487</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; From: Dean Willis =
[SMTP:dwillis@dynamicsoft.com]</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Sent: Sunday, June 04, 2000 2:42 =
AM</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; To:&nbsp;&nbsp; Linden =
deCarmo</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Cc:&nbsp;&nbsp; 'Glenn Morrow'; =
sip@lists.bell-labs.com</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; =
Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Re: [SIP] Suspend a =
session??</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; I suspect I don't like what =
you're attempting to achieve, but methinks</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; the friendly UA could always =
submit an appropriate INFO method which the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &quot;billing proxy&quot; could =
use to determine an appropriate action.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; AS has been pointed out here, it =
is that UA that is responsible for</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; translating user action into =
appropriate signaling.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; One might also ask WHY and end =
device would choose to use a &quot;time</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; billing proxy&quot; rather than =
routing around said proxy. There are probably</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; some cases involving =
proxy-moderated firewalls and QoS sessions, but I</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; still suspect the fundamental =
approach is somehow flawed.</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; Dean</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; Linden deCarmo wrote:</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; Here's the problem with the =
UA doing it:</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; User goes on hook (time 0 =
).&nbsp; UA starts timer.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; Timer expires (time 30 =
seconds). UA realizes session expires so it</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; issues a</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; BYE.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; Since the proxy sees the =
BYE 30 seconds later, thats when it stops</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; billing.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; I do not believe your =
solution addresses how the UA signals to the proxy</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; that billing should stop at =
time 0 rather than time 30.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; I realize I can do this =
with a proprietary method or header, but I'd</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; rather</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; not.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; Linden deCarmo</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; Netspeak Corporation</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; 902 Clint Moore Road</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; Suite 104</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; Boca Raton, FL 33487</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; -----Original =
Message-----</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; From: Glenn Morrow =
[SMTP:gmorrow@nortelnetworks.com]</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; Sent: Thursday, June =
01, 2000 11:38 AM</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; To:&nbsp;&nbsp; =
sip@lists.bell-labs.com</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; =
Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RE: [SIP] Suspend a =
session??</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; It seems to me that =
this should be handled by the UA itself. Once the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; UA</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; determines the correct =
functionilty that the real end-device/user</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; wishes</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; to achieve, it should =
signal proxies and far-end stations. For</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; instance a</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; legacy device may =
which to indicate flash and collect more digits for</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; a</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; 3-way call by the user =
using the hook as you say.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; This would result in a =
separate transaction but I believe it doesn't</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; require something like =
a&nbsp; &quot;suspend&quot; method or indication at the SIP</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; level.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; I suggest that the UA =
itself should handle this with timers, etc..</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; based</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; upon the physical =
device and human interface it is trying to achieve.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; For the billing issue =
the UA could simply use a timer started based on</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; physical off-hook =
stimulus to determine when to actually send a BYE or</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; not.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; If this doesn't =
address your concerns please let me know.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; -----Original =
Message-----</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; From:&nbsp;&nbsp; =
Linden deCarmo [SMTP:ldeCarmo@netspeak.com]</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; Sent:&nbsp;&nbsp; =
Wednesday, May 31, 2000 1:54 PM</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; =
To:&nbsp;&nbsp;&nbsp;&nbsp; sip@lists.bell-labs.com</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; =
Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; [SIP] Suspend a =
session??</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Right now, SIP smoothly =
handles one party putting another party</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; on</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; hold via</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; a re-INVITE with =
modified SDP.&nbsp; However, we have a scenario where the</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; session needs to be =
suspended (not just placed on hold), and I'm not</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; sure</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; how this can be done =
without using proprietary headers.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Here are ther details.&nbsp; =
If party A calls party B , party B</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; should be</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; able to</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; hang up without =
immediately terminating the session.&nbsp; If party B picks</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; up</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; the phone within some =
prescribed time window, the session continues.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; However, if they don't =
pick it up within the time window, the session</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; terminates (via a =
BYE).</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; We're looking for a mechanism =
for party B to communicate with</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; either</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; party A</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; or a proxy server, to =
inform the distant party that the session has</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; been</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; suspended, and not =
just placed on hold.&nbsp; The primary motivator for</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; this</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; notification is =
billing.&nbsp; Without notification of suspension, its</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; impossible</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; to determine when to =
safely stop billing.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Thanks.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Linden deCarmo</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; Netspeak =
Corporation</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; 902 Clint Moore =
Road</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; Suite 104</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt; Boca Raton, FL =
33487</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; &gt;</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">&gt; &gt; =
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
_______________________________________________</FONT>
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With the more general formulation, I immediately have the question -
why? If the application is not for this billing thing, what is this
useful for? I don't see why the originating UA could not simply wait the
required time and then issue the request.

-Jonathan R.

Linden deCarmo wrote:
> 
> Ok, you've forced to to evaluate what I'm really requesting.  If you throw
> out my billing/on hook/off hook application, and examine the root
> functionality I'd like to see added, it boils down to extended transactional
> roll backs.
> 
> Basically, for certain methods, it would be convenient if the requesting
> entity could request the amount of time a CANCEL could roll back the
> transaction.  The recipient of the request would determine if it supported
> extended roll backs for the method in question.
> 
> If it supported these extended roll backs, it would send a provisional
> (18x-type) response to acknowlege the receipt of the request.  After the
> specified time period passed, a final response (2xx) could be issued.
> Should a CANCEL be received during this intermediate time period, the
> request would be cancelled.
> 
> If the UA didn't support extended roll backs, it could immediately return a
> final response (2xx) so the solution is completely optional and backwards
> compatible.
> 
> If you apply this solution to my scenario, here's how it would map out:
> 
> UA A calls UA B.
> B goes on hook. UA B issues BYE (with rollback: 30 seconds) to UA A.
> UA A sends provisional response 18x to UA B.
> B goes off hook within 30 seconds.  UA B sends CANCEL to UA A and session
> continues.
> 
> scenario two:
> 
> UA A calls UA B.
> B goes on hook. UA B issues BYE (with rollback: 30 seconds) to UA A.
> UA A sends provisional response 18x to UA B.
> B remains on hook for 30 seconds.  UA A sends final 200 OK response to BYE
> and session terminates.
> 
> I don't think this solution is useful (or relevant) for an INVITE, but I
> believe there are applications for it with OPTIONS, INFO, and BYE.
> 


-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Sean Olson wrote:
> 
> > What is the proper behavior of a Proxy in the following scenarios:
> >
> > Scenario1:
> >
> > OUA-----INVITE----->PS------INVITE----->TUA
> > OUA2-----CANCEL---->PS
> >
> > In other words - the CANCEL comes from a different address than the original
> > INVITE came from.  The same question applies to a "retransmitted" INVITE
> > request prior to the final response.
> >
> >
> > Scenario2:
> >
> > OUA---INVITE---->PS-----INVITE---->TUA
> > TUA2--->200--->PS
> >
> > In other words - the final response (or any provisional response) comes from
> > a different address than the one that the INVITE was sent to.
> >
> > I know that AFTER the final response, any request can legitimately come from
> > a different source than the original request, but what is the proper
> > behavior when this situation occurs PRIOR to the final response?
> >
> >
> > Thanks,
> > Kathleen McMurry
> 
> By different address you mean different source IP address, I'm assuming(?)
> (A different From: of course would not cause a problem -- the proxy could return a 481)
> 
> I see no reason to state absolutely that the CANCEL or final response MUST come from
> the same IP address. There are obvious applications for exactly this scenario.

For security purposes, actually, you really do need to check that the
CANCEL came from the same place as the request being cancelled (see the
minutes of the security task force on this). CANCELs cannot usefully be
authenticated directly, since they may come from proxies. That aside,
CANCEL can come from anywhere.

As for responses, I'm really not sure. The spec says nothing about this,
as we had not considered it. It clearly requires some out-of-band
communications between the entity which received the request, and the
one which sent the response. I'm not sure what the application of that
would be. That aside, there may be problems. For requests sent over TCP,
the response is supposed to go over the same connection. Clearly it
cannot if the response is generated somewhere else. I suspect some
proxies may be upset if the connection to the server is open, and the
response comes on a different connection, opened just to send the
response. In the case of UDP, this will probably work, but you never
know. 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Sean Olson wrote:
> 
> >
> > Wait up ... quick back pedal ... now I'm not so sure. As written in Section
> > 6.45.4 it does suggest that you would use the source port over the Via port
> > (as the "sent-by" includes the port part) but is this the best thing to do ?
> > We know this Via scheme won't work for a NAPT (without a SIP proxy sitting
> > on the firewall as well) so perhaps we should be using the port in the Via.
> > This at least allows SIP user-agent spawning to function normally across a
> > NAT.
> >
> > Anyway, there is definitely a need for clarification here.
> > Cheers,
> >
> > Robert.
> >
> 
> Definitely confusing. Since 6.45.4 references 6.45.2 which describes the receiver-tagged
> Via:, I would assume that you would use the source IP address, BUT, use the port
> specified in the sent-by, since this is basically the behaviour specified for proxies
> that receive a request from a different address than the top-most Via: specifies.

Correct. The response is sent to the IP address the request came from,
and the port in the Via header. We will clean up to make this
consistent.

Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From: "McMurry, Kathleen" <Kathleen.McMurry@wcom.com>
Subject: RE: [SIP] Request/Responses from other addresses
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-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Monday, June 05, 2000 4:26 PM
To: Sean Olson
Cc: sip@lists.bell-labs.com; McMurry, Kathleen
Subject: Re: [SIP] Request/Responses from other addresses




Sean Olson wrote:
> > 
> > > What is the proper behavior of a Proxy in the following scenarios:
> > >
> > > Scenario1:
> > >
> > > OUA-----INVITE----->PS------INVITE----->TUA
> > > OUA2-----CANCEL---->PS
> > >
> > > In other words - the CANCEL comes from a different address than the
original
> > > INVITE came from.  The same question applies to a "retransmitted"
INVITE
> > > request prior to the final response.
> > >
> > >
> > > Scenario2:
> > >
> > > OUA---INVITE---->PS-----INVITE---->TUA
> > > TUA2--->200--->PS
> > >
> > > In other words - the final response (or any provisional response)
comes from
> > > a different address than the one that the INVITE was sent to.
> > >
> > > I know that AFTER the final response, any request can legitimately
come from
> > > a different source than the original request, but what is the proper
> > > behavior when this situation occurs PRIOR to the final response?
> > >
> > >
> > > Thanks,
> > > Kathleen McMurry
> > 
> > By different address you mean different source IP address, I'm
assuming(?)
> > (A different From: of course would not cause a problem -- the proxy
could return a 481)
> > 
> > I see no reason to state absolutely that the CANCEL or final response
MUST come from
> > the same IP address. There are obvious applications for exactly this
scenario.
>
> For security purposes, actually, you really do need to check that the
> CANCEL came from the same place as the request being cancelled (see the
> minutes of the security task force on this). CANCELs cannot usefully be
> authenticated directly, since they may come from proxies. That aside,
> CANCEL can come from anywhere.
>
> As for responses, I'm really not sure. The spec says nothing about this,
> as we had not considered it. It clearly requires some out-of-band
> communications between the entity which received the request, and the
> one which sent the response. I'm not sure what the application of that
> would be. That aside, there may be problems. For requests sent over TCP,
> the response is supposed to go over the same connection. Clearly it
> cannot if the response is generated somewhere else. I suspect some
> proxies may be upset if the connection to the server is open, and the
> response comes on a different connection, opened just to send the
> response. In the case of UDP, this will probably work, but you never
> know. 

Wouldn't you have the same security concerns with the response, since it
would allow an entity to redirect a call that was not sent to it?


-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Section 4.3: "A TRANSFER method must not contain a body."
 
I'm wondering about the motivation for this blanket restriction.  It seems
to me that just like a TRANSFER tells the transferee whom to invite, there
are also cases where the transferor would like to indicate some of the
parameters of that new call.  Using the ATM-enabled SDP being worked in the
mmusic group now, for example, the transferor might want to indicate some
suggested QoS parameters that the transferee should use for the new call.
Or perhaps the transferor (particularly if there has been contact with the
target, as in a transfer-with-consultation) wishes to suggest some
particular codecs for the new call.  At the very least, the transferor might
want to indicate whether the transferee should initiate an audio-only call
or a multimedia call to the target.  Moving away from SDP, one could imagine
sending an infamous .jpg file of the target, or perhaps a URL to view while
waiting for a transfer to complete.
 
-----
 
Section 4.5: "If the transferee's agent decides to attempt the transfer, a
200 OK response must be returned by the [sic] if it was able to establish a
session with the transferor, otherwise a 603 Decline must be returned."
 
There's a missing word in here, hence the "sic" above.  And do you mean
"establish a session with the target", not establish a session with the
*transferor*?
 
In any case, there are four possible outcomes that I see for a TRANSFER: 
 
(1) There's something wrong with the request.  I think this is dealt with in
that any appropriate 400-600 class response may be returned.  (This is
specified just before the quote I extracted above.)
(2) The transfer is attempted, and succeeds.  The section quoted above seems
to say that 200 OK should be returned in this case, and I agree.
(3) The transfer is attempted, but fails.
(4) The transfer is not even attempted.
 
The section quoted above seems to mandate 603 Decline for both cases (3) and
(4).  I would think that there are cases where the transferor would like to
distinguish between the cases.  For example, I can try to transfer someone
to party A, and if that fails try to transfer them to party B, etc., perhaps
even indefinitely.  Whereas if the transferor doesn't want to be
transferred, there's no point in trying another transfer.  In fact, I would
think (4) is actually (4a) "this particular transfer will not be attempted"
and (4b) "no transfer will be attempted."
 
I'd like to see 603 Decline used only for case (4), and a different
"transfer failed" code returned for case (3).  And I'd like to see some way
to distinguish between cases (4a) and (4b), along the lines of "Busy Here"
vs. "Busy Everywhere".  Probably 603 Decline should mean Decline All (a la
600 Busy Everywhere), and a new 400-class code would be the simple transfer
Decline (a la 486 Busy Here).  I thought about the Retry-After header, but
it wouldn't be clear whether to retry this transfer, or a different one.
 
-----
 
Section 4.6: "UA receiving a well-formed TRANSFER request should request
approval from the user to proceed.   In the absence of that request, or upon
receiving approval, the UA must submit an INVITE to the resource ..."
 
I don't think that sending the INVITE is necessarily the correct behavior if
the user is not queried.  Certainly if the user is queried and replies, the
action is obvious.  But if the user is not queried, I think there are
scenarios where the INVITE should not be done -- in secure environments,
with a transfer target outside the security boundary, for example.  I think
in the absence of a query or response from the user, the behavior should be
up to the UA implementer, who will probably make it some sort of
configurable option.  But I think "MUST" is too strong here for the default
behavior.  If you do want to mandate some default behavior, I think in the
absence of a user query/response or known user configuration, it would be to
respond with a Decline All.
 
-----

Tim Schroeder 
tschroeder@tri.sbc.com 
SBC Technology Resources, 9505 Arboretum Blvd., Austin, TX  78759 

 


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"McMurry, Kathleen" wrote:
> 
> > > By different address you mean different source IP address, I'm
> assuming(?)
> > > (A different From: of course would not cause a problem -- the proxy
> could return a 481)
> > >
> > > I see no reason to state absolutely that the CANCEL or final response
> MUST come from
> > > the same IP address. There are obvious applications for exactly this
> scenario.
> >
> > For security purposes, actually, you really do need to check that the
> > CANCEL came from the same place as the request being cancelled (see the
> > minutes of the security task force on this). CANCELs cannot usefully be
> > authenticated directly, since they may come from proxies. That aside,
> > CANCEL can come from anywhere.
> >
> > As for responses, I'm really not sure. The spec says nothing about this,
> > as we had not considered it. It clearly requires some out-of-band
> > communications between the entity which received the request, and the
> > one which sent the response. I'm not sure what the application of that
> > would be. That aside, there may be problems. For requests sent over TCP,
> > the response is supposed to go over the same connection. Clearly it
> > cannot if the response is generated somewhere else. I suspect some
> > proxies may be upset if the connection to the server is open, and the
> > response comes on a different connection, opened just to send the
> > response. In the case of UDP, this will probably work, but you never
> > know.
> 
> Wouldn't you have the same security concerns with the response, since it
> would allow an entity to redirect a call that was not sent to it?

Yes, actually. Like CANCEL, a redirect response can be generated by a
proxy and also by a user agent. So, doing source address checks there is
probably a good idea. 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Mon Jun  5 23:13:07 2000
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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        "McMurry, Kathleen" <Kathleen.McMurry@wcom.com>
Cc: Sean Olson <eussean@exu.ericsson.se>, sip@lists.bell-labs.com
Subject: RE: [SIP] Request/Responses from other addresses
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> > For security purposes, actually, you really do need to check that the
> > CANCEL came from the same place as the request being cancelled (see the
> > minutes of the security task force on this). CANCELs cannot usefully be
> > authenticated directly, since they may come from proxies. That aside,
> > CANCEL can come from anywhere.
> >
I take it then that this doesn't apply to BYE requests because they can be
authenticated directly? Is there a weaker source address check you could
apply in the absence of authentication? I can't think of any infallible
ones. Requests will not always come from the original source address and the
Contact address doesn't necessarily infer that the source of future requests
will be from that address... although it would help if we could assume that!

So I guess that without authentication, SIP is very vulnerable to someone
sniffing packets and then generating their own requests to re-INVITE the
call to themselves.

Am I missing something?

Robert.

-----Original Message-----
From:	Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent:	Tuesday, June 06, 2000 6:11 AM
To:	McMurry, Kathleen
Cc:	Sean Olson; sip@lists.bell-labs.com
Subject:	Re: [SIP] Request/Responses from other addresses



"McMurry, Kathleen" wrote:
> 
> > > By different address you mean different source IP address, I'm
> assuming(?)
> > > (A different From: of course would not cause a problem -- the proxy
> could return a 481)
> > >
> > > I see no reason to state absolutely that the CANCEL or final response
> MUST come from
> > > the same IP address. There are obvious applications for exactly this
> scenario.
> >
> > As for responses, I'm really not sure. The spec says nothing about this,
> > as we had not considered it. It clearly requires some out-of-band
> > communications between the entity which received the request, and the
> > one which sent the response. I'm not sure what the application of that
> > would be. That aside, there may be problems. For requests sent over TCP,
> > the response is supposed to go over the same connection. Clearly it
> > cannot if the response is generated somewhere else. I suspect some
> > proxies may be upset if the connection to the server is open, and the
> > response comes on a different connection, opened just to send the
> > response. In the case of UDP, this will probably work, but you never
> > know.
> 
> Wouldn't you have the same security concerns with the response, since it
> would allow an entity to redirect a call that was not sent to it?

Yes, actually. Like CANCEL, a redirect response can be generated by a
proxy and also by a user agent. So, doing source address checks there is
probably a good idea. 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Tue Jun  6 00:20:59 2000
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"Fairlie-Cuninghame, Robert" wrote:
> 
> > > For security purposes, actually, you really do need to check that the
> > > CANCEL came from the same place as the request being cancelled (see the
> > > minutes of the security task force on this). CANCELs cannot usefully be
> > > authenticated directly, since they may come from proxies. That aside,
> > > CANCEL can come from anywhere.
> > >
> I take it then that this doesn't apply to BYE requests because they can be
> authenticated directly?

Yes.

> Is there a weaker source address check you could
> apply in the absence of authentication? I can't think of any infallible
> ones.

THere is no such thing as infallible security anyway.

> Requests will not always come from the original source address and the
> Contact address doesn't necessarily infer that the source of future requests
> will be from that address... although it would help if we could assume that!

In the case of mobility, the address definitely won't be the same.

> 
> So I guess that without authentication, SIP is very vulnerable to someone
> sniffing packets and then generating their own requests to re-INVITE the
> call to themselves.

Yes; but thats why we have it. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From: Postmann Erwin <erwin.postmann@siemens.at>
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Subject: AW: [SIP] SIP Application Server
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I have followed the discussion on the SIP application server with big
interest. My understanding of the application server is that it is a more
complex SIP proxy, redirect server or user agent which provides applications
and services to the users.
Is my understanding right?


I have the following questions for my clarification:

How can I ensure that the application server is always in the route of my
SIP requests?, 
Have to registered at the application server or is it enough to use the
route header in my requests (e.g. INVITE)?

If I want to use more then one application server (with different services)
in parallel how can I instruct the proxy/client to select the appropriate
one (e.g. based, called party, the time, day, location, etc.)? Has this
selection to be done at the client or could I have some intelligence stored
in the network?



Best regards


Erwin Postmann
SIEMENS AG Austria
Program and System Engineering
Mobile Networks Solutions - Product Marketing 

Autokaderstrasse 29
A-1210 Vienna

Tel:	+43.5.1707.21398
Fax:	+43.5.1707.51924
GSM:	+43.664.1010034
email:	erwin.postmann@siemens.at 




-----Ursprüngliche Nachricht-----
Von: Neil Deason [mailto:ndeason@ubiquity.net]
Gesendet am: Freitag, 02. Juni 2000 10:20
An: Jonathan Rosenberg
Cc: sip@lists.bell-labs.com
Betreff: Re: [SIP] SIP Application Server

Jonathan Rosenberg wrote:

> Neil Deason wrote:
> >
> > Yes of course a separate SIP network element can talk SIP to a smart
proxy/application
> > server. But the point I was trying to make is that the application
server encapsulates a
> > vanilla proxy server, i.e. the proxy functionality as described by
rfc2543, plus service
> > logic accessed through CPL/SIP-CGI/Servlet API/whatever. My difficulty
is in seeing how
> > this fits with the ISC application working group premise of separating
call signalling from
> > service control via a SIP messaging interface. This seems like an
unnecessary IN model of
> > separation. In the App Server the proxy part does signalling but it is
clearly not
> > separated from controlling service logic by a SIP messaging interface.
>
> I think you are over-analyzing the meaning of separation of
> call-signaling from service control. In reality, all services are
> delivered eventually through signaling, and are based on information
> provided by signaling. Thus, service logic is quite connected to call
> signaling, and embedding a vanilla proxy in an app server is quite a
> good way to access it. I would argue this is quite contrary to the IN
> model, in which there is an attempt to use a separate protocol for
> service control to a separate box. Note however that these protocols
> carry much the same information as the signaling protocols themselves.

Agreed. It was not my analysis that SIP should be used to try and separate
out service control
from call control. However, as was pointed out earlier in this thread this
could be achieved if
SIP is used to separate a non-SIP signalling protocol from service logic in
a 'Softswitch'
network. But then it appears that you would have a separate protocol (SIP)
for service control to
a separate box (App Server). Such separation doesn't really apply in the
same way to the SIP part
of the network and it sounds a lot like the Softswitch network is a SIP
network at the core.
There you find SIP conversant App Servers and around them are SIP Proxy
Servers, SIP Redirect
Servers and SIP User Agents. Some of these UAs may in fact represent
interfaces to PSTN or even
H.323 gateways. Though how do protocols that don't use SIP for signalling
deliver all the rich
new services that SIP enables when, for example, a text/html payload gets
discarded at the
SIP/foo gateway. The point of SIP is surely not to create LNP, 800
translation services in an App
Server for protocols such as SS7, H.323 etc Yes, you could recreate the IN
this way but why not
start out intending to build something better. You achieve this by building
Application Servers
that empower the simple creation of innovative services not Application
Servers that merely
recreate old services.

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



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Postmann Erwin wrote:
> 
> I have followed the discussion on the SIP application server with big
> interest. My understanding of the application server is that it is a more
> complex SIP proxy, redirect server or user agent which provides applications
> and services to the users.
> Is my understanding right?

Yes.

> 
> I have the following questions for my clarification:
> 
> How can I ensure that the application server is always in the route of my
> SIP requests?,
> Have to registered at the application server or is it enough to use the
> route header in my requests (e.g. INVITE)?

Excellent question.

There is no easy answer. The problem is that "service triggers" can be
fairly complex. It is not as simple as checking the request URI of a
request, and sending it to an application server if that user has some
services enabled. Some services are provisioned for any user (such as
administrator services). Some services are triggered based on the From
or To field. Some services are triggered based on domain names in the
request URI rather than on the complete URI. Even more complicated is
the fact that service triggers can be extremely dynamic; a particular
service may require installation of triggers for a wide range of
different requests.

So, how does one do this? One way is to simply route everything to your
applications server, and let it decide what to do. Another way is to
assume that this information exists in some database, and use a query
protocol to fetch it, or use a replication protocol to distribute it.
Taking a step back, you can argue that this is just a more complex form
of routing (remember I made the statement that the beauty of using SIP
for service invocation is that feature selection looks like a routing
problem). So, what we are talking about is distribution of very complex
routing tables between proxies. So, how does one do route distribution
in SIP? Many ways, but a colleague suggested to me once that the closest
thing may well be TRIP. One can envision extending TRIP to cover a wider
range of information for describing the routes. Now, TRIP isn't ideal,
since its really engineered for inter-domain operation, and supporting
distribution of multidimensional routing criteria will destroy any hope
of aggregation, which TRIP spends a lot of time on. But, we have
realized the value of TRIP for intra-domain exchange of routes between a
gateway and a TRIP speaker
(http://search.ietf.org/internet-drafts/draft-rs-trip-gw-00.txt), and I
think much of the arguments about why it makes sense there, may also
apply here. I'm not proposing this, but I think its worth further
consideration. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Subject: [SIP] subscribe SIP Michael Quan
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Dear Sir/Madam,

Please could put me in your distribution list for the SIP working group.

Thanks,


Michael Quan

International Voice Engineering
Level (3) Communications
66 Prescot St
London, E1 8HG

Tel:                +44 (0)20 7864 4444
Direct Line:  +44 (0)20 7864 0941
Mobile:         +44 (0)788 0506108
Fax:                +44 (0)20 7864 4488
Email:             michael.quan@level3.com





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Hi:
    I have a problem regarding Security considerations in RFC2543. Now IETF
have many working groups in security area which consider the security in
lower layer.  so is the security considerations in RFC2543 redundant? In
other words, if having the security in Transport and Network layer, we do
not need the Security consideration of SIP. Am I correct?

ShaoHua Yin

2000/6/4




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From sip-admin@lists.bell-labs.com  Tue Jun  6 09:46:06 2000
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While verifying CANCEL source address is somewhat useful, I wouldn't put
too much faith in it (or feel too bad about not implementing it). In the
realistic cases where somebody can intercept the original INVITE (and,
thus, know the To, From, CSeq and Call-ID needed to spoof the CANCEL),
the same person can probably also fake the source address. Intercept is
a likely possibility in old-style shared-medium (hubbed) LANs, either at
the source or destination. A rogue proxy server could obviously also do
intercept messages, but there are far easier and less conspicuous ways
to prevent a call from happening for such a server. An ISP could
intercept INVITEs, but once you are in the "middle" of the network,
source address filtering is rather difficult, so I doubt that relying on
source addresses in that environment helps much.

While CANCELs cannot be verified in the same way that INVITEs can, I
don't see why proxies couldn't sign them. At least, that would give you
some way to call in the Feds for illegal interference with
telecommunications services when all your calls are being cancelled. (I
suppose rogue CANCELs offer new ways for ILECs to convince their
departing customers that maybe the CLEC wasn't trustworthy to begin
with...) Also, if you have a signed CANCEL, you could, if sufficiently
paranoid, verify that it comes from a domain in the Via list of the
original INVITE.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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Most applications that were invisaged by the pure end to end security
mecanisms did not have chains of proxies, firewalls and NATs as a
consideration. 

Certain applications that can be built using the SIP protocol will require
routing intelligence at the application level and a secure mechanism for hop
by hop traversal through application entities based upon more complex trust
relationships than simply end to end which is the basis for the transport
and network level security mechanisms.

So these SIP security mechanisms are not redudant. 

I would like to mention that:

You do not have to use them if you do not wish to in your applications.
You can use the end to end transport/network security mechanisms with SIP.



> -----Original Message-----
> From:	yinshaohua [SMTP:yin@huawei.com.cn]
> Sent:	Monday, June 05, 2000 8:10 AM
> To:	sip
> Subject:	[SIP] About Security
> 
> Hi:
>     I have a problem regarding Security considerations in RFC2543. Now
> IETF
> have many working groups in security area which consider the security in
> lower layer.  so is the security considerations in RFC2543 redundant? In
> other words, if having the security in Transport and Network layer, we do
> not need the Security consideration of SIP. Am I correct?
> 
> ShaoHua Yin
> 
> 2000/6/4
> 
> 
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

------_=_NextPart_001_01BFCFC1.29CBD448
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>RE: [SIP] About Security</TITLE>
</HEAD>
<BODY>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Most applications =
that were invisaged by the pure end to end security mecanisms did not =
have chains of proxies, firewalls and NATs as a consideration. =
</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Certain applications =
that can be built using the SIP protocol will require routing =
intelligence at the application level and a secure mechanism for hop by =
hop traversal through application entities based upon more complex =
trust relationships than simply end to end which is the basis for the =
transport and network level security mechanisms.</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">So these SIP =
security mechanisms are not redudant. </FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">I would like to =
mention that:</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">You do not have to =
use them if you do not wish to in your applications.</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">You can use the end =
to end transport/network security mechanisms with SIP.</FONT>
</P>
<BR>
<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">yinshaohua [SMTP:yin@huawei.com.cn]</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Monday, June 05, 2000 8:10 AM</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">sip</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">[SIP] About Security</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"MS Song">Hi:</FONT>
<BR><FONT SIZE=3D2 FACE=3D"MS Song">&nbsp;&nbsp;&nbsp; I have a problem =
regarding Security considerations in RFC2543. Now IETF</FONT>
<BR><FONT SIZE=3D2 FACE=3D"MS Song">have many working groups in =
security area which consider the security in</FONT>
<BR><FONT SIZE=3D2 FACE=3D"MS Song">lower layer.&nbsp; so is the =
security considerations in RFC2543 redundant? In</FONT>
<BR><FONT SIZE=3D2 FACE=3D"MS Song">other words, if having the security =
in Transport and Network layer, we do</FONT>
<BR><FONT SIZE=3D2 FACE=3D"MS Song">not need the Security consideration =
of SIP. Am I correct?</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"MS Song">ShaoHua Yin</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"MS Song">2000/6/4</FONT>
</P>
<BR>
<BR>
<BR>

<P><FONT SIZE=3D2 FACE=3D"MS =
Song">_______________________________________________</FONT>
<BR><FONT SIZE=3D2 FACE=3D"MS Song">SIP mailing list</FONT>
<BR><FONT SIZE=3D2 FACE=3D"MS Song">SIP@lists.bell-labs.com</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"MS Song"><A =
HREF=3D"http://lists.bell-labs.com/mailman/listinfo/sip" =
TARGET=3D"_blank">http://lists.bell-labs.com/mailman/listinfo/sip</A></F=
ONT></U>
</P>
</UL>
</BODY>
</HTML>
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Date: Tue, 06 Jun 2000 09:25:28 -0500
From: "Vijay K. Gurbani" <vkg@lucent.com>
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Original-CC: Postmann Erwin <erwin.postmann@siemens.at>, sip@lists.bell-labs.com
Subject: Re: AW: [SIP] SIP Application Server
References: <D9F2B9CD7BD5D21196BC0800060D9ED601E8AFE1@vies186a.sie.siemens.at> <393C8A3F.1657FE78@dynamicsoft.com>
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Jonathan Rosenberg wrote:
> 
> Postmann Erwin wrote:
> >
> > I have followed the discussion on the SIP application server with 
> > big interest. My understanding of the application server is that it 
> > is a more complex SIP proxy, redirect server or user agent which 
> > provides applications and services to the users.
> > Is my understanding right?
> 
> Yes.

I have followed this thread with some interest as well.  However, IMHO,
we need to delineate an application server from a proxy server clearly.
There are functions an application server can do that a proxy MUST not.
One case I can think of is that of third party call control where some 
network entity, based on some application logic, asynchronously creates 
both legs of a call.  Clearly, a proxy MUST not do this since a proxy
MUST not create INVITE requests on its own. 

Regards,

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
IN Architecture and Internet Software Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184


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To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: Sean Olson <eussean@exu.ericsson.se>, sip@lists.bell-labs.com
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So I am assuming that after performing the source address checks, the
correct behavior is to ignore the message that came from a bad source?

--Kathleen McMurry

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Monday, June 05, 2000 5:11 PM
To: McMurry, Kathleen
Cc: Sean Olson; sip@lists.bell-labs.com
Subject: Re: [SIP] Request/Responses from other addresses




"McMurry, Kathleen" wrote:
> 
> > > By different address you mean different source IP address, I'm
> assuming(?)
> > > (A different From: of course would not cause a problem -- the proxy
> could return a 481)
> > >
> > > I see no reason to state absolutely that the CANCEL or final response
> MUST come from
> > > the same IP address. There are obvious applications for exactly this
> scenario.
> >
> > For security purposes, actually, you really do need to check that the
> > CANCEL came from the same place as the request being cancelled (see the
> > minutes of the security task force on this). CANCELs cannot usefully be
> > authenticated directly, since they may come from proxies. That aside,
> > CANCEL can come from anywhere.
> >
> > As for responses, I'm really not sure. The spec says nothing about this,
> > as we had not considered it. It clearly requires some out-of-band
> > communications between the entity which received the request, and the
> > one which sent the response. I'm not sure what the application of that
> > would be. That aside, there may be problems. For requests sent over TCP,
> > the response is supposed to go over the same connection. Clearly it
> > cannot if the response is generated somewhere else. I suspect some
> > proxies may be upset if the connection to the server is open, and the
> > response comes on a different connection, opened just to send the
> > response. In the case of UDP, this will probably work, but you never
> > know.
> 
> Wouldn't you have the same security concerns with the response, since it
> would allow an entity to redirect a call that was not sent to it?

Yes, actually. Like CANCEL, a redirect response can be generated by a
proxy and also by a user agent. So, doing source address checks there is
probably a good idea. 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From: "Robert Sparks" <rsparks@dynamicsoft.com>
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> Section 4.3: "A TRANSFER method must not contain a body."
>
> I'm wondering about the motivation for this blanket restriction.

There was list discussion on that matter near the end of March (look
to the archives for my original motivation) and discussion at the
Adelaide meeting. The decision reached was to remove the restriction.

> -----
>
> Section 4.5: "If the transferee's agent decides to attempt the transfer, a
> 200 OK response must be returned by the [sic] if it was able to
> establish a
> session with the transferor, otherwise a 603 Decline must be returned."
>
> There's a missing word in here, hence the "sic" above.

Actually, there was an extra "the"

>  And do you mean
> "establish a session with the target", not establish a session with the
> *transferor*?

Definitely!


> In any case, there are four possible outcomes that I see for a TRANSFER:

More on that section later.

>
> Section 4.6: "UA receiving a well-formed TRANSFER request should request
> approval from the user to proceed.   In the absence of that
> request, or upon
> receiving approval, the UA must submit an INVITE to the resource ..."
>
> I don't think that sending the INVITE is necessarily the correct
> behavior if
> the user is not queried.  Certainly if the user is queried and
> replies, the
> action is obvious.  But if the user is not queried, I think there are
> scenarios where the INVITE should not be done -- in secure environments,
> with a transfer target outside the security boundary, for
> example.  I think
> in the absence of a query or response from the user, the behavior
> should be
> up to the UA implementer, who will probably make it some sort of
> configurable option.  But I think "MUST" is too strong here for
> the default
> behavior.  If you do want to mandate some default behavior, I think in the
> absence of a user query/response or known user configuration, it
> would be to
> respond with a Decline All.

I will reword this so that it is obvious that local configuration
can affect the behavior.

For the last statement - if an implementation doesn't (or can't) query
a user, and is by policy not going to accept transfers, why not have
it return a Method Not Allowed?

Robert Sparks



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From sip-admin@lists.bell-labs.com  Tue Jun  6 14:57:38 2000
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From: "Farahmand, Fardad" <FFarahmand@sonusnet.com>
To: vkg@lucent.com, Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: RE: AW: [SIP] SIP Application Server
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Good question. The best example can be found in draft-rosenberg-sip-3pcc-00.
The draft proposes a scenario in which a "controller" initiates both legs of
a call in a third party call control model.  The controller can be an
Application Server initiating both legs of a call (either directly as in the
3pcc draft or through the SoftSwitch as in the ISC model). This brings up
another question.

Jonathan,

It seems that the 3pcc draft uses a mechanism which does not require any
extensions to rfc 2543. So is the "controller" in this draft a special case
of one (or more) of the existing clients and servers?  Or is the controller
a form of a server that needs to be specifically identified in the rfc?

Thanks,
Fardad Farahmand
ffarahmand@sonusnet.com

> -----Original Message-----
> From: Vijay K. Gurbani [mailto:vkg@lucent.com]
> Sent: Tuesday, June 06, 2000 10:25 AM
> To: Jonathan Rosenberg
> Cc: Postmann Erwin; sip@lists.bell-labs.com
> Subject: Re: AW: [SIP] SIP Application Server
> 
> 
> Jonathan Rosenberg wrote:
> > 
> > Postmann Erwin wrote:
> > >
> > > I have followed the discussion on the SIP application server with 
> > > big interest. My understanding of the application server 
> is that it 
> > > is a more complex SIP proxy, redirect server or user agent which 
> > > provides applications and services to the users.
> > > Is my understanding right?
> > 
> > Yes.
> 
> I have followed this thread with some interest as well.  
> However, IMHO,
> we need to delineate an application server from a proxy 
> server clearly.
> There are functions an application server can do that a proxy 
> MUST not.
> One case I can think of is that of third party call control 
> where some 
> network entity, based on some application logic, 
> asynchronously creates 
> both legs of a call.  Clearly, a proxy MUST not do this since a proxy
> MUST not create INVITE requests on its own. 
> 
> Regards,
> 
> - vijay
> -- 
> Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com 
> vkg@acm.org
> IN Architecture and Internet Software Group
> Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
> Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 
> 630 713 0184
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 


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From sip-admin@lists.bell-labs.com  Tue Jun  6 16:14:02 2000
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From: "Dave Oran" <oran@cisco.com>
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Date: Tue, 6 Jun 2000 16:13:45 -0400
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Subject: [SIP] Porposal for providing additional error information to UACs
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Besides PC-like devices and big-hunking PSTN gateways, there are going to be
a gaggle of apppliance-like SIP UACs: phones, cable modems, bricks, and
other real "line-side" user devices out there.

Some of these have limited flash and essentially no user programmability. It
also is pretty important to keep the amount of provisioning and
configuration stuff on such devices to a minimum.

One area I've been playing around with is the issue of how to report SIP
error responses to the user of such a device. The basic philosophy of SIP,
of course, is to return the error
response, with possibly some ascii text, and let the UAC figure out what to
do. Letting the UAC control this is great - just what we want - so I've been
considering the options for "what to do".

One thing I don't want to do is have some fixed mapping table of error codes
to tones and announcmeents in the UA iteself. This is hard to provision
(national variants, multiple languages, etc.), takes lots of memory, etc.
etc. and while tractable for, say a SIP phone with a display, what about:
  - black phone attached to a SIP brick, or
  - a SIP phone with no display, or
  - a blind user.

One idea I find rather appealing is to allow a Contact header on any
response (not just redirects) and use that to point the UAC via a URL to
some kind of server which can give more feedback about the error. Servers
might include an RTSP server, a fancy speech synthesizer with computed
content, or just an ftp: url to an audio file. So, for example, you could
do:

     SIP/2.0  480 No Response
     Via: SIP/2.0/UDP here.com:5060
     From: BigGuy <sip:UserA@here.com>
     To: LittleGuy <sip:UserB@there.com>
     Call-ID: 12345600@here.com
     CSeq: 1 INVITE
     Contact:
rtsp://annoucements.here.com/english/you-lose.mp3?UserB@there.com
     Content-Length: 0

and the SIP UA would then use the URL to get to the rtsp server, which would
play the announcement "sorry, but the number you have reached,
UserB@there.com, is out of service, if you need assistance, tough luck - you
think this is AT&T? (just kidding)".

I thought of some alternative mechanisms, but rejected them as follows:

a) putting the url in the text area of the response header: not good because
we don't want to pre-empt that fallback, and because there might be parsing
difficulties in deciding whether to treat the text as a url.
b) use a URL pointer body part: how do you indicate that the semantics of
this body part is something to contact? The contact just seems more natural
for this.
c) invent a new header field: then we get into debates whether that other
thing belongs to 200/300 responses
d) use a new redirect code, something like "387 More Information": this
takes the control out of the hands of the UAC, since now the policy for
whether to just return the "original" error response or remap it for this
particular UAC now has to be made by a proxy, or worse, by the UAS that
rejected the call.

If people think this is a good idea, we could add this to some existing
ongoing SIP WG item, or alternatively I could write a brief I.D. describing
this. A whole Internet-draft for something so minimal seems overkill, but
whatever works for folks...

Thanks to Jonathan and Henning for screening an earlier version of this and
providing some initial feedback (don't blame them!).

Comment? Bricks? "380 Go to Jail, go directly to jail, do not pass go, do
not collect $200"?

Dave Oran.



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Tentative wording is in the current (June 5) 2543bis-00 draft:

\begin{changebar}
\item[4xx, 5xx and 6xx responses:] The \header{Contact} response-header
field can be used with a 4xx, 5xx or 6xx response to indicate the
location where additional information about the error can be found.
\end{changebar}

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


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Dave Oran wrote:
[...]
> One idea I find rather appealing is to allow a Contact header on any
> response (not just redirects) and use that to point the UAC via a URL 
> to some kind of server which can give more feedback about the error. 
> Servers might include an RTSP server, a fancy speech synthesizer with 
> computed content, or just an ftp: url to an audio file. So, for 
> example, you could do:
> 
>      SIP/2.0  480 No Response
>      Via: SIP/2.0/UDP here.com:5060
>      From: BigGuy <sip:UserA@here.com>
>      To: LittleGuy <sip:UserB@there.com>
>      Call-ID: 12345600@here.com
>      CSeq: 1 INVITE
>      Contact:
> rtsp://annoucements.here.com/english/you-lose.mp3?UserB@there.com
>      Content-Length: 0
> 
> and the SIP UA would then use the URL to get to the rtsp server, which 
> would play the announcement 
[...]

Cool.  That is similar to storing AVP in LDAP directories, where the
value of the attribute is a generalized URL from where the actual
information can be retrieved.  Been lucky enough to use such frameworks
with great success...

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
IN Architecture and Internet Software Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184


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From: "Dave Oran" <oran@cisco.com>
To: "Henning Schulzrinne" <schulzrinne@cs.columbia.edu>
Cc: "IETF SIP WG" <sip@lists.bell-labs.com>
Subject: RE: [SIP] Proposal for providing additional error information to UACs
Date: Tue, 6 Jun 2000 16:54:47 -0400
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Works for me!!!!

> -----Original Message-----
> From: hgs@cs.columbia.edu [mailto:hgs@cs.columbia.edu]On Behalf Of
> Henning Schulzrinne
> Sent: Tuesday, June 06, 2000 4:37 PM
> To: Dave Oran
> Cc: IETF SIP WG
> Subject: Re: [SIP] Porposal for providing additional error information
> to UACs
> 
> 
> Tentative wording is in the current (June 5) 2543bis-00 draft:
> 
> \begin{changebar}
> \item[4xx, 5xx and 6xx responses:] The \header{Contact} response-header
> field can be used with a 4xx, 5xx or 6xx response to indicate the
> location where additional information about the error can be found.
> \end{changebar}
> 
> -- 
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> 
> 


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From: Jon Steer <jsteer@tsemantics.com>
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At 04:13 PM 6/6/00 -0400, Dave Oran wrote:

Why isn't a redirect to a regular announcement server by
the destination the preference here?

Isn't there a problem with redirecting a UA device?

Wouldn't it be simpler to instead define an
announcement server that could use the
error message as an address which is
more likely in the UA's domain?

So in my scenario,

  UA1 -> calls UA2 gets an error message
  UA1 then calls Announcement server A using
  the error message/error number as a destination
  and gets a media stream returned in the
  language of choice.


This way, the announcement server shows up
in the normal SLP/DNS directory service
and the UA can use it if it wants to.

regards,
jon



>Besides PC-like devices and big-hunking PSTN gateways, there are going to be
>a gaggle of apppliance-like SIP UACs: phones, cable modems, bricks, and
>other real "line-side" user devices out there.
>
>Some of these have limited flash and essentially no user programmability. It
>also is pretty important to keep the amount of provisioning and
>configuration stuff on such devices to a minimum.
>
>One area I've been playing around with is the issue of how to report SIP
>error responses to the user of such a device. The basic philosophy of SIP,
>of course, is to return the error
>response, with possibly some ascii text, and let the UAC figure out what to
>do. Letting the UAC control this is great - just what we want - so I've been
>considering the options for "what to do".
>
>One thing I don't want to do is have some fixed mapping table of error codes
>to tones and announcmeents in the UA iteself. This is hard to provision
>(national variants, multiple languages, etc.), takes lots of memory, etc.
>etc. and while tractable for, say a SIP phone with a display, what about:
>   - black phone attached to a SIP brick, or
>   - a SIP phone with no display, or
>   - a blind user.
>
>One idea I find rather appealing is to allow a Contact header on any
>response (not just redirects) and use that to point the UAC via a URL to
>some kind of server which can give more feedback about the error. Servers
>might include an RTSP server, a fancy speech synthesizer with computed
>content, or just an ftp: url to an audio file. So, for example, you could
>do:
>
>      SIP/2.0  480 No Response
>      Via: SIP/2.0/UDP here.com:5060
>      From: BigGuy <sip:UserA@here.com>
>      To: LittleGuy <sip:UserB@there.com>
>      Call-ID: 12345600@here.com
>      CSeq: 1 INVITE
>      Contact:
>rtsp://annoucements.here.com/english/you-lose.mp3?UserB@there.com
>      Content-Length: 0
>
>and the SIP UA would then use the URL to get to the rtsp server, which would
>play the announcement "sorry, but the number you have reached,
>UserB@there.com, is out of service, if you need assistance, tough luck - you
>think this is AT&T? (just kidding)".
>
>I thought of some alternative mechanisms, but rejected them as follows:
>
>a) putting the url in the text area of the response header: not good because
>we don't want to pre-empt that fallback, and because there might be parsing
>difficulties in deciding whether to treat the text as a url.
>b) use a URL pointer body part: how do you indicate that the semantics of
>this body part is something to contact? The contact just seems more natural
>for this.
>c) invent a new header field: then we get into debates whether that other
>thing belongs to 200/300 responses
>d) use a new redirect code, something like "387 More Information": this
>takes the control out of the hands of the UAC, since now the policy for
>whether to just return the "original" error response or remap it for this
>particular UAC now has to be made by a proxy, or worse, by the UAS that
>rejected the call.
>
>If people think this is a good idea, we could add this to some existing
>ongoing SIP WG item, or alternatively I could write a brief I.D. describing
>this. A whole Internet-draft for something so minimal seems overkill, but
>whatever works for folks...
>
>Thanks to Jonathan and Henning for screening an earlier version of this and
>providing some initial feedback (don't blame them!).
>
>Comment? Bricks? "380 Go to Jail, go directly to jail, do not pass go, do
>not collect $200"?
>
>Dave Oran.
>
>
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip



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From sip-admin@lists.bell-labs.com  Tue Jun  6 17:11:52 2000
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From: Christian Huitema <huitema@microsoft.com>
To: "'Dave Oran'" <oran@cisco.com>
Cc: IETF SIP WG <sip@lists.bell-labs.com>
Subject: RE: [SIP] Porposal for providing additional error information to 
	UACs
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Dave,

I understand the intent, but find a few design problems with your solution.
First, I would rather not overload the "contact" header with this kind of
information. There are error cases in which it is legitimate to use Contact
for its intended purpose, e.g. a suggestion of an alternative target for the
call. If you want to point to some explanatory text, or voice message, you
can always use another header, e.g. "Comment."

Then, we have to think a little more about the actual intent of the comment.
It seems to me that you want to use SIP in a semi autonomous appliance, that
cannot really function well without the help of a big brother in the net. I
would rather use MGCP for such apliances, but that is another story. In any
case, you are looking at some form of distributed system. So, you could just
as well use a non-SIP channel to get the information. For example, get the
voice message for error code 107 by looking for the URL:
	 http://my.big.brother.in.the.net/explain/?code=107.
If you insist in making that part of SIP, you have to analyze who should
insert the "comment" line. Is the comment a direct function of the error
code, or does it provide additional information? Is this an end-to-end
field, or a field inserted by the last proxy? Is it covered by the
authentication checksums? Who decides in which language to put the
explanation? 

Christian Huitema

> -----Original Message-----
> From: Dave Oran [mailto:oran@cisco.com]
> Sent: Tuesday, June 06, 2000 1:14 PM
> To: IETF SIP WG
> Subject: [SIP] Porposal for providing additional error information to
> UACs
> 
> 
> Besides PC-like devices and big-hunking PSTN gateways, there 
> are going to be
> a gaggle of apppliance-like SIP UACs: phones, cable modems, 
> bricks, and
> other real "line-side" user devices out there.
> 
> Some of these have limited flash and essentially no user 
> programmability. It
> also is pretty important to keep the amount of provisioning and
> configuration stuff on such devices to a minimum.
> 
> One area I've been playing around with is the issue of how to 
> report SIP
> error responses to the user of such a device. The basic 
> philosophy of SIP,
> of course, is to return the error
> response, with possibly some ascii text, and let the UAC 
> figure out what to
> do. Letting the UAC control this is great - just what we want 
> - so I've been
> considering the options for "what to do".
> 
> One thing I don't want to do is have some fixed mapping table 
> of error codes
> to tones and announcmeents in the UA iteself. This is hard to 
> provision
> (national variants, multiple languages, etc.), takes lots of 
> memory, etc.
> etc. and while tractable for, say a SIP phone with a display, 
> what about:
>   - black phone attached to a SIP brick, or
>   - a SIP phone with no display, or
>   - a blind user.
> 
> One idea I find rather appealing is to allow a Contact header on any
> response (not just redirects) and use that to point the UAC 
> via a URL to
> some kind of server which can give more feedback about the 
> error. Servers
> might include an RTSP server, a fancy speech synthesizer with computed
> content, or just an ftp: url to an audio file. So, for 
> example, you could
> do:
> 
>      SIP/2.0  480 No Response
>      Via: SIP/2.0/UDP here.com:5060
>      From: BigGuy <sip:UserA@here.com>
>      To: LittleGuy <sip:UserB@there.com>
>      Call-ID: 12345600@here.com
>      CSeq: 1 INVITE
>      Contact:
> rtsp://annoucements.here.com/english/you-lose.mp3?UserB@there.com
>      Content-Length: 0
> 
> and the SIP UA would then use the URL to get to the rtsp 
> server, which would
> play the announcement "sorry, but the number you have reached,
> UserB@there.com, is out of service, if you need assistance, 
> tough luck - you
> think this is AT&T? (just kidding)".
> 
> I thought of some alternative mechanisms, but rejected them 
> as follows:
> 
> a) putting the url in the text area of the response header: 
> not good because
> we don't want to pre-empt that fallback, and because there 
> might be parsing
> difficulties in deciding whether to treat the text as a url.
> b) use a URL pointer body part: how do you indicate that the 
> semantics of
> this body part is something to contact? The contact just 
> seems more natural
> for this.
> c) invent a new header field: then we get into debates 
> whether that other
> thing belongs to 200/300 responses
> d) use a new redirect code, something like "387 More 
> Information": this
> takes the control out of the hands of the UAC, since now the 
> policy for
> whether to just return the "original" error response or remap 
> it for this
> particular UAC now has to be made by a proxy, or worse, by 
> the UAS that
> rejected the call.
> 
> If people think this is a good idea, we could add this to 
> some existing
> ongoing SIP WG item, or alternatively I could write a brief 
> I.D. describing
> this. A whole Internet-draft for something so minimal seems 
> overkill, but
> whatever works for folks...
> 
> Thanks to Jonathan and Henning for screening an earlier 
> version of this and
> providing some initial feedback (don't blame them!).
> 
> Comment? Bricks? "380 Go to Jail, go directly to jail, do not 
> pass go, do
> not collect $200"?
> 
> Dave Oran.
> 
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 


_______________________________________________
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From sip-admin@lists.bell-labs.com  Tue Jun  6 17:42:22 2000
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To: "Jon Steer" <jsteer@tsemantics.com>,
        "IETF SIP WG" <sip@lists.bell-labs.com>
Subject: RE: [SIP] Porposal for providing additional error information to UACs
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> -----Original Message-----
> From: Jon Steer [mailto:jsteer@tsemantics.com]
> Sent: Tuesday, June 06, 2000 5:05 PM
> To: Dave Oran; IETF SIP WG
> Subject: Re: [SIP] Porposal for providing additional error information
> to UACs
>
>
> At 04:13 PM 6/6/00 -0400, Dave Oran wrote:
>
> Why isn't a redirect to a regular announcement server by
> the destination the preference here?
>
Because that moves the policy for what to do from the UAC to a proxy or UAS.
I want the UAC to decide what to do based on what it finds most convenient
without other elements in the network trying to second guess it's
capabilities or the preferences of the user using it at the moment.




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Date: Tue, 06 Jun 2000 16:45:35 -0500
From: Henry Sinnreich <Henry.Sinnreich@WCom.com>
Subject: RE: [SIP] Porposal for providing additional error information to UACs
In-reply-to: <NDBBKHCGKKIOOIJEGCOEOEHNDJAA.oran@cisco.com>
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Dave Oran wrote:
> or alternatively I could write a
>brief I.D. describing
>this. A whole Internet-draft for something so minimal
>seems overkill, but
>whatever works for folks...

The lenght of an I-D says nothing about its value.
Go for it!

Henry

>-----Original Message-----
>From: sip-admin@lists.bell-labs.com
>[mailto:sip-admin@lists.bell-labs.com]On Behalf Of Dave Oran
>Sent: Tuesday, June 06, 2000 3:14 PM
>To: IETF SIP WG
>Subject: [SIP] Porposal for providing additional error
>information to
>UACs
>
>
>Besides PC-like devices and big-hunking PSTN gateways,
>there are going to be
>a gaggle of apppliance-like SIP UACs: phones, cable
>modems, bricks, and
>other real "line-side" user devices out there.
>
>Some of these have limited flash and essentially no
>user programmability. It
>also is pretty important to keep the amount of provisioning and
>configuration stuff on such devices to a minimum.
>
>One area I've been playing around with is the issue of
>how to report SIP
>error responses to the user of such a device. The
>basic philosophy of SIP,
>of course, is to return the error
>response, with possibly some ascii text, and let the
>UAC figure out what to
>do. Letting the UAC control this is great - just what
>we want - so I've been
>considering the options for "what to do".
>
>One thing I don't want to do is have some fixed
>mapping table of error codes
>to tones and announcmeents in the UA iteself. This is
>hard to provision
>(national variants, multiple languages, etc.), takes
>lots of memory, etc.
>etc. and while tractable for, say a SIP phone with a
>display, what about:
>  - black phone attached to a SIP brick, or
>  - a SIP phone with no display, or
>  - a blind user.
>
>One idea I find rather appealing is to allow a Contact
>header on any
>response (not just redirects) and use that to point
>the UAC via a URL to
>some kind of server which can give more feedback about
>the error. Servers
>might include an RTSP server, a fancy speech
>synthesizer with computed
>content, or just an ftp: url to an audio file. So, for
>example, you could
>do:
>
>     SIP/2.0  480 No Response
>     Via: SIP/2.0/UDP here.com:5060
>     From: BigGuy <sip:UserA@here.com>
>     To: LittleGuy <sip:UserB@there.com>
>     Call-ID: 12345600@here.com
>     CSeq: 1 INVITE
>     Contact:
>rtsp://annoucements.here.com/english/you-lose.mp3?UserB
>@there.com
>     Content-Length: 0
>
>and the SIP UA would then use the URL to get to the
>rtsp server, which would
>play the announcement "sorry, but the number you have reached,
>UserB@there.com, is out of service, if you need
>assistance, tough luck - you
>think this is AT&T? (just kidding)".
>
>I thought of some alternative mechanisms, but rejected
>them as follows:
>
>a) putting the url in the text area of the response
>header: not good because
>we don't want to pre-empt that fallback, and because
>there might be parsing
>difficulties in deciding whether to treat the text as a url.
>b) use a URL pointer body part: how do you indicate
>that the semantics of
>this body part is something to contact? The contact
>just seems more natural
>for this.
>c) invent a new header field: then we get into debates
>whether that other
>thing belongs to 200/300 responses
>d) use a new redirect code, something like "387 More
>Information": this
>takes the control out of the hands of the UAC, since
>now the policy for
>whether to just return the "original" error response
>or remap it for this
>particular UAC now has to be made by a proxy, or
>worse, by the UAS that
>rejected the call.
>
>If people think this is a good idea, we could add this
>to some existing
>ongoing SIP WG item, or alternatively I could write a
>brief I.D. describing
>this. A whole Internet-draft for something so minimal
>seems overkill, but
>whatever works for folks...
>
>Thanks to Jonathan and Henning for screening an
>earlier version of this and
>providing some initial feedback (don't blame them!).
>
>Comment? Bricks? "380 Go to Jail, go directly to jail,
>do not pass go, do
>not collect $200"?
>
>Dave Oran.
>
>
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip



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From sip-admin@lists.bell-labs.com  Tue Jun  6 17:52:30 2000
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From: "Robert Sparks" <rsparks@dynamicsoft.com>
To: "Schroeder, Tim" <schroeder@tri.sbc.com>, <sip@lists.bell-labs.com>
Subject: RE: [SIP] comments on draft-sparks-sip-cc-transfer-00.txt
Date: Tue, 6 Jun 2000 16:51:20 -0500
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> In any case, there are four possible outcomes that I see for a TRANSFER:
>
> (1) There's something wrong with the request.  I think this is
> dealt with in
> that any appropriate 400-600 class response may be returned.  (This is
> specified just before the quote I extracted above.)
> (2) The transfer is attempted, and succeeds.  The section quoted
> above seems
> to say that 200 OK should be returned in this case, and I agree.
> (3) The transfer is attempted, but fails.
> (4) The transfer is not even attempted.
>
> The section quoted above seems to mandate 603 Decline for both
> cases (3) and
> (4).  I would think that there are cases where the transferor
> would like to
> distinguish between the cases.  For example, I can try to transfer someone
> to party A, and if that fails try to transfer them to party B,
> etc., perhaps
> even indefinitely.  Whereas if the transferor doesn't want to be
> transferred, there's no point in trying another transfer.  In
> fact, I would
> think (4) is actually (4a) "this particular transfer will not be
> attempted"
> and (4b) "no transfer will be attempted."
>
> I'd like to see 603 Decline used only for case (4), and a different
> "transfer failed" code returned for case (3).  And I'd like to
> see some way
> to distinguish between cases (4a) and (4b), along the lines of "Busy Here"
> vs. "Busy Everywhere".  Probably 603 Decline should mean Decline All (a la
> 600 Busy Everywhere), and a new 400-class code would be the
> simple transfer
> Decline (a la 486 Busy Here).  I thought about the Retry-After header, but
> it wouldn't be clear whether to retry this transfer, or a different one.
>

Addressing 4b) first : if the transferee has instructed his UA to accept
no transfers, then the UA could return a 405 Method not Allowed.

Can you sketch an example where it would make a difference to the
transferor's UA if a 603 response was due to 3) or 4a)? From its
perspective, should it care if the response came from the transferee
deciding "I don't want to talk to _that_ person" or "I tried, but
nobody answered".

RjS



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From sip-admin@lists.bell-labs.com  Tue Jun  6 19:19:39 2000
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"Farahmand, Fardad" wrote:
> 
> It seems that the 3pcc draft uses a mechanism which does not require any
> extensions to rfc 2543. So is the "controller" in this draft a special case
> of one (or more) of the existing clients and servers?  Or is the controller
> a form of a server that needs to be specifically identified in the rfc?

The controller is a UAC, actually. It requires no extensions at all.

People seem to get caught up in whether a box is a proxy or a redirect
or a UA or whatever. These are logical roles. If a box does something
which defines it as a proxy, then its a proxy. If it does something
which is characteristic of a UA (like initiating its own requests), its
a UA.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Tue Jun  6 19:31:45 2000
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I read the draft, and I am left wondering exactly what aspects (or features)
of call control the Transfer method adds which can not be accomplished using
existing SIP methods. The draft cites that:

"...these changes facilitate recovery of failed transfers and
   clarify state management in the participating entities"

Looking back over the SIP email there are additional points the author and
others make in favor of the Transfer method, but I am (as I'm sure others
are) unsure if these alone justify implementation of a new method. I believe
the draft, discussions, and group thinking would all benefit from a thorough
comparison of call control models and features made possible using the
Transfer method vs. existing SIP methods (such as BYE Also).

Many thanks,
Mike Fox


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From sip-admin@lists.bell-labs.com  Tue Jun  6 19:40:02 2000
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Date: Tue, 06 Jun 2000 19:40:08 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Folks,

I am pleased to finally announce the availability of source code for
RTPlib, and RTP library in C developed jointly by myself, Daniel
Rubenstein from U.Mass Amherst, and Henning Schulzrinne and Jonathan
Lennox from Columbia University. The library is fairly complete,
implementing some of the harder things, such as SSRC collision, mixers,
and RTCP timer reconsideration, in a transparent manner. It comes with
HTML documentation and example applications. 

The library is free for non-commercial and research usage. It has been
run and tested primarily on Solaris, but has also been used on NT.

You can obtain the software at:
http://www.bell-labs.com/topic/swdist/

Thanks!

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
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From sip-admin@lists.bell-labs.com  Tue Jun  6 20:26:08 2000
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Subject: Re: [SIP] Record-Route (Was: Contact in OPTIONS)
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Jonathan Rosenberg wrote:
> 

> Via hiding is not stricly needed; only if you want to be stateless AND
> hide the route. The far easier approach is simply strip off the via
> list, store it, and insert a fresh Via header with some public address
> you are not afraid to show to the world, in order to get the response.
> When the response comes, slap back on the old Via list, and off you go.
> 
> I'll also note that Via hiding always worried me because it relied on
> the *next hop* to encrypt for you; what if the trust relationship is
> such that they might not do this for you? Better off protect yourself
> than trust the next hop to do so.

Actually, that's pretty similar in either case. One could think of the
"public address that you are not afraid to show" as the one doing the
Via hiding. The SIP server at the public address would have to be
trusted in either case.



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


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From sip-admin@lists.bell-labs.com  Tue Jun  6 22:01:09 2000
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Subject: Re: AW: [SIP] SIP Application Server
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Jonathan Rosenberg wrote:
> 
> "Farahmand, Fardad" wrote:
> >
> > It seems that the 3pcc draft uses a mechanism which does not require any
> > extensions to rfc 2543. So is the "controller" in this draft a special case
> > of one (or more) of the existing clients and servers?  Or is the controller
> > a form of a server that needs to be specifically identified in the rfc?
> 
> The controller is a UAC, actually. It requires no extensions at all.
> 
> People seem to get caught up in whether a box is a proxy or a redirect
> or a UA or whatever. These are logical roles. If a box does something
> which defines it as a proxy, then its a proxy. If it does something
> which is characteristic of a UA (like initiating its own requests), its
> a UA.
> 
> -Jonathan R.
> 

It seems that I am not the only one that have to explain that a SIP
proxy is not necessary a box to add on the network but some simple rules
that can be added to an implementation to gain this logical role.
Perhaps it will be a good thing to add some explanations about this in
the SIP draft?

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"


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From: Scott Petrack <scott.petrack@metatel.com>
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Subject: Re: [SIP] Porposal for providing additional error information to
 UACs
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Sorry that I've not been following this too closely, so apologies if
the following makes no sense or has already been rejected: but this is
really what I wanted a Warning: header to do. I agree with you that the
extra indirection can be very useful, but if all you want is some minimal
extra information, then a single line of text inserted by the last SIP
proxy before the SIP appliance can be very useful. 

Remember, the text in this Warning: header can be tagged for what
language it's in, etc, so that the appliance can be configured to show on
the screen only messages in swahili, whatever. 

If you are going to put a URL to allow the appliance to retrieve the
error, I suggest a new header, rather than the Contact: header. Something
like Warning-URL: or Error-URL:
or even just Display-URL: (the appliance doesn't really need to know if
the stuff to be retrieved and displayed is a warning or some extra info.
The same mechanism could be used for "extra lookups" sort of as if there
were an end-to-end calling-party identification service).

Scott

On Tue, 6 Jun 2000, Dave Oran wrote:

> Besides PC-like devices and big-hunking PSTN gateways, there are going to be
> a gaggle of apppliance-like SIP UACs: phones, cable modems, bricks, and
> other real "line-side" user devices out there.
> 
> Some of these have limited flash and essentially no user programmability. It
> also is pretty important to keep the amount of provisioning and
> configuration stuff on such devices to a minimum.
> 
> One area I've been playing around with is the issue of how to report SIP
> error responses to the user of such a device. The basic philosophy of SIP,
> of course, is to return the error
> response, with possibly some ascii text, and let the UAC figure out what to
> do. Letting the UAC control this is great - just what we want - so I've been
> considering the options for "what to do".
> 
> One thing I don't want to do is have some fixed mapping table of error codes
> to tones and announcmeents in the UA iteself. This is hard to provision
> (national variants, multiple languages, etc.), takes lots of memory, etc.
> etc. and while tractable for, say a SIP phone with a display, what about:
>   - black phone attached to a SIP brick, or
>   - a SIP phone with no display, or
>   - a blind user.
> 
> One idea I find rather appealing is to allow a Contact header on any
> response (not just redirects) and use that to point the UAC via a URL to
> some kind of server which can give more feedback about the error. Servers
> might include an RTSP server, a fancy speech synthesizer with computed
> content, or just an ftp: url to an audio file. So, for example, you could
> do:
> 
>      SIP/2.0  480 No Response
>      Via: SIP/2.0/UDP here.com:5060
>      From: BigGuy <sip:UserA@here.com>
>      To: LittleGuy <sip:UserB@there.com>
>      Call-ID: 12345600@here.com
>      CSeq: 1 INVITE
>      Contact:
> rtsp://annoucements.here.com/english/you-lose.mp3?UserB@there.com
>      Content-Length: 0
> 
> and the SIP UA would then use the URL to get to the rtsp server, which would
> play the announcement "sorry, but the number you have reached,
> UserB@there.com, is out of service, if you need assistance, tough luck - you
> think this is AT&T? (just kidding)".
> 
> I thought of some alternative mechanisms, but rejected them as follows:
> 
> a) putting the url in the text area of the response header: not good because
> we don't want to pre-empt that fallback, and because there might be parsing
> difficulties in deciding whether to treat the text as a url.
> b) use a URL pointer body part: how do you indicate that the semantics of
> this body part is something to contact? The contact just seems more natural
> for this.
> c) invent a new header field: then we get into debates whether that other
> thing belongs to 200/300 responses
> d) use a new redirect code, something like "387 More Information": this
> takes the control out of the hands of the UAC, since now the policy for
> whether to just return the "original" error response or remap it for this
> particular UAC now has to be made by a proxy, or worse, by the UAS that
> rejected the call.
> 
> If people think this is a good idea, we could add this to some existing
> ongoing SIP WG item, or alternatively I could write a brief I.D. describing
> this. A whole Internet-draft for something so minimal seems overkill, but
> whatever works for folks...
> 
> Thanks to Jonathan and Henning for screening an earlier version of this and
> providing some initial feedback (don't blame them!).
> 
> Comment? Bricks? "380 Go to Jail, go directly to jail, do not pass go, do
> not collect $200"?
> 
> Dave Oran.
> 
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 




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Scott Petrack wrote:
> 
> If you are going to put a URL to allow the appliance to retrieve the
> error, I suggest a new header, rather than the Contact: header. Something
> like Warning-URL: or Error-URL:
> or even just Display-URL: (the appliance doesn't really need to know if
> the stuff to be retrieved and displayed is a warning or some extra info.
> The same mechanism could be used for "extra lookups" sort of as if there
> were an end-to-end calling-party identification service).

Whether to use Contact or a new header depends a lot on how you want to
define the precise semantics of the header. If you use the loosest
definition, which is "go here", then Contact will suffice. In fact,
using Contact has the advantage of being able to also use the caller
preferences capabilities. The UA can, in the INVITE, indicate that it
only knows about sip URL schemes. This way, your redirecting device
would not try to send the UA to an rtsp URL, rather it might redirect
the UA to an app server which contacts the RTSP server and streams the
content to the UA. 

However, if your definition of Contact is "try this address to reach the
desired party", probably Contact is not the right thing. As Dave has
pointed out, proxies can recurse on 3xx and attempt to reach the
addresses in a Contact header because the meaning is precisely "try this
address to reach the desired party", and thus its fine for the proxy to
try that address. But, Dave doesn't want proxies to go to these
addresses - they should only be used by the UA upon receipt of the
reject response.

So, I'm a bit on the fence, but probably inclined to define a new header
in order to avoid further overload of exactly what Contact means. 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Henning Schulzrinne wrote:
> 
> While CANCELs cannot be verified in the same way that INVITEs can, I
> don't see why proxies couldn't sign them. At least, that would give you
> some way to call in the Feds for illegal interference with
> telecommunications services when all your calls are being cancelled. 

Right; cancel signatures wouldn't give you any online protection since
you can't easily verify that the entity in the signature is, in fact,
the proxy which sent you the original request.

> Also, if you have a signed CANCEL, you could, if sufficiently
> paranoid, verify that it comes from a domain in the Via list of the
> original INVITE.

This gets confusing if IP addresses are used in the Via header
instead.... 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Hi,

It seems that currently there is no mechanism to inform other parties (e.g.
other proxies) about the supported proxy capabilities.
This is needed especially in REGISTER messages but useful also in INVITEs as
well.

Currently there exists (proposed) Supported header in SIP, but I suppose it
is meant for terminal capabilities only. 
It is impossible to advertize both terminal and proxy capabilities at the
same time in one INVITE/REGISTER message.

Therefore, I'd like to propose "Proxy-Supported" header to be included into
RFC2543bis draft.
The syntax is identical to Supported header.

Comments?


--
Petri Koskelainen
Nokia Research Center


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Subject: RE: AW: [SIP] SIP Application Server
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An alternative model is that all proxy/redirect
servers in a network are "application" servers in 
the sense that they have an application programming
interface. Some of these, say stage 1, application 
servers are specialized for providing "no application 
needed" services and for routing to the application
servers that house applications. The stage 1 & stage 2 
"applications" might then access a common 
(possibly dynamic) data store via mechanisms outside 
the scope of IETF protocol discussions.

This works if the stage 1 & 2 servers are operated
by the same service provider (and provided by the
same vendor?). Probably simpler and more efficient 
than involving another protocol in those common
cases, but is it a generally enough solution?

Steven Gardell
GTE Laboratories
Voice: 781-466-2681, Fax: 781-466-3375

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Tuesday, June 06, 2000 1:21 AM
> To: Postmann Erwin
> Cc: sip@lists.bell-labs.com
> Subject: Re: AW: [SIP] SIP Application Server
> 
> 
> 
> 
> Postmann Erwin wrote:
> > 
> > I have followed the discussion on the SIP application 
> server with big
> > interest. My understanding of the application server is 
> that it is a more
> > complex SIP proxy, redirect server or user agent which 
> provides applications
> > and services to the users.
> > Is my understanding right?
> 
> Yes.
> 
> > 
> > I have the following questions for my clarification:
> > 
> > How can I ensure that the application server is always in 
> the route of my
> > SIP requests?,
> > Have to registered at the application server or is it 
> enough to use the
> > route header in my requests (e.g. INVITE)?
> 
> Excellent question.
> 
> There is no easy answer. The problem is that "service triggers" can be
> fairly complex. It is not as simple as checking the request URI of a
> request, and sending it to an application server if that user has some
> services enabled. Some services are provisioned for any user (such as
> administrator services). Some services are triggered based on the From
> or To field. Some services are triggered based on domain names in the
> request URI rather than on the complete URI. Even more complicated is
> the fact that service triggers can be extremely dynamic; a particular
> service may require installation of triggers for a wide range of
> different requests.
> 
> So, how does one do this? One way is to simply route 
> everything to your
> applications server, and let it decide what to do. Another way is to
> assume that this information exists in some database, and use a query
> protocol to fetch it, or use a replication protocol to distribute it.
> Taking a step back, you can argue that this is just a more 
> complex form
> of routing (remember I made the statement that the beauty of using SIP
> for service invocation is that feature selection looks like a routing
> problem). So, what we are talking about is distribution of 
> very complex
> routing tables between proxies. So, how does one do route distribution
> in SIP? Many ways, but a colleague suggested to me once that 
> the closest
> thing may well be TRIP. One can envision extending TRIP to 
> cover a wider
> range of information for describing the routes. Now, TRIP isn't ideal,
> since its really engineered for inter-domain operation, and supporting
> distribution of multidimensional routing criteria will 
> destroy any hope
> of aggregation, which TRIP spends a lot of time on. But, we have
> realized the value of TRIP for intra-domain exchange of 
> routes between a
> gateway and a TRIP speaker
> (http://search.ietf.org/internet-drafts/draft-rs-trip-gw-00.tx
t), and I
think much of the arguments about why it makes sense there, may also
apply here. I'm not proposing this, but I think its worth further
consideration. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Marc Petit-Huguenin wrote:
> 
> Jonathan Rosenberg wrote:
> >
> > "Farahmand, Fardad" wrote:
> > >
> > > It seems that the 3pcc draft uses a mechanism which does not require any
> > > extensions to rfc 2543. So is the "controller" in this draft a special case
> > > of one (or more) of the existing clients and servers?  Or is the controller
> > > a form of a server that needs to be specifically identified in the rfc?
> >
> > The controller is a UAC, actually. It requires no extensions at all.
> >
> > People seem to get caught up in whether a box is a proxy or a redirect
> > or a UA or whatever. These are logical roles. If a box does something
> > which defines it as a proxy, then its a proxy. If it does something
> > which is characteristic of a UA (like initiating its own requests), its
> > a UA.
> >
> > -Jonathan R.
> >
> 
> It seems that I am not the only one that have to explain that a SIP
> proxy is not necessary a box to add on the network but some simple rules
> that can be added to an implementation to gain this logical role.
> Perhaps it will be a good thing to add some explanations about this in
> the SIP draft?

If you look at the June 5th edition of the 2543bis draft, you'll see
that the definition now includes wording about per-request behavior.
This is also explained in the SIP FAQ.


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


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Date: Wed, 07 Jun 2000 08:32:37 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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Subject: Re: [SIP] Request/Responses from other addresses
References: <75C79E507864D3118AFC00805FEAB7D85B392B@ripexch001.mcit.com> <393CFB7E.B7A901BF@cs.columbia.edu> <393DE3B8.2C65762F@dynamicsoft.com>
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Jonathan Rosenberg wrote:
> 
> Henning Schulzrinne wrote:
> >
> > While CANCELs cannot be verified in the same way that INVITEs can, I
> > don't see why proxies couldn't sign them. At least, that would give you
> > some way to call in the Feds for illegal interference with
> > telecommunications services when all your calls are being cancelled.
> 
> Right; cancel signatures wouldn't give you any online protection since
> you can't easily verify that the entity in the signature is, in fact,
> the proxy which sent you the original request.
> 
> > Also, if you have a signed CANCEL, you could, if sufficiently
> > paranoid, verify that it comes from a domain in the Via list of the
> > original INVITE.
> 
> This gets confusing if IP addresses are used in the Via header
> instead....
> 
But probably still manageable, with somewhat more work (you'd have to
translate the domain of the signer to see if it matches the IP address).
Isn't perfect, but given that the attack requires a fair amount of
sophistication to begin with, I suspect that this will be an issue only
in very limited (high-assurance) situations. A paranoid implementation
could simply say "acme.com (signed) wants to cancel your call from
aol.com". Unless you (or the original destination) are in either the
acme or aol domain, it's very likely that this is a fake, regardless of
the precise IP address.


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


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From sip-admin@lists.bell-labs.com  Wed Jun  7 08:48:00 2000
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If you have a SIP SRV record, please let me know so that I can update
http://www.cs.columbia.edu/~hgs/sip/assignments.html. Note in particular
the paragraph about the transition from sip.udp.foo.com to
_sip._udp.foo.com. (There is no harm in maintaining both "old" and "new"
RR for now.) It is quite useful to have these for testing purposes.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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From sip-admin@lists.bell-labs.com  Wed Jun  7 09:01:03 2000
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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        IETF SIP WG <sip@lists.bell-labs.com>
Subject: Re: [SIP] Porposal for providing additional error information toUACs
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Jonathan Rosenberg wrote:
> 
> Scott Petrack wrote:
> >
> > If you are going to put a URL to allow the appliance to retrieve the
> > error, I suggest a new header, rather than the Contact: header. Something
> > like Warning-URL: or Error-URL:
> > or even just Display-URL: (the appliance doesn't really need to know if
> > the stuff to be retrieved and displayed is a warning or some extra info.
> > The same mechanism could be used for "extra lookups" sort of as if there
> > were an end-to-end calling-party identification service).
> 
> Whether to use Contact or a new header depends a lot on how you want to
> define the precise semantics of the header. If you use the loosest
> definition, which is "go here", then Contact will suffice. In fact,
> using Contact has the advantage of being able to also use the caller
> preferences capabilities. The UA can, in the INVITE, indicate that it
> only knows about sip URL schemes. This way, your redirecting device
> would not try to send the UA to an rtsp URL, rather it might redirect
> the UA to an app server which contacts the RTSP server and streams the
> content to the UA.
> 
> However, if your definition of Contact is "try this address to reach the
> desired party", probably Contact is not the right thing. As Dave has
> pointed out, proxies can recurse on 3xx and attempt to reach the
> addresses in a Contact header because the meaning is precisely "try this
> address to reach the desired party", and thus its fine for the proxy to
> try that address. But, Dave doesn't want proxies to go to these
> addresses - they should only be used by the UA upon receipt of the
> reject response.

I'm not sure this is different from "this address is only to be used in
case the original address isn't reachable" (3xx).

> 
> So, I'm a bit on the fence, but probably inclined to define a new header
> in order to avoid further overload of exactly what Contact means.
> 

Since there is no ambiguity in the action to be taken, I'm not in favor
of defining a new header. Defining a new header then means worrying
about whether *it* has meaning for 3xx or 2xx responses. Thus, in terms
of economy of definition, this one-paragraph version seems to be the
lowest-cost way of achieving additional error explanation capability.
Any new header also increases the chances for misunderstandings, with
various combinations of the new and Contact header.


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


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From sip-admin@lists.bell-labs.com  Wed Jun  7 09:46:03 2000
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Subject: RE: [SIP] Proposal for providing additional error information to UACs
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This has been a good discussion. Sonce I'm basically interested in getting
the functionality, I'm not strongly wedded to whether we define a separate
header, or use Contact. Contact certainly seems the course of least
resistance, as Henning and Jonathan point out, and I agree.

To respond to Christian Heuitema's message, if all I was trying to do was
get access to an announcement server that played error-response-dependent
audio or did something fixed, then I agree that there's no need for
enahncing SIP itself. However, I want to factor out the UAC policy about
what it does when it gets an error reponse from how the alternatives for
what it could do are presented. It seems for flexibility it makes sense to
allow the protocol to carry the alternatives.

Getting back to thge issue of Contact vs. new header, I did, however, think
of one reason why a separate header provides more flexibility, which may
also alleviate Henning's (and my) concern about what such a header would
mean on a 2xx or 3xx.

Let's suppose you do get a 3xx-style redirect. If we had a separate header,
like Scott's proposed "Error-URL:" the UAC could *both* act on the URL,
*and* retry the invite to the parties listed in the Contact: header.

For example, let's suppose I'm on vacation, and I set my UAS (or my local
proxy) to reject the Invite with a 302 Moved Temporarily, with the contact
field set to, say, my voicemail server, and the new header set a URL which
points to an audio file that says "Sorry I'm away, you'll have to talk to my
voicemail". The UAC could then access the message either before retrying, or
in parallel with retrying to the voicemail.

Don't know if people find this example useful, but it might stimulate
further discussion and hopefully consensus.

Dave.

> -----Original Message-----
> From: hgs@cs.columbia.edu [mailto:hgs@cs.columbia.edu]On Behalf Of
> Henning Schulzrinne
> Sent: Wednesday, June 07, 2000 9:00 AM
> To: Jonathan Rosenberg
> Cc: Scott Petrack; Dave Oran; IETF SIP WG
> Subject: Re: [SIP] Porposal for providing additional error information
> toUACs
>
>
> Jonathan Rosenberg wrote:
> >
> > Scott Petrack wrote:
> > >
> > > If you are going to put a URL to allow the appliance to retrieve the
> > > error, I suggest a new header, rather than the Contact:
> header. Something
> > > like Warning-URL: or Error-URL:
> > > or even just Display-URL: (the appliance doesn't really need
> to know if
> > > the stuff to be retrieved and displayed is a warning or some
> extra info.
> > > The same mechanism could be used for "extra lookups" sort of
> as if there
> > > were an end-to-end calling-party identification service).
> >
> > Whether to use Contact or a new header depends a lot on how you want to
> > define the precise semantics of the header. If you use the loosest
> > definition, which is "go here", then Contact will suffice. In fact,
> > using Contact has the advantage of being able to also use the caller
> > preferences capabilities. The UA can, in the INVITE, indicate that it
> > only knows about sip URL schemes. This way, your redirecting device
> > would not try to send the UA to an rtsp URL, rather it might redirect
> > the UA to an app server which contacts the RTSP server and streams the
> > content to the UA.
> >
> > However, if your definition of Contact is "try this address to reach the
> > desired party", probably Contact is not the right thing. As Dave has
> > pointed out, proxies can recurse on 3xx and attempt to reach the
> > addresses in a Contact header because the meaning is precisely "try this
> > address to reach the desired party", and thus its fine for the proxy to
> > try that address. But, Dave doesn't want proxies to go to these
> > addresses - they should only be used by the UA upon receipt of the
> > reject response.
>
> I'm not sure this is different from "this address is only to be used in
> case the original address isn't reachable" (3xx).
>
> >
> > So, I'm a bit on the fence, but probably inclined to define a new header
> > in order to avoid further overload of exactly what Contact means.
> >
>
> Since there is no ambiguity in the action to be taken, I'm not in favor
> of defining a new header. Defining a new header then means worrying
> about whether *it* has meaning for 3xx or 2xx responses. Thus, in terms
> of economy of definition, this one-paragraph version seems to be the
> lowest-cost way of achieving additional error explanation capability.
> Any new header also increases the chances for misunderstandings, with
> various combinations of the new and Contact header.
>
>
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
>
>



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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
Organization: Oy L M Ericsson Ab
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To: jdrosen@dynamicsoft.com
Cc: Henry.Sinnreich@WCOM.Com, sip@lists.bell-labs.com
Subject: Re: [SIP] SIP Rules!
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Hi,

Are we receiving the "Computer Telephony Magazine"?? Does any of you
have a copy?

Thanks,

Gonzalo

jdrosen@dynamicsoft.com wrote:
> 
> Let me add that that story was the cover article of the May Computer
> Telephony Magazine. The cover itself is very amusing.... I encourage
> people to pick up a copy.
> 
> -Jonathan R.
> 
> "Sinnreich, Henry" wrote:
> >
> > You may have seen it, but here is the URL:
> > http://commweb.com/features/comptel/2k0522.sip.html
> >
> > Henry
> >
> > >SIP mailing list
> > >SIP@lists.bell-labs.com
> > >http://lists.bell-labs.com/mailman/listinfo/sip
> > >
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland


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Date: Wed, 07 Jun 2000 17:32:30 +0200
To: petri.koskelainen@nokia.com
From: Romeo Zwart <Romeo.Zwart@icoe.att.com>
Subject: Re: [SIP] Proxy-Supported header
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Hi Petri,

At 02:07 PM 07-06-00 +0300, petri.koskelainen@nokia.com wrote:
 > It seems that currently there is no mechanism to inform other parties (e.g.
 > other proxies) about the supported proxy capabilities.
 > This is needed especially in REGISTER messages but useful also in INVITEs
 > as well.

I agree with you that there is a need for a mechanism to exchange
capability info between proxies, in addition to capabilities exchange
between UA (C and S).

 > Currently there exists (proposed) Supported header in SIP, but I suppose
 > it is meant for terminal capabilities only.
 > It is impossible to advertize both terminal and proxy capabilities at the
 > same time in one INVITE/REGISTER message.

According to 2543bis:
"The Supported general header field enumerates all the capabilities
of the client or server".

This text ("server") does not seem to exclude using Supported by
proxies, however, you are right that you run into problems with
expressing both UA and server capabilities in the same (forwarded)
message. Therefore Proxy-Supported seems to make sense.

 > Therefore, I'd like to propose "Proxy-Supported" header to be included
 > into RFC2543bis draft.
 > The syntax is identical to Supported header.
 >
 > Comments?

The same argumentation applies to a Proxy-Unsupported header
as well. If we need Proxy-Supported, do we also need
Proxy-Unsupported?

Romeo

 > --
 > Petri Koskelainen
 > Nokia Research Center



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Subject: Re: [SIP] Request/Responses from other addresses
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Henning Schulzrinne wrote:
> 
> > This gets confusing if IP addresses are used in the Via header
> > instead....
> >
> But probably still manageable, with somewhat more work (you'd have to
> translate the domain of the signer to see if it matches the IP address).
> Isn't perfect, but given that the attack requires a fair amount of
> sophistication to begin with, I suspect that this will be an issue only
> in very limited (high-assurance) situations. A paranoid implementation
> could simply say "acme.com (signed) wants to cancel your call from
> aol.com". Unless you (or the original destination) are in either the
> acme or aol domain, it's very likely that this is a fake, regardless of
> the precise IP address.

True. So, sounds like we have two solutions for CANCEL security:

1. source address validation, marginally useful
2. only accept CANCEL on an SA, and that CANCEL must be for a request
received over the SA
3. proxies sign CANCEL

Like any other security mechanism, there are pros and cons. Probably the
spec should mention all three.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Wed Jun  7 13:30:04 2000
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From: Jonathan Lennox <lennox@cs.columbia.edu>
To: sip@lists.bell-labs.com
Subject: Re: [SIP] CANCEL security (was: Request/Responses from other addresses)
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On Wed, June 7 2000, "Jonathan Rosenberg" wrote to "Henning Schulzrinne, McMurry, Kathleen, Sean Olson, sip@lists.bell-labs.com" saying:

> True. So, sounds like we have two solutions for CANCEL security:
> 
> 1. source address validation, marginally useful
> 2. only accept CANCEL on an SA, and that CANCEL must be for a request
> received over the SA
> 3. proxies sign CANCEL
> 
> Like any other security mechanism, there are pros and cons. Probably the
> spec should mention all three.

There's another way to do CANCEL authentication; it's much lighter weight,
and more robust, but it requires some spec extensions.

When a proxy server proxies a request, it adds a hop-by-hop header
"Cancel-Lock:" identifying its domain and containing a secure hash (MD5 or
SHA/1) of some random string.  If that proxy server subsequently wants to
cancel the request, it inserts a "Cancel-Key:" containing the original
string.  A downstream server wanting to validate the CANCEL message simply
verifies that the Cancel-Key does indeed hash to the Cancel-Lock that was in
the original request.

As an example, if a UAS receives the following request:

INVITE sip:lennox@cs.columbia.edu SIP/2.0
....
Cancel-Lock: example.com md5 d1a6fb27aa2a3c6ff4a181001dba4a20
....

Then the following CANCEL request should be accepted for this transaction:

CANCEL sip:lennox@cs.columbia.edu SIP/2.0
....
Cancel-Key: example.com md5 This-is-my-cancel-key-129781234701924701234
....


This has, I believe, the correct properties for CANCEL: any device which has
been on the proxy chain for the initial request (including the initial UAC)
can cancel it, but no one else can.  No one has to have any security
associations or relationships outside of this particular transaction.

The main problem with this proposal is backward compatibility; it's not
clear what the right thing to do is if some devices on the proxy chain
support Cancel-Locks and others don't.

(The idea for this scheme is taken from a similar proposal for securing the
NNTP cancel cmsg.)

-- 
Jonathan Lennox
lennox@cs.columbia.edu


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From sip-admin@lists.bell-labs.com  Wed Jun  7 13:57:58 2000
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Jonathan Lennox wrote:
> 
> On Wed, June 7 2000, "Jonathan Rosenberg" wrote to "Henning Schulzrinne, McMurry, Kathleen, Sean Olson, sip@lists.bell-labs.com" saying:
> 
> > True. So, sounds like we have two solutions for CANCEL security:
> >
> > 1. source address validation, marginally useful
> > 2. only accept CANCEL on an SA, and that CANCEL must be for a request
> > received over the SA
> > 3. proxies sign CANCEL
> >
> > Like any other security mechanism, there are pros and cons. Probably the
> > spec should mention all three.
> 
> There's another way to do CANCEL authentication; it's much lighter weight,
> and more robust, but it requires some spec extensions.
> 
> When a proxy server proxies a request, it adds a hop-by-hop header
> "Cancel-Lock:" identifying its domain and containing a secure hash (MD5 or
> SHA/1) of some random string.  If that proxy server subsequently wants to
> cancel the request, it inserts a "Cancel-Key:" containing the original
> string.  A downstream server wanting to validate the CANCEL message simply
> verifies that the Cancel-Key does indeed hash to the Cancel-Lock that was in
> the original request.
> 
> As an example, if a UAS receives the following request:
> 
> INVITE sip:lennox@cs.columbia.edu SIP/2.0
> ....
> Cancel-Lock: example.com md5 d1a6fb27aa2a3c6ff4a181001dba4a20
> ....
> 
> Then the following CANCEL request should be accepted for this transaction:
> 
> CANCEL sip:lennox@cs.columbia.edu SIP/2.0
> ....
> Cancel-Key: example.com md5 This-is-my-cancel-key-129781234701924701234
> ....
> 
> This has, I believe, the correct properties for CANCEL: any device which has
> been on the proxy chain for the initial request (including the initial UAC)
> can cancel it, but no one else can.  No one has to have any security
> associations or relationships outside of this particular transaction.
> 
> The main problem with this proposal is backward compatibility; it's not
> clear what the right thing to do is if some devices on the proxy chain
> support Cancel-Locks and others don't.
> 
> (The idea for this scheme is taken from a similar proposal for securing the
> NNTP cancel cmsg.)

Assuming somebody can modify the INVITE request (or create a copy that
outraces the original to the UAS), this doesn't help much.

Presumably, you'd have to do the same thing with the BYE (assuming you
don't do end system authentication, i.e., you are only concerned about
an uninterrupted call once you've decided to speak to a person). It
would be nice if this worked for re-INVITE, too...

Frankly, given that there are for more "interesting" denial-of-service
attacks that require far less snooping effort, I wonder if this a
first-order problem. News is a bit different since anybody has all of
the information required to cancel anybody else's article, unless you
apply this proof-of-possession mechanism.




> 
> --
> Jonathan Lennox
> lennox@cs.columbia.edu
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

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


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From sip-admin@lists.bell-labs.com  Wed Jun  7 15:26:52 2000
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From: "Schroeder, Tim" <schroeder@tri.sbc.com>
To: sip@lists.bell-labs.com
Subject: RE: [SIP] comments on draft-sparks-sip-cc-transfer-00.txt
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Sorry about missing the decision to remove the blanket restriction against
bodies in TRANSFER messages.

> > Section 4.5: "If the transferee's agent decides to attempt the transfer,
a
> > 200 OK response must be returned by the [sic] if it was able to
establish a
> > session with the transferor, otherwise a 603 Decline must be returned."
> >
> > There's a missing word in here, hence the "sic" above.
> 
> Actually, there was an extra "the".

My copy still doesn't read right after taking out "the", but I trust you
will make the final copy make sense. :-)

> > I think in the absence of a user query/response or known user
configuration, it
> > would be to respond with a Decline All.
>
> For the last statement - if an implementation doesn't (or can't) query
> a user, and is by policy not going to accept transfers, why not have
> it return a Method Not Allowed?

That sounds good.  I suppose you could even use it if the transferee was
queried and the response was "for this session, I am unwilling to consider
being transferred".  As far as the particular session was concerned, the
TRANSFER method would no longer be allowed.  The transferee could even try a
few proposed transfers that turned out to fail (or decline them one-by-one,
or a mixture of both) and eventually give up in disgust and send back a
Method Not Allowed.  

There is a difference between TRANSFER being unsupported by the software and
TRANFERs being declined wholesale by the user, but I'm not sure it's
important to relay that information to the transferor.  It might even be
considered a privacy issue to convey it.

Tim Schroeder
tschroeder@tri.sbc.com
SBC Technology Resources, 9505 Arboretum Blvd., Austin, TX  78759
> 


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From: "Schroeder, Tim" <schroeder@tri.sbc.com>
To: "'sip@lists.bell-labs.com'" <sip@lists.bell-labs.com>
Subject: FW: [SIP] Proposal for providing additional error information to 
	UACs
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> Comment? Bricks? "380 Go to Jail, go directly to jail, do not 
> pass go, do not collect $200"?

Should that be "do not collect 200 OK"? :-}

Tim Schroeder
tschroeder@tri.sbc.com
SBC Technology Resources, 9505 Arboretum Blvd., Austin, TX  78759

 


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From: "Schroeder, Tim" <schroeder@tri.sbc.com>
To: sip@lists.bell-labs.com
Subject: RE: [SIP] comments on draft-sparks-sip-cc-transfer-00.txt
Date: Wed, 7 Jun 2000 15:18:33 -0500 
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> > In any case, there are four possible outcomes that I see for a TRANSFER:
> >
> > (1) There's something wrong with the request.  I think this is dealt
with in
> > that any appropriate 400-600 class response may be returned.
> > (2) The transfer is attempted, and succeeds.  The section quoted above
seems
> > to say that 200 OK should be returned in this case, and I agree.
> > (3) The transfer is attempted, but fails.
> > (4) The transfer is not even attempted.
> >
> > The section quoted above seems to mandate 603 Decline for both cases (3)
and (4).
> >
> > (4) is actually (4a) "this particular transfer will not be attempted"
> > and (4b) "no transfer will be attempted."
 
> Addressing 4b) first : if the transferee has instructed his UA to accept
> no transfers, then the UA could return a 405 Method not Allowed.

Agreed.

> Can you sketch an example where it would make a difference to the
> transferor's UA if a 603 response was due to 3) or 4a)? From its
> perspective, should it care if the response came from the transferee
> deciding "I don't want to talk to _that_ person" or "I tried, but
> nobody answered".

Maybe. :-)

One example might be if the transferor might consider retrying the TRANSFER,
perhaps after some time had passed, perhaps after doing some out-of-band
thing to make it more likely that the transfer would succeed.  Maybe the
Bonds Department tries to TRANSFER the call to the IRA Department, and it
fails because everyone is busy.  But when a busy response is generated, a
page goes out to have all the loiterers get back to the phones, so a little
later the call might succeed.  That TRANSFER failure would be different than
a Decline (meaning "don't you dare transfer me to the IRA Department -- they
transferred me to you!").

The difference might also be useful in planning one's call handling
scenarios.  If a high proportion of transfers from "General Support" to "OS
Support" are not succeeding, it's different if they are failing because "OS
Support" is understaffed and the calls are failing, vs. callers are
declining to be transferred to "OS Support" (because "General Support" is
cavalierly transferring callers there when the callers are sure it's an
application problem, or because they've had bad experiences with OS Support
in the past, or ...).

I could maybe contrive some more examples, but these are the two that come
to mind first.  I guess if the transferor and the target are un-related, it
doesn't make much difference if the transfer fails because it was tried and
failed, or because it was declined.  But in the cases where the transferor
and the target are more tightly coupled (in the same umbrella organization,
perhaps), it does seems to make a difference.  The user-to-user session can
proceed differently if the TRANSFER failed because the transferee
declined/balked, or whether the TRANSFER failed despite the agreeableness of
the transferee.

So, using my numbering scheme, I still like
(1)  400-600-class failure [problem with the TRANSFER request]
(2)  200 OK [transfer attempted and succeeded]
(3)  ??? [some response that specifically means transfer attempted but
failed]
(4a) 603 Decline [this particular transfer will not be attempted]
(4b) 405 Method Not Allowed [no transfer method will be attempted]

(If the transfer was actually attempted, even knowing *why* the transfer
failed could be useful in some cases.  If Boss transfers calls to Peon, Boss
may be interested in knowing whether the failure was because Peon is
responding with 603 Decline or because of 500 Internal Server Error.  I
suppose in case (3), the transferee's UA could optionally include a reason
phrase conveying any desired information about the transfer failure, while
still using the status code that means only "transfer actually attempted but
failed".)

Tim Schroeder
tschroeder@tri.sbc.com
SBC Technology Resources, 9505 Arboretum Blvd., Austin, TX  78759


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From sip-admin@lists.bell-labs.com  Wed Jun  7 17:13:39 2000
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Subject: [SIP] Distinctive ringing feature
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Hi,
	I would like to know if it would be possible to use SIP to instruct
the endpoints/UA's to provide different ringing sequences. If someone has
implemented this feature I would love to know more about it. I am talking
about the distinctive ringing feature as it is present in the PSTN network
today.

Thanks

Sekhar


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From sip-admin@lists.bell-labs.com  Wed Jun  7 19:32:03 2000
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To: sip@lists.bell-labs.com, oran@cisco.com
Subject: RE: [SIP] Proposal for providing additional error information to UACs
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> Let's suppose you do get a 3xx-style redirect. If we had a separate header,
> like Scott's proposed "Error-URL:" the UAC could *both* act on the URL,
> *and* retry the invite to the parties listed in the Contact: header.
> 
> For example, let's suppose I'm on vacation, and I set my UAS (or my local
> proxy) to reject the Invite with a 302 Moved Temporarily, with the contact
> field set to, say, my voicemail server, and the new header set a URL which
> points to an audio file that says "Sorry I'm away, you'll have to talk to my
> voicemail". The UAC could then access the message either before retrying, or
> in parallel with retrying to the voicemail.
> 
> Don't know if people find this example useful, but it might stimulate
> further discussion and hopefully consensus.
> 
> Dave.

This is a perfect example why Contact: should not be used. I personally
think a URI in the body is the best solution provided enough implementations
support multipart MIME bodies. The allows you to easily specify additional
details about the nature of the content of the URI such as language. This is
also in line with specifying additional information as a text/html body in 
a 4xx/6xx response. Consider this as analagous to the SDP payload of a 
183 Session Progress response: you are specifying additional information or an
alternate form of that information.

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


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From sip-admin@lists.bell-labs.com  Thu Jun  8 00:35:42 2000
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Hi,
you can definetely implement distinctive ringing from a SIP endpoint.
The distinctive ring implementation is purely an endpoint side feature (see
SIP FAQ on Henning's site)
and does not need any special SIP signalling.
For example, if say your phone has two alloted phone numbers - and you want
a different ringing tone depending
on the number the caller dialed - your implementation would simply see the
phone number from the SIP headers
and generate the appropriate ring for that phone.


Regds
Arjun
--
Arjun Roychowdhury @ Hughes Software Systems






Sekhar Banerjee <SBanerjee@lboard.com> on 06/08/2000 02:52:13 AM

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

Subject:  [SIP] Distinctive ringing feature




Hi,
     I would like to know if it would be possible to use SIP to instruct
the endpoints/UA's to provide different ringing sequences. If someone has
implemented this feature I would love to know more about it. I am talking
about the distinctive ringing feature as it is present in the PSTN network
today.

Thanks

Sekhar


_______________________________________________
SIP mailing list
SIP@lists.bell-labs.com
http://lists.bell-labs.com/mailman/listinfo/sip






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From sip-admin@lists.bell-labs.com  Thu Jun  8 00:39:06 2000
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Sekhar Banerjee wrote:
> 
> Hi,
>         I would like to know if it would be possible to use SIP to instruct
> the endpoints/UA's to provide different ringing sequences.

Distinctive ringing is generally implemented locally (and I know of at
least one implementation which does it locally). Conveying it in SIP
doesn't make a whole lot of sense, since SIP is a peer to peer protocol.
There is no notion of "the network" telling the phone how to ring. 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Thu Jun  8 00:48:53 2000
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Folks,

Unfortunately, rfc2543 has a few places where we really need extension
placeholders in the grammar, but they are not there. We would like to
add these in to the revised draft; please shout if you have a problem
with this. Some of the places:

1. transport parameter of SIP URL (we need sctp and tls plus who knows
what else..)
2. user parameter in SIP URL
3. Hide header
4. Content-Disposition

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
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From sip-admin@lists.bell-labs.com  Thu Jun  8 06:48:21 2000
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From: Gethin Liddell <gethin@ubiquity.net>
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To: <adam.roach@ericsson.com>, Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Jonathan Rosenberg wrote:
>"Adam B. Roach" wrote: 

*snip*

>> 
>> 3) This is actually a disagreement I have with RFC2543, and 
>> one that I raised some time ago; however, no one seemed 
>> interested at the time. Since it actively prevented us from 
>>running one of the test cases we wanted to run at this bakeoff, 
>> I'm going to raise the issue again. Take discussion on this 
>> topic to the SIP working group mailing list, please. 
>> 
>> According to section 6.37, an implementation should reject 
>> messages containing unknown URIs in the "To:" field with 
>> a 400 response. Many implementations actually did this. 
>> The test case we were attempting was as follows: 
>> 
>> UAC1 ----> tel: mapping proxy -----> UAC2 
>> 

*snip*

>> UAC2 definitely has quite enough information to service the 
>> call. In fact, if not for this clause in 2543, the call would 
>> be set up perfectly. I contend that clients shouldn't 
>> care if they understand the URI in the To: field, as long 
>> as they can process the call, and I encourage implementations 
>> to exhibit this behavior. 
>> 
>> This also follows naturally from what I suggested above: don't 
>> validate fields unless you intend to use the information 
>> contained in them. 
>
>Hmm. This is a good point. Given the increasing use of tel URLs, its 
>probably worth changing this in rfc2543bis. This doesn't introduce 
>backwards compatibility problems; it just means that calls to UAs 
>compliant to the bis version will get set up successfully, when they 
>won't with rfc2543. 
>
>In general, the *request URI* is what is important in processing a 
>request. The To field is used only for (1) call leg identification 
>(along with From and Call-ID), and (2) FYI to the called party. The 
>request URI represents the desired user at the server being contacted. 
>The To field may contain a URI which is meaningless within the namespace 
>served by the server. 

I agree with the sentiments here but I have a question.

How is it possible to correctly equate two separate unknown 
URI's in a transaction ID?

For example lets say the server (UA or Proxy etc..) receives a 
To: header that contains an Unknown URI as follows:

To: <newuri:unknown;uri:format?>

A URI allows escaping, may be case sensitive, may be case 
insensitive or may be both. So how is it possible to equate the 
URI above to:

To: <newuri:UNKNOWN;uri:f%4Frmat?>

they may or may not be equal.  it all depends on the URI specification itself.

So if two INVITE's are received, identical apart from the To headers shown above,
what is the server to do.

If it does byte by byte comparison, then it will consider them two separate messages.
If it is really one message then a UA may provide two alertings for the one call or 
may accept one and send a BUSY back on the other.  Whatever, both are wrong.

Conversely, if it unescapes the chars and does a case-insensitive comparison then 
it will consider the two messages identical.  If however, the difference in case is
important to the URI spec then they are in fact two different messages and again the 
action taken will be wrong.

This has been added to the bis draft (Section F, top of page 117, although i can't 
find where it refers to) but it only considers the To header.  This does in fact have
broader implications and is an issue for every element that uses URI's (From, Via,
Request-uri), and subsequently every transaction id, Call-Leg and thus message.

As i see it, the original action (RFC 2543) of sending an error back on Unknown URI's
was the correct one.  It would be nice to handle unknown URI's properly but i don't
see how this is possible.

BTW sorry for bringing up a problem with a thread that is over a month old. Big up the 
massive ;)

-- 
Gethin Liddell
Team Geffin
Ubiquity Software Corporation

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


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Subject: Re: [SIP] how to retransmit final response ?
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Its very much a part of the bis draft
See section 1.5.1

Quote:

"A server which transmits a provisional response should retransmit it upon
reception of a duplicate re-quest.
A server which transmits a final response should retransmit it with an
interval that starts at T1
seconds, and doubles for each subsequent packet until it reaches T2
seconds. Response retransmissions
cease when any one of the following occurs:

1. An ACK request for the same transaction is received;
2. a BYE request for the same call leg is received;
3. a CANCEL request for the same call leg is received and the final
response status was equal or greater
to 300;
4. the response has been transmitted 7 times."


Regds
Arjun
--
Arjun Roychowdhury @ Hughes Software Systems
Technical Leader, VOIP Solutions
Off: Prestige Opal,146 Infantry Road,Blore - 560001  Ph:+91-080-2286390/1/2
Res: 68/G 2nd Narayanappa Block, RT Nagar,Blore - 560032  Ph:3430565
http://arjun.cjb.net





Taji Rachid <Rachid.Taji@srit.siemens.fr> on 06/08/2000 05:44:10 PM

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

Subject:  [SIP] how to retransmit final response ?




 Hi All;

What is the policy for reliability of final responses in UDP  ?

I didn't find anywhere in the SIP RFC the way to retransmit final
responses.

Can anyone of you help me  ?

Thanks

Rachid



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Hi,

Final responses (UDP) for BYE, REGISTER, CANCEL & OPTIONS: RFC 2543,
section 10.4.1
Final responses (UDP) for INVITE: RFC 2543, section 10.5.1

Gonzalo

Taji Rachid wrote:
> 
>  Hi All;
> 
> What is the policy for reliability of final responses in UDP  ?
> 
> I didn't find anywhere in the SIP RFC the way to retransmit final responses.
> 
> Can anyone of you help me  ?
> 
> Thanks
> 
> Rachid
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland


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Subject: Re: [SIP] how to retransmit final response ?
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Hi Rachid,

     Please refer to section 10.4.1 in the SIP RFC.

Regards
Rafi Assadi H.M.


----- Original Message -----
From: Taji Rachid <Rachid.Taji@srit.siemens.fr>
To: <sip@lists.research.bell-labs.com>
Sent: Thursday, June 08, 2000 5:44 PM
Subject: [SIP] how to retransmit final response ?


> Hi All;
>
> What is the policy for reliability of final responses in UDP  ?
>
> I didn't find anywhere in the SIP RFC the way to retransmit final
responses.
>
> Can anyone of you help me  ?
>
> Thanks
>
> Rachid
>
>
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip



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Date: Thu, 8 Jun 2000 14:31:41 +0100
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        sip <sip@lists.bell-labs.com>
From: Lawrence Conroy <lwc@roke.co.uk>
Subject: Re: [SIP] missing extension placeholders
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At 12:44 am -0400 8/6/00, Jonathan Rosenberg wrote:
>Folks,
>
>Unfortunately, rfc2543 has a few places where we really need extension
>placeholders in the grammar, but they are not there. We would like to
>add these in to the revised draft; please shout if you have a problem
>with this. Some of the places:
>
>1. transport parameter of SIP URL (we need sctp and tls plus who knows
>what else..)
>2. user parameter in SIP URL
>3. Hide header
>4. Content-Disposition
To which I reply:
Absolutely fine, but how will this fit with TEL: urls?
If we expand the parameters of a SIP URL, then it would be nice if they
could be done in a way that still fits with the current RFC 2806 BNF
for future-extension (I doubt it will as 2806's BNF for this is 
pretty generic, but...).
-- 
All the best, Lawrence
-----------------------------------------------------------------------
| Lawrence Conroy,    | "These Opinions must be mine, 'cos if they    |
| Roke Manor Research |  were my Company's they'd pay me for them"    |
|- lwc@roke.co.uk  ---+- Tel: +44 1794 833666  Fax: +44 1794 833434 --|


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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        Sean Olson <eussean@exu.ericsson.se>, sip@lists.bell-labs.com
Subject: Re: [SIP] Request/Responses from other addresses
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Henning Schulzrinne wrote:
> 
> While verifying CANCEL source address is somewhat useful, I wouldn't put
> too much faith in it (or feel too bad about not implementing it). In the
> realistic cases where somebody can intercept the original INVITE (and,
> thus, know the To, From, CSeq and Call-ID needed to spoof the CANCEL),
> the same person can probably also fake the source address.

I agree. Source address validation does not seem that useful. If loose
source routing or strict source routing (provided by IP) are used the
source IP address is not even the last "host" that handled the packet.

Gonzalo
-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland


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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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Gethin Liddell wrote:
> 

> >In general, the *request URI* is what is important in processing a
> >request. The To field is used only for (1) call leg identification
> >(along with From and Call-ID), and (2) FYI to the called party. The
> >request URI represents the desired user at the server being contacted.
> >The To field may contain a URI which is meaningless within the namespace
> >served by the server.

This was changed April 23 and has been in the 2543bis since that time.

> 
> I agree with the sentiments here but I have a question.
> 
> How is it possible to correctly equate two separate unknown
> URI's in a transaction ID?
> 
> For example lets say the server (UA or Proxy etc..) receives a
> To: header that contains an Unknown URI as follows:
> 
> To: <newuri:unknown;uri:format?>
> 
> A URI allows escaping, may be case sensitive, may be case
> insensitive or may be both. So how is it possible to equate the
> URI above to:
> 
> To: <newuri:UNKNOWN;uri:f%4Frmat?>
> 
> they may or may not be equal.  it all depends on the URI specification itself.
> 
> So if two INVITE's are received, identical apart from the To headers shown above,
> what is the server to do.

Practically speaking, this would only occur if the other side
intentionally mangled the URL on either responses or for requests in the
other direction. Thus, the correct solution (particularly for URL types
other than SIP) is that the callee copies the URL in the From field into
the To field and vice versa if it wants to send a request in the other
direction (for the same call).

For request-URI, comparison only matters to compute branch ids for
loops. However, request URIs are not likely to be random URI types as
they have to be understood by the receiving proxy. In addition, you'd
have to have a truly strange proxy that does random equivalence
transformations on URLs looked up locally, within the same transaction.

Via's don't contain URLs and obviously have nothing to do with call
legs.

Contact are copied into Request-URIs, so they don't matter for call leg
computation.

Thus, byte-by-byte comparison of unknown URL types would seem to work
fine in practice.

> 
> If it does byte by byte comparison, then it will consider them two separate messages.
> If it is really one message then a UA may provide two alertings for the one call or
> may accept one and send a BUSY back on the other.  Whatever, both are wrong.
> 
> Conversely, if it unescapes the chars and does a case-insensitive comparison then
> it will consider the two messages identical.  If however, the difference in case is
> important to the URI spec then they are in fact two different messages and again the
> action taken will be wrong.
> 
> This has been added to the bis draft (Section F, top of page 117, although i can't
> find where it refers to) but it only considers the To header.  This does in fact have
> broader implications and is an issue for every element that uses URI's (From, Via,
> Request-uri), and subsequently every transaction id, Call-Leg and thus message.
> 
> As i see it, the original action (RFC 2543) of sending an error back on Unknown URI's
> was the correct one.  It would be nice to handle unknown URI's properly but i don't
> see how this is possible.
> 
> BTW sorry for bringing up a problem with a thread that is over a month old. Big up the
> massive ;)
> 
> --
> Gethin Liddell
> Team Geffin
> Ubiquity Software Corporation
> 
> http://www.ubiquity.net
> mailto:gethin@ubiquity.net
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

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


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> I agree with the sentiments here but I have a question.
> 
> How is it possible to correctly equate two separate unknown 
> URI's in a transaction ID?
> 

You have three options: 

1) Treat the URI as "generic" in the sense of RFC2396. Then you can
   create a canonical form using basic heuristics like the "host" 
   should be CI, the "userinfo" should CS, etc. Make a best effort
   guess as to whether or not various portions of the URI should be
   excluded and/or considered CI.
 
   Before anyone comments that this won't work in all situations, let
   me assure you that I know this :) It does, however, work fairly
   well.

2) Treat the URI as an opaque string and do a byte-for-byte comparison.
   This also won't work in all cases. It is a best effort attempt.
   (This would work for most tel: URIs.)

3) Cry foul and reject with a 4xx response. 

It is VERY unfortunate that you can't treat the To: and From: URIs as immutable 
strings and so here we are (I hope that anyone writing a SIP implementation out
there will take this into account when they are considering whether or not to 
add the port to the URI or change the case of the hostname or other things like
that).  What Adam is arguing for (and I agree) is to do
your best and not be overly concerned about details that may not be 
meaningful to you and your application anyways.

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


> -- 
> Gethin Liddell
> Team Geffin
> Ubiquity Software Corporation
> 
> http://www.ubiquity.net
> mailto:gethin@ubiquity.net
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 


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> >Unfortunately, rfc2543 has a few places where we really need extension
> >placeholders in the grammar, but they are not there. We would like to
> >add these in to the revised draft; please shout if you have a problem
> >with this. Some of the places:
> >
> >1. transport parameter of SIP URL (we need sctp and tls plus who knows
> >what else..)
> >2. user parameter in SIP URL
> >3. Hide header
> >4. Content-Disposition
> To which I reply:
> Absolutely fine, but how will this fit with TEL: urls?
> If we expand the parameters of a SIP URL, then it would be nice if they
> could be done in a way that still fits with the current RFC 2806 BNF
> for future-extension (I doubt it will as 2806's BNF for this is 
> pretty generic, but...).

The transport and user parameters don't need to be added to the tel: URI
so this should be ok.

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

> -- 
> All the best, Lawrence
> -----------------------------------------------------------------------
> | Lawrence Conroy,    | "These Opinions must be mine, 'cos if they    |
> | Roke Manor Research |  were my Company's they'd pay me for them"    |
> |- lwc@roke.co.uk  ---+- Tel: +44 1794 833666  Fax: +44 1794 833434 --|
>  


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From: Taji Rachid <Rachid.Taji@srit.siemens.fr>
To: sip@lists.research.bell-labs.com
Date: Thu, 8 Jun 2000 14:14:10 +0200 
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Subject: [SIP] how to retransmit final response ?
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 Hi All;
 
What is the policy for reliability of final responses in UDP  ?

I didn't find anywhere in the SIP RFC the way to retransmit final responses.
 
Can anyone of you help me  ?
 
Thanks

Rachid



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From: "Dave Oran" <oran@cisco.com>
To: "Sekhar Banerjee" <SBanerjee@lboard.com>, <SIP@lists.bell-labs.com>
Subject: RE: [SIP] Distinctive ringing feature
Date: Thu, 8 Jun 2000 10:50:52 -0400
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Why can't the SIP user agent just figure it out for itself? The "PSTN
version" of this can just play a different audio file (or built-in ting
tone) based on the To: field or the Request-URI field in the Invite. More
sophisticated UAs can play different rings based on the From: field (say to
ring "outside" and "inside" calls differently, or just about anything it's
clever enough to figure out from the Invite message.

The possibilities are limitless once you decouple the user interface from
the protocol!

Dave.

> -----Original Message-----
> From: sip-admin@lists.bell-labs.com
> [mailto:sip-admin@lists.bell-labs.com]On Behalf Of Sekhar Banerjee
> Sent: Wednesday, June 07, 2000 5:22 PM
> To: 'SIP@lists.bell-labs.com'
> Subject: [SIP] Distinctive ringing feature
>
>
> Hi,
> 	I would like to know if it would be possible to use SIP to instruct
> the endpoints/UA's to provide different ringing sequences. If someone has
> implemented this feature I would love to know more about it. I am talking
> about the distinctive ringing feature as it is present in the PSTN network
> today.
>
> Thanks
>
> Sekhar
>
>
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From sip-admin@lists.bell-labs.com  Thu Jun  8 10:58:48 2000
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From: "Dave Oran" <oran@cisco.com>
To: "Sean Olson" <eussean@exu.ericsson.se>, <sip@lists.bell-labs.com>
Subject: RE: [SIP] Proposal for providing additional error information to UACs
Date: Thu, 8 Jun 2000 10:52:24 -0400
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The body approach does seem themost flexible, but it also is the
heaviest-weight. The way I see this used it's likely to be a proxy which
inserts the information and I don't relish having to write proxy code that
tears apart and re-constructs a MIME multi-part body to add this
information.

We'd probably also need some MIME notation to indicate what the purpose of
the body part was, at least via conventions documented somewhere.

I'm on the fence on this one, like Jonathan. Any suggestions how to reach
closure?

Dave

> -----Original Message-----
> From: Sean Olson [mailto:eussean@exu.ericsson.se]
> Sent: Wednesday, June 07, 2000 7:32 PM
> To: sip@lists.bell-labs.com; oran@cisco.com
> Subject: RE: [SIP] Proposal for providing additional error information
> to UACs
>
>
> > Let's suppose you do get a 3xx-style redirect. If we had a
> separate header,
> > like Scott's proposed "Error-URL:" the UAC could *both* act on the URL,
> > *and* retry the invite to the parties listed in the Contact: header.
> >
> > For example, let's suppose I'm on vacation, and I set my UAS
> (or my local
> > proxy) to reject the Invite with a 302 Moved Temporarily, with
> the contact
> > field set to, say, my voicemail server, and the new header set
> a URL which
> > points to an audio file that says "Sorry I'm away, you'll have
> to talk to my
> > voicemail". The UAC could then access the message either before
> retrying, or
> > in parallel with retrying to the voicemail.
> >
> > Don't know if people find this example useful, but it might stimulate
> > further discussion and hopefully consensus.
> >
> > Dave.
>
> This is a perfect example why Contact: should not be used. I personally
> think a URI in the body is the best solution provided enough
> implementations
> support multipart MIME bodies. The allows you to easily specify additional
> details about the nature of the content of the URI such as
> language. This is
> also in line with specifying additional information as a
> text/html body in
> a 4xx/6xx response. Consider this as analagous to the SDP payload of a
> 183 Session Progress response: you are specifying additional
> information or an
> alternate form of that information.
>
> sean
> --
> Sean Olson <sean.olson@ericsson.com>
>
>



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From sip-admin@lists.bell-labs.com  Thu Jun  8 15:26:10 2000
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petri.koskelainen@nokia.com wrote:
> 

> What do you think about Proxy-Supported and Proxy-Unsupported
> headers to be included into 2543bis ?
> 
> The syntax would be similar to Supported header.
> 
> We will very likely need those (at least Proxy-Supported) in 3G
> registration.

This isn't quite symmetric to Require. Would proxies add or subtract
features from the "Proxy-Supported" and "Proxy-Unsupported" list?
(Presumably subtract and add, respectively...) Would only those proxies
that are on the record-route be allowed to do this? 

> 
> Best Regards,
> --
> Petri Koskelainen
> Nokia Research Center

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


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From sip-admin@lists.bell-labs.com  Thu Jun  8 16:14:25 2000
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Henning Schulzrinne wrote:
> 
> > We will very likely need those (at least Proxy-Supported) in 3G
> > registration.
> 
> This isn't quite symmetric to Require. Would proxies add or subtract
> features from the "Proxy-Supported" and "Proxy-Unsupported" list?
> (Presumably subtract and add, respectively...) Would only those proxies
> that are on the record-route be allowed to do this?

We discussed these and other issues in the past on the list (in fact, I
posted a note outlining every possible combination of who needs what
features from who else); it turned out to be overly complicated, and
there were no real applications for it. So, we solved the basic problem
which there was a clear need for.

You mention some application in 3G registration. Can you elaborate on
that?

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
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From sip-admin@lists.bell-labs.com  Thu Jun  8 16:56:39 2000
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Date: Thu, 08 Jun 2000 15:55:28 -0500
From: Henry Sinnreich <Henry.Sinnreich@WCom.com>
Subject: RE: [SIP] Distinctive ringing feature
In-reply-to: <652568F8.00193A95.00@sampark.hss.hns.com>
To: archow@hss.hns.com, Sekhar Banerjee <SBanerjee@lboard.com>
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Could distinctive ringing be a caller/called preference?

Thanks, Henry

>-----Original Message-----
>From: sip-admin@lists.bell-labs.com
>[mailto:sip-admin@lists.bell-labs.com]On Behalf Of 
>archow@hss.hns.com
>Sent: Wednesday, June 07, 2000 11:36 PM
>To: Sekhar Banerjee
>Cc: 'SIP@lists.bell-labs.com'
>Subject: Re: [SIP] Distinctive ringing feature
>
>
>
>
>Hi,
>you can definetely implement distinctive ringing from 
>a SIP endpoint.
>The distinctive ring implementation is purely an 
>endpoint side feature (see
>SIP FAQ on Henning's site)
>and does not need any special SIP signalling.
>For example, if say your phone has two alloted phone 
>numbers - and you want
>a different ringing tone depending
>on the number the caller dialed - your implementation 
>would simply see the
>phone number from the SIP headers
>and generate the appropriate ring for that phone.
>
>
>Regds
>Arjun
>--
>Arjun Roychowdhury @ Hughes Software Systems
>
>
>
>
>
>
>Sekhar Banerjee <SBanerjee@lboard.com> on 06/08/2000 
>02:52:13 AM
>
>To:   "'SIP@lists.bell-labs.com'" <SIP@lists.bell-labs.com>
>cc:
>
>Subject:  [SIP] Distinctive ringing feature
>
>
>
>
>Hi,
>     I would like to know if it would be possible to 
>use SIP to instruct
>the endpoints/UA's to provide different ringing 
>sequences. If someone has
>implemented this feature I would love to know more 
>about it. I am talking
>about the distinctive ringing feature as it is present 
>in the PSTN network
>today.
>
>Thanks
>
>Sekhar
>
>
>_______________________________________________
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>
>
>
>
>
>
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"Schroeder, Tim" wrote:
> 
> I could maybe contrive some more examples, but these are the two that come
> to mind first.  I guess if the transferor and the target are un-related, it
> doesn't make much difference if the transfer fails because it was tried and
> failed, or because it was declined.  But in the cases where the transferor
> and the target are more tightly coupled (in the same umbrella organization,
> perhaps), it does seems to make a difference.  The user-to-user session can
> proceed differently if the TRANSFER failed because the transferee
> declined/balked, or whether the TRANSFER failed despite the agreeableness of
> the transferee.
> 
> So, using my numbering scheme, I still like
> (1)  400-600-class failure [problem with the TRANSFER request]
> (2)  200 OK [transfer attempted and succeeded]
> (3)  ??? [some response that specifically means transfer attempted but
> failed]

I would say this is a 500 class response. I believe 500 class responses
generally mean, "this request failed this time, but the same request
might succeed later on". Thats exactly the case here. You could even
include a Retry-After in the response indicating when to try.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Message-ID: <202F03744BDCD31194270000F803CA9E76FD32@cxoexc2.cxo.dec.com>
From: "Maddux, Michel" <Michel.Maddux@COMPAQ.com>
To: "'Henry Sinnreich'" <Henry.Sinnreich@WCom.com>, archow@hss.hns.com,
        Sekhar Banerjee <SBanerjee@lboard.com>
Cc: SIP@lists.bell-labs.com
Subject: RE: [SIP] Distinctive ringing feature
Date: Thu, 8 Jun 2000 16:20:34 -0500 
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Greetings:

It occurs to me that this is an exmple of the ability of the called party to
establish a personal filter based upon multiple characteristics; e.g. From:
list (preference to girlfriend/wife/mother) etc... Alarm tone for inbound
solicitors.  Isn't
this up to the endpoint to supply multiple ring capability?  Obvious fun
with .wav files include mooing from certain callers,
even up to a personalized voice announcment ("Hi honey, it's me.  Please
pick up.")  Sort of changes the whole distinctive ringing concept.  

Is there any problem with this sort of concept?  thanks. /m.

> -----Original Message-----
> From:	Henry Sinnreich [SMTP:Henry.Sinnreich@WCom.com]
> Sent:	Thursday, June 08, 2000 2:55 PM
> To:	archow@hss.hns.com; Sekhar Banerjee
> Cc:	SIP@lists.bell-labs.com
> Subject:	RE: [SIP] Distinctive ringing feature
> 
> Could distinctive ringing be a caller/called preference?
> 
> Thanks, Henry
> 
> >-----Original Message-----
> >From: sip-admin@lists.bell-labs.com
> >[mailto:sip-admin@lists.bell-labs.com]On Behalf Of 
> >archow@hss.hns.com
> >Sent: Wednesday, June 07, 2000 11:36 PM
> >To: Sekhar Banerjee
> >Cc: 'SIP@lists.bell-labs.com'
> >Subject: Re: [SIP] Distinctive ringing feature
> >
> >
> >
> >
> >Hi,
> >you can definetely implement distinctive ringing from 
> >a SIP endpoint.
> >The distinctive ring implementation is purely an 
> >endpoint side feature (see
> >SIP FAQ on Henning's site)
> >and does not need any special SIP signalling.
> >For example, if say your phone has two alloted phone 
> >numbers - and you want
> >a different ringing tone depending
> >on the number the caller dialed - your implementation 
> >would simply see the
> >phone number from the SIP headers
> >and generate the appropriate ring for that phone.
> >
> >
> >Regds
> >Arjun
> >--
> >Arjun Roychowdhury @ Hughes Software Systems
> >
> >
> >
> >
> >
> >
> >Sekhar Banerjee <SBanerjee@lboard.com> on 06/08/2000 
> >02:52:13 AM
> >
> >To:   "'SIP@lists.bell-labs.com'" <SIP@lists.bell-labs.com>
> >cc:
> >
> >Subject:  [SIP] Distinctive ringing feature
> >
> >
> >
> >
> >Hi,
> >     I would like to know if it would be possible to 
> >use SIP to instruct
> >the endpoints/UA's to provide different ringing 
> >sequences. If someone has
> >implemented this feature I would love to know more 
> >about it. I am talking
> >about the distinctive ringing feature as it is present 
> >in the PSTN network
> >today.
> >
> >Thanks
> >
> >Sekhar
> >
> >
> >_______________________________________________
> >SIP mailing list
> >SIP@lists.bell-labs.com
> >http://lists.bell-labs.com/mailman/listinfo/sip
> >
> >
> >
> >
> >
> >
> >_______________________________________________
> >SIP mailing list
> >SIP@lists.bell-labs.com
> >http://lists.bell-labs.com/mailman/listinfo/sip
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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From sip-admin@lists.bell-labs.com  Thu Jun  8 17:31:48 2000
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: "Maddux, Michel" <Michel.Maddux@COMPAQ.com>
Cc: "'Henry Sinnreich'" <Henry.Sinnreich@WCom.com>, archow@hss.hns.com,
        Sekhar Banerjee <SBanerjee@lboard.com>, SIP@lists.bell-labs.com
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"Maddux, Michel" wrote:
> 
> Greetings:
> 
> It occurs to me that this is an exmple of the ability of the called party to
> establish a personal filter based upon multiple characteristics; e.g. From:
> list (preference to girlfriend/wife/mother) etc... Alarm tone for inbound
> solicitors.  Isn't
> this up to the endpoint to supply multiple ring capability?  Obvious fun
> with .wav files include mooing from certain callers,
> even up to a personalized voice announcment ("Hi honey, it's me.  Please
> pick up.")  Sort of changes the whole distinctive ringing concept.
> 
> Is there any problem with this sort of concept?  thanks. /m.

No problem at all. Thats what people have been saying by "its a matter
of local implementation." What you describe is the approach supported in
the SIP model.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Thu Jun  8 17:39:51 2000
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Date: Thu, 08 Jun 2000 16:36:10 -0500
From: Henry Sinnreich <Henry.Sinnreich@WCom.com>
Subject: RE: [SIP] Distinctive ringing feature
In-reply-to: <202F03744BDCD31194270000F803CA9E76FD32@cxoexc2.cxo.dec.com>
To: "Maddux, Michel" <Michel.Maddux@COMPAQ.com>, archow@hss.hns.com,
        Sekhar Banerjee <SBanerjee@lboard.com>
Cc: SIP@lists.bell-labs.com
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>>Is there any problem with this sort of concept?  thanks. /m.

Thought the same , but wanted some confirmation.

Thanks, Henry

>-----Original Message-----
>From: Maddux, Michel [mailto:Michel.Maddux@COMPAQ.com]
>Sent: Thursday, June 08, 2000 4:21 PM
>To: 'Henry Sinnreich'; archow@hss.hns.com; Sekhar Banerjee
>Cc: SIP@lists.bell-labs.com
>Subject: RE: [SIP] Distinctive ringing feature
>
>
>Greetings:
>
>It occurs to me that this is an exmple of the ability 
>of the called party to
>establish a personal filter based upon multiple 
>characteristics; e.g. From:
>list (preference to girlfriend/wife/mother) etc... 
>Alarm tone for inbound
>solicitors.  Isn't
>this up to the endpoint to supply multiple ring 
>capability?  Obvious fun
>with .wav files include mooing from certain callers,
>even up to a personalized voice announcment ("Hi 
>honey, it's me.  Please
>pick up.")  Sort of changes the whole distinctive 
>ringing concept.  
>
>Is there any problem with this sort of concept?  thanks. /m.
>
>> -----Original Message-----
>> From:	Henry Sinnreich [SMTP:Henry.Sinnreich@WCom.com]
>> Sent:	Thursday, June 08, 2000 2:55 PM
>> To:	archow@hss.hns.com; Sekhar Banerjee
>> Cc:	SIP@lists.bell-labs.com
>> Subject:	RE: [SIP] Distinctive ringing feature
>> 
>> Could distinctive ringing be a caller/called preference?
>> 
>> Thanks, Henry
>> 
>> >-----Original Message-----
>> >From: sip-admin@lists.bell-labs.com
>> >[mailto:sip-admin@lists.bell-labs.com]On Behalf Of 
>> >archow@hss.hns.com
>> >Sent: Wednesday, June 07, 2000 11:36 PM
>> >To: Sekhar Banerjee
>> >Cc: 'SIP@lists.bell-labs.com'
>> >Subject: Re: [SIP] Distinctive ringing feature
>> >
>> >
>> >
>> >
>> >Hi,
>> >you can definetely implement distinctive ringing from 
>> >a SIP endpoint.
>> >The distinctive ring implementation is purely an 
>> >endpoint side feature (see
>> >SIP FAQ on Henning's site)
>> >and does not need any special SIP signalling.
>> >For example, if say your phone has two alloted phone 
>> >numbers - and you want
>> >a different ringing tone depending
>> >on the number the caller dialed - your implementation 
>> >would simply see the
>> >phone number from the SIP headers
>> >and generate the appropriate ring for that phone.
>> >
>> >
>> >Regds
>> >Arjun
>> >--
>> >Arjun Roychowdhury @ Hughes Software Systems
>> >
>> >
>> >
>> >
>> >
>> >
>> >Sekhar Banerjee <SBanerjee@lboard.com> on 06/08/2000 
>> >02:52:13 AM
>> >
>> >To:   "'SIP@lists.bell-labs.com'" <SIP@lists.bell-labs.com>
>> >cc:
>> >
>> >Subject:  [SIP] Distinctive ringing feature
>> >
>> >
>> >
>> >
>> >Hi,
>> >     I would like to know if it would be possible to 
>> >use SIP to instruct
>> >the endpoints/UA's to provide different ringing 
>> >sequences. If someone has
>> >implemented this feature I would love to know more 
>> >about it. I am talking
>> >about the distinctive ringing feature as it is present 
>> >in the PSTN network
>> >today.
>> >
>> >Thanks
>> >
>> >Sekhar
>> >
>> >
>> >_______________________________________________
>> >SIP mailing list
>> >SIP@lists.bell-labs.com
>> >http://lists.bell-labs.com/mailman/listinfo/sip
>> >
>> >
>> >
>> >
>> >
>> >
>> >_______________________________________________
>> >SIP mailing list
>> >SIP@lists.bell-labs.com
>> >http://lists.bell-labs.com/mailman/listinfo/sip
>> 
>> 
>> _______________________________________________
>> SIP mailing list
>> SIP@lists.bell-labs.com
>> http://lists.bell-labs.com/mailman/listinfo/sip


_______________________________________________
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From sip-admin@lists.bell-labs.com  Thu Jun  8 18:07:12 2000
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From: "Maddux, Michel" <Michel.Maddux@COMPAQ.com>
To: "'Henry Sinnreich'" <Henry.Sinnreich@WCom.com>, archow@hss.hns.com,
        Sekhar Banerjee <SBanerjee@lboard.com>
Cc: SIP@lists.bell-labs.com
Subject: RE: [SIP] Distinctive ringing feature
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Actually, having the phone moo when mother in law calls might be the next
killer app.  ;^) /m.



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Date: Thu, 08 Jun 2000 16:41:37 -0500
From: Henry Sinnreich <Henry.Sinnreich@WCom.com>
In-reply-to: <39401009.CB1DCD4A@dynamicsoft.com>
To: Henry.Sinnreich@WCom.com
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http://www.infoworld.com/articles/hn/xml/00/06/06/000606hnworldc
om.xml

Enjoy!

Henry




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Subject: [SIP] Reg: SipUrl Grammmer
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Hi everyone!

Can some explain me with example about the grammer in SipUrl;
hostname= *(domainlabel".")toplabel["."]

domainlabel & toplabel  correct usage examples ?

thanks
--
Krishan veer



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From sip-admin@lists.bell-labs.com  Fri Jun  9 00:20:56 2000
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To: Henry Sinnreich <Henry.Sinnreich@WCom.com>
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Subject: RE: [SIP] Distinctive ringing feature
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why not ? :-)

lots of interesting things can be put in.

Since this is totally a client side implemenation that does not affect the
SIP signalling in any way,
we also did some cute(?!?) things like the ring volume being dynamically
modulated as duration of ring increases etc (you know,
like those alarms clocks that start with a whisper and keep screaming
louder and louder (till a point) till you hit on the button....

On the lighter side - here's some rings my limited imagination brings up:

Girl/BoyFriend Ring - sweet chimes (or doomsday bongs - depending on what
state you are with her/him)
Beer pal ring - hic..hic...ring.ring...hic..hic
Boss Ring - no ring :-p
Wife Ring - continuous monotonic ring :-))
etc......

Oh well :-) Back to work - I enjoyed keying in this mail :-p

Regds
Arjun

--
Arjun Roychowdhury @ Hughes Software Systems
Technical Leader, VOIP Solutions
Off: Prestige Opal,146 Infantry Road,Blore - 560001  Ph:+91-080-2286390/1/2
Res: 68/G 2nd Narayanappa Block, RT Nagar,Blore - 560032  Ph:3430565
http://arjun.cjb.net











Henry Sinnreich <Henry.Sinnreich@WCom.com> on 06/09/2000 02:25:28 AM

To:   archow, Sekhar Banerjee <SBanerjee@lboard.com>
cc:   SIP@lists.bell-labs.com

Subject:  RE: [SIP] Distinctive ringing feature




Could distinctive ringing be a caller/called preference?

Thanks, Henry

>-----Original Message-----
>From: sip-admin@lists.bell-labs.com
>[mailto:sip-admin@lists.bell-labs.com]On Behalf Of
>archow@hss.hns.com
>Sent: Wednesday, June 07, 2000 11:36 PM
>To: Sekhar Banerjee
>Cc: 'SIP@lists.bell-labs.com'
>Subject: Re: [SIP] Distinctive ringing feature
>
>
>
>
>Hi,
>you can definetely implement distinctive ringing from
>a SIP endpoint.
>The distinctive ring implementation is purely an
>endpoint side feature (see
>SIP FAQ on Henning's site)
>and does not need any special SIP signalling.
>For example, if say your phone has two alloted phone
>numbers - and you want
>a different ringing tone depending
>on the number the caller dialed - your implementation
>would simply see the
>phone number from the SIP headers
>and generate the appropriate ring for that phone.
>
>
>Regds
>Arjun
>--
>Arjun Roychowdhury @ Hughes Software Systems
>
>
>
>
>
>
>Sekhar Banerjee <SBanerjee@lboard.com> on 06/08/2000
>02:52:13 AM
>
>To:   "'SIP@lists.bell-labs.com'" <SIP@lists.bell-labs.com>
>cc:
>
>Subject:  [SIP] Distinctive ringing feature
>
>
>
>
>Hi,
>     I would like to know if it would be possible to
>use SIP to instruct
>the endpoints/UA's to provide different ringing
>sequences. If someone has
>implemented this feature I would love to know more
>about it. I am talking
>about the distinctive ringing feature as it is present
>in the PSTN network
>today.
>
>Thanks
>
>Sekhar
>
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip
>
>
>
>
>
>
>_______________________________________________
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>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip






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From: Igor Slepchin <islepchin@dynamicsoft.com>
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To: Krishan Veer <kveer@vovida.com>
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Subject: Re: [SIP] Reg: SipUrl Grammmer
References: <39401EAF.B5544846@vovida.com>
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vovida.com would be an example of a host name complying with the BNF:)

Just in case: vovida is a domainlabel, com is a toplabel.

---
Igor Slepchin

Krishan Veer wrote:
> 
> Hi everyone!
> 
> Can some explain me with example about the grammer in SipUrl;
> hostname= *(domainlabel".")toplabel["."]
> 
> domainlabel & toplabel  correct usage examples ?
> 
> thanks
> --
> Krishan veer
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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From sip-admin@lists.bell-labs.com  Fri Jun  9 04:50:59 2000
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From: "rafi" <rafi@sasi.com>
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Subject: [SIP] [SIP]: Caching of Final Response(s)
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Hi,

Consider the following paragraph in RFC 2543,
section 10.4.1

     .......  the server sends a final response, it  cannot
 be sure the client has received the response, and thus
SHOULD  cache the results for at least 10*T2 seconds
to avoid having to, for example, contact the user or location
server again upon receiving a request retransmission.

  Now consider the following scenarios.

Scenario 1 :  Consider a UAS supporting many calls
                   (each call's media stream may terminate
                   at different IP  address). If server caches the
                   final response then in worst case server may be
                   caching all the responses almost at same time. This
caching
                   may  cause the server run out of buffer space thereby
                   preventing the server  from serving  new requests.

Scenario 2:  Consider a misbehaved client which is sending request in
                    such way that each time it sends a request it challenges
the
                    server to send final response.  Client can quickly
                    make the server run out of buffer space and bring down
its
                    operation.



  Any comment are welcome.

Thank You

 Rafi Assadi H.M.
 Silicon Automation Systems
 Bangalore, INDIA.





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From sip-admin@lists.bell-labs.com  Fri Jun  9 06:06:18 2000
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From: dmalhotra@hss.hns.com
To: "rafi" <rafi@sasi.com>, sip@lists.bell-labs.com
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Date: Fri, 9 Jun 2000 15:34:47 +0530
Subject: Re: [SIP] [SIP]: Caching of Final Response(s)
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Hi Rafi,
                 In scenario 1, you are right that if multiple responses are
cached then the server
may go out of buffer space but I have a point here that all the calls do not
appear simultaneously
generally they have some pattern and even if it is not there then the
implementation can put a limit
on the number of simultaneous calls for which it will support caching. After
that it may not support the caching and
may release the call or whatever specific action it wants to take.
As far as the scenario 2 is concerned,
                You have probably missed one point that server is going to keep
this info maximum upto 10*T2 seconds.
T2 being 4 seconds so worst case it is 40 seconds.
Regards
Dhiraj Malhotra





"rafi" <rafi@sasi.com> on 06/09/2000 12:08:53 PM

To:   sip@lists.bell-labs.com
cc:    (bcc: Dhiraj Malhotra/HSS)

Subject:  [SIP] [SIP]: Caching of Final Response(s)




Hi,

Consider the following paragraph in RFC 2543,
section 10.4.1

     .......  the server sends a final response, it  cannot
 be sure the client has received the response, and thus
SHOULD  cache the results for at least 10*T2 seconds
to avoid having to, for example, contact the user or location
server again upon receiving a request retransmission.

  Now consider the following scenarios.

Scenario 1 :  Consider a UAS supporting many calls
                   (each call's media stream may terminate
                   at different IP  address). If server caches the
                   final response then in worst case server may be
                   caching all the responses almost at same time. This
caching
                   may  cause the server run out of buffer space thereby
                   preventing the server  from serving  new requests.

Scenario 2:  Consider a misbehaved client which is sending request in
                    such way that each time it sends a request it challenges
the
                    server to send final response.  Client can quickly
                    make the server run out of buffer space and bring down
its
                    operation.



  Any comment are welcome.

Thank You

 Rafi Assadi H.M.
 Silicon Automation Systems
 Bangalore, INDIA.





_______________________________________________
SIP mailing list
SIP@lists.bell-labs.com
http://lists.bell-labs.com/mailman/listinfo/sip






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To: jdrosen@dynamicsoft.com, schulzrinne@cs.columbia.edu
Cc: petri.koskelainen@nokia.com, sip@lists.bell-labs.com
Subject: RE: [SIP] Re: About Proxy-Supported and Proxy-Unsupported
Date: Fri, 9 Jun 2000 14:30:20 +0300 
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Hi,
> > This isn't quite symmetric to Require. Would proxies add or subtract
> > features from the "Proxy-Supported" and "Proxy-Unsupported" list?
> > (Presumably subtract and add, respectively...) Would only 
> those proxies
> > that are on the record-route be allowed to do this?
> 
> We discussed these and other issues in the past on the list 
> (in fact, I
> posted a note outlining every possible combination of who needs what
> features from who else); it turned out to be overly complicated, and
> there were no real applications for it. So, we solved the 
> basic problem
> which there was a clear need for.
> 
> You mention some application in 3G registration. Can you elaborate on
> that?
> 
> -Jonathan R.


I have been thinking that problem also. Initially, the idea behind
Proxy-Supported was that only one proxy (in the visited network) describes
the capabilites available in the visited domain (capabilities specific to
3G) and that REGISTER message is then forwarded into the home network.  

Other proxies should not be allowed to touch the Proxy-Supported header.

If Proxy-Supported/UnSupported are used to describe the capabilities of
several 
SIP proxies along the route, then subtract/add should be applied 
by each proxy (but then we end up with problems, e.g. proxies in 
different domains can not be trusted as much as local proxies etc).

Maybe the name of the proposed Proxy-Supported header should be changed 
to something like "Local-Proxy-Capabilities", or "3G-local-capabilities".


--
Petri


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Subject: Re: [SIP] Re: About Proxy-Supported and Proxy-Unsupported
To: petri.koskelainen@nokia.com
Date: Fri, 9 Jun 2000 06:46:45 -0500 (CDT)
Cc: jdrosen@dynamicsoft.com, schulzrinne@cs.columbia.edu,
        sip@lists.bell-labs.com
In-Reply-To: <25B79E9476BAD211811B0008C7894CDC02F794CB@treis04nok> from "petri.koskelainen@nokia.com" at Jun 09, 2000 02:30:20 PM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>If Proxy-Supported/UnSupported are used to describe the capabilities of
>several 
>SIP proxies along the route, then subtract/add should be applied 
>by each proxy (but then we end up with problems, e.g. proxies in 
>different domains can not be trusted as much as local proxies etc).
>
>Maybe the name of the proposed Proxy-Supported header should be changed 
>to something like "Local-Proxy-Capabilities", or "3G-local-capabilities".

Perhaps. I beleive you're a bit confused about the semantics of
the Supported: header. It is used to denote that certain SIP extension
drafts are supported, not capability sets within a service architecture.

For example, you may want to define an extension (the name of
which would go in the "Supported" header) which defines how to
negotiate the version of a service triggering protocol (this information
would go in a new header defined by the above mentioned extension draft).

You can't really mix the ideas.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA


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Date: Fri, 09 Jun 2000 10:50:39 -0400
From: "Gregory W. Bond" <bond@research.att.com>
Organization: AT&T Labs - Research, Florham Park, NJ
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Subject: [SIP] 3rd CFP (EXTENDED DEADLINE): IP Telecom Services Workshop 2000
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[Please accept our apologies for multiple copies of this call.]

-------------   THIRD CALL FOR PAPERS  ---------------
-------------     EXTENDED DEADLINE    ---------------

       IP TELECOM SERVICES WORKSHOP 2000 (IPTS 2000)

                    co-located with
             Fall 2000 Voice on the Net (VON)

                     organized by
                  AT&T Labs Research
                      pulver.com

         Cobb Galleria, Atlanta, Georgia, U.S.A
                  September 11, 2000

            submission deadline: June 23, 2000
         http://www.research.att.com/conf/ipts2000/

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

IPTS 2000 is a new workshop dedicated to the important, emerging field of IP
Telecom Services research.

In contrast to the services available on the PSTN, a new world of telecom
services is possible on an IP-based infrastructure. This world presents new
challenges for service development, deployment and management. This one-day
workshop will serve as a forum for the dissemination of research relating to
these challenges.

The IPTS 2000 Program Committee invites submission of papers on substantial,
original and previously unpublished research in IP telecom services, including,
but not limited to:

- service architectures
  (e.g. AIN, SoftSwitch, JAIN, Parlay, DFC)
- service creation environments
- protocols, standards and interoperability
  (e.g. H.323, SIP, MGCP, Megaco/H.248, etc.)
- service programming languages
- service management - billing, fault tolerance, monitoring
- novel services

We welcome papers describing original research; position papers describing
research interests in the field, work in progress, or future directions of
research; project or system descriptions.


IMPORTANT DATES

June 23, 2000        Submission deadline
July 21, 2000        Notification of acceptance
August 25, 2000      Final version of papers
September 11, 2000   IPTS 2000


PAPER SUBMISSIONS

Papers must describe original, previously unpublished work that has not been
simultaneously submitted for publication elsewhere. They must be written in
English, not exceeding 6 pages including figures, tables and references in 10-12
point font on 8.5x11-inch (US letter) paper. The first page must include an
abstract of up to 200 words, keywords, and email address of the corresponding
author.

All papers must be submitted electronically unless specifically approved by the
Workshop Co-Chairs. Submissions in PostScript or PDF format should be sent to
mailto:ipts2000@research.att.com.

Authors of accepted papers are expected to present their contribution at the
workshop.  The proceedings will be published in hardcopy and on the web.


REVIEW PROCEDURE

All submissions will be subject to academic peer review by the IPTS 2000 Program
Committee under the chairmanship of the Workshop Co-Chairs. Each paper will
receive three written reviews, and reviews will be returned to the authors.  The
Workshop Co-Chairs have final authority over the review process and all
decisions relating to acceptance of papers. Review criteria include originality
of ideas, technical soundness, significance of results, and quality of
presentation.


CONFERENCE VENUE AND RELATED EVENTS

IPTS 2000 will be held on September 11, 2000 in Atlanta, Georgia. The workshop
will be co-located with Fall 2000 Voice on the Net (VON).  Details about Fall
2000 VON can be found at http://pulver.com.

IPTS 2000 is organized by AT&T Labs Research (http://www.research.att.com) and
pulver.com (http://pulver.com).


WORKSHOP CO-CHAIRS

Greg Bond, AT&T Labs Research
Eric Cheung, AT&T Labs Research

PROGRAM COMMITTEE

Mauricio Arango, Sun Microsystems Labs
Joanne Atlee, University of Waterloo
Ralph Blumenthal, Daewoo Telecom
Scott Hoffpauir, BroadSoft
Evan Magill, University of Strathclyde
Peter Mataga, dynamicsoft
Bernie Pagurek, Carleton University
Jonathan Rosenberg, dynamicsoft
Henning Schulzrinne, Columbia University
Greg Utas, Nortel Networks
Pamela Zave, AT&T Labs Research


FURTHER GENERAL INFORMATION

http://www.research.att.com/conf/ipts2000


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From sip-admin@lists.bell-labs.com  Fri Jun  9 11:45:55 2000
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From: William Marshall <wtm@research.att.com>
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To: sip@lists.bell-labs.com
Subject: [SIP] updated manyfolks draft
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I have submitted an updated version of the 'manyfolks' draft
on resource preconditions for SIP sessions.

Until it appears in the normal ietf directories, it may be
obtained from
ftp://ftp.research.att.com/dist/wtm/draft-manyfolks-sip-resource-01.txt

The most important change in this version of the draft is the
full compliance with Section 10 of RFC 2026.

This also incorporates all the comments received in Adelaide, and
on the email list subsequent to the last IETF meeting.

The PacketCable DCS group has produced an updated specification
incorporating all of the draft-dcsgroup-* extensions.  It
provides numerous detailed call flow examples, and message-by-
message requirements and test sequences for DCS clients and proxies.

A copy of the latest DCS document is available at
http://www.softarmor.com/sipwg/drafts/dcsdraft2.pdf
	and also available at
ftp://ftp.cablelabs.com/pub/ietfdocs/dcsdraft2.pdf

Bill Marshall
wtm@research.att.com


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Subject: [SIP] Re: [PINT] UDP support changed from SHOULD to MUST
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Please see my comments inline.

Regards,
Alec

Lawrence Conroy wrote:

> *       UDP support changed from SHOULD to MUST in SIP 2453 bis draft.
> It looks like support for UDP is gradually moving from a SHOULD to a
> MUST in SIP.
>
> In PINT we may not have that assumption. We can have a good reason
> for using TCP only (inline content, security using TLS, ...), so *for
> PINT* I'd prefer to leave this as a SHOULD.
>

I would like to add to what Lawrence is saying that some SIP applications
might need a more flexible and rapid session set up than others. PINT is a
good example, SPIRITS might as well be another one, assuming that the WG
will decide to use SIP.

>
> I wonder if the move to a MUST in the core spec will cause grief for
> any other extension or application.
>
> The key point here is ->in the core spec<-

Looks like the core spec. needs to be more accommodating to the needs of
extensions (PINT). SHOULD is just fine. Why break something that actually
works?

>
>
> I had the same feeling at the Interim meeting in D.C. when the
> subject of "Busy Line Verification" was raised - there's a lot of
> focus on VoIP, when I can easily imagine SIP being used for a much
> wider set of applications even for "straight" conferences; for
> example, Quake, as Dean suggests.
> Let's imagine an Operator listening in on the sampled Quake audio
> track that I'm conferencing. How long before the Police arrive? An
> over-emphasis on VoIP can lead to bad assumptions, IMHO - SIP is much
> wider than that.
> --
> lwc@roke.co.uk -- +44 1794 833666
> -------------------------------------------------------------
> Herewith a pointless waste of a few lines, stating that
> Roke Manor Research Limited is NOT responsible for my rantings
> and that they would very much like people to sue me rather than
> them if this email contains racist or sexist comments, please.
> Also, I can't do anything; only our lawyers can, so speak to them.
> Oh, and by the way, if our I.T. department has so mangled our
> email system that this has been misdirected, beware that having
> read this far, your eyes are about to fall out from reading
> this highly sensitive information, unless you immediately
> forget that this ever darkened your mailbox. Do it now.
> -------------------------------------------------------------
>
> _______________________________________________
> PINT mailing list
> PINT@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/pint

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Alec Brusilovsky wrote:
> 
> Please see my comments inline.
> 
> Regards,
> Alec
> 
> Lawrence Conroy wrote:
> 
> > *       UDP support changed from SHOULD to MUST in SIP 2453 bis draft.
> > It looks like support for UDP is gradually moving from a SHOULD to a
> > MUST in SIP.
> >
> > In PINT we may not have that assumption. We can have a good reason
> > for using TCP only (inline content, security using TLS, ...), so *for
> > PINT* I'd prefer to leave this as a SHOULD.
> >
> 
> I would like to add to what Lawrence is saying that some SIP applications
> might need a more flexible and rapid session set up than others. PINT is a
> good example, SPIRITS might as well be another one, assuming that the WG
> will decide to use SIP.

Right; both UDP and TCP are still there. You can still do TCP if you
want, or just UDP, but just not ONLY TCP.

> 
> >
> > I wonder if the move to a MUST in the core spec will cause grief for
> > any other extension or application.
> >
> > The key point here is ->in the core spec<-
> 
> Looks like the core spec. needs to be more accommodating to the needs of
> extensions (PINT). SHOULD is just fine. Why break something that actually
> works?

Because it doesn't - thats the problem. UA to UA interoperability is not
possible in all cases unless there is a baseline transport. The "core
spec" must be a workable protocol in its own right, without
clarifications and specifics being defined only in extensions, as pint
is.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Sat Jun 10 01:41:53 2000
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Dave Oran wrote:
> 
> The body approach does seem themost flexible, but it also is the
> heaviest-weight. The way I see this used it's likely to be a proxy which
> inserts the information and I don't relish having to write proxy code that
> tears apart and re-constructs a MIME multi-part body to add this
> information.
> 
> We'd probably also need some MIME notation to indicate what the purpose of
> the body part was, at least via conventions documented somewhere.
> 
> I'm on the fence on this one, like Jonathan. Any suggestions how to reach
> closure?

Well, one way would be to move the pros/cons from theory to something
concrete. Can anyone come up with an example where using Contact would
cause something to go wrong somewhere?

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Sat Jun 10 13:03:51 2000
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At 04:13 PM 6/6/00 -0400, Dave Oran wrote:
>One idea I find rather appealing is to allow a Contact header on any
>response (not just redirects) and use that to point the UAC via a URL to
>some kind of server which can give more feedback about the error. Servers
>might include an RTSP server, a fancy speech synthesizer with computed
>content, or just an ftp: url to an audio file. So, for example, you could
>do:

This is a great idea. I can see how implementers would have had specialized 
hooks in their phones to do this anyway. That would have led to a lot of 
user confusion and complaints of too much complexity with SIP phones. This 
makes things easy and perhaps more important even for everyone by allowing 
for it in the protocol.



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From sip-admin@lists.bell-labs.com  Sun Jun 11 19:24:27 2000
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Date: Sun, 11 Jun 2000 19:26:47 -0400
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Subject: [SIP] DCS privacy update - open issues
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Having cleared the RFC 2026 issues, I'm in the process of updating the
dcs-privacy draft (draft-dcsgroup-sip-privacy-01.txt) based on the
comments received in Adelaide, but would like some list discussion on
some of these first.


1. Remote-Party-ID authenticity
======================
Currently, the Remote-Party-ID may include an "rpi-auth" token intended
to provide authenticity service for the Remote-Party-ID specified. The
current scheme has some shortcomings, most notably that it is not linked
to the INVITE, which can be rectified fairly easily. However, upon
thinking further about this and also based on the SIP Security meeting,
it seems to me that this is simply a special case of a more general
security service, i.e. to provide authenticity (or other security)
service for one or more fields in a SIP message. The proposal is
therefore to simply to remove the "rpi-auth" token.


2. Remote-Party-ID type
=================
The "rpi-type" is intended to identify users by type rather than by name
under the assumption that some types of users may be granted privileges
different than others. One particular example we have identified and
defined in the draft is the type "operator" which we need for the
Busy-Line-Very end Emergency Interrupt. As has been noted, these are
specific service examples and the "operator" type tends to fall in the
service specific category as well. As the concept seems to apply in
general, but the particular application of it may depend on the
environment, the proposal is therefore simply to define "rpi-type" as a
token and have local policy determine particular values that may or may
not be supported.


3. OSPS header
============
As noted above, the OSPS header is not particularly general. but rather
service specific. In the case of BLV, busy call processing should not
result in a busy indication being returned but rather a duplicate media
stream. In the case of EI, a conversation/conference should be the
result. Taking a step back, the OSPS header can be viewed as a caller
request for some kind of specifial call processing - as has been noted,
this looks rather similar to caller preferences. The Request-Disposition
"ring-feature" already achieves parts of this, and an alternative to the
OSPS header would therefore be to simply expand upon this, e.g. by
adding a "bridge-feature" (which might be further described in terms of
simplex or duplex) or something similar.


All comments appreciated.


Thanks and regards

                Flemming Andreasen



--
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Cisco Systems                     Fax:   +1 240 359 5839




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From sip-admin@lists.bell-labs.com  Mon Jun 12 05:29:37 2000
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Subject: [SIP] mobility support in SIP - basic questions
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Hi,
I just went thru draft-itsumo-sip-mobility-req-01.txt and have the
following very basic Qs:

1. Is it correct to assume that no extensions are required in the SIP
protocol to support mobility applications ?
2. Are there any other drafts/papers that address SIP & mobility which I
need to see ?
3. Is there any doc that describes suggested SIP message exchanges b/w the
various entities in a mobile environment ? (ie something  more detailed wrt
message flows )


Thx
Regds
Arjun

--
Arjun Roychowdhury @ Hughes Software Systems





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From sip-admin@lists.bell-labs.com  Mon Jun 12 06:00:46 2000
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Hi,

archow@hss.hns.com wrote:
> 
> Hi,
> I just went thru draft-itsumo-sip-mobility-req-01.txt and have the
> following very basic Qs:
> 
> 1. Is it correct to assume that no extensions are required in the SIP
> protocol to support mobility applications ?

It might be a correct assumption depending on what it is understood by
mobility. If you want just a mobile SIP terminal in your research
network vanilla SIP is probably enough. 

Extensions might be needed if you are thinking of using a UMTS access,
or some services that are typically related to mobility, or being
interoperable with other mobile systems...


> 2. Are there any other drafts/papers that address SIP & mobility which I
> need to see ?

http://www.cs.columbia.edu/~hgs/papers/Wedl9908_Mobility.pdf

> 3. Is there any doc that describes suggested SIP message exchanges b/w the
> various entities in a mobile environment ? (ie something  more detailed wrt
> message flows )

You might want to follow ongoing work in 3GPP.

Regards,

Gonzalo

> 
> Thx
> Regds
> Arjun
> 
> --
> Arjun Roychowdhury @ Hughes Software Systems
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland


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From sip-admin@lists.bell-labs.com  Mon Jun 12 08:06:00 2000
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Besides 3GPP, there is also some interest on SIP mobility in Mobile Wireless
Internet Forum (www.mwif.org) which has started  recently. SIP-2000 in Paris had
a presentation on SIP mobility which addresses some SIP mobility issues as well.

Besides the original work on SIP mobility by Henning and Elin, there are couple
of related drafts in IETF for issues  covering CDMA soft-handoff, and HMMP.

Thanks
Ashutosh




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Hi

I was wondering if someone could point me to a good source of example
SIP messages.

Any information much appreciated.

Thank you

Christopher
-- 
**********************************************************************
Christopher Downey (Student Software Engineer)

Agilent Technologies
Research & Development
Telecoms Systems Division	Email	 :  cdowney@sqf.hp.com
South Queensferry		Phone	 :  +44 31 331 6137
Scotland			Internal :  33137
**********************************************************************


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From sip-admin@lists.bell-labs.com  Mon Jun 12 09:38:17 2000
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Subject: [SIP] Re: Picky SIP bis draft comment
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Billy Biggs wrote:
> 
>   Hi,
> 
>   I noticed you added SCTP and TLS to the transport URL parameter.  On
> page 19, when describing URL parameters, you don't mention them at all.
> 
>   This should be updated to say something like:
> 
>   "The transport parameter determines the transport mechanism to be used
>    for communications.  SIP is currently defined to operate over TCP,
>    UDP, SCTP, and TLS, although others are possible."

Thanks for catching this; fixed (with slightly different wording).

> 
>   One problem with this is that transport unfortunately isn't a list.
> So, I guess if you support all of them, you have to publish each
> transport you support as a seperate URL.  Correct?

Interesting point. A list would be more convenient, particularly when
the URL is included on a web page. For Contact, enumeration will
probably do, except that this won't work when Record-Route is used - the
rules only allow to copy one Contact entry and the UAC has no clue which
of the transport parameters might be appropriate for the last proxy
before the Contact header address. Unfortunately, a list is not
backward-compatible. I suppose we could define a new parameters, say,
transports, which allows a list and deprecate the old version.

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


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From sip-admin@lists.bell-labs.com  Mon Jun 12 09:40:22 2000
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As a follow-up comment, this whole issue could be avoided if we could
rely on DNS SRV. It seems far better to store transport availability in
DNS rather than trying to update URLs on web pages...
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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From sip-admin@lists.bell-labs.com  Mon Jun 12 09:45:19 2000
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Hello,

I'm a little confused as to the proxy behaviour implied by section
6.26 of the June bis.

In the case of OPTIONS, how should I (as a proxy) be responding?
 * Should I literally be responding as to _my_ Server capabilities?
   (I guess this can be useful with Supported, but I'm not sure
   how useful any of { Allow, Accept, Accept-Encoding,
   Accept-Language } are.)  Or is this perhaps dependent on
   the Request-URI being me (the proxy)?
 * Should I return the response I would potentially have sent
   (e.g., do a provisional lookup to see if I could have proxied
   it further; if not, 404)?
 * Should I guess what the desired server (UA?) can do? &:)

In the case of REGISTER, if the domain of the Request-URI
is foreign, what should I return?

Finally, if the request is neither OPTIONS nor REGISTER, I am
supposed to respond with 483.  If I do a lookup on the Request-URI,
and discover that the target resolves in my Registrar with an
action of "redirect", is it sensible to return a 3xx class?
I guess what I'm asking is if there's any need to mandate the 483;
aren't there some instances where a Proxy can return something
that's maybe a little bit more useful?  (Even a 404 could be said
to be useful...no?)


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



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From sip-admin@lists.bell-labs.com  Mon Jun 12 09:46:48 2000
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Check out the SIP call flows draft. There is a copy at Henning's
site.

http://www.cs.columbia.edu/%7Ehgs/sip/drafts/draft-ietf-sip-call-flows-00.txt

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

Christopher Downey wrote:
> 
> Hi
> 
> I was wondering if someone could point me to a good source of example
> SIP messages.
> 
> Any information much appreciated.
> 
> Thank you
> 
> Christopher
> --
> **********************************************************************
> Christopher Downey (Student Software Engineer)
> 
> Agilent Technologies
> Research & Development
> Telecoms Systems Division       Email    :  cdowney@sqf.hp.com
> South Queensferry               Phone    :  +44 31 331 6137
> Scotland                        Internal :  33137
> **********************************************************************
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

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Date: Mon, 12 Jun 2000 09:52:14 -0500
From: Henry Sinnreich <Henry.Sinnreich@WCom.com>
Subject: RE: [SIP] SIP Message Examples
In-reply-to: <3944E770.4857@sqf.hp.com>
To: Christopher Downey <cdowney@sqf.hp.com>, sip@lists.bell-labs.com
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>I was wondering if someone could point me to a good
>source of example
>SIP messages.

http://search.ietf.org/internet-drafts/draft-ietf-sip-call-flows
-00.txt

and

http://search.ietf.org/internet-drafts/draft-sparks-sip-service-
examples-00.txt

Henry

>-----Original Message-----
>From: sip-admin@lists.bell-labs.com
>[mailto:sip-admin@lists.bell-labs.com]On Behalf Of
>Christopher Downey
>Sent: Monday, June 12, 2000 8:37 AM
>To: sip@lists.bell-labs.com
>Subject: [SIP] SIP Message Examples
>
>
>Hi
>
>I was wondering if someone could point me to a good
>source of example
>SIP messages.
>
>Any information much appreciated.
>
>Thank you
>
>Christopher
>--
>*******************************************************
>***************
>Christopher Downey (Student Software Engineer)
>
>Agilent Technologies
>Research & Development
>Telecoms Systems Division	Email	 :  cdowney@sqf.hp.com
>South Queensferry		Phone	 :  +44 31 331 6137
>Scotland			Internal :  33137
>*******************************************************
>***************
>
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip



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From sip-admin@lists.bell-labs.com  Mon Jun 12 11:03:44 2000
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Subject: [SIP] Forking Proxy Behaviour Question
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Hullo,

I have a couple of questions about section 12.4 in the June bis.

The first paragraph after the C fragment contains the following
sentence:
   "A proxy MAY send a CANCEL to all incomplete branches
    and return the best available final status to the
    client if not all responses have been received after
    60 seconds or the timeout period specified in the
    Timeout header field of the request."
    ^^^^^^^
s/Timeout/Expires/?

The next sentence says:
   "When forwarding responses, a proxy MUST forward all
    header fields of the selected response."
Shouldn't this read more along the lines of forwarding the whole
message, since there could be useful information in the message
body?

The next sentence says:
   "1xx: The proxy MAY forward the response upstream
         towards the client."
Would it be better to whack this up to "SHOULD" (as long as
the response is not 100)?  (I guess this also applies to section
10.1.2.)

Finally, is the notion of the "best response" strictly the lowest-
numbered response (excluding 1xx), or is it more about the class
of the response?  The reason I ask this is because if this is the
case, then I don't feel that the order of status codes (especially
in the 4xx region) is necessarily very helpful.

For instance, if a proxy forks to two locations, and one returns
486 and one returns 404, I would think that the 486 was a more
useful response to return?

Further, say a proxy receives an INVITE with a Request-URI that
is unknown (i.e., not in the proxy's domain/locationserver), and
thus the proxy just tries to route as best it can based on some
DNS resolution.  Now there's no server listening at the
transport:IP:port that the proxy tries, so the proxy times-out [0].
As per 12.4, I guess one would expect the proxy to return a 408
response, but perhaps something like a 404 would be better?

Comments?

Cheers,


 - Jo.

[0] Not particularly relevant for TCP, perhaps, but unfortunately
    it looks like we don't get the ICMP clues on Win32 Java. &:(
-- 
  Jo Hornsby; IT Professional         --       mailto:jhornsby@ubiquity.net
 Ubiquity Software Corporation        --         http://www.ubiquity.net/



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From: Billy Biggs <vektor@div8.net>
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Cc: sip@lists.bell-labs.com
Subject: Re: [SIP] Re: Picky SIP bis draft comment
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Henning Schulzrinne (schulzrinne@cs.columbia.edu):

> As a follow-up comment, this whole issue could be avoided if we could
> rely on DNS SRV. It seems far better to store transport availability in
> DNS rather than trying to update URLs on web pages...

  I'm not sure I follow.  You're suggesting that if we could rely on DNS
SRV records, then we could announce URLs without specifying transport,
and have the remote end use DNS to query what transports we support?

  That doesn't work very well with clients on machines that don't have
access to their DNS entries, or to cases where many SIP clients (with
seperate transport capabilities) are running on one machine...

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


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Henning Schulzrinne wrote:
> 
> As a follow-up comment, this whole issue could be avoided if we could
> rely on DNS SRV. It seems far better to store transport availability in
> DNS rather than trying to update URLs on web pages...

I agree. I think we should keep transport as listing a single transport,
which represents the "preferred" mechanism, but still allow for others
discovered as a result of DNS and/or simply trying other transports to
see what happens.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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On Mon, 12 Jun 2000, Jo Hornsby wrote:

> Finally, is the notion of the "best response" strictly the lowest-
> numbered response (excluding 1xx), or is it more about the class
> of the response?  The reason I ask this is because if this is the

i.e. is the code fragment or the text to be taken as true. The code
fragment returns the first response from the lowest "hundred group" of the
responses. Yet the text says to return the lowest non 1xx error code seen.

> case, then I don't feel that the order of status codes (especially
> in the 4xx region) is necessarily very helpful.

Although baseing response codes on HTTP was never going to produce a
ordering of meaningful to SIP. Bearing this in mind the lowest non 1xx
response will produce a consistent result, however, the lack of meaning in
this makes it no better than using the first response from the lowest
category of codes (excluding 1xx).

Do other people feel this is a non-issue or something that ought to be
standardised.

James Undery



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From sip-admin@lists.bell-labs.com  Mon Jun 12 12:23:57 2000
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Subject: Re: [SIP] Re: Picky SIP bis draft comment
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Billy Biggs wrote:
> 
> Henning Schulzrinne (schulzrinne@cs.columbia.edu):
> 
> > As a follow-up comment, this whole issue could be avoided if we could
> > rely on DNS SRV. It seems far better to store transport availability in
> > DNS rather than trying to update URLs on web pages...
> 
>   I'm not sure I follow.  You're suggesting that if we could rely on DNS
> SRV records, then we could announce URLs without specifying transport,
> and have the remote end use DNS to query what transports we support?

That would be one model that works some of the time.

> 
>   That doesn't work very well with clients on machines that don't have
> access to their DNS entries, or to cases where many SIP clients (with
> seperate transport capabilities) are running on one machine...

Yes, but the alternative of changing dozens of web pages and an untold
number of address book entries is probably higher. In the end, I'd
imagine we end up with a hybrid system, where SRV is used for servers
(where you typically wouldn't have a bunch running on the same system,
at least not under the same externally visible name) and URL parameters
for specifying the "favorite" transport and, like Jonathan said,
try-others-if-necessary approach.

To conclude this item: Unless somebody speaks up, the single-transport
mechanism stays as is, rather than complicating things.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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From sip-admin@lists.bell-labs.com  Mon Jun 12 12:45:30 2000
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Jo Hornsby wrote:
> 
> Hullo,
> 
> I have a couple of questions about section 12.4 in the June bis.
> 
> The first paragraph after the C fragment contains the following
> sentence:
>    "A proxy MAY send a CANCEL to all incomplete branches
>     and return the best available final status to the
>     client if not all responses have been received after
>     60 seconds or the timeout period specified in the
>     Timeout header field of the request."
>     ^^^^^^^
> s/Timeout/Expires/?

Yes; the 'timeout' word just got propagated inappropriately.

> 
> The next sentence says:
>    "When forwarding responses, a proxy MUST forward all
>     header fields of the selected response."
> Shouldn't this read more along the lines of forwarding the whole
> message, since there could be useful information in the message
> body?

Proxies always forward the body, but I guess it doesn't hurt to belabor
the (supposedly) obvious. Changed.

> 
> The next sentence says:
>    "1xx: The proxy MAY forward the response upstream
>          towards the client."
> Would it be better to whack this up to "SHOULD" (as long as
> the response is not 100)?  (I guess this also applies to section
> 10.1.2.)

I'm not sure it makes a big difference in practice, but I believe that's
what most implementations do, so SHOULD seems appropriate.

> 
> Finally, is the notion of the "best response" strictly the lowest-
> numbered response (excluding 1xx), or is it more about the class
> of the response?  The reason I ask this is because if this is the
> case, then I don't feel that the order of status codes (especially
> in the 4xx region) is necessarily very helpful.
> 
> For instance, if a proxy forks to two locations, and one returns
> 486 and one returns 404, I would think that the 486 was a more
> useful response to return?

There are two cases: If the proxy "knows", it can apply whatever
optimization it deems desirable - this seems like an implementation
decision. In the absence of detailed status code knowledge, either
first-in-lowest-class or number seems about as good.

> 
> Further, say a proxy receives an INVITE with a Request-URI that
> is unknown (i.e., not in the proxy's domain/locationserver), and
> thus the proxy just tries to route as best it can based on some
> DNS resolution.  Now there's no server listening at the
> transport:IP:port that the proxy tries, so the proxy times-out [0].
> As per 12.4, I guess one would expect the proxy to return a 408
> response, but perhaps something like a 404 would be better?

I don't think you can distinguish the two with any degree of accuracy.
If I'm not in my office when you knock, and you wait five minutes, was I
"not found" or did you just "time out". Note that 404 indicates that you
have definitive information that I don't exist (at that address).
Proving the non-existence of people is hard. Just ask Elvis or the Loch
Ness monster.

> 
> Comments?
> 


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


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Jonathan Rosenberg wrote: 
>
> > I'm on the fence on this one, like Jonathan. Any suggestions how to reach
> > closure?
> 
> Well, one way would be to move the pros/cons from theory to something
> concrete. Can anyone come up with an example where using Contact would
> cause something to go wrong somewhere?

Yes: Contact already has some well-defined meaning in certain contexts,
e.g., in 300 responses or responses to REGISTER. If we were to reuse it
as a link to accessibility info, this would automatically mean that we
are disabling that info in 300 responses or responses to REGISTER. I see
no reason for this: defining a new header is no harder than overloading
the old one. It even has an added advantage: the new header doesn't have
to be defined in the base spec, which is thick enough as it is.

---
Igor Slepchin


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From sip-admin@lists.bell-labs.com  Mon Jun 12 17:02:11 2000
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Hi Folks,
  If one can specify a transport other than UDP using a transport-parameter,
then why does one HAVE TO support UDP in the first place?

It looks like the bis draft now means that one MUST support it, even if
only for an initial negotiation.

Within enterprise environments, I can imagine providing a system which
only supports TCP. No retransmit support, and using SSL/TLS for security
for ALL session setups.

I've been trying to see under what circumstances this is a problem.
Of course, if I plug in a UA that only supports UDP, then certainly
there's no success. However...this is not going to work anyway where
we want to use SSL/TLS (in current implementations, at least).

The new requirement for UDP support "come hell or high water" seems to make
the current discussion on transport-parameter negotiation moot. You have
to support UDP, so the initial negotiation will be done using UDP, so it's
expensive to change to TCP - setting up all calls this way ends up with
message exchanges worthy of H.323.

UDP is a fine transport, and SIP's efficient support for reliable exchanges
using UDP is one of its beauties, but making it MANDATORY for every
implementation is not a good move, IMHO.
-- 
lwc@roke.co.uk -- +44 1794 833666
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and that they would very much like people to sue me rather than
them if this email contains racist or sexist comments, please.
Also, I can't do anything; only our lawyers can, so speak to them.
Oh, and by the way, if our I.T. department has so mangled our
email system that this has been misdirected, beware that having
read this far, your eyes are about to fall out from reading
this highly sensitive information, unless you immediately
forget that this ever darkened your mailbox. Do it now.
-------------------------------------------------------------


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From: "Piotr S. Kossowski" <P.Kossowski@elka.pw.edu.pl>
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Hi,

About n-way conferences:

There is an example only 3-way conference
in draft-ietf-sip-call-flows-00.txt.
Did conisder anyone more participants (in any articles, drafts).

Especialy, What's going with RTP streams then ? 
(I think about difference between 2-way and 3-way or more
streaming)

BR, 
Piotr


-- 
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"Conroy, Lawrence (SMTP)" wrote:
> 
> Hi Folks,
>   If one can specify a transport other than UDP using a 
> transport-parameter, then why does one HAVE TO support UDP in the 
> first place?

So that two UACs can talk to each other without involoving a UDP->TCP
proxy.  In other words, if UDP is the de-facto standard that all SIP
entities MUST support, then any entity can talk with any other one.  If 
on the other hand, a UAC supports only UDP, then it can only talk to 
other UACs who support UDP.  It cannot talk to a UAC that only supports 
TCP without necessarily involving a proxy that does UDP->TCP con- 
version.
[...]
> UDP is a fine transport, and SIP's efficient support for reliable 
> exchanges using UDP is one of its beauties, but making it MANDATORY 
> for every implementation is not a good move, IMHO.

I belive most SIP implementations support UDP anyway.  I also believe
that under the new bis draft, UACs MUST support UDP and MAY support
TCP; proxies on the other hand MUST support both UDP and TCP.  

I am not sure how the new requirements of support SCTP, etc. will affect
the proxy.

Regards,

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
IN Architecture and Internet Software Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184


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From sip-admin@lists.bell-labs.com  Mon Jun 12 17:56:25 2000
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Date: Mon, 12 Jun 2000 22:56:08 +0100
To: vkg@lucent.com
From: Lawrence Conroy <lwc@roke.co.uk>
Subject: Re: [SIP] Transports again
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At 4:21 pm -0500 12/6/00, Vijay K. Gurbani wrote:
>
>I belive most SIP implementations support UDP anyway.  I also believe
>that under the new bis draft, UACs MUST support UDP and MAY support
>TCP; proxies on the other hand MUST support both UDP and TCP. 
>
>I am not sure how the new requirements of support SCTP, etc. will affect
>the proxy.
To which I reply
I agree that most SIP implementations support UDP.
I understand from your comment that, if one has a TCP-only UA, it's
non-standard. This is stronger than the original SHOULD for UDP support.
If one wants to use a UDP-only UA in an enterprise with a set of TCP-only
UAs then one has to go through a proxy - this was always the case and I'd
expect to have such a beast even where the vast majority of UAs use TCP.

The difference now is that the TCP-only UAs are non-standard.
                                                 ^^^^^^^^^^^^
-- 
All the best, Lawrence
-----------------------------------------------------------------------
| Lawrence Conroy,    | "These Opinions must be mine, 'cos if they    |
| Roke Manor Research |  were my Company's they'd pay me for them"    |
|- lwc@roke.co.uk  ---+- Tel: +44 1794 833666  Fax: +44 1794 833434 --|


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From sip-admin@lists.bell-labs.com  Mon Jun 12 17:57:00 2000
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"Conroy, Lawrence (SMTP)" wrote:
> You have
> to support UDP, so the initial negotiation will be done using UDP, so it's
> expensive to change to TCP - setting up all calls this way ends up with
> message exchanges worthy of H.323.

This isn't the case: there is no requirement for the protocol of initial
transaction in SIP. If you know that the callee supports TCP, SCTP,
pigeon carriers etc. (by examining callee's SRV record or by any other
means), you are free to use that protocol for initial transaction. The
requirement to implement UDP ensures that there is always a common
protocol for communication.

If a particular implementation/network configuration/PINT wants to use
TCP only due to any reason (security considerations, inline contents
etc), it is free to publish TCP as the preferred (or the only) protocol
in its SRV records or SIP URLs. If it still receives a UDP request, it
can either reject it or redirect with transport=tcp in the Contact.

---
Igor Slepchin


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From sip-admin@lists.bell-labs.com  Mon Jun 12 18:06:37 2000
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Date: Mon, 12 Jun 2000 18:06:11 -0400
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Subject: Re: [SIP] Transports again
References: <p04310101b56af7c6c6ec@[193.118.192.41]>
	 <39455451.91BF9CFE@lucent.com> <p04310101b56b0ae74a4d@[193.118.192.80]>
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Lawrence Conroy wrote:
> 
> At 4:21 pm -0500 12/6/00, Vijay K. Gurbani wrote:
> >
> >I belive most SIP implementations support UDP anyway.  I also believe
> >that under the new bis draft, UACs MUST support UDP and MAY support
> >TCP; proxies on the other hand MUST support both UDP and TCP.
> >
> >I am not sure how the new requirements of support SCTP, etc. will affect
> >the proxy.
> To which I reply
> I agree that most SIP implementations support UDP.
> I understand from your comment that, if one has a TCP-only UA, it's
> non-standard. This is stronger than the original SHOULD for UDP support.
> If one wants to use a UDP-only UA in an enterprise with a set of TCP-only
> UAs then one has to go through a proxy - this was always the case and I'd
> expect to have such a beast even where the vast majority of UAs use TCP.
> 
> The difference now is that the TCP-only UAs are non-standard.
>                                                  ^^^^^^^^^^^^

Yes, that's intentional. The last thing I want to happen is that you and
I both buy a SIP gadget, with a nice "Powered by SIP" logo pasted on the
box, we plug them in and we can't communicate. Internet protocol
implementations should always have a minimal way to communicate with
each other to be standards-compliant. That is a common theme for all
implementation aspects, from transport to security mechanisms. The
effort in supporting both is minimal. The only time it would conceivably
matter is for embedded systems - they would have a problem supporting
TCP, not UDP. Yes, you may spend a few hours extra building UDP logic
into your software. Standards compliance has its cost.

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


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From sip-admin@lists.bell-labs.com  Mon Jun 12 18:26:19 2000
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Date: Mon, 12 Jun 2000 18:23:37 -0400
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From: Jay Batson <jbatson@pingtel.com>
Subject: RE: [SIP] Distinctive ringing feature
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<trying to avoid shameless self promotion>

For those who are titillated by such behaviors as this, there's a 
SIP/Ethernet/Java phone that'll be available in the marketplace "real soon" 
to which this behavior can be added via a Java applet created by you or 
your favorite Java developer and which runs directly on the phone.

We've created ring files with a bunch of old Hanna  Barbera cartoon sound 
effects, like the Jetson's telephone ring sound, Fred Flintstone's 
bongo-feet-runaway, etc.  They do catch your attention better than an old 
fashioned ring.

Do our venture capitalists think it'll be the killer app?  Nahh-- they're 
too boring.  Now do *we* think it'll be cool?   What do you think...?  ;-)

</trying to avoid shameless self promotion>

<shameless self promotion>

Go see http://www.pingtel.com for more info.

<shamless self promotion>

-jb

At 09:53 AM 6/9/00 +0530, you wrote:


>why not ? :-)
>
>lots of interesting things can be put in.
>
>Since this is totally a client side implemenation that does not affect the
>SIP signalling in any way,
>we also did some cute(?!?) things like the ring volume being dynamically
>modulated as duration of ring increases etc (you know,
>like those alarms clocks that start with a whisper and keep screaming
>louder and louder (till a point) till you hit on the button....
>
>On the lighter side - here's some rings my limited imagination brings up:
>
>Girl/BoyFriend Ring - sweet chimes (or doomsday bongs - depending on what
>state you are with her/him)
>Beer pal ring - hic..hic...ring.ring...hic..hic
>Boss Ring - no ring :-p
>Wife Ring - continuous monotonic ring :-))
>etc......
>
>Oh well :-) Back to work - I enjoyed keying in this mail :-p
>
>Regds
>Arjun
>
>--
>Arjun Roychowdhury @ Hughes Software Systems
>Technical Leader, VOIP Solutions
>Off: Prestige Opal,146 Infantry Road,Blore - 560001  Ph:+91-080-2286390/1/2
>Res: 68/G 2nd Narayanappa Block, RT Nagar,Blore - 560032  Ph:3430565
>http://arjun.cjb.net
>
>
>Henry Sinnreich <Henry.Sinnreich@WCom.com> on 06/09/2000 02:25:28 AM
>
>To:   archow, Sekhar Banerjee <SBanerjee@lboard.com>
>cc:   SIP@lists.bell-labs.com
>
>Subject:  RE: [SIP] Distinctive ringing feature
>
>Could distinctive ringing be a caller/called preference?
>
>Thanks, Henry
>
> >-----Original Message-----
> >From: sip-admin@lists.bell-labs.com
> >[mailto:sip-admin@lists.bell-labs.com]On Behalf Of
> >archow@hss.hns.com
> >Sent: Wednesday, June 07, 2000 11:36 PM
> >To: Sekhar Banerjee
> >Cc: 'SIP@lists.bell-labs.com'
> >Subject: Re: [SIP] Distinctive ringing feature
> >
> >Hi,
> >you can definetely implement distinctive ringing from
> >a SIP endpoint.
> >The distinctive ring implementation is purely an
> >endpoint side feature (see
> >SIP FAQ on Henning's site)
> >and does not need any special SIP signalling.
> >For example, if say your phone has two alloted phone
> >numbers - and you want
> >a different ringing tone depending
> >on the number the caller dialed - your implementation
> >would simply see the
> >phone number from the SIP headers
> >and generate the appropriate ring for that phone.
> >
> >
> >Regds
> >Arjun
> >--
> >Arjun Roychowdhury @ Hughes Software Systems
> >
> >Sekhar Banerjee <SBanerjee@lboard.com> on 06/08/2000
> >02:52:13 AM
> >
> >To:   "'SIP@lists.bell-labs.com'" <SIP@lists.bell-labs.com>
> >cc:
> >
> >Subject:  [SIP] Distinctive ringing feature
> >
> >
> >
> >
> >Hi,
> >     I would like to know if it would be possible to
> >use SIP to instruct
> >the endpoints/UA's to provide different ringing
> >sequences. If someone has
> >implemented this feature I would love to know more
> >about it. I am talking
> >about the distinctive ringing feature as it is present
> >in the PSTN network
> >today.
> >
> >Thanks
> >
> >Sekhar




PingMe!
Jay Batson
President, CEO
Pingtel Corp.
jbatson@pingtel.com

781-938-5306 (office)
781-938-9650 (fax)



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Date: Mon, 12 Jun 2000 23:33:44 +0100
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
From: Lawrence Conroy <lwc@roke.co.uk>
Subject: Re: [SIP] Transports again
Cc: vkg@lucent.com, sip@lists.bell-labs.com
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At 6:06 pm -0400 12/6/00, Henning Schulzrinne wrote:
>  > The difference now is that the TCP-only UAs are non-standard.
>  >                                                  ^^^^^^^^^^^^
>Yes, that's intentional. The last thing I want to happen is that you and
>I both buy a SIP gadget, with a nice "Powered by SIP" logo pasted on the
>box, we plug them in and we can't communicate. Internet protocol
>implementations should always have a minimal way to communicate with
>each other to be standards-compliant. That is a common theme for all
>implementation aspects, from transport to security mechanisms. The
>effort in supporting both is minimal. The only time it would conceivably
>matter is for embedded systems - they would have a problem supporting
>TCP, not UDP. Yes, you may spend a few hours extra building UDP logic
>into your software. Standards compliance has its cost.

Hmm... embedded systems may well have as hard a time supporting the
retransmit-logic and storage implied by SIP-over-UDP - That software
is not *yet* available completely off the shelf, whilst TCP stacks are.
It's not the development time that's a problem here - it's the size and
application processing requirements of the eventual product.

I've realised that it may be my misunderstanding of the word SHOULD.
I had thought that this could be translated as "you had better do this
unless you have a good reason for doing otherwise".

If I buy a "Powered by SIP" (apologies if this is someone's trademark :)
device, then I would expect it to talk over UDP, unless it states that
it only talks over TCP. I just believe that a TCP-only device should be
a valid sub-set - but any such product had better state that it supports
only this sub-set. Isn't that the sense of SHOULD?

-- 
All the best, Lawrence
-----------------------------------------------------------------------
| Lawrence Conroy,    | "These Opinions must be mine, 'cos if they    |
| Roke Manor Research |  were my Company's they'd pay me for them"    |
|- lwc@roke.co.uk  ---+- Tel: +44 1794 833666  Fax: +44 1794 833434 --|


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> 
> Hmm... embedded systems may well have as hard a time supporting the
> retransmit-logic and storage implied by SIP-over-UDP - That software
> is not *yet* available completely off the shelf, whilst TCP stacks are.
> It's not the development time that's a problem here - it's the size and
> application processing requirements of the eventual product.

Speaking from personal experience, with a lab that has developed a stack
for both networking and SIP: TCP would have been far more complex than
the trivial retransmit timer in SIP. For us, that's as much a code space
problem as a development time problem.

> 
> I've realised that it may be my misunderstanding of the word SHOULD.
> I had thought that this could be translated as "you had better do this
> unless you have a good reason for doing otherwise".
> 
> If I buy a "Powered by SIP" (apologies if this is someone's trademark :)

It should be, with a nice logo to go along with it :-)

> device, then I would expect it to talk over UDP, unless it states that
> it only talks over TCP. I just believe that a TCP-only device should be
> a valid sub-set - but any such product had better state that it supports
> only this sub-set. Isn't that the sense of SHOULD?

My perception is that items crucial for basic interoperability are
labeled MUST, while those more on the margins can be SHOULD.


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From sip-admin@lists.bell-labs.com  Mon Jun 12 23:26:45 2000
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Date: Mon, 12 Jun 2000 23:26:00 -0400
From: Faramak Vakil <farm@research.telcordia.com>
Organization: Telcordia Technologies
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To: Ashutosh Dutta <adutta@telcordia.com>
Cc: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>, archow@hss.hns.com,
        sip@lists.bell-labs.com
Subject: Re: [SIP] mobility support in SIP - basic questions
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Ashutosh,

MWIF has not considered/discussed any protocol for
mobility management so far, and the statement about
SIP mobility in MWIF is not accurate.

Regards! Faramak

Ashutosh Dutta wrote:

> Besides 3GPP, there is also some interest on SIP mobility in Mobile Wireless
> Internet Forum (www.mwif.org) which has started  recently. SIP-2000 in Paris had
> a presentation on SIP mobility which addresses some SIP mobility issues as well.
>
> Besides the original work on SIP mobility by Henning and Elin, there are couple
> of related drafts in IETF for issues  covering CDMA soft-handoff, and HMMP.
>
> Thanks
> Ashutosh
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip



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From sip-admin@lists.bell-labs.com  Mon Jun 12 23:44:23 2000
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Subject: Re: [SIP] Transports again
References: <p04310101b56af7c6c6ec@[193.118.192.41]>
		 <39455451.91BF9CFE@lucent.com> <p04310101b56b0ae74a4d@[193.118.192.80]>
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Henning Schulzrinne wrote:
> 
> > device, then I would expect it to talk over UDP, unless it states that
> > it only talks over TCP. I just believe that a TCP-only device should be
> > a valid sub-set - but any such product had better state that it supports
> > only this sub-set. Isn't that the sense of SHOULD?
> 
> My perception is that items crucial for basic interoperability are
> labeled MUST, while those more on the margins can be SHOULD.

SHOULD means an implementation can not do it and still be compliant.

As has been pointed out, you can still build your network which always
uses TCP/TLS when talking to each other. No UDP under normal conditions.
But, you just need to be prepared for that UDP call from someone elses
phone.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Mon Jun 12 23:51:12 2000
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Igor Slepchin wrote:
> 
> Jonathan Rosenberg wrote:
> >
> > > I'm on the fence on this one, like Jonathan. Any suggestions how to reach
> > > closure?
> >
> > Well, one way would be to move the pros/cons from theory to something
> > concrete. Can anyone come up with an example where using Contact would
> > cause something to go wrong somewhere?
> 
> Yes: Contact already has some well-defined meaning in certain contexts,
> e.g., in 300 responses or responses to REGISTER. If we were to reuse it
> as a link to accessibility info, this would automatically mean that we
> are disabling that info in 300 responses or responses to REGISTER. I see
> no reason for this: defining a new header is no harder than overloading
> the old one. It even has an added advantage: the new header doesn't have
> to be defined in the base spec, which is thick enough as it is.

The big question is whether the current usage in 300 responses isn't
what Dave wants anyway. I see no problem in returning an RTSP URL in a
300 redirect. This 300 is forwarded up to the UAC, which can act on the
URL and fetch the streaming media content. No different than if the URL
where an HTTP URL.

One issue is if the UA wants to present the fetched data in some way
which is different because the UA knows that the data is for the
purposes of error reporting. Another issue, perhaps, is if a UA thinks
that redirects are kind of optional, and so it ignores Contact headers
in 300 class responses. These special error messages are probably more
important and should be fetched by the UA.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Tue Jun 13 00:16:39 2000
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Hi, thought Id put in my 2c worth:
(responses preceded by ARC> )

>
> Hmm... embedded systems may well have as hard a time supporting the
> retransmit-logic and storage implied by SIP-over-UDP - That software
> is not *yet* available completely off the shelf, whilst TCP stacks are.
> It's not the development time that's a problem here - it's the size and
> application processing requirements of the eventual product.

Speaking from personal experience, with a lab that has developed a stack
for both networking and SIP: TCP would have been far more complex than
the trivial retransmit timer in SIP. For us, that's as much a code space
problem as a development time problem.

ARC> We have done a lot of porting of the SIP Stack on various embedded
systems
ARC> and the retransmission logic/ availability of a UDP stack was of no
concern
ARC> at all. As far as the retrans logic was concerned, it did not affect
performance
ARC> or run time overhead enough for us to look back at it.
ARC> However,  I would like to know of common RTOSes which dont have UDP
ARC> support (so I can be prepared when we attack that RTOS next ;-)

>

> device, then I would expect it to talk over UDP, unless it states that
> it only talks over TCP. I just believe that a TCP-only device should be
> a valid sub-set - but any such product had better state that it supports
> only this sub-set. Isn't that the sense of SHOULD?


My perception is that items crucial for basic interoperability are
labeled MUST, while those more on the margins can be SHOULD.

ARC> Couldnt agree more. Its great to know that we have atleast one
protocol
ARC> which we know will be supported by others. If both are made SHOULD, im
pretty
ARC> sure in the next bakeoff we will have a lot of new ones. half
supporting only TCP
ARC> and the other only UDP who wont be able to talk to each other. I do
understand
ARC> your point that why use UDP if my network doesnt need it ? Its a
tussle b/w
ARC> your needs and making a minimal set of SIP fixed so that in broader
interworking
ARC> scenarios, there is lesser of ambiguity. If you say your n/w doesnt
need UDP
ARC> how about doing only TCP, and returning a final response Error of
something
ARC> like 300 (Protocol not allowed) (actually that was for SDP - sometinng
similar ?)
ARC> so that the sending terminal at least knows you are up and dont accept
UDP without
ARC> him trying to figure out if you are really up, or putting in extra
logic at the
ARC> sender to try TCP next... Not sure if this would be of any help though
(?)
ARC> However, im against suggesting this behaviour in the base spec for
reasons I mentioned
ARC> previously.

Regds
Arjun
--
Arjun Roychowdhury @ Hughes Software Systems










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From sip-admin@lists.bell-labs.com  Tue Jun 13 00:37:11 2000
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Subject: Re: [SIP] Forking Proxy Behaviour Question
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Henning Schulzrinne wrote:
> 
> >
> > The next sentence says:
> >    "1xx: The proxy MAY forward the response upstream
> >          towards the client."
> > Would it be better to whack this up to "SHOULD" (as long as
> > the response is not 100)?  (I guess this also applies to section
> > 10.1.2.)
> 
> I'm not sure it makes a big difference in practice, but I believe that's
> what most implementations do, so SHOULD seems appropriate.

Actually, you really want to make this a SHOULD, since reliable
provisional responses may not work otherwise.

> 
> >
> > Finally, is the notion of the "best response" strictly the lowest-
> > numbered response (excluding 1xx), or is it more about the class
> > of the response?  The reason I ask this is because if this is the
> > case, then I don't feel that the order of status codes (especially
> > in the 4xx region) is necessarily very helpful.

Hmm... actually, I always thought that the ordering was:

200 better than 600 better than 300 better than 400 better than 500

which is different from the code. A 600 response is a fairly strong
statement, and I think it makes sense to make it higher than 300-599.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Tue Jun 13 01:08:36 2000
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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
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I have a query about the CSeq matching logic.

I don't think I have seen anything in the INFO or 2543bis drafts explicitly
forbidding having multiple outstanding requests: eg a re-INVITE and INFO
both outstanding. If this is the case then the CSeq matching logic is flawed
as you cannot rely on the value CSeq monotonically increasing.

Section 11.5 of RFC2543bis "Receiving Subsequent Requests" states that if
the Cseq doesn 't match any previous requests AND the CSeq is higher than
the last received value, then it is a new request.

This doesn't work if can you have multiple outstanding requests. For
instance, consider the case where one side sends an INFO request (eg
application/mgcp-event) and then shortly after sends a re-INVITE.
[Independent mid-call signaling is most immediate use I can see for multiple
requests]. Now consider if the first INFO request UDP message is lost but
the INVITE is not -> the INVITE can arrive before the INFO retransmission.
Using the above logic, the UAS will now never respond to the INFO request.

I remember on Henning's page, a CSeq clarification alluded to disallowing
multiple outstanding requests but my interpretation of that (after
subsequent discussion) was that this was referring to INVITE responses from
either side (ie the glare condition).

If I have missed some crucial piece of text then I think it should be moved
to a more logical place in the draft (eg Section 11). Do we need multiple
outstanding requests (per UA) anyway ? It certainly complicates matters
significantly but also has a number of benefits!!
It opens a whole can of worms: how many simultaneous requests? one per
method? how do you handle CSeq? 

Comments?

Robert.


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From sip-admin@lists.bell-labs.com  Tue Jun 13 09:59:37 2000
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> > (1) 400-600-class failure [problem with the TRANSFER request]
> > (2) 200 OK [transfer attempted and succeeded]
> > (3) ??? [response that specifically means transfer attempted but failed]

[Jonathan R. responds:]
> I would say this is a 500 class response. I believe 500 class responses
> generally mean, "this request failed this time, but the same request
> might succeed later on". That's exactly the case here. You could even
> include a Retry-After in the response indicating when to try.

With the Retry-After probably copied from the response to the failed
transfer INVITE, if present.  I say "probably" because one could likely
contrive an example where the transferee has additional knowledge that
enables it to generate a better Retry-After value.  But the normal case
could be to copy the Retry-After value, in effect saying, "I tried the
transfer and it failed; they told me to try again in this many minutes; send
me another TRANSFER then if you'd like me to do that."

Tim Schroeder
tschroeder@tri.sbc.com
SBC Technology Resources, 9505 Arboretum Blvd., Austin, TX  78759


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From sip-admin@lists.bell-labs.com  Tue Jun 13 11:46:20 2000
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Are these suppose to be two distinct response messages? If so, it would be
less confusing if
the text also related that these are two unique responses.

William Lee
williaml@nortelnetworks.com

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<P><FONT SIZE=2 FACE="Arial">Are these suppose to be two distinct response messages? If so, it would be less confusing if</FONT>
<BR><FONT SIZE=2 FACE="Arial">the text also related that these are two unique responses.</FONT>
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<P><FONT SIZE=2 FACE="Arial">William Lee</FONT>
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From sip-admin@lists.bell-labs.com  Tue Jun 13 16:13:16 2000
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Hello all,

There have been a couple IDs written a short while back on DTMF transport,
collection, etc. each using the SIP INFO method.  I see out-of-band DTMF
reporting as a "problem" to be solved to support carrying telecommunications
traffic over SIP networks.  IMO, Out-of-band "event" reporting is needed to
minimize latency and provide services to PSTN-based users within SIP
proxy/application servers that do not also "own" the gateways.  MGCP and
Megaco compliant gateways support DTMF collection and out-of-band reporting
but there is no "agreed upon" method or interworking proposal in SIP/SIP-T
that I have found.

Is there a solution to this "problem" that I have missed or are others
interested in addressing this area?  As of this week, I've asked for a
"slot" at the upcoming IETF meeting to present one ID on the subject but
don't want to take others' time if there's no need.

Thanks,
Bert


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From sip-admin@lists.bell-labs.com  Tue Jun 13 16:19:12 2000
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Subject: Re: [SIP] mobility support in SIP - basic questions
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Thanks Faramak for clarifying that.

My impression about MWIF's interest about SIP was based on the  two MWIF
references that were mentioned in SIP-2000 presentation. But those date back to
January though.

Thanks
Ashutosh




Faramak Vakil <farm@research.telcordia.com> on 06/12/2000 11:26:00 PM

To:   Ashutosh Dutta <adutta@telcordia.com>
cc:   Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>, archow@hss.hns.com,
      sip@lists.bell-labs.com (bcc: Ashutosh Dutta/Telcordia)
Subject:  Re: [SIP] mobility support in SIP - basic questions




Ashutosh,

MWIF has not considered/discussed any protocol for
mobility management so far, and the statement about
SIP mobility in MWIF is not accurate.

Regards! Faramak

Ashutosh Dutta wrote:

> Besides 3GPP, there is also some interest on SIP mobility in Mobile Wireless
> Internet Forum (www.mwif.org) which has started  recently. SIP-2000 in Paris
had
> a presentation on SIP mobility which addresses some SIP mobility issues as
well.
>
> Besides the original work on SIP mobility by Henning and Elin, there are
couple
> of related drafts in IETF for issues  covering CDMA soft-handoff, and HMMP.
>
> Thanks
> Ashutosh
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip







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From sip-admin@lists.bell-labs.com  Tue Jun 13 18:09:10 2000
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Subject: Re: [SIP] Forking Proxy Behaviour Question
References: <000c01bfd47f$067fff10$4e34c3c1@ubiquity.co.uk> <39451377.1B6201E8@cs.columbia.edu> <3945B9A9.88605438@dynamicsoft.com>
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Jonathan Rosenberg wrote:
> 
> Henning Schulzrinne wrote:

> 
> Actually, you really want to make this a SHOULD, since reliable
> provisional responses may not work otherwise.

Changed to SHOULD.

> 
> >
> > >
> > > Finally, is the notion of the "best response" strictly the lowest-
> > > numbered response (excluding 1xx), or is it more about the class
> > > of the response?  The reason I ask this is because if this is the
> > > case, then I don't feel that the order of status codes (especially
> > > in the 4xx region) is necessarily very helpful.
> 
> Hmm... actually, I always thought that the ordering was:
> 
> 200 better than 600 better than 300 better than 400 better than 500
> 
> which is different from the code. A 600 response is a fairly strong
> statement, and I think it makes sense to make it higher than 300-599.
> 

Maybe I'm missing something, but I read the code as: If a class '6'
arrives, best is set to this response and the search loop is terminated
by 'break', so that the 6xx is returned.


      else if (class == 6) {
        best = r;
        break;
      }


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


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--------------C9FF0E6D9ADC0203C2F25F28
Content-Type: text/plain; charset=us-ascii
Content-Transfer-Encoding: 7bit

Hi:

If a message goes through a proxy twice , ex:


client A----------> proxy1-------------> proxy 2
-------------------------->    proxy1.
                 sends  invite                forwards
this                           sends this back to proxy 1



proxy1----------------> client B.


In the above scenario, proxy 1 needs to distinguish between clientA-->
proxy1 INVITE, and proxy2--->proxy1 INVITE. This can be done in SIP by
putting a different branch in the Via.

My concern was that in the standard it is said that when a proxy sees
its own address in the Via, we need to
remove it , otherwise this may cause a loop. Now if we change the
branch, this will indeed not cause the loop.

However, I wanted to know if there was any other repurcussions if a
proxy does  this.  The SIP examples or the RFC as such does not mention
this scenario any where.

Thanks much !

--
Sunitha Kumar
http://www.vovida.com



--------------C9FF0E6D9ADC0203C2F25F28
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Hi:
<p>If a message goes through a proxy twice , ex:
<br>&nbsp;
<p>client A----------> proxy1-------------> proxy 2 -------------------------->&nbsp;&nbsp;&nbsp;
proxy1.
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
sends&nbsp; invite&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
forwards this&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
sends this back to proxy 1
<br>&nbsp;
<br>&nbsp;
<p>proxy1----------------> client B.
<br>&nbsp;
<p>In the above scenario, proxy 1 needs to distinguish between clientA-->
proxy1 INVITE, and proxy2--->proxy1 INVITE. This can be done in&nbsp;SIP
by putting a different branch in the Via.
<p>My concern was that in the standard it is said that when a proxy sees
its own address in the Via, we need to
<br>remove it , otherwise this may cause a loop. Now if we change the branch,
this will indeed not cause the loop.
<p>However, I&nbsp;wanted to know if there was any other repurcussions
if a proxy does&nbsp; this.&nbsp; The SIP examples or the RFC as such does
not mention this scenario any where.
<p>Thanks much !
<pre>--&nbsp;
Sunitha Kumar
<A HREF="http://www.vovida.com">http://www.vovida.com</A></pre>
&nbsp;</html>

--------------C9FF0E6D9ADC0203C2F25F28--



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From sip-admin@lists.bell-labs.com  Tue Jun 13 23:57:52 2000
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Date: Tue, 13 Jun 2000 23:58:19 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
Cc: "'sip@lists.bell-labs.com'" <sip@lists.bell-labs.com>
Subject: Re: [SIP] CSeq matching and multiple requests.
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"Fairlie-Cuninghame, Robert" wrote:
> 
> I have a query about the CSeq matching logic.
> 
> I don't think I have seen anything in the INFO or 2543bis drafts explicitly
> forbidding having multiple outstanding requests: eg a re-INVITE and INFO
> both outstanding. If this is the case then the CSeq matching logic is flawed
> as you cannot rely on the value CSeq monotonically increasing.

It has been discussed, but I don't think a concrete conclusion was ever
reached on whether it was allowed.

> 
> Section 11.5 of RFC2543bis "Receiving Subsequent Requests" states that if
> the Cseq doesn 't match any previous requests AND the CSeq is higher than
> the last received value, then it is a new request.
> 
> This doesn't work if can you have multiple outstanding requests. For
> instance, consider the case where one side sends an INFO request (eg
> application/mgcp-event) and then shortly after sends a re-INVITE.
> [Independent mid-call signaling is most immediate use I can see for multiple
> requests]. Now consider if the first INFO request UDP message is lost but
> the INVITE is not -> the INVITE can arrive before the INFO retransmission.
> Using the above logic, the UAS will now never respond to the INFO request.

This is true assuming the behavior for requests that are not new is to
ignore them. If you are supporting multiple outstanding requests, you
would not ignore such requests. You could determine them to be
retransmissions if the CSeq matched a CSeq you had seen before, "old" if
the CSeq is lower than the highest received CSeq, and "new" if its
higher than the highest received CSeq. Both old and new requests would
need to be processed.


> If I have missed some crucial piece of text then I think it should be moved
> to a more logical place in the draft (eg Section 11). Do we need multiple
> outstanding requests (per UA) anyway ? 

It would be nice to hear if anyone has any concrete examples of a need
for this.

> It certainly complicates matters
> significantly but also has a number of benefits!!

No doubt, but nothing that isn't solvable.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Wed Jun 14 00:14:21 2000
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To: Sunitha Kumar <skumar@vovida.com>
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Subject: Re: [SIP] Clarification on the proxy behaviour.
References: <3946D8F3.86230CC6@vovida.com>
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This issue has been discussed numerous times on the list. It is referred
to as request "spiraling" and is not a loop. The mechanisms for
constructing the Via to support this are defined in the bis draft,
section 6.46. The text still needs some work though; a request is only a
loop if your IP address is there, AND, when you compute the hash over
the request URI, To, From, Call-ID, it matches the component in the
branch parameter.

-Jonathan R.


Sunitha Kumar wrote:
> 
> Hi:
> 
> If a message goes through a proxy twice , ex:
> 
> 
> client A----------> proxy1-------------> proxy 2
> -------------------------->    proxy1.
>                  sends  invite                forwards
> this                           sends this back to proxy 1
> 
> 
> 
> proxy1----------------> client B.
> 
> 
> In the above scenario, proxy 1 needs to distinguish between clientA-->
> proxy1 INVITE, and proxy2--->proxy1 INVITE. This can be done in SIP by
> putting a different branch in the Via.
> 
> My concern was that in the standard it is said that when a proxy sees
> its own address in the Via, we need to
> remove it , otherwise this may cause a loop. Now if we change the
> branch, this will indeed not cause the loop.
> 
> However, I wanted to know if there was any other repurcussions if a
> proxy does  this.  The SIP examples or the RFC as such does not
> mention this scenario any where.
> 
> Thanks much !
> 
> --
> Sunitha Kumar
> http://www.vovida.com
> 
> 

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Wed Jun 14 00:23:24 2000
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Date: Wed, 14 Jun 2000 00:23:20 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Cc: Jo Hornsby <jhornsby@ubiquity.net>, sip@lists.bell-labs.com
Subject: Re: [SIP] Forking Proxy Behaviour Question
References: <000c01bfd47f$067fff10$4e34c3c1@ubiquity.co.uk> <39451377.1B6201E8@cs.columbia.edu> <3945B9A9.88605438@dynamicsoft.com> <3946B093.4C71B5C2@cs.columbia.edu>
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Henning Schulzrinne wrote:
> 
> Maybe I'm missing something, but I read the code as: If a class '6'
> arrives, best is set to this response and the search loop is terminated
> by 'break', so that the 6xx is returned.
> 
>       else if (class == 6) {
>         best = r;
>         break;
>       }

You're right. Too much blood in my caffeine when I read that code again
;)

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Wed Jun 14 02:37:45 2000
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Date: Wed, 14 Jun 2000 09:36:50 +0300
From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
Organization: Oy L M Ericsson Ab
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Subject: [SIP] [Fwd: Status of SDPng]
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Hi,

I sent this mail to the MMUSIC mailing list but I got no answer... maybe
here somebody knows...

Thanks,

Gonzalo

-------- Original Message --------
Subject: Status of SDPng
Date: Mon, 12 Jun 2000 09:29:44 +0300
From: gonzalo.camarillo@lmf.ericsson.se
Organization: Oy L M Ericsson Ab
To: mmusic <confctrl@ISI.EDU>

Hi,

I would like to know the status of the so called, SDPng. Is there any
draft available about it? Are there folks working on it? Time frame for
this work, goals...

Thanks,

Gonzalo
-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland


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From sip-admin@lists.bell-labs.com  Wed Jun 14 02:54:33 2000
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From: "rafi" <rafi@sasi.com>
To: <sip@lists.bell-labs.com>
Date: Wed, 14 Jun 2000 12:12:56 +0530
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Subject: [SIP] Handling multiple Transfer Target
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Hi,

           Is it possible for the transferor to inform the transferee
 to try more than one transfer target serially or simultaneously?
 Probably one way to do is to include more than one Transfer-to-Header
 in the TRANSFER request message because transferee take decision=20
based on this field  (section 4.6 of draft  =
draft-sparks-sip-cc-transfer-00.txt).
 May be order of the of each name-addr / addr-spec in  =
Transfer-to-Header implies=20
their priority. Having more than one Transfer-to-Header header
is not allowed in the draft, draft-sparks-sip-cc-transfer-00.txt.

     Please note that contact field in the TRANSFER method is optional. =
So
my understanding is that only Transfer-to-Header is used to identify the =
transfer target.


Thanks
Rafi AssadiH.M.
Silicon Automation Systems
Bangalore, INDIA




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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2614.3500" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Hi,</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Is =
it=20
possible for the transferor to inform the</FONT><FONT face=3DArial =
size=3D2>=20
transferee</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;to try more than one transfer=20
target</FONT><FONT face=3DArial size=3D2> serially or =
simultaneously?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;Probably one way to </FONT><FONT =
face=3DArial=20
size=3D2>do is to include more than one Transfer-to-Header</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;in</FONT><FONT face=3DArial =
size=3D2> the=20
TRANSFER request message</FONT><FONT face=3DArial size=3D2> because =
transferee take=20
decision </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>based on this field&nbsp; (section 4.6 =
of=20
draft&nbsp; draft-sparks-sip-cc-transfer-00.txt).</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;May be order of the =
of</FONT><FONT face=3DArial=20
size=3D2> each name-addr / addr-spec in&nbsp; Transfer-to-Header implies =

</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>their priority.</FONT><FONT =
face=3DArial size=3D2>=20
Having more than one Transfer-to-Header header</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>is not allowed in the draft,=20
draft-sparks-sip-cc-transfer-00.txt.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp;&nbsp; Please note =
that contact=20
field in the TRANSFER method is optional. So</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>my understanding is that only =
Transfer-to-Header is=20
used to identify the transfer target.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Rafi AssadiH.M.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Silicon Automation Systems</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Bangalore, INDIA</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

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From: Neil Deason <ndeason@ubiquity.net>
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Cc: Jo Hornsby <jhornsby@ubiquity.net>, sip@lists.bell-labs.com
Subject: Re: [SIP] Forking Proxy Behaviour Question
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Henning Schulzrinne wrote:
> 
> Jo Hornsby wrote:

[snip]

> > Finally, is the notion of the "best response" strictly the lowest-
> > numbered response (excluding 1xx), or is it more about the class
> > of the response?  The reason I ask this is because if this is the
> > case, then I don't feel that the order of status codes (especially
> > in the 4xx region) is necessarily very helpful.
> >
> > For instance, if a proxy forks to two locations, and one returns
> > 486 and one returns 404, I would think that the 486 was a more
> > useful response to return?
> 
> There are two cases: If the proxy "knows", it can apply whatever
> optimization it deems desirable - this seems like an implementation
> decision. In the absence of detailed status code knowledge, either
> first-in-lowest-class or number seems about as good.

Is it clear that this is the intention? The text in 12.4 about
sending back responses only refers to returning lowest-numbered
responses within the 3xx, 4xx, 5xx categories. While the "C-code
[which] describes the behaviour of a proxy" would send back the
last 4xx or 5xx class response it receives, if there was nothing
lower.

Cheers,
Neil
-- 
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From sip-admin@lists.bell-labs.com  Wed Jun 14 10:00:11 2000
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What would you like to see?  I don't think that agreement on the 
MIME-type is going to be easy to get.  Your draft suggested MGCP
events, others might prefer megaco.  A SIP phone may not speak
either, but if it uses RFC2833 it would be easy to send audio/tone 
MIME-type inside of INFO.  We already have the Accept headers for
this kind of thing.  Would you like to see an option-tag defined
for it?

Regards,

Dave Walker
SS8 Networks
Ottawa, Canada.

"Culpepper, Bert" wrote:
> 
> Hello all,
> 
> There have been a couple IDs written a short while back on DTMF transport,
> collection, etc. each using the SIP INFO method.  I see out-of-band DTMF
> reporting as a "problem" to be solved to support carrying telecommunications
> traffic over SIP networks.  IMO, Out-of-band "event" reporting is needed to
> minimize latency and provide services to PSTN-based users within SIP
> proxy/application servers that do not also "own" the gateways.  MGCP and
> Megaco compliant gateways support DTMF collection and out-of-band reporting
> but there is no "agreed upon" method or interworking proposal in SIP/SIP-T
> that I have found.
> 
> Is there a solution to this "problem" that I have missed or are others
> interested in addressing this area?  As of this week, I've asked for a
> "slot" at the upcoming IETF meeting to present one ID on the subject but
> don't want to take others' time if there's no need.
> 
> Thanks,
> Bert
>


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From sip-admin@lists.bell-labs.com  Wed Jun 14 10:10:15 2000
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Hi all,

The rcf2543bis states in section 6.45.2

"Every host that sends or forwards a SIP request adds a Via field indicating the
host’s address. However, it is possible that Network Address Translators (NATs)
change the source address and port of the request (e.g.,from a net-10 to a
globally routable address), in which case the Via header field cannot be relied
on to route replies. To prevent this, a proxy SHOULD check the top-most Via
header field to ensure that it contains the sender’s correct network address, as
seen from that proxy. If the sender’s address is incorrect, the proxy MUST add a
“received” parameter to the Via header field inserted by the previous hop. Such
a modified Via header field is known as a receiver-tagged Via header field."

So, it is possible that the port is also changed.  Yet the syntax for
via-received looks like this:

via-received = ”received” ”=” host

shouldn't  it be

via-received = ”received” ”=” hostport
or
via-received = ”received” ”=” host [":" port]

Thanks,
Hisham



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From sip-admin@lists.bell-labs.com  Wed Jun 14 10:20:56 2000
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        SIP List <sip@lists.bell-labs.com>
Subject: Re: [SIP] Out-of-Band DTMF in SIP
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Dave Walker wrote:
> 
> What would you like to see?  I don't think that agreement on the
> MIME-type is going to be easy to get.  Your draft suggested MGCP
> events, others might prefer megaco.  A SIP phone may not speak
> either, but if it uses RFC2833 it would be easy to send audio/tone
> MIME-type inside of INFO.  We already have the Accept headers for
> this kind of thing.  Would you like to see an option-tag defined
> for it?
> 

While this not be a bad idea (I might be biased, here...), it does need
a bit more work, since carrying RTP packets in INFO would require some
way of mapping RTP timestamps to absolute time [RTCP does this] and,
less importantly, decide what significance the payload type (PT) field
has in this context. Obviously, if you only need a duration, not
absolute timing, sticking RTP packets into INFO is sufficient. However,
you'd probably not want to use every such RTP packet, only when a new
digit has been detected.


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


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From sip-admin@lists.bell-labs.com  Wed Jun 14 10:25:24 2000
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Subject: Re: [SIP] Receiver-tagged Via header
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Correct, this is noted towards the end of Appendix G - Changes To
Be Made as	"recieved-port for NAPT support in Via header field".

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

Hisham Khartabil wrote:
> 
> Hi all,
> 
> The rcf2543bis states in section 6.45.2
> 
> "Every host that sends or forwards a SIP request adds a Via field indicating the
> host’s address. However, it is possible that Network Address Translators (NATs)
> change the source address and port of the request (e.g.,from a net-10 to a
> globally routable address), in which case the Via header field cannot be relied
> on to route replies. To prevent this, a proxy SHOULD check the top-most Via
> header field to ensure that it contains the sender’s correct network address, as
> seen from that proxy. If the sender’s address is incorrect, the proxy MUST add a
> “received” parameter to the Via header field inserted by the previous hop. Such
> a modified Via header field is known as a receiver-tagged Via header field."
> 
> So, it is possible that the port is also changed.  Yet the syntax for
> via-received looks like this:
> 
> via-received = ”received” ”=” host
> 
> shouldn't  it be
> 
> via-received = ”received” ”=” hostport
> or
> via-received = ”received” ”=” host [":" port]
> 
> Thanks,
> Hisham
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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From sip-admin@lists.bell-labs.com  Wed Jun 14 10:29:49 2000
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From: Billy Biggs <vektor@div8.net>
To: Hisham Khartabil <hisham.khartabil@lmf.ericsson.se>
Cc: "'sip@lists.bell-labs.com'" <sip@lists.bell-labs.com>
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Hisham Khartabil (hisham.khartabil@lmf.ericsson.se):

> "Every host that sends or forwards a SIP request adds a Via field indicating the
> host’s address. However, it is possible that Network Address Translators (NATs)
> change the source address and port of the request (e.g.,from a net-10 to a
> globally routable address), in which case the Via header field cannot be relied
> on to route replies. To prevent this, a proxy SHOULD check the top-most Via
> header field to ensure that it contains the sender’s correct network address, as
> seen from that proxy. If the sender’s address is incorrect, the proxy MUST add a
> “received” parameter to the Via header field inserted by the previous hop. Such
> a modified Via header field is known as a receiver-tagged Via header field."
> 
> So, it is possible that the port is also changed.  Yet the syntax for
> via-received looks like this:
> 
> via-received = "received" "=" host
> 
> shouldn't  it be
> 
> via-received = "received" "=" hostport
> or
> via-received = "received" "=" host [":" port]

  Wouldn't that be nice.  You could put it in, but it won't help the NAPT
problem.

  Unfortunately, responses in SIP always go back to the port listed in
the sent-by of the Via, NOT the source of the message.  This means you
are free to open a UDP socket just to send one message and then nuke the
socket (since the response will come back on your listening socket).

  This means that you don't care what port you received it on, you just
care what port is listed in the sent-by of the Via.  If the SIP message
went through an NAPT, then the sent-by will list the wrong port anyways,
and there won't be much use in indicating which port the message came
from.

  The solution would be to recommend that clients use the same socket
(source port) that they receive messages in as the socket they send
messages from, and therefore sending a response to the source port will
'usually' work.

  It might be worth it.....

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


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From sip-admin@lists.bell-labs.com  Wed Jun 14 10:39:22 2000
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Neil Deason (ndeason@ubiquity.net):

> Correct, this is noted towards the end of Appendix G - Changes To
> Be Made as	"recieved-port for NAPT support in Via header field".

  Cute, I hadn't noticed that.

  The spec would have to be changed to suggest that clients always send
from the port that they intend to receive the message on, where possible.
That breaks stuff pretty bad.

  I know my client (for one) opens a new socket for every outgoing
message, and then closes it after.  This is very very bad, and actually
breaks my own NAT ALG (Linux doesn't like it when you do this).  I'm
going to fix my client, and I guess it's probably a good idea if we
discourage unnecessary source port changes.

  Thoughts?

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


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From:  rafi
Hi,
 
           Is it possible for the transferor to inform the transferee
 to try more than one transfer target serially or simultaneously?
 Probably one way to do is to include more than one Transfer-to-Header
 in the TRANSFER request message because transferee take decision 
based on this field  (section 4.6 of draft
draft-sparks-sip-cc-transfer-00.txt).
 May be order of the of each name-addr / addr-spec in  Transfer-to-Header
implies 
their priority. Having more than one Transfer-to-Header header
is not allowed in the draft, draft-sparks-sip-cc-transfer-00.txt.
 
     Please note that contact field in the TRANSFER method is optional. So
my understanding is that only Transfer-to-Header is used to identify the
transfer target.
 
 
Thanks
Rafi AssadiH.M.
Silicon Automation Systems
Bangalore, INDIA
 
 
 

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<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=952412216-14062000>From:&nbsp; rafi</SPAN></FONT></DIV>
<DIV><FONT face=Arial size=2>Hi,</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial 
size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Is it 
possible for the transferor to inform the</FONT><FONT face=Arial size=2> 
transferee</FONT></DIV>
<DIV><FONT face=Arial size=2>&nbsp;to try more than one transfer 
target</FONT><FONT face=Arial size=2> serially or simultaneously?</FONT></DIV>
<DIV><FONT face=Arial size=2>&nbsp;Probably one way to </FONT><FONT face=Arial 
size=2>do is to include more than one Transfer-to-Header</FONT></DIV>
<DIV><FONT face=Arial size=2>&nbsp;in</FONT><FONT face=Arial size=2> the 
TRANSFER request message</FONT><FONT face=Arial size=2> because transferee take 
decision </FONT></DIV>
<DIV><FONT face=Arial size=2>based on this field&nbsp; (section 4.6 of 
draft&nbsp; draft-sparks-sip-cc-transfer-00.txt).</FONT></DIV>
<DIV><FONT face=Arial size=2>&nbsp;May be order of the of</FONT><FONT face=Arial 
size=2> each name-addr / addr-spec in&nbsp; Transfer-to-Header implies 
</FONT></DIV>
<DIV><FONT face=Arial size=2>their priority.</FONT><FONT face=Arial size=2> 
Having more than one Transfer-to-Header header</FONT></DIV>
<DIV><FONT face=Arial size=2>is not allowed in the draft, 
draft-sparks-sip-cc-transfer-00.txt.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>&nbsp;&nbsp;&nbsp;&nbsp; Please note that contact 
field in the TRANSFER method is optional. So</FONT></DIV>
<DIV><FONT face=Arial size=2>my understanding is that only Transfer-to-Header is 
used to identify the transfer target.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>Thanks</FONT></DIV>
<DIV><FONT face=Arial size=2>Rafi AssadiH.M.</FONT></DIV>
<DIV><FONT face=Arial size=2>Silicon Automation Systems</FONT></DIV>
<DIV><FONT face=Arial size=2>Bangalore, INDIA</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

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hello everybody!
I'm working in a research group wich is starting a development about =
SIP. Probably, we will need somethig help and we would like know whether =
this list is properly to do this or there is another developers' list.
=20
thanks a lot
=20
Mauricio Bola=F1os Arturo
ITEC-TELECOM
Research Division
Bogot=E1 - Colombia

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about SIP. Probably, we will need somethig help and we would like know =
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this list is properly to do this or there is another developers'=20
list.</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>thanks a lot</FONT></DIV>
<DIV><FONT size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2>Mauricio Bola&ntilde;os Arturo</FONT></DIV>
<DIV><FONT size=3D2></FONT><FONT color=3D#000000 =
size=3D2>ITEC-TELECOM</FONT></DIV>
<DIV><FONT color=3D#000000 size=3D2></FONT><FONT size=3D2>Research=20
Division</FONT></DIV>
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Does SIP support Music on hold feature, If so please tell me where do I find
its details
 
thanks
amar 


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<BLOCKQUOTE style="MARGIN-RIGHT: 0px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT color=#0000ff 
  face=Arial size=2><SPAN class=648345117-14062000>Does SIP support Music 
  on&nbsp;hold feature, If so please tell me where do I&nbsp;find its 
  details</SPAN></FONT></DIV>
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT color=#0000ff 
  face=Arial size=2><SPAN class=648345117-14062000></SPAN></FONT>&nbsp;</DIV>
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT color=#0000ff 
  face=Arial size=2><SPAN class=648345117-14062000>thanks</SPAN></FONT></DIV>
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT color=#0000ff 
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From sip-admin@lists.bell-labs.com  Wed Jun 14 15:34:11 2000
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From: "Culpepper, Bert" <bert.culpepper@intervoice-brite.com>
To: Dave Walker <drwalker@ss8networks.com>
Cc: SIP List <sip@lists.bell-labs.com>
Subject: RE: [SIP] Out-of-Band DTMF in SIP
Date: Wed, 14 Jun 2000 14:29:50 -0500
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I understand agreement on MIME-type may be difficult to get, however ISUP
and QSIG did ;-) or maybe I'm wrong about that.  My ID did discuss Megaco
(it actually provides better support than MGCP for DTMF and other terminal
events).

Anyway, I'd like to have a way to ask the MGC to "turn on" digit collection
at the gateway during a call and report the collected digits.  I'd also like
to ask the MGC/gateway to report any/all detected terminal events.  And I'm
looking at PSTN-based terminals (telephones) and  SIP-based application
servers independent of the MGC (or softswitch).  Granted, this is a
"specialized" network architecture but one that is drawing a great deal of
interest by many.  Without the above functionality it is nearly impossible
to deploy many of today's enhanced services in a VoIP network without having
direct control of the gateways.

Regards,
Bert

> -----Original Message-----
> From: Dave Walker [mailto:drwalker@ss8networks.com]
> Sent: Wednesday, June 14, 2000 9:55 AM
> To: Culpepper, Bert
> Cc: SIP List
> Subject: Re: [SIP] Out-of-Band DTMF in SIP
> 
> 
> What would you like to see?  I don't think that agreement on the 
> MIME-type is going to be easy to get.  Your draft suggested MGCP
> events, others might prefer megaco.  A SIP phone may not speak
> either, but if it uses RFC2833 it would be easy to send audio/tone 
> MIME-type inside of INFO.  We already have the Accept headers for
> this kind of thing.  Would you like to see an option-tag defined
> for it?
> 
> Regards,
> 
> Dave Walker
> SS8 Networks
> Ottawa, Canada.
> 
> "Culpepper, Bert" wrote:
> > 
> > Hello all,
> > 
> > There have been a couple IDs written a short while back on DTMF
transport,
> > collection, etc. each using the SIP INFO method.  I see out-of-band DTMF
> > reporting as a "problem" to be solved to support carrying
telecommunications
> > traffic over SIP networks.  IMO, Out-of-band "event" reporting is needed
to
> > minimize latency and provide services to PSTN-based users within SIP
> > proxy/application servers that do not also "own" the gateways.  MGCP and
> > Megaco compliant gateways support DTMF collection and out-of-band
reporting
> > but there is no "agreed upon" method or interworking proposal in
SIP/SIP-T
> > that I have found.
> > 
> > Is there a solution to this "problem" that I have missed or are others
> > interested in addressing this area?  As of this week, I've asked for a
> > "slot" at the upcoming IETF meeting to present one ID on the subject but
> > don't want to take others' time if there's no need.
> > 
> > Thanks,
> > Bert
> >
> 


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Date: Wed, 14 Jun 2000 15:38:16 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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        "'sip@lists.bell-labs.com'" <sip@lists.bell-labs.com>
Subject: Re: [SIP] Receiver-tagged Via header
References: <B16E9BA540A0D211A11D00105A65571F9DAF44@EXCHANGESVR> <394791E4.CD32B405@ericsson.fi> <20000614102850.A24951@div8.net>
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Billy Biggs wrote:
> 
> Hisham Khartabil (hisham.khartabil@lmf.ericsson.se):
> 
> > "Every host that sends or forwards a SIP request adds a Via field indicating the
> > host?s address. However, it is possible that Network Address Translators (NATs)
> > change the source address and port of the request (e.g.,from a net-10 to a
> > globally routable address), in which case the Via header field cannot be relied
> > on to route replies. To prevent this, a proxy SHOULD check the top-most Via
> > header field to ensure that it contains the sender?s correct network address, as
> > seen from that proxy. If the sender?s address is incorrect, the proxy MUST add a
> > ?received? parameter to the Via header field inserted by the previous hop. Such
> > a modified Via header field is known as a receiver-tagged Via header field."
> >
> > So, it is possible that the port is also changed.  Yet the syntax for
> > via-received looks like this:
> >
> > via-received = "received" "=" host
> >
> > shouldn't  it be
> >
> > via-received = "received" "=" hostport
> > or
> > via-received = "received" "=" host [":" port]
> 
>   Wouldn't that be nice.  You could put it in, but it won't help the NAPT
> problem.
> 
>   Unfortunately, responses in SIP always go back to the port listed in
> the sent-by of the Via, NOT the source of the message.  This means you
> are free to open a UDP socket just to send one message and then nuke the
> socket (since the response will come back on your listening socket).
> 
>   This means that you don't care what port you received it on, you just
> care what port is listed in the sent-by of the Via.  If the SIP message
> went through an NAPT, then the sent-by will list the wrong port anyways,
> and there won't be much use in indicating which port the message came
> from.
> 
>   The solution would be to recommend that clients use the same socket
> (source port) that they receive messages in as the socket they send
> messages from, and therefore sending a response to the source port will
> 'usually' work.
> 
>   It might be worth it.....

The only problem is that it does require listening to more sockets
(every sending socket), unless you use a single socket to send and
receive all messages (that socket would have to be on port 5060,
typically).

Having a single socket is difficult since it can't be a connected socket
(which you need to detect ICMP errors).

Thus, effectively, this means that you need to listen to every sending
socket. I don't think that's a big deal (thread-based applications
probably do that anyway), but it is a minor cost.

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


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From: "Karen Miller" <karen@estara.com>
To: "Amar Singh" <ASingh@shoretel.com>, <sip@lists.bell-labs.com>
References: <E41C84FFA720D4118F760050DAD7D73022D0B6@pop-rocks.shoretel.com>
Subject: Re: [SIP] Music On Hold
Date: Wed, 14 Jun 2000 15:45:48 -0400
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In the paper "Comparison of H.323 and SIP for IP Telephony Signaling" by =
Ismail Dalgic and Hanlin Fang, it says on page 6
"the holding side needs to send an INVITE message to the other side, =
indicating a NULL set of
receiving capability for any kind of media.  MOH can be implemented by =
asking an RTSP server to
play to the IP address or phone number provided in the RTSP SETUP =
request.

- Karen
  ----- Original Message -----=20
  From: Amar Singh=20
  To: sip@lists.bell-labs.com=20
  Sent: Wednesday, June 14, 2000 1:53 PM
  Subject: [SIP] Music On Hold


    Does SIP support Music on hold feature, If so please tell me where =
do I find its details
    =20
    thanks
    amar=20

------=_NextPart_000_0071_01BFD617.9C1B2600
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2314.1000" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>In the paper "Comparison of H.323 and =
SIP for IP=20
Telephony Signaling" by Ismail Dalgic and Hanlin Fang, it says on page=20
6</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>"the holding side needs to send an =
INVITE message=20
to the other side, indicating a NULL set of</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>receiving capability for any kind of =
media.&nbsp;=20
MOH can be implemented by asking an RTSP server to</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>play to the IP address or phone number =
provided in=20
the RTSP SETUP request.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>- Karen</FONT></DIV>
<BLOCKQUOTE=20
style=3D"BORDER-LEFT: #000000 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: =
0px; PADDING-LEFT: 5px; PADDING-RIGHT: 0px">
  <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV=20
  style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
  <A href=3D"mailto:ASingh@shoretel.com" =
title=3DASingh@shoretel.com>Amar Singh</A>=20
  </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A=20
  href=3D"mailto:sip@lists.bell-labs.com"=20
  title=3Dsip@lists.bell-labs.com>sip@lists.bell-labs.com</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Wednesday, June 14, 2000 =
1:53=20
  PM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> [SIP] Music On =
Hold</DIV>
  <DIV><BR></DIV>
  <BLOCKQUOTE style=3D"MARGIN-RIGHT: 0px">
    <DIV align=3Dleft class=3DOutlookMessageHeader dir=3Dltr><FONT =
color=3D#0000ff=20
    face=3DArial size=3D2><SPAN class=3D648345117-14062000>Does SIP =
support Music=20
    on&nbsp;hold feature, If so please tell me where do I&nbsp;find its=20
    details</SPAN></FONT></DIV>
    <DIV align=3Dleft class=3DOutlookMessageHeader dir=3Dltr><FONT =
color=3D#0000ff=20
    face=3DArial size=3D2><SPAN =
class=3D648345117-14062000></SPAN></FONT>&nbsp;</DIV>
    <DIV align=3Dleft class=3DOutlookMessageHeader dir=3Dltr><FONT =
color=3D#0000ff=20
    face=3DArial size=3D2><SPAN =
class=3D648345117-14062000>thanks</SPAN></FONT></DIV>
    <DIV align=3Dleft class=3DOutlookMessageHeader dir=3Dltr><FONT =
color=3D#0000ff=20
    face=3DArial size=3D2><SPAN=20
    =
class=3D648345117-14062000>amar&nbsp;</SPAN></FONT></DIV></BLOCKQUOTE></B=
LOCKQUOTE></BODY></HTML>

------=_NextPart_000_0071_01BFD617.9C1B2600--



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From sip-admin@lists.bell-labs.com  Wed Jun 14 15:56:07 2000
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To: Amar Singh <ASingh@shoretel.com>
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Subject: Re: [SIP] Music On Hold
References: <E41C84FFA720D4118F760050DAD7D73022D0B6@pop-rocks.shoretel.com>
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Take a look at draft-ietf-sip-183-00.txt.

---
Igor Slepchin

> Does SIP support Music on hold feature, 
> If so please tell me where do I find its details
>     
>    thanks
>    amar


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From sip-admin@lists.bell-labs.com  Wed Jun 14 15:59:09 2000
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From: "Bratakos, Maroula" <mbratakos@nuera.com>
To: "'Henning Schulzrinne'" <schulzrinne@cs.columbia.edu>,
        Dave Walker <drwalker@ss8networks.com>
Cc: "Culpepper, Bert" <bert.culpepper@intervoice-brite.com>,
        SIP List <sip@lists.bell-labs.com>
Subject: RE: [SIP] Out-of-Band DTMF in SIP
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Even devices that don't speak mgcp or megaco could implement the out-of-band
DTMF using the method described in the culpepper draft because it only uses
a very small subset of mgcp/megaco commands.  I think it is a cleaner method
than carrying RTP in the INFO message because you don't have to worry about
absolute time or the payload type, which for avt-tones is dynamic.  And the
mgcp/megaco commands provide a way to request DTMF via INFO to be turned
on/off, which is a major plus.  I'm not sure how you would turn INFO on/off
with RTP.

Regards,
Maroula Bratakos

-----Original Message-----
From: Henning Schulzrinne [mailto:schulzrinne@cs.columbia.edu]
Sent: Wednesday, June 14, 2000 7:20 AM
To: Dave Walker
Cc: Culpepper, Bert; SIP List
Subject: Re: [SIP] Out-of-Band DTMF in SIP


Dave Walker wrote:
> 
> What would you like to see?  I don't think that agreement on the
> MIME-type is going to be easy to get.  Your draft suggested MGCP
> events, others might prefer megaco.  A SIP phone may not speak
> either, but if it uses RFC2833 it would be easy to send audio/tone
> MIME-type inside of INFO.  We already have the Accept headers for
> this kind of thing.  Would you like to see an option-tag defined
> for it?
> 

While this not be a bad idea (I might be biased, here...), it does need
a bit more work, since carrying RTP packets in INFO would require some
way of mapping RTP timestamps to absolute time [RTCP does this] and,
less importantly, decide what significance the payload type (PT) field
has in this context. Obviously, if you only need a duration, not
absolute timing, sticking RTP packets into INFO is sufficient. However,
you'd probably not want to use every such RTP packet, only when a new
digit has been detected.


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


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From sip-admin@lists.bell-labs.com  Wed Jun 14 16:01:52 2000
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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        "Culpepper, Bert" <bert.culpepper@intervoice-brite.com>,
        SIP List <sip@lists.bell-labs.com>
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"Bratakos, Maroula" wrote:
> 
> Even devices that don't speak mgcp or megaco could implement the out-of-band
> DTMF using the method described in the culpepper draft because it only uses
> a very small subset of mgcp/megaco commands.  I think it is a cleaner method
> than carrying RTP in the INFO message because you don't have to worry about
> absolute time or the payload type, which for avt-tones is dynamic.  And the

Since the PT doesn't matter, you can ignore it. The timestamp is a wash:
with MGCP-lite, you don't get any better timing information than the
arrival of the MGCP packet, which is what RTP also provides by simply
ignoring the timestamp. RTP gives you better relative timing between
tones, based on timestamps.

> mgcp/megaco commands provide a way to request DTMF via INFO to be turned
> on/off, which is a major plus.  I'm not sure how you would turn INFO on/off
> with RTP.

Easy: don't put the MIME type in Accept. It would be more difficult to
change your mind mid-call.

> 
> Regards,
> Maroula Bratakos
> 
>
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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From sip-admin@lists.bell-labs.com  Wed Jun 14 16:30:20 2000
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From: "Culpepper, Bert" <bert.culpepper@intervoice-brite.com>
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
        "Bratakos, Maroula" <mbratakos@nuera.com>
Cc: Dave Walker <drwalker@ss8networks.com>,
        "Culpepper, Bert" <bert.culpepper@intervoice-brite.com>,
        SIP List <sip@lists.bell-labs.com>
Subject: RE: [SIP] Out-of-Band DTMF in SIP
Date: Wed, 14 Jun 2000 15:25:59 -0500
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What I'd like to avoid is "processing" RTP during a session for scenarios
similar to those in draft-rosenberg-sip-3pcc-00.txt.  Although having the
SIP network edge device multicast RTP is a possibility (but is this possible
when a media gateway doesn't speak SIP?).

Thanks,
Bert

-----Original Message-----
From: Henning Schulzrinne [mailto:schulzrinne@cs.columbia.edu]
Sent: Wednesday, June 14, 2000 4:01 PM
To: Bratakos, Maroula
Cc: Dave Walker; Culpepper, Bert; SIP List
Subject: Re: [SIP] Out-of-Band DTMF in SIP


"Bratakos, Maroula" wrote:
> 
> Even devices that don't speak mgcp or megaco could implement the
out-of-band
> DTMF using the method described in the culpepper draft because it only
uses
> a very small subset of mgcp/megaco commands.  I think it is a cleaner
method
> than carrying RTP in the INFO message because you don't have to worry
about
> absolute time or the payload type, which for avt-tones is dynamic. And the

Since the PT doesn't matter, you can ignore it. The timestamp is a wash:
with MGCP-lite, you don't get any better timing information than the
arrival of the MGCP packet, which is what RTP also provides by simply
ignoring the timestamp. RTP gives you better relative timing between
tones, based on timestamps.

> mgcp/megaco commands provide a way to request DTMF via INFO to be turned
> on/off, which is a major plus.  I'm not sure how you would turn INFO
on/off
> with RTP.

Easy: don't put the MIME type in Accept. It would be more difficult to
change your mind mid-call.

> 
> Regards,
> Maroula Bratakos
> 
>
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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This subject has come up periodically over the past 5 years or so. At the
risk of beating dead horses:

1. In the Internet we don't really have "bands" to be "out of" or "in to."
We have hosts, and the requirement to transport information between (or
among) those hosts. You have stated a very common and important
requirement -- the need to transport DTMF information from some hosts on
the net to "SIP proxy/application servers that do not also 'own' the
gateways" -- aka "other hosts". You have stated the requirement that the
transport method minimizes latency. There are other requirements you might
have mentioned, which have to do with keeping the time synchrony of the
DTMF tones aligned to the voice media. (Think of systems where you can
"press 1 or say 'yes'", and what might happen if the dtmf tone path gets
out of time synch with the voice path.) 

2. There is actually an RFC 2833 which describes how to use RTP to
transport DTMF tones in such a way that these requirements are met. 

Please don't think that somehow it is illegal or even strange to use RTP
to transport information to a SIP server. If you need the realtime
information, you need the realtime information. It is not a difficult
payload format to parse, and you might just find that if you do something
else, you will miss some point that when into the godzillion too many
hours that went into getting agreement on that payload format.

I once suggested to someone as a joke that if they really really needed to
use SIP, they should just take the RTP packet and put it into a SIP INFO
message with a body of type application/rtp. Just like the ISUP-within-SIP 
people do. I have since heard that this got implemented. Well, at least
they used the RTP payload format....

Scott

On Tue, 13 Jun 2000, Culpepper, Bert wrote:

> Hello all,
> 
> There have been a couple IDs written a short while back on DTMF transport,
> collection, etc. each using the SIP INFO method.  I see out-of-band DTMF
> reporting as a "problem" to be solved to support carrying telecommunications
> traffic over SIP networks.  IMO, Out-of-band "event" reporting is needed to
> minimize latency and provide services to PSTN-based users within SIP
> proxy/application servers that do not also "own" the gateways.  MGCP and
> Megaco compliant gateways support DTMF collection and out-of-band reporting
> but there is no "agreed upon" method or interworking proposal in SIP/SIP-T
> that I have found.
> 
> Is there a solution to this "problem" that I have missed or are others
> interested in addressing this area?  As of this week, I've asked for a
> "slot" at the upcoming IETF meeting to present one ID on the subject but
> don't want to take others' time if there's no need.
> 
> Thanks,
> Bert
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 




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From sip-admin@lists.bell-labs.com  Thu Jun 15 01:29:06 2000
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Subject: [SIP] Receiver-tagged Via header
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Hi all,

The rcf2543bis states in section 6.45.2

"Every host that sends or forwards a SIP request adds a Via field
indicating the
host’s address. However, it is possible that Network Address Translators
(NATs)
change the source address and port of the request (e.g.,from a net-10 to
a
globally routable address), in which case the Via header field cannot be
relied
on to route replies. To prevent this, a proxy SHOULD check the top-most
Via
header field to ensure that it contains the sender’s correct network
address, as
seen from that proxy. If the sender’s address is incorrect, the proxy
MUST add a
“received” parameter to the Via header field inserted by the previous
hop. Such
a modified Via header field is known as a receiver-tagged Via header
field."

So, it is possible that the port is also changed.  Yet the syntax for
via-received looks like this:

via-received = ”received” ”=” host

shouldn't  it be

via-received = ”received” ”=” hostport
or
via-received = ”received” ”=” host [":" port]

Thanks,
Hisham





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From sip-admin@lists.bell-labs.com  Thu Jun 15 02:14:49 2000
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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: "'Scott Petrack'" <scott.petrack@metatel.com>
Cc: SIP List <sip@lists.bell-labs.com>
Subject: RE: [SIP] Out-of-Band DTMF in SIP
Date: Wed, 14 Jun 2000 23:14:31 -0700
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Scott,

Here we use in-band and out-of-band to indicate whether the DTMF packets are
sent along the media path (with the RTP voice packets) or whether they are
sent along the call processing path. You may quite often want to send DTMF
notification along _both_ paths. A classic example is with pre-paid calling
card systems. When the user makes a call through the system, the RTP media
is connected to the remote SIP gateway. For obvious reasons you also need to
pass avt-tones RTP packets through to the remote gateway; however, if the
user presses and holds the '#' key then they should be taken back to the IVR
system. The IVR application server is not in the media path and so the DTMF
notification needs to be sent along the SIP call processing path for this to
work.

The requirements for each DTMF transport path are different. In most cases
the INFO DTMF events do not need to be matched to the media timestamp (as
they are independent to the media path). Additionally, a low number of INFO
DTMF event packets is desirable and so an event driven event package (such
as the suggestions in the Culpepper draft) seems a good fit.

As other people have said, using RTP avt-tones packets in INFO requests has
no way of enabling/disabling INFO event transmission mid-call (so far).

Regards,

Robert.
-----Original Message-----
From:	Scott Petrack [mailto:scott.petrack@metatel.com]
Sent:	Thursday, June 15, 2000 12:33 PM
To:	Culpepper, Bert
Cc:	SIP List
Subject:	Re: [SIP] Out-of-Band DTMF in SIP


This subject has come up periodically over the past 5 years or so. At the
risk of beating dead horses:

1. In the Internet we don't really have "bands" to be "out of" or "in to."
We have hosts, and the requirement to transport information between (or
among) those hosts. You have stated a very common and important
requirement -- the need to transport DTMF information from some hosts on
the net to "SIP proxy/application servers that do not also 'own' the
gateways" -- aka "other hosts". You have stated the requirement that the
transport method minimizes latency. There are other requirements you might
have mentioned, which have to do with keeping the time synchrony of the
DTMF tones aligned to the voice media. (Think of systems where you can
"press 1 or say 'yes'", and what might happen if the dtmf tone path gets
out of time synch with the voice path.) 

2. There is actually an RFC 2833 which describes how to use RTP to
transport DTMF tones in such a way that these requirements are met. 

Please don't think that somehow it is illegal or even strange to use RTP
to transport information to a SIP server. If you need the realtime
information, you need the realtime information. It is not a difficult
payload format to parse, and you might just find that if you do something
else, you will miss some point that when into the godzillion too many
hours that went into getting agreement on that payload format.

I once suggested to someone as a joke that if they really really needed to
use SIP, they should just take the RTP packet and put it into a SIP INFO
message with a body of type application/rtp. Just like the ISUP-within-SIP 
people do. I have since heard that this got implemented. Well, at least
they used the RTP payload format....

Scott

On Tue, 13 Jun 2000, Culpepper, Bert wrote:

> Hello all,
> 
> There have been a couple IDs written a short while back on DTMF transport,
> collection, etc. each using the SIP INFO method.  I see out-of-band DTMF
> reporting as a "problem" to be solved to support carrying
telecommunications
> traffic over SIP networks.  IMO, Out-of-band "event" reporting is needed
to
> minimize latency and provide services to PSTN-based users within SIP
> proxy/application servers that do not also "own" the gateways.  MGCP and
> Megaco compliant gateways support DTMF collection and out-of-band
reporting
> but there is no "agreed upon" method or interworking proposal in SIP/SIP-T
> that I have found.
> 
> Is there a solution to this "problem" that I have missed or are others
> interested in addressing this area?  As of this week, I've asked for a
> "slot" at the upcoming IETF meeting to present one ID on the subject but
> don't want to take others' time if there's no need.
> 
> Thanks,
> Bert
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 




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From sip-admin@lists.bell-labs.com  Thu Jun 15 03:26:22 2000
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        mmusic <confctrl@ISI.EDU>
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Gonzalo,

the last available draft is draft-ott-mmusic-caps-01.txt which is currently
being heavily revised (note that the syntax section in the end is just giving
examaples).  At the moment we working on completing the modeling aspects
taking into account the various signaling protocols available (particularly
H.323 and SIP, also MEGACO to some degree) to provide a sufficiently rich yet
reasonably simple means for description that can satisfy the needs of all these
protocols.  We also consider local interaction between system components
(e.g. via the Mbus) for the modeling looking into how to model call,
conferences, applications (=media sessions), and individual media streams.
(this has to fit with SIP, SIP call control services, a future conference control
protocol, RTSP, and who knows what).  This model then needs to be linked to
the concrete syntax (a step that is entirely omitted in the old draft).

From a syntax point of view, we are yet undecided.  There is good reasoning to
stick close to the current SDP syntax to allow smooth introduction of new features
into implementations and retain backward compatibility.  And this is what we would
like to achieve given the increasing installed base of SDP.  Nevertheless, a little
more expressiveness for grouping, alternatives and the like would be nice to have.
There are ideas how you can do this without breaking SDP syntax but the semantics
typically changes with such extensions so that a new version numbers allowing for
new features does not seem unreasonable either.

We plan to publish the revised draft as basis for discussion by the end of June
or the beginning of July.

Of course, we welcome any input on all of this.

Joerg


>Hi,
>
>I would like to know the status of the so called, SDPng. Is there any
>draft available about it? Are there folks working on it? Time frame for
>this work, goals...
>
>Thanks,
>
>Gonzalo
>-- 
>Gonzalo Camarillo         Phone :  +358  9 299 33 71
>Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
>Telecom R&D               Fax   :  +358  9 299 30 52
>FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
>Finland
> 

-------------------------------------------------------------------------------
Joerg Ott                                                  jo@tzi.uni-bremen.de
Universitaet Bremen, Germany                              fax + 49 421 218-7000
Center for Computing Technology (TZI)                   voice + 49 421 201-7028
Digital Media and Networks                     http://www.tzi.uni-bremen.de/dmn



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From sip-admin@lists.bell-labs.com  Thu Jun 15 06:52:59 2000
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From: Loita Jahnsson <Loita.Jahnsson@uab.ericsson.se>
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Hi,

   I kind of new in SIP design and have a couple of
   basic questions regarding proxy protocol conversion.

   Assuming that "proxy protocol conversion" means
   something like this:  

   ---transport-1--> proxy ----transport-2-->

   For which cases are protocol conversion applicable
   in a proxy:

   * when the proxy adds more data to the message, so it 
     cannot be stored within a UDP package when proxying 
     the message?

   * when an outbound proxy receives a UDP package from
     a UAC, stating transport=tcp in the Request-URI?

   * when a proxy is configured to always go through
     another proxy using TCP, disregarding any transport   
     parameters in the SIP message?

   * when a proxy has one client which only knowns UDP
     on one side and a client who only knows TCP on the
     other (ie the attempt to contact the called client
     with UDP fails)?

   * ... ?

   Which of those (if applicable) scenarios was tested
   at the 4th bakeoff?

   If "proxy protocol conversions" refers to scenarios
   like the ones mentioned above, what is it called when 
   you need to do this:


   caller-client   ----request, transport1 ----> server
   
   caller-client   <-- response, transport2 ---- same server :)

   (For example if the response for some reason would be
   bigger than max UDP size).

   Regards,

      Loita Jahnsson


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From sip-admin@lists.bell-labs.com  Thu Jun 15 06:55:48 2000
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Date: Thu, 15 Jun 2000 12:53:50 +0200
From: Loita Jahnsson <Loita.Jahnsson@uab.ericsson.se>
Organization: Ericsson Utvecklings AB
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Hi,

   On what basis does the proxy decide whether to fork
   in parallell or use sequential forking?

   Regards,

      Loita Jahnsson


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Based on the internal policies for the network/originating user/dialed number
etc.

Anoop




Loita Jahnsson <Loita.Jahnsson@uab.ericsson.se> on 06/15/2000 05:53:50 AM

Sent by:  Loita Jahnsson <Loita.Jahnsson@uab.ericsson.se>


To:   sip@lists.bell-labs.com
cc:    (Anoop Tripathi/MW/US/3Com)
Subject:  [SIP] sequential/parallell forking



Hi,

   On what basis does the proxy decide whether to fork
   in parallell or use sequential forking?

   Regards,

      Loita Jahnsson


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Date: Thu, 15 Jun 2000 10:25:51 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: Anoop_Tripathi@3com.com
Cc: Loita Jahnsson <Loita.Jahnsson@uab.ericsson.se>, sip@lists.bell-labs.com
Subject: Re: [SIP] sequential/parallell forking
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They can also be based on caller preferences expressed with the
Request-Disposition header, defined in the caller preferences extension.
Honoring this preference is, of course, at the policy discretion of the
proxy server.

-Jonathan R.

Anoop_Tripathi@3com.com wrote:
> 
> Based on the internal policies for the network/originating user/dialed number
> etc.
> 
> Anoop
> 
> Loita Jahnsson <Loita.Jahnsson@uab.ericsson.se> on 06/15/2000 05:53:50 AM
> 
> Sent by:  Loita Jahnsson <Loita.Jahnsson@uab.ericsson.se>
> 
> To:   sip@lists.bell-labs.com
> cc:    (Anoop Tripathi/MW/US/3Com)
> Subject:  [SIP] sequential/parallell forking
> 
> Hi,
> 
>    On what basis does the proxy decide whether to fork
>    in parallell or use sequential forking?
> 
>    Regards,
> 
>       Loita Jahnsson
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 
> _______________________________________________
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> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Subject: [SIP] Question on Tag for To and From headers
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Content-Type: text/plain


The syntax for the To and the From headers is: ( name-addr | addr-spec )
*(";"addr-params) where
addr-params = tag-param | addr-extension

Does this mean that you can have zero or more tag-params followed or
inter-mingled among zero or more
addr-extensions? This is very unclear.


There is a similar situation with the Server and User-Agent headers 1*(
product | comment ).  Is there a
one to one relationship between product and comment, or can there be say one
product followed by
several comments, etc. ?





------_=_NextPart_001_01BFD6E4.7AA4210C
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> Question on Tag for To and From headers</TITLE>
</HEAD>
<BODY>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Arial">The syntax for the To and the From =
headers is: ( name-addr | addr-spec ) *(&quot;;&quot;addr-params) =
where</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">addr-params =3D tag-param | =
addr-extension</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Does this mean that you can have zero =
or more tag-params followed or inter-mingled among zero or more</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">addr-extensions? This is very =
unclear.</FONT>
</P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Arial">There is a similar situation with the =
Server and User-Agent headers 1*( product | comment ).&nbsp; Is there =
a</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">one to one relationship between =
product and comment, or can there be say one product followed by</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">several comments, etc. ?</FONT>
</P>
<BR>
<BR>
<BR>

</BODY>
</HTML>
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From sip-admin@lists.bell-labs.com  Thu Jun 15 12:16:35 2000
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To: sip@lists.bell-labs.com
Date: Thu, 15 Jun 2000 11:16:09 -0500 (CDT)
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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Subject: [SIP] Static Images
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I'm trying to get a feeling for what most people think the most
appropriate way of sending static pictures (generally, at a frame
rate too slow to constitute animation, and definitely not at a
fixed frequency) in the course of a SIP session.

For a frame of reference, some of you may have seen consumer phones
that use some extrodinarily slow and proprietary transfer protocol
to send greyscale images during the course of a phone conversation.
("I've got a picture of Annie right here. She's just started walking.
I'll send it to you" sort of applications).

As I see it, there are a few options for doing this sort of thing:

1) In the signalling path: use a SIP INFO (or similar) message with a
   type image/jpeg body. Since SIP isn't really meant as a transport
   protocol, this doesn't necessarily seem appropriate.

2) As a new, variable frame-rate video codec with packet-loss
   correction (without multiple frames per image, packet loss
   can be a serious issue).

3) By using an existing video codec, but just sending the same frame
   over and over (until a new picture is sent). This is, of course,
   a bit heavyweight for what should be a simple application.

4) As a short-lived TCP "stream" set up with a re-INVITE message;
   this would require an additional transport type to be defined 
   for SDP, but could easily look something like:

   INVITE sip:ejsmith@hotmail.com SIP/2.0
   To: "Mom and Dad" <sip:ejsmith@hotmail.com>;tag=abcde
   From: "John Smith" <sip:john_smith@yahoo.com>;tag=99999
   Contact: <sip:138.85.60.124:5060>
   Call-ID: 31f-39046c6b-119d5b79-1@138.85.60.124
   CSeq: 9986 INVITE
   Accept: application/sdp
   Content-Type: application/sdp
   Content-Length: 119

   v=0
   o=jsmith 95659121152363266 0 IN IP4 138.85.60.124
   s=-
   p=+1 214 555 1212
   c=IN IP4 138.85.60.124
   t=0 0
   m=audio 5000 RTP/AVP 0
   m=image 5098 tcp jpeg
   a=sendonly

5) As a special function of a whiteboard program. I consider this
   punting, as it wouldn't work very well for the embeded type
   phones I mention in the introduction to this mail.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA


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From sip-admin@lists.bell-labs.com  Thu Jun 15 12:27:34 2000
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To: William Lee <williaml@nortelnetworks.com>
Cc: sip@lists.bell-labs.com
Subject: Re: [SIP] Question on Tag for To and From headers
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> William Lee wrote:
> 
> The syntax for the To and the From headers is: ( name-addr | addr-spec
> ) *(";"addr-params) where
> addr-params = tag-param | addr-extension
> 
> Does this mean that you can have zero or more tag-params followed or
> inter-mingled among zero or more
> addr-extensions? This is very unclear.

Yes, that's what the BNF says. As has been discussed here in different
contexts, BNFs are not good at (and not meant for) conveying semantic
restrictions such as "no duplication". That said, since there's only one
tag, it makes sense to label it as 

[ tag-param ] *addr-extension

> 
> There is a similar situation with the Server and User-Agent headers
> 1*( product | comment ).  Is there a
> one to one relationship between product and comment, or can there be
> say one product followed by
> several comments, etc. ?

Yes.

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


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From sip-admin@lists.bell-labs.com  Thu Jun 15 12:28:34 2000
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To: "Adam B. Roach" <Adam.Roach@Ericsson.com>
Cc: sip@lists.bell-labs.com
Subject: Re: [SIP] Static Images
In-reply-to: Your message of "Thu, 15 Jun 2000 11:16:09 CDT." <200006151616.LAA12031@b04a24.exu.ericsson.se>
Date: Thu, 15 Jun 2000 17:27:00 +0100
Message-ID: <6847.961086420@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> "Adam B. Roach" writes:
>I'm trying to get a feeling for what most people think the most
>appropriate way of sending static pictures (generally, at a frame
>rate too slow to constitute animation, and definitely not at a
>fixed frequency) in the course of a SIP session.
...
>As I see it, there are a few options for doing this sort of thing:
...
>4) As a short-lived TCP "stream" set up with a re-INVITE message;
>   this would require an additional transport type to be defined 
>   for SDP, but could easily look something like:
>
>   INVITE sip:ejsmith@hotmail.com SIP/2.0
>   To: "Mom and Dad" <sip:ejsmith@hotmail.com>;tag=abcde
>   From: "John Smith" <sip:john_smith@yahoo.com>;tag=99999
>   Contact: <sip:138.85.60.124:5060>
>   Call-ID: 31f-39046c6b-119d5b79-1@138.85.60.124
>   CSeq: 9986 INVITE
>   Accept: application/sdp
>   Content-Type: application/sdp
>   Content-Length: 119
>
>   v=0
>   o=jsmith 95659121152363266 0 IN IP4 138.85.60.124
>   s=-
>   p=+1 214 555 1212
>   c=IN IP4 138.85.60.124
>   t=0 0
>   m=audio 5000 RTP/AVP 0
>   m=image 5098 tcp jpeg
>   a=sendonly

Is it too disgusting to use...

	INVITE sip:ejsmith@hotmail.com SIP/2.0
	To: "Mom and Dad" <sip:ejsmith@hotmail.com>;tag=abcde
	From: "John Smith" <sip:john_smith@yahoo.com>;tag=99999
	Contact: <sip:138.85.60.124:5060>
	Call-ID: 31f-39046c6b-119d5b79-1@138.85.60.124
	CSeq: 9986 INVITE
	Accept: application/sdp
	Content-Type: application/sdp
	Content-Length: 119

	v=0
	o=jsmith 95659121152363266 0 IN IP4 138.85.60.124
	s=-
	p=+1 214 555 1212
	c=IN IP4 138.85.60.124
	t=0 0
	m=audio 5000 RTP/AVP 0
	m=image 80 http jpeg
	a=sendonly

to indicate fetchming image/jpeg using http? Weird, but fits existing usage
of the m= line...

Colin


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In-Reply-To: <394702DB.3C479937@dynamicsoft.com> from "Jonathan Rosenberg" at Jun 13, 2000 11:58:19 PM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>> If I have missed some crucial piece of text then I think it should be moved
>> to a more logical place in the draft (eg Section 11). Do we need multiple
>> outstanding requests (per UA) anyway ? 
>
>It would be nice to hear if anyone has any concrete examples of a need
>for this.

There are some thoughts that such a need might arise during ISUP
tunneling for some national variants of ISUP. From a practical 
point of view, you also have the issue that PRACKs are also 
being sent and completeing before the INVITE receives a final 
response.  I would strongly prefer that PRACK isn't an exception 
to an overall rule of "one pending request per direction".

The simple way to get around the problem being described is to say 
that you can't send a new request until the previous request has 
received a response. Note the subtle difference: not completed, 
just received a response (including provisionals).

This may cause problems at proxies, who can generate 100s
without the far end necessarily receiving the message. You
*could* have the proxies hold onto the messages until the next
hop responds to the previous messages to ensure in-order delivery,
but this seems overly complicated.  Perhaps, as in most other 
cases, we can just say that 100 responses are an exception?

-- 
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: Billy Biggs <vektor@div8.net>
To: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
Cc: "Adam B. Roach" <Adam.Roach@Ericsson.com>, sip@lists.bell-labs.com
Subject: Re: [SIP] Static Images
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Colin Perkins (C.Perkins@cs.ucl.ac.uk):

> --> "Adam B. Roach" writes:
> >I'm trying to get a feeling for what most people think the most
> >appropriate way of sending static pictures (generally, at a frame
> >rate too slow to constitute animation, and definitely not at a
> >fixed frequency) in the course of a SIP session.
> ...
> >As I see it, there are a few options for doing this sort of thing:
> ...
> >4) As a short-lived TCP "stream" set up with a re-INVITE message;
> >   this would require an additional transport type to be defined 
> >   for SDP, but could easily look something like:
> >
> >   [...]
> >   m=image 5098 tcp jpeg
> >   a=sendonly
> 
> Is it too disgusting to use...
> 
>       [...]
> 	m=image 80 http jpeg
> 	a=sendonly
> 
> to indicate fetchming image/jpeg using http? Weird, but fits existing usage
> of the m= line...

  I agree that it is definitely a session description issue and not a SIP
issue.

  You need to define a protocol for the exchange of a slow stream of
images.  One method is to use http.  Then you need to describe it in the
description protocol.

  It really is a new protocol, since you want to be able to indicate when
the next image is available, if at all, etc...  Definitely do-able.  This
is one of the applications I was working on last summer when I was
investigating alternate description protocols.

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


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Why couldn't an RTP-over-TCP stream be used here? Telling the client to
go fetch it seems strange, particularly since it also requires that you
somehow upload the image to another server or run a web server on your
PC/phone.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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Subject: [SIP] Looking for  SIP client and server implementation
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Hi,
   I am looking for a SIP client and server implementations which I can use
to establish a SIP
call (including media). I came across a few freely available
stacks/libraries but no  integrated 
product (something like NetMeeting for H.323).
Is anyone aware of any such product ? Any help would be appreciated.
 
Regards,
aseem

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Content-Type: text/html;
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso-8859-1">
<TITLE>Question on Tag for To and From headers</TITLE>

<META content="MSHTML 5.00.2314.1000" name=GENERATOR></HEAD>
<BODY>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=420285916-15062000>Hi,</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=420285916-15062000>&nbsp;&nbsp;&nbsp;I am looking for a SIP client and 
server implementations which I can use to establish a SIP</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=420285916-15062000>call 
(including media). I came across a few freely available stacks/libraries but 
no&nbsp; integrated </SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=420285916-15062000>product</SPAN></FONT><FONT color=#0000ff face=Arial 
size=2><SPAN class=420285916-15062000> (something like NetMeeting for 
H.323).</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN class=420285916-15062000>Is 
anyone aware of any such product ?</SPAN></FONT><FONT color=#0000ff face=Arial 
size=2><SPAN class=420285916-15062000> Any help would be 
appreciated.</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=420285916-15062000></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=420285916-15062000>Regards,</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff face=Arial size=2><SPAN 
class=420285916-15062000>aseem</SPAN></FONT></DIV></BODY></HTML>

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Subject: Re: [SIP] Question on Tag for To and From headers
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I'd even argue for taking it one step further. Since tag-param fits into the
syntactic defintion of addr-extension, why not:

( name-addr | addr-spec ) *(";"addr-extension)

and leave it up to the semantic desription of what address parameters are
allowed, what are the semantics of duplicates, and what they actually mean
at the application level (as opposed to what they look like). I generally
think that applying semantic restrictions in formal syntax specifications is
not a good idea, even though it's sometimes possible.

Rick Workman
Nortel Networks


Henning Schulzrinne wrote:

> > William Lee wrote:
> >
> > The syntax for the To and the From headers is: ( name-addr | addr-spec
> > ) *(";"addr-params) where
> > addr-params = tag-param | addr-extension
> >
> > Does this mean that you can have zero or more tag-params followed or
> > inter-mingled among zero or more
> > addr-extensions? This is very unclear.
>
> Yes, that's what the BNF says. As has been discussed here in different
> contexts, BNFs are not good at (and not meant for) conveying semantic
> restrictions such as "no duplication". That said, since there's only one
> tag, it makes sense to label it as
>
> [ tag-param ] *addr-extension
>
> >
> > There is a similar situation with the Server and User-Agent headers
> > 1*( product | comment ).  Is there a
> > one to one relationship between product and comment, or can there be
> > say one product followed by
> > several comments, etc. ?
>
> Yes.
>
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
<tt>I'd even argue for taking it one step further. Since tag-param fits
into the syntactic defintion of addr-extension, why not:</tt><tt></tt>
<p><tt>( name-addr | addr-spec ) *(";"addr-extension)</tt><tt></tt>
<p><tt>and leave it up to the semantic desription of what address parameters
are allowed, what are the semantics of duplicates, and what they actually
mean at the application level (as opposed to what they look like). I generally
think that applying semantic restrictions in formal syntax specifications
is not a good idea, even though it's sometimes possible.</tt><tt></tt>
<p><tt>Rick Workman</tt>
<br><tt>Nortel Networks</tt>
<br><tt></tt>&nbsp;<tt></tt>
<p><tt>Henning Schulzrinne wrote:</tt>
<blockquote TYPE=CITE><tt>> William Lee wrote:</tt>
<br><tt>></tt>
<br><tt>> The syntax for the To and the From headers is: ( name-addr |
addr-spec</tt>
<br><tt>> ) *(";"addr-params) where</tt>
<br><tt>> addr-params = tag-param | addr-extension</tt>
<br><tt>></tt>
<br><tt>> Does this mean that you can have zero or more tag-params followed
or</tt>
<br><tt>> inter-mingled among zero or more</tt>
<br><tt>> addr-extensions? This is very unclear.</tt><tt></tt>
<p><tt>Yes, that's what the BNF says. As has been discussed here in different</tt>
<br><tt>contexts, BNFs are not good at (and not meant for) conveying semantic</tt>
<br><tt>restrictions such as "no duplication". That said, since there's
only one</tt>
<br><tt>tag, it makes sense to label it as</tt><tt></tt>
<p><tt>[ tag-param ] *addr-extension</tt><tt></tt>
<p><tt>></tt>
<br><tt>> There is a similar situation with the Server and User-Agent headers</tt>
<br><tt>> 1*( product | comment ).&nbsp; Is there a</tt>
<br><tt>> one to one relationship between product and comment, or can there
be</tt>
<br><tt>> say one product followed by</tt>
<br><tt>> several comments, etc. ?</tt><tt></tt>
<p><tt>Yes.</tt><tt></tt>
<p><tt>--</tt>
<br><tt>Henning Schulzrinne&nbsp;&nbsp; <a href="http://www.cs.columbia.edu/~hgs">http://www.cs.columbia.edu/~hgs</a></tt><tt></tt>
<p><tt>_______________________________________________</tt>
<br><tt>SIP mailing list</tt>
<br><tt>SIP@lists.bell-labs.com</tt>
<br><tt><a href="http://lists.bell-labs.com/mailman/listinfo/sip">http://lists.bell-labs.com/mailman/listinfo/sip</a></tt></blockquote>
<tt></tt></html>

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i'm using kphone from kdenonbeta.  see http://www.div8.net/dissipate/.
it runs fine on linux.  i don't know nor care about bill gates uas.
send him email and ask to include sip support in the next version of
netmeeting.

-- juha


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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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Cc: "'sip@lists.bell-labs.com'" <sip@lists.bell-labs.com>,
        Srini Beereddy <srini@trillium.com>
Subject: Re: [SIP] Looking for  SIP client and server implementation
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> Aseem Agarwal wrote:
> 
> Hi,
>    I am looking for a SIP client and server implementations which I
> can use to establish a SIP
> call (including media). I came across a few freely available
> stacks/libraries but no  integrated
> product (something like NetMeeting for H.323).
> Is anyone aware of any such product ? Any help would be appreciated.

See http://www.cs.columbia.edu/~hgs/sip for a list.

> 
> Regards,
> aseem

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


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From sip-admin@lists.bell-labs.com  Thu Jun 15 14:46:01 2000
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Date: Thu, 15 Jun 2000 20:45:26 +0200
From: Jiri Kuthan <kuthan@fokus.gmd.de>
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Subject: [SIP] Firewall Control Protocol
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Hi,

I submitted the Firewall Control Protocol (FCP) Internet Draft 
to IETF tonight. This protocol may be used by SIP proxies to open 
pinholes in firewalls so that the SIP sessions get through
them. Feel free to review/comment/etc. Until the draft
appears in the IETF directory, it is available at

http://www.fokus.gmd.de/research/cc/glone/employees/jiri.kuthan/private/fcp/draft-kuthan-fcp-01.txt

The FCP discussion is taking place at the foglamps mailing list:
- foglamps@lists.panix.com
- To subscribe, send email to majordomo@lists.panix.com with “subscribe foglamps” 
  in the body of message

Regards,

Jiri

-- 
Jiri Kuthan               Phone: +49-30-3463 7271
GMD FOKUS                 Fax  : +49-30-3463 8271
Kaiserin-Augusta-Allee 31 E-Mail : kuthan@fokus.gmd.de
D-10589 Berlin, Germany   WWW    : http://www.fokus.gmd.de/usr/kuthan
--
Curious about Internet Telephony? 
Check http://www.fokus.gmd.de/glone/projects/ipt today!


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From sip-admin@lists.bell-labs.com  Thu Jun 15 14:51:22 2000
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: Re: [SIP] Clarification on the proxy behaviour.
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Loop detection is described in 12.3.1 of the 2543bis draft. Suggestions
for improved wording are welcome. I added a reference to this section in
the Via description.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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From sip-admin@lists.bell-labs.com  Thu Jun 15 16:40:19 2000
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Date: Thu, 15 Jun 2000 16:40:30 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Folks,

Some colleagues and I have put together a suite of protocols for IMPP,
based partly on SIP. Its actually seven separate drafts, althouth the
core is just two. They have been submitted to the I-D editor, but you
can grab them right now from:

http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-impp-presence-00.txt

   Core presence protocol. We've written it to be understandable by non
SIP experts,
   and to also indicate what parts of SIP you do and don't need for
presence.
   Don't be scared by the size (70 pages), nearly half are example
message flows.

http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-impp-im-00.txt

   Core IM protocol. Also written to be both a tutorial on SIP and
explain how
   to do IM.

http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-impp-qauth-00.txt

   A mini-protocol for authorizing subscriptions, to support, among
other things,
   the ability for subscriptions to happen when the presentity is
offline.

http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-impp-pidf-00.txt

   An XML PIDF format, optional in our architecture.

http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-impp-lpidf-00.txt

   A really thin pidf, good for wireless devices, since it uses the same
parser
   SIP uses. This is mandatory to support in our architecture.

http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-impp-watcherinfo-00.txt

   A format for watcher info. 

http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-impp-buddylist-00.txt

   Something we never discussed on the list before; a buddylist
document, much 
   akin to bookmarks, which you can store on a server. This way, when
you go
   to a different machine, you can fetch your buddylist, and get your
presence
   services there.


Comments and questions are welcome, of course.



-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Thu Jun 15 18:19:37 2000
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Why stop at XML for IMPP? Why not XML for SIP? Simple Object Access Protocol
may be the right way. Call it SIP Control by Unified Methods.

SOAP SCUM 


Best regards,
Mike Fox


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From sip-admin@lists.bell-labs.com  Thu Jun 15 18:32:40 2000
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Date: Thu, 15 Jun 2000 17:29:15 -0500 (CDT)
Message-Id: <200006152229.RAA00788@b04a45.exu.ericsson.se>
To: vektor@div8.net, schulzrinne@cs.columbia.edu
Subject: Re: [SIP] Static Images
Cc: C.Perkins@cs.ucl.ac.uk, Adam.Roach@Ericsson.com, sip@lists.bell-labs.com
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> Why couldn't an RTP-over-TCP stream be used here? Telling the client to
> go fetch it seems strange, particularly since it also requires that you
> somehow upload the image to another server or run a web server on your
> PC/phone.
> -- 
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
>

That is of course assuming that you don't support HTTP over port 5060 :)
If you can build SIP, why not HTTP as well? (I'm being sarcastic of course.
HTTP 1.1 is not to be underestimated)

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


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From sip-admin@lists.bell-labs.com  Thu Jun 15 19:21:47 2000
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Cc: Billy Biggs <vektor@div8.net>, "Adam B. Roach" <Adam.Roach@Ericsson.com>,
        sip@lists.bell-labs.com
Subject: Re: [SIP] Static Images
In-Reply-To: Message from Henning Schulzrinne <schulzrinne@cs.columbia.edu> of "Thu, 15 Jun 2000 13:04:48 EDT." <39490CB0.52860D1F@cs.columbia.edu>
Date: Thu, 15 Jun 2000 23:38:45 +0100
From: Colin Perkins <c.perkins@cs.ucl.ac.uk>
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--> Henning Schulzrinne writes:
>Why couldn't an RTP-over-TCP stream be used here? Telling the client to
>go fetch it seems strange, particularly since it also requires that you
>somehow upload the image to another server or run a web server on your
>PC/phone.

Yeah, that's probably a better solution for the audio with accompanying
video snapshot scenario...

We probably have to think about how to include downloadable content for
other scenarios, but I guess that's something for the SDP++ folks to worry
about, not a SIP problem.

Colin


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From sip-admin@lists.bell-labs.com  Thu Jun 15 23:53:15 2000
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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: "'Henning Schulzrinne'" <schulzrinne@cs.columbia.edu>,
        Billy Biggs <vektor@div8.net>
Cc: Colin Perkins <C.Perkins@cs.ucl.ac.uk>,
        "Adam B. Roach" <Adam.Roach@Ericsson.com>, sip@lists.bell-labs.com
Subject: RE: [SIP] Static Images
Date: Thu, 15 Jun 2000 20:52:55 -0700
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Do you think that will require a new SDP profile (like RTP/AVP) or does an
adequate "tcp" SDP profile already exist ?

If this is a multiparty SIP conference you've got the problem of reliable
multicast transport and so simple tcp may not cut it (there are other
multicast transport protocols which address this problem). 

Cheers,

Robert.

-----Original Message-----
From:	Henning Schulzrinne [mailto:schulzrinne@cs.columbia.edu]
Sent:	Friday, June 16, 2000 1:05 AM
To:	Billy Biggs
Cc:	Colin Perkins; Adam B. Roach; sip@lists.bell-labs.com
Subject:	Re: [SIP] Static Images

Why couldn't an RTP-over-TCP stream be used here? Telling the client to
go fetch it seems strange, particularly since it also requires that you
somehow upload the image to another server or run a web server on your
PC/phone.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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From sip-admin@lists.bell-labs.com  Fri Jun 16 03:32:37 2000
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Subject: Re: [SIP] Static Images
To: schulzrinne@cs.columbia.edu (Henning Schulzrinne)
Date: Fri, 16 Jun 2000 02:32:26 -0500 (CDT)
Cc: vektor@div8.net (Billy Biggs), C.Perkins@cs.ucl.ac.uk (Colin Perkins),
        sip@lists.bell-labs.com
In-Reply-To: <39490CB0.52860D1F@cs.columbia.edu> from "Henning Schulzrinne" at Jun 15, 2000 01:04:48 PM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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Henning Schulzrinne writes:
>Why couldn't an RTP-over-TCP stream be used here? Telling the client to
>go fetch it seems strange, particularly since it also requires that you
>somehow upload the image to another server or run a web server on your
>PC/phone.

If you're using TCP, I don't quite see what RTP gains you. I agree
with Sean that HTTP may be a bit heavy.

Anyway, I offered 5 scenarios, and everyone seems to have latched onto
one. If I don't hear otherwise, I'll assume that the method of
doing something special in the SDP to indicate a TCP connection of
some sort (possibly with some protocol on top of it, like RTP or HTTP)
is the preferred choice, and take this discussion to MMUSIC.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA


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From sip-admin@lists.bell-labs.com  Fri Jun 16 03:40:31 2000
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From: A Srinath <srinath@sasi.com>
To: Loita Jahnsson <Loita.Jahnsson@uab.ericsson.se>
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Subject: Re: [SIP] sequential/parallell forking
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On Thu, 15 Jun 2000, Loita Jahnsson wrote:

> Hi,
> 
>    On what basis does the proxy decide whether to fork
>    in parallell or use sequential forking?

I think that it is an implementation issue, i.e. each proxy has a fixed
way in which it handles all messages. If it is stateless then it has to be
sequential.  

Srinath A
Silicon Automation Systems
Bangalore, India. 

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> http://lists.bell-labs.com/mailman/listinfo/sip
> 



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From sip-admin@lists.bell-labs.com  Fri Jun 16 04:08:45 2000
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To: A Srinath <srinath@sasi.com>
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A Srinath wrote:
> 
> On Thu, 15 Jun 2000, Loita Jahnsson wrote:
> 
> > Hi,
> >
> >    On what basis does the proxy decide whether to fork
> >    in parallell or use sequential forking?
> 
> I think that it is an implementation issue, i.e. each proxy has a fixed
> way in which it handles all messages. If it is stateless then it has to be
> sequential.

Stateless proxies do not fork (12.3.7). Sequential forking means 
issuing one request and then waiting for a final response before 
issuing the next request. This necessitates the retention of 
state information. As previously mentioned a stateful proxy may 
fork in parallel or sequentially based on internal configuration 
policies. Its decision could be influenced by callee 
preferences, for example through a CPL script, and/or caller 
preferences through the Request-Disposition header.

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


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From sip-admin@lists.bell-labs.com  Fri Jun 16 04:29:44 2000
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Subject: Re: [SIP] Proposal for providing additional error information to UACs
To: oran@cisco.com (Dave Oran)
Date: Fri, 16 Jun 2000 03:29:24 -0500 (CDT)
Cc: eussean@exu.ericsson.se (Sean Olson), sip@lists.bell-labs.com
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From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>The body approach does seem themost flexible, but it also is the
>heaviest-weight. The way I see this used it's likely to be a proxy which
>inserts the information and I don't relish having to write proxy code that
>tears apart and re-constructs a MIME multi-part body to add this
>information.

You don't have to, if people implementing multipart do it correctly.
MIME multipart allows nesting (i.e. one of the parts of a MIME
multipart body can be a MIME multipart body). So all your proxy needs
to do is compose a two-part MIME multipart body; the first part is
taken from the original body (regardless of its type), and the 
second part is your URI.

But I agree that the exact purpose of a URI in the body would need
to be specified as a header of some sort.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA


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From sip-admin@lists.bell-labs.com  Fri Jun 16 04:34:27 2000
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From: Neil Deason <ndeason@ubiquity.net>
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To: Loita Jahnsson <Loita.Jahnsson@uab.ericsson.se>
Cc: sip@lists.bell-labs.com
Subject: Re: [SIP] proxy protocol conversion
References: <3948B56D.E1821A57@uab.ericsson.se>
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Loita Jahnsson wrote:

>    I kind of new in SIP design and have a couple of
>    basic questions regarding proxy protocol conversion.
> 
>    Assuming that "proxy protocol conversion" means
>    something like this:
> 
>    ---transport-1--> proxy ----transport-2-->
> 
>    For which cases are protocol conversion applicable
>    in a proxy:
> 
>    * when the proxy adds more data to the message, so it
>      cannot be stored within a UDP package when proxying
>      the message?
> 
>    * when an outbound proxy receives a UDP package from
>      a UAC, stating transport=tcp in the Request-URI?
> 
>    * when a proxy is configured to always go through
>      another proxy using TCP, disregarding any transport
>      parameters in the SIP message?
> 
>    * when a proxy has one client which only knowns UDP
>      on one side and a client who only knows TCP on the
>      other (ie the attempt to contact the called client
>      with UDP fails)?
> 
>    * ... ?
> 
>    Which of those (if applicable) scenarios was tested
>    at the 4th bakeoff?

Unless I missed it "proxy protocol conversion" is not a term 
defined within the RFC. However the classification system 
for the 4th bakeoff did include the ability for a proxy to do 
UDP to TCP, and vice versa, conversion. AFAIK this simply means
the proxy accepts a SIP message on one of the protocols and 
proxies it on using the other, for whatever reason. I know 
we did this in the advanced scenario testing, where the reason 
was that our proxy was configured to use TCP to contact the 
next downstream proxy.

>    If "proxy protocol conversions" refers to scenarios
>    like the ones mentioned above, what is it called when
>    you need to do this:
> 
>    caller-client   ----request, transport1 ----> server
> 
>    caller-client   <-- response, transport2 ---- same server :)
> 
>    (For example if the response for some reason would be
>    bigger than max UDP size).

Again I don't know of a specific term for this behaviour within 
the RFC.

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


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From sip-admin@lists.bell-labs.com  Fri Jun 16 04:58:02 2000
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To: "Fox, Mike" <mfox@nuera.com>
Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        sip <sip@lists.bell-labs.com>
Subject: Re: [SIP] RE: proposal for IMPP
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"Fox, Mike" wrote:
> 
> Why stop at XML for IMPP? Why not XML for SIP? Simple Object Access Protocol
> may be the right way. Call it SIP Control by Unified Methods.
> 
> SOAP SCUM

All joking aside, SOAP within SIP is already possible as it is
just a 
text/html payload. As SOAP has been designed with HTTP as the
base 
transport using SIP instead, with it's HTTP ancestry, is straight
forward. 
Perhaps the flexibility and extensibility of SOAP could be an
alternative 
to some of the more proprietary/out of SIP's solution space
method/headers extensions.

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


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From sip-admin@lists.bell-labs.com  Fri Jun 16 13:47:40 2000
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From: "Fox, Mike" <mfox@nuera.com>
To: Neil Deason <ndeason@ubiquity.net>
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Subject: [SIP] SIP->XML->SOAP
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I agree. I believe (SIP -> XML) -> SOAP(like)HTTP is unstoppable. OSP and
IMPP require XML awareness in SIP devices, so the door has already been
opened for XML convergence of the SIP domain. HTTP is the Internet
Application Protocol. The problems of HTTP using TCP can be fixed with SCTP
(I hope!). Of course this will take some time (12+ months).

Shouldn't this be the direction of SIP?
Comments?


Mike Fox

> -----Original Message-----
> From: Neil Deason [mailto:ndeason@ubiquity.net]
> Sent: Friday, June 16, 2000 1:55 AM
> To: Fox, Mike
> Cc: Jonathan Rosenberg; sip
> Subject: Re: [SIP] RE: proposal for IMPP
> 
> 
> "Fox, Mike" wrote:
> > 
> > Why stop at XML for IMPP? Why not XML for SIP? Simple 
> Object Access Protocol
> > may be the right way. Call it SIP Control by Unified Methods.
> > 
> > SOAP SCUM
> 
> All joking aside, SOAP within SIP is already possible as it is
> just a 
> text/html payload. As SOAP has been designed with HTTP as the
> base 
> transport using SIP instead, with it's HTTP ancestry, is straight
> forward. 
> Perhaps the flexibility and extensibility of SOAP could be an
> alternative 
> to some of the more proprietary/out of SIP's solution space
> method/headers extensions.
> 
> Cheers,
> Neil.
> -- 
> Ubiquity Software Corporation, UK          
> http://www.ubiquity.net
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 


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From sip-admin@lists.bell-labs.com  Fri Jun 16 14:37:22 2000
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> I agree. I believe (SIP -> XML) -> SOAP(like)HTTP is unstoppable. OSP and
> IMPP require XML awareness in SIP devices, so the door has already been
> opened for XML convergence of the SIP domain. HTTP is the Internet
> Application Protocol. The problems of HTTP using TCP can be fixed with SCTP
> (I hope!). Of course this will take some time (12+ months).
> 
> Shouldn't this be the direction of SIP?
> Comments?
> 
> 
> Mike Fox

If you mean using XML as a common body format, I agree. If you mean SOAP,
I think this is better solved with HTTP. SOAP is an interesting alternative
for service execution in a SIP network. I don't think carrying it via SIP
is necessary or really buys you anything in this case. But it certainly
is a possibility.

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


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From sip-admin@lists.bell-labs.com  Fri Jun 16 14:43:34 2000
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Subject: RE: [SIP] SIP->XML->SOAP
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SIP+XML is where SIP is at now. Additional objects are added and tagged by
MIME headers.

SIP->XML would converge the SIP methods and required parameters to XML and
allows the introduction of a DTD or schema to define the syntax of a valid
SIP message. Transport is UDP to the well known SIP port. Additional
information objects could still use MIME tags and individual syntax, however
many objects would converge to XML and reference their own dtd ( as MLPP
does). The MIME tags are redundant(obsolete) if a xml dtd reference is used.


(SIP->XML)->SOAP would change the transport to HTTP ( I know HTTP may need
some fixing ie. TCP -> SCTP), and change SIP from a protocol to a set of
object interfaces exposed by the SOAP schema. The SIP "methods" would become
object interface methods. Events could be defined through event object
interface methods. Device control can be defined similarly.


Mike

> -----Original Message-----
> From: Sean Olson [mailto:eussean@exu.ericsson.se]
> Sent: Friday, June 16, 2000 11:37 AM
> To: mfox@nuera.com
> Cc: sip@lists.bell-labs.com
> Subject: Re: [SIP] SIP->XML->SOAP
> 
> 
> > I agree. I believe (SIP -> XML) -> SOAP(like)HTTP is 
> unstoppable. OSP and
> > IMPP require XML awareness in SIP devices, so the door has 
> already been
> > opened for XML convergence of the SIP domain. HTTP is the Internet
> > Application Protocol. The problems of HTTP using TCP can be 
> fixed with SCTP
> > (I hope!). Of course this will take some time (12+ months).
> > 
> > Shouldn't this be the direction of SIP?
> > Comments?
> > 
> > 
> > Mike Fox
> 
> If you mean using XML as a common body format, I agree. If 
> you mean SOAP,
> I think this is better solved with HTTP. SOAP is an 
> interesting alternative
> for service execution in a SIP network. I don't think 
> carrying it via SIP
> is necessary or really buys you anything in this case. But it 
> certainly
> is a possibility.
> 
> --
> Sean Olson <sean.olson@ericsson.com>
> 


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From sip-admin@lists.bell-labs.com  Sat Jun 17 00:54:48 2000
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My understanding is that if initial INVITE is from A to B, B will append
a tag to the To header of the final response ( assuming ofcourse that To
header does not initially have a tag).

Now, for the re-INVITE from B to A, A is not supposed to append a tag 
to the To header of the final response to this re-INVITE (although 
the To header of re-INVITE from B to A does may not have tag). 

My question is that if for some reason, a stateful proxy generates the
final response to this re-INVITE from B to A, should it append a tag to
the To header. 

Thanks,
Shail


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From sip-admin@lists.bell-labs.com  Sat Jun 17 07:26:17 2000
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Subject: [SIP] Re: "Static" Images
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I understand that the idea is a "low-data-rate" image transfer
scheme (so Static is possibly the wrong word :).

Agree it's a SDP issue rather than purely a SIP issue.

In those cases where there's only a single group of items, one could 
use the same
approach we used in PINT - having a fmtp line that indicates a URI
and/or an "in-line" item carried in another numbered sub-part.

This is kind of similar to the INFO/INVITE/... approach, and Colin's 
http suggestion,
but allows more than one item and could be a set of URIs rather than 
just in-line
(or a combination).

Remember that PINT has defined extensions that may WELL be useful in a
"normal" IP net conference.

If it's a *really* low refresh rate, then why not Re-Invite using the
same scheme? It doesn't have to be an INFO.

You *may* not even need this - what's wrong with a URI pointing to a
Progressive JPEG/Interlaced GIF/...? I had assumed that every ad-based
service would have some of these wasting the end-users bandwidth.

If, OTOH, it's a low but significant refresh rate, then it is a kind
of animated image//video stream, so RTP-over-whatever seems appropriate.

Final point - who said HTTP == HTTP/1.1?
HTTP 1.0 is a doddle, and this kind of request is not exactly complex.
-- 
All the best, Lawrence
-----------------------------------------------------------------------
| Lawrence Conroy,    | "These Opinions must be mine, 'cos if they    |
| Roke Manor Research |  were my Company's they'd pay me for them"    |
|- lwc@roke.co.uk  ---+- Tel: +44 1794 833666  Fax: +44 1794 833434 --|


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From sip-admin@lists.bell-labs.com  Sat Jun 17 23:48:30 2000
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"Fox, Mike" wrote:
> 
> SIP+XML is where SIP is at now. Additional objects are added and tagged by
> MIME headers.
> 
> SIP->XML would converge the SIP methods and required parameters to XML and
> allows the introduction of a DTD or schema to define the syntax of a valid
> SIP message. 

Why?

SIP is already standardized. A syntax is already defined. There are well
over 50 implementations (more likely around 100 by now). Any future
versions of SIP need to be maximally compatible. So, for better or for
worse, and NOT for debate, SIPs syntax is already done. Any discussions
on migration to XML, binary, ASN.1 or the next flavor du jour are
academic, fruitless, and a waste of space in my inbox, IMHO.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
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From sip-admin@lists.bell-labs.com  Sat Jun 17 23:50:42 2000
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"Adam B. Roach" wrote:
> 
> Henning Schulzrinne writes:
> >Why couldn't an RTP-over-TCP stream be used here? Telling the client to
> >go fetch it seems strange, particularly since it also requires that you
> >somehow upload the image to another server or run a web server on your
> >PC/phone.
> 
> If you're using TCP, I don't quite see what RTP gains you. I agree
> with Sean that HTTP may be a bit heavy.
> 
> Anyway, I offered 5 scenarios, and everyone seems to have latched onto
> one. If I don't hear otherwise, I'll assume that the method of
> doing something special in the SDP to indicate a TCP connection of
> some sort (possibly with some protocol on top of it, like RTP or HTTP)
> is the preferred choice, and take this discussion to MMUSIC.

I think it depends on what the purpose is.

If you want to include an image to be displayed before the call is even
accepted (like a jpeg snapshot for new caller ID), I believe this to be
very much a SIP thing, not SDP. You can think of it as a really long
"subject" header - information for the purposes of determining what to
do with the call. We have discussed these things on the list as well,
and debated including the content in sip vs. indirection via a URL, and
also whether it goes in a body or a header. There was no conclusion to
this debate; it would be nice to finish it.

If you want to do some kind of slow motion video after the call is set
up, I'm unsure why this is anything different from very traditional
video over RTP (UDP even). Why TCP? Sounds like you want a motion JPEG
codec.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
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From sip-admin@lists.bell-labs.com  Sat Jun 17 23:52:38 2000
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Loita Jahnsson wrote:
> 
> Hi,
> 
>    I kind of new in SIP design and have a couple of
>    basic questions regarding proxy protocol conversion.
> 
>    Assuming that "proxy protocol conversion" means
>    something like this:
> 
>    ---transport-1--> proxy ----transport-2-->
> 
>    For which cases are protocol conversion applicable
>    in a proxy:
> 
>    * when the proxy adds more data to the message, so it
>      cannot be stored within a UDP package when proxying
>      the message?

Generally, determination of the transport to use for outgoing requests
will be driven by preferences set in the SRV records. In this case, they
would not be dependent on a change in size of the message. Normal SIP
procesing of a message will normally not make it much bigger. In any
case, the maximum UDP size is 64k, which you are unlikely to ever see in
SIP.

> 
>    * when an outbound proxy receives a UDP package from
>      a UAC, stating transport=tcp in the Request-URI?

Yes, in this case the protocol goes out TCP. This is spelled out (along
with the above) in Section 1.4.2 of the bis draft. This section applies
independnt of the protocol the request came in on, so it may result in
a  "conversion" if the outgoing protocol selected was different from the
incoming.

> 
>    * when a proxy is configured to always go through
>      another proxy using TCP, disregarding any transport
>      parameters in the SIP message?

You can always do that, although its not specified to work this way.
You'll run into troubles if the next hop doesn't support that protocol.

> 
>    * when a proxy has one client which only knowns UDP
>      on one side and a client who only knows TCP on the
>      other (ie the attempt to contact the called client
>      with UDP fails)?

This would be determined through transport parameters in registered
Contacts, so conversion might take place here, yes.



>    If "proxy protocol conversions" refers to scenarios
>    like the ones mentioned above, what is it called when
>    you need to do this:
> 
>    caller-client   ----request, transport1 ----> server
> 
>    caller-client   <-- response, transport2 ---- same server :)
> 
>    (For example if the response for some reason would be
>    bigger than max UDP size).

This is not supported. You could try to do it by putting a different
transport in the Via header than the one you actually used, but I don't
recommend doing that, since its unlikely to work. In any case, it sounds
like  you want the server to dynamically change the tranpsort for the
response if the response comes in and its big; having the client set the
protocol in the Via won't help that. 


-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Sat Jun 17 23:55:18 2000
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> Helpdesk MIS wrote:
> 
> From:  rafi
> Hi,
> 
>            Is it possible for the transferor to inform the transferee
>  to try more than one transfer target serially or simultaneously?
>  Probably one way to do is to include more than one Transfer-to-Header
>  in the TRANSFER request message because transferee take decision
> based on this field  (section 4.6 of draft
> draft-sparks-sip-cc-transfer-00.txt).
>  May be order of the of each name-addr / addr-spec in
> Transfer-to-Header implies
> their priority. Having more than one Transfer-to-Header header
> is not allowed in the draft, draft-sparks-sip-cc-transfer-00.txt.

I'm a bit reluctant to allow that. I'm not sure its needed. If you want
the transferred party to try one address, then the next, then the next,
etc., once each fails, you can do that without having multiple
transfer-tos. Logic in the transferring party can simply send multiple
transfer requests, one after a failure response from the previous.

So, having multiple addresses is only useful for parallel transfers, but
I doubt that this is useful. What happens if multiple acceptances come
back? Does the transferred party speak to all, or just one? Was the
TRANSFER successful if only some of the addresses in the transfer-to
header yield successful results? Lots of complexity for no real value. 

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Sat Jun 17 23:59:50 2000
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Billy Biggs wrote:
> 
> Neil Deason (ndeason@ubiquity.net):
> 
> > Correct, this is noted towards the end of Appendix G - Changes To
> > Be Made as    "recieved-port for NAPT support in Via header field".
> 
>   Cute, I hadn't noticed that.
> 
>   The spec would have to be changed to suggest that clients always send
> from the port that they intend to receive the message on, where possible.
> That breaks stuff pretty bad.

Exactly. This is a significant change from current behavior. I do not
think we can do it, since it is not backwards compatible.

Anyway, getting SIP through a NAT, with media and all working, is much
harder than just this, and requires an ALG anyway. So, I'm not sure
there is much gained in adding this port change. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Sun Jun 18 00:33:38 2000
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Shailendra Bhatnagar wrote:
> 
> My understanding is that if initial INVITE is from A to B, B will append
> a tag to the To header of the final response ( assuming ofcourse that To
> header does not initially have a tag).
> 
> Now, for the re-INVITE from B to A, A is not supposed to append a tag
> to the To header of the final response to this re-INVITE (although
> the To header of re-INVITE from B to A does may not have tag).

I think it is worth discussing whether or not A would/should append a
tag in the To header of this final response.

I'll note that there is no problem if the initial INVITE from A to B had
a tag in the From field.

The difficulty with not appending the tag in the response to re-INVITE
is a loss of idempotency. This will come and bite you when you have
devices that don't maintain call state (i.e., your proxy), which will
insert the tag since they don't know this is a re-INVITE. It will also
complicate recovery from crash/reboot cycles during a call. The
difficulty in appending the tag to the response is, I believe, purely
one of implementation. Most folks probably use some kind of key, hashed
on various message fields, to index call state. This key would basically
change as a result of this re-INVITE. Kind of ugly.

I have been gradually reaching the conclusion that idempotency is a
really important property, and thus I believe worth the implementation
pain.

Comments?

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Sun Jun 18 01:38:43 2000
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To: "Piotr S. Kossowski" <P.Kossowski@elka.pw.edu.pl>
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Work on multiparty conferencing is in progress. There are several ways
to do it with SIP now (dial up conference bridge, 3-way calling with end
system mixing, multicast) which require no extensions; work is in
progress on defining extensions for other models. Note that the 3-way
model described in the call flows draft does extend to n-way, but the
scalability is limited.

For some drafts on the subject:
http://www.softarmor.com/sipwg/drafts/draft-ietf-mmusic-sip-cc-01.txt
(EXPIRED)
http://search.ietf.org/internet-drafts/draft-campbell-sip-cc-framework-00.txt
http://search.ietf.org/internet-drafts/draft-mark-sip-dmcs-00.txt


-Jonathan R.
"Piotr S. Kossowski" wrote:
> 
> Hi,
> 
> About n-way conferences:
> 
> There is an example only 3-way conference
> in draft-ietf-sip-call-flows-00.txt.
> Did conisder anyone more participants (in any articles, drafts).
> 
> Especialy, What's going with RTP streams then ?
> (I think about difference between 2-way and 3-way or more
> streaming)
> 
> BR,
> Piotr
> 
> --
> All messages sent to P.Kossowski are delivered immediately also via SMS.
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Sun Jun 18 02:10:41 2000
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Date: Sun, 18 Jun 2000 07:09:04 -0500
From: Henry Sinnreich <Henry.Sinnreich@WCom.com>
Subject: RE: [SIP] SIP->XML->SOAP
In-reply-to: <394B0817.4999E642@dynamicsoft.com>
To: "Fox, Mike" <mfox@nuera.com>, Sean Olson <eussean@exu.ericsson.se>
Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@lists.bell-labs.com
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>> SIP->XML would converge the SIP methods and required
>parameters to XML and
>> allows the introduction of a DTD or schema to define
>the syntax of a valid
>> SIP message.

Do not forget that endangering the deployment of SIP as is,
would set IP telephony back for a generation and leave the field
for "softswitches" and other such nonsense. Nobody on this
mailing list would then live to see true/open Internet
communications, but just the old circuit switch central control
model with its terrible signaling. Just using IP as a cheap
wire.

My two cents.

Henry



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Henry,

Don't lump Mike in with "those softswitch vendors" just because
he mentioned XML, (as you could if he had mentioned ASN.1).

Also, I believe he was talking about SIP's ``direction'' not the
current situation.

XML approaches have some strong arguments: it would take many
IETF WG's away from defining new syntaxes and allow them to
concentrate on the semantics.  Combined with an XSL, message
parsers and generators could be made to run of the specification
instead of being specially developed.  Bindings to application
level frameworks (such as Java) could be possible in general.

It's just running the thing (necessarily) over HTTP which doesn't
make sense to me.

--Brian

Henry Sinnreich wrote:                 (07:09:04 -0500 Sun, 18 Jun 2000)
>
> >> SIP->XML would converge the SIP methods and required
> >> parameters to XML and allows the introduction of a DTD or
> >> schema to define the syntax of a valid SIP message.
> 
> Do not forget that endangering the deployment of SIP as is,
> would set IP telephony back for a generation and leave the
> field for "softswitches" and other such nonsense. Nobody on
> this mailing list would then live to see true/open Internet
> communications, but just the old circuit switch central control
> model with its terrible signaling. Just using IP as a cheap
> wire.
> 
> My two cents.
> 
> Henry

-- 
Brian F. G. Bidulock   ¦ The reasonable man adapts himself to the ¦
bidulock@dallas.net    ¦ world; the unreasonable one persists in  ¦
                       ¦ trying  to adapt the  world  to himself. ¦
                       ¦ Therefore  all  progress  depends on the ¦
M.O.A.M.O.N.N.T.O.M.E. ¦ unreasonable man. -- George Bernard Shaw ¦


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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: "Adam B. Roach" <Adam.Roach@Ericsson.com>,
        Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
        Billy Biggs <vektor@div8.net>, Colin Perkins <C.Perkins@cs.ucl.ac.uk>,
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Subject: Re: [SIP] Static Images
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> 
> If you want to include an image to be displayed before the call is even
> accepted (like a jpeg snapshot for new caller ID), I believe this to be
> very much a SIP thing, not SDP. You can think of it as a really long
> "subject" header - information for the purposes of determining what to
> do with the call. We have discussed these things on the list as well,
> and debated including the content in sip vs. indirection via a URL, and
> also whether it goes in a body or a header. There was no conclusion to
> this debate; it would be nice to finish it.

There is one additional argument (not sure whether this was made last
time around...) for a URI header: body parts are very difficult to add
by proxies, but it might be nice to allow a proxy to go into (say) an
LDAP database or private photo collection and add this information. 

> 
> If you want to do some kind of slow motion video after the call is set
> up, I'm unsure why this is anything different from very traditional
> video over RTP (UDP even). Why TCP? Sounds like you want a motion JPEG
> codec.
> 
The only reason to use TCP here is that since delay isn't that important
in this image transfer application, one might as well get the benefit of
reliability. The only reason to use RTP rather than just "raw bits" is
that you might want the encapsulation (e.g., for JPEG or H.261 frames)
and might want to re-use existing fast-motion code [I don't see these as
particularly compelling arguments]. As a side remark, one should
probably weigh the advantage of having a separate mechanism vs. only
using RTP/UDP in very low frame-rate mode. The latter gives you this
frame-on-demand or one-frame-a-minute pretty much for free, without
having to wait to upgrade everyone.




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From sip-admin@lists.bell-labs.com  Sun Jun 18 23:14:48 2000
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: Re: [SIP] Static Images
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Henning Schulzrinne wrote:
> 
> >
> > If you want to include an image to be displayed before the call is even
> > accepted (like a jpeg snapshot for new caller ID), I believe this to be
> > very much a SIP thing, not SDP. You can think of it as a really long
> > "subject" header - information for the purposes of determining what to
> > do with the call. We have discussed these things on the list as well,
> > and debated including the content in sip vs. indirection via a URL, and
> > also whether it goes in a body or a header. There was no conclusion to
> > this debate; it would be nice to finish it.
> 
> There is one additional argument (not sure whether this was made last
> time around...) for a URI header: body parts are very difficult to add
> by proxies, but it might be nice to allow a proxy to go into (say) an
> LDAP database or private photo collection and add this information.

This point was made last time around. We had several candidate ways to
include data by reference:

1. a body type of application/uri-list
2. a new header, URI, as you suggest

for content included directly, there is clearly only one way, in the
body with multipart MIME.

I initially supported (1) for indirect references to content. This was
because it would be nice to have the same basic encapsulation
independent of whether the body is indirectly references or actually
included. However, you are correct that practically speaking, this makes
it very hard for manipulation by proxies and application servers. As you
point out, adding something like a URL to a prestored picture of the
caller, after doing proxy-authentication to verify the identity of the
caller, would be a nice feature, and much harder if the proxy has to
much with bodies. Related to that, APIs like CGI and servlets don't play
that well with multipart, which then becomes mandatory for such
services.

So, I am basically saying that I have pretty much (although not 100%
sure yet) changed my mind on this, and would prefer a header like URI:
or something, as Henning proposes. Other comments?




> > If you want to do some kind of slow motion video after the call is set
> > up, I'm unsure why this is anything different from very traditional
> > video over RTP (UDP even). Why TCP? Sounds like you want a motion JPEG
> > codec.
> >
> The only reason to use TCP here is that since delay isn't that important
> in this image transfer application, one might as well get the benefit of
> reliability. The only reason to use RTP rather than just "raw bits" is
> that you might want the encapsulation (e.g., for JPEG or H.261 frames)
> and might want to re-use existing fast-motion code [I don't see these as
> particularly compelling arguments]. As a side remark, one should
> probably weigh the advantage of having a separate mechanism vs. only
> using RTP/UDP in very low frame-rate mode. The latter gives you this
> frame-on-demand or one-frame-a-minute pretty much for free, without
> having to wait to upgrade everyone.

I really don't see the need for having a completely separate way to
support streaming media that is high latency, as opposed to low. If you
want TCP, how about:

m=video 21375 RTP/AVPT 31

where AVPT refers to a TCP encapsulating profile of RTP. 

You can then use a pitifully slow motion JPEG, and reuse the RTP
encapsulations, code, codecs, etc.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Sun Jun 18 23:22:39 2000
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From: Shail Bhatnagar <shbhatna@cisco.com>
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Subject: Re: [SIP] Tags again
To: jdrosen@dynamicsoft.com (Jonathan Rosenberg)
Date: Sun, 18 Jun 2000 23:22:14 -0400 (EDT)
Cc: sip@lists.bell-labs.com
In-Reply-To: <394C514B.983E1BA9@dynamicsoft.com> from "Jonathan Rosenberg" at Jun 18, 2000 12:34:19 AM
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Jonathan, Precisely because of the stateful proxy case, I asked the 
question. However, I think on an earlier thread it was suggested that
re-INVITE should not cause any tag to be appended ( I think even
the bis version says this). 
As far as the hash key is concerned, in my humble opinion, lookup
logic should not include tags. Tag comparison should be the state
machine code.

Thanks,
Shail

> 
> 
> 
> Shailendra Bhatnagar wrote:
> > 
> > My understanding is that if initial INVITE is from A to B, B will append
> > a tag to the To header of the final response ( assuming ofcourse that To
> > header does not initially have a tag).
> > 
> > Now, for the re-INVITE from B to A, A is not supposed to append a tag
> > to the To header of the final response to this re-INVITE (although
> > the To header of re-INVITE from B to A does may not have tag).
> 
> I think it is worth discussing whether or not A would/should append a
> tag in the To header of this final response.
> 
> I'll note that there is no problem if the initial INVITE from A to B had
> a tag in the From field.
> 
> The difficulty with not appending the tag in the response to re-INVITE
> is a loss of idempotency. This will come and bite you when you have
> devices that don't maintain call state (i.e., your proxy), which will
> insert the tag since they don't know this is a re-INVITE. It will also
> complicate recovery from crash/reboot cycles during a call. The
> difficulty in appending the tag to the response is, I believe, purely
> one of implementation. Most folks probably use some kind of key, hashed
> on various message fields, to index call state. This key would basically
> change as a result of this re-INVITE. Kind of ugly.
> 
> I have been gradually reaching the conclusion that idempotency is a
> really important property, and thus I believe worth the implementation
> pain.
> 
> Comments?
> 
> -Jonathan R.
> -- 
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 
> 
> _______________________________________________
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> SIP@lists.bell-labs.com
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> 
> 



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From sip-admin@lists.bell-labs.com  Sun Jun 18 23:43:58 2000
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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        "Adam B. Roach" <Adam.Roach@Ericsson.com>
Cc: Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
        Billy Biggs <vektor@div8.net>, Colin Perkins <C.Perkins@cs.ucl.ac.uk>,
        sip@lists.bell-labs.com
Subject: RE: [SIP] Static Images
Date: Sun, 18 Jun 2000 20:43:40 -0700
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This discussion is not only limited to images but also the transferring of
executable objects, scripts and just about any other non-streaming entity in
a SIP session. I can foresee that some objects (perhaps call preference
scripts or display objects) may be desired during call setup (as Jonathan
mentioned) and others form part of the session content. 

For transferring content during call setup:
Inline content has undesirable size limitations and deleterious effects on
call-setup time. For referenced content: using HTTP to _fetch_ the content
during call setup seems the way to go as no two-way session information
needs to be exchanged (which defeats the purpose of SIP).
The question is should the URL be in a MIME body or SIP header ? MIME seems
the most flexible but also required multipart support in all proxies. Do we
need both ?

The same problem exists for file-based session content: If MIME multipart
support is not mandatory on all proxies then we need to look at SDP.
Transferring objects in SDP is somewhat limited unless we come up with a=
extensions as someone suggested (from PINT). Otherwise, let's consider the
case where each object transferred is separate and so requires a different
SDP m= line (although not all at once). Now you cannot reuse PT values
within the same SIP call (correct?) and so you are limited to 120 or so
object transfers (if you use the static RTP payload range as well). Of
course what we really need is SDPng but until then I think its is worthwhile
in stretching the current SDP a= usage to perform object based transfer -
this is a potentially huge application area of a SIP session. 

.... Just my thoughts.

Robert.


-----Original Message-----
From:	Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent:	Saturday, June 17, 2000 2:02 PM
To:	Adam B. Roach
Cc:	Henning Schulzrinne; Billy Biggs; Colin Perkins;
sip@lists.bell-labs.com
Subject:	Re: [SIP] Static Images



"Adam B. Roach" wrote:
> 
> Henning Schulzrinne writes:
> >Why couldn't an RTP-over-TCP stream be used here? Telling the client to
> >go fetch it seems strange, particularly since it also requires that you
> >somehow upload the image to another server or run a web server on your
> >PC/phone.
> 
> If you're using TCP, I don't quite see what RTP gains you. I agree
> with Sean that HTTP may be a bit heavy.
> 
> Anyway, I offered 5 scenarios, and everyone seems to have latched onto
> one. If I don't hear otherwise, I'll assume that the method of
> doing something special in the SDP to indicate a TCP connection of
> some sort (possibly with some protocol on top of it, like RTP or HTTP)
> is the preferred choice, and take this discussion to MMUSIC.

I think it depends on what the purpose is.

If you want to include an image to be displayed before the call is even
accepted (like a jpeg snapshot for new caller ID), I believe this to be
very much a SIP thing, not SDP. You can think of it as a really long
"subject" header - information for the purposes of determining what to
do with the call. We have discussed these things on the list as well,
and debated including the content in sip vs. indirection via a URL, and
also whether it goes in a body or a header. There was no conclusion to
this debate; it would be nice to finish it.

If you want to do some kind of slow motion video after the call is set
up, I'm unsure why this is anything different from very traditional
video over RTP (UDP even). Why TCP? Sounds like you want a motion JPEG
codec.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com




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From sip-admin@lists.bell-labs.com  Mon Jun 19 00:13:49 2000
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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: sip@lists.bell-labs.com
Subject: RE: [SIP] Request/Responses from other addresses
Date: Sun, 18 Jun 2000 21:13:43 -0700
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Hi Jonathan,

It would be nice though if the source authentication could be provided at
the IP layer through IPSEC AH or similar methods, such that the SIP UA could
trust the source address if an SA exists without having to implement
authentication itself (assuming hop-by-hop authentication). Is this a
pipe-dream ? Can anyone see any obvious holes ? There are many issues to
consider but one of them is: who is allowed to send BYE & re-INVITE requests
for a call. I guess one answer is "anyone with an SA" but I was hoping thing
we could narrow it down a bit.

Robert.

-----Original Message-----
From:	Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent:	Tuesday, June 06, 2000 12:21 PM
To:	Fairlie-Cuninghame, Robert
Cc:	McMurry, Kathleen; Sean Olson; sip@lists.bell-labs.com
Subject:	Re: [SIP] Request/Responses from other addresses



"Fairlie-Cuninghame, Robert" wrote:
> 
> > > For security purposes, actually, you really do need to check that the
> > > CANCEL came from the same place as the request being cancelled (see
the
> > > minutes of the security task force on this). CANCELs cannot usefully
be
> > > authenticated directly, since they may come from proxies. That aside,
> > > CANCEL can come from anywhere.
> > >
> I take it then that this doesn't apply to BYE requests because they can be
> authenticated directly?

Yes.

> Is there a weaker source address check you could
> apply in the absence of authentication? I can't think of any infallible
> ones.

THere is no such thing as infallible security anyway.

> Requests will not always come from the original source address and the
> Contact address doesn't necessarily infer that the source of future
requests
> will be from that address... although it would help if we could assume
that!

In the case of mobility, the address definitely won't be the same.

> 
> So I guess that without authentication, SIP is very vulnerable to someone
> sniffing packets and then generating their own requests to re-INVITE the
> call to themselves.

Yes; but thats why we have it. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Date: Mon, 19 Jun 2000 00:16:07 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [SIP] Reliable provisional responses
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Folks,

WG last call has pretty much ended on the reliable provisional responses
draft. The only open issue seems to be whether prack should be
cumulative or not. There was one note arguing against it, and I weakly
argued in favor of it. To follow a recent trend I seem to be having
regarding about-faces, I think I have changed my mind on this one too.

Why? Cumulative prack is purely an optimization; you can always send N
separate PRACKs instead of a single cumulative PRACK. The reduction in
messaging is the only benefit; it is still possible to support multiple
outstanding reliable provisional responses without cumulative PRACK. The
primary drawback is complexity. One of the ones which concerns me is
determining the set of outstanding reliable provisionals covered by
prack. If people place the reliable provisional state machine objects
into a hash table, indexed by, among other things, the rseq number, this
makes it hard to match up transactions with PRACK. Finding the a set of
things matching a rule, where those things are stored in a hash table,
is a pain. 

So, given that (1) we have no specific application which really is a
killer app for cumulative PRACK, (2) its a message reduction
optimization for something which is probably not a common case, and (3)
there are complexity issues that make an implementation difficult, I am
proposing to remove this feature from the spec before submitting the
version that will go to IESG.

So, if you want to speak in favor of keeping this feature, speak NOW
(well, within 3 days, say), or forever hold your peace....

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Mon Jun 19 04:50:34 2000
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Date: Mon, 19 Jun 2000 09:45:04 +0100
From: Neil Deason <ndeason@ubiquity.net>
Organization: Ubiquity Software Corporation Limited
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To: Sean Olson <eussean@exu.ericsson.se>
Cc: mfox@nuera.com, sip@lists.bell-labs.com
Subject: Re: [SIP] SIP->XML->SOAP
References: <200006161836.NAA02855@b04a45.exu.ericsson.se>
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Sean Olson wrote:
> 
> > I agree. I believe (SIP -> XML) -> SOAP(like)HTTP is unstoppable. OSP and
> > IMPP require XML awareness in SIP devices, so the door has already been
> > opened for XML convergence of the SIP domain. HTTP is the Internet
> > Application Protocol. The problems of HTTP using TCP can be fixed with SCTP
> > (I hope!). Of course this will take some time (12+ months).
> >
> > Shouldn't this be the direction of SIP?
> > Comments?

I fully agree with Jonathan about us being well past the time to
define the basic syntax of SIP and Henry's point of what we would
stand to lose if we did. However, no large scale disturbance of
the protocol is necessary just to carry an XML payload encoded by
the SOAP schema within a SIP message.
 
> If you mean using XML as a common body format, I agree. If you mean SOAP,
> I think this is better solved with HTTP. SOAP is an interesting alternative
> for service execution in a SIP network. I don't think carrying it via SIP
> is necessary or really buys you anything in this case. But it certainly
> is a possibility.

Within a SIP network how about something like this, say between a
SIP Proxy (shown as the client) and an Application Services
Broker (the server), for a trivial credit check service. Probably
not a brilliant example but it shows the sort of thing SOAP could
be used for, i.e. alleviating the need for lots of header/method
extensions to SIP by providing a well defined way for such info
to be encoded in the SOAP/XML payload. I have taken the liberty
of introducing a new SIP method, SERVICE, to carry the SOAP/XML
payload for simplicity, though even this might not be strictly
necessary.

C->S:

SERVICE sip:application_service_broker.ubiquity.net SIP/2.0
To: sip:application_service_broker.ubiquity.net 
From: sip:proxy.ubiquity.net
Call-ID:648324@193.195.52.30
CSeq: 1 SERVICE
Via: SIP/2.0/UDP proxy.ubiquity.net
Content-Type: text/xml
Content-Length: nnnn

<SOAP-ENV:Envelope 
  xmlns:SOAP-ENV="http://schemas.xmlsoap-org/soap/envelope" 
  SOAPENV:encodingStyle=
  "http://schemas.xmlsoap.org/soap/encoding/">
    <SOAP-ENV:Body>
        <m:GetCreditStatus xmlns:m="Some-Namespace-URI">
            <user>sip:jo@ubiquity.net</user>
        </m:GetCreditStatus>
    </SOAP-ENV:Body>
</SOAP-ENV:Envelope>

S->C:

SIP/2.0 200 OK
To: sip:application_service_broker.ubiquity.net 
From: sip:proxy.ubiquity.net
Call-ID:648324@193.195.52.30
CSeq: 1 SERVICE
Via: SIP/2.0/UDP proxy.ubiquity.net
Content-Type: text/xml
Content-Length: nnnn

<SOAP-ENV:Envelope 
  xmlns:SOAP-ENV="http://schemas.xmlsoap-org/soap/envelope" 
  SOAP-ENV:encodingStyle=
  "http://schemas.xmlsoap.org/soap/encoding/">
    <SOAP-ENV:Body>
        <m:GetCreditStatus xmlns:m="Some-Namespace-URI">
            <return>34.5</return>
        </m:GetCreditStatus>
    </SOAP-ENV:Body>
</SOAP-ENV:Envelope>

I could extend this into a short Internet Draft if there is any
interest in doing so.
Cheers,
Neil.
-- 
Ubiquity Software Corporation, UK          
http://www.ubiquity.net


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        Henning Schulzrinne <hgs@cs.columbia.edu>
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What is the URL for the I-D on SIP and SCTP?

Sorry for the trouble, Henry


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Date: Mon, 19 Jun 2000 14:50:48 -0500
From: Henry Sinnreich <Henry.Sinnreich@WCom.com>
Subject: RE: [SIP] SIP->XML->SOAP
In-reply-to: <394DDD90.A82076DE@ubiquity.net>
To: Neil Deason <ndeason@ubiquity.net>, Sean Olson <eussean@exu.ericsson.se>
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>I could extend this into a short Internet Draft

Would suggest to have such a draft and open it for discussion.

Thanks, Henry

>-----Original Message-----
>From: sip-admin@lists.bell-labs.com
>[mailto:sip-admin@lists.bell-labs.com]On Behalf Of Neil Deason
>Sent: Monday, June 19, 2000 3:45 AM
>To: Sean Olson
>Cc: mfox@nuera.com; sip@lists.bell-labs.com
>Subject: Re: [SIP] SIP->XML->SOAP
>
>
>Sean Olson wrote:
>>
>> > I agree. I believe (SIP -> XML) -> SOAP(like)HTTP
>is unstoppable. OSP and
>> > IMPP require XML awareness in SIP devices, so the
>door has already been
>> > opened for XML convergence of the SIP domain. HTTP
>is the Internet
>> > Application Protocol. The problems of HTTP using
>TCP can be fixed with SCTP
>> > (I hope!). Of course this will take some time (12+ months).
>> >
>> > Shouldn't this be the direction of SIP?
>> > Comments?
>
>I fully agree with Jonathan about us being well past
>the time to
>define the basic syntax of SIP and Henry's point of
>what we would
>stand to lose if we did. However, no large scale disturbance of
>the protocol is necessary just to carry an XML payload
>encoded by
>the SOAP schema within a SIP message.
>
>> If you mean using XML as a common body format, I
>agree. If you mean SOAP,
>> I think this is better solved with HTTP. SOAP is an
>interesting alternative
>> for service execution in a SIP network. I don't
>think carrying it via SIP
>> is necessary or really buys you anything in this
>case. But it certainly
>> is a possibility.
>
>Within a SIP network how about something like this,
>say between a
>SIP Proxy (shown as the client) and an Application Services
>Broker (the server), for a trivial credit check
>service. Probably
>not a brilliant example but it shows the sort of thing
>SOAP could
>be used for, i.e. alleviating the need for lots of
>header/method
>extensions to SIP by providing a well defined way for such info
>to be encoded in the SOAP/XML payload. I have taken the liberty
>of introducing a new SIP method, SERVICE, to carry the SOAP/XML
>payload for simplicity, though even this might not be strictly
>necessary.
>
>C->S:
>
>SERVICE sip:application_service_broker.ubiquity.net SIP/2.0
>To: sip:application_service_broker.ubiquity.net
>From: sip:proxy.ubiquity.net
>Call-ID:648324@193.195.52.30
>CSeq: 1 SERVICE
>Via: SIP/2.0/UDP proxy.ubiquity.net
>Content-Type: text/xml
>Content-Length: nnnn
>
><SOAP-ENV:Envelope
>  xmlns:SOAP-ENV="http://schemas.xmlsoap-org/soap/envelope"
>  SOAPENV:encodingStyle=
>  "http://schemas.xmlsoap.org/soap/encoding/">
>    <SOAP-ENV:Body>
>        <m:GetCreditStatus xmlns:m="Some-Namespace-URI">
>            <user>sip:jo@ubiquity.net</user>
>        </m:GetCreditStatus>
>    </SOAP-ENV:Body>
></SOAP-ENV:Envelope>
>
>S->C:
>
>SIP/2.0 200 OK
>To: sip:application_service_broker.ubiquity.net
>From: sip:proxy.ubiquity.net
>Call-ID:648324@193.195.52.30
>CSeq: 1 SERVICE
>Via: SIP/2.0/UDP proxy.ubiquity.net
>Content-Type: text/xml
>Content-Length: nnnn
>
><SOAP-ENV:Envelope
>  xmlns:SOAP-ENV="http://schemas.xmlsoap-org/soap/envelope"
>  SOAP-ENV:encodingStyle=
>  "http://schemas.xmlsoap.org/soap/encoding/">
>    <SOAP-ENV:Body>
>        <m:GetCreditStatus xmlns:m="Some-Namespace-URI">
>            <return>34.5</return>
>        </m:GetCreditStatus>
>    </SOAP-ENV:Body>
></SOAP-ENV:Envelope>
>
>I could extend this into a short Internet Draft if there is any
>interest in doing so.
>Cheers,
>Neil.
>--
>Ubiquity Software Corporation, UK
>http://www.ubiquity.net
>
>
>_______________________________________________
>SIP mailing list
>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip



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Subject: Re: [SIP] Tags again
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> I have been gradually reaching the conclusion that idempotency is a
> really important property, and thus I believe worth the implementation
> pain.
> 
> Comments?
> 
> -Jonathan R.
> -- 
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 

I second that. Plus I think it's more of pain not to add a tag
(a special case) in this event.

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


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From sip-admin@lists.bell-labs.com  Mon Jun 19 11:40:49 2000
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Cc: Henning Schulzrinne <hgs@cs.columbia.edu>, sip@lists.bell-labs.com
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Subject: [SIP] Re: SIP and SCTP
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I have written a preliminary draft on the subject, which I hope to send
in shortly (next few days or so).

-Jonathan R.

Henry Sinnreich wrote:
> 
> What is the URL for the I-D on SIP and SCTP?
> 
> Sorry for the trouble, Henry

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Mon Jun 19 12:34:12 2000
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From: "Fox, Mike" <mfox@nuera.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>, sip@lists.bell-labs.com
Subject: RE: [SIP] SIP->XML->SOAP
Date: Mon, 19 Jun 2000 09:33:54 -0700
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As SOAP is defined. It runs over HTTP, sure you could run it over SIP, but
why? Mapping SIP to XML alone is not compelling, but I think it is good for
SIP. I believe, that the Software art is beyond lumping a protocol and its
application functions together, as many CORBA, Java, C++, and OO programmers
will understand. SOAP is a candidate infrastructure for a object oriented
solution to session and device control.

I respect that the chair does not like the idea. The work can be taken
elsewhere.

Best regards,
Mike 



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From sip-admin@lists.bell-labs.com  Mon Jun 19 16:56:18 2000
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Subject: Re: [SIP] comments on draft-sparks-sip-cc-transfer-00.txt
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"Fox, Mike" wrote:

> I read the draft, and I am left wondering exactly what aspects (or features)
> of call control the Transfer method adds which can not be accomplished using
> existing SIP methods. The draft cites that:
> Looking back over the SIP email there are additional points the author and
> others make in favor of the Transfer method, but I am (as I'm sure others
> are) unsure if these alone justify implementation of a new method. I believe
> the draft, discussions, and group thinking would all benefit from a thorough
> comparison of call control models and features made possible using the
> Transfer method vs. existing SIP methods (such as BYE Also).

I was hoping that somebody else would point out that there IS no existing SIP
BYE Also method. That came from a draft which has expired and I believe is
considered abandoned by the authors . . .

--
Dean Willis




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From sip-admin@lists.bell-labs.com  Mon Jun 19 17:04:24 2000
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        Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
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Subject: Re: [SIP] Static Images
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The problem with having proxies add body parts is (a) complexity, (b)
that now all clients have to understand multipart, even if they have no
interest in the information (I would guess that most current SIP clients
don't understand multipart) and (c) that it doesn't work with signed
requests. Headers containing a reference (http:, say) avoid these
problems. The only drawback is that such references require http support
in clients (which may be an issue for a very light-weight clients, but
most of these are probably graphically challenged anyway) and requires
that the information is available on a publically accessible web page. I
suspect that most corporate users would have difficult uploading their
likeness or other images to a web page, and not everyone wants to have
their likeness framed by an advertisement from geocity or kin. These
are, more or less, the arguments that were made the last time this topic
came around, I believe.




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Subject: [SIP] Questions on Figure 11 of RFC2543bis
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In this diagram does "give up" mean the caller gave up and hung=20
up the call or that the state machine gave up ?

Since there is another arrow from the "Calling" state showing=20
"7 INVITE sent", I assume the arrow to the right of "Calling" state=20
happens when the user hangs up. In this case shouldn't the state machine
wait for the response to INVITE and send an ACK ?

The arrow to the right of "Call Proceeding" state could happen if either
the caller hangs up or the state machine hangs up. Is the timeout for
when=20
the state machine will give up in this state specified in the draft or
left=20
to the implementation. If the user hangs up, shouldn't the state machine
wait for the response to INVITE and send an ACK ?

Also section 4.2.4 states :
   A BYE request from either called or calling party terminates any=20
   pending INVITE, but the INVITE request transaction MUST be completed=20
   with a final response.=20

Does this mean that the ACK also needs to be sent after getting=20
the response ?=20

Thanks
Ajay.

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<P><FONT SIZE=3D2>In this diagram does &quot;give up&quot; mean the =
caller gave up and hung </FONT>

<BR><FONT SIZE=3D2>up the call or that the state machine gave up =
?</FONT>
</P>

<P><FONT SIZE=3D2>Since there is another arrow from the =
&quot;Calling&quot; state showing </FONT>

<BR><FONT SIZE=3D2>&quot;7 INVITE sent&quot;, I assume the arrow to the =
right of &quot;Calling&quot; state </FONT>

<BR><FONT SIZE=3D2>happens when the user hangs up. In this case =
shouldn't the state machine</FONT>

<BR><FONT SIZE=3D2>wait for the response to INVITE and send an ACK =
?</FONT>
</P>

<P><FONT SIZE=3D2>The arrow to the right of &quot;Call Proceeding&quot; =
state could happen if either</FONT>

<BR><FONT SIZE=3D2>the caller hangs up or the state machine hangs up. Is =
the timeout for when </FONT>

<BR><FONT SIZE=3D2>the state machine will give up in this state =
specified in the draft or left </FONT>

<BR><FONT SIZE=3D2>to the implementation. If the user hangs up, =
shouldn't the state machine</FONT>

<BR><FONT SIZE=3D2>wait for the response to INVITE and send an ACK =
?</FONT>
</P>

<P><FONT SIZE=3D2>Also section 4.2.4 states :</FONT>

<BR><FONT SIZE=3D2>&nbsp;&nbsp; A BYE request from either called or =
calling party terminates any </FONT>

<BR><FONT SIZE=3D2>&nbsp;&nbsp; pending INVITE, but the INVITE request =
transaction MUST be completed </FONT>

<BR><FONT SIZE=3D2>&nbsp;&nbsp; with a final response. </FONT>
</P>

<P><FONT SIZE=3D2>Does this mean that the ACK also needs to be sent =
after getting </FONT>

<BR><FONT SIZE=3D2>the response ? </FONT>
</P>

<P><FONT SIZE=3D2>Thanks</FONT>

<BR><FONT SIZE=3D2>Ajay.</FONT>
</P>

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From sip-admin@lists.bell-labs.com  Mon Jun 19 22:55:23 2000
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From: "Ajay Chitturi" <ajaych@Exchange.Microsoft.com>
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Subject: [SIP] Questions on section 10.1.1 of RFC2543bis
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The following is a paragraph in section 10.1.1 of the RFC2543bis draft

    When a user agent server receives a request, it checks the Call-ID
    against those of in-progress calls. If the Call-ID was found, it
    compares the tag value of To with the user's tag and rejects
    the request if the two do not match. If the From header, including
    any tag value, matches the value for an existing call leg, the
    server compares the CSeq header field value. If less than or equal
    to the current sequence number, the request is a retransmission.
    Otherwise, it is a new request. If the From header does not match
    an existing call leg, a new call leg is created.

Why is just the tag value of To compared (while the whole of From
header is compared).

Also could you please explain the scenario where the following
happens: "If the From header does not match an existing call leg, a
new call leg is created."


Thanks
Ajay.


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<P><FONT SIZE=3D2>The following is a paragraph in section 10.1.1 of the =
RFC2543bis draft</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; When a user agent server receives a =
request, it checks the Call-ID</FONT>

<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; against those of in-progress =
calls. If the Call-ID was found, it</FONT>

<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; compares the tag value of To with =
the user&#8217;s tag and rejects</FONT>

<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; the request if the two do not =
match. If the From header, including</FONT>

<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; any tag value, matches the value =
for an existing call leg, the</FONT>

<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; server compares the CSeq header =
field value. If less than or equal</FONT>

<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; to the current sequence number, =
the request is a retransmission.</FONT>

<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; Otherwise, it is a new request. If =
the From header does not match</FONT>

<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp; an existing call leg, a new call =
leg is created.</FONT>
</P>

<P><FONT SIZE=3D2>Why is just the tag value of To compared (while the =
whole of From</FONT>

<BR><FONT SIZE=3D2>header is compared).</FONT>
</P>

<P><FONT SIZE=3D2>Also could you please explain the scenario where the =
following</FONT>

<BR><FONT SIZE=3D2>happens: &quot;If the From header does not match an =
existing call leg, a</FONT>

<BR><FONT SIZE=3D2>new call leg is created.&quot;</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Thanks</FONT>

<BR><FONT SIZE=3D2>Ajay.</FONT>
</P>

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Subject: RE: [SIP] Questions on section 10.1.1 of RFC2543bis
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> The following is a paragraph in section 10.1.1 of the RFC2543bis draft
>     When a user agent server receives a request, it checks the Call-ID
>     against those of in-progress calls. If the Call-ID was found, it
>     compares the tag value of To with the user’s tag and rejects
>     the request if the two do not match. If the From header, including
>     any tag value, matches the value for an existing call leg, the
>     server compares the CSeq header field value. If less than or equal
>     to the current sequence number, the request is a retransmission.
>     Otherwise, it is a new request. If the From header does not match
>     an existing call leg, a new call leg is created.
> Why is just the tag value of To compared (while the whole of From
> header is compared).

Presumably because even if UACs are generating non-unique Call-IDs,
they are far less likely to be non-unique within the callee's space
(one would hope!).

The reason the tag of the To is considered is to ensure that this
Request really is for this user; if the tag does not match, then
the callee's request probably forked, and the request is for some
other destination (along another branch).

> Also could you please explain the scenario where the following
> happens: "If the From header does not match an existing call leg, a
> new call leg is created."

I guess this vaguely confirms my first paragraph.  That's a relief. &:)


 - Jo.



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From sip-admin@lists.bell-labs.com  Tue Jun 20 05:23:22 2000
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Subject: [SIP] How to ping Prxoy Server ?
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Hi,

                  Consider a  UAC which  uses a default proxy server
(say proxy 1)  whenever it has to send SIP request. It may so happen
that though transport connection is available at server where proxy is
suppose to run but proxy  may not  be responding due to
some reason ( may be due to temporary failure).  UAC keeps =
retransmitting
 the request for a given number of time and when it decides to give up =
it=20
sends  a CANCEL  message and then try other proxy server. Thus whenever
 a new SIP request is made UAC  first tries default proxy server  and  =
then everything=20
 repeats once again. This may create some problem  because it increase =
the call set
 up time ( in worst case it may happen at every intermediate proxy =
server). Is there=20
any method/way available in SIP using which status of proxy server can =
be known in advance ?.
 This  information helps the UAC in keeping track current active list of =
default=20
proxy server(s).

Thank You
Rafi Assadi H.M.
Silicon Automation Sysytems
Bangalore, INDIA



------=_NextPart_000_00D6_01BFDAC6.F7784F60
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	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2614.3500" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Hi,</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; Consider a&nbsp; UAC which&nbsp; =
uses a=20
default proxy server</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>(say proxy 1)  whenever it has to send =
SIP request.=20
It may so happen</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>that though</FONT><FONT face=3DArial =
size=3D2>=20
transport connection is available at server where proxy is</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>suppose to run but proxy&nbsp; may =
not&nbsp; be=20
responding due to</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>some reason ( may be due to temporary =
failure). =20
UAC keeps retransmitting</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;the request for a given =
</FONT><FONT=20
face=3DArial size=3D2>number of time and when it decides to give up it =
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>sends</FONT><FONT face=3DArial =
size=3D2>&nbsp; a=20
CANCEL&nbsp; message and</FONT><FONT face=3DArial size=3D2> then try =
other proxy=20
server. Thus whenever</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;a new SIP request</FONT><FONT =
face=3DArial=20
size=3D2>&nbsp;is made UAC</FONT><FONT face=3DArial size=3D2>&nbsp; =
first tries=20
default proxy server&nbsp; and&nbsp; then everything&nbsp;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;repeats once again. This may =
create=20
some</FONT><FONT face=3DArial size=3D2> problem&nbsp; because it =
increase the call=20
set</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;up time ( in worst case it may=20
happen</FONT><FONT face=3DArial size=3D2> at every intermediate proxy =
server). Is=20
there </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>any method/way available in =
SIP</FONT><FONT=20
face=3DArial size=3D2> using which status of proxy server can be known =
in advance=20
?.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;This&nbsp; </FONT><FONT =
face=3DArial=20
size=3D2>information helps the UAC in keeping track current active list =
of default=20
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>proxy server(s).</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thank You</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Rafi Assadi H.M.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Silicon Automation =
Sysytems</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Bangalore, INDIA</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

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From sip-admin@lists.bell-labs.com  Tue Jun 20 06:26:18 2000
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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: sip@lists.bell-labs.com
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Subject: [SIP] SIP URL telephone-subscriber syntax.
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RFC2543bis now states that the user part may be a "telephone-subscriber" as
defined in RFC2806 ( http://www.ietf.org/rfc/rfc2806.txt
<http://www.ietf.org/rfc/rfc2806.txt> ). And that this is not discernable by
BNF (hence the user=phone parameter).
 
So is the following a valid SIP URL:
 
sip:0w003585551234567; phone-context=+3585551234@gateway.com;user=phone
<mailto:phone-context=+3585551234@gateway.com;user=phone> 
 
Cheers,
 
Robert.

 



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From sip-admin@lists.bell-labs.com  Tue Jun 20 06:31:08 2000
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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: sip@lists.bell-labs.com
Subject: RE: [SIP] SIP URL telephone-subscriber syntax.
Date: Tue, 20 Jun 2000 03:29:51 -0700
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Please ignore the <mailto:> section Microsoft Outpatient 98 inserted in my
last email:
 
sip:0w003585551234567; phone-context=+3585551234@gateway.com;user=phone
<mailto:phone-context=+3585551234@gateway.com;user=phone> 
 
Cheers,
 
Robert.

 



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From sip-admin@lists.bell-labs.com  Tue Jun 20 06:34:46 2000
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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: sip@lists.bell-labs.com
Subject: RE: [SIP] SIP URL telephone-subscriber syntax.
Date: Tue, 20 Jun 2000 03:34:15 -0700
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Argh. Sorry everyone for filling up your Inbox's.

I also noticed it inserted a space into the URL. There weren't meant to be
any spaces.
 
sip:0w003585551234567;phone-context=+3585551234@gateway.com;user=phone
 
Cheers,
 
Robert.

 



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From sip-admin@lists.bell-labs.com  Tue Jun 20 06:47:38 2000
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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		: MIME media types for ISUP and QSIG Objects
	Author(s)	: E. Zimmerer, A. Vemuri, L. Ong, M. Zonoun,
                          M. Watson
	Filename	: draft-ietf-sip-isup-mime-02.txt
	Pages		: 6
	Date		: 19-Jun-00
	
This document describes MIME types for application/ISUP and 
application/QSIG objects for use in SIP applications, according to 
the rules defined in RFC 2048 [1].  These types can be used to identify 
ISUP and QSIG objects within a SIP message such as INVITE or INFO, 
as might be implemented when using SIP between legacy systems.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sip-isup-mime-02.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
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or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
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In the body type:
	"FILE /internet-drafts/draft-ietf-sip-isup-mime-02.txt".
	
NOTE:	The mail server at ietf.org can return the document in
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	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
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Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
Content-Type: Message/External-body;
	access-type="mail-server";
	server="mailserv@ietf.org"

Content-Type: text/plain
Content-ID:	<20000619125522.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-ietf-sip-isup-mime-02.txt

--OtherAccess
Content-Type: Message/External-body;
	name="draft-ietf-sip-isup-mime-02.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<20000619125522.I-D@ietf.org>

--OtherAccess--

--NextPart--




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From sip-admin@lists.bell-labs.com  Tue Jun 20 07:32:45 2000
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From: Gethin Liddell <gethin@ubiquity.net>
Organization: Ubiquity Software Corp.
To: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>, sip@lists.bell-labs.com
Subject: RE: [SIP] SIP URL telephone-subscriber syntax.
Date: Tue, 20 Jun 2000 12:03:15 +0100
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On Tue, 20 Jun 2000, Robert wrote:
> RFC2543bis now states that the user part may be a "telephone-subscriber" as
> defined in RFC2806 ( http://www.ietf.org/rfc/rfc2806.txt
> <http://www.ietf.org/rfc/rfc2806.txt> ). And that this is not discernable by
> BNF (hence the user=phone parameter).
>  
> So is the following a valid SIP URL:
>  
> sip:0w003585551234567;phone-context=+3585551234@gateway.com;user=phone
>  
> Cheers,
>  
> Robert.

as i understand it this is not legal.  there is a semi-colon
in the user part.  i think that all characters that appear in the
userinfo section have to belong to the character set defined by user

therefore the ; character needs to be escaped as follows:

sip:0w003585551234567%3bphone-context=+3585551234@gateway.com;user=phone

when this is parsed, the %3b should be unescaped back to ; to obtain
the correct meaning.

the bis does not make this clear in section 2 in the bnf.  perhaps some
explanatory text of the semantics of the bis is required within the
explanation of telephone-subscriber to the effect of.

"Any characters of the telephone-subscriber that do not exist in the
character set of `user' MUST be escaped."

HTH

gethin

-- 
Gethin Liddell
Ubiquity Software Corporation

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



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From sip-admin@lists.bell-labs.com  Tue Jun 20 10:39:41 2000
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Subject: Re: [SIP] How to ping Prxoy Server ?
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<snip>
> Is there 
> any method/way available in SIP using which status of proxy server can be known in advance ?.
>  This  information helps the UAC in keeping track current active list of default 
> proxy server(s).
> 

The OPTIONS method provides a very crude sort of ping for SIP.
Combined with the Max-Forwards: header, you can ping various nodes along 
a call signalling path. But this may still not give you the information
you desire; just because a proxy responds to an OPTIONS request, doesn't
mean it is capable of handling calls.


> Thank You
> Rafi Assadi H.M.
> Silicon Automation Sysytems
> Bangalore, INDIA
>  

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



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> rafi wrote:
> 
> 
> Hi,
> 
>                   Consider a  UAC which  uses a default proxy server
> (say proxy 1) whenever it has to send SIP request. It may so happen
> that though transport connection is available at server where proxy is
> suppose to run but proxy  may not  be responding due to
> some reason ( may be due to temporary failure). UAC keeps 
> retransmitting the request for a given number of time and when it 
> decides to give up it sends  a CANCEL  message and then try other 
> proxy server. Thus whenever  a new SIP request is made UAC  first 
> tries default proxy server  and  then everything repeats once again. 
> This may create some problem  because it increase the call set
>  up time ( in worst case it may happen at every intermediate proxy 
> server). Is there any method/way available in SIP using which status 
> of proxy server can be known in advance ?.  This  information helps 
> the UAC in keeping track current active list of default proxy 
> server(s).

It appears that you are talking of some heart-beat mechanism; clearly,
if you bind your UDP socket to the destination address, the second
write should return ECONNREFUSED (see bis, Section 1.4.2).  On TCP
sockets, the connect() will fail.  However, it appears that you are
assuming that the transport is available between a UAC and the outbound
proxy, just that the outbound proxy is "in a wierd state" (for the lack
of a better description).

SIP does not talk about any such heart-beat mechanisms to detect the
health of a proxy from a UAC.  You can build such a mechanism into your
proxy and UAC, but it will only be useful within your domain.  If all
proxies in the world supported heart-beat across domains, I would 
presume that the resulting traffic would be something to contend with.
Not to mention security.  If a proxy in domain A wants to heart-beat a
proxy in domain B, how does it authenticate itself?  Is the heart-beat
message encrypted?  You run into a set formidable problems once you go
down this path.

Note that TCP provides a heart-beat mechanism via the SO_KEEPALIVE sock-
et option.  When this option is set for a TCP socket and no data has
been exchanged across the socket (in either direction) for 2 hours, TCP
sends out a probe to its peer, to which the peer must respond.  However,
this is of no help to your case unless your SIP transactions last for
such a long duration.

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
IN Architecture and Internet Software Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184


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From sip-admin@lists.bell-labs.com  Tue Jun 20 11:44:56 2000
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According to 6.26 of the bis draft if a proxy receives a request
that has a zero value in the Max-Forwards header field it MUST
NOT forward the request. Instead, for the OPTIONS or REGISTER
methods, it MUST respond as the final recipient. For all other
methods, the server returns 483 (Too many hops).

How exactly should a Proxy respond as the final recipient to an
OPTIONS?

How should a proxy respond to a REGISTER if it is not responsible
for registrations within the domain named in the Request-URI?

In the case of other methods is sensible to always return 483?
For example what if it was an INVITE and the Proxy knows from a
location service a final response would have been 3XX/404.

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


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From sip-admin@lists.bell-labs.com  Tue Jun 20 12:06:20 2000
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Folks:

In the June 9th bis draft, Section 12.3.4 (Stateful Proxy: Receiving
Requests) states:

   When a stateful proxy receives a request, it checks the To,
   From (including tags), Call-ID and CSeq against existing
   request records.

I presume the To check does NOT include tags.  Can someone corroborate
this, please.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
IN Architecture and Internet Software Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
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From sip-admin@lists.bell-labs.com  Tue Jun 20 12:20:15 2000
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Vijay, I think it includes tags, since a brand new re-INVITE or BYE
may have a tag in the To header.


"Vijay K. Gurbani" wrote:
> 
> Folks:
> 
> In the June 9th bis draft, Section 12.3.4 (Stateful Proxy: Receiving
> Requests) states:
> 
>    When a stateful proxy receives a request, it checks the To,
>    From (including tags), Call-ID and CSeq against existing
>    request records.
> 
> I presume the To check does NOT include tags.  Can someone corroborate
> this, please.
> 
> Thanks,
> 
> - vijay
> --
> Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
> IN Architecture and Internet Software Group
> Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
> Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184
> 
> _______________________________________________
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Hi.

I would like to follow up on this.

Let's assume we have two users, which both have clients which only support
UDP.  They have different home proxies, and those proxies are stateful, use
TCP when possible and always insist on being included throughout any call
going through them.

Thus we have this setup with transport:

Client A  -------  UDP -------- Proxy A --------- TCP --------- Proxy B
-------- UDP -------- Client B

User A calls user B, and a call is successfully set up.  Both proxy A and
proxy B have include Record-Route headers in any proxied requests.

Now let us assume that the code implementing the underlying session being
established has a critical bug, and that when client A and B act on the
agreed session description, they both crash violently.  How will the
proxies in-between ever discover that?  After the ACK is sent by client A,
the proxies never hear from the clients again!

How is this handled?

Best regards,
Hörður K

--
Hordur Kvaran <hordurk@oz.com>
Software Developer - OZ.COM
Snorrabraut 54 - 105 Reykjavik - Iceland
Tel: +354 535 0000 - Direct tel: +354 535 0013






Jonathan Rosenberg <jdrosen@dynamicsoft.com>@lists.bell-labs.com on
17.06.2000 06:55:47

Sent by:  sip-admin@lists.bell-labs.com


To:   Loita Jahnsson <Loita.Jahnsson@uab.ericsson.se>
cc:   sip@lists.bell-labs.com

Subject:  Re: [SIP] proxy protocol conversion




Loita Jahnsson wrote:
>
> Hi,
>
>    I kind of new in SIP design and have a couple of
>    basic questions regarding proxy protocol conversion.
>
>    Assuming that "proxy protocol conversion" means
>    something like this:
>
>    ---transport-1--> proxy ----transport-2-->
>
>    For which cases are protocol conversion applicable
>    in a proxy:
>
>    * when the proxy adds more data to the message, so it
>      cannot be stored within a UDP package when proxying
>      the message?

Generally, determination of the transport to use for outgoing requests
will be driven by preferences set in the SRV records. In this case, they
would not be dependent on a change in size of the message. Normal SIP
procesing of a message will normally not make it much bigger. In any
case, the maximum UDP size is 64k, which you are unlikely to ever see in
SIP.

>
>    * when an outbound proxy receives a UDP package from
>      a UAC, stating transport=tcp in the Request-URI?

Yes, in this case the protocol goes out TCP. This is spelled out (along
with the above) in Section 1.4.2 of the bis draft. This section applies
independnt of the protocol the request came in on, so it may result in
a  "conversion" if the outgoing protocol selected was different from the
incoming.

>
>    * when a proxy is configured to always go through
>      another proxy using TCP, disregarding any transport
>      parameters in the SIP message?

You can always do that, although its not specified to work this way.
You'll run into troubles if the next hop doesn't support that protocol.

>
>    * when a proxy has one client which only knowns UDP
>      on one side and a client who only knows TCP on the
>      other (ie the attempt to contact the called client
>      with UDP fails)?

This would be determined through transport parameters in registered
Contacts, so conversion might take place here, yes.



>    If "proxy protocol conversions" refers to scenarios
>    like the ones mentioned above, what is it called when
>    you need to do this:
>
>    caller-client   ----request, transport1 ----> server
>
>    caller-client   <-- response, transport2 ---- same server :)
>
>    (For example if the response for some reason would be
>    bigger than max UDP size).

This is not supported. You could try to do it by putting a different
transport in the Via header than the one you actually used, but I don't
recommend doing that, since its unlikely to work. In any case, it sounds
like  you want the server to dynamically change the tranpsort for the
response if the response comes in and its big; having the client set the
protocol in the Via won't help that.


-Jonathan R.
--
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
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Hi All,
   I have a question about the usage of "action" parameter in REGISTER
request. According to the rfc2543bis [Sec 6.13]
"It indicates whether the client wishes that the server proxy or redirect 
future requests intended for the client"

Does this imply that a Network Server implementing a Registrar should
also be able to function both as a proxy and a redirect server ?

thanks,
aseem 


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From sip-admin@lists.bell-labs.com  Wed Jun 21 04:49:03 2000
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From: "Jo Hornsby" <jhornsby@ubiquity.net>
To: <ozsip@oz.com>
Cc: <sip@lists.bell-labs.com>
Subject: RE: [SIP] proxy protocol conversion
Date: Wed, 21 Jun 2000 09:45:32 +0100
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> Let's assume we have two users, which both have clients which
> only support UDP.  They have different home proxies, and those
> proxies are stateful, use TCP when possible and always insist on
> being included throughout any call going through them.
>
> Thus we have this setup with transport:
> 
> Client A  -------  UDP -------- Proxy A --------- TCP --------- Proxy B
> -------- UDP -------- Client B
> 
> User A calls user B, and a call is successfully set up.  Both proxy
> A and proxy B have include Record-Route headers in any proxied
> requests.
>
> Now let us assume that the code implementing the underlying session
> being established has a critical bug, and that when client A and B
> act on the agreed session description, they both crash violently.
> How will the proxies in-between ever discover that?  After the ACK
> is sent by client A, the proxies never hear from the clients again!
> 
> How is this handled?

I'm not massively confident that I'm understanding where there's
a problem; is it because you are concerned that the TCP connection
might persist unnecessarily indefinitely?  Else I can't see where
the "proxy protocol conversion" comes in?  I guess the solution
that jumps out at me is to have some sort of timer on a TCP connection,
which causes it to be closed if there's been no activity for some
period of time.

If this is more a Call-Stateful issue, probably the best solution
is for the proxies to mandate use of the "SIP Session Timer"; the
draft on this is available from Henning's site.

HTH,


 - Jo.



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To: "Aseem Agarwal" <aseem@trillium.com>, <sip@lists.bell-labs.com>
Cc: "Srini Beereddy" <srini@trillium.com>
Subject: RE: [SIP] Question on "action" parameter
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> Hi All,
>    I have a question about the usage of "action" parameter in REGISTER
> request. According to the rfc2543bis [Sec 6.13]
> "It indicates whether the client wishes that the server proxy or redirect
> future requests intended for the client"
>
> Does this imply that a Network Server implementing a Registrar should
> also be able to function both as a proxy and a redirect server ?

Not necessarily.  If you look at the definition of a Registrar in
section 1.3, you'll see that:
    "A registrar is a server that accepts REGISTER requests.
     A registrar is typically co-located with a proxy or redirect
     server and MAY offer location services."

As you suggest, a Registrar is a logical role; the physical server
may actually also look like a stateful proxy, in which case it
will probably obey the action prescribed.

However, in other cases a Registrar may be all that the physical
server is.  In this case, more-than-likely it's talking to some
location server, which proxies/redirectservers also talk to.

HTH,


 - Jo.



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From sip-admin@lists.bell-labs.com  Wed Jun 21 09:43:41 2000
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From: "McMurry, Kathleen" <Kathleen.McMurry@wcom.com>
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Subject: [SIP] tags in ACKs in rfc2543bis-00
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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.

--Boundary_(ID_fJ1I5oc1mj29MOTO2zDmBg)
Content-type: text/plain; charset=ISO-8859-1

I am confused about the new statement in Section 11.4 that states "A UAC
copies the tag from the final 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 request."

This seems to conflict with Section 4.2.2 which states "A proxy server
receiving an ACK request after having send a 3xx, 4xx, 5xx, or 6xx response
must make a determination about whether the ACK is for it, or for some user
agent or proxy server further downstream.  This determination is made by
examining the tag in the To field."

If the UAC doesn't copy the tag into the ACK, how is the proxy server
supposed to make the determination that the ACK request is for it?

Thanks, 

Kathleen McMurry


--Boundary_(ID_fJ1I5oc1mj29MOTO2zDmBg)
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charset=3DISO-8859-1">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2652.35">
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<BODY>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">I am confused about =
the new statement in Section 11.4 that states &quot;A UAC copies the =
tag from the final 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 request.&quot;</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">This seems to =
conflict with Section 4.2.2 which states &quot;A proxy server receiving =
an ACK request after having send a 3xx, 4xx, 5xx, or 6xx response must =
make a determination about whether the ACK is for it, or for some user =
agent or proxy server further downstream.&nbsp; This determination is =
made by examining the tag in the To field.&quot;</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">If the UAC doesn't =
copy the tag into the ACK, how is the proxy server supposed to make the =
determination that the ACK request is for it?</FONT></P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Thanks, </FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Kathleen =
McMurry</FONT>
</P>

</BODY>
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From sip-admin@lists.bell-labs.com  Wed Jun 21 09:58:57 2000
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To: "McMurry, Kathleen" <Kathleen.McMurry@wcom.com>
Cc: <sip@lists.bell-labs.com>
Subject: RE: [SIP] tags in ACKs in rfc2543bis-00
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> I am confused about the new statement in Section 11.4 that states "A
> UAC copies the tag from the final 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 request."
>
> This seems to conflict with Section 4.2.2 which states "A proxy
> server receiving an ACK request after having send a 3xx, 4xx, 5xx,
> or 6xx response must make a determination about whether the ACK is
> for it, or for some user agent or proxy server further downstream.
> This determination is made by examining the tag in the To field."
>
> If the UAC doesn't copy the tag into the ACK, how is the proxy server
> supposed to make the determination that the ACK request is for it?

It does copy the tag into the ACK; it just doesn't copy the tag into
any other requests.  For instance, if the UA was to retry an INVITE
(with a different CSeq, or something) after having received a non-2xx
final response, it would be wrong for it to update the To header to
include a tag that was in the response; only if the INVITE succeeds
should the To header be updated.

HTH,


 - Jo.



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From sip-admin@lists.bell-labs.com  Wed Jun 21 11:40:54 2000
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Date: Wed, 21 Jun 2000 16:39:09 +0100
From: Phil Hoffer <phoffer@ubiquity.net>
Organization: Ubiquity Software Corporation
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Hi,

Section 15.1.1 of the rfc2543bis draft is missing information, and a BNF entry with respect to pgp-pubalgolrithm.

Presumably, pgp-pubalgolrithm should be included in the BNF for pgp-params in section 15.1.1

Also some text about the parameter is required, e.g. what is the default value of pgp-pubalgolrithm if it is omitted from WWW-Authenticate header.
Is it assumed to be RSA?

Hope that helps
Regards
--
Phil Hoffer                         Ubiquity Software Corporation
Software Developer                  Ubiquity House
Tel: +44 (0) 1633-765600            Langstone Park
Fax: +44 (0) 1633-765601            Newport   NP18 2LH
URL: http://www.ubiquity.net
mailto:phoffer@ubiquity.net



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From sip-admin@lists.bell-labs.com  Wed Jun 21 13:00:15 2000
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Hi,
I just wanted to invite those interested to participate in the SIP Forum
work. 

SIP Forum was announced yesterday at the Voice on the net Europe conference
to be a natural meeting place for those companies and individuals working
with SIP. The forum will NOT do any technical work, all such ideas should be
forwarded to the IETF, so there will be no overlap between the
organizations. SIP Forum will spread information on how to utilize SIP as a
tool for service creation, how to deploy SIP in real networks etc. It will
also list available products making use of SIP. If you have such products
available for purchase right now, please let the SIP forum webmaster know
about them. The organization is an open non profit Swedish association, open
to anyone that can contribute on the SIP information sharing, on a not so
very technical level. 

More information is found at http://www.sipforum.com

Welcome!
Jörgen Björkner, Chairman of SIP Forum


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From sip-admin@lists.bell-labs.com  Wed Jun 21 13:47:20 2000
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Subject: RE: [SIP] Announcement of SIP Forum
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Interesting...there is nothing at www.sipforum.com .... however, a site much
like what you describe is at www.sipforum.org.

-----Original Message-----
From: Jörgen Björkner [mailto:Jorgen.Bjorkner@hotsip.com]
Sent: Wednesday, June 21, 2000 11:55 AM
To: 'sip@lists.bell-labs.com'
Subject: [SIP] Announcement of SIP Forum


Hi,
I just wanted to invite those interested to participate in the SIP Forum
work. 

SIP Forum was announced yesterday at the Voice on the net Europe conference
to be a natural meeting place for those companies and individuals working
with SIP. The forum will NOT do any technical work, all such ideas should be
forwarded to the IETF, so there will be no overlap between the
organizations. SIP Forum will spread information on how to utilize SIP as a
tool for service creation, how to deploy SIP in real networks etc. It will
also list available products making use of SIP. If you have such products
available for purchase right now, please let the SIP forum webmaster know
about them. The organization is an open non profit Swedish association, open
to anyone that can contribute on the SIP information sharing, on a not so
very technical level. 

More information is found at http://www.sipforum.com

Welcome!
Jörgen Björkner, Chairman of SIP Forum


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http://lists.bell-labs.com/mailman/listinfo/sip


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From sip-admin@lists.bell-labs.com  Wed Jun 21 17:56:24 2000
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Date: Wed, 21 Jun 2000 17:56:13 -0400
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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One of the ideas that was discussed at the Telia dinner in Stockholm was
whether there was a need for a basic SIP certification program. There
are obvious pluses and minuses for this, but my concern is that if this
is a real need, I can think of a few organizations (looking for
post-H.323 work...) that I would rather *not* see doing this.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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From sip-admin@lists.bell-labs.com  Wed Jun 21 19:01:44 2000
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Date: Wed, 21 Jun 2000 13:31:39 -0700
To: Henry Sinnreich <Henry.Sinnreich@WCom.com>
From: Matt Holdrege <matt@ipverse.com>
Subject: RE: [SIP] SIP->XML->SOAP
Cc: sip@lists.bell-labs.com
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At 07:09 AM 6/18/00 -0500, Henry Sinnreich wrote:
> >> SIP->XML would converge the SIP methods and required
> >parameters to XML and
> >> allows the introduction of a DTD or schema to define
> >the syntax of a valid
> >> SIP message.
>
>Do not forget that endangering the deployment of SIP as is,
>would set IP telephony back for a generation and leave the field
>for "softswitches" and other such nonsense. Nobody on this
>mailing list would then live to see true/open Internet
>communications, but just the old circuit switch central control
>model with its terrible signaling. Just using IP as a cheap
>wire.

Henry,

This is the first time I recall seeing "softswitches" being denigrated on 
an IETF list. I would hope you realize that they are an enabler to get the 
circuit switched world to migrate to open communications? Saying that 
Softswitches will preserve the circuit switched world with it's "terrible 
signalling" is a bit short-sighted.

You should also realize that SIP is doing quite well in the area of market 
acceptance. The more people buy into SIP the more evolution will be 
required. Change is inevitable even though we must try to manage it well. 
XML is one such change that seems (to me) to be inevitable. So it would be 
better to try and manage the interactions with SIP and XML here in the IETF 
rather than elsewhere. Besides, it's part of the openess that we all should 
encourage.



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From sip-admin@lists.bell-labs.com  Wed Jun 21 22:04:31 2000
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From: "Brett Tate" <brett@broadsoft.com>
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Date: Wed, 21 Jun 2000 22:07:18 -0400
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Subject: [SIP] Hold & then re-INVITE without media
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------=_NextPart_000_065F_01BFDBCD.106FE350
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	charset="iso-8859-1"
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A -- Invite (hold media) --> B
  <-- 200 (hold media) --   B
   --- Ack -------------------->  B

After A requests B to hold, how can A without providing media request=20
that B stop holding and provide media?

I just wanted to verify that the following should work unless B=20
has also pushed the hold button.

A -- Invite -------------------> B
  <-- 200 (media) ---------   B
   --- Ack (media) -------->  B



------=_NextPart_000_065F_01BFDBCD.106FE350
Content-Type: text/html;
	charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2722.2800" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT size=3D2>
<DIV><FONT size=3D2>A -- Invite (hold media) --&gt; B</FONT></DIV>
<DIV><FONT size=3D2>&nbsp; &lt;-- 200 (hold media) --&nbsp;&nbsp; =
B</FONT></DIV>
<DIV><FONT size=3D2>&nbsp; &nbsp;--- Ack --------------------&gt;&nbsp;=20
B</FONT></DIV></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2>After A requests B to hold, how can A without =
providing media=20
</FONT><FONT size=3D2>request </FONT></DIV>
<DIV><FONT size=3D2>that B</FONT><FONT size=3D2> stop holding and =
provide=20
media?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2>I just wanted to verify that the following should =
work unless=20
B </FONT></DIV>
<DIV><FONT size=3D2>has also pushed the hold button.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT size=3D2>A -- Invite -------------------&gt; B</FONT></DIV>
<DIV><FONT size=3D2>&nbsp; &lt;-- 200 (media) ---------&nbsp;&nbsp; =
B</FONT></DIV>
<DIV><FONT size=3D2>&nbsp; &nbsp;--- Ack (media) --------&gt;&nbsp; =
B</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

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From sip-admin@lists.bell-labs.com  Thu Jun 22 01:11:50 2000
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From: "rafi" <rafi@sasi.com>
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Date: Thu, 22 Jun 2000 10:34:27 +0530
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Subject: [SIP] Phone user in Buddy List ?
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Hi,


     Can I have phone user in my buddy list ?.=20
Thus,  is it possible for a gateway to act as PA?
In that case  gateway may  need some support from
PSTN world.=20


Thank You
Rafi Assadi H.M
Silicon Automation Systems
Bangalore, INDIA

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi,</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp;&nbsp; Can I have =
phone user in=20
my buddy list ?. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Thus,&nbsp; is it possible for a =
gateway to act as=20
PA?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>In that case&nbsp; gateway may  need =
some support=20
from</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>PSTN world. </FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thank You</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Rafi Assadi H.M</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Silicon Automation Systems</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Bangalore, =
INDIA</FONT></DIV></BODY></HTML>

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From sip-admin@lists.bell-labs.com  Thu Jun 22 02:55:49 2000
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Date: Thu, 22 Jun 2000 12:22:51 +0530 (IST)
From: A Srinath <srinath@sasi.com>
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hi,
	I have a doubt regarding the number of responses that a caller may
receive after having sent an INVITE request. 

	Section 11.3 of RFC 2543 says 
	Multiple responses may arrive at a UAC for a single INVITE request 
due to a forking proxy. 

	But the algorithm outlined in section 12.4 ensures that only one
response is sent by the proxy upstream. 
	
	If this behavior is followed by the proxy then how does a UAC
receive multiple responses, or section 11.3 means all responses including 
provisional? 


Regards,

 Srinath A
 Silicon Automation Systems       Phone: 5276100, 5276108 extn 4220
 #1309, 10th Main, HAL III Stage   
 Bangalore 560 008, India                    



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From sip-admin@lists.bell-labs.com  Thu Jun 22 05:22:59 2000
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Hi all,
I haven't found anything related to sending DTMFs in the documentation I've
read about SIP. Are you expected to send DTMFs in band within the audio
stream?
Thanks in advance,
Fernando Orus



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From sip-admin@lists.bell-labs.com  Thu Jun 22 05:28:50 2000
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From: "Jo Hornsby" <jhornsby@ubiquity.net>
To: "A Srinath" <srinath@sasi.com>, <sip@lists.bell-labs.com>
Subject: RE: [SIP] Caller receives responses
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> 	I have a doubt regarding the number of responses that a caller may
> receive after having sent an INVITE request. 
> 
> 	Section 11.3 of RFC 2543 says 
> 	Multiple responses may arrive at a UAC for a single INVITE request 
> due to a forking proxy. 
> 
> 	But the algorithm outlined in section 12.4 ensures that only one
> response is sent by the proxy upstream. 

Not quite.  The problem with the code in section 12.4 is that it
doesn't tell the whole story (else proxy-implementors would be
getting a free ride, eh? (:&).  It waits until either the request
is CANCELled, a 2xx response is received (on any branch), or non-2xx
final responses are received on all branches; you don't get to
see what happens after that.

If you read the text following the code, you'll see under the 2xx
entry that "The proxy MUST forward the response upstream towards
the client".  Now I would guess that most proxies CANCEL outstanding
branches once they've seen a 2xx (even though it's only a "MAY");
however, this does not preclude a 2xx and CANCEL "crossing on the
wire".

Furthermore, in certain conditions, a client may see multiple
non-2xx class responses to a request as the result of a downstream
forking proxy being naughty.  (This scenario is documented in section
4.2.2.)

Therefore, you cannot assume that a request will result in at
most one final response.  The multiple non-2xx class case may be
reasonably unlikely, but the multiple 2xx case is most definitely
a reality.

> 	If this behavior is followed by the proxy then how does a UAC
> receive multiple responses, or section 11.3 means all responses
> including provisional? 

My feeling that 11.3 is leaning more towards "Multiple final
responses".

Of course, multiple (in the sense that they're from different
servers) provisional (non-100) is also possible, and probably
desirable in the face of Session Progress.

Me -- I'm just glad I pulled the proxy straw. &:)

HTH,


 - Jo.



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Fernando Orus wrote:
> 
> Hi all,
> I haven't found anything related to sending DTMFs in the documentation I've
> read about SIP. Are you expected to send DTMFs in band within the audio
> stream?
> Thanks in advance,
> Fernando Orus

There is no definitive agreed answer, but the basic
options are that dtmf signalling gets sent in a SIP INFO 
message (draft-culpepper-sip-info-event-00.txt) or
is carried encoded within the RTP stream (RFC 2833).
Look back at the list archives for a discussion of their 
relative merits, both methods could even coexist.

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


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From sip-admin@lists.bell-labs.com  Thu Jun 22 05:53:12 2000
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Subject: RE: [SIP] SIP URL telephone-subscriber syntax.
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Hi Gethin,

telephone-subscriber can also contain escaped private prefixes - is there
any overlap in the character set that must be escaped for a SIP user field
and those that can be escaped for a private prefix ? I couldn't see any at
first glance - in fact the semi-colon is the only character I could find
that needs to be escaped (if you ignore possibility of quoted string in the
future-extension field).

Robert.

-----Original Message-----
From: Gethin Liddell [mailto:gethin@ubiquity.net]
Sent: Tuesday, June 20, 2000 7:03 PM
To: Fairlie-Cuninghame, Robert; sip@lists.bell-labs.com
Subject: RE: [SIP] SIP URL telephone-subscriber syntax.


On Tue, 20 Jun 2000, Robert wrote:
> RFC2543bis now states that the user part may be a "telephone-subscriber"
as
> defined in RFC2806 ( http://www.ietf.org/rfc/rfc2806.txt
> <http://www.ietf.org/rfc/rfc2806.txt> ). And that this is not discernable
by
> BNF (hence the user=phone parameter).
>  
> So is the following a valid SIP URL:
>  
> sip:0w003585551234567;phone-context=+3585551234@gateway.com;user=phone
>  
> Cheers,
>  
> Robert.

as i understand it this is not legal.  there is a semi-colon
in the user part.  i think that all characters that appear in the
userinfo section have to belong to the character set defined by user

therefore the ; character needs to be escaped as follows:

sip:0w003585551234567%3bphone-context=+3585551234@gateway.com;user=phone

when this is parsed, the %3b should be unescaped back to ; to obtain
the correct meaning.

the bis does not make this clear in section 2 in the bnf.  perhaps some
explanatory text of the semantics of the bis is required within the
explanation of telephone-subscriber to the effect of.

"Any characters of the telephone-subscriber that do not exist in the
character set of `user' MUST be escaped."

HTH

gethin

-- 
Gethin Liddell
Ubiquity Software Corporation

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



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From sip-admin@lists.bell-labs.com  Thu Jun 22 05:55:41 2000
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UBIQUITY SOFTWARE LAUNCHES WWW.SIPCENTER.COM 
WITH CO-SPONSORSHIP FROM SUN MICROSYSTEMS AND TELECOM TECHNOLOGIES

SIP Center  - an independent portal dedicated to the acceleration of the
commercial development of Session Initiation Protocol software.

VON, Stockholm, Sweden, June 20th, 2000 - Ubiquity Software Corporation,
a major developer of 'End to End' SIP Service Solutions, today announced
the launch of the new web site: http://www.sipcenter.com - a portal
dedicated to the development and deployment of SIP (Session Initiation
Protocol) based products. Ubiquity's co-sponsors, Sun Microsystems and
telecom technologies, will be assisting with promotion, next generation
technologies and services. The SIP Center web site, designed for both
technical and commercial use, is an open and independent site where SIP
ideas and views can be exchanged. With a unique Test Bed Area, SIP
software developers can access to SIP software for interoperability
testing 24hrs a day, 7 days a week, 365 days a year.

There are no membership fees or joining process and all companies are
invited to participate in the exchange of ideas and news items related
to the SIP community.  Interactive forums have been setup to facilitiate
rapid exchange of information.

The SIP Center is actively seeking sponsorship from other companies and
organisations.

-- 
Michael Doyle                         Ubiquity Software Corporation
Chief Technology Officer              Ubiquity House
Tel: +44 (0) 1633-765600              Langstone Park
Fax: +44 (0) 1633-765601              Newport   NP18 2LH
URL: http://www.ubiquity.net


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From sip-admin@lists.bell-labs.com  Thu Jun 22 07:06:10 2000
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From: Gethin Liddell <gethin@ubiquity.net>
Organization: Ubiquity Software Corp.
To: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>, sip@lists.bell-labs.com
Subject: RE: [SIP] SIP URL telephone-subscriber syntax.
Date: Thu, 22 Jun 2000 11:01:48 +0100
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Hi,

Sorry, not 100% sure what you mean here.

Yes, private-prefix (rfc 2806 - TelURI) does allow characters that will
need to be escaped when printed/sent or what ever because of the
escaping defined in rfc 2396 (Generic URI).  Examples of these
characters are `"', `#',  `<' and `>' because they are used as
delimiters else where. (in fact `%' also cannot exist in its own right
as it is defined as being the escape character delimiter). 

However, as far as the Tel URI rfc (2806) is concerend, the character @
(for example) will not have to be escaped.  This, of course, will not be
valid in the sipurl as it is used to separate the `userinfo' and
`hostinfo' elements and will need to be escaped.

For example, if the phone-context is actually stored as:

"Context1@Context2"

then the tel url will have to look like:

tel:+12345;phone-context=%22Context1@Context2%22

however, it would NOT be valid to just enter this as a sip url as
follows:

sip:+12345;phone-context=%22Context1@Context2%22@company.com;user=phone

for the obvious problem of the @. this would need to be translated to:

sip:+12345;phone-context=%22Context1%40Context2%22@company.com;user=phone

as for other characters that need to be escaped?  The tel URI set of
escaped characters will be a subset of the characters that need to be
escaped for the `user' definition.  So as long as the escaping is done
to the more restrictive `user' element, all will be OK.

This property is explaind in the rfc 2543 (and bis) in section 2:

"Note that reserved characters have to be escaped and that the  set of
characters reserved within any given URI component is defined by that
component. In general, a character is reserved if the semantics of the
URI changes if the character is replaced with its escaped US-ASCII
encoding."

As for the future-extension and the quoted string, this could still not
exist in its original format.  i.e. the quotes and any other characters
that are not valid URI characters (as defined in rfc 2396) within the
quotes will have to be escaped.

HTH 

gethin

On Thu, 22 Jun 2000, Fairlie-Cuninghame, Robert wrote:
> Hi Gethin,
> 
> telephone-subscriber can also contain escaped private prefixes - is there
> any overlap in the character set that must be escaped for a SIP user field
> and those that can be escaped for a private prefix ? I couldn't see any at
> first glance - in fact the semi-colon is the only character I could find
> that needs to be escaped (if you ignore possibility of quoted string in the
> future-extension field).
> 
> Robert.
> 
> -----Original Message-----
> From: Gethin Liddell [mailto:gethin@ubiquity.net]
> Sent: Tuesday, June 20, 2000 7:03 PM
> To: Fairlie-Cuninghame, Robert; sip@lists.bell-labs.com
> Subject: RE: [SIP] SIP URL telephone-subscriber syntax.
> 
> 
> On Tue, 20 Jun 2000, Robert wrote:
> > RFC2543bis now states that the user part may be a "telephone-subscriber"
> as
> > defined in RFC2806 ( http://www.ietf.org/rfc/rfc2806.txt
> > <http://www.ietf.org/rfc/rfc2806.txt> ). And that this is not discernable
> by
> > BNF (hence the user=phone parameter).
> >  
> > So is the following a valid SIP URL:
> >  
> > sip:0w003585551234567;phone-context=+3585551234@gateway.com;user=phone
> >  
> > Cheers,
> >  
> > Robert.
> 
> as i understand it this is not legal.  there is a semi-colon
> in the user part.  i think that all characters that appear in the
> userinfo section have to belong to the character set defined by user
> 
> therefore the ; character needs to be escaped as follows:
> 
> sip:0w003585551234567%3bphone-context=+3585551234@gateway.com;user=phone
> 
> when this is parsed, the %3b should be unescaped back to ; to obtain
> the correct meaning.
> 
> the bis does not make this clear in section 2 in the bnf.  perhaps some
> explanatory text of the semantics of the bis is required within the
> explanation of telephone-subscriber to the effect of.
> 
> "Any characters of the telephone-subscriber that do not exist in the
> character set of `user' MUST be escaped."
> 
> HTH
> 
> gethin
> 
> -- 
> Gethin Liddell
> Ubiquity Software Corporation
> 
> http://www.ubiquity.net
> mailto:gethin@ubiquity.net
> 
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
-- 
Gethin Liddell
Ubiquity Software Corporation

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


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Subject: Re: [SIP] SIP URL telephone-subscriber syntax.
Date: Thu, 22 Jun 2000 19:42:09 +1000
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Hi

I am new to SIP
Is there any open SIP stack available for SIP?

pls help
thanks in advance..

Jose Cherian. P.




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Date: Thu, 22 Jun 2000 10:32:58 -0400
From: Shail Bhatnagar <shbhatna@cisco.com>
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To: Jo Hornsby <jhornsby@ubiquity.net>
Cc: A Srinath <srinath@sasi.com>, sip@lists.bell-labs.com
Subject: Re: [SIP] Caller receives responses
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While we are here, I would like to hear what proxy implementors think
about receiving multiple final respones on a single branch. Is it 
reasonable to simply drop final responses received after the first
one ?? I guess, dropping 200 response is not good, but it is very
unlikely that a 200 response is received on a branch after a non-200
final response. If the proxy decides to forward all 200 responses
upstream, regardless of the state of the branch and transaction, the
branch state gets kind of messy.

Also, it was suggested on this list that proxies should not drop 
final responses which do not append a tag to the To header. If a proxy
tries to be lenient and does this, it complicates matters even more.

Comments ??

Shail


Jo Hornsby wrote:
> 
> >       I have a doubt regarding the number of responses that a caller may
> > receive after having sent an INVITE request.
> >
> >       Section 11.3 of RFC 2543 says
> >       Multiple responses may arrive at a UAC for a single INVITE request
> > due to a forking proxy.
> >
> >       But the algorithm outlined in section 12.4 ensures that only one
> > response is sent by the proxy upstream.
> 
> Not quite.  The problem with the code in section 12.4 is that it
> doesn't tell the whole story (else proxy-implementors would be
> getting a free ride, eh? (:&).  It waits until either the request
> is CANCELled, a 2xx response is received (on any branch), or non-2xx
> final responses are received on all branches; you don't get to
> see what happens after that.
> 
> If you read the text following the code, you'll see under the 2xx
> entry that "The proxy MUST forward the response upstream towards
> the client".  Now I would guess that most proxies CANCEL outstanding
> branches once they've seen a 2xx (even though it's only a "MAY");
> however, this does not preclude a 2xx and CANCEL "crossing on the
> wire".
> 
> Furthermore, in certain conditions, a client may see multiple
> non-2xx class responses to a request as the result of a downstream
> forking proxy being naughty.  (This scenario is documented in section
> 4.2.2.)
> 
> Therefore, you cannot assume that a request will result in at
> most one final response.  The multiple non-2xx class case may be
> reasonably unlikely, but the multiple 2xx case is most definitely
> a reality.
> 
> >       If this behavior is followed by the proxy then how does a UAC
> > receive multiple responses, or section 11.3 means all responses
> > including provisional?
> 
> My feeling that 11.3 is leaning more towards "Multiple final
> responses".
> 
> Of course, multiple (in the sense that they're from different
> servers) provisional (non-100) is also possible, and probably
> desirable in the face of Session Progress.
> 
> Me -- I'm just glad I pulled the proxy straw. &:)
> 
> HTH,
> 
>  - Jo.
> 
> _______________________________________________


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To: <shbhatna@cisco.com>
Cc: "A Srinath" <srinath@sasi.com>, <sip@lists.bell-labs.com>
Subject: RE: [SIP] Caller receives responses
Date: Thu, 22 Jun 2000 16:02:27 +0100
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> While we are here, I would like to hear what proxy implementors think
> about receiving multiple final respones on a single branch. Is it 
> reasonable to simply drop final responses received after the first
> one ??

If you have state, and the response is not 2xx, then maybe so.

The reason I say this is because I would guess that if this situation
has arisen, then it's probably due to some server responding in a
very tardy fashion; if this is the case, then it's not so unlikely
that any proxies along the route have dropped transaction state, and
thus they'll be forwarded the response statelessly.

One reason against this behaviour is that I guess you could still
be pending results on some branches when you receive a different
final response on a branch that was marked as done.  This probably
would tend to indicate that a downstream server is very ill, however.

Of course, if the response is to an INVITE, you really should ACK
if it's non-2xx class...

> I guess, dropping 200 response is not good, but it is very
> unlikely that a 200 response is received on a branch after a non-200
> final response.

I don't see that it's any less likely than receiving any other
different response.

> If the proxy decides to forward all 200 responses
> upstream, regardless of the state of the branch and transaction, the
> branch state gets kind of messy.

How so?  If the proxy still has state on the transaction, then the
branch will either be done or it won't.  Forwarding 2xx class
upstream in a stateless fashion doesn't really complicate this;
you just set the boolean unconditionally. &:)

> Also, it was suggested on this list that proxies should not drop 
> final responses which do not append a tag to the To header. If a proxy
> tries to be lenient and does this, it complicates matters even more.

I can see ambiguity arising with ACKs...is there anything else?


 - Jo.



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-----Original Message-----
From: ozsip@oz.com [mailto:ozsip@oz.com ]
Sent: 20 June 2000 21:16
To: sip@lists.bell-labs.com
Subject: Re: [SIP] proxy protocol conversion



Hi.

I would like to follow up on this.

Let's assume we have two users, which both have clients which only support
UDP.  They have different home proxies, and those proxies are stateful, use
TCP when possible and always insist on being included throughout any call
going through them.


Thus we have this setup with transport:

Client A  -------  UDP -------- Proxy A --------- TCP --------- Proxy B
-------- UDP -------- Client B

User A calls user B, and a call is successfully set up.  Both proxy A and
proxy B have include Record-Route headers in any proxied requests.

Now let us assume that the code implementing the underlying session being
established has a critical bug, and that when client A and B act on the
agreed session description, they both crash violently.  How will the
proxies in-between ever discover that?  After the ACK is sent by client A,
the proxies never hear from the clients again!

How is this handled?

Best regards,
Hörður K

--
Hordur Kvaran <hordurk@oz.com>
Software Developer - OZ.COM
Snorrabraut 54 - 105 Reykjavik - Iceland
Tel: +354 535 0000 - Direct tel: +354 535 0013






Jonathan Rosenberg <jdrosen@dynamicsoft.com>@lists.bell-labs.com on
17.06.2000 06:55:47

Sent by:  sip-admin@lists.bell-labs.com


To:   Loita Jahnsson <Loita.Jahnsson@uab.ericsson.se>
cc:   sip@lists.bell-labs.com

Subject:  Re: [SIP] proxy protocol conversion




Loita Jahnsson wrote:
>
> Hi,
>
>    I kind of new in SIP design and have a couple of
>    basic questions regarding proxy protocol conversion.
>
>    Assuming that "proxy protocol conversion" means
>    something like this:
>
>    ---transport-1--> proxy ----transport-2-->
>
>    For which cases are protocol conversion applicable
>    in a proxy:
>
>    * when the proxy adds more data to the message, so it
>      cannot be stored within a UDP package when proxying
>      the message?

Generally, determination of the transport to use for outgoing requests
will be driven by preferences set in the SRV records. In this case, they
would not be dependent on a change in size of the message. Normal SIP
procesing of a message will normally not make it much bigger. In any
case, the maximum UDP size is 64k, which you are unlikely to ever see in
SIP.

>
>    * when an outbound proxy receives a UDP package from
>      a UAC, stating transport=tcp in the Request-URI?

Yes, in this case the protocol goes out TCP. This is spelled out (along
with the above) in Section 1.4.2 of the bis draft. This section applies
independnt of the protocol the request came in on, so it may result in
a  "conversion" if the outgoing protocol selected was different from the
incoming.

>
>    * when a proxy is configured to always go through
>      another proxy using TCP, disregarding any transport
>      parameters in the SIP message?

You can always do that, although its not specified to work this way.
You'll run into troubles if the next hop doesn't support that protocol.

>
>    * when a proxy has one client which only knowns UDP
>      on one side and a client who only knows TCP on the
>      other (ie the attempt to contact the called client
>      with UDP fails)?

This would be determined through transport parameters in registered
Contacts, so conversion might take place here, yes.



>    If "proxy protocol conversions" refers to scenarios
>    like the ones mentioned above, what is it called when
>    you need to do this:
>
>    caller-client   ----request, transport1 ----> server
>
>    caller-client   <-- response, transport2 ---- same server :)
>
>    (For example if the response for some reason would be
>    bigger than max UDP size).

This is not supported. You could try to do it by putting a different
transport in the Via header than the one you actually used, but I don't
recommend doing that, since its unlikely to work. In any case, it sounds
like  you want the server to dynamically change the tranpsort for the
response if the response comes in and its big; having the client set the
protocol in the Via won't help that.


-Jonathan R.
--
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Thu Jun 22 11:32:59 2000
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Subject: RE: [SIP] Announcement of SIP Forum
Date: Thu, 22 Jun 2000 17:20:03 +0200
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Hi,
yes of course you are right, it is www.sipforum.org. It should be an alias
for www.sipforum.com point there, but it is appearantly not working.

Thanks!
/Jörgen

-----Original Message-----
From: Kevin Summers
To: Jörgen Björkner; 'sip@lists.bell-labs.com'
Sent: 2000-06-21 19:46
Subject: RE: [SIP] Announcement of SIP Forum

Interesting...there is nothing at www.sipforum.com .... however, a site
much
like what you describe is at www.sipforum.org.

-----Original Message-----
From: Jörgen Björkner [mailto:Jorgen.Bjorkner@hotsip.com]
Sent: Wednesday, June 21, 2000 11:55 AM
To: 'sip@lists.bell-labs.com'
Subject: [SIP] Announcement of SIP Forum


Hi,
I just wanted to invite those interested to participate in the SIP Forum
work. 

SIP Forum was announced yesterday at the Voice on the net Europe
conference
to be a natural meeting place for those companies and individuals
working
with SIP. The forum will NOT do any technical work, all such ideas
should be
forwarded to the IETF, so there will be no overlap between the
organizations. SIP Forum will spread information on how to utilize SIP
as a
tool for service creation, how to deploy SIP in real networks etc. It
will
also list available products making use of SIP. If you have such
products
available for purchase right now, please let the SIP forum webmaster
know
about them. The organization is an open non profit Swedish association,
open
to anyone that can contribute on the SIP information sharing, on a not
so
very technical level. 

More information is found at http://www.sipforum.com

Welcome!
Jörgen Björkner, Chairman of SIP Forum


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        "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>,
        sip@lists.bell-labs.com
Subject: Re: [SIP] SIP URL telephone-subscriber syntax.
References: <B16E9BA540A0D211A11D00105A65571F9DAFA1@exchangesvr.nuera.com> <00062211545801.21137@gethin> <00a701bfdc2e$28fc1a40$a164a8c0@josetel>
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"Jose Cherian. P." wrote:
> 
> Hi
> 
> I am new to SIP
> Is there any open SIP stack available for SIP?
> 
> pls help
> thanks in advance..

Jose:

See http://www.cs.columbia.edu/sip/implementations.html for more info.

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
IN Architecture and Internet Software Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184


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From sip-admin@lists.bell-labs.com  Thu Jun 22 11:41:23 2000
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To: "'Fernando Orus'" <forus@insynergy.com>
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Subject: RE: [SIP] Sending DTMF
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Fernando:

Take a look at: draft-choudhuri-sip-info-digit-00.txt

It discusses transporting DTMF digits mid-session using the SIP INFO method
with an
alternate extensible payload description.

William Lee
williaml@nortenetworks.com

> -----Original Message-----
> From:	Fernando Orus [SMTP:forus@insynergy.com]
> Sent:	Thursday, June 22, 2000 4:21 AM
> To:	sip
> Subject:	[SIP] Sending DTMF
> 
> Hi all,
> I haven't found anything related to sending DTMFs in the documentation
> I've
> read about SIP. Are you expected to send DTMFs in band within the audio
> stream?
> Thanks in advance,
> Fernando Orus
> 
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

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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>RE: [SIP] Sending DTMF</TITLE>
</HEAD>
<BODY>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Fernando:</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Take a look at: =
draft-choudhuri-sip-info-digit-00.txt</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">It discusses =
transporting DTMF digits mid-session using the SIP INFO method with =
an</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">alternate =
extensible payload description.</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">William Lee</FONT>
<BR><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">williaml@nortenetworks.com</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">Fernando Orus [SMTP:forus@insynergy.com]</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Thursday, June 22, 2000 4:21 AM</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">sip</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">[SIP] Sending DTMF</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Hi all,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">I haven't found anything related to =
sending DTMFs in the documentation I've</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">read about SIP. Are you expected to =
send DTMFs in band within the audio</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">stream?</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Thanks in advance,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Fernando Orus</FONT>
</P>
<BR>
<BR>

<P><FONT SIZE=3D2 =
FACE=3D"Arial">_______________________________________________</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">SIP mailing list</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">SIP@lists.bell-labs.com</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://lists.bell-labs.com/mailman/listinfo/sip" =
TARGET=3D"_blank">http://lists.bell-labs.com/mailman/listinfo/sip</A></F=
ONT></U>
</P>
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From sip-admin@lists.bell-labs.com  Thu Jun 22 12:17:40 2000
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To: Jo Hornsby <jhornsby@ubiquity.net>
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I am not sure if a proxy should forward a retransmitted 200 OK to
a non-INVITE request. Request retransmission by the upstream ua/proxy
will take care of it. The first 200 OK may have already reached
the upstream guy.


Jo Hornsby wrote:
> 
> > While we are here, I would like to hear what proxy implementors think
> > about receiving multiple final respones on a single branch. Is it
> > reasonable to simply drop final responses received after the first
> > one ??
> 
> If you have state, and the response is not 2xx, then maybe so.
> 
> The reason I say this is because I would guess that if this situation
> has arisen, then it's probably due to some server responding in a
> very tardy fashion; if this is the case, then it's not so unlikely
> that any proxies along the route have dropped transaction state, and
> thus they'll be forwarded the response statelessly.
> 
> One reason against this behaviour is that I guess you could still
> be pending results on some branches when you receive a different
> final response on a branch that was marked as done.  This probably
> would tend to indicate that a downstream server is very ill, however.
> 
> Of course, if the response is to an INVITE, you really should ACK
> if it's non-2xx class...
> 
> > I guess, dropping 200 response is not good, but it is very
> > unlikely that a 200 response is received on a branch after a non-200
> > final response.
> 
> I don't see that it's any less likely than receiving any other
> different response.
> 
> > If the proxy decides to forward all 200 responses
> > upstream, regardless of the state of the branch and transaction, the
> > branch state gets kind of messy.
> 
> How so?  If the proxy still has state on the transaction, then the
> branch will either be done or it won't.  Forwarding 2xx class
> upstream in a stateless fashion doesn't really complicate this;
> you just set the boolean unconditionally. &:)
> 
> > Also, it was suggested on this list that proxies should not drop
> > final responses which do not append a tag to the To header. If a proxy
> > tries to be lenient and does this, it complicates matters even more.
> 
> I can see ambiguity arising with ACKs...is there anything else?
> 
>  - Jo.


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From sip-admin@lists.bell-labs.com  Thu Jun 22 12:37:35 2000
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> I am not sure if a proxy should forward a retransmitted 200 OK to
> a non-INVITE request. Request retransmission by the upstream ua/proxy
> will take care of it. The first 200 OK may have already reached
> the upstream guy.

Interesting point -- I'd missed that.  (Although I guess all it
implies is that there will be some extra message transmissions.)

So I guess we look to the tag to distinguish response retransmissions?
But if the tag is missing...

Oh dear.


 - Jo.



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> Brett Tate wrote:
> 
> A -- Invite (hold media) --> B
>   <-- 200 (hold media) --   B
>    --- Ack -------------------->  B
> 
> After A requests B to hold, how can A without providing media request
> that B stop holding and provide media?

I'm having a bit of trouble parsing that sentence; I think you are
asking how to take someone off hold? Simply send a re-INVITE with the
connection address non-zero once more.

> 
> I just wanted to verify that the following should work unless B
> has also pushed the hold button.
> 
> A -- Invite -------------------> B
>   <-- 200 (media) ---------   B
>    --- Ack (media) -------->  B

Assuming (media) means non-zero address, yes.

Note that hold is not necessarily bi-directional, if A sends an INVITE
to B with a zero IP address, B won't send media to A, and is "on hold",
but B is not obligated to return zero in the response. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
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From sip-admin@lists.bell-labs.com  Thu Jun 22 14:21:15 2000
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In case anyone is not aware of it, I believe that this type
of certification program is currently being considered by the
SIP Working Group in the Softswitch Consortium (which is *not*
looking for post-H.323 work!).

Dave Walker
SS8 Networks
Ottawa, Canada


Henning Schulzrinne wrote:
> 
> One of the ideas that was discussed at the Telia dinner in Stockholm was
> whether there was a need for a basic SIP certification program. There
> are obvious pluses and minuses for this, but my concern is that if this
> is a real need, I can think of a few organizations (looking for
> post-H.323 work...) that I would rather *not* see doing this.
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> 
> _______________________________________________
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If it were bi-directional, then how would you get to hear how 
wonderful company X is when someone you phoned there puts you on
hold?  In other words, MOH doesn't work very well with bi-directional
hold.  On the other hand, it'd be nice for my app to automatically 
switch over to my own MOH when I'm on hold.  But then maybe I'd miss 
important information from the far end (like "there are approximately 
20 people before you in the queue", or like "press pound or else I'm 
going to drop your call" - pound to be sent in an INFO of course).

Dave Walker
(who enjoys being on hold)


Jonathan Rosenberg wrote:
> 
[snip]
> 
> Note that hold is not necessarily bi-directional, if A sends an INVITE
> to B with a zero IP address, B won't send media to A, and is "on hold",
> but B is not obligated to return zero in the response.
> 
> -Jonathan R.
> 
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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Shail Bhatnagar wrote:
> 
> I am not sure if a proxy should forward a retransmitted 200 OK to
> a non-INVITE request. Request retransmission by the upstream ua/proxy
> will take care of it. The first 200 OK may have already reached
> the upstream guy.

No, it should not. Responses to non-INVITEs are only retransmitted upon
request retransmissions. See 10.4.1 for details.

---
Igor Slepchin


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Subject: [SIP] Announcement of Small Group Multicast (SGM) BoF
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I thought people on this list might be interested in the
"Small Group Multicast" (SGM) BoF that will be held at
the IETF in Pittsburgh.

The idea here is to develop a multicast solution that can
support very large numbers (eg tens of thousands or millions)
of relatively small multicast groups (e.g. up to 10 or 20 or 40
members) so that we can make multicast practical for
important classes of applications such as IP telephony,
videoconferencing, "collaborative" applications, networked
games etc.

I'm including the note that went out on the IETF-announce
list earlier in the week.  Note that there's a mailing list you
can subscribe to (instructions below) if you're interested
in this.

Thanks.

Rick Boivie
rhboivie@us.ibm.com
---------------------- Forwarded by Rick Boivie/Hawthorne/IBM on 06/22/2000
11:32 AM ---------------------------


"David Oran" <oran@cisco.com>@cnri.reston.va.us on 06/19/2000 02:27:49 PM

Sent by:  scoya@cnri.reston.va.us


To:   IETF-Announce: ;
cc:
Subject:  Small Group Multicast (SGM) BoF (CORRECTED)



This is a corrected version of the previous announcement.


Small Group Multicast (SGM) BoF
-----------------------------

A number of people in the Internet community have been advocating that the
IETF consider standardizing a method for handling multicast to small
groups,
where "small" is generally understood to be in the range of 3-10 or
possibly
up to 20 or so participants. The rationale for this is twofold:

1) there are alleged to be a number of very important applications which
have a small numbers of participants per group (e.g. classic audio and
video
conferencing, collaboration tools, etc.) with an extremely large number of
such groups simultaneously active on the Internet.

2) Existing IP multicast methods, including newer specialized schemes like
Source Specific Multicast (SSM) are generally thought to have scaling
problems when there are a very large nubmer of groups, because of per-group
state in all the intermediate routers.

Because of the range of possible approaches to this problem and the
potential widespread impact on portions of the IP protocol suite, router
implementation, operational practices, etc. of the Internet, the BoF is
being jointly hosted by the Routing and Internet areas, and will be chaired
by Randy Bush of the OPS area.

The BoF's goals are to:
a) assess interest in forming a Working Group on Small Group Multicast
b) produce a crisp problem statement which could be used as the
   basis for a Working Group Charter
c) reach consensus on a set of constraints on the solution space (see
below)
d) examine existing work for possible solutions (see below)

In order to have a technically sound and deployable solution, it is
particularly important the scope of changes to the Internet architecture,
protocols, host & router implementations, and operational management be
understood and hopefully minimized. Some of these possible constraints
include:

- Host API (e.g. sockets interface) changes allowed?
- Host protocol implementation changes allowed?
- IP packet header changes allowed? If so, effect on router forwarding
  speed?
- New routing protocols needed? Modifications to existing multicast
  and/or unicast routing protocols? Changes to or new group membership
  and rendezvous protocols?
- Incremental deployment possible? Impact on deployed infrastructure?
  Is the new capability worth the level of disruption, if any?
- need to re-use existing congestion control and security mechanisms?

In order to stimulate discussion in advance of the BoF, we will use an
existing mailing list set up by the folks who have been working on a set of
approaches to SGM which are collectively known as explicit multicast, or
"xcast". The mailing list is:
     xcast@public.alcatel.com
and may be subscribed to by via xcast-request@public.alcatel.com

A detailed agenda will be provided in the next couple of weeks after
initial
discussion on the mailing list.

An initial reading list for people interested in the discussion follows:

    1) "Connectionless Multicast", draft-ooms-cl-multicast-02.txt

    2) "Small Group Multicast",
       http://www.icair.org/main_projects_infrast_sgm.html
       http://www.ietf.org/internet-drafts/draft-boivie-sgm-00.txt

    3) "Multiple Destination option on IPv6", draft-imai-mdo6-01.txt

    4) "Distributed Core Multicast: a routing protocol for many small
        groups with application to mobile IP telephony"
        draft-blazevic-dcm-mobility-00.txt(Expired)

    5) "Distributed Core Multicast(DCM): a multicast routing protocol
        for many groups with few receivers", In ACM SIGCOMM Computer
        Communication Review, October 1999, Vol.29, No.5, pp.6-21

    6) "Datagram Routing for Internet Multicasting", L.  Aguilar,
        Sigcomm84, March 1984

    7) "IPv4 option for Sender Directed Multi-Destination Delivery",
        C. Graff, RFC1770

    8) "Explicit Multicast(xcast) proposed working group"
       http://www.alcatel.com/xcast




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From sip-admin@lists.bell-labs.com  Thu Jun 22 14:47:41 2000
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To: rafi <rafi@sasi.com>
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It is theoretically possible, but gateways have no useful way to
determine the on-hook/off-hook status of a phone. Presuming you exposed
some entity that could know this (for example if a 5e had an IP
interface and would accept SUBSCRIBE), then you could do it. It would
all depend on there being sufficient naming/routing information to get a
SUBSCRIBE request to that 5E. I suspect that ENUM would be helpful
here...


-Jonathan R.

> rafi wrote:
> 
> Hi,
> 
> 
>      Can I have phone user in my buddy list ?.
> Thus,  is it possible for a gateway to act as PA?
> In that case  gateway may need some support from
> PSTN world.
> 
> 
> Thank You
> Rafi Assadi H.M
> Silicon Automation Systems
> Bangalore, INDIA

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Thu Jun 22 16:56:22 2000
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Subject: Re: [SIP] Hold & then re-INVITE without media
Date: Thu, 22 Jun 2000 16:59:11 -0400
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Thanks for the response.  

A rewording of the question is presented within the following text.

----- Original Message ----- 
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
To: Brett Tate <brett@broadsoft.com>
Cc: SIPbell-labs <sip@lists.bell-labs.com>
Sent: Thursday, June 22, 2000 12:52 PM
Subject: Re: [SIP] Hold & then re-INVITE without media


> 
> 
> > Brett Tate wrote:
> > 
> > A -- Invite (hold media) --> B
> >   <-- 200 (hold media) --   B
> >    --- Ack -------------------->  B
> > 
> > After A requests B to hold, how can A without providing media request
> > that B stop holding and provide media?
> 
> I'm having a bit of trouble parsing that sentence; I think you are
> asking how to take someone off hold? Simply send a re-INVITE with the
> connection address non-zero once more.
> 
> > 
> > I just wanted to verify that the following should work unless B
> > has also pushed the hold button.
> > 
> > A -- Invite -------------------> B
> >   <-- 200 (media) ---------   B
> >    --- Ack (media) -------->  B
> 
> Assuming (media) means non-zero address, yes.

I think you already answered my question, but maybe this
will provide some clarity.  

For the following question, assume A originally 
sent the INVITE containing an SDP 
with connection address 0.0.0.0 to B; B replied with 200; 
A replied with an ACK.  Can an INVITE without SDP 
indicate to B that A is no longer "holding B" and will 
provide the complete SDP in the ACK?

A --------------- Invite (no SDP) ----------------------------> B
  <-- 200 (SDP with connection address not 0.0.0.0) ---
    --- Ack (SDP with connection address not 0.0.0.0) -->

> 
> Note that hold is not necessarily bi-directional, if A sends an INVITE
> to B with a zero IP address, B won't send media to A, and is "on hold",
> but B is not obligated to return zero in the response.
> 
> -Jonathan R.
> 
> -- 
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 



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From sip-admin@lists.bell-labs.com  Thu Jun 22 17:22:08 2000
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Actually, this is in RFC2543 also.

I don't like this, in fact. I believe method-based special handling of
headers is confusing, and I would prefer that the response always be
483, for any method. I guess the objective (for OPTIONS at least), is to
query a proxy for capabilities. But, its not really clear that this
means the same thing as a user agent. For REGISTER, its not likely to do
anything useful, except result in the request being rejected, as it
probably can't be accepted locally at some intermediate proxy.

Anyone object to having this sentence stricken?

-Jonathan R.


Neil Deason wrote:
> 
> According to 6.26 of the bis draft if a proxy receives a request
> that has a zero value in the Max-Forwards header field it MUST
> NOT forward the request. Instead, for the OPTIONS or REGISTER
> methods, it MUST respond as the final recipient. For all other
> methods, the server returns 483 (Too many hops).
> 
> How exactly should a Proxy respond as the final recipient to an
> OPTIONS?
> 
> How should a proxy respond to a REGISTER if it is not responsible
> for registrations within the domain named in the Request-URI?
> 
> In the case of other methods is sensible to always return 483?
> For example what if it was an INVITE and the Proxy knows from a
> location service a final response would have been 3XX/404.
> 
> Cheers,
> Neil.
> --
> Ubiquity Software Corporation, UK
> http://www.ubiquity.net
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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> Ajay Chitturi wrote:
> 
> In this diagram does "give up" mean the caller gave up and hung
> up the call or that the state machine gave up ?

I believe it means that the caller gives up.

> 
> Since there is another arrow from the "Calling" state showing
> "7 INVITE sent", I assume the arrow to the right of "Calling" state
> happens when the user hangs up. In this case shouldn't the state
> machine
> wait for the response to INVITE and send an ACK ?

I believe it should. This is reflected in the server state machine on
the pages following. In fact, hanging up has no effect on the
transaction at all (its termination should be sped up, since the far
side will send a final response to the INVITE).

> 
> The arrow to the right of "Call Proceeding" state could happen if
> either
> the caller hangs up or the state machine hangs up. Is the timeout for
> when
> the state machine will give up in this state specified in the draft or
> left
> to the implementation. If the user hangs up, shouldn't the state
> machine
> wait for the response to INVITE and send an ACK ?

I don't think you can usefully distinguish between the "state machine"
giving up, and the user giving up. In any case, yes, you should wait for
the response and send the ACK. Of course, from a UI perspective, the
call is hung up and that should be reflected; the transaction kind of
hangs around (much like TCP state for closed connections) to clean up.

> 
> Also section 4.2.4 states :
>    A BYE request from either called or calling party terminates any
>    pending INVITE, but the INVITE request transaction MUST be
> completed
>    with a final response.
> 
> Does this mean that the ACK also needs to be sent after getting
> the response ?

Once again, yes.


Transactions are independent entities that should always complete by
themselves.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Jo Hornsby wrote:
> 
> > The following is a paragraph in section 10.1.1 of the RFC2543bis draft
> >     When a user agent server receives a request, it checks the Call-ID
> >     against those of in-progress calls. If the Call-ID was found, it
> >     compares the tag value of To with the user’s tag and rejects
> >     the request if the two do not match. If the From header, including
> >     any tag value, matches the value for an existing call leg, the
> >     server compares the CSeq header field value. If less than or equal
> >     to the current sequence number, the request is a retransmission.
> >     Otherwise, it is a new request. If the From header does not match
> >     an existing call leg, a new call leg is created.
> > Why is just the tag value of To compared (while the whole of From
> > header is compared).
> 
> Presumably because even if UACs are generating non-unique Call-IDs,
> they are far less likely to be non-unique within the callee's space
> (one would hope!).

No; nothing like that. I actually think the complete To field should be
included in the comparison, so as to make this consistent with the
definition of a call leg (which does include To). In practice it will
never make a difference, since, as you point out, Call-ID are unique
anyway.

> 
> > Also could you please explain the scenario where the following
> > happens: "If the From header does not match an existing call leg, a
> > new call leg is created."
> 
> I guess this vaguely confirms my first paragraph.  That's a relief. &:)

No, this case is to support multiparty conferences, so that someone else
can join in on a call.... some more definition work is needed for this
as part of the call control work.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Jonathan Rosenberg wrote:
> 
> Actually, this is in RFC2543 also.
> 
> I don't like this, in fact. I believe method-based special handling of
> headers is confusing, and I would prefer that the response always be
> 483, for any method. I guess the objective (for OPTIONS at least), is
> to query a proxy for capabilities. 
             ^^^^^
Is it?  How does a proxy know that this OPTIONS request is destined for
it, and not for proxying downstream?  The RFC (and the bis updates)
state that "Proxy and redirect servers simply forward the request with-
out indicating their capabilities."  I simply assumed that a proxy will
forward the request downstream and not respond upstream with _it's_ 
capability set.

> Anyone object to having this sentence stricken?

Not me, for one.  Anything stricken is good :-)

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
IN Architecture and Internet Software Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184


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If someone else wants to join a conference, wouldn't that be a new Call-Id ?

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
>  
> No, this case is to support multiparty conferences, so that 
> someone else
> can join in on a call.... some more definition work is needed for this
> as part of the call control work.
> 
> -Jonathan R.
> -- 
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 


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From sip-admin@lists.bell-labs.com  Thu Jun 22 18:15:51 2000
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A UA could also cache the fact that there was a failure; next time it
tries to contact that host, it can continue using the backup address. I
will note that rfc2543bis does actually forbid this behavior, though:

From 1.4.2:

> A client MAY cache a successful DNS query result. A successful query is one which contained records
> in the answer, and a server was contacted at one of the addresses from the answer. When the client wishes
> to send a request to the same host, it MUST start the search as if it had just received this answer from the
> name server. The client MUST follow the procedures in RFC1035 [16] regarding DNS cache invalidation
> when the DNS time-to-live expires.

I don't remember why we made it a MUST to start the search from
scratch.... Henning?

-Jonathan R.

Sean Olson wrote:
> 
> 
> <snip>
> > Is there
> > any method/way available in SIP using which status of proxy server can be known in advance ?.
> >  This  information helps the UAC in keeping track current active list of default
> > proxy server(s).
> >
> 
> The OPTIONS method provides a very crude sort of ping for SIP.
> Combined with the Max-Forwards: header, you can ping various nodes along
> a call signalling path. But this may still not give you the information
> you desire; just because a proxy responds to an OPTIONS request, doesn't
> mean it is capable of handling calls.
> 
> > Thank You
> > Rafi Assadi H.M.
> > Silicon Automation Sysytems
> > Bangalore, INDIA
> >
> 
> --
> Sean Olson <sean.olson@ericsson.com>
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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In the original call control specification, fully distributed multiparty
conferencing used the Call-ID to identify the entire conference. For
other conferencing models, like dialup conference bridges, the call-ID
is not used, but rather the request URI. In any case, fully distributed
multiparty conferencing is still being worked out, so this may or may
not still be the case.

-Jonathan R.

Ajay Chitturi wrote:
> 
> If someone else wants to join a conference, wouldn't that be a new Call-Id ?
> 
> > -----Original Message-----
> > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> >
> > No, this case is to support multiparty conferences, so that
> > someone else
> > can join in on a call.... some more definition work is needed for this
> > as part of the call control work.
> >
> > -Jonathan R.
> > --
> > Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> > Chief Scientist                             First Floor
> > dynamicsoft                                 East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> > http://www.dynamicsoft.com
> >

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Also note that caching these results is a SHOULD; a proxy that is
beginning to run out of memory, or is under attack, can simply elect to
stop caching the results to get itself out of trouble. The caching is an
optimziation, after all.

-Jonathan R.

dmalhotra@hss.hns.com wrote:
> 
> Hi Rafi,
>                  In scenario 1, you are right that if multiple responses are
> cached then the server
> may go out of buffer space but I have a point here that all the calls do not
> appear simultaneously
> generally they have some pattern and even if it is not there then the
> implementation can put a limit
> on the number of simultaneous calls for which it will support caching. After
> that it may not support the caching and
> may release the call or whatever specific action it wants to take.
> As far as the scenario 2 is concerned,
>                 You have probably missed one point that server is going to keep
> this info maximum upto 10*T2 seconds.
> T2 being 4 seconds so worst case it is 40 seconds.
> Regards
> Dhiraj Malhotra
> 
> "rafi" <rafi@sasi.com> on 06/09/2000 12:08:53 PM
> 
> To:   sip@lists.bell-labs.com
> cc:    (bcc: Dhiraj Malhotra/HSS)
> 
> Subject:  [SIP] [SIP]: Caching of Final Response(s)
> 
> Hi,
> 
> Consider the following paragraph in RFC 2543,
> section 10.4.1
> 
>      .......  the server sends a final response, it  cannot
>  be sure the client has received the response, and thus
> SHOULD  cache the results for at least 10*T2 seconds
> to avoid having to, for example, contact the user or location
> server again upon receiving a request retransmission.
> 
>   Now consider the following scenarios.
> 
> Scenario 1 :  Consider a UAS supporting many calls
>                    (each call's media stream may terminate
>                    at different IP  address). If server caches the
>                    final response then in worst case server may be
>                    caching all the responses almost at same time. This
> caching
>                    may  cause the server run out of buffer space thereby
>                    preventing the server  from serving  new requests.
> 
> Scenario 2:  Consider a misbehaved client which is sending request in
>                     such way that each time it sends a request it challenges
> the
>                     server to send final response.  Client can quickly
>                     make the server run out of buffer space and bring down
> its
>                     operation.
> 
>   Any comment are welcome.
> 
> Thank You
> 
>  Rafi Assadi H.M.
>  Silicon Automation Systems
>  Bangalore, INDIA.
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 
> _______________________________________________
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> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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To: jdrosen@dynamicsoft.com, vkg@lucent.com
Subject: Re: [SIP] Max-Forwards + Proxies
Cc: ndeason@ubiquity.net, sip@lists.bell-labs.com
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> Jonathan Rosenberg wrote:
> > 
> > Actually, this is in RFC2543 also.
> > 
> > I don't like this, in fact. I believe method-based special handling of
> > headers is confusing, and I would prefer that the response always be
> > 483, for any method. I guess the objective (for OPTIONS at least), is
> > to query a proxy for capabilities.

>              ^^^^^
> Is it?  How does a proxy know that this OPTIONS request is destined for
> it, and not for proxying downstream?  The RFC (and the bis updates)
> state that "Proxy and redirect servers simply forward the request with-
> out indicating their capabilities."  I simply assumed that a proxy will
> forward the request downstream and not respond upstream with _it's_ 
> capability set.

This is the point of the Max-Forwards: header usage in connection with
the OPTIONS method.

> 
> > Anyone object to having this sentence stricken?
> Not me, for one.  Anything stricken is good :-)
> 

It is nice to be able to specify that an OPTIONS/REGISTER be directed
to a particular node, at least for certain diagnostic purposes. I will
probably continue to support this behavior, even if it becomes
"non-standard". But I wouldn't mind if no one else implements it.

sean <sean.olson@ericsson.com>

> - vijay
> -- 
> Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
> IN Architecture and Internet Software Group
> Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
> Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184


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From sip-admin@lists.bell-labs.com  Fri Jun 23 01:24:03 2000
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Subject: Re: [SIP] Hold & then re-INVITE without media
References: <066201bfdbee$979b73f0$3102a8c0@broadsoft.com> <39524453.6CEE67EF@dynamicsoft.com> <070a01bfdc8c$b7d1b7d0$3102a8c0@broadsoft.com>
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Brett Tate wrote:
> 
> I think you already answered my question, but maybe this
> will provide some clarity.
> 
> For the following question, assume A originally
> sent the INVITE containing an SDP
> with connection address 0.0.0.0 to B; B replied with 200;
> A replied with an ACK.  Can an INVITE without SDP
> indicate to B that A is no longer "holding B" and will
> provide the complete SDP in the ACK?
> 
> A --------------- Invite (no SDP) ----------------------------> B
>   <-- 200 (SDP with connection address not 0.0.0.0) ---
>     --- Ack (SDP with connection address not 0.0.0.0) -->

This should work. This behavior (no SDP in INVITE, but SDP in ACK) is
allowed, actually. Its main use is for the first INVITE, for interop
with H.323v1 amongst other things. However, if you can do something in a
first INVITE, it is allowed in a re-INVITE, so there you go.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Sean Olson wrote:
> 
> It is nice to be able to specify that an OPTIONS/REGISTER be directed
> to a particular node, at least for certain diagnostic purposes. I will
> probably continue to support this behavior, even if it becomes
> "non-standard". But I wouldn't mind if no one else implements it.

I'm assuming by diagnostic you refer to the ability to tell if the proxy
is alive. You can still do that with the consistent behavior I am
proposing. It works exactly like traceroute; you set max-forwards to
some number, and the proxy who gets it when its zero returns an error.
If you get no response, you know that this particular device is down.
The difference is whether the Supported, Accept, Allow headers are
there. Allow and Accept don't make sense for proxies anyway.

Just trying to simplify things... special cases are annoying.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Fri Jun 23 01:49:00 2000
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Subject: Re: [SIP] Caller receives responses
References: <000701bfdc5a$e1972690$4e34c3c1@ubiquity.co.uk> <39523BF6.7E7E648D@cisco.com> <39525B55.FD8B7F11@dynamicsoft.com>
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Igor is correct. Handling of non-INVITE is much simpler. You can (and
MUST) only ever return a single response upstream. This is in contrast
to INVITE where you can (and SHOULD/MUST (not sure which)) forward all
200 OKs you receive. Forwarding multiple 200 OK for non-INVITE doesn't
work because of the lack of ACK; reliability of responses for non-INVITE
is guaranteed by request retransmissions. So, when a request
retransmission arrives, which response do you send? No way to know, and
thus no way to guarantee reliability. For INVITE, each final response
has its own retransmissions and its own acknowledgement with ACK. As a
result, reliability for multiple 200 OK for INVITE works.

Shail writes:
> While we are here, I would like to hear what proxy implementors think
> about receiving multiple final respones on a single branch. Is it 
> reasonable to simply drop final responses received after the first
> one ?? I guess, dropping 200 response is not good, but it is very
> unlikely that a 200 response is received on a branch after a non-200
> final response. If the proxy decides to forward all 200 responses
> upstream, regardless of the state of the branch and transaction, the
> branch state gets kind of messy.

Here is the scenario where you could get a 200 OK after a non-200 OK
response. Proxy A sends INVITE to B. B forks, and then times out, never
getting any responses. So, it sends CANCEL and also sends a 408 response
upstream. Now, after that CANCEL, a 200 OK arrives. Its then forwarded
upstream also. Voila. A now gets a 408 followed by 200 OK.

As Jo has pointed out, forwarding the additional 200 OK upstream is not
hard.

-Jonathan R.


Igor Slepchin wrote:
> 
> Shail Bhatnagar wrote:
> >
> > I am not sure if a proxy should forward a retransmitted 200 OK to
> > a non-INVITE request. Request retransmission by the upstream ua/proxy
> > will take care of it. The first 200 OK may have already reached
> > the upstream guy.
> 
> No, it should not. Responses to non-INVITEs are only retransmitted upon
> request retransmissions. See 10.4.1 for details.
> 
> ---
> Igor Slepchin
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Fri Jun 23 03:49:51 2000
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Cc: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>, sip@lists.bell-labs.com
Subject: Re: [SIP] SIP URL telephone-subscriber syntax.
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Gethin Liddell wrote:
> 
> However, as far as the Tel URI rfc (2806) is concerend, the character @
> (for example) will not have to be escaped.  This, of course, will not be
> valid in the sipurl as it is used to separate the `userinfo' and
> `hostinfo' elements and will need to be escaped.
> 
> For example, if the phone-context is actually stored as:
> 
> "Context1@Context2"
> 
> then the tel url will have to look like:
> 
> tel:+12345;phone-context=%22Context1@Context2%22
> 
> however, it would NOT be valid to just enter this as a sip url as
> follows:
> 
> sip:+12345;phone-context=%22Context1@Context2%22@company.com;user=phone
> 
> for the obvious problem of the @. this would need to be translated to:
> 
> sip:+12345;phone-context=%22Context1%40Context2%22@company.com;user=phone

Hmm. I wonder if the actual translation would be:

sip:+12345;phone-context=%2522Context1%40Context2%2522@company.com;user=phone

The difference is that here, I have treated the subscriber portion of
the tel URL as an opaque entity. When placing this into the user portion
of the SIP URL, I need to escape code reserved characters AND the %
itself. Effectively, I re-encoded the string, so that the two % signs
already there are treated as normal %, and thus escape encoded as %25.

Which way to do it depends on whether you assume that the SIP entity has
knowledge of the structure of the tel URL subscriber syntax, in which
case it would know that this would have already had its % url-encoded,
and thus there is no need to redo that encoding, or whether it is
treated as a random username that happens to have some bad characters in
it.

I tend to think the right answer is probably what Gethin has suggested.
In either case we should clarify this in the spec.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Fri Jun 23 04:06:40 2000
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Subject: Re: [SIP] Static Images
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So, in the interests of coming to consensus, let me propose that:

1. inline content MAY be included, possibly using multipart. This would
obviously include MIME types like application/URI-list which are
themselves indirect references
2. we define a new header, called URI or something, which contains
indirect references to content that is to be displayed. The header is
totally optional to process at UAS. It cannot be used for content that
is critical for processing the request properly.

We also RECOMMEND that proxies use the URI header when inserting
content. A UA inserting content can use either, but indirect references
are RECOMMENDED for content over 500 bytes (or some other number). In
this way, if a UAC sends content inline using multipart, and the UAS
doesn't support it, the 415 response will indicate that multipart isn't
supported. The UAC can then resubmit the request using the URI header. 

This keeps things open and flexible, but makes concrete recommendations
to guide implementors (i.e, supporting the URI header would be your
first choice).

-Jonathan R.

Henning Schulzrinne wrote:
> 
> The problem with having proxies add body parts is (a) complexity, (b)
> that now all clients have to understand multipart, even if they have no
> interest in the information (I would guess that most current SIP clients
> don't understand multipart) and (c) that it doesn't work with signed
> requests. Headers containing a reference (http:, say) avoid these
> problems. The only drawback is that such references require http support
> in clients (which may be an issue for a very light-weight clients, but
> most of these are probably graphically challenged anyway) and requires
> that the information is available on a publically accessible web page. I
> suspect that most corporate users would have difficult uploading their
> likeness or other images to a web page, and not everyone wants to have
> their likeness framed by an advertisement from geocity or kin. These
> are, more or less, the arguments that were made the last time this topic
> came around, I believe.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Jo Hornsby wrote:
> 
> > Let's assume we have two users, which both have clients which
> > only support UDP.  They have different home proxies, and those
> > proxies are stateful, use TCP when possible and always insist on
> > being included throughout any call going through them.
> >
> > Thus we have this setup with transport:
> >
> > Client A  -------  UDP -------- Proxy A --------- TCP --------- Proxy B
> > -------- UDP -------- Client B
> >
> > User A calls user B, and a call is successfully set up.  Both proxy
> > A and proxy B have include Record-Route headers in any proxied
> > requests.
> >
> > Now let us assume that the code implementing the underlying session
> > being established has a critical bug, and that when client A and B
> > act on the agreed session description, they both crash violently.
> > How will the proxies in-between ever discover that?  After the ACK
> > is sent by client A, the proxies never hear from the clients again!
> >
> > How is this handled?
> 
> I'm not massively confident that I'm understanding where there's
> a problem; is it because you are concerned that the TCP connection
> might persist unnecessarily indefinitely?  Else I can't see where
> the "proxy protocol conversion" comes in?  I guess the solution
> that jumps out at me is to have some sort of timer on a TCP connection,
> which causes it to be closed if there's been no activity for some
> period of time.

Persistence of TCP connections is totally independent of determination
of whether the UAC and UAS are still in the session set up with the
INVITE. Session timer is used exclusively for the latter; how long to
keep a TCP connection open for is a matter of policy. However, I have a
draft pending shortly which discusses some issues with TCP connection
management.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Fri Jun 23 04:14:01 2000
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Yes, Shail is correct.

-Jonathan R.

Shail Bhatnagar wrote:
> 
> Vijay, I think it includes tags, since a brand new re-INVITE or BYE
> may have a tag in the To header.
> 
> "Vijay K. Gurbani" wrote:
> >
> > Folks:
> >
> > In the June 9th bis draft, Section 12.3.4 (Stateful Proxy: Receiving
> > Requests) states:
> >
> >    When a stateful proxy receives a request, it checks the To,
> >    From (including tags), Call-ID and CSeq against existing
> >    request records.
> >
> > I presume the To check does NOT include tags.  Can someone corroborate
> > this, please.
> >
> > Thanks,
> >
> > - vijay
> > --
> > Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
> > IN Architecture and Internet Software Group
> > Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
> > Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Fri Jun 23 04:20:37 2000
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From: "Alexander Agranov" <sagranov@comgates.co.il>
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Cc: <sip@lists.bell-labs.com>
Date: Fri, 23 Jun 2000 11:21:02 +0200
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Subject: [SIP] H323 Fast Connect to SIP interworking
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Session 5.2.1 in draft-singh-sip-h323-01, that describes interworking
of H323 Fast Connect with SIP says that there is "one-to-one mapping
between H.323 and SIP call establishment messages".

Nevertheless let's have a look at the scenario when H.323 Fast Start
message contains set of more than one alternative channels:

H.323 terminal                GW                         SIP UA
128.59.21.152                                         128.59.19.194
|                             |                            |
|            SETUP            |                            |
|---------------------------->|                            |
| destination:hgs@cs.columbia.edu                          |
| fastStart={g711Ulaw,Tx},    |                            |
|  {{g711Ulaw,Rx,128.59.21.152:10000},                     |
|   {g711Alaw,Rx,128.59.21.152:10000}}                     |
|                             |                            |
|                             |          INVITE            |
|                             |===========================>|
|                             | To:hgs@cs.columbia.edu     |
|                             | c=IN IP4 128.59.21.152     |
|                             | m=audio 10000 RTP/AVP 0 8  |
|                             |                            |
|                             |          200 OK            |
|                             |<===========================|
|                             | c=IN IP4 128.59.19.194     |
|                             | m=audio 8000 RTP/AVP 0 8   |
                           ????????

In this case (according to 8.1.1) IWF should generate INVITE
SIP request with SDP that contains multiple payloads.

Now it's my understanding of RFC2543 that SIP UA may respond
to such INVITE with "200 OK" that also contains SDP with
multiple payloads.

But we can't acknowledge both FastConnect alternative channels,
can we? 

So GW will need to acknowledge only one of the channels and 
issue reINVITE to SIP UA (like in H323v1), right?

This situation is applicable even when FastConnect contains
only one channel, since RFC2543 states that "200 OK" message's
"list of codecs may or may not be a subset of" INVITE message
codecs.

Please correct me if I'm wrong.

Best regards,
                Alex Agranov
---
ComGates Ltd.
Tel: +972 (9) 9500-404, Ext: 228
Mobile: +972 (54) 928435
Fax: +972 (9) 9500-385
E-mail: sagranov@comgates.co.il





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From sip-admin@lists.bell-labs.com  Fri Jun 23 04:31:40 2000
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"Vijay K. Gurbani" wrote:
> 
> Jonathan Rosenberg wrote:
> >
> > Actually, this is in RFC2543 also.
> >
> > I don't like this, in fact. I believe method-based special handling of
> > headers is confusing, and I would prefer that the response always be
> > 483, for any method. I guess the objective (for OPTIONS at least), is
> > to query a proxy for capabilities.
>              ^^^^^
> Is it?  How does a proxy know that this OPTIONS request is destined for
> it, and not for proxying downstream?  

Because the case we were discussing is where the Max-Forwards
header field of the OPTIONS request contains a zero.

> The RFC (and the bis updates)
> state that "Proxy and redirect servers simply forward the request with-
> out indicating their capabilities."  I simply assumed that a proxy will
> forward the request downstream and not respond upstream with _it's_
> capability set.

Yes, this is the normal behaviour when Max-Forwards doesn't come
in to play.
Cheers,
Neil.
-- 
Ubiquity Software Corporation, UK          
http://www.ubiquity.net

> > Anyone object to having this sentence stricken?
> 
> Not me, for one.  Anything stricken is good :-)


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From sip-admin@lists.bell-labs.com  Fri Jun 23 04:37:57 2000
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Folks,

I've just submitted a draft on using SCTP for SIP transport. It defines
how to do it (its very easy) and some of the benefits. Its mostly to
encourage implementation experience and experimentation; my own belief
is that much experimentation is needed before we can make concrete
statements about its applicability to SIP.

Comments welcome, of course, particularly from the SCTP experts.

Until it appears in the archives, you can grab a copy from:
http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-sip-sctp-00.txt

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Fri Jun 23 04:52:14 2000
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Jonathan Rosenberg wrote:
> 
> Sean Olson wrote:
> >
> > It is nice to be able to specify that an OPTIONS/REGISTER be directed
> > to a particular node, at least for certain diagnostic purposes. I will
> > probably continue to support this behavior, even if it becomes
> > "non-standard". But I wouldn't mind if no one else implements it.
> 
> I'm assuming by diagnostic you refer to the ability to tell if the proxy
> is alive. You can still do that with the consistent behavior I am
> proposing. It works exactly like traceroute; you set max-forwards to
> some number, and the proxy who gets it when its zero returns an error.
> If you get no response, you know that this particular device is down.
> The difference is whether the Supported, Accept, Allow headers are
> there. Allow and Accept don't make sense for proxies anyway.

Also the Supported header which might be useful SHOULD
be included in the response according to 6.40.

> Just trying to simplify things... special cases are annoying.

I agree that simplification would be good in this area. There
doesn't seem to be any loss in saying that the standard response 
from a proxy/gateway recipient to any request containing 
Max-Forwards: 0 is 483.

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


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From: "Jo Hornsby" <jhornsby@ubiquity.net>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "Igor Slepchin" <islepchin@dynamicsoft.com>
Cc: "Shail Bhatnagar" <shbhatna@cisco.com>, "A Srinath" <srinath@sasi.com>,
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Subject: RE: [SIP] Caller receives responses
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> Igor is correct. Handling of non-INVITE is much simpler. You can (and
> MUST) only ever return a single response upstream.

Is this explicitly documented anywhere -- have I missed it?  The
only forking documentation I'm aware of is 12.4, and that's specific
to INVITEs.

If you fork REGISTERs (I'm not sure why you'd do this -- perhaps
towards some sort of redundancy?), are there any potential
problems with only returning one 200?  The only other request
I can possibly see problems with (because you might get a response
from a UA that ultimately you wouldn't have been interested in)
is OPTIONS.  (Of course, unknown methods are forwarded like
OPTIONS.)

This has actually raised another point.  Upon re-reading of 12.4,
I note that 6xx MUST be forwarded if a 2xx has not been seen.  Is
this true regardless of whether-or-not a different 6xx has already
be forwarded?


 - Jo.



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From sip-admin@lists.bell-labs.com  Fri Jun 23 06:06:31 2000
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Subject: [SIP] Coco/R-MGCP/SIP
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Hi,

My name is Catalin and I work for ERICSSON Romania.

Now I am working to implement a parser for the MGCP protocol (and later for the SIP protocol).
I am using the Coco/R parser for this, but I am actually a beginner in doing this kind of things.
The Coco/R is a nice tool, but sometimes I need to ask questions, to get some hints.

Is anyone having experience in using the Coco/R parser for MGCP/SIP protocols ?

Thank you.



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From sip-admin@lists.bell-labs.com  Fri Jun 23 06:07:20 2000
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Subject: Re: [SIP] Announcement of SIP Forum
References: <2160ABC55998D311821900508B2C14BB269D71@webmail.speedventures.com> <395139FD.45921C74@cs.columbia.edu> <3952587E.3930A139@ss8networks.com>
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Dave Walker wrote:
> 
> In case anyone is not aware of it, I believe that this type
> of certification program is currently being considered by the
> SIP Working Group in the Softswitch Consortium (which is *not*
> looking for post-H.323 work!).

Personally I wouldn't be in favour of this 
consortium, or any other membership charging 
forum, seeking such a role. The problems of 
such groups having influence over protocols 
has been demonstrated by the recent criticisms 
of the WAP Forum. Let's keep the focus for SIP 
within the truly open environment of the IETF.

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

> Henning Schulzrinne wrote:
> >
> > One of the ideas that was discussed at the Telia dinner in Stockholm was
> > whether there was a need for a basic SIP certification program. There
> > are obvious pluses and minuses for this, but my concern is that if this
> > is a real need, I can think of a few organizations (looking for
> > post-H.323 work...) that I would rather *not* see doing this.
> > --
> > Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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From sip-admin@lists.bell-labs.com  Fri Jun 23 08:00:33 2000
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In-reply-to: Your message of "Thu, 22 Jun 2000 17:58:51 EDT."
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Date: Fri, 23 Jun 2000 07:59:57 -0400
From: Gopal <gopal@research.att.com>
Subject: [SIP] >   12. Re: Questions on section 10.1.1 of RFC2543bis (Jonathan Rosenberg)
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> Return-Path: <owner-sip@lists.bell-labs.com>
> Delivered-To: sip@lists.bell-labs.com
> Message-ID: <39528C60.7F0BE88D@dynamicsoft.com>
> Date: Thu, 22 Jun 2000 18:00:00 -0400
> From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
> Organization: dynamicsoft
> MIME-Version: 1.0
> To: Jo Hornsby <jhornsby@ubiquity.net>
> Cc: Ajay Chitturi <ajaych@Exchange.Microsoft.com>,
>       sip@lists.bell-labs.com
> Subject: Re: [SIP] Questions on section 10.1.1 of RFC2543bis
> References: <000e01bfda91$07b6d940$4e34c3c1@ubiquity.co.uk>
> Content-Type: text/plain; charset=iso-8859-1
> Content-Transfer-Encoding: 8bit
> 
> 
> 
> Jo Hornsby wrote:
> > 
> > > The following is a paragraph in section 10.1.1 of the RFC2543bis draft
> > >     When a user agent server receives a request, it checks the Call-ID
> > >     against those of in-progress calls. If the Call-ID was found, it
> > >     compares the tag value of To with the users tag and rejects
> > >     the request if the two do not match. If the From header, including
--- snipped 4 lines ----
> > > Why is just the tag value of To compared (while the whole of From
> > > header is compared).

> No; nothing like that. I actually think the complete To field should be
> included in the comparison, so as to make this consistent with the
> definition of a call leg (which does include To). In practice it will

There is another potential problem though with comparing tag value of To.
Consider the foll. scenario when a UAC talks directly to a UAS  (no proxy)

	UAC				UAS
INVITE (no tag value for To)----------->Generate new Tag value for To
		        (lost)X <------ 100 Trying (MAY include tag in To)
		        (lost)X <------ 180 Ringing (MAY include tag in To)

(timeout. Retransmit INVITE)
INVITE ((no tag value for To)---------->Rejected since Tag value of To does not
					match

10.1.1 of RFC2543bis I think (wrongfully) suggests the above behavior. Ideally,
re-transmission of the INVITE should trigger the 180 response from the UAS. I
think the tag value of To should be compared only if the To field in the
incoming message has a tag. Sec. 6.23 actually states that tag values of 2
URIs are compared only if both URIs have tags. How about changing the sentence to

"If the Call-ID was found, it compares the tag value of To (if present)
 with the users tag and rejects the request if the two do not match."

-gopal


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From sip-admin@lists.bell-labs.com  Fri Jun 23 08:38:57 2000
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Hi,

Section 15.1.1 of the rfc2543bis draft is missing information, and a BNF entry with respect to pgp-pubalgolrithm.

Presumably, pgp-pubalgolrithm should be included in the BNF for pgp-params in section 15.1.1

Also some text about the parameter is required, e.g. what is the default value of pgp-pubalgolrithm if it is omitted from WWW-Authenticate header.
Is it assumed to be RSA?

Hope that helps
Regards
--
Phil Hoffer                         Ubiquity Software Corporation
Software Developer                  Ubiquity House
Tel: +44 (0) 1633-765600            Langstone Park
Fax: +44 (0) 1633-765601            Newport   NP18 2LH
URL: http://www.ubiquity.net
mailto:phoffer@ubiquity.net


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From sip-admin@lists.bell-labs.com  Fri Jun 23 08:57:40 2000
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Subject: Re: [SIP] Announcement of SIP Forum
References: <2160ABC55998D311821900508B2C14BB269D71@webmail.speedventures.com> <395139FD.45921C74@cs.columbia.edu> <3952587E.3930A139@ss8networks.com> <39533662.AEAED0C8@ubiquity.net>
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Well, I don't know about the WAP Forum, but I would think that
a certification program should certainly be open to non-members
if it's run by a forum that charges for membership.  I'd be
surprised if a certification program could be run without 
costs that have to be covered somehow (volunteer organizations
aren't too reliable in the long term).  And remember, the reason
people will want such a certification for their products is to
make money.  And it takes money to make money.  (or it used to!)

I also note that SIP Forum indicates that it will be charging
a fee to organizations (didn't see how much), and while it's
free to individuals, it is typically organizations that have 
more of a motivation for acquiring certification.

Regards,

Dave Walker
SS8 Networks
Ottawa, Canada

Neil Deason wrote:
> 
> Dave Walker wrote:
> >
> > In case anyone is not aware of it, I believe that this type
> > of certification program is currently being considered by the
> > SIP Working Group in the Softswitch Consortium (which is *not*
> > looking for post-H.323 work!).
> 
> Personally I wouldn't be in favour of this
> consortium, or any other membership charging
> forum, seeking such a role. The problems of
> such groups having influence over protocols
> has been demonstrated by the recent criticisms
> of the WAP Forum. Let's keep the focus for SIP
> within the truly open environment of the IETF.
> 
> Cheers,
> Neil.
> --
> Ubiquity Software Corporation, UK
> http://www.ubiquity.net
>


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From sip-admin@lists.bell-labs.com  Fri Jun 23 09:37:46 2000
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This sounds find (I'll incorporate it), but I do wonder if one wouldn't
want to be more specific in the purpose of the header, rather than just
"URI". For the 'visual caller id', something like '*-Info' (with a
suitable value of *, TBD) seems more evocative. I don't want to follow
the INFO tradition of dumping everything into one header. This may be
relevant, for example, as to whether the UA uses this information to
update some kind of address book.

Jonathan Rosenberg wrote:
> 
> So, in the interests of coming to consensus, let me propose that:
> 
> 1. inline content MAY be included, possibly using multipart. This would
> obviously include MIME types like application/URI-list which are
> themselves indirect references
> 2. we define a new header, called URI or something, which contains
> indirect references to content that is to be displayed. The header is
> totally optional to process at UAS. It cannot be used for content that
> is critical for processing the request properly.
> 
> We also RECOMMEND that proxies use the URI header when inserting
> content. A UA inserting content can use either, but indirect references
> are RECOMMENDED for content over 500 bytes (or some other number). In
> this way, if a UAC sends content inline using multipart, and the UAS
> doesn't support it, the 415 response will indicate that multipart isn't
> supported. The UAC can then resubmit the request using the URI header.
> 
> This keeps things open and flexible, but makes concrete recommendations
> to guide implementors (i.e, supporting the URI header would be your
> first choice).
> 
> -Jonathan R.
> 
> Henning Schulzrinne wrote:
> >
> > The problem with having proxies add body parts is (a) complexity, (b)
> > that now all clients have to understand multipart, even if they have no
> > interest in the information (I would guess that most current SIP clients
> > don't understand multipart) and (c) that it doesn't work with signed
> > requests. Headers containing a reference (http:, say) avoid these
> > problems. The only drawback is that such references require http support
> > in clients (which may be an issue for a very light-weight clients, but
> > most of these are probably graphically challenged anyway) and requires
> > that the information is available on a publically accessible web page. I
> > suspect that most corporate users would have difficult uploading their
> > likeness or other images to a web page, and not everyone wants to have
> > their likeness framed by an advertisement from geocity or kin. These
> > are, more or less, the arguments that were made the last time this topic
> > came around, I believe.
> 
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com



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Jo Hornsby wrote:
> 
> > Igor is correct. Handling of non-INVITE is much simpler. You can (and
> > MUST) only ever return a single response upstream.
> 
> Is this explicitly documented anywhere -- have I missed it?  The
> only forking documentation I'm aware of is 12.4, and that's specific
> to INVITEs.

Hmmm; I somehow thought the code handled this case as well, but perhaps
not.

The behavior is more or less identical, excepting you forward only one
200 OK response upstream if more than one come.

> 
> If you fork REGISTERs (I'm not sure why you'd do this -- perhaps
> towards some sort of redundancy?), are there any potential
> problems with only returning one 200?

If all registrars receiving the REGISTER are identical (in the sense
that they maintain a shared database of registrations), receiving only
one 200 OK is just fine, and the right thing. If they were different, it
would anyway be confusing... if the responses contained different
Contacts, which ones are actually in place in which servers?


> This has actually raised another point.  Upon re-reading of 12.4,
> I note that 6xx MUST be forwarded if a 2xx has not been seen.  Is
> this true regardless of whether-or-not a different 6xx has already
> be forwarded?

No; you forward a single 600 only. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From: Kundan Singh <kns10@cs.columbia.edu>
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One method is to send the updated SDP in ACK to SIP UA.

After receiving the 200 OK from SIP UA, the IWF
recalculates the capability sets and operating modes in
both directions and sends ACK to SIP UA.

IWF may send a reINVITE to SIP UA to inform about the 
updated modes, instead of changing the session description in ACK.

As far as call establishment is concerned, there can be
a one-to-one mapping between SIP and H.323 messages with fast
connect (in simple case).

Regards

Kundan

> Session 5.2.1 in draft-singh-sip-h323-01, that describes interworking
> of H323 Fast Connect with SIP says that there is "one-to-one mapping
> between H.323 and SIP call establishment messages".
> 
> Nevertheless let's have a look at the scenario when H.323 Fast Start
> message contains set of more than one alternative channels:
> 
> H.323 terminal                GW                         SIP UA
> 128.59.21.152                                         128.59.19.194
> |                             |                            |
> |            SETUP            |                            |
> |---------------------------->|                            |
> | destination:hgs@cs.columbia.edu                          |
> | fastStart={g711Ulaw,Tx},    |                            |
> |  {{g711Ulaw,Rx,128.59.21.152:10000},                     |
> |   {g711Alaw,Rx,128.59.21.152:10000}}                     |
> |                             |                            |
> |                             |          INVITE            |
> |                             |===========================>|
> |                             | To:hgs@cs.columbia.edu     |
> |                             | c=IN IP4 128.59.21.152     |
> |                             | m=audio 10000 RTP/AVP 0 8  |
> |                             |                            |
> |                             |          200 OK            |
> |                             |<===========================|
> |                             | c=IN IP4 128.59.19.194     |
> |                             | m=audio 8000 RTP/AVP 0 8   |
>                            ????????
> 
> In this case (according to 8.1.1) IWF should generate INVITE
> SIP request with SDP that contains multiple payloads.
> 
> Now it's my understanding of RFC2543 that SIP UA may respond
> to such INVITE with "200 OK" that also contains SDP with
> multiple payloads.
> 
> But we can't acknowledge both FastConnect alternative channels,
> can we? 
> 
> So GW will need to acknowledge only one of the channels and 
> issue reINVITE to SIP UA (like in H323v1), right?
> 
> This situation is applicable even when FastConnect contains
> only one channel, since RFC2543 states that "200 OK" message's
> "list of codecs may or may not be a subset of" INVITE message
> codecs.
> 
> Please correct me if I'm wrong.
> 
> Best regards,
>                 Alex Agranov
> ---
> ComGates Ltd.
> Tel: +972 (9) 9500-404, Ext: 228
> Mobile: +972 (54) 928435
> Fax: +972 (9) 9500-385
> E-mail: sagranov@comgates.co.il
> 
> 
> 
> 
> ------------------------------------------------------------------------
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From sip-admin@lists.bell-labs.com  Fri Jun 23 10:46:10 2000
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Subject: Re: [SIP] How to ping Prxoy Server ?
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The paragraph you quote refers to caching *DNS* query results, not
whether the host was contacted or not. The idea here is just to make
sure that SRV load balancing is observed. I don't think the text says
anything about caching addresses, but it does have the potential
drawback of bypassing load balancing. Since caching is likely only to be
useful for large servers with lots of traffic, this could be a problem.
You don't want AOL to always go to the same Earthlink server.

Jonathan Rosenberg wrote:
> 
> A UA could also cache the fact that there was a failure; next time it
> tries to contact that host, it can continue using the backup address. I
> will note that rfc2543bis does actually forbid this behavior, though:
> 
> >From 1.4.2:
> 
> > A client MAY cache a successful DNS query result. A successful query is one which contained records
> > in the answer, and a server was contacted at one of the addresses from the answer. When the client wishes
> > to send a request to the same host, it MUST start the search as if it had just received this answer from the
> > name server. The client MUST follow the procedures in RFC1035 [16] regarding DNS cache invalidation
> > when the DNS time-to-live expires.
> 
> I don't remember why we made it a MUST to start the search from
> scratch.... Henning?
> 
> -Jonathan R.
> 
> Sean Olson wrote:
> >
> >
> > <snip>
> > > Is there
> > > any method/way available in SIP using which status of proxy server can be known in advance ?.
> > >  This  information helps the UAC in keeping track current active list of default
> > > proxy server(s).
> > >
> >
> > The OPTIONS method provides a very crude sort of ping for SIP.
> > Combined with the Max-Forwards: header, you can ping various nodes along
> > a call signalling path. But this may still not give you the information
> > you desire; just because a proxy responds to an OPTIONS request, doesn't
> > mean it is capable of handling calls.
> >
> > > Thank You
> > > Rafi Assadi H.M.
> > > Silicon Automation Sysytems
> > > Bangalore, INDIA
> > >
> >
> > --
> > Sean Olson <sean.olson@ericsson.com>
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> 
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip


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From sip-admin@lists.bell-labs.com  Fri Jun 23 10:52:23 2000
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From: "rafi" <rafi@sasi.com>
To: <sip@lists.bell-labs.com>
Date: Fri, 23 Jun 2000 20:19:47 +0530
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Subject: [SIP] Is SIP Transperent in this case ?
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Hi,


   Consider a call setup involving PSTN-IP-PSTN  network.
How much (SIP is used in IP network -:)  )
transparent SIP can be in this case ?.If features like
calling party presentation  restriction  are desired then how=20
can SIP handle/carry  these  kind of  information?


  =20
Thank You
Rafi Assadi H.M.
Silicon Automation Systems, Bangalore




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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2614.3500" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi,</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp; Consider a call setup =
involving=20
PSTN-IP-PSTN&nbsp; network.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>How much (<FONT face=3DArial =
size=3D2>SIP is used in IP=20
network -:)  )</FONT></FONT></DIV>
<DIV><FONT face=3DArial size=3D2>transparent SIP can be in this case =
?.If features=20
like</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>calling party presentation&nbsp; =
restriction&nbsp;=20
</FONT><FONT face=3DArial size=3D2>are desired then how </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>can SIP handle/carry  these&nbsp; kind =
of&nbsp;=20
information?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp; </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Thank You</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Rafi Assadi H.M.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Silicon Automation Systems, =
Bangalore</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

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From sip-admin@lists.bell-labs.com  Fri Jun 23 10:54:24 2000
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        Sean Olson <eussean@exu.ericsson.se>, vkg@lucent.com,
        sip@lists.bell-labs.com
Subject: Re: [SIP] Max-Forwards + Proxies
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Neil Deason wrote:

> I agree that simplification would be good in this area. There
> doesn't seem to be any loss in saying that the standard response
> from a proxy/gateway recipient to any request containing
> Max-Forwards: 0 is 483.


I think we still need 2 special cases here 
    * A REGISTER destined for a proxy may have a Max-Forwards of 0
      (registrar/proxy should still process it )
    * A CANCEL for an existing call leg is allowed to have Max-Forwards
      value of 0

Shail


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Sean Olson wrote:
       ^^^^^
> > Is it?  How does a proxy know that this OPTIONS request is destined 
> > for it, and not for proxying downstream?  The RFC (and the bis 
> > updates) state that "Proxy and redirect servers simply forward the 
> > request without indicating their capabilities."  I simply assumed 
> > that a proxy will forward the request downstream and not respond 
> > upstream with _it's_ capability set.
> 
> This is the point of the Max-Forwards: header usage in connection with
> the OPTIONS method.

Oh, okay.  Kind of like the traceroute command with the TTL being 1
greater then the previous packet.  

I agree with Jonathan's belief that method-based special handling of
headers is confusing.  I had to re-read the Max-Forwards section of
the RFC before it sunk in that the semantics the authors wanted to
impose if (Max-Forwards == 0 && Request == OPTIONS) was to have the 
current proxy respond with _it's_ capabilities.

> It is nice to be able to specify that an OPTIONS/REGISTER be directed
> to a particular node, at least for certain diagnostic purposes. I will
> probably continue to support this behavior, even if it becomes
> "non-standard". But I wouldn't mind if no one else implements it.

I'm not sure if there is a better way to achieve this.  I don't think
having a new request header will help in any way.

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
IN Architecture and Internet Software Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184


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Can anybody point me at a draft that discusses call feature processing in SIP?
In particular, I'm interested in anything that separates feature
processing from initial connection management. Thanx.

		jak



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From sip-admin@lists.bell-labs.com  Fri Jun 23 11:36:35 2000
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Not everything converts to SIP header equivalents; however, SIP-T allows
you to tunnel everything that doesn;t convery, resulting in complete
transparency. See:

http://www.softarmor.com/sipwg/teams/sipt/index.html

-Jonathan R.

> rafi wrote:
> 
> Hi,
> 
> 
>    Consider a call setup involving PSTN-IP-PSTN  network.
> How much (SIP is used in IP network -:) )
> transparent SIP can be in this case ?.If features like
> calling party presentation  restriction  are desired then how
> can SIP handle/carry these  kind of  information?
> 
> 
> 
> Thank You
> Rafi Assadi H.M.
> Silicon Automation Systems, Bangalore
> 
> 
> 

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Shail Bhatnagar wrote:
> 
> Neil Deason wrote:
> 
> > I agree that simplification would be good in this area. There
> > doesn't seem to be any loss in saying that the standard response
> > from a proxy/gateway recipient to any request containing
> > Max-Forwards: 0 is 483.
> 
> I think we still need 2 special cases here
>     * A REGISTER destined for a proxy may have a Max-Forwards of 0
>       (registrar/proxy should still process it )

Why? I really don't like method specific processing of Max-Forwards; I
think this will introduce oddball effects, where the topology of the
proxies may effect whether registrations can placed in a server or not. 

>     * A CANCEL for an existing call leg is allowed to have Max-Forwards
>       value of 0

Also, why?

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Fri Jun 23 11:43:01 2000
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> rafi wrote:
> 
> Hi,
> 
> 
>    Consider a call setup involving PSTN-IP-PSTN  network.
> How much (SIP is used in IP network -:) )
> transparent SIP can be in this case ?.If features like
> calling party presentation  restriction  are desired then how
> can SIP handle/carry these  kind of  information?

Rafi:

Some thoughts on using SIP for PSTN-like service delivery are outlined 
in the following I-Ds:

Title: Accessing IN services from SIP networks
URL: 
http://search.ietf.org/internet-drafts/draft-gurbani-iptel-sip-to-in-02.txt

Title: SIP enabled IN Services - an implementation report
URL:
http://search.ietf.org/internet-drafts/draft-gurbani-iptel-sip-to-in-02.txt

Title: Frameworks and Requirements for Internet Intelligent Networks (IIN)
URL: 
http://search.ietf.org/internet-drafts/draft-lslutsman-sip-iin-framework-00.txt

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
IN Architecture and Internet Software Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184


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From sip-admin@lists.bell-labs.com  Fri Jun 23 11:46:19 2000
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Henning Schulzrinne wrote:
> 
> The paragraph you quote refers to caching *DNS* query results, not
> whether the host was contacted or not. 

The quote indirectly says you shouldn't cache whether the host was
contacted or not; it says you should start the search afresh when
contacting the same host. To me, this means you are not allowed to
remember that you previously reached the host at address X.


The idea here is just to make
> sure that SRV load balancing is observed. I don't think the text says
> anything about caching addresses, but it does have the potential
> drawback of bypassing load balancing. Since caching is likely only to be
> useful for large servers with lots of traffic, this could be a problem.
> You don't want AOL to always go to the same Earthlink server.

The cache would need to honor DNS TTLs, as the paragraph says, so it
would eventually time out. Not caching means you may increase call setup
times waiting to fail against servers you have been failing against for
the last hour...

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Jonathan Rosenberg wrote:
> 
> Shail Bhatnagar wrote:
> >
> > Neil Deason wrote:
> >
> > > I agree that simplification would be good in this area. There
> > > doesn't seem to be any loss in saying that the standard response
> > > from a proxy/gateway recipient to any request containing
> > > Max-Forwards: 0 is 483.
> >
> > I think we still need 2 special cases here
> >     * A REGISTER destined for a proxy may have a Max-Forwards of 0
> >       (registrar/proxy should still process it )
> 
> Why? I really don't like method specific processing of Max-Forwards; I
> think this will introduce oddball effects, where the topology of the
> proxies may effect whether registrations can placed in a server or not.

I agree. If you make an exception for REGISTER though why not
INVITE? If a proxy receives an INVITE and it knows from it's
location service that the response would be a 3XX redirect then
why not return this.

> >     * A CANCEL for an existing call leg is allowed to have Max-Forwards
> >       value of 0
> 
> Also, why?

Yeah, this seems like allowing an unnecessarily odd way of doing
a CANCEL. For me the simplicity of 483 for all methods, which
allows the apparent fault diagnosis of the Max-Forwards tracing,
still wins out.

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


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From sip-admin@lists.bell-labs.com  Fri Jun 23 11:52:36 2000
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Subject: Re: [SIP] Max-Forwards + Proxies
References: <200006222312.SAA19957@b04a45.exu.ericsson.se> <3952F3E5.982D5382@dynamicsoft.com> <39532457.C5EF0266@ubiquity.net> <395379EA.1D187EED@cisco.com> <3953847A.27A61A4B@dynamicsoft.com>
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I don't have a problem with this simplification - but the spec says
an incoming request with Max-Forwards of 0 MUST NOT be forwarded.
It does not say that the request cannot be locally processed and a 
response returned. Since REGISTER (for the registrar) and CANCEL
will be absorbed by the proxy, I thought it is okay to process these
requests. 
The relevant statement in the spec should be appropriately changed -

request MUST NOT be forward and MUST NOT be processed.

Thanks,
Shail

Jonathan Rosenberg wrote:
> 
> Shail Bhatnagar wrote:
> >
> > Neil Deason wrote:
> >
> > > I agree that simplification would be good in this area. There
> > > doesn't seem to be any loss in saying that the standard response
> > > from a proxy/gateway recipient to any request containing
> > > Max-Forwards: 0 is 483.
> >
> > I think we still need 2 special cases here
> >     * A REGISTER destined for a proxy may have a Max-Forwards of 0
> >       (registrar/proxy should still process it )
> 
> Why? I really don't like method specific processing of Max-Forwards; I
> think this will introduce oddball effects, where the topology of the
> proxies may effect whether registrations can placed in a server or not.
> 
> >     * A CANCEL for an existing call leg is allowed to have Max-Forwards
> >       value of 0
> 
> Also, why?
> 
> -Jonathan R.


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> 
> If you fork REGISTERs (I'm not sure why you'd do this -- perhaps
> towards some sort of redundancy?), are there any potential
> problems with only returning one 200?  The only other request
> I can possibly see problems with (because you might get a response
> from a UA that ultimately you wouldn't have been interested in)
> is OPTIONS.  (Of course, unknown methods are forwarded like
> OPTIONS.)


I can think of a couple of good applications for forking REGISTER and other
non-INVITE requests where you definitely DO care about receiving
ALL of the 200 class responses. If a UA already has to deal with
multiple 2xx response for an INVITE, is it really so different
to do this for non-INVITE requests as well?

sean


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From sip-admin@lists.bell-labs.com  Fri Jun 23 12:12:19 2000
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To: ndeason@ubiquity.net, drwalker@ss8networks.com
Subject: Re: [SIP] Announcement of SIP Forum
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Dave Walker wrote:
> Well, I don't know about the WAP Forum, but I would think that
> a certification program should certainly be open to non-members
> if it's run by a forum that charges for membership.  I'd be
> surprised if a certification program could be run without 
> costs that have to be covered somehow (volunteer organizations
> aren't too reliable in the long term).  

Oops. I think you have a typo in there. There a number of volunteer
organizations in the OpenSource community that have been very 
reliable, even in the long term. I do agree, however, that it is
reasonable for an organization to charge a reasonable fee for
certification (provided that don't make a business out of it)

> And remember, the reason
> people will want such a certification for their products is to
> make money.  And it takes money to make money.  (or it used to!)
>

Many of the best implementations of IETF protocols were not done
for profit and were done with little or no commercial support.
Is there a reason for discouraging research or non-profit implementations?
 
> I also note that SIP Forum indicates that it will be charging
> a fee to organizations (didn't see how much), and while it's
> free to individuals, it is typically organizations that have 
> more of a motivation for acquiring certification.
>
True.

my two cents
sean


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From sip-admin@lists.bell-labs.com  Fri Jun 23 13:00:21 2000
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From: Neil Deason <ndeason@ubiquity.net>
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Sean Olson wrote:
> 
> Dave Walker wrote:
> > Well, I don't know about the WAP Forum, but I would think that
> > a certification program should certainly be open to non-members
> > if it's run by a forum that charges for membership.  I'd be
> > surprised if a certification program could be run without
> > costs that have to be covered somehow (volunteer organizations
> > aren't too reliable in the long term).
> 
> Oops. I think you have a typo in there. There a number of volunteer
> organizations in the OpenSource community that have been very
> reliable, even in the long term. 

Indeed - the IETF is made up of volunteers.

> I do agree, however, that it is
> reasonable for an organization to charge a reasonable fee for
> certification (provided that don't make a business out of it)

Yes, this wasn't my major concern. That is that any 
closed organisation (a membership oriented 
group is by definition closed to non-members, be 
it because they can't afford the fees or even the 
governing body reserves the right to deny membership 
to them) might be able to decide what is certified 
as being the standard for an open protocol, such as SIP.

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


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From sip-admin@lists.bell-labs.com  Fri Jun 23 13:07:40 2000
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From: "Culpepper, Bert" <bert.culpepper@intervoice-brite.com>
To: rafi <rafi@sasi.com>, sip@lists.bell-labs.com
Subject: RE: [SIP] Is SIP Transperent in this case ?
Date: Fri, 23 Jun 2000 12:03:59 -0500
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Using SIP in the network architecture you describe requires some
"extensions" to the protocol.  There is a solution to the situation in your
example though.  That is to use the SIP INFO method to carry ISUP messages.
See http://www.ietf.org/internet-drafts/draft-ietf-sip-info-method-03.txt
and http://www.ietf.org/internet-drafts/draft-ietf-sip-isup-mime-02.txt.

Regards,
Bert

> -----Original Message-----
> From: rafi [mailto:rafi@sasi.com]
> Sent: Friday, June 23, 2000 10:50 AM
> To: sip@lists.bell-labs.com
> Subject: [SIP] Is SIP Transperent in this case ?
> 
> 
> Hi,
> 
> 
>    Consider a call setup involving PSTN-IP-PSTN  network.
> How much (SIP is used in IP network -:)  )
> transparent SIP can be in this case ?.If features like
> calling party presentation  restriction  are desired then how 
> can SIP handle/carry  these  kind of  information?
> 
> 
>    
> Thank You
> Rafi Assadi H.M.
> Silicon Automation Systems, Bangalore
> 
> 
> 
> 
> 


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From sip-admin@lists.bell-labs.com  Fri Jun 23 15:33:27 2000
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Subject: Re: [SIP] Caller receives responses
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Sean Olson wrote:
> 
> >
> > If you fork REGISTERs (I'm not sure why you'd do this -- perhaps
> > towards some sort of redundancy?), are there any potential
> > problems with only returning one 200?  The only other request
> > I can possibly see problems with (because you might get a response
> > from a UA that ultimately you wouldn't have been interested in)
> > is OPTIONS.  (Of course, unknown methods are forwarded like
> > OPTIONS.)
> 
> I can think of a couple of good applications for forking REGISTER and other
> non-INVITE requests where you definitely DO care about receiving
> ALL of the 200 class responses. If a UA already has to deal with
> multiple 2xx response for an INVITE, is it really so different
> to do this for non-INVITE requests as well?

Yes, because it is impossible.

Multiple 200 OK for INVITE only works because you can ACK each one
individually. With non-INVITE, the only way to retrigger a response
retransmission is a request retransmission. But, request retransmissions
are not for a specific response; thus, it would be impossible to
reliably transmit multiple 200 OK to non-INVITE.

I do not believe there is really a need for multiple 200 OK for
non-INVITE, in any case.

-Jonathan R.


-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Subject: Re: [SIP] Max-Forwards + Proxies
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Shail Bhatnagar wrote:
> 
> I don't have a problem with this simplification - but the spec says
> an incoming request with Max-Forwards of 0 MUST NOT be forwarded.
> It does not say that the request cannot be locally processed and a
> response returned. Since REGISTER (for the registrar) and CANCEL
> will be absorbed by the proxy, I thought it is okay to process these
> requests.
> The relevant statement in the spec should be appropriately changed -
> 
> request MUST NOT be forward and MUST NOT be processed.

That is the proposal; as I said, I think it is very confusing otherwise. 

-Jonathan R.


-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Fri Jun 23 15:46:51 2000
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Subject: Re: [SIP] Max-Forwards + Proxies
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Jonathan Rosenberg wrote:
> 
> Shail Bhatnagar wrote:
> >
> > I don't have a problem with this simplification - but the spec says
> > an incoming request with Max-Forwards of 0 MUST NOT be forwarded.
> > It does not say that the request cannot be locally processed and a
> > response returned. Since REGISTER (for the registrar) and CANCEL
> > will be absorbed by the proxy, I thought it is okay to process these
> > requests.
> > The relevant statement in the spec should be appropriately changed -
> >
> > request MUST NOT be forward and MUST NOT be processed.
> 
> That is the proposal; as I said, I think it is very confusing >otherwise.

I thought behavior of registrars/redirect servers closely matches
user agents, in terms of request handling. Of course
there are differences. Per this proposal, a user agent will process a
request with Max-Forwards of 0 but registrars/redirect servers and
proxies will not. Is my understanding correct ??



Thanks,
Shail


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From sip-admin@lists.bell-labs.com  Fri Jun 23 16:08:28 2000
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The draft  draft-dcsgroup-sip-state-01.txt was presented in Adelaide.  In
brief, the draft  discusses the "State" header extension that may be used to
transfer call state from the proxy to the UAC. The proxy can then be
stateless for the duration of the call.  The state is stored at the
UAC/endpoint for the duration of the call and passed back to the proxy if
mid-call changes are required. The draft also includes rules at endpoints
and proxies on inclusion and processing of  State headers. Comments and
suggestions on the State header extension as presented in draft-01 are
welcome. (this is available at
http://search.ietf.org/internet-drafts/draft-dcsgroup-sip-state-01.txt  ). 

I am in the process of updating the draft and submitting it as
draft-dcsgroup-sip-state-02.txt before the Pittsburgh IETF. The dcsgroup has
cleared the RFC2026 issues and the updated draft will reflect that and
comments received on this list on the proposed State header extension and
associated rules. 


Thanks and regards,
Poornima Lalwaney



------------------------------------------------
Poornima Lalwaney
Nokia
poornima.lalwaney@nokia.com
Tel: +1 858-831-4727
-----------------------------------------------


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From sip-admin@lists.bell-labs.com  Fri Jun 23 21:09:07 2000
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Subject: RE: [SIP] Static Images
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Perhaps an accompanying *-Disposition or *-Type or *-Function header to
describe the purpose of *-Info would be helpful. This stops a UA trying to
display something that was not supposed to be. Actually, IMO, this goal
would perhaps be better served by incorporating the
function/type/disposition information into the actual *-Info field. This
would make the provisioning of multiple *-Info headers much simpler. 

Display-Info (simple but can be misleading), Optional-Info,
Optional-Content, Fluff-Info? :)

Robert.

-----Original Message-----
From: Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
Sent: Friday, June 23, 2000 9:42 PM
To: Jonathan Rosenberg
Cc: Fairlie-Cuninghame, Robert; Adam B. Roach; Henning Schulzrinne;
Billy Biggs; Colin Perkins; sip@lists.bell-labs.com
Subject: Re: [SIP] Static Images


This sounds find (I'll incorporate it), but I do wonder if one wouldn't
want to be more specific in the purpose of the header, rather than just
"URI". For the 'visual caller id', something like '*-Info' (with a
suitable value of *, TBD) seems more evocative. I don't want to follow
the INFO tradition of dumping everything into one header. This may be
relevant, for example, as to whether the UA uses this information to
update some kind of address book.

Jonathan Rosenberg wrote:
> 
> So, in the interests of coming to consensus, let me propose that:
> 
> 1. inline content MAY be included, possibly using multipart. This would
> obviously include MIME types like application/URI-list which are
> themselves indirect references
> 2. we define a new header, called URI or something, which contains
> indirect references to content that is to be displayed. The header is
> totally optional to process at UAS. It cannot be used for content that
> is critical for processing the request properly.
> 
> We also RECOMMEND that proxies use the URI header when inserting
> content. A UA inserting content can use either, but indirect references
> are RECOMMENDED for content over 500 bytes (or some other number). In
> this way, if a UAC sends content inline using multipart, and the UAS
> doesn't support it, the 415 response will indicate that multipart isn't
> supported. The UAC can then resubmit the request using the URI header.
> 
> This keeps things open and flexible, but makes concrete recommendations
> to guide implementors (i.e, supporting the URI header would be your
> first choice).
> 
> -Jonathan R.
> 
> Henning Schulzrinne wrote:
> >
> > The problem with having proxies add body parts is (a) complexity, (b)
> > that now all clients have to understand multipart, even if they have no
> > interest in the information (I would guess that most current SIP clients
> > don't understand multipart) and (c) that it doesn't work with signed
> > requests. Headers containing a reference (http:, say) avoid these
> > problems. The only drawback is that such references require http support
> > in clients (which may be an issue for a very light-weight clients, but
> > most of these are probably graphically challenged anyway) and requires
> > that the information is available on a publically accessible web page. I
> > suspect that most corporate users would have difficult uploading their
> > likeness or other images to a web page, and not everyone wants to have
> > their likeness framed by an advertisement from geocity or kin. These
> > are, more or less, the arguments that were made the last time this topic
> > came around, I believe.
> 
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Sat Jun 24 04:23:53 2000
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From: "rafi" <rafi@sasi.com>
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Date: Sat, 24 Jun 2000 13:50:42 +0530
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Subject: [SIP] How Expire field works  in this case ?
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Hi,

                 Consider a call set up between  UAC and=20
     UAS  passing through 'n' number of proxies. When
    UAC  generated  INVITE message it used   Expire
    field with time  't'.  All the intermediate proxies  upon receiving
    INVITE message decided  to use time 't' in order to limit their=20
    search. When  INVITE finally arrives at UAS (assuming user is
   presently unavailable),   it decides to wait for time duration 't'.
    Now, before timer of duration 't' really expires  at UAS, timer at =
proxy (first proxy=20
    in the call setup) expires first resulting in call being released. =
Thus call=20
    MAY be released  much earlier than call waiting time promised at =
UAS.=20
    Thus, how  should  UAS use  time 't' present in INVITE request when=20
    user is  currently not available?.


       Second question is how does UA handles the=20
case of INVITE crossing each other for a given pair of
user ?.  Is this decision is left to user of SIP stack ?


Thank You
Rafi Assadi H.M.
Silicon Automation Systems
Bangalore,  INDIA



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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2614.3500" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi,</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
Consider a call set up between&nbsp; UAC and </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp;&nbsp; UAS&nbsp; =
passing through=20
'n' number of proxies. When</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; UAC&nbsp; generated  =
INVITE=20
message it used&nbsp;&nbsp; Expire</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; field with =
time&nbsp; 't'.&nbsp;=20
All the intermediate proxies</FONT><FONT face=3DArial size=3D2>&nbsp; =
upon=20
receiving</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; INVITE message=20
decide</FONT><FONT face=3DArial size=3D2>d &nbsp;to use time 't' in =
order to limit=20
their </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; search. =
When</FONT><FONT=20
face=3DArial size=3D2>&nbsp; INVITE finally</FONT><FONT face=3DArial =
size=3D2> arrives=20
at UAS (assuming user is</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp; presently =
unavailable),&nbsp; &nbsp;it=20
decides to wait for time duration 't'.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp; &nbsp;Now, before =
t</FONT><FONT=20
face=3DArial size=3D2>imer of duration 't' really expires</FONT><FONT =
face=3DArial=20
size=3D2> &nbsp;at UAS,</FONT><FONT face=3DArial size=3D2> timer at =
proxy (first proxy=20
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; in the call setup)=20
expires</FONT><FONT face=3DArial size=3D2> first</FONT><FONT =
face=3DArial size=3D2>=20
resulting in call being released. Thus call&nbsp;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp; &nbsp;MAY be=20
released&nbsp;</FONT><FONT face=3DArial size=3D2> much earlier =
than</FONT><FONT=20
face=3DArial size=3D2> call waiting time promised at =
UAS.&nbsp;</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;</FONT><FONT face=3DArial =
size=3D2>&nbsp;=20
Thus, how&nbsp;</FONT><FONT face=3DArial size=3D2> should&nbsp; UAS use  =
time 't'=20
present in INVITE</FONT><FONT face=3DArial size=3D2> request when =
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; user is</FONT><FONT =
face=3DArial=20
size=3D2>&nbsp; currently not available?.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Second=20
question is how does UA handles the </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>case of INVITE crossing each other for =
a given pair=20
of</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>user ?.&nbsp; Is this decision is left =
to user of=20
SIP stack ?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thank You</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Rafi Assadi H.M.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Silicon Automation Systems</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Bangalore,&nbsp; INDIA</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

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Subject: Re: [SIP] How Expire field works  in this case ?
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> rafi wrote:
> 
> Hi,
> 
>                  Consider a call set up between  UAC and
>      UAS  passing through 'n' number of proxies. When
>     UAC  generated INVITE message it used   Expire
>     field with time  't'.  All the intermediate proxies  upon
> receiving
>     INVITE message decided  to use time 't' in order to limit their
>     search. When  INVITE finally arrives at UAS (assuming user is
>    presently unavailable),   it decides to wait for time duration 't'.
>     Now, before timer of duration 't' really expires  at UAS, timer at
> proxy (first proxy
>     in the call setup) expires first resulting in call being released.
> Thus call
>     MAY be released  much earlier than call waiting time promised at
> UAS.
>     Thus, how  should  UAS use time 't' present in INVITE request when
>     user is  currently not available?.
> 

The timer is not suitable for microsecond timing. If forwarding your
call takes more than a second, something is seriously broken (and your
call throughput is going to be pretty low). Thus, the window where the
call gets terminated by the first proxy rather than the UAS is going to
be measured in units of milliseconds. In addition, even if the first
proxy cancels the call, the total search time is still the expiration
time, so this is correct behavior. The phone will stop ringing either
way. (This can happen in any event if a proxy has a shorter
pre-configured patience than the UAS.)



> 
>        Second question is how does UA handles the
> case of INVITE crossing each other for a given pair of
> user ?.  Is this decision is left to user of SIP stack ?

This question has been discussed several times on the list, at great
length. Please take a look at the archives.

> 
> 
> Thank You
> Rafi Assadi H.M.
> Silicon Automation Systems
> Bangalore,  INDIA
> 
>


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Can you explain what you mean by "feature processing"? That's not really
a SIP term...

James Kempf wrote:
> 
> Can anybody point me at a draft that discusses call feature processing in SIP?
> In particular, I'm interested in anything that separates feature
> processing from initial connection management. Thanx.
> 
>                 jak
> 
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> 
> The quote indirectly says you shouldn't cache whether the host was
> contacted or not; it says you should start the search afresh when
> contacting the same host. To me, this means you are not allowed to
> remember that you previously reached the host at address X.

I have no (serious) problem with caching results of failed requests, but
the paragraph cited is simply silent on this issue. It is not clear to
me whether this implementation decision needs to be specified, since the
"bad server" avoidance policy can be as complex as "if this is a
first-priority server, retry with exponential back-off every 5, 10, 20
seconds; if it is a lower-priority server, retry with some other delay".
I don't think that the DNS time-out is necessarily the optimal choice,
since validity of name translations and likely downtime durations are
not inherently related.

Might be useful to see if the MX record RFC (and RFC 821/822) says
anything about this. I believe SMTP MTAs have fairly sophisticated retry
algorithms for failed servers, but I'm not sure if these are for a
single message or per server. (Due to aggregation, probably makes no
difference.)

> 
> The idea here is just to make
> > sure that SRV load balancing is observed. I don't think the text says
> > anything about caching addresses, but it does have the potential
> > drawback of bypassing load balancing. Since caching is likely only to be
> > useful for large servers with lots of traffic, this could be a problem.
> > You don't want AOL to always go to the same Earthlink server.
> 
> The cache would need to honor DNS TTLs, as the paragraph says, so it
> would eventually time out. Not caching means you may increase call setup
> times waiting to fail against servers you have been failing against for
> the last hour...
> 
> -Jonathan R.
> 
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip




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While this is getting into hypotheticals (it is unlikely, say, that the
SIP Forum would deny membership to any semi-reasonable candidate, as the
likely "stink" generated is going to be far in excess of whatever damage
the membership could conceivably cause), one low-cost model might be to
provide a generally agreed upon set of call flows and sample messages,
similar to the torture tests and call flows in the existing I-D, that
makes the test open to community inspection and allows
self-certification. Since this is not a health-and-safety issue,
self-"certification" is less of a risk. I doubt that any company would
want to be caught claiming "we 'passed' the set of message flows" if
that was not the truth.

Neil Deason wrote:
> 
> Sean Olson wrote:
> >
> > Dave Walker wrote:
> > > Well, I don't know about the WAP Forum, but I would think that
> > > a certification program should certainly be open to non-members
> > > if it's run by a forum that charges for membership.  I'd be
> > > surprised if a certification program could be run without
> > > costs that have to be covered somehow (volunteer organizations
> > > aren't too reliable in the long term).
> >
> > Oops. I think you have a typo in there. There a number of volunteer
> > organizations in the OpenSource community that have been very
> > reliable, even in the long term.
> 
> Indeed - the IETF is made up of volunteers.
> 
> > I do agree, however, that it is
> > reasonable for an organization to charge a reasonable fee for
> > certification (provided that don't make a business out of it)
> 
> Yes, this wasn't my major concern. That is that any
> closed organisation (a membership oriented
> group is by definition closed to non-members, be
> it because they can't afford the fees or even the
> governing body reserves the right to deny membership
> to them) might be able to decide what is certified
> as being the standard for an open protocol, such as SIP.
> 
> Cheers,
> Neil.
> --
> Ubiquity Software Corporation, UK
> http://www.ubiquity.net
> 
> _______________________________________________
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> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip




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Date: Sun, 25 Jun 2000 15:14:55 -0500
From: Henry Sinnreich <Henry.Sinnreich@WCom.com>
Subject: RE: [SIP] Launch of www.sipcenter.com
In-reply-to: <3951D29D.BFA3E6C4@ubiquity.net>
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Hearty congratulations and success with the SIP Center!

Henry

Henry Sinnreich
MCI WorldCom
400 International Parkway
Richardson, Texas 75081
USA

>-----Original Message-----
>From: sip-admin@lists.bell-labs.com
>[mailto:sip-admin@lists.bell-labs.com]On Behalf Of
>Michael Doyle
>Sent: Thursday, June 22, 2000 3:47 AM
>To: sip-implementors@cs.columbia.edu; sip@lists.bell-labs.com
>Subject: [SIP] Launch of www.sipcenter.com
>
>
>UBIQUITY SOFTWARE LAUNCHES WWW.SIPCENTER.COM
>WITH CO-SPONSORSHIP FROM SUN MICROSYSTEMS AND TELECOM
>TECHNOLOGIES
>
>SIP Center  - an independent portal dedicated to the
>acceleration of the
>commercial development of Session Initiation Protocol software.
>
>VON, Stockholm, Sweden, June 20th, 2000 - Ubiquity
>Software Corporation,
>a major developer of 'End to End' SIP Service
>Solutions, today announced
>the launch of the new web site:
>http://www.sipcenter.com - a portal
>dedicated to the development and deployment of SIP
>(Session Initiation
>Protocol) based products. Ubiquity's co-sponsors, Sun
>Microsystems and
>telecom technologies, will be assisting with
>promotion, next generation
>technologies and services. The SIP Center web site,
>designed for both
>technical and commercial use, is an open and
>independent site where SIP
>ideas and views can be exchanged. With a unique Test
>Bed Area, SIP
>software developers can access to SIP software for
>interoperability
>testing 24hrs a day, 7 days a week, 365 days a year.
>
>There are no membership fees or joining process and
>all companies are
>invited to participate in the exchange of ideas and
>news items related
>to the SIP community.  Interactive forums have been
>setup to facilitiate
>rapid exchange of information.
>
>The SIP Center is actively seeking sponsorship from
>other companies and
>organisations.
>
>--
>Michael Doyle                         Ubiquity
>Software Corporation
>Chief Technology Officer              Ubiquity House
>Tel: +44 (0) 1633-765600              Langstone Park
>Fax: +44 (0) 1633-765601              Newport   NP18 2LH
>URL: http://www.ubiquity.net
>
>
>_______________________________________________
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>SIP@lists.bell-labs.com
>http://lists.bell-labs.com/mailman/listinfo/sip



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Date: Sun, 25 Jun 2000 16:22:54 -0500
From: Henry Sinnreich <Henry.Sinnreich@WCom.com>
Subject: RE: [SIP] Announcement of SIP Forum
In-reply-to: <3952587E.3930A139@ss8networks.com>
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I have serious misgivings about a certification program.
The SIP bakeoffs provide excellent and discreet info that helps
vendors correct their bugs.
Would rather see vendors stating compliance with the published
I-D's with call flows.

Henry

>-----Original Message-----
>From: sip-admin@lists.bell-labs.com
>[mailto:sip-admin@lists.bell-labs.com]On Behalf Of Dave Walker
>Sent: Thursday, June 22, 2000 1:19 PM
>To: 'sip@lists.bell-labs.com'
>Subject: Re: [SIP] Announcement of SIP Forum
>
>
>In case anyone is not aware of it, I believe that this type
>of certification program is currently being considered by the
>SIP Working Group in the Softswitch Consortium (which is *not*
>looking for post-H.323 work!).
>
>Dave Walker
>SS8 Networks
>Ottawa, Canada
>
>
>Henning Schulzrinne wrote:
>>
>> One of the ideas that was discussed at the Telia
>dinner in Stockholm was
>> whether there was a need for a basic SIP
>certification program. There
>> are obvious pluses and minuses for this, but my
>concern is that if this
>> is a real need, I can think of a few organizations
>(looking for
>> post-H.323 work...) that I would rather *not* see doing this.
>> --
>> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
>>
>> _______________________________________________
>> SIP mailing list
>> SIP@lists.bell-labs.com
>> http://lists.bell-labs.com/mailman/listinfo/sip
>
>
>_______________________________________________
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>http://lists.bell-labs.com/mailman/listinfo/sip



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From: James Kempf <James.Kempf@eng.sun.com>
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To: "Henning Schulzrinne" <hgs@cs.columbia.edu>,
        "James Kempf" <James.Kempf@eng.sun.com>
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Subject: Re: [SIP] Feature Processing Draft?
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For example, suppose the SIP proxy server gets an INVITE from a client that's a prepaid
cellular caller, and the company that sold the service is not the operator.
The operator's SIP proxy would like to pass authorization and accounting
(and potentially handling the call itself) to the third party server.

This *particular* example involves authorization and accounting, but one could imagine other
such features that do not involve them but are actually implemented by
a third party. 

I'm familiar with OPTIONS, which seems to be a way for the SIP server itself
to advertise that it supports particular features. Is SIP the proper way for
the SIP server to communicate with a server implementing the actual options
themselves or would that be some other protocol?

			jak

>Can you explain what you mean by "feature processing"? That's not really
>a SIP term...
>
>James Kempf wrote:
>> 
>> Can anybody point me at a draft that discusses call feature processing in
>SIP?
>> In particular, I'm interested in anything that separates feature
>> processing from initial connection management. Thanx.
>> 
>>                 jak
>> 
>> _______________________________________________
>> SIP mailing list
>> SIP@lists.bell-labs.com
>> http://lists.bell-labs.com/mailman/listinfo/sip
>
>




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Rather than repackaging SIP requests into another protocol (since
probably almost information in the request might be useful for some
service or another), it would seem easiest if the SIP proxy server
simply proxies the request to the server that has the necessary
information. Elsewhere, one might to need to worry about sending
signaling around, but this seems easiest in the SIP case. Clearly, for
some specialized applications, such as QoS and AAA, the mechanisms
described in draft-sinnreich-sip-qos-osp might be useful.

James Kempf wrote:
> 
> For example, suppose the SIP proxy server gets an INVITE from a client that's a prepaid
> cellular caller, and the company that sold the service is not the operator.
> The operator's SIP proxy would like to pass authorization and accounting
> (and potentially handling the call itself) to the third party server.
> 
> This *particular* example involves authorization and accounting, but one could imagine other
> such features that do not involve them but are actually implemented by
> a third party.
> 
> I'm familiar with OPTIONS, which seems to be a way for the SIP server itself
> to advertise that it supports particular features. Is SIP the proper way for
> the SIP server to communicate with a server implementing the actual options
> themselves or would that be some other protocol?
> 
>                         jak
> 
> >Can you explain what you mean by "feature processing"? That's not really
> >a SIP term...
> >
> >James Kempf wrote:
> >>
> >> Can anybody point me at a draft that discusses call feature processing in
> >SIP?
> >> In particular, I'm interested in anything that separates feature
> >> processing from initial connection management. Thanx.
> >>
> >>                 jak
> >>
> >> _______________________________________________
> >> SIP mailing list
> >> SIP@lists.bell-labs.com
> >> http://lists.bell-labs.com/mailman/listinfo/sip
> >
> >


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From sip-admin@lists.bell-labs.com  Mon Jun 26 00:16:38 2000
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
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Subject: Re: [SIP] Static Images
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"Fairlie-Cuninghame, Robert" wrote:
> 
> Perhaps an accompanying *-Disposition or *-Type or *-Function header to
> describe the purpose of *-Info would be helpful. This stops a UA trying to
> display something that was not supposed to be. Actually, IMO, this goal
> would perhaps be better served by incorporating the
> function/type/disposition information into the actual *-Info field. This
> would make the provisioning of multiple *-Info headers much simpler.

Agree; sounds like a parameter:

Display-Info: http://www.example.com/picture;purpose=picture,
  http://www.example.com/hello.au;purpose=speakname

We would need to define some basic purpose values, and then define a
procedure for specifying new ones with IANA.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Mon Jun 26 03:02:51 2000
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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>, sip@lists.bell-labs.com
Subject: RE: [SIP] Static Images
Date: Mon, 26 Jun 2000 00:02:59 -0700
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Perhaps this Display-Info field could also be used to describe inline
content. This means that all information regarding optional content is in
the same place.

Display-Info:
http://www.example.com/picture;purpose=picture,inline:2;purpose=aboutme

where inline:2 means that the content is contained in the second multipart
body. [Although this would place a requirement on proxies adding multipart
bodies to place them at the end.]

Do you think that a mime type parameter is required (especially for URL's)?
eg 
Display-Info: http://www.example.com/picture;purpose=picture;type=image/tiff

Robert.

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Monday, June 26, 2000 12:18 PM
To: Fairlie-Cuninghame, Robert
Cc: 'Henning Schulzrinne'; sip@lists.bell-labs.com
Subject: Re: [SIP] Static Images




"Fairlie-Cuninghame, Robert" wrote:
> 
> Perhaps an accompanying *-Disposition or *-Type or *-Function header to
> describe the purpose of *-Info would be helpful. This stops a UA trying to
> display something that was not supposed to be. Actually, IMO, this goal
> would perhaps be better served by incorporating the
> function/type/disposition information into the actual *-Info field. This
> would make the provisioning of multiple *-Info headers much simpler.

Agree; sounds like a parameter:

Display-Info: http://www.example.com/picture;purpose=picture,
  http://www.example.com/hello.au;purpose=speakname

We would need to define some basic purpose values, and then define a
procedure for specifying new ones with IANA.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Mon Jun 26 04:44:25 2000
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Date: Mon, 26 Jun 2000 09:41:36 +0100
To: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>,
        "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
From: Lawrence Conroy <lwc@roke.co.uk>
Subject: RE: [SIP] Static Images
Cc: "'Henning Schulzrinne'" <hgs@cs.columbia.edu>, sip@lists.bell-labs.com
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At 12:02 am -0700 26/6/00, Fairlie-Cuninghame, Robert wrote:
>Perhaps this Display-Info field could also be used to describe inline
>content. This means that all information regarding optional content is in
>the same place.
>
>Display-Info:
>http://www.example.com/picture;purpose=picture,inline:2;purpose=aboutme
>
>where inline:2 means that the content is contained in the second multipart
>body. [Although this would place a requirement on proxies adding multipart
>bodies to place them at the end.]
>
>Do you think that a mime type parameter is required (especially for URL's)?
>eg
>Display-Info: http://www.example.com/picture;purpose=picture;type=image/tiff
>
>Robert.
>
Problem is, there is an ASSUMPTION here that the body parts are never 
re-ordered.
This would require a prohibition or re-ordering, as implied above.

If you use the body part content-ID field instead of just a plain 
part index, then
re-ordering is no longer a problem, and we don't have to add such a 
prohibition.
Thus, inline:2@3456 indicates the subpart with a content-id:2@3456.
MIME Content-ID is defined Appendix A of RFC 2045, and that in turn 
took it from
RFC822. We stand on the shoulders of giants.
-- 
All the best, Lawrence
-----------------------------------------------------------------------
| Lawrence Conroy,    | "These Opinions must be mine, 'cos if they    |
| Roke Manor Research |  were my Company's they'd pay me for them"    |
|- lwc@roke.co.uk  ---+- Tel: +44 1794 833666  Fax: +44 1794 833434 --|


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From sip-admin@lists.bell-labs.com  Mon Jun 26 08:42:35 2000
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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To: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
Cc: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>, sip@lists.bell-labs.com
Subject: Re: [SIP] Static Images
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"Fairlie-Cuninghame, Robert" wrote:
> 
> Perhaps this Display-Info field could also be used to describe inline
> content. This means that all information regarding optional content is in
> the same place.
> 
> Display-Info:
> http://www.example.com/picture;purpose=picture,inline:2;purpose=aboutme
> 
> where inline:2 means that the content is contained in the second multipart
> body. [Although this would place a requirement on proxies adding multipart
> bodies to place them at the end.]
> 
> Do you think that a mime type parameter is required (especially for URL's)?
> eg
> Display-Info: http://www.example.com/picture;purpose=picture;type=image/tiff

I know this is tempting, but can we please make this implementable in
less than a GHz Pentium workstation and describable in less than 30
pages? 

Adding MIME types to a URL is a really bad idea. URLs may well map to
several different MIME types, depending on content negotiation. (This is
implemented in Apache, for example.)

Since it doesn't matter, semantically, whether you speak something or
display it, I would also be disinclined about being too specific about
the rendering method used. We don't do this for the web, either.


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


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Subject: Re: [SIP] Static Images
To: jdrosen@dynamicsoft.com (Jonathan Rosenberg)
Date: Mon, 26 Jun 2000 07:58:07 -0500 (CDT)
Cc: rfairlie@nuera.com (Fairlie-Cuninghame Robert),
        hgs@cs.columbia.edu ('Henning Schulzrinne'), sip@lists.bell-labs.com
In-Reply-To: <3956D96F.AE5A9495@dynamicsoft.com> from "Jonathan Rosenberg" at Jun 26, 2000 12:17:51 AM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>Agree; sounds like a parameter:
>
>Display-Info: http://www.example.com/picture;purpose=picture,
>  http://www.example.com/hello.au;purpose=speakname
>
>We would need to define some basic purpose values, and then define a
>procedure for specifying new ones with IANA.

I'm with Henning on this one: your values seem to imply a certain
type of *rendering*, not a certain type of information -- and,
in HTTP at least, it's the client's responsiblity to render the
content type fetched in a appropriate form.

Display-Info: http://www.example.com/call-id/adam.roach

For example, if you fetch this URI with an Accept of
"audio/*", and no video parameters, the HTTP server
may return an audio clip. If you send both of
"audio/*, image/*", it may return a GIF instead.
This has to do with trusting the way HTTP currently
handles browser capabilities.

I know this isn't widely used right now, even though
servers (and browsers) support it; I'm sure
this type of use of HTTP will expand as terminals
evolve away from 17" screens with 101-key keyboards.

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA


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From sip-admin@lists.bell-labs.com  Mon Jun 26 09:17:07 2000
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Subject: Re: [SIP] Static Images
To: jdrosen@dynamicsoft.com (Jonathan Rosenberg)
Date: Mon, 26 Jun 2000 08:16:24 -0500 (CDT)
Cc: hgs@cs.columbia.edu (Henning Schulzrinne),
        rfairlie@nuera.com (Fairlie-Cuninghame Robert),
        schulzrinne@cs.columbia.edu (Henning Schulzrinne),
        vektor@div8.net (Billy Biggs), C.Perkins@cs.ucl.ac.uk (Colin Perkins),
        sip@lists.bell-labs.com
In-Reply-To: <3953189E.4125273D@dynamicsoft.com> from "Jonathan Rosenberg" at Jun 23, 2000 03:58:22 AM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>So, in the interests of coming to consensus, let me propose that:
>
>1. inline content MAY be included, possibly using multipart. This would
>obviously include MIME types like application/URI-list which are
>themselves indirect references

I agree with your goal, but my originally posed question
seems a bit unanswered still. If you'll remember, the goal
was to send arbitrary-sized still images (or a single
image) after a conversation has been set up. This can be
generalised into the problem of sending arbitrary data
during the course of a conversation, where the data
has no realtime characteristics. I prefer this problem
description, since it avoids being mired down with
the realtime stuff like low bitrate codecs (which I don't
think address my problem; they all aim towards error
concealment instead of error correction).

If I read the ongoing conversation so far, the consensus
would seem to point to a solution of sending an INFO
midcall (or possibly a re-INVITE, to keep the number of
required extensions somewhat lower); this INFO or INVITE
would contain a new header (Display-Info has been proposed)
with a URL. Since the image in this case is expected to
originate from the terminal sending the INFO or INVITE,
it would be an HTTP URL pointing to itself, with some sort
of identifier to tie back to the call or resource 
(implementation dependant, but chosen to be the call ID in
this example).

INVITE sip:bob@terminal1273.example.com SIP/2.0
From: Adam Roach <sip:adam.roach@ericsson.com>;tag=99999
To: Bob Example <sip:bob@example.com>;tag=44444
Call-ID: 12345@b04a24.exu.ericsson.se
Display-Info: http://b04a24.exu.ericsson.se/12345

Bob's terminal would then contact mine using HTTP to get the
picture:

GET /12345 HTTP/1.0
Accept: image/jpeg, image/png, video/mpeg, audio/ulaw

So, the terminals would need a minimal HTTP implementation
for this purpose. I have no problem with this; I just wanted
to make sure that, given this information, the consensus
remains the same.

Finally, I pose a quick straw poll: INFO or INVITE? INFO 
requires much less processing (since it can't change 
callstate), while INVITE will already be implemented on 
all clients. (Consider: what does your implementation
do if it gets a re-INVITE with no SDP present? With unchanged
SDP? Are you sure everyone else has done the same thing?)

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA


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From sip-admin@lists.bell-labs.com  Mon Jun 26 09:24:34 2000
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Subject: Re: [SIP] Static Images
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> "Fairlie-Cuninghame, Robert" wrote:
> > 
> > Perhaps this Display-Info field could also be used to describe inline
> > content. This means that all information regarding optional content is in
> > the same place.
> > 
> > Display-Info:
> > http://www.example.com/picture;purpose=picture,inline:2;purpose=aboutme
> > 
> > where inline:2 means that the content is contained in the second multipart
> > body. [Although this would place a requirement on proxies adding multipart
> > bodies to place them at the end.]
> > 
> > Do you think that a mime type parameter is required (especially for URL's)?
> > eg
> > Display-Info: http://www.example.com/picture;purpose=picture;type=image/tiff
> 

This is not a road we want to go down. Especially with deeply nested
multipart bodies. 

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


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From sip-admin@lists.bell-labs.com  Mon Jun 26 13:38:19 2000
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Date: Mon, 26 Jun 2000 13:39:37 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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To: "Adam B. Roach" <Adam.Roach@Ericsson.com>
Cc: Fairlie-Cuninghame Robert <rfairlie@nuera.com>,
        "'Henning Schulzrinne'" <hgs@cs.columbia.edu>, sip@lists.bell-labs.com
Subject: Re: [SIP] Static Images
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"Adam B. Roach" wrote:
> 
> >Agree; sounds like a parameter:
> >
> >Display-Info: http://www.example.com/picture;purpose=picture,
> >  http://www.example.com/hello.au;purpose=speakname
> >
> >We would need to define some basic purpose values, and then define a
> >procedure for specifying new ones with IANA.
> 
> I'm with Henning on this one: your values seem to imply a certain
> type of *rendering*, not a certain type of information -- and,
> in HTTP at least, it's the client's responsiblity to render the
> content type fetched in a appropriate form.

No, they don't imply any kind of mandated behavior. I was trying to
address a concern raised previously on the list, which was that just the
MIME type wasn't enough. Is this image a thumbnail of the caller? Is it
a picture of a phone for a "graphical custom ringing" or something? The
idea with this purpose thing was just a hint to indicate the purpose of
the content, so that the device could do with it something reasonable,
as it saw fit. 

-Jonathan R.



-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Mon Jun 26 13:45:50 2000
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        Fairlie-Cuninghame Robert <rfairlie@nuera.com>,
        Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
        Billy Biggs <vektor@div8.net>, Colin Perkins <C.Perkins@cs.ucl.ac.uk>,
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"Adam B. Roach" wrote:
> 
> >So, in the interests of coming to consensus, let me propose that:
> >
> >1. inline content MAY be included, possibly using multipart. This would
> >obviously include MIME types like application/URI-list which are
> >themselves indirect references
> 
> I agree with your goal, but my originally posed question
> seems a bit unanswered still. If you'll remember, the goal
> was to send arbitrary-sized still images (or a single
> image) after a conversation has been set up. This can be
> generalised into the problem of sending arbitrary data
> during the course of a conversation, where the data
> has no realtime characteristics. I prefer this problem
> description, since it avoids being mired down with
> the realtime stuff like low bitrate codecs (which I don't
> think address my problem; they all aim towards error
> concealment instead of error correction).

The difference is artificial. You can no doubt use RTP over TCP and
obtain reliable, non-real time transport.


> 
> If I read the ongoing conversation so far, the consensus
> would seem to point to a solution of sending an INFO
> midcall (or possibly a re-INVITE, to keep the number of
> required extensions somewhat lower); this INFO or INVITE
> would contain a new header (Display-Info has been proposed)
> with a URL. Since the image in this case is expected to
> originate from the terminal sending the INFO or INVITE,
> it would be an HTTP URL pointing to itself, with some sort
> of identifier to tie back to the call or resource
> (implementation dependant, but chosen to be the call ID in
> this example).

The URL need not point to itself, of course.

If you are doing a slide show of your kids, I still think that TCP/RTP
is better. If it is something more aynchronous and "one shot", I see no
fault in using INFO. In fact, INFO already allows this with content in
the body:

> If a server receives an INFO request with a body it understands,
>       but it has no knowledge of INFO associated processing rules for
>       the body, the body MAY be rendered and displayed to the user. The
>       INFO is responded to with a 200 OK.


I would prefer INFO to re-INVITE in this case; re-INVITE is meant to
have some affect on session state. This does not.


> Finally, I pose a quick straw poll: INFO or INVITE? INFO
> requires much less processing (since it can't change
> callstate), while INVITE will already be implemented on
> all clients. (Consider: what does your implementation
> do if it gets a re-INVITE with no SDP present? With unchanged
> SDP? Are you sure everyone else has done the same thing?)

re-INVITE with no SDP mid-call is probably an error. If you want to add
a stream, you add an m line. If you want to remove one, you set its port
to zero and re-INVITE. Sending no SDP at all is not right. If the SDP is
unchanged, its unchanged. This is a session refresh, as per session
timer.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Subject: Re: [SIP] Max-Forwards + Proxies
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Shail Bhatnagar wrote:
> 
> Jonathan Rosenberg wrote:
> >
> > Shail Bhatnagar wrote:
> > >
> > > I don't have a problem with this simplification - but the spec says
> > > an incoming request with Max-Forwards of 0 MUST NOT be forwarded.
> > > It does not say that the request cannot be locally processed and a
> > > response returned. Since REGISTER (for the registrar) and CANCEL
> > > will be absorbed by the proxy, I thought it is okay to process these
> > > requests.
> > > The relevant statement in the spec should be appropriately changed -
> > >
> > > request MUST NOT be forward and MUST NOT be processed.
> >
> > That is the proposal; as I said, I think it is very confusing >otherwise.
> 
> I thought behavior of registrars/redirect servers closely matches
> user agents, in terms of request handling. Of course
> there are differences. Per this proposal, a user agent will process a
> request with Max-Forwards of 0 but registrars/redirect servers and
> proxies will not. Is my understanding correct ??

No. Any server (UA, proxy, redirect, registrar) receiving a request with
Max-Forwards zero responds with a 483 and does not process.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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Subject: [SIP] Compact form for CSeq
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Hi:

Any particular reason there isn't a compact form representation of
CSeq?  It is a mandatory parameter in all SIP requests.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
IN Architecture and Internet Software Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184


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"Vijay K. Gurbani" wrote:
> 
> Hi:
> 
> Any particular reason there isn't a compact form representation of
> CSeq?  It is a mandatory parameter in all SIP requests.
> 

Mostly oversight, secondly because it would only save 3 bytes. Thirdly,
it's too late to change it...

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


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From sip-admin@lists.bell-labs.com  Mon Jun 26 14:39:01 2000
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Subject: Re: [SIP] Compact form for CSeq
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"Vijay K. Gurbani" wrote:
> 
> Henning Schulzrinne wrote:
> >

> 
> > Thirdly, it's too late to change it...
> 
> Is it?  Older UAC's can still use the full form.  In any case, it is no
> big deal, really.  I was just looking for some orthogonality with other
> required headers (like To, From, Via) and since a big drive for compact
> form appears to be memory-constrained devices, CSeq, as a required
> header appeared to be left out.

This discussion appears to occur occasionally. We can't add
abbreviations for "old" headers since old UAs wouldn't understand the
compact form. (That they can't generate them is not the issue.)

If you do compression, CSeq is likely to be transformed to one byte or
less anyway.

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


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Subject: Re: [SIP] >   12. Re: Questions on section 10.1.1 of RFC2543bis 
 (Jonathan Rosenberg)
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Gopal wrote:
> 
> > No; nothing like that. I actually think the complete To field should be
> > included in the comparison, so as to make this consistent with the
> > definition of a call leg (which does include To). In practice it will
> 
> There is another potential problem though with comparing tag value of To.
> Consider the foll. scenario when a UAC talks directly to a UAS  (no proxy)
> 
>         UAC                             UAS
> INVITE (no tag value for To)----------->Generate new Tag value for To
>                         (lost)X <------ 100 Trying (MAY include tag in To)
>                         (lost)X <------ 180 Ringing (MAY include tag in To)
> 
> (timeout. Retransmit INVITE)
> INVITE ((no tag value for To)---------->Rejected since Tag value of To does not
>                                         match
> 
> 10.1.1 of RFC2543bis I think (wrongfully) suggests the above behavior. Ideally,
> re-transmission of the INVITE should trigger the 180 response from the UAS. I
> think the tag value of To should be compared only if the To field in the
> incoming message has a tag. Sec. 6.23 actually states that tag values of 2
> URIs are compared only if both URIs have tags. How about changing the sentence to
> 
> "If the Call-ID was found, it compares the tag value of To (if present)
>  with the users tag and rejects the request if the two do not match."
> 
> -gopal

Right. This also allows you to send a BYE without knowing the tag,
something that can cause problems, but people seem very insistent on.
We'll try to clean up the description. Tags are an unending source of
confusion.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Phil Hoffer wrote:
> 
> Hi,
> 
> Section 15.1.1 of the rfc2543bis draft is missing information, and a BNF entry with respect to pgp-pubalgolrithm.
> 
> Presumably, pgp-pubalgolrithm should be included in the BNF for pgp-params in section 15.1.1

Yes, I believe it should.

> 
> Also some text about the parameter is required, e.g. what is the default value of pgp-pubalgolrithm if it is omitted from WWW-Authenticate header.
> Is it assumed to be RSA?

I presume... I guess it depends on what the default is for PGP in
general, if there is a default...?

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
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From sip-admin@lists.bell-labs.com  Mon Jun 26 14:49:49 2000
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Subject: Re: [SIP] Compact form for CSeq
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Henning Schulzrinne wrote:
> 
> "Vijay K. Gurbani" wrote:
> >
> > Hi:
> >
> > Any particular reason there isn't a compact form representation of
> > CSeq?  It is a mandatory parameter in all SIP requests.
> >
> 
> Mostly oversight, secondly because it would only save 3 bytes. 

That's what I thought, but since we have a compact form for To and Via 
(at a savings of 1 and 2 bytes respectively), I figured maybe CSeq 
should be included as well since it is a required header in all 
requests.

> Thirdly, it's too late to change it...

Is it?  Older UAC's can still use the full form.  In any case, it is no
big deal, really.  I was just looking for some orthogonality with other
required headers (like To, From, Via) and since a big drive for compact 
form appears to be memory-constrained devices, CSeq, as a required
header appeared to be left out.

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
IN Architecture and Internet Software Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
Naperville, Illinois 60566  Voice: +1 630 224 0216 Fax: +1 630 713 0184


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Hi:

A collegue asked this question of me, for which I have an answer (see 
below), but would like to validate it.

Question: Why does the RFCbis (June 9, 2000) mandate that the tag be 
"globally unique" (Section 6.23)?  Can't it be unique within the Call-ID 
space it occupies?  The Call-ID already is "a globally unique 
identifier".  Why have 2 globally unique identifiers to identify a 
single call leg?

One reason I can think of is that if multiple UA servers are contacted
for a call, this wording guarantees that each of the UA servers use an
algorithm to generate a globally unique tag; otherwise, if each of them
used, say, the time of day to generate the tag (and they happen to be
in the same time zone etc. etc.), there could be a potential to generate
the same tag at each UA server.

Is this accurate?  Any other reasons?

Thanks,

- vijay
-- 
Vijay K. Gurbani  vkg@lucent.com vkg@research.bell-labs.com vkg@acm.org
IN Architecture and Internet Software Group
Lucent Technologies/Bell Labs Innovations, 263 Shuman Blvd, Rm 1A-418
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> > I thought behavior of registrars/redirect servers closely matches
> > user agents, in terms of request handling. Of course
> > there are differences. Per this proposal, a user agent will process a
> > request with Max-Forwards of 0 but registrars/redirect servers and
> > proxies will not. Is my understanding correct ??
> 
> No. Any server (UA, proxy, redirect, registrar) receiving a request with
> Max-Forwards zero responds with a 483 and does not process.
> 
> -Jonathan R.
> 

Okay, so any proxy should not forward a request with incoming
Max-Forwards 1, since if it forwards it, Max-Forwards will become
0 and the downstream guy will return 483.

Right ??


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Shail Bhatnagar wrote:
> 
> > > I thought behavior of registrars/redirect servers closely matches
> > > user agents, in terms of request handling. Of course
> > > there are differences. Per this proposal, a user agent will process a
> > > request with Max-Forwards of 0 but registrars/redirect servers and
> > > proxies will not. Is my understanding correct ??
> >
> > No. Any server (UA, proxy, redirect, registrar) receiving a request with
> > Max-Forwards zero responds with a 483 and does not process.
> >
> > -Jonathan R.
> >
> 
> Okay, so any proxy should not forward a request with incoming
> Max-Forwards 1, since if it forwards it, Max-Forwards will become
> 0 and the downstream guy will return 483.
> 
> Right ??

I suppose so, yes.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Date: Mon, 26 Jun 2000 15:36:00 -0500
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        "Adam B. Roach" <Adam.Roach@Ericsson.com>
From: Richard Shockey <rshockey@ix.netcom.com>
Subject: Re: [SIP] Static Images
Cc: Fairlie-Cuninghame Robert <rfairlie@nuera.com>,
        "'Henning Schulzrinne'" <hgs@cs.columbia.edu>, sip@lists.bell-labs.com
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>
> > >Display-Info: http://www.example.com/picture;purpose=picture,
> > >  http://www.example.com/hello.au;purpose=speakname
> > >
> > >We would need to define some basic purpose values, and then define a
> > >procedure for specifying new ones with IANA.
> >
> > I'm with Henning on this one: your values seem to imply a certain
> > type of *rendering*, not a certain type of information -- and,
> > in HTTP at least, it's the client's responsiblity to render the
> > content type fetched in a appropriate form.
>
>No, they don't imply any kind of mandated behavior. I was trying to
>address a concern raised previously on the list, which was that just the
>MIME type wasn't enough. Is this image a thumbnail of the caller? Is it
>a picture of a phone for a "graphical custom ringing" or something? The
>idea with this purpose thing was just a hint to indicate the purpose of
>the content, so that the device could do with it something reasonable,
>as it saw fit.


Gentlemen ... I'd like to offer a "RATHOLE ALERT" here.

The problem of defining content-types and the priority of those 
content-types in processing has been thoroughly debated in the VPIM WG with 
no conclusion.

Please note  { draft-burger-vpim-cc-00.txt } for some background discussion 
of what you are getting into.

The context of each application is different but the problem statement is 
quite similar.


 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
Please Note New Contact Information:

Richard Shockey
Shockey Consulting LLC
5237 Sutherland
St. Louis, MO 63109
Voice 314.503.0640
eFAX Fax to EMail 815.333.1237 (Preferred for Fax)
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From sip-admin@lists.bell-labs.com  Mon Jun 26 16:37:52 2000
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        Sean Olson <eussean@exu.ericsson.se>, vkg@lucent.com,
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Subject: Re: [SIP] Max-Forwards + Proxies
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Jonathan Rosenberg wrote:
> 
> Shail Bhatnagar wrote:
> >
> > > > I thought behavior of registrars/redirect servers closely matches
> > > > user agents, in terms of request handling. Of course
> > > > there are differences. Per this proposal, a user agent will process a
> > > > request with Max-Forwards of 0 but registrars/redirect servers and
> > > > proxies will not. Is my understanding correct ??
> > >
> > > No. Any server (UA, proxy, redirect, registrar) receiving a request with
> > > Max-Forwards zero responds with a 483 and does not process.
> > >
> > > -Jonathan R.
> > >
> >
> > Okay, so any proxy should not forward a request with incoming
> > Max-Forwards 1, since if it forwards it, Max-Forwards will become
> > 0 and the downstream guy will return 483.
> >
> > Right ??
> 
> I suppose so, yes.

But that would defeat the whole point of being able to figure out the
downstream capabilities, returned in the 483. Thus, a proxy should *not*
try to be too smart. After all, if IP routers did this, traceroute
wouldn't work...


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


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From sip-admin@lists.bell-labs.com  Mon Jun 26 18:01:51 2000
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Date: Tue, 27 Jun 2000 00:01:40 +0200 (CEST)
From: Lars Berggren <lars@intertex.se>
To: Poornima.Lalwaney@nokia.com
Cc: sip@lists.bell-labs.com
Subject: Re: [SIP] Distributing call state : draft-dcsgroup-sip-state-01.txt
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Some comments:

1. The extension does not define an option-tag to be used with the Require
and Supported header fields. How would a proxy know that the UAC or UAS
support the 'state' extension?

2. Section 4.2 mentions that the state header is included in subsequent
INVITEs. What about BYE requests? I think it could be useful to include it
in BYEs also, since BYEs certainly affects call state.

/Lars


On Fri, 23 Jun 2000 Poornima.Lalwaney@nokia.com wrote:
> 
> The draft  draft-dcsgroup-sip-state-01.txt was presented in Adelaide.  In
> brief, the draft  discusses the "State" header extension that may be used to
> transfer call state from the proxy to the UAC. The proxy can then be
> stateless for the duration of the call.  The state is stored at the
> UAC/endpoint for the duration of the call and passed back to the proxy if
> mid-call changes are required. The draft also includes rules at endpoints
> and proxies on inclusion and processing of  State headers. Comments and
> suggestions on the State header extension as presented in draft-01 are
> welcome. (this is available at
> http://search.ietf.org/internet-drafts/draft-dcsgroup-sip-state-01.txt  ). 
> 
> I am in the process of updating the draft and submitting it as
> draft-dcsgroup-sip-state-02.txt before the Pittsburgh IETF. The dcsgroup has
> cleared the RFC2026 issues and the updated draft will reflect that and
> comments received on this list on the proposed State header extension and
> associated rules. 
> 
> 
> Thanks and regards,
> Poornima Lalwaney
> 
> 
> 
> ------------------------------------------------
> Poornima Lalwaney
> Nokia
> poornima.lalwaney@nokia.com
> Tel: +1 858-831-4727
> -----------------------------------------------
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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



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        Sean Olson <eussean@exu.ericsson.se>, vkg@lucent.com,
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Subject: Re: [SIP] Max-Forwards + Proxies
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Henning Schulzrinne wrote:
> 
> > > Okay, so any proxy should not forward a request with incoming
> > > Max-Forwards 1, since if it forwards it, Max-Forwards will become
> > > 0 and the downstream guy will return 483.
> > >
> > > Right ??
> >
> > I suppose so, yes.
> 
> But that would defeat the whole point of being able to figure out the
> downstream capabilities, returned in the 483. Thus, a proxy should *not*
> try to be too smart. After all, if IP routers did this, traceroute
> wouldn't work...

I was assuming that the proxy wouldn't say anything useful in the 483.
If it does contain a Supported header, then yes, you are right. Sooo,
seems like in general it is NOT a good idea to drop the request with
Max-Forwards 1. 


-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Mon Jun 26 23:52:12 2000
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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: "'Henning Schulzrinne'" <schulzrinne@cs.columbia.edu>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: sip@lists.bell-labs.com
Subject: RE: [SIP] Max-Forwards + Proxies
Date: Mon, 26 Jun 2000 20:52:32 -0700
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It is my understanding that the IP TTL operates in the way that Johnathan
described. That is, a router should not forward packets with a TTL of 0 or 1
(returning an ICMP in both cases). A Host however is free to process a
received packet with a TTL of 0 or 1 (although it should not normally
receive a packet with a TTL of 0). Traceroute knows that it reached the
destination because the endpoint returns an ICMP Echo reply (rather than
ICMp Time Exceeded).

This is not necessarily the behavior we want in this case but I thought I
would straighten out the analogy.

Cheers,

Robert.

-----Original Message-----
From: Henning Schulzrinne [mailto:schulzrinne@cs.columbia.edu]
Sent: Tuesday, June 27, 2000 4:34 AM
To: Jonathan Rosenberg
Cc: Shail Bhatnagar; Neil Deason; Sean Olson; vkg@lucent.com;
sip@lists.bell-labs.com
Subject: Re: [SIP] Max-Forwards + Proxies


Jonathan Rosenberg wrote:
> 
> Shail Bhatnagar wrote:
> >
> > > > I thought behavior of registrars/redirect servers closely matches
> > > > user agents, in terms of request handling. Of course
> > > > there are differences. Per this proposal, a user agent will process
a
> > > > request with Max-Forwards of 0 but registrars/redirect servers and
> > > > proxies will not. Is my understanding correct ??
> > >
> > > No. Any server (UA, proxy, redirect, registrar) receiving a request
with
> > > Max-Forwards zero responds with a 483 and does not process.
> > >
> > > -Jonathan R.
> > >
> >
> > Okay, so any proxy should not forward a request with incoming
> > Max-Forwards 1, since if it forwards it, Max-Forwards will become
> > 0 and the downstream guy will return 483.
> >
> > Right ??
> 
> I suppose so, yes.

But that would defeat the whole point of being able to figure out the
downstream capabilities, returned in the 483. Thus, a proxy should *not*
try to be too smart. After all, if IP routers did this, traceroute
wouldn't work...


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


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From: "rafi" <rafi@sasi.com>
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Date: Tue, 27 Jun 2000 11:32:43 +0530
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Couple of comment on the draft.

Syntax for Anonymity is given as

(1)     Anonymity       =3D "Anonymity" ":"  *privacy-tag
         privacy-tag     =3D "Full" | "URI" | "Name" | "IPAddr" | "Off"

I think it should have been,

       Anonymity       =3D "Anonymity" ":"  1#privacy-tag
       privacy-tag     =3D "Full" | "URI" | "Name" | "IPAddr" | "Off"


(2)  When DCS-proxy-t  encrypts the "addr-spec" in the Remote-Party-ID
       I think it should use different key each time it encrypts.


Thank You
Rafi Assadi H.M.
Silicon Automation Systems
Bangalore, INDIA



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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2614.3500" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Couple of comment on the =
draft.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Syntax for Anonymity is given =
as</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>(1)&nbsp;&nbsp; =20
&nbsp;Anonymity&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =3D "Anonymity" =
":"&nbsp;=20
*privacy-tag</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;=20
privacy-tag&nbsp;&nbsp;&nbsp;&nbsp; =3D "Full" | "URI" | "Name" | =
"IPAddr" |=20
"Off"</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I think it should have =
been,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Anonymity&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;=20
=3D "Anonymity" ":"&nbsp; =
1#privacy-tag<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
&nbsp;privacy-tag&nbsp;&nbsp;&nbsp;&nbsp; =3D "Full" | "URI" | "Name" | =
"IPAddr" |=20
"Off"</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>(2)&nbsp; When DCS-proxy-t&nbsp; =
encrypts the=20
"addr-spec" in the Remote-Party-ID</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I =
think it=20
should use different key each time it encrypts.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Thank You</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Rafi Assadi H.M.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Silicon Automation Systems</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Bangalore, INDIA</FONT></DIV>
<DIV><FONT face=3DArial size=3D2><BR>&nbsp;</DIV></FONT></BODY></HTML>

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Subject: Re: [SIP] Max-Forwards + Proxies
References: <200006222312.SAA19957@b04a45.exu.ericsson.se> <3952F3E5.982D5382@dynamicsoft.com> <39532457.C5EF0266@ubiquity.net> <395379EA.1D187EED@cisco.com> <3953847A.27A61A4B@dynamicsoft.com> <39538776.E603ABAF@cisco.com> <3953BC06.59B9275E@dynamicsoft.com> <3953BD98.8545FFB9@cisco.com> <39579B64.BD7C4A04@dynamicsoft.com> <3957B290.F14A4BE8@cisco.com> <3957BB81.F3888271@dynamicsoft.com> <3957BE1F.5A892769@cs.columbia.edu> <3958238F.FD826DD9@dynamicsoft.com>
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Jonathan Rosenberg wrote:
> 
> Henning Schulzrinne wrote:

> >
> > But that would defeat the whole point of being able to figure out the
> > downstream capabilities, returned in the 483. Thus, a proxy should *not*
> > try to be too smart. After all, if IP routers did this, traceroute
> > wouldn't work...
> 
> I was assuming that the proxy wouldn't say anything useful in the 483.
> If it does contain a Supported header, then yes, you are right. Sooo,
> seems like in general it is NOT a good idea to drop the request with
> Max-Forwards 1.

> 
> -Jonathan R.
> 

Another semi-interesting question is whether it is useful for any server
that returns a final response to include a "Server" header. That way,
one could find out which types of servers are along the path. (I suspect
that some people are including this in the Via header comment, but even
so, that information doesn't get back to the UAC.)
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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Not to belabor the point, but please see RFC 1009 (Requirements for
Internet Gateways), Section 4.8. Your statement about TTL appears to be
at variance with the facts. (Not forwarding a packet with TTL 1 also
wouldn't work since receiving a packet with TTL 0 is perfectly valid.)
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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> ÀÌ°æÈñ wrote:
> 
> hi.
> 
> In <The SIP INFO Method> Document...
> 
> Page 3
> 
> 1.1 Example Uses
> 
> - Carrying images or other non streaming information between the
> participatnts of a session.
> 
> 
> What is "image" ?
> Microsoft Netmeeting's board send image from user to user.
> Is that same??
> Otherwise, what is "image"?

Any MIME type of type image/* whose purpose is to render on the
recipients machine.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From: "Nils Appeldoorn (EMN)" <Nils.Appeldoorn@emn.ericsson.se>
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Subject: [SIP] How do these programs work?
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Hello everybody, 

I hope this is an approperiate mailinglist, please correct me if it's not. Thanks anyway.

I'm doing some research on the current use of VoIP (voice and multimedia). I want to get the state of the art on VoIP applications that are available on the net. 

Now, I know there are a few applications available. I've seen Net2Phone, DialPad. But the problem is, I want to know what functionality these programs offer, where are they used for? And how do they work? Are they SIP or H.323 based? 

I can't look into the programs, maybe some of you know the answer? 

Can someone explain to me how these programs work? What's the principle behind these services? 

Thanks, 

Nils 



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From sip-admin@lists.bell-labs.com  Tue Jun 27 09:44:50 2000
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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Subject: [SIP] SIP bake-off technical program committee
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I'm looking for volunteers to assist with planning the technical aspects
of the upcoming SIP bake-off, including interoperability scenarios,
grouping of implementations, etc.
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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From sip-admin@lists.bell-labs.com  Tue Jun 27 09:57:24 2000
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Subject: Re: [SIP] Static Images
To: jdrosen@dynamicsoft.com (Jonathan Rosenberg)
Date: Tue, 27 Jun 2000 08:56:38 -0500 (CDT)
Cc: rfairlie@nuera.com (Fairlie-Cuninghame Robert),
        hgs@cs.columbia.edu ('Henning Schulzrinne'), sip@lists.bell-labs.com
In-Reply-To: <39579559.71A27F7B@dynamicsoft.com> from "Jonathan Rosenberg" at Jun 26, 2000 01:39:37 PM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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Jonathan Rosenberg writes:
>"Adam B. Roach" wrote:
>> 
>> >Agree; sounds like a parameter:
>> >
>> >Display-Info: http://www.example.com/picture;purpose=picture,
>> >  http://www.example.com/hello.au;purpose=speakname
>> >
>> >We would need to define some basic purpose values, and then define a
>> >procedure for specifying new ones with IANA.
>> 
>> I'm with Henning on this one: your values seem to imply a certain
>> type of *rendering*, not a certain type of information -- and,
>> in HTTP at least, it's the client's responsiblity to render the
>> content type fetched in a appropriate form.
>
>No, they don't imply any kind of mandated behavior. I was trying to
>address a concern raised previously on the list, which was that just the
>MIME type wasn't enough. Is this image a thumbnail of the caller? Is it
>a picture of a phone for a "graphical custom ringing" or something? The
>idea with this purpose thing was just a hint to indicate the purpose of
>the content, so that the device could do with it something reasonable,
>as it saw fit. 

I propose, then, that the purpose for *both* of your URIs
is the same: caller identification. I beleive the distinction you're
trying to make would be better illustrated by, for example, the
difference between a spoken name to identify the caller and
a custom alerting tone provided (or, more appropriately, suggested)
by the caller. The callee's device may then choose to present this 
content appropriately.

Display-Info: http://www.example.com/picture;purpose=callerid,
  http://www.example.com/hello.au;purpose=callerid,
  http://www.example.com/chimes.au;purpose=alerting

-- 
Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA


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From sip-admin@lists.bell-labs.com  Tue Jun 27 10:04:24 2000
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"Adam B. Roach" wrote:
> 
> Jonathan Rosenberg writes:
> >"Adam B. Roach" wrote:
> >>
> >> >Agree; sounds like a parameter:
> >> >
> >> >Display-Info: http://www.example.com/picture;purpose=picture,
> >> >  http://www.example.com/hello.au;purpose=speakname
> >> >
> >> >We would need to define some basic purpose values, and then define a
> >> >procedure for specifying new ones with IANA.
> >>
> >> I'm with Henning on this one: your values seem to imply a certain
> >> type of *rendering*, not a certain type of information -- and,
> >> in HTTP at least, it's the client's responsiblity to render the
> >> content type fetched in a appropriate form.
> >
> >No, they don't imply any kind of mandated behavior. I was trying to
> >address a concern raised previously on the list, which was that just the
> >MIME type wasn't enough. Is this image a thumbnail of the caller? Is it
> >a picture of a phone for a "graphical custom ringing" or something? The
> >idea with this purpose thing was just a hint to indicate the purpose of
> >the content, so that the device could do with it something reasonable,
> >as it saw fit.
> 
> I propose, then, that the purpose for *both* of your URIs
> is the same: caller identification. I beleive the distinction you're
> trying to make would be better illustrated by, for example, the
> difference between a spoken name to identify the caller and
> a custom alerting tone provided (or, more appropriately, suggested)
> by the caller. The callee's device may then choose to present this
> content appropriately.
> 
> Display-Info: http://www.example.com/picture;purpose=callerid,
>   http://www.example.com/hello.au;purpose=callerid,
>   http://www.example.com/chimes.au;purpose=alerting
> 

Given that we don't have to pay taxes for headers, and given that the
number of purposes suggested so far is small, why all these parameters?

Using, for example,

To-Info:
From-Info:
Alert:

is much shorter and it's much easier to tell what's going on. If we go
the parameter route, I'd suggest being more consistent and rewriting SIP
as

Protocol-Info: some-url;purpose=to-address
Protocol-Info: some-other-url;purpose=from-address
Protocol-Info: 1234 INVITE;purpose=sequence-number
Protocol-Info: INVITE;purpose=method

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


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From sip-admin@lists.bell-labs.com  Tue Jun 27 14:24:45 2000
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From: Poornima.Lalwaney@nokia.com
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Lars,

Thanks for your comments.

> 1. The extension does not define an option-tag to be used 
> with the Require
> and Supported header fields. How would a proxy know that the 
> UAC or UAS
> support the 'state' extension?
> 

I agree that we need a Requires: header so the proxy can depend on the
UAC/UAS storing the state headers and returning them. Perhaps Requires:
State (or any other name besides "dcs" which is being proposed for proxy to
proxy signaling extensions) would work. I will update the draft to include a
discussion on Requires and Supports with an option-tag.

> 2. Section 4.2 mentions that the state header is included in 
> subsequent
> INVITEs. What about BYE requests? I think it could be useful 
> to include it
> in BYEs also, since BYEs certainly affects call state.
> 

As currently used in DCS, the state header includes information that is
relevant to proxies only in subsequent INVITE's that go through proxies
(i.e. INVITE's that affect mid call "resource" or "authorization" changes).
As you correctly point out, in a general model if the state information sent
by the proxy to the client is required by the proxy at call termination, and
if the BYE goes through proxies, the state header can be sent back in a BYE.
The updated draft will include some discussion on the general case. 


Thanks..
Poornima Lalwaney


> On Fri, 23 Jun 2000 Poornima.Lalwaney@nokia.com wrote:
> > 
> > The draft  draft-dcsgroup-sip-state-01.txt was presented in 
> Adelaide.  In
> > brief, the draft  discusses the "State" header extension 
> that may be used to
> > transfer call state from the proxy to the UAC. The proxy can then be
> > stateless for the duration of the call.  The state is stored at the
> > UAC/endpoint for the duration of the call and passed back 
> to the proxy if
> > mid-call changes are required. The draft also includes 
> rules at endpoints
> > and proxies on inclusion and processing of  State headers. 
> Comments and
> > suggestions on the State header extension as presented in 
> draft-01 are
> > welcome. (this is available at
> > 
> http://search.ietf.org/internet-drafts/draft-dcsgroup-sip-stat
e-01.txt  ). 
> 
> I am in the process of updating the draft and submitting it as
> draft-dcsgroup-sip-state-02.txt before the Pittsburgh IETF. The dcsgroup
has
> cleared the RFC2026 issues and the updated draft will reflect that and
> comments received on this list on the proposed State header extension and
> associated rules. 
> 
> 
> Thanks and regards,
> Poornima Lalwaney
> 
> 
> 
> ------------------------------------------------
> Poornima Lalwaney
> Nokia
> poornima.lalwaney@nokia.com
> Tel: +1 858-831-4727
> -----------------------------------------------
> 
> 
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
> 

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



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        IMPP WG <impp@iastate.edu>
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--=====================_1302240==_.ALT
Content-Type: text/plain; charset="us-ascii"


Mike

How long did it take you to think of this acronym? If less than an hour -- you
need a life, man!

;-)

cheers

At 03:18 PM 6/15/2000 -0700, Fox, Mike wrote:
>Why stop at XML for IMPP? Why not XML for SIP? Simple Object Access Protocol
>may be the right way. Call it SIP Control by Unified Methods.
>
>SOAP SCUM 
>
>
>Best regards,
>Mike Fox
>
> 
*************************************
"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
--=====================_1302240==_.ALT
Content-Type: text/html; charset="us-ascii"

<html><div>Mike</div>
<br>
<div>How long did it take you to think of this acronym? If less than an
hour -- you need a life, man!</div>
<br>
<div>;-)</div>
<br>
<div>cheers</div>
<br>
<div>At 03:18 PM 6/15/2000 -0700, Fox, Mike wrote:</div>
<div>&gt;Why stop at XML for IMPP? Why not XML for SIP? Simple Object
Access Protocol</div>
<div>&gt;may be the right way. Call it SIP Control by Unified
Methods.</div>
<div>&gt;</div>
<div>&gt;SOAP SCUM </div>
<div>&gt;</div>
<div>&gt;</div>
<div>&gt;Best regards,</div>
<div>&gt;Mike Fox</div>
<div>&gt;</div>
&gt;
<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>

--=====================_1302240==_.ALT--



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From sip-admin@lists.bell-labs.com  Tue Jun 27 21:40:28 2000
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Date: Tue, 27 Jun 2000 21:41:36 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Cc: "Adam B. Roach" <Adam.Roach@Ericsson.com>,
        Fairlie-Cuninghame Robert <rfairlie@nuera.com>,
        sip@lists.bell-labs.com
Subject: Re: [SIP] Static Images
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Henning Schulzrinne wrote:
> 
> > >No, they don't imply any kind of mandated behavior. I was trying to
> > >address a concern raised previously on the list, which was that just the
> > >MIME type wasn't enough. Is this image a thumbnail of the caller? Is it
> > >a picture of a phone for a "graphical custom ringing" or something? The
> > >idea with this purpose thing was just a hint to indicate the purpose of
> > >the content, so that the device could do with it something reasonable,
> > >as it saw fit.
> >
> > I propose, then, that the purpose for *both* of your URIs
> > is the same: caller identification. I beleive the distinction you're
> > trying to make would be better illustrated by, for example, the
> > difference between a spoken name to identify the caller and
> > a custom alerting tone provided (or, more appropriately, suggested)
> > by the caller. The callee's device may then choose to present this
> > content appropriately.
> >
> > Display-Info: http://www.example.com/picture;purpose=callerid,
> >   http://www.example.com/hello.au;purpose=callerid,
> >   http://www.example.com/chimes.au;purpose=alerting
> >
> 
> Given that we don't have to pay taxes for headers, and given that the
> number of purposes suggested so far is small, why all these parameters?
> 
> Using, for example,
> 
> To-Info:
> From-Info:
> Alert:

The difference is that with a single header, if you don't understand the
parameter that gives a "hint" on what its for, you still know to display
it. If you use a separate header for each "hint", then as we come up
with new purposes, they won't be handled at all by older UAs.

In general, parameters make sense when they further qualify the
semantics implied by the header. I think this is the case here. The
header says "render this". The parameter provides an application layer
hint, which may be ignored, saying for what purpose this should be
rendered.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From sip-admin@lists.bell-labs.com  Wed Jun 28 02:14:53 2000
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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: "'Henning Schulzrinne'" <schulzrinne@cs.columbia.edu>
Cc: sip@lists.bell-labs.com
Subject: RE: [SIP] Max-Forwards + Proxies
Date: Tue, 27 Jun 2000 23:14:32 -0700
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I got this from Stevens, TCP/IP Illustrated Vol. 1 chapter on Traceroute. It
is normally a fairly good source. I read RFC1009 and it also says that you
should never receive a packet with TTL 0 (cf. Section 4.8). Likewise,
RFC1122(Host Requirements) states
	"A host MUST NOT discard a datagram just because it was
      received with TTL less than 2.
	.... The intent is that TTL expiration will cause a datagram
      to be discarded by a gateway but not by the destination
      host; however, hosts that act as gateways by forwarding
      datagrams must follow the gateway rules for TTL.

So I am unsure as to what is in dispute here. Anyway this is getting fairly
offtopic and so we should probably take the discussion offline.

Regards,

Robert.

-----Original Message-----
From: Henning Schulzrinne [mailto:schulzrinne@cs.columbia.edu]
Sent: Tuesday, June 27, 2000 8:23 PM
To: Fairlie-Cuninghame, Robert
Cc: Jonathan Rosenberg; sip@lists.bell-labs.com
Subject: Re: [SIP] Max-Forwards + Proxies


Not to belabor the point, but please see RFC 1009 (Requirements for
Internet Gateways), Section 4.8. Your statement about TTL appears to be
at variance with the facts. (Not forwarding a packet with TTL 1 also
wouldn't work since receiving a packet with TTL 0 is perfectly valid.)
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs


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From sip-admin@lists.bell-labs.com  Wed Jun 28 06:49:05 2000
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--NextPart

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


	Title		: SDP media alignment in SIP
	Author(s)	: G. Camarillo, J. Holler, G. Eriksson
	Filename	: draft-camarillo-sip-sdp-00.txt
	Pages		: 7
	Date		: 27-Jun-00
	
This document defines an SDP media attribute. This attribute is
intended to be used in conjunction with SIP in order to align
different media streams belonging to a session. The use of this
attribute allows sending media from a single media stream, encoded
in different formats during the session, to different ports and host
interfaces.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-camarillo-sip-sdp-00.txt

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-camarillo-sip-sdp-00.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-camarillo-sip-sdp-00.txt".
	
NOTE:	The mail server at ietf.org can return the document in
	MIME-encoded form by using the "mpack" utility.  To use this
	feature, insert the command "ENCODING mime" before the "FILE"
	command.  To decode the response(s), you will need "munpack" or
	a MIME-compliant mail reader.  Different MIME-compliant mail readers
	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
Content-Type: Multipart/Alternative; Boundary="OtherAccess"

--OtherAccess
Content-Type: Message/External-body;
	access-type="mail-server";
	server="mailserv@ietf.org"

Content-Type: text/plain
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ENCODING mime
FILE /internet-drafts/draft-camarillo-sip-sdp-00.txt

--OtherAccess
Content-Type: Message/External-body;
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	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

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--OtherAccess--

--NextPart--




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From: Elizabeth George <ElizabethG@sedonanetworks.com>
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Does anybody know of a Sip Server by MCI WorldCom which i can talk to

Thanks
EG



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From sip-admin@lists.bell-labs.com  Wed Jun 28 11:26:30 2000
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Date: Wed, 28 Jun 2000 16:07:02 +0100
From: Neil Deason <ndeason@ubiquity.net>
Organization: Ubiquity Software Corporation Limited
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To: Elizabeth George <ElizabethG@sedonanetworks.com>
Cc: "'sip@lists.research.bell-labs.com'" <sip@lists.research.bell-labs.com>
Subject: Re: [SIP] SIP server by mci
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Elizabeth George wrote:
> 
> Does anybody know of a Sip Server by MCI WorldCom which i can talk to
> 
> Thanks
> EG

Not an MCI one specifically but there is a list of public SIP
Servers here ...

	http://www.cs.columbia.edu/%7Ehgs/sip/servers.html

There is also a new public SIP Server not yet listed on that page
at http://www.sipcenter.com (you can actually reach it at
sip.sipcenter.com port 5060) but I would check the web site for
related info first.

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


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To: <sip@lists.bell-labs.com>
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The English version of Mr. Olivier Dubuisson's book "ASN.1 - Communication
between heterogeneous systems" is now available as a free download in pdf
format from:

http://www.oss.com/asn1/booksintro.html


Zipora Hartov




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From: "David R. Oran" <oran@cisco.com>
To: "Henning Schulzrinne" <schulzrinne@cs.columbia.edu>,
        "Adam B. Roach" <Adam.Roach@Ericsson.com>
Cc: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
        "Fairlie-Cuninghame Robert" <rfairlie@nuera.com>,
        <sip@lists.bell-labs.com>
References: <200006271356.IAA02779@b04a24.exu.ericsson.se> <3958B3ED.75C26A89@cs.columbia.edu>
Subject: Re: [SIP] Static Images
Date: Wed, 28 Jun 2000 20:25:12 -0400
Organization: Cisco Systems
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Henning's reductio ad absurdum really drove this home for me. If we want to
do something that needs a header, just invent a header.

We could go even further than Henning suggests and just cast SIP as a single
Lamba function in Common Lisp. Or better still, use my favorite
one-instruction programming language:

DWIM(arg1,arg2...)

where DWIM = "Do what I mean".

Dave.

> > I propose, then, that the purpose for *both* of your URIs
> > is the same: caller identification. I beleive the distinction you're
> > trying to make would be better illustrated by, for example, the
> > difference between a spoken name to identify the caller and
> > a custom alerting tone provided (or, more appropriately, suggested)
> > by the caller. The callee's device may then choose to present this
> > content appropriately.
> >
> > Display-Info: http://www.example.com/picture;purpose=callerid,
> >   http://www.example.com/hello.au;purpose=callerid,
> >   http://www.example.com/chimes.au;purpose=alerting
> >
>
> Given that we don't have to pay taxes for headers, and given that the
> number of purposes suggested so far is small, why all these parameters?
>
> Using, for example,
>
> To-Info:
> From-Info:
> Alert:
>
> is much shorter and it's much easier to tell what's going on. If we go
> the parameter route, I'd suggest being more consistent and rewriting SIP
> as
>
> Protocol-Info: some-url;purpose=to-address
> Protocol-Info: some-other-url;purpose=from-address
> Protocol-Info: 1234 INVITE;purpose=sequence-number
> Protocol-Info: INVITE;purpose=method
>
> :-)
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
>
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>
>



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From sip-admin@lists.bell-labs.com  Thu Jun 29 06:06:55 2000
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From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: "'David R. Oran'" <oran@cisco.com>,
        Henning Schulzrinne <schulzrinne@cs.columbia.edu>,
        "Adam B. Roach" <Adam.Roach@Ericsson.com>
Cc: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
        "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>,
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Subject: RE: [SIP] Static Images
Date: Thu, 29 Jun 2000 03:06:52 -0700
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However Dave, although Display-Info could be done using multiple headers
(anything is possible), I  don't believe this would be helpful. The parsing
is much more difficult and order is now important for multiple headers.
Adding purpose-specific parameters is IMO option a lot simpler and more
localized than adding new headers.

Come on, would you prefer

Display-Info: http:blah
Display-Purpose: picture
Display-Info: http:blah2
Display-Purpose: picture
Display-Info: http:blah3
Display-Purpose: myagent
Display-Other: whatever
Display-Info: inline:25487852
Display-Purpose: aboutme

or 

Display-Info: http:blah;purpose=picture
Display-Info: http:blah2;purpose=picture
Display-Info: http:blah;purpose=myagent;other=whatever
Display-Info: inline:25487852;purpose=aboutme
[or just comma separate them]

This is a very flexible & powerful feature and like anything that is very
flexible you want to make sure that flexibility does not turn into
widespread chaos - where there are a zillion SIP headers defined. Using only
only header 
a) abstracts/prevents the Display-Info flexibility from cluttering the core
SIP functionality and parsing.
b) localizes future extensions (the scope of any future parameter is limited
to that purpose type and not to ALL SIP processing!!)
c) removes the need correlate all the Display-* headers. Each header is
self-contained and order is unimportant. All Display info can be found in
one place - your Walmart of Display content.

Henning, so when are we going to see the new SIP draft with your syntax
proposal? <grin>

Cheers,

Robert.

-----Original Message-----
From: David R. Oran [mailto:oran@cisco.com]
Sent: Thursday, June 29, 2000 8:25 AM
To: Henning Schulzrinne; Adam B. Roach
Cc: Jonathan Rosenberg; Fairlie-Cuninghame Robert;
sip@lists.bell-labs.com
Subject: Re: [SIP] Static Images


Henning's reductio ad absurdum really drove this home for me. If we want to
do something that needs a header, just invent a header.

We could go even further than Henning suggests and just cast SIP as a single
Lamba function in Common Lisp. Or better still, use my favorite
one-instruction programming language:

DWIM(arg1,arg2...)

where DWIM = "Do what I mean".

Dave.

> > I propose, then, that the purpose for *both* of your URIs
> > is the same: caller identification. I beleive the distinction you're
> > trying to make would be better illustrated by, for example, the
> > difference between a spoken name to identify the caller and
> > a custom alerting tone provided (or, more appropriately, suggested)
> > by the caller. The callee's device may then choose to present this
> > content appropriately.
> >
> > Display-Info: http://www.example.com/picture;purpose=callerid,
> >   http://www.example.com/hello.au;purpose=callerid,
> >   http://www.example.com/chimes.au;purpose=alerting
> >
>
> Given that we don't have to pay taxes for headers, and given that the
> number of purposes suggested so far is small, why all these parameters?
>
> Using, for example,
>
> To-Info:
> From-Info:
> Alert:
>
> is much shorter and it's much easier to tell what's going on. If we go
> the parameter route, I'd suggest being more consistent and rewriting SIP
> as
>
> Protocol-Info: some-url;purpose=to-address
> Protocol-Info: some-other-url;purpose=from-address
> Protocol-Info: 1234 INVITE;purpose=sequence-number
> Protocol-Info: INVITE;purpose=method
>
> :-)
> --
> Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
>
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>
>



_______________________________________________
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SIP@lists.bell-labs.com
http://lists.bell-labs.com/mailman/listinfo/sip


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From sip-admin@lists.bell-labs.com  Thu Jun 29 08:04:48 2000
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From: "David R. Oran" <oran@cisco.com>
To: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>,
        "Henning Schulzrinne" <schulzrinne@cs.columbia.edu>,
        "Adam B. Roach" <Adam.Roach@Ericsson.com>
Cc: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>,
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        <sip@lists.bell-labs.com>
References: <B16E9BA540A0D211A11D00105A65571F9DAFFF@exchangesvr.nuera.com>
Subject: Re: [SIP] Static Images
Date: Thu, 29 Jun 2000 08:06:42 -0400
Organization: Cisco Systems
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Robert, perhaps you missed my point amidst the humor element. I was
advocating that the information actually be semantically tagged by the name
of the header so we don't have so much overloading and an extra identifier
space to manage. The tag method introduces a whole new identifier space with
its own semantics for the UA to understand, when the header would work just
fine. As Henning and others have ponted out, the rendering method (picture,
audio, video, smell, etc.) should be deduced by the UA from the content of
the URL when it's accessed rather than via URI names or purpose tags.

So, in my view, if you wanted a something to be presented to the user during
alerting, we'd have headers like:
Alerting-Info: http://doggie-heaven.aol.com/~mydogskip/mug.jpg
Alerting-Info: ftp://doggie-heaven.aol.com/~mydogskip/woof-woof.mp3
Alerting-Info: rtsp://doggie-heaven.aol.com/~mydogskip/phew-phew.sml
From-Info: http://doggie-heaven.aol.com/~mydogskip/skip.vcf

Depending on what the UA was capable of and the UA owner's preferences,
during alerting it could do one or more of:
a) display the picture of Skip
b) play Skip barking as the alerting sound
c) emit chemicals to simulate a very muddy unwashed skip walking into the
room.

and make available via some UI function Skip's vCard information to allow it
to be added to an address book, or whatever.

This making more sense?

Dave.

----- Original Message -----
From: "Fairlie-Cuninghame, Robert" <rfairlie@nuera.com>
To: "'David R. Oran'" <oran@cisco.com>; "Henning Schulzrinne"
<schulzrinne@cs.columbia.edu>; "Adam B. Roach" <Adam.Roach@Ericsson.com>
Cc: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>; "Fairlie-Cuninghame,
Robert" <rfairlie@nuera.com>; <sip@lists.bell-labs.com>
Sent: Thursday, June 29, 2000 6:06 AM
Subject: RE: [SIP] Static Images


> However Dave, although Display-Info could be done using multiple headers
> (anything is possible), I  don't believe this would be helpful. The
parsing
> is much more difficult and order is now important for multiple headers.
> Adding purpose-specific parameters is IMO option a lot simpler and more
> localized than adding new headers.
>
> Come on, would you prefer
>
> Display-Info: http:blah
> Display-Purpose: picture
> Display-Info: http:blah2
> Display-Purpose: picture
> Display-Info: http:blah3
> Display-Purpose: myagent
> Display-Other: whatever
> Display-Info: inline:25487852
> Display-Purpose: aboutme
>
> or
>
> Display-Info: http:blah;purpose=picture
> Display-Info: http:blah2;purpose=picture
> Display-Info: http:blah;purpose=myagent;other=whatever
> Display-Info: inline:25487852;purpose=aboutme
> [or just comma separate them]
>
> This is a very flexible & powerful feature and like anything that is very
> flexible you want to make sure that flexibility does not turn into
> widespread chaos - where there are a zillion SIP headers defined. Using
only
> only header
> a) abstracts/prevents the Display-Info flexibility from cluttering the
core
> SIP functionality and parsing.
> b) localizes future extensions (the scope of any future parameter is
limited
> to that purpose type and not to ALL SIP processing!!)
> c) removes the need correlate all the Display-* headers. Each header is
> self-contained and order is unimportant. All Display info can be found in
> one place - your Walmart of Display content.
>
> Henning, so when are we going to see the new SIP draft with your syntax
> proposal? <grin>
>
> Cheers,
>
> Robert.
>
> -----Original Message-----
> From: David R. Oran [mailto:oran@cisco.com]
> Sent: Thursday, June 29, 2000 8:25 AM
> To: Henning Schulzrinne; Adam B. Roach
> Cc: Jonathan Rosenberg; Fairlie-Cuninghame Robert;
> sip@lists.bell-labs.com
> Subject: Re: [SIP] Static Images
>
>
> Henning's reductio ad absurdum really drove this home for me. If we want
to
> do something that needs a header, just invent a header.
>
> We could go even further than Henning suggests and just cast SIP as a
single
> Lamba function in Common Lisp. Or better still, use my favorite
> one-instruction programming language:
>
> DWIM(arg1,arg2...)
>
> where DWIM = "Do what I mean".
>
> Dave.
>
> > > I propose, then, that the purpose for *both* of your URIs
> > > is the same: caller identification. I beleive the distinction you're
> > > trying to make would be better illustrated by, for example, the
> > > difference between a spoken name to identify the caller and
> > > a custom alerting tone provided (or, more appropriately, suggested)
> > > by the caller. The callee's device may then choose to present this
> > > content appropriately.
> > >
> > > Display-Info: http://www.example.com/picture;purpose=callerid,
> > >   http://www.example.com/hello.au;purpose=callerid,
> > >   http://www.example.com/chimes.au;purpose=alerting
> > >
> >
> > Given that we don't have to pay taxes for headers, and given that the
> > number of purposes suggested so far is small, why all these parameters?
> >
> > Using, for example,
> >
> > To-Info:
> > From-Info:
> > Alert:
> >
> > is much shorter and it's much easier to tell what's going on. If we go
> > the parameter route, I'd suggest being more consistent and rewriting SIP
> > as
> >
> > Protocol-Info: some-url;purpose=to-address
> > Protocol-Info: some-other-url;purpose=from-address
> > Protocol-Info: 1234 INVITE;purpose=sequence-number
> > Protocol-Info: INVITE;purpose=method
> >
> > :-)
> > --
> > Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
> >
> >
> > _______________________________________________
> > SIP mailing list
> > SIP@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/sip
> >
> >
>
>
>
> _______________________________________________
> SIP mailing list
> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
>
>



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Message-Id: <200006291243.HAA08045@b04a24.exu.ericsson.se>
Subject: Re: [SIP] Static Images
To: oran@cisco.com (David R. Oran)
Date: Thu, 29 Jun 2000 07:43:16 -0500 (CDT)
Cc: rfairlie@nuera.com (Fairlie-Cuninghame Robert),
        schulzrinne@cs.columbia.edu (Henning Schulzrinne),
        jdrosen@dynamicsoft.com (Jonathan Rosenberg), sip@lists.bell-labs.com
In-Reply-To: <0cfa01bfe1c2$7dd02040$07d245ab@cisco.com> from "David R. Oran" at Jun 29, 2000 08:06:42 AM
From: "Adam B. Roach" <Adam.Roach@Ericsson.com>
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>So, in my view, if you wanted a something to be presented to the user during
>alerting, we'd have headers like:
>Alerting-Info: http://doggie-heaven.aol.com/~mydogskip/mug.jpg
>Alerting-Info: ftp://doggie-heaven.aol.com/~mydogskip/woof-woof.mp3
>Alerting-Info: rtsp://doggie-heaven.aol.com/~mydogskip/phew-phew.sml
>From-Info: http://doggie-heaven.aol.com/~mydogskip/skip.vcf
>
>Depending on what the UA was capable of and the UA owner's preferences,
>during alerting it could do one or more of:
>a) display the picture of Skip
>b) play Skip barking as the alerting sound
>c) emit chemicals to simulate a very muddy unwashed skip walking into the
>room.
>
>and make available via some UI function Skip's vCard information to allow it
>to be added to an address book, or whatever.

I think the point that Jonathan is trying to make (and one
with which I agree) is that, for example, if you define a future
extension for some other semantics than "alerting" or "from," new
*headers* are suboptimal.

Henning's argument holds *only* if the header itself has
no associated semantics. In this case, it isn't true.
"Display-Info" conveys the concept that "this is additional 
content to be rendered for the user." The purpose 
parameters are merely giving hints as to how this 
information would most effectively be conveyed.

I'll demonstrate this point.

Sprog-Info: http://www.xyz.com/image.jpg

If your client didn't understand this new header type
of "Sprog-Info," it would be ignored. 

Display-Info: http://www.xyz.com/image.jpg;purpose=sprog

On the other hand, a client which understood the general
"Display-Info" header, but had no idea what this new "sprog" 
thing was would have a much better recovery. It could either 
present the image (possibly in a box labeled "sprog"), or at 
least let the user know: "I have an image here labeled 'sprog.' 
Do you want to see it?")

I see the second set of behavior as far preferable.

-- 
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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Here are some comments on the SIP RFC2543bis Draft.

1. Section 16.8 OPTIONS Request. The example SDP here does not have a c=
line. I believe it needs one, either at the session level or as part of each
media description in order to be compliant with the SDP spec.

2. Appendix B.7 Subject and SDP "s=" Line. This section indicates that the
s= line may be left empty for invitations to two-party sessions. This is not
compliant with the grammar in the SDP RFC (Appendix A) which requires at
least one character in the session name (which may be a blank). 

3. Appendix B.8 SDP "o=" Line. I had trouble with the wording "not strictly
necessary". Does this mean that the o= line is not always required or that
it is always required, but sometimes not very useful? Wouldn't it be best to
always be compliant with the SDP RFC, and just require that it be present?

4. Appendix C Summary of Augmented BNF. The *rule description seems to have
a problem with the < and > characters in <n>, etc. I see upside down ! and ?
(Spanish punctuation marks?). Don't know if this is just occurring at my end
or there is actually a problem with the document source.

My apologies if I have misunderstood any of these points.

Regards,
Jean Jervis
Nortel Networks

------_=_NextPart_001_01BFE1C3.FCCAB59E
Content-Type: text/html;
	charset="windows-1252"
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=3Dwindows-1252">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2651.65">
<TITLE>[SIP] RFC 2543bis - SDP Compliance, ABNF</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2 FACE=3D"Arial">Here are some comments on the SIP =
RFC2543bis Draft.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">1. Section 16.8 OPTIONS Request. The =
example SDP here does not have a c=3D line. I believe it needs one, =
either at the session level or as part of each media description in =
order to be compliant with the SDP spec.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">2. Appendix B.7 Subject and SDP =
&quot;s=3D&quot; Line. This section indicates that the s=3D line may be =
left empty for invitations to two-party sessions. This is not compliant =
with the grammar in the SDP RFC (Appendix A) which requires at least =
one character in the session name (which may be a blank). </FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">3. Appendix B.8 SDP &quot;o=3D&quot; =
Line. I had trouble with the wording &quot;not strictly =
necessary&quot;. Does this mean that the o=3D line is not always =
required or that it is always required, but sometimes not very useful? =
Wouldn't it be best to always be compliant with the SDP RFC, and just =
require that it be present?</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">4. Appendix C Summary of Augmented =
BNF. The *rule description seems to have&nbsp; a problem with the &lt; =
and &gt; characters in &lt;n&gt;, etc. I see upside down ! and ? =
(Spanish punctuation marks?). Don't know if this is just occurring at =
my end or there is actually a problem with the document =
source.</FONT></P>

<P><FONT SIZE=3D2 FACE=3D"Arial">My apologies if I have misunderstood =
any of these points.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Regards,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Jean Jervis</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Nortel Networks</FONT>
</P>

</BODY>
</HTML>
------_=_NextPart_001_01BFE1C3.FCCAB59E--


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From sip-admin@lists.bell-labs.com  Thu Jun 29 12:11:52 2000
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If anyone needs an ASN.1 Syntax Checker, or wants to update theirs, Version
5.2 of the OSS ASN.1 Syntax Checker is available as a free download from:
http://www.nokalva.com/products/checksyntax.html

Zipora Hartov




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From sip-admin@lists.bell-labs.com  Thu Jun 29 12:15:14 2000
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> If anyone needs an ASN.1 Syntax Checker, or wants to update theirs, Version
> 5.2 of the OSS ASN.1 Syntax Checker is available as a free download from:
> http://www.nokalva.com/products/checksyntax.html
> 
> Zipora Hartov
> 

Interesting, but I hope to never have to use ASN.1 with SIP :)
sean


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> SIP@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/sip
 


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From sip-admin@lists.bell-labs.com  Thu Jun 29 12:34:41 2000
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Date: Thu, 29 Jun 2000 09:33:31 -0700
To: "Adam B. Roach" <Adam.Roach@Ericsson.com>
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [SIP] Static Images
Cc: oran@cisco.com (David R. Oran),
        rfairlie@nuera.com (Fairlie-Cuninghame Robert),
        schulzrinne@cs.columbia.edu (Henning Schulzrinne),
        jdrosen@dynamicsoft.com (Jonathan Rosenberg), sip@lists.bell-labs.com
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Adam,

The "sprog" behavior reminds me of push advertising content.  I am 
incredibly alarmed at the idea of displaying "sprog", or playing "sprog", 
or continually prompting me if I want to see "sprog".  It's bad enough when 
I surf with cookie protection turned on.  IMO, This non-semantically 
defined rendering is more appropriate for Instant Messaging.

Now if you really want your UA to always display whatever content your peer 
barfs down to you, then I suggest we use headers, but create a convention 
that all the header names end with -Info.  Then, when you get the headers:

Sprog-Info: http://www.spam.com/annoying-animated.gif
Sprog-Info: http://www.spam.com/annoying.wav

Your UA can match against unknown *-Info headers and a) happily render 
them, b) prompt for them, or c) ignore them.

Alerting-Info:  special ring or display
From-Info:      CallerID photos, .vcfs, spoken names, etc.
Error-Info:     Dave's pointer to error messages
Context-Info:   stuff relevant to the call (presentation, web page, photo, 
etc.)
etc...

thanks,
-rohan


At 05:43 AM 6/29/00 , Adam B. Roach wrote:
> >So, in my view, if you wanted a something to be presented to the user during
> >alerting, we'd have headers like:
> >Alerting-Info: http://doggie-heaven.aol.com/~mydogskip/mug.jpg
> >Alerting-Info: ftp://doggie-heaven.aol.com/~mydogskip/woof-woof.mp3
> >Alerting-Info: rtsp://doggie-heaven.aol.com/~mydogskip/phew-phew.sml
> >From-Info: http://doggie-heaven.aol.com/~mydogskip/skip.vcf
> >
> >Depending on what the UA was capable of and the UA owner's preferences,
> >during alerting it could do one or more of:
> >a) display the picture of Skip
> >b) play Skip barking as the alerting sound
> >c) emit chemicals to simulate a very muddy unwashed skip walking into the
> >room.
> >
> >and make available via some UI function Skip's vCard information to allow it
> >to be added to an address book, or whatever.
>
>I think the point that Jonathan is trying to make (and one
>with which I agree) is that, for example, if you define a future
>extension for some other semantics than "alerting" or "from," new
>*headers* are suboptimal.
>
>Henning's argument holds *only* if the header itself has
>no associated semantics. In this case, it isn't true.
>"Display-Info" conveys the concept that "this is additional
>content to be rendered for the user." The purpose
>parameters are merely giving hints as to how this
>information would most effectively be conveyed.
>
>I'll demonstrate this point.
>
>Sprog-Info: http://www.xyz.com/image.jpg
>
>If your client didn't understand this new header type
>of "Sprog-Info," it would be ignored.
>
>Display-Info: http://www.xyz.com/image.jpg;purpose=sprog
>
>On the other hand, a client which understood the general
>"Display-Info" header, but had no idea what this new "sprog"
>thing was would have a much better recovery. It could either
>present the image (possibly in a box labeled "sprog"), or at
>least let the user know: "I have an image here labeled 'sprog.'
>Do you want to see it?")
>
>I see the second set of behavior as far preferable.
>
>--
>Adam Roach, Ericsson Inc. |  Ph: +1 972 583 7594 | 1010 E. Arapaho, MS L-04
>adam.roach@ericsson.com   | Fax: +1 972 669 0154 | Richardson, TX 75081 USA
>
>
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From sip-admin@lists.bell-labs.com  Thu Jun 29 12:57:44 2000
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Subject: Re: [SIP] Static Images
Cc: oran@cisco.com, rfairlie@nuera.com, schulzrinne@cs.columbia.edu,
        jdrosen@dynamicsoft.com, sip@lists.bell-labs.com
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> Adam,
> 
> The "sprog" behavior reminds me of push advertising content.  I am 
> incredibly alarmed at the idea of displaying "sprog", or playing "sprog", 
> or continually prompting me if I want to see "sprog".  It's bad enough when 
> I surf with cookie protection turned on.  IMO, This non-semantically 
> defined rendering is more appropriate for Instant Messaging.
> 
> Now if you really want your UA to always display whatever content your peer 
> barfs down to you, then I suggest we use headers, but create a convention 
> that all the header names end with -Info.  Then, when you get the headers:
> 
> Sprog-Info: http://www.spam.com/annoying-animated.gif
> Sprog-Info: http://www.spam.com/annoying.wav
> 
> Your UA can match against unknown *-Info headers and a) happily render 
> them, b) prompt for them, or c) ignore them.

It can quite easily do the same when matching unknown "purpose=" parameters
on a Display-Info: header. This is just a syntactical difference, not a
semantic one. You might argue that it can be more efficient to filter based
on the header name. You can also argue that it's easier to just look for
all the Display-Info: headers and not have to worry about possible namespace
collisions if someone adds a SIP extension header that uses a *-Info header.
Both approaches have problems. 

There seems to be some confusion here about what you are trying to convey
either with a header or with a parameter on that header. There seem
to be three pieces of information that are needed and only two are being
addressed.

1) A URL identifying some resource potentially to be accessed
2) The purpose of that resource
3) The mechanism for presenting that resource

What I would propose then is a compromise:

Use the header name to indicate the purpose of the resource as in:

Alerting-Info:  special ring or display
From-Info:      CallerID photos, .vcfs, spoken names, etc.
Error-Info:     Dave's pointer to error messages
Context-Info:   stuff relevant to the call (presentation, web page, photo, 
etc.)

Use a "disposition=" parameter to indicate the mechanism for presenting that
resource.

For example, a caller ID image that is meant to be rendered in parallel
with call setup would be

From-Info: http://www.cow.com/~bob/id.jpg ; disposition=inline 

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


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Date: Thu, 29 Jun 2000 10:05:40 -0700
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
From: Rohan Mahy <rohan@cisco.com>
Subject: Re: [SIP] Handling multiple Transfer Target
Cc: Helpdesk MIS <AdminLogs@globespan.net>, sip@lists.bell-labs.com
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At 12:09 AM 6/17/00 , Jonathan Rosenberg wrote:
> > From:  rafi
> >            Is it possible for the transferor to inform the transferee
> >  to try more than one transfer target serially or simultaneously?
> >  Probably one way to do is to include more than one Transfer-to-Header
> >  in the TRANSFER request message because transferee take decision
> > based on this field  (section 4.6 of draft
> > draft-sparks-sip-cc-transfer-00.txt).
> >  May be order of the of each name-addr / addr-spec in
> > Transfer-to-Header implies
> > their priority. Having more than one Transfer-to-Header header
> > is not allowed in the draft, draft-sparks-sip-cc-transfer-00.txt.
>
>I'm a bit reluctant to allow that. I'm not sure its needed. If you want
>the transferred party to try one address, then the next, then the next,
>etc., once each fails, you can do that without having multiple
>transfer-tos. Logic in the transferring party can simply send multiple
>transfer requests, one after a failure response from the previous.
>
>So, having multiple addresses is only useful for parallel transfers, but
>I doubt that this is useful. What happens if multiple acceptances come
>back? Does the transferred party speak to all, or just one? Was the
>TRANSFER successful if only some of the addresses in the transfer-to
>header yield successful results? Lots of complexity for no real value.

Rafi,

If you want message-level efficiency instead of byte level efficiency, you 
could bundle multiple TRANSFER requests into a single UDP datagram (size 
permitting).

thanks,
-rohan




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Hi,

I'm looking for information about dissipate.
Where can I download the latest revision of this library.
Best regards
Tony








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Tonino Villani (villani@coritel.it):

> I'm looking for information about dissipate.  Where can I download the
> latest revision of this library.

  The homepage is at http://www.div8.net/dissipate/

  If you have questions about it, please contact me directly.

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


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Folks,

Based on comments during last call, I have updated the reliable
provisional responses draft, and resubmitted. I have removed cumulative
PRACK from the document. Until it appears, you can find a new version
at:

http://www.cs.columbia.edu/~jdrosen/papers/draft-ietf-sip-100rel-02.txt

as this represents a technical change, I will not send to IESG until the
end of next week. This will give people a chance to comment on this
final version.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From sip-admin@lists.bell-labs.com  Fri Jun 30 18:56:08 2000
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Generally, it seems that the cost of generating globally unique
identifiers is low, but the cost of not doing it potentially high. Thus,
this seems to be a safe choice.

The tag is used to distinguish several "versions" of the same To/From
identifier, so it doesn't have to chang when the Call-ID changes. There
doesn't seem to be any advantage of making it unique within a call-id,
since you'd have to be sure that all "versions" with the same call-id
choose a different identifier - at that point, you're likely back at a
globally unique identifier, based on (say) the MAC address, CPU
identifier and who knows what else.

Why is this a problem?


> Vijay Gurbani wrote:
> Hi:
> 
> A collegue asked this question of me, for which I have an answer (see
> below), but would like to validate it.
> 
> Question: Why does the RFCbis (June 9, 2000) mandate that the tag be
> "globally unique" (Section 6.23)?  Can't it be unique within the Call-ID
> space it occupies?  The Call-ID already is "a globally unique
> identifier".  Why have 2 globally unique identifiers to identify a
> single call leg?
> 
> One reason I can think of is that if multiple UA servers are contacted
> for a call, this wording guarantees that each of the UA servers use an
> algorithm to generate a globally unique tag; otherwise, if each of them
> used, say, the time of day to generate the tag (and they happen to be
> in the same time zone etc. etc.), there could be a potential to generate
> the same tag at each UA server.


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I can't tell for sure from the PGP documentation, but rsa does seem to
be the default. (The documentation for version 6.5, the latest, almost
makes it sound like that this depends on the version of the software.) I
added the missing parameter.


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