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Subject: RE : [sipping] comment on draft-polk-sipping-location-requirements -01.txt
Date: Fri, 19 Dec 2003 12:32:22 +0100
Message-ID: <57F140DC5CDA10438C41674629DE0D8701602BB7@parmhs4.rd.francetelecom.fr>
Thread-Topic: [sipping] comment on draft-polk-sipping-location-requirements -01.txt
thread-index: AcPFsM9u//0JqP5LSPWafP5ZbNRk8gAco20Q
From: =?iso-8859-1?Q?PROUVOST_S=E9bastien_FTRD/DAC/ISS?= <sebastien.prouvost@francetelecom.com>
To: "Rosen, Brian" <Brian.Rosen@marconi.com>, <Mpierce1@aol.com>,
        <mhammer@cisco.com>
Cc: <sipping@ietf.org>, "Henning Schulzrinne" <hgs@cs.columbia.edu>,
        "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
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Two points here :
=20
First I doubt that there will be a single world wide set of requirements =
for emergency calls.

Second, a requirement listed in the original draft was for a proxy to be =
able to give its idea of the terminal location (originally by asserting =
the location given by the terminal). However, it may be easier for a =
proxy to insert its own info about the location of the terminal rather =
than to assert the info given by the terminal. It also let the =
possibility to the called party to get both pieces of information which =
is useful information as pointed out by Jonathan. Both pieces of =
information shall be passed on.


S=E9bastien Prouvost
France T=E9l=E9com R&D/DAC/CAR


-----Message d'origine-----
De : Rosen, Brian [mailto:Brian.Rosen@marconi.com]=20
Envoy=E9 : jeudi 18 d=E9cembre 2003 22:48
=C0 : 'Mpierce1@aol.com'; mhammer@cisco.com
Cc : sipping@ietf.org
Objet : RE: [sipping] comment on =
draft-polk-sipping-location-requirements -01.txt


And NENA is working on such requirements, and is starting to engage =
other=20
organizations in other parts of the world.  It has not been decided =
whether=20
there will be multiple, possibly conflicting requirements sent in from=20
different organizations, or whether an effort will be mounted to =
negotiate=20
a single set of requirements.  I favor the latter, because we don't have =

the expertise in the IETF to deal with conflicts if they were to arise.  =


I'd ask that anyone who knows people who should be involved (NENA=20
equivalents in other parts of the world), please contact me.=20

Brian


-----Original Message-----
From: Mpierce1@aol.com [mailto:Mpierce1@aol.com]
Sent: Thursday, December 18, 2003 3:47 PM
To: mhammer@cisco.com
Cc: sipping@ietf.org
Subject: Re: [sipping] comment on =
draft-polk-sipping-location-requirements
-01.txt


In a message dated 12/18/2003 3:05:36 PM Eastern Standard Time, =
mhammer@cisco.com writes:=20



It doesn't say you can't provide more than one type of measurement, does =
it?


What liability would a carrier assume if the handset using GPS provided =
a=20
location to 10 meters, and that got over-ridden by a carrier network =
using=20
TOA providing an accuracy of 100 meters, and the ambulance arrives too =
late?





More than one type of location is not the problem. I believe all the =
past discussion was that the "handset" and the "carrier" came up with =
conflicting information. If they were not conflicting, then why have =
two. If they are conflicting, that is a problem. If the handset gives a =
physical location to 10 meters, why would the carrier network bother to =
include a value of 100 meters accuracy, if the value from the end user =
were within that 100 meters? If it isn't, what should the PSAP do, send =
an ambulance to both?=20

If the handset gave a value within 10 meters and the carrier overwrote =
it with a value accurate to only 100 meters, there would be serious =
liability issues, especially if the requirement were for an accuracy of =
50 feet.=20

Like I said, NENA (or some such organization) needs to say what the =
requirements are (in the US). It is silly and hopeless for the IETF to =
try to agree on "requirements".=20

Mike=20

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip Use =
sip@ietf.org for new developments of core SIP

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: [Sipping] RFC 3665 on Session Initiation Protocol (SIP) Basic Call Flow Examples
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--NextPart


A new Request for Comments is now available in online RFC libraries.


        BCP 75
        RFC 3665

        Title:      Session Initiation Protocol (SIP) Basic Call Flow
                    Examples
        Author(s):  A. Johnston, S. Donovan, R. Sparks, C. Cunningham,
                    K. Summers
        Status:     Best Current Practice
        Date:       December 2003
        Mailbox:    alan.johnston@mci.com, sdonovan@dynamicsoft.com,
                    rsparks@dynamicsoft.com,
                    ccunningham@dynamicsoft.com,
                    kevin.summers@sonusnet.com
        Pages:      94
        Characters: 163159
        SeeAlso:    BCP 75

        I-D Tag:    draft-ietf-sipping-basic-call-flows-02.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3665.txt


This document gives examples of Session Initiation Protocol (SIP) call
flows.  Elements in these call flows include SIP User Agents and
Clients, SIP Proxy and Redirect Servers.  Scenarios include SIP
Registration and SIP session establishment.  Call flow diagrams and
message details are shown.  

This document is a product of the Session Initiation Proposal
Investigation Working Group of the IETF.

This document specifies an Internet Best Current Practices for the
Internet Community, and requests discussion and suggestions for
improvements.  Distribution of this memo is unlimited.

This announcement is sent to the IETF list and the RFC-DIST list.
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Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

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_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: [Sipping] BCP 76, RFC 3666 on Session Initiation Protocol (SIP) Public Switched Telephone Network (PSTN) Call Flows
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--NextPart


A new Request for Comments is now available in online RFC libraries.


        BCP 76
        RFC 3666

        Title:      Session Initiation Protocol (SIP)
                    Public Switched Telephone Network (PSTN) Call Flows
        Author(s):  A. Johnston, S. Donovan, R. Sparks, C. Cunningham,
                    K. Summers
        Status:     Best Current Practice
        Date:       December 2003
        Mailbox:    alan.johnston@mci.com, sdonovan@dynamicsoft.com,
                    rsparks@dynamicsoft.com,
                    ccunningham@dynamicsoft.com,
                    kevin.summers@sonusnet.com
        Pages:      118
        Characters: 200478
        SeeAlso:    BCP 76

        I-D Tag:    draft-ietf-sipping-pstn-call-flows-02.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3666.txt


This document contains best current practice examples of Session
Initiation Protocol (SIP) call flows showing interworking with the
Public Switched Telephone Network (PSTN).  Elements in these call
flows include SIP User Agents, SIP Proxy Servers, and PSTN
Gateways.  Scenarios include SIP to PSTN, PSTN to SIP, and PSTN to
PSTN via SIP.  PSTN telephony protocols are illustrated using ISDN
(Integrated Services Digital Network), ISUP (ISDN User Part), and FGB
(Feature Group B) circuit associated signaling.  PSTN calls are
illustrated using global telephone numbers from the PSTN and private
extensions served on by a PBX (Private Branch Exchange).  Call flow
diagrams and message details are shown.

This document is a product of the Session Initiation Proposal
Investigation Working Group of the IETF.

This document specifies an Internet Best Current Practices for the
Internet Community, and requests discussion and suggestions for
improvements.  Distribution of this memo is unlimited.

This announcement is sent to the IETF list and the RFC-DIST list.
Requests to be added to or deleted from the IETF distribution list
should be sent to IETF-REQUEST@IETF.ORG.  Requests to be
added to or deleted from the RFC-DIST distribution list should
be sent to RFC-DIST-REQUEST@RFC-EDITOR.ORG.

Details on obtaining RFCs via FTP or EMAIL may be obtained by sending
an EMAIL message to rfc-info@RFC-EDITOR.ORG with the message body 
help: ways_to_get_rfcs.  For example:

        To: rfc-info@RFC-EDITOR.ORG
        Subject: getting rfcs

        help: ways_to_get_rfcs

Requests for special distribution should be addressed to either the
author of the RFC in question, or to RFC-Manager@RFC-EDITOR.ORG.  Unless
specifically noted otherwise on the RFC itself, all RFCs are for
unlimited distribution.echo 
Submissions for Requests for Comments should be sent to
RFC-EDITOR@RFC-EDITOR.ORG.  Please consult RFC 2223, Instructions to RFC
Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...

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_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Wed Jan  7 06:08:51 2004
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From: "Peter Lewis" <peter.lewis@upperside.fr>
To: <sipping@ietf.org>
Date: Wed, 7 Jan 2004 12:08:14 +0100
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X-Priority: 3
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X-Mailer: Microsoft Outlook Express 5.00.2919.6600
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Subject: [Sipping] WiFi Voice Conference - Paris
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C'est un message de format MIME en plusieurs parties.

------=_NextPart_000_0249_01C3D516.ED802160
Content-Type: text/plain;
	charset="iso-8859-1"
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The WiFi Voice Conference will stand in Paris next May 25 to 28, 2004.
QoS, security, roaming, interoperability, architectures : distinguished =
experts=20
and key players in the field will address these questions with detailed =
and=20
technical explanations.=20
=20
More details at:
http://www.upperside.fr/wifivoice04/wifivoice04intro.htm

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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 5.00.2919.6307" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT size=3D2>The WiFi Voice Conference will stand in Paris next =
May 25 to=20
28, 2004.<BR>QoS, security, roaming, interoperability, architectures :=20
distinguished experts <BR>and key players in the field will address =
these=20
questions with detailed and <BR>technical explanations. =
<BR>&nbsp;<BR>More=20
details at:<BR></FONT><A=20
href=3D"http://www.upperside.fr/wifivoice04/wifivoice04intro.htm"><FONT=20
size=3D2>http://www.upperside.fr/wifivoice04/wifivoice04intro.htm</FONT><=
/A></DIV></BODY></HTML>

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_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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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
Proposal Investigation Working Group of the IETF.

	Title           : Best Current Practices for Third Party Call
                          Control in the Session Initiation Protocol
	Author(s)	: J. Rosenberg, J. Peterson, H. Schulzrinne, 
                          G. Camarillo
	Filename	: draft-ietf-sipping-3pcc-06.txt
	Pages		: 32
	Date		: 2004-1-6
	
Third party call control refers to the ability of one entity to create
a call in which communications is actually between other parties. Third
party call control is possible using the mechanisms specified within
the Session Initiation Protocol (SIP). However, there are several
possible approaches, each with different benefits and drawbacks. This
document discusses best current practices for the usage of the SIP for
third party call control.

A URL for this Internet-Draft is:
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_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Wed Jan  7 09:50:34 2004
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Date: Wed, 7 Jan 2004 15:43:44 +0100
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Thread-Topic: Completing URIs with port
Thread-Index: AcPVLKar8sGqGJigR9qH3I0AWuxTlA==
From: =?iso-8859-2?Q?Szel=E9nyi_Csaba?= <Csaba.Szelenyi@t-systems.co.hu>
To: <sipping@ietf.org>
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Hi,

According to RFC3261 chapter 19.1.4 two URIs are not the same if one of =
them contains no port description and the other one does contain the =
default port. E.g.:

   sip:bob@biloxi.com                   (can resolve to different ports)
   sip:bob@biloxi.com:5060

The CISCO 7960 completes the received URI with the ':5060' string in the =
following cases:
 - when the Contact URI is used as Request-URI in the next request (e.g. =
in case of redirection)
 - when the Record-Route is copied to the Route header in case of loose =
routing.

Is it legal? IMHO it is not, but I would like to be sure.

Thank you very mush for any input in advance!

Regards,

Szel=E9nyi Csaba
T-Systems RIC
mailto:Csaba.Szelenyi@t-systems.co.hu
+36 1 456 5507
+36 20 477 7778


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Wed Jan  7 13:46:45 2004
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Reply-To: <brett@broadsoft.com>
From: "Brett Tate" <brett@broadsoft.com>
To: "=?iso-8859-2?Q?'Szel=E9nyi_Csaba'?=" <Csaba.Szelenyi@t-systems.co.hu>,
        <sipping@ietf.org>
Subject: RE: [Sipping] Completing URIs with port
Date: Wed, 7 Jan 2004 13:40:41 -0500
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> According to RFC3261 chapter 19.1.4 two URIs are not the same 
> if one of them contains no port description and the other one 
> does contain the default port. E.g.:
> 
>    sip:bob@biloxi.com        (can resolve to different ports)
>    sip:bob@biloxi.com:5060
> 
> The CISCO 7960 completes the received URI with the ':5060' 
> string in the following cases:
>  - when the Contact URI is used as Request-URI in the next 
> request (e.g. in case of redirection)
>  - when the Record-Route is copied to the Route header in 
> case of loose routing.
> 
> Is it legal?

Within rfc2543, sip:bob@biloxi.com and
sip:bob@biloxi.com:5060 were considered equivalent.
Within rfc3261, they are no longer considered equivalent.
This is one of the various non backward compatibilities.

Because the adding of an extraneous port can break 
rfc3263 functionality within your example, you should
inquire when the product intends to become rfc3261
compliant concerning not extraneously adding port 5060.


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Date: Wed, 7 Jan 2004 15:19:28 -0800
Subject: Re: [Sipping] Completing URIs with port
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From: Rohan Mahy <rohan@cisco.com>
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The Cisco 7960 does not claim to be fully RFC-3261 compliant.

thanks,
-rohan


On Wednesday, January 7, 2004, at 06:43 AM, Szel=E9nyi Csaba wrote:

> Hi,
>
> According to RFC3261 chapter 19.1.4 two URIs are not the same if one=20=

> of them contains no port description and the other one does contain=20
> the default port. E.g.:
>
>    sip:bob@biloxi.com                   (can resolve to different=20
> ports)
>    sip:bob@biloxi.com:5060
>
> The CISCO 7960 completes the received URI with the ':5060' string in=20=

> the following cases:
>  - when the Contact URI is used as Request-URI in the next request=20
> (e.g. in case of redirection)
>  - when the Record-Route is copied to the Route header in case of=20
> loose routing.
>
> Is it legal? IMHO it is not, but I would like to be sure.
>
> Thank you very mush for any input in advance!
>
> Regards,
>
> Szel=E9nyi Csaba
> T-Systems RIC
> mailto:Csaba.Szelenyi@t-systems.co.hu
> +36 1 456 5507
> +36 20 477 7778
>
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Thu Jan  8 01:50:58 2004
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From: "Banibrata Dutta" <dutta@india.hp.com>
To: "Sipping Mailing-List" <sipping@ietf.org>
Date: Thu, 8 Jan 2004 12:13:35 +0530
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Subject: [Sipping] any info about "push-to-talk" service using SIP
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hi,

i've gone thru few old posts about the push-to-talk service
that i could find via google, and it the idea that i got was
it's "just another service" and need not be discussed on the
mailing list!! also, another view was that, the service itself
is not properly defined.

wanted to know if these 2 view-points have changed since then?
has there been any effort to standardize the implementation,
or definition of this service w.r.t. SIP ?

push-to-talk, as applicable to mobile devices (i.e.
the thing i am interested in), is a half-duplex mode of
communication where a mobile user can push-a-special-button
(keep it pressed) and after a go-ahead signal (i.e. after
seizing control of the floor) can speak out. this he does
after selecting one particular individual or a group in
his/her buddy-list. buddy-list is just like in case of 
text IM clients, which may be provisioned off-line/on-line.
the addressed party, i.e. individual or group will hear
what the originator said, via their spearker-phone mode
enabled mobile devices, if they are available and connected,
else not. i.e. a best effort delivery. this is like the
walkie-talkies. only one person can have floor-control at
a time. while a floor is under control of an individual,
others can only listen, but cannot speak. they must wait
for their turn.

having given this rough description of the service, may i
know, where we are ? i.e. any standardization / draft making
effort underway (current or expired or planned) ??

another thing that i am interested in finding out is, what 
role can SIP play in this, given the fact that the target
network is basically GSM/GPRS network, and not a pure SIP 
network?

thanks & regards,
Banibrata Dutta
  Hewlett-Packard ISO Pvt. Ltd,
  29, Cunningham Road,
  Bangalore - 560052, India.
  Tel: +91-80-2051796
 


_______________________________________________
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Use sip@ietf.org for new developments of core SIP



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From: "Garland Sharratt" <gsharratt@convedia.com>
To: "Banibrata Dutta" <dutta@india.hp.com>
Cc: "Sipping Mailing-List" <sipping@ietf.org>
Subject: RE: [Sipping] any info about "push-to-talk" service using SIP
Date: Wed, 7 Jan 2004 23:01:51 -0800
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The Open Mobile Alliance (www.openmobilealliance.org) has a large
standardization effort underway for PTT.  The project is called POC,
Push-to-talk Over Cellular.  They are using SIP over the data channel.

Regards,

Garland

-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of
Banibrata Dutta
Sent: Wednesday, January 07, 2004 22:44
To: Sipping Mailing-List
Subject: [Sipping] any info about "push-to-talk" service using SIP


hi,

i've gone thru few old posts about the push-to-talk service
that i could find via google, and it the idea that i got was
it's "just another service" and need not be discussed on the
mailing list!! also, another view was that, the service itself
is not properly defined.

wanted to know if these 2 view-points have changed since then?
has there been any effort to standardize the implementation,
or definition of this service w.r.t. SIP ?

push-to-talk, as applicable to mobile devices (i.e.
the thing i am interested in), is a half-duplex mode of
communication where a mobile user can push-a-special-button
(keep it pressed) and after a go-ahead signal (i.e. after
seizing control of the floor) can speak out. this he does
after selecting one particular individual or a group in
his/her buddy-list. buddy-list is just like in case of
text IM clients, which may be provisioned off-line/on-line.
the addressed party, i.e. individual or group will hear
what the originator said, via their spearker-phone mode
enabled mobile devices, if they are available and connected,
else not. i.e. a best effort delivery. this is like the
walkie-talkies. only one person can have floor-control at
a time. while a floor is under control of an individual,
others can only listen, but cannot speak. they must wait
for their turn.

having given this rough description of the service, may i
know, where we are ? i.e. any standardization / draft making
effort underway (current or expired or planned) ??

another thing that i am interested in finding out is, what
role can SIP play in this, given the fact that the target
network is basically GSM/GPRS network, and not a pure SIP
network?

thanks & regards,
Banibrata Dutta
  Hewlett-Packard ISO Pvt. Ltd,
  29, Cunningham Road,
  Bangalore - 560052, India.
  Tel: +91-80-2051796



_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Message-ID: <001101c3d5d0$12ac4720$0200000a@tkw>
From: "Pete Cordell" <pete@tech-know-ware.com>
To: "Cullen Jennings" <fluffy@cisco.com>,
        "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: "sipping" <sipping@ietf.org>
References: <BC16F709.2AD0D%fluffy@cisco.com>
Subject: Re: [Sipping] Problem with reg-event that needs immediate fix
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Jonathan, Cullen,

One concern I have about many XML schemas defined in this space is how they
will be extended for subsequent versions.  Has a strategy been defined?

The reason I mention it is that I read an article on xml.com about
versioning (http://www.xml.com/pub/a/2003/12/03/versioning.html) and it
proposes adding additional syntax to a schema to allow an instance
conforming to a new version of a schema to be correctly validated against a
previous version of the schema.

It uses <xs:any namespace="##other"> (the form used in a lot of the SIP
schemas) to allow for 'third party' extensions.

But, to allow the schema to be extended with new elements it suggests the
following format (I've tried to mark the relevant bits with ****s):

<s:complexType name="CallbackType">
 <s:sequence>
 <s:element name="callbackLocation" type="s:anyURI"
minOccurs="1" maxOccurs="1"/>
**** <s:element name="Extension" type="wscb:ExtensionType"
            minOccurs="0" maxOccurs="1"/>*****
 <s:any processContents="lax" minOccurs="0"
maxOccurs="unbounded" namespace="##other"/>
 </s:sequence>
 <s:anyAttribute/>
</s:complexType>

*****
<s:complexType name="ExtensionType">
 <s:sequence>
 <s:any processContents="lax" minOccurs="1"
maxOccurs="unbounded" namespace="##targetnamespace"/>
 </s:sequence>
******

 <s:anyAttribute/>
</s:complexType>


If an extension is added to the schema, (I think!) it would become:


<s:complexType name="CallbackType">
 <s:sequence>
 <s:element name="callbackLocation" type="s:anyURI"
minOccurs="1" maxOccurs="1"/>
<s:element name="Extension" type=******"wscb:MyExtension"*****
            minOccurs="0" maxOccurs="1"/>
 <s:any processContents="lax" minOccurs="0"
maxOccurs="unbounded" namespace="##other"/>
 </s:sequence>
 <s:anyAttribute/>
</s:complexType>

*****
<s:complexType name="MyExtension">
 <s:sequence>
  <s:element name="myNewElement" type="xs:string"/>
<s:element name="Extension" type="wscb:ExtensionType"
            minOccurs="0" maxOccurs="1"/>
</s:sequence>
</s:complexType>
*******

<s:complexType name="ExtensionType">
 <s:sequence>
 <s:any processContents="lax" minOccurs="1"
maxOccurs="unbounded" namespace="##targetnamespace"/>
 </s:sequence>
 <s:anyAttribute/>
</s:complexType>

An alternative would be to define all new extensions in a separate
namespace.  However, this becomes a bit imprecise because its not easy to
say where the new material plugs into the original schema.

So my question is this an issue, or have I just not found the right
document?!

Happy New Year,

Pete.

=============================================
Pete Cordell
Tech-Know-Ware
pete@tech-know-ware.com
http://www.tech-know-ware.com
+44 1473 635863
=============================================

----- Original Message -----
From: "Cullen Jennings" <fluffy@cisco.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>; "sipping"
<sipping@ietf.org>
Sent: 30 December 2003 17:34
Subject: Re: [Sipping] Problem with reg-event that needs immediate fix


>
> I would prefer to change the Schema to be right for this things we know
are
> coming even if that broke backwards compatibility.
>
> The more I look at the schema, I think the whole the things could be
> structured a little better for extensibility. I had the same comment on
some
> of the other schema that have been in sipish drafts. We should probably
make
> this part of our NITS review.
>
> Cullen
>
> On 12/18/03 9:13 AM, "Jonathan Rosenberg" <jdrosen@dynamicsoft.com> wrote:
>
> > While implementing the sipping reg-event draft, we ran into a serious
> > technical problem.
> >
> > The problem is that the specification has not left sufficient room for
> > extensibility within the XML schema. It allows for new information to
> > be added to the <registration> element (which describes the AOR), but
> > does not allow any additional information to be provided about a
> > contact. We ran into this problem because we needed to add path
> > information (ala rfc 3327) to the notifications, but this information
> > is per contact. There is no way to do this with the current
specification.
> >
> > The essence of the problem is that the contact element has, as
> > content, the actual URI. This makes it difficult to add other
> > information about the contact. A better approach would have been to
> > make the contact element a complex type, which would include an
> > element called <uri> that contains the uri.
> >
> > For example, today we have:
> >
> > <?xml version="1.0"?>
> >   <reginfo xmlns="urn:ietf:params:xml:ns:reginfo"
> >                version="1" state="partial">
> >     <registration aor="sip:joe@bar.com" id="a7" state="active">
> >       <contact id="76" state="active" event="registered"
> >                duration-registered="0">sip:joe@pc34.bar.com</contact>
> >     </registration>
> >   </reginfo>
> >
> > whereas it ought to have been:
> >
> > <?xml version="1.0"?>
> >   <reginfo xmlns="urn:ietf:params:xml:ns:reginfo"
> >                version="1" state="partial">
> >     <registration aor="sip:joe@bar.com" id="a7" state="active">
> >       <contact id="76" state="active" event="registered"
> >                duration-registered="0">
> >                   <uri>sip:joe@pc34.bar.com</uri>
> >       </contact>
> >     </registration>
> >   </reginfo>
> >
> >
> > The schema for the contact element could then allow for other children
> > in other namespaces, giving us the needed extensibility.
> >
> > Unfortunately, there is other per-contact data we might want to add in
> > the not-too-distant future, such as callee-caps information. That kind
> > of data would be very useful for presence applications, where the
> > presence server could obtain it from the registrar using reg-event.
> >
> > I believe that this problem is significant enough that we should try
> > and address it during auth48. For those of you not familiar with it,
> > auth48 is a brief period just before publication of the actual RFC
> > where the author can correct typos or minor other problems. It is
> > possible, with AD approval, to make more substantive changes as I am
> > suggesting (and indeed I already ran this by Allison and she is OK).
> > Of course, I still need group consensus on any change. So, here is my
> > proposal.
> >
> > PROPOSAL
> >
> > My proposal for fixing this, as outlined above, is to modify the XML
> > schema. The <contact> element will instead become a complex type. It
> > will have, as children, a sequence of <uri> followed by an optional
> > element from any other namespace. The <uri> element will contain the
> > actual contact URI.
> >
> > Unfortunately, this proposed solution above is not backwards
> > compatible with the current draft.
> >
> > I considered other solutions as well:
> >
> >
> > ALTERNATE 1
> >
> > Modify the schema to allow attributes in the contact element from
> > other namespaces. Adds some extensibility, but is very bad for more
> > complex data, such as path or callee-cap data. Using attributes for
> > that data will require us to define a lot of structure within each
> > attribute. This approach is backwards compatible, but not forwards
> > compatible. That is, an implementation compliant to the schema before
> > this change would reject documents created by an implementation
> > compliant to the schema after the change.
> >
> > ALTERNATE 2
> >
> > Modify the schema to contain a new element, child to the
> > <registration> element. This element would be called "alt-contact",
> > and would be defined as I suggest above for contact. Whenever new
> > information is needed about a contact, the "alt-contact" element would
> > be present, and contain that extension data. The document would also
> > still contain a contact element, repeating the URI and other
> > attributes. This would provide forwards and backwards compatibility,
> > at the expense of a horrible hack that would repeat a lot of
> > information in the document.
> >
> >
> > I found neither of the two alternatives very palatable, which is why
> > my proposal is to fix this now, and do it properly, while we still can.
> >
> > Please provide input on the proposal ASAP. I'd like to come to
> > consensus in the next week or so (before everyone disappears for
> > vacations).
> >
> > Thanks,
> > Jonathan R.
>
>
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Fri, 09 Jan 2004 01:47:43 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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To: Pete Cordell <pete@tech-know-ware.com>
CC: Cullen Jennings <fluffy@cisco.com>, sipping <sipping@ietf.org>
Subject: Re: [Sipping] Problem with reg-event that needs immediate fix
References: <BC16F709.2AD0D%fluffy@cisco.com> <001101c3d5d0$12ac4720$0200000a@tkw>
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inline.

Pete Cordell wrote:

> Jonathan, Cullen,
> 
> One concern I have about many XML schemas defined in this space is how they
> will be extended for subsequent versions.  Has a strategy been defined?

The general approach is to add <xs:any namespace="##other"> in places 
where we believe more data will be needed in the future. Nothing 
beyond that.

> 
> The reason I mention it is that I read an article on xml.com about
> versioning (http://www.xml.com/pub/a/2003/12/03/versioning.html) and it
> proposes adding additional syntax to a schema to allow an instance
> conforming to a new version of a schema to be correctly validated against a
> previous version of the schema.


Right.

> 
> It uses <xs:any namespace="##other"> (the form used in a lot of the SIP
> schemas) to allow for 'third party' extensions.
> 
> But, to allow the schema to be extended with new elements it suggests the
> following format (I've tried to mark the relevant bits with ****s):
> 
> <s:complexType name="CallbackType">
>  <s:sequence>
>  <s:element name="callbackLocation" type="s:anyURI"
> minOccurs="1" maxOccurs="1"/>
> **** <s:element name="Extension" type="wscb:ExtensionType"
>             minOccurs="0" maxOccurs="1"/>*****
>  <s:any processContents="lax" minOccurs="0"
> maxOccurs="unbounded" namespace="##other"/>
>  </s:sequence>
>  <s:anyAttribute/>
> </s:complexType>
> 
> *****
> <s:complexType name="ExtensionType">
>  <s:sequence>
>  <s:any processContents="lax" minOccurs="1"
> maxOccurs="unbounded" namespace="##targetnamespace"/>
>  </s:sequence>
> ******
> 
>  <s:anyAttribute/>
> </s:complexType>

I'm not sure why this is a better idea. The general technique of just 
adding an <xs:any> tag directly gets you the same result. The above 
approach merely wraps the extension element in another element called 
<Extension>. I dont see why thats useful.

> 
> 
> If an extension is added to the schema, (I think!) it would become:
> 
> 
> <s:complexType name="CallbackType">
>  <s:sequence>
>  <s:element name="callbackLocation" type="s:anyURI"
> minOccurs="1" maxOccurs="1"/>
> <s:element name="Extension" type=******"wscb:MyExtension"*****
>             minOccurs="0" maxOccurs="1"/>
>  <s:any processContents="lax" minOccurs="0"
> maxOccurs="unbounded" namespace="##other"/>
>  </s:sequence>
>  <s:anyAttribute/>
> </s:complexType>
> 
> *****
> <s:complexType name="MyExtension">
>  <s:sequence>
>   <s:element name="myNewElement" type="xs:string"/>
> <s:element name="Extension" type="wscb:ExtensionType"
>             minOccurs="0" maxOccurs="1"/>
> </s:sequence>
> </s:complexType>
> *******
> 
> <s:complexType name="ExtensionType">
>  <s:sequence>
>  <s:any processContents="lax" minOccurs="1"
> maxOccurs="unbounded" namespace="##targetnamespace"/>
>  </s:sequence>
>  <s:anyAttribute/>
> </s:complexType>

I dont see why you simply cant define the new schema with the new 
element anywhere you previously, in the old schema, allowed elements 
from any namespace.

> 
> An alternative would be to define all new extensions in a separate
> namespace.

Each extension normally has its own namespace.

>  However, this becomes a bit imprecise because its not easy to
> say where the new material plugs into the original schema.

I don't think we've planned specifically on how one might specify 
where to plug the new material into the existing schema. RPID, for 
example, which is an extension to PIDF, merely defines the new 
elements. Validation that they are placed in the right places is not 
done via the resulting schema validation, but through normative text 
in the doc. That may be OK.

If you want to validate, via schema, that the elements are placed in 
the right place, you would define a new schema with the new element as 
optional, as above. Again, I dont see the need for wrapping it in a 
new element called <extension>.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
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Message-ID: <004601c3d69f$e869d560$0200000a@tkw>
From: "Pete Cordell" <pete@tech-know-ware.com>
To: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
Cc: "Cullen Jennings" <fluffy@cisco.com>, "sipping" <sipping@ietf.org>
References: <BC16F709.2AD0D%fluffy@cisco.com> <001101c3d5d0$12ac4720$0200000a@tkw> <3FFE4E8F.7060803@dynamicsoft.com>
Subject: XML schema versioning strategy (Was: Re: [Sipping] Problem with reg-event that needs immediate fix)
Date: Fri, 9 Jan 2004 11:01:11 -0000
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As you know, the main issue of versioning is allowing devices working to =
the
old version of a schema to operate successfully when they receive instanc=
es
conforming to a new version of a schema.  (I'm assuming that we are not
expecting UAs to try and locate the latest version when a received schema
fails to validate.)

Unfortunately the use of <s:any namespace=3D"##other"/> doesn't get you t=
he
same result as the article proposes.

If the old version of the schema contains this fragment:

<s:element name=3D"callbackLocation" type=3D"s:anyURI"/>
  <s:any processContents=3D"lax" minOccurs=3D"0"
 maxOccurs=3D"unbounded" namespace=3D"##other"/>

An instance of a document conforming to a new version of the schema that
contains the following fragment would clearly not be valid against the ol=
d
schema if it contained the new element "newItem":

<s:element name=3D"callbackLocation" type=3D"s:anyURI"/>
<s:element name=3D"newItem" type=3D"newitem" minOccurs=3D"0"/>
  <s:any processContents=3D"lax" minOccurs=3D"0"
 maxOccurs=3D"unbounded" namespace=3D"##other"/>

At first sight, using the ##any option to xs:any might seem the way to go=
,
e.g.:

<s:element name=3D"callbackLocation" type=3D"s:anyURI"/>
  <s:any processContents=3D"lax" minOccurs=3D"0"
 maxOccurs=3D"unbounded" namespace=3D"##any"/>

However, this violates the Unique Particle Attribution Constraint, the
relevant bits being:

- Two non-group particles overlap if ... One is a wildcard and the other =
an
element declaration, and the {target namespace} of the element declaratio=
n,
or of any member of its =B7substitution group=B7, is =B7valid=B7 with res=
pect to the
{namespace constraint} of the wildcard.

- A content model will violate the unique attribution constraint if it
contains two particles which =B7overlap=B7 and may =B7validate=B7 adjacen=
t
information items and ...  the first has {min occurs} less than {max
occurs}.

The syntax in the original post was a way to get around this, allowing bo=
th
external additions, and new syntax in the existing schema without violati=
ng
this Unique Particle Attribution constraint.  (e.g. any variable cardinal=
ity
element would conflict with the <xs:any ##any> particle, but this wouldn'=
t
be the case if an optional element named "Extension" wrapped local
extensions.)

One option might appear to be to define the new element in a completely n=
ew
schema, in a completely new namespace.  The fragment in the existing sche=
ma
would then be modified to include a new reference to the element of the
form:

 <s:element name=3D"callbackLocation" type=3D"s:anyURI"/>
<s:element xmlns:v2=3D"urn:...:v2" ref=3D"v2:newItem" minOccurs=3D"0"/>
  <s:any processContents=3D"lax" minOccurs=3D"0"
 maxOccurs=3D"unbounded" namespace=3D"##other"/>

Unfortunately this seems to violate the Unique Particle Attribution
Constraint also; same reason, different namespace.

So without doing anything we are left with defining version 2 extensions =
in
separate schemas with different namespaces, and specifying where they sho=
uld
be plugged in using narrative (or possibly some form of XPath in
non-standard xs:appinfo sections).  This manual process seems a major sou=
rce
of potential errors and defeats some of the benefit of using schemas in t=
he
first place.

Adding in the extra syntax seems the correct way to go.  Much of the burd=
en
of adding the syntax could potentially be saved by defining suitable grou=
ps
(one for sequences, one for choices).

Pete.
--
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Pete Cordell
Tech-Know-Ware
pete@tech-know-ware.com
http://www.tech-know-ware.com
http://www.xml2cpp.com
+44 1473 635863
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

----- Original Message -----
From: "Jonathan Rosenberg" <jdrosen@dynamicsoft.com>
To: "Pete Cordell" <pete@tech-know-ware.com>
Cc: "Cullen Jennings" <fluffy@cisco.com>; "sipping" <sipping@ietf.org>
Sent: 09 January 2004 06:47
Subject: Re: [Sipping] Problem with reg-event that needs immediate fix


> inline.
>
> Pete Cordell wrote:
>
> > Jonathan, Cullen,
> >
> > One concern I have about many XML schemas defined in this space is ho=
w
they
> > will be extended for subsequent versions.  Has a strategy been define=
d?
>
> The general approach is to add <xs:any namespace=3D"##other"> in places
> where we believe more data will be needed in the future. Nothing
> beyond that.
>
> >
> > The reason I mention it is that I read an article on xml.com about
> > versioning (http://www.xml.com/pub/a/2003/12/03/versioning.html) and =
it
> > proposes adding additional syntax to a schema to allow an instance
> > conforming to a new version of a schema to be correctly validated
against a
> > previous version of the schema.
>
>
> Right.
>
> >
> > It uses <xs:any namespace=3D"##other"> (the form used in a lot of the=
 SIP
> > schemas) to allow for 'third party' extensions.
> >
> > But, to allow the schema to be extended with new elements it suggests
the
> > following format (I've tried to mark the relevant bits with ****s):
> >
> > <s:complexType name=3D"CallbackType">
> >  <s:sequence>
> >  <s:element name=3D"callbackLocation" type=3D"s:anyURI"
> > minOccurs=3D"1" maxOccurs=3D"1"/>
> > **** <s:element name=3D"Extension" type=3D"wscb:ExtensionType"
> >             minOccurs=3D"0" maxOccurs=3D"1"/>*****
> >  <s:any processContents=3D"lax" minOccurs=3D"0"
> > maxOccurs=3D"unbounded" namespace=3D"##other"/>
> >  </s:sequence>
> >  <s:anyAttribute/>
> > </s:complexType>
> >
> > *****
> > <s:complexType name=3D"ExtensionType">
> >  <s:sequence>
> >  <s:any processContents=3D"lax" minOccurs=3D"1"
> > maxOccurs=3D"unbounded" namespace=3D"##targetnamespace"/>
> >  </s:sequence>
> > ******
> >
> >  <s:anyAttribute/>
> > </s:complexType>
>
> I'm not sure why this is a better idea. The general technique of just
> adding an <xs:any> tag directly gets you the same result. The above
> approach merely wraps the extension element in another element called
> <Extension>. I dont see why thats useful.
>
> >
> >
> > If an extension is added to the schema, (I think!) it would become:
> >
> >
> > <s:complexType name=3D"CallbackType">
> >  <s:sequence>
> >  <s:element name=3D"callbackLocation" type=3D"s:anyURI"
> > minOccurs=3D"1" maxOccurs=3D"1"/>
> > <s:element name=3D"Extension" type=3D******"wscb:MyExtension"*****
> >             minOccurs=3D"0" maxOccurs=3D"1"/>
> >  <s:any processContents=3D"lax" minOccurs=3D"0"
> > maxOccurs=3D"unbounded" namespace=3D"##other"/>
> >  </s:sequence>
> >  <s:anyAttribute/>
> > </s:complexType>
> >
> > *****
> > <s:complexType name=3D"MyExtension">
> >  <s:sequence>
> >   <s:element name=3D"myNewElement" type=3D"xs:string"/>
> > <s:element name=3D"Extension" type=3D"wscb:ExtensionType"
> >             minOccurs=3D"0" maxOccurs=3D"1"/>
> > </s:sequence>
> > </s:complexType>
> > *******
> >
> > <s:complexType name=3D"ExtensionType">
> >  <s:sequence>
> >  <s:any processContents=3D"lax" minOccurs=3D"1"
> > maxOccurs=3D"unbounded" namespace=3D"##targetnamespace"/>
> >  </s:sequence>
> >  <s:anyAttribute/>
> > </s:complexType>
>
> I dont see why you simply cant define the new schema with the new
> element anywhere you previously, in the old schema, allowed elements
> from any namespace.
>
> >
> > An alternative would be to define all new extensions in a separate
> > namespace.
>
> Each extension normally has its own namespace.
>
> >  However, this becomes a bit imprecise because its not easy to
> > say where the new material plugs into the original schema.
>
> I don't think we've planned specifically on how one might specify
> where to plug the new material into the existing schema. RPID, for
> example, which is an extension to PIDF, merely defines the new
> elements. Validation that they are placed in the right places is not
> done via the resulting schema validation, but through normative text
> in the doc. That may be OK.
>
> If you want to validate, via schema, that the elements are placed in
> the right place, you would define a new schema with the new element as
> optional, as above. Again, I dont see the need for wrapping it in a
> new element called <extension>.
>
> -Jonathan R.
>
> --
> Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
> Chief Technology Officer                    Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.jdrosen.net                      PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Hi All,

It seems to me, that the branch parameter is case-sensitive (the magic =
cookie contains upper and lower case characters). I could find only the =
following paragraph regarding the case-sensitivity of the branch =
parameter in rfc3261:

7.3.1 Header Field Format
...
   When comparing header fields, field names are always case-
   insensitive.  Unless otherwise stated in the definition of a
   particular header field, field values, parameter names, and parameter
   values are case-insensitive.  Tokens are always case-insensitive.
   Unless specified otherwise, values expressed as quoted strings are
   case-sensitive.  For example,

Neither chapter 8.1.1.7, nor chapter 20.42 states anything regarding the =
case-sensitivity of branch. So, according to rfc3261, the branch is case =
insensitive!=20

Is the Via branch parameter case sensitive?

Thank you very much in anticipation for any input regarding this topic!

Regards,
Csaba


Szel=E9nyi Csaba
T-Systems RIC
mailto:Csaba.Szelenyi@t-systems.co.hu
+36 1 456 5507
+36 20 477 7778


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Subject: RE: [Sipping] Case-sensitivty of the branch parameter
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Regards,
Nataraju A.B.=20

-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf =
Of Szel=E9nyi Csaba
Sent: Tuesday, January 13, 2004 4:35 PM
To: sipping@ietf.org
Subject: [Sipping] Case-sensitivty of the branch parameter

Hi All,

It seems to me, that the branch parameter is case-sensitive (the magic =
cookie contains upper and lower case characters). I could find only the =
following paragraph regarding the case-sensitivity of the branch =
parameter in rfc3261:

7.3.1 Header Field Format
...
   When comparing header fields, field names are always case-
   insensitive.  Unless otherwise stated in the definition of a
   particular header field, field values, parameter names, and parameter
   values are case-insensitive.  Tokens are always case-insensitive.
   Unless specified otherwise, values expressed as quoted strings are
   case-sensitive.  For example,

Neither chapter 8.1.1.7, nor chapter 20.42 states anything regarding the =
case-sensitivity of branch. So, according to rfc3261, the branch is case =
insensitive!=20

Is the Via branch parameter case sensitive?

 [ABN] Yes Branch parameter MUST be case sensitive. It's not been =
explicitly   mentioned in any section. I guess it should have been =
mentioned in some section about the case sensitiveness...

Thank you very much in anticipation for any input regarding this topic!

Regards,
Csaba


Szel=E9nyi Csaba
T-Systems RIC
mailto:Csaba.Szelenyi@t-systems.co.hu
+36 1 456 5507
+36 20 477 7778


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

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<P ALIGN=3DLEFT><SPAN LANG=3D"en"></SPAN><A NAME=3D""><SPAN =
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New">Regards,</FONT></SPAN></A></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en"><FONT SIZE=3D2 FACE=3D"Courier =
New">Nataraju A.B. </FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">-----Original Message-----<BR>
From: sipping-admin@ietf.org [<A =
HREF=3D"mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</A>]=
 On Behalf Of Szel=E9nyi Csaba<BR>
Sent: Tuesday, January 13, 2004 4:35 PM<BR>
To: sipping@ietf.org<BR>
Subject: [Sipping] Case-sensitivty of the branch =
parameter</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Hi All,</FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">It seems to me, that the branch parameter is case-sensitive (the =
magic cookie contains upper and lower case characters). I could find =
only the following paragraph regarding the case-sensitivity of the =
branch parameter in rfc3261:</FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">7.3.1 Header Field Format</FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">...</FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; When comparing header fields, field names are always =
case-</FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; insensitive.&nbsp; Unless otherwise stated in the =
definition of a</FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; particular header field, field values, parameter =
names, and parameter</FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; values are case-insensitive.&nbsp; Tokens are always =
case-insensitive.</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; Unless specified otherwise, values expressed as quoted =
strings are</FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;&nbsp; case-sensitive.&nbsp; For example,</FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Neither chapter 8.1.1.7, nor chapter 20.42 states anything =
regarding the case-sensitivity of branch. So, according to rfc3261, the =
branch is case insensitive! </FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Is the Via branch parameter case sensitive?</FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">&nbsp;[ABN]</FONT></SPAN><SPAN LANG=3D"en-us"><B> <FONT =
COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Courier =
New">Yes</FONT></B></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New"> Branch parameter</FONT></SPAN><SPAN =
LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier =
New">MUST</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 =
FACE=3D"Courier New">be case sensitive.</FONT></SPAN><SPAN =
LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier =
New">It's</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New"> not been explicitly&nbsp;&nbsp; mentioned in any =
section.</FONT></SPAN><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Courier New"> I guess it should have been mentioned in some =
section about</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 =
FACE=3D"Courier New">the</FONT></SPAN><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Courier New"> case sensitiveness</FONT></SPAN><SPAN =
LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">...</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Thank you very much in anticipation for any input regarding this =
topic!</FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Regards,</FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Csaba</FONT></SPAN></P>
<BR>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Szel=E9nyi Csaba</FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">T-Systems RIC</FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New"><A =
HREF=3D"mailto:Csaba.Szelenyi@t-systems.co.hu">mailto:Csaba.Szelenyi@t-sy=
stems.co.hu</A></FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">+36 1 456 5507</FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">+36 20 477 7778</FONT></SPAN></P>
<BR>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">_______________________________________________</FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Sipping mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sipping">https://www1.ietf=
.org/mailman/listinfo/sipping</A></FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">This list is for NEW development of the application of =
SIP</FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Use sip-implementors@cs.columbia.edu for questions on current =
sip</FONT></SPAN></P>

<P ALIGN=3DLEFT><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Courier =
New">Use sip@ietf.org for new developments of core SIP</FONT></SPAN></P>

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_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: Steve Pellegrino <Steve.Pellegrino@ulticom.com>
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To: nataraju.alilaghatta@wipro.com
CC: Csaba.Szelenyi@t-systems.co.hu, sipping@ietf.org
Subject: Re: [Sipping] Case-sensitivty of the branch parameter
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Actually, the branch parameter is case insensitive.  This has been=20
logged as bug 661:
http://www.sipwg.org/sipwg/show_bug.cgi?id=3D661

Regards,
Steve


nataraju.alilaghatta@wipro.com wrote:

>
>
> Regards,
>
> Nataraju A.B.
>
> -----Original Message-----
> From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf=20
> Of Szel=E9nyi Csaba
> Sent: Tuesday, January 13, 2004 4:35 PM
> To: sipping@ietf.org
> Subject: [Sipping] Case-sensitivty of the branch parameter
>
> Hi All,
>
> It seems to me, that the branch parameter is case-sensitive (the magic=20
> cookie contains upper and lower case characters). I could find only=20
> the following paragraph regarding the case-sensitivity of the branch=20
> parameter in rfc3261:
>
> 7.3.1 Header Field Format
>
> ...
>
>    When comparing header fields, field names are always case-
>
>    insensitive.  Unless otherwise stated in the definition of a
>
>    particular header field, field values, parameter names, and paramete=
r
>
>    values are case-insensitive.  Tokens are always case-insensitive.
>
>    Unless specified otherwise, values expressed as quoted strings are
>
>    case-sensitive.  For example,
>
> Neither chapter 8.1.1.7, nor chapter 20.42 states anything regarding=20
> the case-sensitivity of branch. So, according to rfc3261, the branch=20
> is case insensitive!
>
> Is the Via branch parameter case sensitive?
>
>  [ABN] Yes Branch parameter MUST be case sensitive. It's not been=20
> explicitly   mentioned in any section. I guess it should have been=20
> mentioned in some section about the case sensitiveness...
>
> Thank you very much in anticipation for any input regarding this topic!
>
> Regards,
>
> Csaba
>
>
> Szel=E9nyi Csaba
>
> T-Systems RIC
>
> mailto:Csaba.Szelenyi@t-systems.co.hu
>
> +36 1 456 5507
>
> +36 20 477 7778
>
>
> _______________________________________________
>
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>
> This list is for NEW development of the application of SIP
>
> Use sip-implementors@cs.columbia.edu for questions on current sip
>
> Use sip@ietf.org for new developments of core SIP
>

--------------040202040804050904020200
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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
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Actually, the branch parameter is case insensitive.&nbsp; This has been
logged as bug 661:<br>
<a class="moz-txt-link-freetext" href="http://www.sipwg.org/sipwg/show_bug.cgi?id=661">http://www.sipwg.org/sipwg/show_bug.cgi?id=661</a><br>
<br>
Regards,<br>
Steve<br>
<br>
<br>
<a class="moz-txt-link-abbreviated" href="mailto:nataraju.alilaghatta@wipro.com">nataraju.alilaghatta@wipro.com</a> wrote:<br>
<blockquote type="cite"
 cite="mid10C4348A1BA43A4FA8D5B767E052CEADDBEDAC@blr-ec-msg04.wipro.com">
  <meta http-equiv="Content-Type" content="text/html; ">
  <meta name="Generator" content="MS Exchange Server version 6.0.6487.1">
  <title>RE: [Sipping] Case-sensitivty of the branch parameter</title>
<!-- Converted from text/rtf format -->
  <br>
  <br>
  <p align="left"><a name=""><span lang="en"><font size="2"
 face="Courier New">Regards,</font></span></a></p>
  <p align="left"><span lang="en"><font size="2" face="Courier New">Nataraju
A.B. </font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">-----Original
Message-----<br>
From: <a class="moz-txt-link-abbreviated" href="mailto:sipping-admin@ietf.org">sipping-admin@ietf.org</a> [<a href="mailto:sipping-admin@ietf.org">mailto:sipping-admin@ietf.org</a>]
On Behalf Of Szel&eacute;nyi Csaba<br>
Sent: Tuesday, January 13, 2004 4:35 PM<br>
To: <a class="moz-txt-link-abbreviated" href="mailto:sipping@ietf.org">sipping@ietf.org</a><br>
Subject: [Sipping] Case-sensitivty of the branch parameter</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">Hi
All,</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">It
seems to me, that the branch parameter is case-sensitive (the magic
cookie contains upper and lower case characters). I could find only the
following paragraph regarding the case-sensitivity of the branch
parameter in rfc3261:</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">7.3.1
Header Field Format</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">...</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">&nbsp;&nbsp;
When comparing header fields, field names are always case-</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">&nbsp;&nbsp;
insensitive.&nbsp; Unless otherwise stated in the definition of a</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">&nbsp;&nbsp;
particular header field, field values, parameter names, and parameter</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">&nbsp;&nbsp;
values are case-insensitive.&nbsp; Tokens are always case-insensitive.</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">&nbsp;&nbsp;
Unless specified otherwise, values expressed as quoted strings are</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">&nbsp;&nbsp;
case-sensitive.&nbsp; For example,</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">Neither
chapter 8.1.1.7, nor chapter 20.42 states anything regarding the
case-sensitivity of branch. So, according to rfc3261, the branch is
case insensitive! </font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">Is
the Via branch parameter case sensitive?</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">&nbsp;[ABN]</font></span><span
 lang="en-us"><b> <font color="#0000ff" size="2" face="Courier New">Yes</font></b></span><span
 lang="en-us"><font size="2" face="Courier New"> Branch parameter</font></span><span
 lang="en-us"> <font size="2" face="Courier New">MUST</font></span><span
 lang="en-us"> <font size="2" face="Courier New">be case sensitive.</font></span><span
 lang="en-us"> <font size="2" face="Courier New">It's</font></span><span
 lang="en-us"><font size="2" face="Courier New"> not been explicitly&nbsp;&nbsp;
mentioned in any section.</font></span><span lang="en-us"><font size="2"
 face="Courier New"> I guess it should have been mentioned in some
section about</font></span><span lang="en-us"> <font size="2"
 face="Courier New">the</font></span><span lang="en-us"><font size="2"
 face="Courier New"> case sensitiveness</font></span><span lang="en-us"><font
 size="2" face="Courier New">...</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">Thank
you very much in anticipation for any input regarding this topic!</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">Regards,</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">Csaba</font></span></p>
  <br>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">Szel&eacute;nyi
Csaba</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">T-Systems
RIC</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New"><a
 href="mailto:Csaba.Szelenyi@t-systems.co.hu">mailto:Csaba.Szelenyi@t-systems.co.hu</a></font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">+36
1 456 5507</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">+36
20 477 7778</font></span></p>
  <br>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">_______________________________________________</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">Sipping
mailing list&nbsp; <a href="https://www1.ietf.org/mailman/listinfo/sipping">https://www1.ietf.org/mailman/listinfo/sipping</a></font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">This
list is for NEW development of the application of SIP</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">Use
<a class="moz-txt-link-abbreviated" href="mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.columbia.edu</a> for questions on current sip</font></span></p>
  <p align="left"><span lang="en-us"><font size="2" face="Courier New">Use
<a class="moz-txt-link-abbreviated" href="mailto:sip@ietf.org">sip@ietf.org</a> for new developments of core SIP</font></span></p>
</blockquote>
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.

	Title		: Extending the Session Initiation Protocol Reason Header for Preemption Events
	Author(s)	: J. Polk
	Filename	: draft-ietf-sipping-reason-header-for-preemption-00.txt
	Pages		: 15
	Date		: 2004-1-13
	
This document proposes an IANA Registration extension to the Session
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Method Request [2] as a result of a session preemption event either 
at a user agent (UA), or somewhere in the network using RSVP [3].  
This document does not attempt to address routers failing in the 
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Date: Wed, 14 Jan 2004 11:37:58 +0900
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Subject: [Sipping] SIP DTMF
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Hello all, 

Who can tell me whether there is a RFC or I-D on DTMF in SIP which is still not expired?
I remember someone proposed using SIP INFO method for DTMF one or two years ago.
Is the DTMF relay implemented only by RTP now?


Regards,
Wendy

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Tue Jan 13 22:59:23 2004
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There is also RFC 2833.

The app interaction framework:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-app-interaction-framework-00.txt

describes the framework for dtmf and other stimulus signaling, and 
KPML, as indicated, is used in conjunction with this framework.

-Jonathan R.

sreeram.kanumuri@wipro.com wrote:

> Wendy,
> 
> There was a discussion in the list some time back and it was decided
> that SIP-INFO method should not be used to 
> Send the DTMF info.
> 
> It was proposed that it is better to you kpxml for sending DTMF tones.
> 
> Please look in to the SIP-Kpxml draft for sending DTMF tomes in-midcall
> processing and all.... 
> 
> 
> Regards,
> Sreeram
> 
> =====================================================
> Sreeram Kanumuri                Plot No 72, Keonics Electronics City
> 			              Hosur Main Road
> Wipro Technologies              Bangalore -29 INDIA
> sreeram.kanumuri@wipro.com      Phone: 8520408 Ex:3208
> skanumury@yahoo.com             ESN  :
> www.wipro.com                   Mobile:
> ======================================================
> 
> 
> -----Original Message-----
> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of wendy
> Sent: Wednesday, January 14, 2004 8:08 AM
> To: sip@ietf.org
> Cc: sipping@ietf.org
> Subject: [Sip] SIP DTMF
> 
> 
> Hello all, 
> 
> Who can tell me whether there is a RFC or I-D on DTMF in SIP which is
> still not expired? I remember someone proposed using SIP INFO method for
> DTMF one or two years ago. Is the DTMF relay implemented only by RTP
> now?
> 
> 
> Regards,
> Wendy
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip Use
> sipping@ietf.org for new developments on the application of sip
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Wendy,

There was a discussion in the list some time back and it was decided
that SIP-INFO method should not be used to=20
Send the DTMF info.

It was proposed that it is better to you kpxml for sending DTMF tones.

Please look in to the SIP-Kpxml draft for sending DTMF tomes in-midcall
processing and all....=20


Regards,
Sreeram

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D
Sreeram Kanumuri                Plot No 72, Keonics Electronics City
			              Hosur Main Road
Wipro Technologies              Bangalore -29 INDIA
sreeram.kanumuri@wipro.com      Phone: 8520408 Ex:3208
skanumury@yahoo.com             ESN  :
www.wipro.com                   Mobile:
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D


-----Original Message-----
From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of wendy
Sent: Wednesday, January 14, 2004 8:08 AM
To: sip@ietf.org
Cc: sipping@ietf.org
Subject: [Sip] SIP DTMF


Hello all,=20

Who can tell me whether there is a RFC or I-D on DTMF in SIP which is
still not expired? I remember someone proposed using SIP INFO method for
DTMF one or two years ago. Is the DTMF relay implemented only by RTP
now?


Regards,
Wendy

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip Use
sipping@ietf.org for new developments on the application of sip

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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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 Proposal Investigation Working Group of the IETF.

	Title		: The Early Session Disposition Type for the Session Initiation Protocol (SIP)
	Author(s)	: G. Camarillo
	Filename	: draft-ietf-sipping-early-disposition-01.txt
	Pages		: 10
	Date		: 2004-1-14
	
This document defines a new disposition type (early-session) for the
Content-Disposition header field in SIP. The treatment of
'early-session' bodies is similar to the treatment of 'session'
bodies. That is, they follow the offer/answer model. Their only
difference is that session descriptions whose disposition type is
'early-session' are used to establish early media sessions within
early dialogs, as opposed to regular sessions within regular dialogs.

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_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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User-Agent: Microsoft-Entourage/10.1.4.030702.0
Date: Sun, 18 Jan 2004 10:17:51 -0800
Subject: Re: [Sipping] Re: [Sip] SIP DTMF
From: Cullen Jennings <fluffy@cisco.com>
To: <sipping@ietf.org>
CC: <xyxie@udtech.com.cn>, <sreeram.kanumuri@wipro.com>,
        Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Message-ID: <BC300DCF.2D0EC%fluffy@cisco.com>
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I have talked to the authors of the various proposal for doing DTMF. For the
case where it is not in the media (ie 2833), I believe the only proposal
that the current authors plan to keep moving forward is the
App-Interaction/KPML stuff. I don't believe there is any intention to
resubmit the DTMF in INFO or DTMF over SUB/NOT proposals.

Cullen

On 1/13/04 7:34 PM, "Jonathan Rosenberg" <jdrosen@dynamicsoft.com> wrote:

> There is also RFC 2833.
> 
> The app interaction framework:
> http://www.ietf.org/internet-drafts/draft-ietf-sipping-app-interaction-framewo
> rk-00.txt
> 
> describes the framework for dtmf and other stimulus signaling, and
> KPML, as indicated, is used in conjunction with this framework.
> 
> -Jonathan R.
> 
> sreeram.kanumuri@wipro.com wrote:
> 
>> Wendy,
>> 
>> There was a discussion in the list some time back and it was decided
>> that SIP-INFO method should not be used to
>> Send the DTMF info.
>> 
>> It was proposed that it is better to you kpxml for sending DTMF tones.
>> 
>> Please look in to the SIP-Kpxml draft for sending DTMF tomes in-midcall
>> processing and all....
>> 
>> 
>> Regards,
>> Sreeram
>> 
>> =====================================================
>> Sreeram Kanumuri                Plot No 72, Keonics Electronics City
>>              Hosur Main Road
>> Wipro Technologies              Bangalore -29 INDIA
>> sreeram.kanumuri@wipro.com      Phone: 8520408 Ex:3208
>> skanumury@yahoo.com             ESN  :
>> www.wipro.com                   Mobile:
>> ======================================================
>> 
>> 
>> -----Original Message-----
>> From: sip-admin@ietf.org [mailto:sip-admin@ietf.org] On Behalf Of wendy
>> Sent: Wednesday, January 14, 2004 8:08 AM
>> To: sip@ietf.org
>> Cc: sipping@ietf.org
>> Subject: [Sip] SIP DTMF
>> 
>> 
>> Hello all, 
>> 
>> Who can tell me whether there is a RFC or I-D on DTMF in SIP which is
>> still not expired? I remember someone proposed using SIP INFO method for
>> DTMF one or two years ago. Is the DTMF relay implemented only by RTP
>> now?
>> 
>> 
>> Regards,
>> Wendy
>> 
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip Use
>> sipping@ietf.org for new developments on the application of sip
>> 
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>> 


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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------------------  Virus Warning Message (on vs02)

tadxlgtfbr.exe is removed from here because it contains a virus.

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The Rockwell Collins Virus Team

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----------354818024557883--


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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This is Something to Think About.

What would we need to do to take advantage of the SACRED work in order to
secure SIP systems? How will this relate to our identity and authentication
models?

--
Dean

-----Original Message-----
From: owner-ietf-announce@ietf.org [mailto:owner-ietf-announce@ietf.org] On
Behalf Of The IESG
Sent: Friday, January 09, 2004 10:07 AM
To: IETF-Announce:
Cc: Internet Architecture Board; RFC Editor; ietf-sacred@imc.org
Subject: Protocol Action: 'Securely Available Credentials Protocol' to
Proposed Standard 


The IESG has approved the following documents:

- 'Securely Available Credentials Protocol '
   <draft-ietf-sacred-protocol-bss-09.txt> as a Proposed Standard
- 'Securely Available Credentials - Credential Server Framework '
   <draft-ietf-sacred-framework-07.txt> as an Informational RFC

These documents are products of the Securely Available Credentials Working 
Group. 

The IESG contact persons are Steve Bellovin and Russ Housley.

Technical Summary

Certificates and other cryptographic credentials are ungainly things, not 
suitable to human memorization.  While smart cards and the like are one 
possible solution, they're not widely deployed or used for general-purpose
computing.  The SACRED protocol permits remote storage and retrieval of 
such credentials.

Working Group Summary

There was working group consensus on this document.

Protocol Quality

This protocol was reviewed for the IESG by Steve Bellovin




_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Tue Jan 20 10:07:55 2004
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From: Internet-Drafts@ietf.org
Reply-to: Internet-Drafts@ietf.org
Date: Tue, 20 Jan 2004 10:05:18 -0500
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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 Proposal Investigation Working Group of the IETF.

	Title		: Interworking between SIP and QSIG
	Author(s)	: J. Elwell
	Filename	: draft-ietf-sipping-qsig2sip-04.txt
	Pages		: 61
	Date		: 2004-1-19
	
This document specifies interworking between the Session Initiation 
   Protocol (SIP) and QSIG within corporate telecommunication networks 
   (also known as enterprise networks). SIP is an Internet application-
   layer control (signalling) protocol for creating, modifying, and 
   terminating sessions with one or more participants. These sessions 
   include, in particular, telephone calls. QSIG is a signalling 
   protocol for creating, modifying and terminating circuit-switched 
   calls, in particular telephone calls, within Private Integrated 
   Services Networks (PISNs). QSIG is specified in a number of ECMA 
   Standards and published also as ISO/IEC standards.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-qsig2sip-04.txt

To remove yourself from the IETF Announcement list, send a message to 
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Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-sipping-qsig2sip-04.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:
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In the body type:
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_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Tue Jan 20 15:07:02 2004
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From: The IESG <iesg-secretary@ietf.org>
To: IETF-Announce:;
Cc: Internet Architecture Board <iab@iab.org>,
        RFC Editor <rfc-editor@rfc-editor.org>, <sipping@ietf.org>
Message-Id: <E1Aj26Z-0001n4-Bc@asgard.ietf.org>
Date: Tue, 20 Jan 2004 15:04:07 -0500
Subject: [Sipping] Protocol Action: 'Best Current Practices for Third Party
 Call Control in the Session Initiation Protocol' to BCP
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The IESG has approved the following document:

- 'Best Current Practices for Third Party Call Control in the Session 
   Initiation Protocol '
   <draft-ietf-sipping-3pcc-06.txt> as a BCP

This document is the product of the Session Initiation Proposal 
Investigation Working Group. 

The IESG contact persons are Allison Mankin and Jon Peterson.

Technical Summary

      Third party call control refers to the ability of one entity to
      create a call in which communications is actually between other
      parties. Third party call control is possible using the mechanisms
      specified within the Session Initiation Protocol (SIP). However,
      there are several possible approaches, each with different benefits
      and drawbacks. This document discusses best current practices for the
      usage of SIP for third party call control.

      Security and identity are important issues in third party call 
      control.  The document provides recommendations for the controller 
      to use a  credentialled identity. It also provides recommendations to 
      aid the two endpoints in the third party controlled call in gaining
      end-to-end secured media.

Working Group Summary

        There was strong working group consensus to advance this document
        it has a significant pull from the community, including 3GPP. There
        were no issues raised during IETF Last Call.   

Protocol Quality

        The document was reviewed for the IESG by Eric Rescorla and Allison 
        Mankin.   Further IESG review included a community poll regarding 
        the use of 0.0.0.0, which is divergent from other Internet usage,
        but was determined to be required due to SDP limitations.


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Wed Jan 21 11:58:43 2004
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From: Robert Sparks <rsparks@dynamicsoft.com>
To: sipping@ietf.org
Content-Type: text/plain
Message-Id: <1074703987.908.82.camel@RjS.localdomain>
Mime-Version: 1.0
X-Mailer: Ximian Evolution 1.4.5 
Date: Wed, 21 Jan 2004 10:53:07 -0600
Content-Transfer-Encoding: 7bit
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Subject: [Sipping] Torture test revision:
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I've just submitted -03 of the torture test document.

It's available at 
http://www.nostrum.com/~rjsparks/draft-ietf-sipping-torture-tests-03.html
and
http://www.nostrum.com/~rjsparks/draft-ietf-sipping-torture-tests-03.txt
until it shows up in the repository.

This version reflects the following bug fixes:

- 3.1.1.1 (wsinv) had a continuation line without
  leading whitespace

- 3.1.1.2 (intmeth) had an unknown header where the
  name was separated from the value with an =, not a :

- 3.1.1.7 (longreq) had a several bad <repeat> end
  tags, causing the binary form of the message to be
  invalid. It also had a continuation line (the last
  Via hfv) with no leading whitespace

- 3.1.2.5 (scalarlg) SIP/2.0.TCP changed to SIP/2.0/TCP

- 3.3.4 (unksm2) and 3.3.5 (bext01) Max-Forward changed to
  Max-Forwards.

- 3.3.7 (regaut) Aligned the Authorization header's unknown
  scheme challenge with the syntax requirements of 3261

- The binary forms of badaspec, badvers, bext01, novelsc, and
  regbadct were missing the null-line.




_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Fri Jan 23 03:45:58 2004
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Thread-Topic: Conference policy URI
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Hi,

There was consensus on the XCON mailing list that the conference state =
event package needs to include in the notification the conference policy =
URI.

There was also consensus that some participants may not want to =
subscribe to the conference package but would still like to learn the =
conference policy URI. In this case, it was suggested that the Call-info =
header carries such information. Perhaps the cc-conferencing draft can =
include examples of that.

Thanks,
Hisham

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Fri Jan 23 05:35:53 2004
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Hi,

It is stated in the changes since section of =
draft-ietf-sipping-cc-conferencing-02.txt:

- Removed inclusion of "isfocus" parameter in Refer-To header field -
   this may be a useful extension to the REFER mechanism in the future,
   however.

Why was that reason for such removal?

Regards,
Hisham

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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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 Proposal Investigation Working Group of the IETF.

	Title		: Session Initiation Protocol Torture Test Messages
	Author(s)	: R. Sparks
	Filename	: draft-ietf-sipping-torture-tests-03.txt
	Pages		: 51
	Date		: 2004-1-22
	
This informational document gives examples of Session Initiation
Protocol (SIP) test messages designed to exercise and 'torture' a
parser.

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_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Fri Jan 23 10:58:09 2004
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Date: Fri, 23 Jan 2004 09:31:12 -0600
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Subject: Re: [Sipping] isfocus in Refer-to header
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Hi Hisham,

The inclusion of feature tags such as "isfocus" was proposed in 
draft-olson-sipping-refer-extensions-00.txt, an individual I-D that has 
expired.

Should this I-D be revised and gain support in the WG, we'd be happy to 
include it in cc-conferencing again as it is a very useful extension.

Thanks,
Alan Johnston
MCI
sip:alan@sipstation.com

At 12:25 PM 1/23/2004 +0200, hisham.khartabil@nokia.com wrote:
>Hi,
>
>It is stated in the changes since section of 
>draft-ietf-sipping-cc-conferencing-02.txt:
>
>- Removed inclusion of "isfocus" parameter in Refer-To header field -
>    this may be a useful extension to the REFER mechanism in the future,
>    however.
>
>Why was that reason for such removal?
>
>Regards,
>Hisham
>
>_______________________________________________
>Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
>This list is for NEW development of the application of SIP
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sip@ietf.org for new developments of core SIP


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Folks,

we have updated the supplemental web page:
http://www.softarmor.com/wgdb/?wg=SIPPING

If you have an *active* individual contribution that does not appear in 
the list above, send me a mail and I will add it to the list.

Thanks,

Gonzalo


_______________________________________________
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This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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I think this is a reasonable usage of Call-Info. However, it can only 
be done if a "purpose" tag value is registered with IANA, as indicated 
by section 20.9. However, it ocurred to me, when reading this, that we 
never set up an IANA registry for call-info purpose tokens! Oops! 
Thats easily rectifiable; when the draft defining the 'policy-uri' (or 
whatever its called) purpose tag is written, it should also create the 
registry and register the other tokens defined in rfc3261.

-Jonathan R.

hisham.khartabil@nokia.com wrote:

> Hi,
> 
> There was consensus on the XCON mailing list that the conference
> state event package needs to include in the notification the
> conference policy URI.
> 
> There was also consensus that some participants may not want to
> subscribe to the conference package but would still like to learn
> the conference policy URI. In this case, it was suggested that the
> Call-info header carries such information. Perhaps the
> cc-conferencing draft can include examples of that.
> 
> Thanks, Hisham
> 
> _______________________________________________ Sipping mailing
> list  https://www1.ietf.org/mailman/listinfo/sipping This list is
> for NEW development of the application of SIP Use
> sip-implementors@cs.columbia.edu for questions on current sip Use
> sip@ietf.org for new developments of core SIP
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "'Paul Kyzivat'" <pkyzivat@cisco.com>,
        Takuya Sawada
	 <tu-sawada@kddi.com>
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Subject: [Sipping] RE: Need to clarify rules for offer/answer usage in sip
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Hi,

I pretty much agree on Takuya's "summary too". However, my understanding is still that you can only send an SDP in a PRACK once (either 18x(offer)/PRACK(answer) or PRACK(offer)/200(answer), as described by Takuya). After that, before the INVITE transactions has completed, you'll have to use UPDATE.

Regards,

Christer Holmberg
Ericsson Finland


> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: 26. tammikuuta 2004 18:43
> To: Takuya Sawada
> Cc: Christer Holmberg (JO/LMF); sipping@ietf.org
> Subject: Need to clarify rules for offer/answer usage in sip
> 
> 
> Takuya,
> 
> You make good points. I think you have convinced me of how this all 
> needs to work. I believe we are also both in agreement that the 
> necessary behavior is currently underspecified in the RFCs.
> 
> I think a useful next step would be to hear from the authors 
> of the RFCs 
> in question, and hopefully obtain their concurrence. Then we 
> can discuss 
> the best way to formalize this. I am moving this to the 
> sipping list and 
> changing the title in order to pick up those authors, some of which 
> don't normally watch sip-implementors.
> 
> 	Paul
> 
> Takuya Sawada wrote:
> > Hi, Paul,
> > 
> > Let me summarize my view of "restrictions" defined in the 
> current SIP RFCs.
> > * Probably I think I may share the same view with Christer 
> in most parts.
> > 
> > Offer/Answer exchange described in RFC 3264 is independent from 
> > higher layer protocol.
> > Higher layer protocol provides means to exchange offer/answer
> > and this needs to provide means for resolving "glare" situation
> > as described in the last paragraph in RFC 3264.
> > 
> > [How to provide means to exchange offer/answer in SIP] 
> > 
> > <RFC 3261>
> > RFC 3261 provides the following two ways to exchange offer/answer;
> > - INVITE/2xx
> > - 2xx/ACK
> > See the page 80 in RFC 3261.
> > 
> > <RFC 3262>
> > RFC 3262 provides the additional three ways to exchange 
> offer/answer;
> > Note that here "1xx" is a reliable provisional response to INVITE.
> > - INVITE/1xx ... a first reliable 1xx with SDP. (may not be 
> the first 1xx)
> > - 1xx/PRACK ... Only when INVITE w/o SDP, 1xx must be the 
> first reliable response.
> > - PRACK/2xx(PRACK) ... Only after a completion of 
> INVITE/1xx or 1xx/PRACK.
> > See chapter 5 in RFC 3262.
> > 
> > <RFC 3311>
> > RFC 3311 provides an additional way to exchange;
> > - UPDATE/2xx(UPDATE)
> > Only when after  any offer/answer exchange defined in RFC 
> 3261 and RFC 3262
> > has completed and no offer/answer exchange in progress.
> > See section 5.1, 5.2 in RFC 3311.
> > 
> > Currently, as far as I understand, no other ways are not 
> defined in any RFCs.
> > Assuming this is correct, let me clarify how to resolve 
> "glare" cases.
> > * I can not imagine how to resolve "glare" if other 
> offer/answer exchanges
> >   are possible beyond my assumption here.
> > 
> > [How to resolve "glare" condition]
> > One reason why the usage of a SIP method to exchange offer/answer 
> > needs to be carefully restricted in RFCs is that UA must 
> detect mis-ordered
> > message delivery and must avoid confusion.
> > In RFC 3264, "it MUST NOT generate a new offer if it has
> > received an offer which it has not yet answered or rejected.
> >  it MUST NOT generate a new offer if it has generated a
> >  prior offer for which it has not yet received an answer or 
> a rejection."
> > So now there seems to be two possible message crossing case
> > to be considered assuming that the above rules are guaranteed.
> > 
> > Case1
> > A                B
> > |offer1          |
> > |--------------->|
> > |         answer1|
> > |<-----\ /-------|
> > |       X  offer2|
> > |<-----/ \-------|
> > 
> > Case2
> > A                B
> > |offer1    offer2|
> > |------\ /-------|
> > |       X        |
> > |<-----/ \------>|
> > 
> > Case1:
> > Assuming that offer/answer exchange follows the 
> restrictions I described
> > above, A can know an SDP in offer2 is not an answer to offer1.
> > Then offer2 must be rejected with a 500 with Retry-After, when 
> > offer2 is UPDATE. And I assume that it can apply to reINVITE,
> > which may not be stated clearly.
> > I am not sure that when this happens when offer2 is in PRACK.
> > But probably 500 with Retry-After will also apply, even it 
> results in
> > delay for reliabile transfer mechanism for provisional response.
> > In the case that offer2 is in the first reliable 
> provisional response
> > or ACK, it seems to me that it never happens, if there is a 
> rule that
> > INVITE, even  w/o SDP, must not be sent if pending offer exists.
> > I am not sure whether it is clearly stated.
> > 
> > Case2:
> > Assuming that offer/answer exchange follows the 
> restrictions I described
> > above, A can know an SDP in offer2 is not an answer to offer1.
> > Then offer2 must be rejected with a 491, if it is UPDATE or 
> (re)INVITE.
> > In the case that offer2 is in PRACK, it may be accepted with 200 or
> > may be rejected with 491. 491 may be adequate for 
> offer/answer but it may
> > delay the completion of a reliable response transfer 
> mechanism or, in
> > worst case, may result in failure to complete SIP 
> transaction because
> > there is no clear retry rule when PRACK is rejected with 491.
> > When RFC 3262 was defined, there were no UPDATE, therefore 
> offer in PRACK
> > never collides with other offers from the other side. I 
> guess this is why
> > there is no rule in RFC 3262 when PRACK offer collides with 
> UPDATE offer.
> > (It is safe to use UPDATE instead of PRACK when UPDATE is supported)
> > Anyway, because there is no PRACK against PRACK collision 
> in the current 
> > RFCs, the other side is always expected to send 491.
> > There is no possibility that offer2 is in the first 
> reliable response or ACK.
> > Because A must not send new offer when it send INVITE w/o offer.
> > 
> > Is this a correct summary?
> > 
> > More comments inline with [TS].
> > 
> > 
> >>Takuya,
> >>
> >>RFC 3312 introduces situations in which multiple 
> offer/answer exchanges 
> >>are required before a single INVITE transaction can 
> complete. Of course 
> >>multiple transactions are involved because PRACK and/or 
> UPDATE are used 
> >>to achieve part of the exchange, but the INVITE transaction 
> can't finish 
> >>until they are done.
> >>
> >>This is at least one case in which offers and answers are 
> spread across 
> >>transactions in complex ways.
> >>
> > 
> > [TS] In my reading so far, there is the only one case which 
> offer/answer is
> > across transaction, that is offer in the first reliable 
> provisional response 
> > of INVITE and answer is in PRACK. There is no other case 
> that an offer is in
> > a transaction and an answer is in another transaction.
> > I agree with that any number of offer/answer exchanges can 
> occur during
> > one INVITE transaction.
> > 
> > 
> >>There are also cases independent of 3312 where there is a 
> need to send a 
> >>subsequent offer/answer before an invite transaction completes. For 
> >>instance, this is one of the techniques for dealing with 
> port conflicts 
> >>introduced as a result of forking.
> >>
> > 
> > [TS] Yes. I did not disagree with that an offer in UPDATE 
> request can
> > be used before the completion of INVITE transaction.
> > 
> > 
> >>Once an invite transaction has completed, including one or more 
> >>offer/answer pairs, additional offer/answer cycles can be initiated 
> >>using either reINVITE or UPDATE.
> >>
> >>In the case of reINVITE, I believe there is a requirement 
> that one or 
> >>more offer/answer cycles be completed before it can 
> succeed. And if it 
> >>fails, the changed tentatively imparted by all of those 
> offer/answer 
> >>cycles are to be rolled back.
> >>
> >>The case of UPDATE is a little less clear. If there is no invite 
> >>transaction in progress, and an offer is sent in an UPDATE, and the 
> >>update is rejected, it must be the case that the dialog 
> state remains as 
> >>it was before the update. If there is no invite in progress and an 
> >>UPDATE is sent without an offer, there seems to be no prohibition 
> >>against sending an offer in the update response. In that case, the 
> >>answerer would have no way to refuse the offer and retain 
> the dialog as 
> >>it is. The only option the answerer would have is to send 
> another UPDATE 
> >>containing the answer. I doubt this is intended behavior, 
> but it isn't 
> >>prohibited.
> >>
> > 
> > [TS] This is the point I can not agree. I can not find any 
> description that
> > UPDATE response can contain an offer in RFC 3311. IMO, this 
> means UPDATE response
> > can not have any offer. SIP RFCs define restrictions to 
> utilize RFC 3264 model.
> > UAs can not do any offer/answer exchange which is not 
> defined clearly in any RFCs.
> > 
> > 
> >>More below.
> >>
> >>Paul
> >>
> >>Takuya Sawada wrote:
> >>
> >>>Hi,
> >>>
> >>>In my understanding,
> >>>
> >>>- Once previous offer/answer completes each side can issue 
> new offer.
> >>>- INVITE request contains the offer, and the first 
> reliable 1xx response
> >>>  which contains answer, offer/answer for the INVITE 
> transaction completes.
> >>>- UPDATE transaction and PRACK transaction can exchange 
> offer/answer.
> >>>- Therefore, once the reliable 1xx response which contains 
> answer is received,
> >>>  caller can start offer/answer with PRACK or UPDATE or 
> any new method.
> >>>  And once the reliable 1xx response which contains answer is sent,
> >>>  callee can start offer/answer with UPDATE.
> >>>
> >>>
> >>>
> >>>>P.S. For those with the patience, below I have tried to 
> enumerate all 
> >>>>the offer/answer cases thru the 2nd offer. It is pretty painful.
> >>>>
> >>>
> >>><snip>
> >>>
> >>>>2) invite with 100rel
> >>>>   a) offer in invite
> >>>>      Possible places for answer:
> >>>>      i)   answer in first reliable 1xx
> >>>>      ii)  answer in subsequent reliable 1xx
> >>>>      iii) answer in 2xx
> >>>>   b) offer not in invite
> >>>>      offer must be in first reliable 1xx
> >>>>      answer must be in PRACK to 1xx w/offer
> >>>>
> >>>>Multiple offer/answer cycle prior to completion of invite.
> >>>>Can only happen in subcases of (2) above.
> >>>>It requires use of UPDATE in some cases.
> >>>>
> >>>>3) original offerer sends 2nd offer
> >>>>   a) following 2.a.i or 2.a.i.i:
> >>>>      Possible places for 2nd offer:
> >>>>      i)   offer2 in PRACK of answer1
> >>>>      ii)  offer2 in PRACK of some subsequent 1xx
> >>>>      iii) offer2 in UPDATE
> >>>>   b) following 2.a.iii:
> >>>>      No 2nd offer possible until after completion of invite.
> >>>>   c) following 2.b:
> >>>>      i)   offer2 in 2xx to PRACK w/answer1
> >>>>      ii)  offer2 in UPDATE
> >>>>
> >>>>4) original answerer sends 2nd offer:
> >>>>   Not clear if this is intended to be legal at all prior to
> >>>>   completion of invite. It appears existing text permits
> >>>>   as follows:
> >>>>   a) following 2.a.i or 2.a.ii:
> >>>>      Possible places for 2nd offer:
> >>>>      i)   offer2 in 2xx to prack of answer1.
> >>>>      ii)  offer2 in subsequent reliable 1xx
> >>>>           following the one carrying answer1.
> >>>>      iii) offer2 in UPDATE following answer1.
> >>>>           (No reason to prefer this to other possibilities.)
> >>>>   b) following 2.a.iii:
> >>>>      No 2nd offer possible until after completion of invite.
> >>>>   c) following 2b:
> >>>>      i)   if another reliable 1xx is received,
> >>>>           could send offer2 in the PRACK.
> >>>>           (but can't count on this happening.)
> >>>>      ii)  offer2 in UPDATE after sending answer1.
> >>>
> >>>
> >>>I do not think offer/answer exchange beyond the SIP transaction
> >>>is possible. I am not sure it is prohibited somewhere in RFC, but
> >>>even if it is not prohibited explicitly, one should not assumed
> >>>that is possible.
> >>
> >>Following the principle of being conservative in what you send and 
> >>liberal in what you receive, I want to assume that I might receive 
> >>anything that isn't prohibited, and do something reasonable with it.
> >>
> > 
> > [TS] I love this principle. IMO, if I receive an SDP which 
> I do not expect
> > to receive in RFC, one possible solution is that I will 
> accept SIP message
> > and ignore the SDP. Content-Disposition header's handling parameter
> > should be checked, though.
> > Anyway, here I do not know what is something reasonable to 
> do with it.
> > 
> > 
> >>I would be willing to be conservative in initiating this stuff if I 
> >>could figure out what that is. Given the current state of 
> >>underspecification I don't know that I will draw the same 
> conclusion 
> >>than anyone else will.
> >>
> > 
> > [TS] There seems to be some uderspecification in RFCs. But 
> I want to 
> > confirm the following;
> >   RFCs only state where and when an offer and an answer can appear.
> >   If RFCs do not state "it can", then it MUST NOT contain 
> offer/answer.
> >   Because RFCs define "restrictions" where and when offer/answer
> >  exchange can occur.
> > First I think we should reach a consensus with this point.
> > 
> > 
> >> > One can achieve the same thing without such a
> >>
> >>>tweak. 
> >>
> >>Not sure what you are referring to here, or whether it is 
> negated in 
> >>your subsequent posting. See above for a number of things 
> that must be 
> >>possible.
> >>
> >>
> >>>Then, 3.c.i and 4.a.1 SHOULD NOT, or hopefully MUST NOT, be used.
> >>
> >>If we can codify a clear, and complete, set of rules for 
> where offers 
> >>and answers can appear, that provides full functionality, and it 
> >>excludes these cases, then fine.
> >>
> >>In the case of 4.a.1, removing it causes no loss of 
> functionality, so I 
> >>am inclined to agree. It might ultimately result in exchanging more 
> >>messages, but the number is probably insignificant.
> >>
> >>In the case of 3.c.1, removing it requires that both sides support 
> >>UPDATE. So its removal may remove functionality in the case 
> where one 
> >>side or the other doesn't support UPDATE. This may or may not be 
> >>significant.
> >>
> > 
> > [TS] As you said, UPDATE can solve the problem.
> > You can use extensions when it wants the functionality that 
> the base 
> > specifications can not provide. This is what the extensions are.
> > 
> > 
> >>>Anyway for INVITE transaction, as explained in my previous mail,
> >>>it is clear in RFC 3261 to me that one INVITE transaction can 
> >>>include only one offer/answer exchange.
> >>>
> >>>Then 4.a.ii is not permitted.
> >>
> >>I think I identified cases above where this is needed.
> >>
> > 
> > [TS] The reason I think it is not permitted is, as Christer 
> pointed our,
> > that RFC 3261 does not permit and any other RFCs do not override it.
> > 
> > 
> >>Takuya Sawada wrote:
> >> > Hi,
> >> >
> >> >
> >> >>I do not think offer/answer exchange beyond the SIP transaction
> >> >>is possible. I am not sure it is prohibited somewhere in RFC, but
> >> >>even if it is not prohibited explicitly, one should not assumed
> >> >>that is possible. One can achieve the same thing without such a
> >> >>tweak.
> >> >
> >> >
> >> > I am not exactly correct here.
> >> > When an offer is in INVITE reliable provisonal response and
> >> > an answer is in PRACK offer/answer exchange is beyond one
> >> > transaction.
> >> > What I want to say is that how offer/answer is used in a certain
> >> > method is explicitly scoped in each RFC which defines the method.
> >>
> >>The trouble with these is that they were written in a 
> certain order. As 
> >>each was written, a broadening of thinking seems to have 
> occurred. This 
> >>is reflected in each, but not necessarily fully reflected 
> in the earlier 
> >>specs.
> >>
> > 
> > [TS] I may agree with this. At least this makes it 
> difficult to understand
> > and share the same view among all. Actually, there seems to 
> be a little 
> > ambiguity with PRACK and UPDATE collision case as I pointed 
> > out in this mail, etc.
> > 
> > 
> >> > How PRACK is used for offer/answer is defined in RFC 3262.
> >> > There seems to be only two cases:
> >> >  - offer/answer is reliable 1xx(INVITE)/PRACK
> >> >  - offer/answer is PRACK/2xx(PRACK)
> >> >    only after INVITE's offer/answer exchange complete.
> >>
> >>I agree that this seems pretty well locked down as it 
> relates to 3261. 
> >>These lock downs of course do apply to the things that came 
> later. But 
> >>they seem at least awkward and arbitrary in some cases.
> >>
> >> > How UPDATE is used for offer/answer is defined in RFC 3311.
> >> >  - offer/answer is UPDATE/2xx(UPDATE)
> >>
> >>Here I disagree. I looked pretty closely at this, and I 
> don't see *any* 
> >>normative language for how offer/answer is used with it. There is a 
> >>bunch of non-normative language which appears to simply restate 
> >>implications of 3261 and 3264 in the presence of update.
> >>
> > 
> > [TS] Again, this is the point I can not agree. I agree with 
> that there is
> > no normative language to prohibit offer in the response of UPDATE
> > like "2xx response to UPDATE MUST NOT contain offer".
> > Actually, response can have an SDP even if UPDATE does not have
> > an offer SDP. But an SDP can be neither an offer nor an answer 
> > unless clear semantics are defined in any RFCs. I can not find 
> > any usage or semantics to use offer/answer exchange beyond
> > one UPDATE transaction.
> > IMO, SDP in a 2xx of UPDATE when an UPDATE request w/o
> > offer is neither an offer nor an answer like SDP in INFO is 
> > neither an offer nor an answer.
> > 
> > 
> >> > How INVITE is used for offer/answer is defined in RFC 3261.
> >> >  - offer/answer is INVITE/non-failure reliable response
> >> >  - offer/answer is the first non-failure first 
> >>response/acknowlegement for the response
> >> >    , which is 2xx(INVITE)/ACK in RFC 3261 and 
> 1xx(INVITE)/PRACK in 
> >>RFC 3262.
> >> >  * Note that the second or later SDP in INVITE 
> non-failure response
> >> >    is not an offer. It is nothing.
> >> >
> >> > There is no other way than explicitly defined in any RFC, IMO.
> >>
> >>I believe you are right that nothing more is explicitly defined.
> >>But 3264 is pretty general and broad. It is not prescriptive in how 
> >>offers and answers are transmitted. 
> > 
> > 
> > [TS] Yes. Therefore, RFC 3264 requires restrictions on how 
> > to exchange offer/answer in higher layer protocol.
> > 
> > 
> >>And while the other RFCs may not 
> >>explicitly specify other ways, there are other ways that aren't 
> >>explicitly prohibited. In general with sip, things that aren't 
> >>explicitly prohibited are permitted.
> >>
> > 
> > [TS] Yes. Any SIP messages can contain SDP at any moment. 
> > But without semantics, it is neither an offer nor an answer.
> > 
> > 
> >>I just want to get this nailed down, so I know how to build 
> a stack that 
> >>can both handle every valid case thrown at it, and that can 
> initiate all 
> >>the needed functionality.
> >>
> >>I am thinking that maybe somebody (probably me) needs to 
> write a best 
> >>practices document that pulls this all together, specifying 
> all the ways 
> >>that offer/answer can be used. Following that, I think it will be 
> >>advisable to tighten up one or more of the RFCs in places 
> where things 
> >>haven't been prohibited but should be.
> >>
> > 
> > [TS] The guideline document that you are proposing can be useful.
> > Collecting all the possible collision cases, recommending UPDATE 
> > offer rather than PRACK offer if UPDATE is supported and no need 
> > to reduce the number of messages, enumerating Warning header code 
> > examples to be included when offer is rejected with 488, etc.
> > 
> > Regards,
> > Takuya
> > 
> > 
> >>Paul
> >>
> >>_______________________________________________
> >>Sip-implementors mailing list
> >>Sip-implementors@cs.columbia.edu
> >>http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> > 
> > 
> > --------
> > Takuya Sawada
> > KDDI Corporation (KDDI)
> > Garden Air Tower, 3-10-10  Iidabashi Chiyoda-ku
> > Tokyo 102-8460, Japan
> > Tel: +81-3-6678-2997
> > Fax: +81-3-6678-0285
> > tu-sawada@kddi.com
> > 
> 

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From: "arunvenk" <arunvenk@cisco.com>
To: <sipping@ietf.org>
Subject: RE: [Sipping] RE: Need to clarify rules for offer/answer usage in sip
Date: Tue, 27 Jan 2004 08:03:13 -0600
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Christer
You state that " you can only send an SDP in a PRACK once".
If you have more than one rel prov response and the most recent one
arrives without an offer and there is no incomplete offer/answer at that
time, a new offer could be sent in the PRACK, I would think.
Venkat


-----Original Message-----
From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On Behalf
Of Christer Holmberg (JO/LMF)
Sent: Tuesday, January 27, 2004 4:55 AM
To: 'Paul Kyzivat'; Takuya Sawada
Cc: sipping@ietf.org
Subject: [Sipping] RE: Need to clarify rules for offer/answer usage in
sip


Hi,

I pretty much agree on Takuya's "summary too". However, my understanding
is still that you can only send an SDP in a PRACK once (either
18x(offer)/PRACK(answer) or PRACK(offer)/200(answer), as described by
Takuya). After that, before the INVITE transactions has completed,
you'll have to use UPDATE.

Regards,

Christer Holmberg
Ericsson Finland


> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> Sent: 26. tammikuuta 2004 18:43
> To: Takuya Sawada
> Cc: Christer Holmberg (JO/LMF); sipping@ietf.org
> Subject: Need to clarify rules for offer/answer usage in sip
> 
> 
> Takuya,
> 
> You make good points. I think you have convinced me of how this all 
> needs to work. I believe we are also both in agreement that the 
> necessary behavior is currently underspecified in the RFCs.
> 
> I think a useful next step would be to hear from the authors 
> of the RFCs 
> in question, and hopefully obtain their concurrence. Then we 
> can discuss 
> the best way to formalize this. I am moving this to the 
> sipping list and 
> changing the title in order to pick up those authors, some of which 
> don't normally watch sip-implementors.
> 
> 	Paul
> 
> Takuya Sawada wrote:
> > Hi, Paul,
> > 
> > Let me summarize my view of "restrictions" defined in the 
> current SIP RFCs.
> > * Probably I think I may share the same view with Christer 
> in most parts.
> > 
> > Offer/Answer exchange described in RFC 3264 is independent from 
> > higher layer protocol.
> > Higher layer protocol provides means to exchange offer/answer
> > and this needs to provide means for resolving "glare" situation
> > as described in the last paragraph in RFC 3264.
> > 
> > [How to provide means to exchange offer/answer in SIP] 
> > 
> > <RFC 3261>
> > RFC 3261 provides the following two ways to exchange offer/answer;
> > - INVITE/2xx
> > - 2xx/ACK
> > See the page 80 in RFC 3261.
> > 
> > <RFC 3262>
> > RFC 3262 provides the additional three ways to exchange 
> offer/answer;
> > Note that here "1xx" is a reliable provisional response to INVITE.
> > - INVITE/1xx ... a first reliable 1xx with SDP. (may not be 
> the first 1xx)
> > - 1xx/PRACK ... Only when INVITE w/o SDP, 1xx must be the 
> first reliable response.
> > - PRACK/2xx(PRACK) ... Only after a completion of 
> INVITE/1xx or 1xx/PRACK.
> > See chapter 5 in RFC 3262.
> > 
> > <RFC 3311>
> > RFC 3311 provides an additional way to exchange;
> > - UPDATE/2xx(UPDATE)
> > Only when after  any offer/answer exchange defined in RFC 
> 3261 and RFC 3262
> > has completed and no offer/answer exchange in progress.
> > See section 5.1, 5.2 in RFC 3311.
> > 
> > Currently, as far as I understand, no other ways are not 
> defined in any RFCs.
> > Assuming this is correct, let me clarify how to resolve 
> "glare" cases.
> > * I can not imagine how to resolve "glare" if other 
> offer/answer exchanges
> >   are possible beyond my assumption here.
> > 
> > [How to resolve "glare" condition]
> > One reason why the usage of a SIP method to exchange offer/answer 
> > needs to be carefully restricted in RFCs is that UA must 
> detect mis-ordered
> > message delivery and must avoid confusion.
> > In RFC 3264, "it MUST NOT generate a new offer if it has
> > received an offer which it has not yet answered or rejected.
> >  it MUST NOT generate a new offer if it has generated a
> >  prior offer for which it has not yet received an answer or 
> a rejection."
> > So now there seems to be two possible message crossing case
> > to be considered assuming that the above rules are guaranteed.
> > 
> > Case1
> > A                B
> > |offer1          |
> > |--------------->|
> > |         answer1|
> > |<-----\ /-------|
> > |       X  offer2|
> > |<-----/ \-------|
> > 
> > Case2
> > A                B
> > |offer1    offer2|
> > |------\ /-------|
> > |       X        |
> > |<-----/ \------>|
> > 
> > Case1:
> > Assuming that offer/answer exchange follows the 
> restrictions I described
> > above, A can know an SDP in offer2 is not an answer to offer1.
> > Then offer2 must be rejected with a 500 with Retry-After, when 
> > offer2 is UPDATE. And I assume that it can apply to reINVITE,
> > which may not be stated clearly.
> > I am not sure that when this happens when offer2 is in PRACK.
> > But probably 500 with Retry-After will also apply, even it 
> results in
> > delay for reliabile transfer mechanism for provisional response.
> > In the case that offer2 is in the first reliable 
> provisional response
> > or ACK, it seems to me that it never happens, if there is a 
> rule that
> > INVITE, even  w/o SDP, must not be sent if pending offer exists.
> > I am not sure whether it is clearly stated.
> > 
> > Case2:
> > Assuming that offer/answer exchange follows the 
> restrictions I described
> > above, A can know an SDP in offer2 is not an answer to offer1.
> > Then offer2 must be rejected with a 491, if it is UPDATE or 
> (re)INVITE.
> > In the case that offer2 is in PRACK, it may be accepted with 200 or
> > may be rejected with 491. 491 may be adequate for 
> offer/answer but it may
> > delay the completion of a reliable response transfer 
> mechanism or, in
> > worst case, may result in failure to complete SIP 
> transaction because
> > there is no clear retry rule when PRACK is rejected with 491.
> > When RFC 3262 was defined, there were no UPDATE, therefore 
> offer in PRACK
> > never collides with other offers from the other side. I 
> guess this is why
> > there is no rule in RFC 3262 when PRACK offer collides with 
> UPDATE offer.
> > (It is safe to use UPDATE instead of PRACK when UPDATE is supported)
> > Anyway, because there is no PRACK against PRACK collision 
> in the current 
> > RFCs, the other side is always expected to send 491.
> > There is no possibility that offer2 is in the first 
> reliable response or ACK.
> > Because A must not send new offer when it send INVITE w/o offer.
> > 
> > Is this a correct summary?
> > 
> > More comments inline with [TS].
> > 
> > 
> >>Takuya,
> >>
> >>RFC 3312 introduces situations in which multiple 
> offer/answer exchanges 
> >>are required before a single INVITE transaction can 
> complete. Of course 
> >>multiple transactions are involved because PRACK and/or 
> UPDATE are used 
> >>to achieve part of the exchange, but the INVITE transaction 
> can't finish 
> >>until they are done.
> >>
> >>This is at least one case in which offers and answers are 
> spread across 
> >>transactions in complex ways.
> >>
> > 
> > [TS] In my reading so far, there is the only one case which 
> offer/answer is
> > across transaction, that is offer in the first reliable 
> provisional response 
> > of INVITE and answer is in PRACK. There is no other case 
> that an offer is in
> > a transaction and an answer is in another transaction.
> > I agree with that any number of offer/answer exchanges can 
> occur during
> > one INVITE transaction.
> > 
> > 
> >>There are also cases independent of 3312 where there is a 
> need to send a 
> >>subsequent offer/answer before an invite transaction completes. For 
> >>instance, this is one of the techniques for dealing with 
> port conflicts 
> >>introduced as a result of forking.
> >>
> > 
> > [TS] Yes. I did not disagree with that an offer in UPDATE 
> request can
> > be used before the completion of INVITE transaction.
> > 
> > 
> >>Once an invite transaction has completed, including one or more 
> >>offer/answer pairs, additional offer/answer cycles can be initiated 
> >>using either reINVITE or UPDATE.
> >>
> >>In the case of reINVITE, I believe there is a requirement 
> that one or 
> >>more offer/answer cycles be completed before it can 
> succeed. And if it 
> >>fails, the changed tentatively imparted by all of those 
> offer/answer 
> >>cycles are to be rolled back.
> >>
> >>The case of UPDATE is a little less clear. If there is no invite 
> >>transaction in progress, and an offer is sent in an UPDATE, and the 
> >>update is rejected, it must be the case that the dialog 
> state remains as 
> >>it was before the update. If there is no invite in progress and an 
> >>UPDATE is sent without an offer, there seems to be no prohibition 
> >>against sending an offer in the update response. In that case, the 
> >>answerer would have no way to refuse the offer and retain 
> the dialog as 
> >>it is. The only option the answerer would have is to send 
> another UPDATE 
> >>containing the answer. I doubt this is intended behavior, 
> but it isn't 
> >>prohibited.
> >>
> > 
> > [TS] This is the point I can not agree. I can not find any 
> description that
> > UPDATE response can contain an offer in RFC 3311. IMO, this 
> means UPDATE response
> > can not have any offer. SIP RFCs define restrictions to 
> utilize RFC 3264 model.
> > UAs can not do any offer/answer exchange which is not 
> defined clearly in any RFCs.
> > 
> > 
> >>More below.
> >>
> >>Paul
> >>
> >>Takuya Sawada wrote:
> >>
> >>>Hi,
> >>>
> >>>In my understanding,
> >>>
> >>>- Once previous offer/answer completes each side can issue 
> new offer.
> >>>- INVITE request contains the offer, and the first 
> reliable 1xx response
> >>>  which contains answer, offer/answer for the INVITE 
> transaction completes.
> >>>- UPDATE transaction and PRACK transaction can exchange 
> offer/answer.
> >>>- Therefore, once the reliable 1xx response which contains 
> answer is received,
> >>>  caller can start offer/answer with PRACK or UPDATE or 
> any new method.
> >>>  And once the reliable 1xx response which contains answer is sent,
> >>>  callee can start offer/answer with UPDATE.
> >>>
> >>>
> >>>
> >>>>P.S. For those with the patience, below I have tried to 
> enumerate all 
> >>>>the offer/answer cases thru the 2nd offer. It is pretty painful.
> >>>>
> >>>
> >>><snip>
> >>>
> >>>>2) invite with 100rel
> >>>>   a) offer in invite
> >>>>      Possible places for answer:
> >>>>      i)   answer in first reliable 1xx
> >>>>      ii)  answer in subsequent reliable 1xx
> >>>>      iii) answer in 2xx
> >>>>   b) offer not in invite
> >>>>      offer must be in first reliable 1xx
> >>>>      answer must be in PRACK to 1xx w/offer
> >>>>
> >>>>Multiple offer/answer cycle prior to completion of invite.
> >>>>Can only happen in subcases of (2) above.
> >>>>It requires use of UPDATE in some cases.
> >>>>
> >>>>3) original offerer sends 2nd offer
> >>>>   a) following 2.a.i or 2.a.i.i:
> >>>>      Possible places for 2nd offer:
> >>>>      i)   offer2 in PRACK of answer1
> >>>>      ii)  offer2 in PRACK of some subsequent 1xx
> >>>>      iii) offer2 in UPDATE
> >>>>   b) following 2.a.iii:
> >>>>      No 2nd offer possible until after completion of invite.
> >>>>   c) following 2.b:
> >>>>      i)   offer2 in 2xx to PRACK w/answer1
> >>>>      ii)  offer2 in UPDATE
> >>>>
> >>>>4) original answerer sends 2nd offer:
> >>>>   Not clear if this is intended to be legal at all prior to
> >>>>   completion of invite. It appears existing text permits
> >>>>   as follows:
> >>>>   a) following 2.a.i or 2.a.ii:
> >>>>      Possible places for 2nd offer:
> >>>>      i)   offer2 in 2xx to prack of answer1.
> >>>>      ii)  offer2 in subsequent reliable 1xx
> >>>>           following the one carrying answer1.
> >>>>      iii) offer2 in UPDATE following answer1.
> >>>>           (No reason to prefer this to other possibilities.)
> >>>>   b) following 2.a.iii:
> >>>>      No 2nd offer possible until after completion of invite.
> >>>>   c) following 2b:
> >>>>      i)   if another reliable 1xx is received,
> >>>>           could send offer2 in the PRACK.
> >>>>           (but can't count on this happening.)
> >>>>      ii)  offer2 in UPDATE after sending answer1.
> >>>
> >>>
> >>>I do not think offer/answer exchange beyond the SIP transaction
> >>>is possible. I am not sure it is prohibited somewhere in RFC, but
> >>>even if it is not prohibited explicitly, one should not assumed
> >>>that is possible.
> >>
> >>Following the principle of being conservative in what you send and 
> >>liberal in what you receive, I want to assume that I might receive 
> >>anything that isn't prohibited, and do something reasonable with it.
> >>
> > 
> > [TS] I love this principle. IMO, if I receive an SDP which 
> I do not expect
> > to receive in RFC, one possible solution is that I will 
> accept SIP message
> > and ignore the SDP. Content-Disposition header's handling parameter
> > should be checked, though.
> > Anyway, here I do not know what is something reasonable to 
> do with it.
> > 
> > 
> >>I would be willing to be conservative in initiating this stuff if I 
> >>could figure out what that is. Given the current state of 
> >>underspecification I don't know that I will draw the same 
> conclusion 
> >>than anyone else will.
> >>
> > 
> > [TS] There seems to be some uderspecification in RFCs. But 
> I want to 
> > confirm the following;
> >   RFCs only state where and when an offer and an answer can appear.
> >   If RFCs do not state "it can", then it MUST NOT contain 
> offer/answer.
> >   Because RFCs define "restrictions" where and when offer/answer
> >  exchange can occur.
> > First I think we should reach a consensus with this point.
> > 
> > 
> >> > One can achieve the same thing without such a
> >>
> >>>tweak. 
> >>
> >>Not sure what you are referring to here, or whether it is 
> negated in 
> >>your subsequent posting. See above for a number of things 
> that must be 
> >>possible.
> >>
> >>
> >>>Then, 3.c.i and 4.a.1 SHOULD NOT, or hopefully MUST NOT, be used.
> >>
> >>If we can codify a clear, and complete, set of rules for 
> where offers 
> >>and answers can appear, that provides full functionality, and it 
> >>excludes these cases, then fine.
> >>
> >>In the case of 4.a.1, removing it causes no loss of 
> functionality, so I 
> >>am inclined to agree. It might ultimately result in exchanging more 
> >>messages, but the number is probably insignificant.
> >>
> >>In the case of 3.c.1, removing it requires that both sides support 
> >>UPDATE. So its removal may remove functionality in the case 
> where one 
> >>side or the other doesn't support UPDATE. This may or may not be 
> >>significant.
> >>
> > 
> > [TS] As you said, UPDATE can solve the problem.
> > You can use extensions when it wants the functionality that 
> the base 
> > specifications can not provide. This is what the extensions are.
> > 
> > 
> >>>Anyway for INVITE transaction, as explained in my previous mail,
> >>>it is clear in RFC 3261 to me that one INVITE transaction can 
> >>>include only one offer/answer exchange.
> >>>
> >>>Then 4.a.ii is not permitted.
> >>
> >>I think I identified cases above where this is needed.
> >>
> > 
> > [TS] The reason I think it is not permitted is, as Christer 
> pointed our,
> > that RFC 3261 does not permit and any other RFCs do not override it.
> > 
> > 
> >>Takuya Sawada wrote:
> >> > Hi,
> >> >
> >> >
> >> >>I do not think offer/answer exchange beyond the SIP transaction
> >> >>is possible. I am not sure it is prohibited somewhere in RFC, but
> >> >>even if it is not prohibited explicitly, one should not assumed
> >> >>that is possible. One can achieve the same thing without such a
> >> >>tweak.
> >> >
> >> >
> >> > I am not exactly correct here.
> >> > When an offer is in INVITE reliable provisonal response and
> >> > an answer is in PRACK offer/answer exchange is beyond one
> >> > transaction.
> >> > What I want to say is that how offer/answer is used in a certain
> >> > method is explicitly scoped in each RFC which defines the method.
> >>
> >>The trouble with these is that they were written in a 
> certain order. As 
> >>each was written, a broadening of thinking seems to have 
> occurred. This 
> >>is reflected in each, but not necessarily fully reflected 
> in the earlier 
> >>specs.
> >>
> > 
> > [TS] I may agree with this. At least this makes it 
> difficult to understand
> > and share the same view among all. Actually, there seems to 
> be a little 
> > ambiguity with PRACK and UPDATE collision case as I pointed 
> > out in this mail, etc.
> > 
> > 
> >> > How PRACK is used for offer/answer is defined in RFC 3262.
> >> > There seems to be only two cases:
> >> >  - offer/answer is reliable 1xx(INVITE)/PRACK
> >> >  - offer/answer is PRACK/2xx(PRACK)
> >> >    only after INVITE's offer/answer exchange complete.
> >>
> >>I agree that this seems pretty well locked down as it 
> relates to 3261. 
> >>These lock downs of course do apply to the things that came 
> later. But 
> >>they seem at least awkward and arbitrary in some cases.
> >>
> >> > How UPDATE is used for offer/answer is defined in RFC 3311.
> >> >  - offer/answer is UPDATE/2xx(UPDATE)
> >>
> >>Here I disagree. I looked pretty closely at this, and I 
> don't see *any* 
> >>normative language for how offer/answer is used with it. There is a 
> >>bunch of non-normative language which appears to simply restate 
> >>implications of 3261 and 3264 in the presence of update.
> >>
> > 
> > [TS] Again, this is the point I can not agree. I agree with 
> that there is
> > no normative language to prohibit offer in the response of UPDATE
> > like "2xx response to UPDATE MUST NOT contain offer".
> > Actually, response can have an SDP even if UPDATE does not have
> > an offer SDP. But an SDP can be neither an offer nor an answer 
> > unless clear semantics are defined in any RFCs. I can not find 
> > any usage or semantics to use offer/answer exchange beyond
> > one UPDATE transaction.
> > IMO, SDP in a 2xx of UPDATE when an UPDATE request w/o
> > offer is neither an offer nor an answer like SDP in INFO is 
> > neither an offer nor an answer.
> > 
> > 
> >> > How INVITE is used for offer/answer is defined in RFC 3261.
> >> >  - offer/answer is INVITE/non-failure reliable response
> >> >  - offer/answer is the first non-failure first 
> >>response/acknowlegement for the response
> >> >    , which is 2xx(INVITE)/ACK in RFC 3261 and 
> 1xx(INVITE)/PRACK in 
> >>RFC 3262.
> >> >  * Note that the second or later SDP in INVITE 
> non-failure response
> >> >    is not an offer. It is nothing.
> >> >
> >> > There is no other way than explicitly defined in any RFC, IMO.
> >>
> >>I believe you are right that nothing more is explicitly defined.
> >>But 3264 is pretty general and broad. It is not prescriptive in how 
> >>offers and answers are transmitted. 
> > 
> > 
> > [TS] Yes. Therefore, RFC 3264 requires restrictions on how 
> > to exchange offer/answer in higher layer protocol.
> > 
> > 
> >>And while the other RFCs may not 
> >>explicitly specify other ways, there are other ways that aren't 
> >>explicitly prohibited. In general with sip, things that aren't 
> >>explicitly prohibited are permitted.
> >>
> > 
> > [TS] Yes. Any SIP messages can contain SDP at any moment. 
> > But without semantics, it is neither an offer nor an answer.
> > 
> > 
> >>I just want to get this nailed down, so I know how to build 
> a stack that 
> >>can both handle every valid case thrown at it, and that can 
> initiate all 
> >>the needed functionality.
> >>
> >>I am thinking that maybe somebody (probably me) needs to 
> write a best 
> >>practices document that pulls this all together, specifying 
> all the ways 
> >>that offer/answer can be used. Following that, I think it will be 
> >>advisable to tighten up one or more of the RFCs in places 
> where things 
> >>haven't been prohibited but should be.
> >>
> > 
> > [TS] The guideline document that you are proposing can be useful.
> > Collecting all the possible collision cases, recommending UPDATE 
> > offer rather than PRACK offer if UPDATE is supported and no need 
> > to reduce the number of messages, enumerating Warning header code 
> > examples to be included when offer is rejected with 488, etc.
> > 
> > Regards,
> > Takuya
> > 
> > 
> >>Paul
> >>
> >>_______________________________________________
> >>Sip-implementors mailing list
> >>Sip-implementors@cs.columbia.edu
> >>http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> > 
> > 
> > --------
> > Takuya Sawada
> > KDDI Corporation (KDDI)
> > Garden Air Tower, 3-10-10  Iidabashi Chiyoda-ku
> > Tokyo 102-8460, Japan
> > Tel: +81-3-6678-2997
> > Fax: +81-3-6678-0285
> > tu-sawada@kddi.com
> > 
> 

_______________________________________________
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Rohan,

Skimmed your draft and have a question... By using the content-disposition
header it seems that we're requiring the request/response to contain
content. In the case where a user agent receives an INVITE with no SDP in it
(slow start call) it would seem that they'll use some sort of default
alerting mechanism until they get the SDP setup (which may not happen until
the 200 OK/ACK). In the case of your draft it would seem that this means the
client may ring one way, then suddenly switch and ring a different way
because it's not aware that there's going to be some alert information in
the future.

Am I correct, and if not, how does the draft approach this problem?

Regards,

Brian

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Wednesday, July 16, 2003 7:08 PM
To: Rohan Mahy
Cc: 'sipping@ietf.org'
Subject: Re: [Sipping] New Tones draft (was: "who plays the tone" or
something like that)


Rohan,

I just got around to reading this draft. I think its neat. You say in 
your note below that its linked to app-interaction, but you don't use 
the app-interaction framework. I believe this is precisely the kind of 
thing we want to do with app-interaction - its a stimulus interface to 
the user. That means that you should be using app-info headers to 
convey the information, for example.

Indeed, there are probably existing xml formats which may even do this.

-Jonathan R.


Rohan Mahy wrote:

> Hi,
> 
> I wrote the following draft in order to provide some discussion on 
> alternatives to in-band early-media ringback, and to discuss the issue 
> of playing tones in general which I believe is inherently linked to 3pcc 
> and application interaction.
> 
> http://www.softarmor.com/sipping/drafts/draft-mahy-sipping-tones-00.html
> http://www.softarmor.com/sipping/drafts/draft-mahy-sipping-tones-00.txt
> 
> comments welcome.
> thx,
> -r
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
> 

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Technology Officer                    Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com


_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

------_=_NextPart_001_01C3E465.CF3CF7D8
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	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

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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 =
5.5.2656.31">
<TITLE>RE: [Sipping] New Tones draft (was: &quot;who plays the =
tone&quot; or something like that)</TITLE>
</HEAD>
<BODY>

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

<P><FONT SIZE=3D2>Skimmed your draft and have a question... By using =
the content-disposition header it seems that we're requiring the =
request/response to contain content. In the case where a user agent =
receives an INVITE with no SDP in it (slow start call) it would seem =
that they'll use some sort of default alerting mechanism until they get =
the SDP setup (which may not happen until the 200 OK/ACK). In the case =
of your draft it would seem that this means the client may ring one =
way, then suddenly switch and ring a different way because it's not =
aware that there's going to be some alert information in the =
future.</FONT></P>

<P><FONT SIZE=3D2>Am I correct, and if not, how does the draft approach =
this problem?</FONT>
</P>

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

<P><FONT SIZE=3D2>Brian</FONT>
</P>

<P><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Jonathan Rosenberg [<A =
HREF=3D"mailto:jdrosen@dynamicsoft.com">mailto:jdrosen@dynamicsoft.com</=
A>]</FONT>
<BR><FONT SIZE=3D2>Sent: Wednesday, July 16, 2003 7:08 PM</FONT>
<BR><FONT SIZE=3D2>To: Rohan Mahy</FONT>
<BR><FONT SIZE=3D2>Cc: 'sipping@ietf.org'</FONT>
<BR><FONT SIZE=3D2>Subject: Re: [Sipping] New Tones draft (was: =
&quot;who plays the tone&quot; or</FONT>
<BR><FONT SIZE=3D2>something like that)</FONT>
</P>
<BR>

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

<P><FONT SIZE=3D2>I just got around to reading this draft. I think its =
neat. You say in </FONT>
<BR><FONT SIZE=3D2>your note below that its linked to app-interaction, =
but you don't use </FONT>
<BR><FONT SIZE=3D2>the app-interaction framework. I believe this is =
precisely the kind of </FONT>
<BR><FONT SIZE=3D2>thing we want to do with app-interaction - its a =
stimulus interface to </FONT>
<BR><FONT SIZE=3D2>the user. That means that you should be using =
app-info headers to </FONT>
<BR><FONT SIZE=3D2>convey the information, for example.</FONT>
</P>

<P><FONT SIZE=3D2>Indeed, there are probably existing xml formats which =
may even do this.</FONT>
</P>

<P><FONT SIZE=3D2>-Jonathan R.</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Rohan Mahy wrote:</FONT>
</P>

<P><FONT SIZE=3D2>&gt; Hi,</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; I wrote the following draft in order to provide =
some discussion on </FONT>
<BR><FONT SIZE=3D2>&gt; alternatives to in-band early-media ringback, =
and to discuss the issue </FONT>
<BR><FONT SIZE=3D2>&gt; of playing tones in general which I believe is =
inherently linked to 3pcc </FONT>
<BR><FONT SIZE=3D2>&gt; and application interaction.</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; <A =
HREF=3D"http://www.softarmor.com/sipping/drafts/draft-mahy-sipping-tones=
-00.html" =
TARGET=3D"_blank">http://www.softarmor.com/sipping/drafts/draft-mahy-sip=
ping-tones-00.html</A></FONT>
<BR><FONT SIZE=3D2>&gt; <A =
HREF=3D"http://www.softarmor.com/sipping/drafts/draft-mahy-sipping-tones=
-00.txt" =
TARGET=3D"_blank">http://www.softarmor.com/sipping/drafts/draft-mahy-sip=
ping-tones-00.txt</A></FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; comments welcome.</FONT>
<BR><FONT SIZE=3D2>&gt; thx,</FONT>
<BR><FONT SIZE=3D2>&gt; -r</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
<BR><FONT SIZE=3D2>&gt; =
_______________________________________________</FONT>
<BR><FONT SIZE=3D2>&gt; Sipping mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sipping" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sipping</A></FO=
NT>
<BR><FONT SIZE=3D2>&gt; This list is for NEW development of the =
application of SIP</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip-implementors@cs.columbia.edu for =
questions on current sip</FONT>
<BR><FONT SIZE=3D2>&gt; Use sip@ietf.org for new developments of core =
SIP</FONT>
<BR><FONT SIZE=3D2>&gt; </FONT>
</P>

<P><FONT SIZE=3D2>-- </FONT>
<BR><FONT SIZE=3D2>Jonathan D. Rosenberg, =
Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; 600 Lanidex Plaza</FONT>
<BR><FONT SIZE=3D2>Chief Technology =
Officer&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Parsippany, NJ =
07054-2711</FONT>
<BR><FONT SIZE=3D2>dynamicsoft</FONT>
<BR><FONT =
SIZE=3D2>jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; FAX:&nbsp;&nbsp; (973) 952-5050</FONT>
<BR><FONT SIZE=3D2><A HREF=3D"http://www.jdrosen.net" =
TARGET=3D"_blank">http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: (973) 952-5000</FONT>
<BR><FONT SIZE=3D2><A HREF=3D"http://www.dynamicsoft.com" =
TARGET=3D"_blank">http://www.dynamicsoft.com</A></FONT>
</P>
<BR>

<P><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>Sipping mailing list&nbsp; <A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/sipping" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/sipping</A></FO=
NT>
<BR><FONT SIZE=3D2>This list is for NEW development of the application =
of SIP</FONT>
<BR><FONT SIZE=3D2>Use sip-implementors@cs.columbia.edu for questions =
on current sip</FONT>
<BR><FONT SIZE=3D2>Use sip@ietf.org for new developments of core =
SIP</FONT>
</P>

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_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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Date: Tue, 27 Jan 2004 16:47:22 -0800
Subject: Re: [Sipping] New Tones draft (was: "who plays the tone" or somet hing like that)
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On Monday, January 26, 2004, at 03:40  PM, Brian Stucker wrote:

> Rohan,
>
> Skimmed your draft and have a question... By using the =20
> content-disposition header it seems that we're requiring the =20
> request/response to contain content. In the case where a user agent =20=

> receives an INVITE with no SDP in it (slow start call) it would seem =20=

> that they'll use some sort of default alerting mechanism until they =20=

> get the SDP setup (which may not happen until the 200 OK/ACK). In the =20=

> case of your draft it would seem that this means the client may ring =20=

> one way, then suddenly switch and ring a different way because it's =20=

> not aware that there's going to be some alert information in the =20
> future.

I'm not really sure of the scope of your question, but here goes...

There is no reason you can't send ringing tones in a 18x before an =20
offer/answer exchange has completed.  The two behaviors are independent.

If you want to be able to control traditional early media though, you =20=

really need the UAs to send early offers and answers (send an offer in =20=

the INVITE and an answer in a reliable 18x, or send an offer in a =20
reliable 18x, send an answer in the PRACK).

I hope this answers your question.

thanks,
-rohan

> Am I correct, and if not, how does the draft approach this problem?
>
> Regards,
>
> Brian
>
> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Wednesday, July 16, 2003 7:08 PM
> To: Rohan Mahy
> Cc: 'sipping@ietf.org'
> Subject: Re: [Sipping] New Tones draft (was: "who plays the tone" or
> something like that)
>
>
> Rohan,
>
> I just got around to reading this draft. I think its neat. You say in
> your note below that its linked to app-interaction, but you don't use
> the app-interaction framework. I believe this is precisely the kind of
> thing we want to do with app-interaction - its a stimulus interface to
> the user. That means that you should be using app-info headers to
> convey the information, for example.
>
> Indeed, there are probably existing xml formats which may even do =
this.
>
> -Jonathan R.
>
>
> Rohan Mahy wrote:
>
> > Hi,
> >
> > I wrote the following draft in order to provide some discussion on
> > alternatives to in-band early-media ringback, and to discuss the =20
> issue
> > of playing tones in general which I believe is inherently linked to =20=

> 3pcc
> > and application interaction.
> >
> > =20
> http://www.softarmor.com/sipping/drafts/draft-mahy-sipping-tones-=20
> 00.html
> > =20
> http://www.softarmor.com/sipping/drafts/draft-mahy-sipping-tones-> =
00.txt
> >
> > comments welcome.
> > thx,
> > -r
> >
> > _______________________________________________
> > Sipping mailing list=A0 =
https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> >
>
> --
> Jonathan D. Rosenberg, Ph.D.=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
 600 Lanidex Plaza
> Chief Technology Officer=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0 Parsippany, NJ 07054-2711
> dynamicsoft
> jdrosen@dynamicsoft.com=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0 FAX:=A0=A0 (973) 952-5050
> http://www.jdrosen.net=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0 PHONE: (973) 952-5000
> http://www.dynamicsoft.com
>
>
> _______________________________________________
> Sipping mailing list=A0 https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
>

--Apple-Mail-4-306878656
Content-Type: text/enriched;
	charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable



On Monday, January 26, 2004, at 03:40  PM, Brian Stucker wrote:


<excerpt><smaller>Rohan,</smaller>


<smaller>Skimmed your draft and have a question... By using the
content-disposition header it seems that we're requiring the
request/response to contain content. In the case where a user agent
receives an INVITE with no SDP in it (slow start call) it would seem
that they'll use some sort of default alerting mechanism until they
get the SDP setup (which may not happen until the 200 OK/ACK). In the
case of your draft it would seem that this means the client may ring
one way, then suddenly switch and ring a different way because it's
not aware that there's going to be some alert information in the
future.</smaller>

</excerpt>

I'm not really sure of the scope of your question, but here goes...


There is no reason you can't send ringing tones in a 18x before an
offer/answer exchange has completed.  The two behaviors are
independent.


If you want to be able to control traditional early media though, you
really need the UAs to send early offers and answers (send an offer in
the INVITE and an answer in a reliable 18x, or send an offer in a
reliable 18x, send an answer in the PRACK).


I hope this answers your question.


thanks,

-rohan


<excerpt><smaller>Am I correct, and if not, how does the draft
approach this problem?</smaller>


<smaller>Regards,</smaller>


<smaller>Brian</smaller>


<smaller>-----Original Message-----

From: Jonathan Rosenberg
=
[<underline><color><param>1999,1999,FFFF</param>mailto:jdrosen@dynamicsoft=
.com</color></underline>]

Sent: Wednesday, July 16, 2003 7:08 PM

To: Rohan Mahy

Cc: 'sipping@ietf.org'

Subject: Re: [Sipping] New Tones draft (was: "who plays the tone" or

something like that)</smaller>



<smaller>Rohan,</smaller>


<smaller>I just got around to reading this draft. I think its neat.
You say in

your note below that its linked to app-interaction, but you don't use

the app-interaction framework. I believe this is precisely the kind of

thing we want to do with app-interaction - its a stimulus interface to

the user. That means that you should be using app-info headers to

convey the information, for example.</smaller>


<smaller>Indeed, there are probably existing xml formats which may
even do this.</smaller>


<smaller>-Jonathan R.</smaller>



<smaller>Rohan Mahy wrote:</smaller>


<smaller>> Hi,

>

> I wrote the following draft in order to provide some discussion on

> alternatives to in-band early-media ringback, and to discuss the
issue

> of playing tones in general which I believe is inherently linked to
3pcc

> and application interaction.

>

>
=
<underline><color><param>1999,1999,FFFF</param>http://www.softarmor.com/si=
pping/drafts/draft-mahy-sipping-tones-00.html

</color></underline>>
=
<underline><color><param>1999,1999,FFFF</param>http://www.softarmor.com/si=
pping/drafts/draft-mahy-sipping-tones-00.txt

</color></underline>>

> comments welcome.

> thx,

> -r

>

> _______________________________________________

> Sipping mailing list=A0
=
<underline><color><param>1999,1999,FFFF</param>https://www1.ietf.org/mailm=
an/listinfo/sipping

</color></underline>> This list is for NEW development of the
application of SIP

> Use sip-implementors@cs.columbia.edu for questions on current sip

> Use sip@ietf.org for new developments of core SIP

></smaller>


<smaller>--

Jonathan D. Rosenberg, Ph.D.=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 =
600 Lanidex Plaza

Chief Technology Officer=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0 Parsippany, NJ 07054-2711

dynamicsoft

jdrosen@dynamicsoft.com=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0 FAX:=A0=A0 (973) 952-5050

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<smaller>_______________________________________________

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</color></underline>This list is for NEW development of the
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Use sip@ietf.org for new developments of core SIP</smaller>


</excerpt>=

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-----Original Message-----
From: Rohan Mahy [mailto:rohan@cisco.com]
Sent: Tuesday, January 27, 2004 6:47 PM
To: Stucker, Brian [NGC:B617:EXCH]
Cc: Jonathan Rosenberg; 'sipping@ietf.org'
Subject: Re: [Sipping] New Tones draft (was: "who plays the tone" or somet
hing like that)





On Monday, January 26, 2004, at 03:40 PM, Brian Stucker wrote: 


Rohan, 


Skimmed your draft and have a question... By using the content-disposition
header it seems that we're requiring the request/response to contain
content. In the case where a user agent receives an INVITE with no SDP in it
(slow start call) it would seem that they'll use some sort of default
alerting mechanism until they get the SDP setup (which may not happen until
the 200 OK/ACK). In the case of your draft it would seem that this means the
client may ring one way, then suddenly switch and ring a different way
because it's not aware that there's going to be some alert information in
the future. 


I'm not really sure of the scope of your question, but here goes... 


There is no reason you can't send ringing tones in a 18x before an
offer/answer exchange has completed. The two behaviors are independent. 
[Stucker, Brian [NGC:B617:EXCH]]  Oh, I agree that this is acceptable. What
I was referring to are clients that start alerting the moment they get the
INVITE (the terminator, not the originator). Basically, using your draft to
ring the terminator in different ways, and not to overcome the extra early
media sessions. 



If you want to be able to control traditional early media though, you really
need the UAs to send early offers and answers (send an offer in the INVITE
and an answer in a reliable 18x, or send an offer in a reliable 18x, send an
answer in the PRACK). 


I hope this answers your question. 


thanks, 

-rohan 


Am I correct, and if not, how does the draft approach this problem? 


Regards, 


Brian 


-----Original Message----- 

From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com] 

Sent: Wednesday, July 16, 2003 7:08 PM 

To: Rohan Mahy 

Cc: 'sipping@ietf.org' 

Subject: Re: [Sipping] New Tones draft (was: "who plays the tone" or 

something like that) 



Rohan, 


I just got around to reading this draft. I think its neat. You say in 

your note below that its linked to app-interaction, but you don't use 

the app-interaction framework. I believe this is precisely the kind of 

thing we want to do with app-interaction - its a stimulus interface to 

the user. That means that you should be using app-info headers to 

convey the information, for example. 


Indeed, there are probably existing xml formats which may even do this. 


-Jonathan R. 



Rohan Mahy wrote: 


> Hi, 

> 

> I wrote the following draft in order to provide some discussion on 

> alternatives to in-band early-media ringback, and to discuss the issue 

> of playing tones in general which I believe is inherently linked to 3pcc 

> and application interaction. 

> 

> http://www.softarmor.com/sipping/drafts/draft-mahy-sipping-tones-00.html 

> http://www.softarmor.com/sipping/drafts/draft-mahy-sipping-tones-00.txt 

> 

> comments welcome. 

> thx, 

> -r 

> 

> _______________________________________________ 

> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping 

> This list is for NEW development of the application of SIP 

> Use sip-implementors@cs.columbia.edu for questions on current sip 

> Use sip@ietf.org for new developments of core SIP 

> 


-- 

Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza 

Chief Technology Officer                    Parsippany, NJ 07054-2711 

dynamicsoft 

jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050 

http://www.jdrosen.net                      PHONE: (973) 952-5000 

http://www.dynamicsoft.com 



_______________________________________________ 

Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping 

This list is for NEW development of the application of SIP 

Use sip-implementors@cs.columbia.edu for questions on current sip 

Use sip@ietf.org for new developments of core SIP 



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<META content="MSHTML 6.00.2800.1264" name=GENERATOR></HEAD>
<BODY>
<DIV><FONT face=Arial color=#0000ff size=2></FONT>&nbsp;</DIV>
<BLOCKQUOTE>
  <DIV class=OutlookMessageHeader dir=ltr align=left><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Rohan Mahy 
  [mailto:rohan@cisco.com]<BR><B>Sent:</B> Tuesday, January 27, 2004 6:47 
  PM<BR><B>To:</B> Stucker, Brian [NGC:B617:EXCH]<BR><B>Cc:</B> Jonathan 
  Rosenberg; 'sipping@ietf.org'<BR><B>Subject:</B> Re: [Sipping] New Tones draft 
  (was: "who plays the tone" or somet hing like 
  that)<BR><BR></FONT></DIV><BR><BR>
  <P>On Monday, January 26, 2004, at 03:40 PM, Brian Stucker wrote: </P><BR>
  <P><FONT size=2>Rohan,</FONT> </P><BR>
  <P><FONT size=2>Skimmed your draft and have a question... By using the 
  content-disposition header it seems that we're requiring the request/response 
  to contain content. In the case where a user agent receives an INVITE with no 
  SDP in it (slow start call) it would seem that they'll use some sort of 
  default alerting mechanism until they get the SDP setup (which may not happen 
  until the 200 OK/ACK). In the case of your draft it would seem that this means 
  the client may ring one way, then suddenly switch and ring a different way 
  because it's not aware that there's going to be some alert information in the 
  future.</FONT> </P><BR>
  <P>I'm not really sure of the scope of your question, but here goes... 
</P><BR>
  <P>There is no reason you can't send ringing tones in a 18x before an 
  offer/answer exchange has completed. The two behaviors are 
  independent.&nbsp;<BR><SPAN class=789155200-28012004><FONT face=Arial 
  color=#0000ff size=2>[Stucker, Brian [NGC:B617:EXCH]]&nbsp; Oh, I agree that 
  this&nbsp;is acceptable. What I was referring to are clients that&nbsp;start 
  alerting the moment they get the INVITE (the terminator, not the 
  originator).&nbsp;Basically,&nbsp;using your draft to ring the terminator in 
  different ways, and not to overcome the extra early media 
  sessions.&nbsp;</FONT></SPAN><BR></P><BR>
  <P>If you want to be able to control traditional early media though, you 
  really need the UAs to send early offers and answers (send an offer in the 
  INVITE and an answer in a reliable 18x, or send an offer in a reliable 18x, 
  send an answer in the PRACK). </P><BR>
  <P>I hope this answers your question. </P><BR>
  <P>thanks, </P>
  <P>-rohan </P><BR>
  <P><FONT size=2>Am I correct, and if not, how does the draft approach this 
  problem?</FONT> </P><BR>
  <P><FONT size=2>Regards,</FONT> </P><BR>
  <P><FONT size=2>Brian</FONT> </P><BR>
  <P><FONT size=2>-----Original Message----- </FONT></P>
  <P><FONT size=2>From: Jonathan Rosenberg [<U></U></FONT><U><FONT color=#199ff0 
  size=2>mailto:jdrosen@dynamicsoft.com</FONT></U><U></U><FONT size=2>] 
  </FONT></P>
  <P><FONT size=2>Sent: Wednesday, July 16, 2003 7:08 PM </FONT></P>
  <P><FONT size=2>To: Rohan Mahy </FONT></P>
  <P><FONT size=2>Cc: 'sipping@ietf.org' </FONT></P>
  <P><FONT size=2>Subject: Re: [Sipping] New Tones draft (was: "who plays the 
  tone" or </FONT></P>
  <P><FONT size=2>something like that)</FONT> </P><BR><BR>
  <P><FONT size=2>Rohan,</FONT> </P><BR>
  <P><FONT size=2>I just got around to reading this draft. I think its neat. You 
  say in </FONT></P>
  <P><FONT size=2>your note below that its linked to app-interaction, but you 
  don't use </FONT></P>
  <P><FONT size=2>the app-interaction framework. I believe this is precisely the 
  kind of </FONT></P>
  <P><FONT size=2>thing we want to do with app-interaction - its a stimulus 
  interface to </FONT></P>
  <P><FONT size=2>the user. That means that you should be using app-info headers 
  to </FONT></P>
  <P><FONT size=2>convey the information, for example.</FONT> </P><BR>
  <P><FONT size=2>Indeed, there are probably existing xml formats which may even 
  do this.</FONT> </P><BR>
  <P><FONT size=2>-Jonathan R.</FONT> </P><BR><BR>
  <P><FONT size=2>Rohan Mahy wrote:</FONT> </P><BR>
  <P><FONT size=2>&gt; Hi, </FONT></P>
  <P><FONT size=2>&gt; </FONT></P>
  <P><FONT size=2>&gt; I wrote the following draft in order to provide some 
  discussion on </FONT></P>
  <P><FONT size=2>&gt; alternatives to in-band early-media ringback, and to 
  discuss the issue </FONT></P>
  <P><FONT size=2>&gt; of playing tones in general which I believe is inherently 
  linked to 3pcc </FONT></P>
  <P><FONT size=2>&gt; and application interaction. </FONT></P>
  <P><FONT size=2>&gt; </FONT></P>
  <P><FONT size=2>&gt;<U> </U></FONT><U><FONT color=#199ff0 
  size=2>http://www.softarmor.com/sipping/drafts/draft-mahy-sipping-tones-00.html 
  </FONT></U></P>
  <P><FONT size=2>&gt;</FONT><U> <FONT color=#199ff0 
  size=2>http://www.softarmor.com/sipping/drafts/draft-mahy-sipping-tones-00.txt 
  </FONT></U></P>
  <P><FONT size=2>&gt; </FONT></P>
  <P><FONT size=2>&gt; comments welcome. </FONT></P>
  <P><FONT size=2>&gt; thx, </FONT></P>
  <P><FONT size=2>&gt; -r </FONT></P>
  <P><FONT size=2>&gt; </FONT></P>
  <P><FONT size=2>&gt; _______________________________________________ 
  </FONT></P>
  <P><FONT size=2>&gt; Sipping mailing list&nbsp;</FONT><U> <FONT color=#199ff0 
  size=2>https://www1.ietf.org/mailman/listinfo/sipping </FONT></U></P>
  <P><FONT size=2>&gt; This list is for NEW development of the application of 
  SIP </FONT></P>
  <P><FONT size=2>&gt; Use sip-implementors@cs.columbia.edu for questions on 
  current sip </FONT></P>
  <P><FONT size=2>&gt; Use sip@ietf.org for new developments of core SIP 
  </FONT></P>
  <P><FONT size=2>&gt;</FONT> </P><BR>
  <P><FONT size=2>-- </FONT></P>
  <P><FONT size=2>Jonathan D. Rosenberg, 
  Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  600 Lanidex Plaza </FONT></P>
  <P><FONT size=2>Chief Technology 
  Officer&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  Parsippany, NJ 07054-2711 </FONT></P>
  <P><FONT size=2>dynamicsoft </FONT></P>
  <P><FONT 
  size=2>jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  FAX:&nbsp;&nbsp; (973) 952-5050 </FONT></P>
  <P><U><FONT color=#199ff0 size=2>http://www.jdrosen.net</FONT></U><U></U><FONT 
  size=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
  PHONE: (973) 952-5000 </FONT></P>
  <P><U><FONT color=#199ff0 size=2>http://www.dynamicsoft.com</FONT></U><U></U> 
  </P><BR><BR>
  <P><FONT size=2>_______________________________________________ </FONT></P>
  <P><FONT size=2>Sipping mailing list&nbsp;<U> </U></FONT><U><FONT 
  color=#199ff0 size=2>https://www1.ietf.org/mailman/listinfo/sipping 
  </FONT></U></P>
  <P><FONT size=2>This list is for NEW development of the application of SIP 
  </FONT></P>
  <P><FONT size=2>Use sip-implementors@cs.columbia.edu for questions on current 
  sip </FONT></P>
  <P><FONT size=2>Use sip@ietf.org for new developments of core SIP</FONT> 
  </P><BR></BLOCKQUOTE></BODY></HTML>

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To: christer.holmberg@ericsson.com, tu-sawada@kddi.com
Cc: sipping@ietf.org
Subject: Re: [Sipping] RE: Need to clarify rules for offer/answer usage in sip
From: Takuya Sawada <tu-sawada@kddi.com>
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Hi,
> 
> Hi,
> 
> I pretty much agree on Takuya's "summary too". However, my understanding is still that you can only send 
an SDP in a PRACK once (either 18x(offer)/PRACK(answer) or PRACK(offer)/200(answer), as described by Takuya).
 After that, before the INVITE transactions has completed, you'll have to use UPDATE.
> 
In my understanding, you can send PRACK(offer)/200(answer) cycles after the first offer/answer
exchange completes and until INVITE transaction completes.
But if both ends can support UPDATE, I recommend to use UPDATE
for offer/answer cycles rather than PRACK.

Drawbacks of PRACK are;
- If PRACK is rejected due to the offer in PRACK, it would delay the reliable mechanism.
- If it wants to send offer in PRACK, it ahs to wait for the reliable response.

The advantage of PRACK over UPDATE is just saving one message cycle.

But I am not sure I can say it "must" use UPDATE.

Regards,
Takuya

> Regards,
> 
> Christer Holmberg
> Ericsson Finland
> 
> 
> > -----Original Message-----
> > From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> > Sent: 26. tammikuuta 2004 18:43
> > To: Takuya Sawada
> > Cc: Christer Holmberg (JO/LMF); sipping@ietf.org
> > Subject: Need to clarify rules for offer/answer usage in sip
> > 
> > 
> > Takuya,
> > 
> > You make good points. I think you have convinced me of how this all 
> > needs to work. I believe we are also both in agreement that the 
> > necessary behavior is currently underspecified in the RFCs.
> > 
> > I think a useful next step would be to hear from the authors 
> > of the RFCs 
> > in question, and hopefully obtain their concurrence. Then we 
> > can discuss 
> > the best way to formalize this. I am moving this to the 
> > sipping list and 
> > changing the title in order to pick up those authors, some of which 
> > don't normally watch sip-implementors.
> > 
> > 	Paul
> > 
> > Takuya Sawada wrote:
> > > Hi, Paul,
> > > 
> > > Let me summarize my view of "restrictions" defined in the 
> > current SIP RFCs.
> > > * Probably I think I may share the same view with Christer 
> > in most parts.
> > > 
> > > Offer/Answer exchange described in RFC 3264 is independent from 
> > > higher layer protocol.
> > > Higher layer protocol provides means to exchange offer/answer
> > > and this needs to provide means for resolving "glare" situation
> > > as described in the last paragraph in RFC 3264.
> > > 
> > > [How to provide means to exchange offer/answer in SIP] 
> > > 
> > > <RFC 3261>
> > > RFC 3261 provides the following two ways to exchange offer/answer;
> > > - INVITE/2xx
> > > - 2xx/ACK
> > > See the page 80 in RFC 3261.
> > > 
> > > <RFC 3262>
> > > RFC 3262 provides the additional three ways to exchange 
> > offer/answer;
> > > Note that here "1xx" is a reliable provisional response to INVITE.
> > > - INVITE/1xx ... a first reliable 1xx with SDP. (may not be 
> > the first 1xx)
> > > - 1xx/PRACK ... Only when INVITE w/o SDP, 1xx must be the 
> > first reliable response.
> > > - PRACK/2xx(PRACK) ... Only after a completion of 
> > INVITE/1xx or 1xx/PRACK.
> > > See chapter 5 in RFC 3262.
> > > 
> > > <RFC 3311>
> > > RFC 3311 provides an additional way to exchange;
> > > - UPDATE/2xx(UPDATE)
> > > Only when after  any offer/answer exchange defined in RFC 
> > 3261 and RFC 3262
> > > has completed and no offer/answer exchange in progress.
> > > See section 5.1, 5.2 in RFC 3311.
> > > 
> > > Currently, as far as I understand, no other ways are not 
> > defined in any RFCs.
> > > Assuming this is correct, let me clarify how to resolve 
> > "glare" cases.
> > > * I can not imagine how to resolve "glare" if other 
> > offer/answer exchanges
> > >   are possible beyond my assumption here.
> > > 
> > > [How to resolve "glare" condition]
> > > One reason why the usage of a SIP method to exchange offer/answer 
> > > needs to be carefully restricted in RFCs is that UA must 
> > detect mis-ordered
> > > message delivery and must avoid confusion.
> > > In RFC 3264, "it MUST NOT generate a new offer if it has
> > > received an offer which it has not yet answered or rejected.
> > >  it MUST NOT generate a new offer if it has generated a
> > >  prior offer for which it has not yet received an answer or 
> > a rejection."
> > > So now there seems to be two possible message crossing case
> > > to be considered assuming that the above rules are guaranteed.
> > > 
> > > Case1
> > > A                B
> > > |offer1          |
> > > |--------------->|
> > > |         answer1|
> > > |<-----\ /-------|
> > > |       X  offer2|
> > > |<-----/ \-------|
> > > 
> > > Case2
> > > A                B
> > > |offer1    offer2|
> > > |------\ /-------|
> > > |       X        |
> > > |<-----/ \------>|
> > > 
> > > Case1:
> > > Assuming that offer/answer exchange follows the 
> > restrictions I described
> > > above, A can know an SDP in offer2 is not an answer to offer1.
> > > Then offer2 must be rejected with a 500 with Retry-After, when 
> > > offer2 is UPDATE. And I assume that it can apply to reINVITE,
> > > which may not be stated clearly.
> > > I am not sure that when this happens when offer2 is in PRACK.
> > > But probably 500 with Retry-After will also apply, even it 
> > results in
> > > delay for reliabile transfer mechanism for provisional response.
> > > In the case that offer2 is in the first reliable 
> > provisional response
> > > or ACK, it seems to me that it never happens, if there is a 
> > rule that
> > > INVITE, even  w/o SDP, must not be sent if pending offer exists.
> > > I am not sure whether it is clearly stated.
> > > 
> > > Case2:
> > > Assuming that offer/answer exchange follows the 
> > restrictions I described
> > > above, A can know an SDP in offer2 is not an answer to offer1.
> > > Then offer2 must be rejected with a 491, if it is UPDATE or 
> > (re)INVITE.
> > > In the case that offer2 is in PRACK, it may be accepted with 200 or
> > > may be rejected with 491. 491 may be adequate for 
> > offer/answer but it may
> > > delay the completion of a reliable response transfer 
> > mechanism or, in
> > > worst case, may result in failure to complete SIP 
> > transaction because
> > > there is no clear retry rule when PRACK is rejected with 491.
> > > When RFC 3262 was defined, there were no UPDATE, therefore 
> > offer in PRACK
> > > never collides with other offers from the other side. I 
> > guess this is why
> > > there is no rule in RFC 3262 when PRACK offer collides with 
> > UPDATE offer.
> > > (It is safe to use UPDATE instead of PRACK when UPDATE is supported)
> > > Anyway, because there is no PRACK against PRACK collision 
> > in the current 
> > > RFCs, the other side is always expected to send 491.
> > > There is no possibility that offer2 is in the first 
> > reliable response or ACK.
> > > Because A must not send new offer when it send INVITE w/o offer.
> > > 
> > > Is this a correct summary?
> > > 
> > > More comments inline with [TS].
> > > 
> > > 
> > >>Takuya,
> > >>
> > >>RFC 3312 introduces situations in which multiple 
> > offer/answer exchanges 
> > >>are required before a single INVITE transaction can 
> > complete. Of course 
> > >>multiple transactions are involved because PRACK and/or 
> > UPDATE are used 
> > >>to achieve part of the exchange, but the INVITE transaction 
> > can't finish 
> > >>until they are done.
> > >>
> > >>This is at least one case in which offers and answers are 
> > spread across 
> > >>transactions in complex ways.
> > >>
> > > 
> > > [TS] In my reading so far, there is the only one case which 
> > offer/answer is
> > > across transaction, that is offer in the first reliable 
> > provisional response 
> > > of INVITE and answer is in PRACK. There is no other case 
> > that an offer is in
> > > a transaction and an answer is in another transaction.
> > > I agree with that any number of offer/answer exchanges can 
> > occur during
> > > one INVITE transaction.
> > > 
> > > 
> > >>There are also cases independent of 3312 where there is a 
> > need to send a 
> > >>subsequent offer/answer before an invite transaction completes. For 
> > >>instance, this is one of the techniques for dealing with 
> > port conflicts 
> > >>introduced as a result of forking.
> > >>
> > > 
> > > [TS] Yes. I did not disagree with that an offer in UPDATE 
> > request can
> > > be used before the completion of INVITE transaction.
> > > 
> > > 
> > >>Once an invite transaction has completed, including one or more 
> > >>offer/answer pairs, additional offer/answer cycles can be initiated 
> > >>using either reINVITE or UPDATE.
> > >>
> > >>In the case of reINVITE, I believe there is a requirement 
> > that one or 
> > >>more offer/answer cycles be completed before it can 
> > succeed. And if it 
> > >>fails, the changed tentatively imparted by all of those 
> > offer/answer 
> > >>cycles are to be rolled back.
> > >>
> > >>The case of UPDATE is a little less clear. If there is no invite 
> > >>transaction in progress, and an offer is sent in an UPDATE, and the 
> > >>update is rejected, it must be the case that the dialog 
> > state remains as 
> > >>it was before the update. If there is no invite in progress and an 
> > >>UPDATE is sent without an offer, there seems to be no prohibition 
> > >>against sending an offer in the update response. In that case, the 
> > >>answerer would have no way to refuse the offer and retain 
> > the dialog as 
> > >>it is. The only option the answerer would have is to send 
> > another UPDATE 
> > >>containing the answer. I doubt this is intended behavior, 
> > but it isn't 
> > >>prohibited.
> > >>
> > > 
> > > [TS] This is the point I can not agree. I can not find any 
> > description that
> > > UPDATE response can contain an offer in RFC 3311. IMO, this 
> > means UPDATE response
> > > can not have any offer. SIP RFCs define restrictions to 
> > utilize RFC 3264 model.
> > > UAs can not do any offer/answer exchange which is not 
> > defined clearly in any RFCs.
> > > 
> > > 
> > >>More below.
> > >>
> > >>Paul
> > >>
> > >>Takuya Sawada wrote:
> > >>
> > >>>Hi,
> > >>>
> > >>>In my understanding,
> > >>>
> > >>>- Once previous offer/answer completes each side can issue 
> > new offer.
> > >>>- INVITE request contains the offer, and the first 
> > reliable 1xx response
> > >>>  which contains answer, offer/answer for the INVITE 
> > transaction completes.
> > >>>- UPDATE transaction and PRACK transaction can exchange 
> > offer/answer.
> > >>>- Therefore, once the reliable 1xx response which contains 
> > answer is received,
> > >>>  caller can start offer/answer with PRACK or UPDATE or 
> > any new method.
> > >>>  And once the reliable 1xx response which contains answer is sent,
> > >>>  callee can start offer/answer with UPDATE.
> > >>>
> > >>>
> > >>>
> > >>>>P.S. For those with the patience, below I have tried to 
> > enumerate all 
> > >>>>the offer/answer cases thru the 2nd offer. It is pretty painful.
> > >>>>
> > >>>
> > >>><snip>
> > >>>
> > >>>>2) invite with 100rel
> > >>>>   a) offer in invite
> > >>>>      Possible places for answer:
> > >>>>      i)   answer in first reliable 1xx
> > >>>>      ii)  answer in subsequent reliable 1xx
> > >>>>      iii) answer in 2xx
> > >>>>   b) offer not in invite
> > >>>>      offer must be in first reliable 1xx
> > >>>>      answer must be in PRACK to 1xx w/offer
> > >>>>
> > >>>>Multiple offer/answer cycle prior to completion of invite.
> > >>>>Can only happen in subcases of (2) above.
> > >>>>It requires use of UPDATE in some cases.
> > >>>>
> > >>>>3) original offerer sends 2nd offer
> > >>>>   a) following 2.a.i or 2.a.i.i:
> > >>>>      Possible places for 2nd offer:
> > >>>>      i)   offer2 in PRACK of answer1
> > >>>>      ii)  offer2 in PRACK of some subsequent 1xx
> > >>>>      iii) offer2 in UPDATE
> > >>>>   b) following 2.a.iii:
> > >>>>      No 2nd offer possible until after completion of invite.
> > >>>>   c) following 2.b:
> > >>>>      i)   offer2 in 2xx to PRACK w/answer1
> > >>>>      ii)  offer2 in UPDATE
> > >>>>
> > >>>>4) original answerer sends 2nd offer:
> > >>>>   Not clear if this is intended to be legal at all prior to
> > >>>>   completion of invite. It appears existing text permits
> > >>>>   as follows:
> > >>>>   a) following 2.a.i or 2.a.ii:
> > >>>>      Possible places for 2nd offer:
> > >>>>      i)   offer2 in 2xx to prack of answer1.
> > >>>>      ii)  offer2 in subsequent reliable 1xx
> > >>>>           following the one carrying answer1.
> > >>>>      iii) offer2 in UPDATE following answer1.
> > >>>>           (No reason to prefer this to other possibilities.)
> > >>>>   b) following 2.a.iii:
> > >>>>      No 2nd offer possible until after completion of invite.
> > >>>>   c) following 2b:
> > >>>>      i)   if another reliable 1xx is received,
> > >>>>           could send offer2 in the PRACK.
> > >>>>           (but can't count on this happening.)
> > >>>>      ii)  offer2 in UPDATE after sending answer1.
> > >>>
> > >>>
> > >>>I do not think offer/answer exchange beyond the SIP transaction
> > >>>is possible. I am not sure it is prohibited somewhere in RFC, but
> > >>>even if it is not prohibited explicitly, one should not assumed
> > >>>that is possible.
> > >>
> > >>Following the principle of being conservative in what you send and 
> > >>liberal in what you receive, I want to assume that I might receive 
> > >>anything that isn't prohibited, and do something reasonable with it.
> > >>
> > > 
> > > [TS] I love this principle. IMO, if I receive an SDP which 
> > I do not expect
> > > to receive in RFC, one possible solution is that I will 
> > accept SIP message
> > > and ignore the SDP. Content-Disposition header's handling parameter
> > > should be checked, though.
> > > Anyway, here I do not know what is something reasonable to 
> > do with it.
> > > 
> > > 
> > >>I would be willing to be conservative in initiating this stuff if I 
> > >>could figure out what that is. Given the current state of 
> > >>underspecification I don't know that I will draw the same 
> > conclusion 
> > >>than anyone else will.
> > >>
> > > 
> > > [TS] There seems to be some uderspecification in RFCs. But 
> > I want to 
> > > confirm the following;
> > >   RFCs only state where and when an offer and an answer can appear.
> > >   If RFCs do not state "it can", then it MUST NOT contain 
> > offer/answer.
> > >   Because RFCs define "restrictions" where and when offer/answer
> > >  exchange can occur.
> > > First I think we should reach a consensus with this point.
> > > 
> > > 
> > >> > One can achieve the same thing without such a
> > >>
> > >>>tweak. 
> > >>
> > >>Not sure what you are referring to here, or whether it is 
> > negated in 
> > >>your subsequent posting. See above for a number of things 
> > that must be 
> > >>possible.
> > >>
> > >>
> > >>>Then, 3.c.i and 4.a.1 SHOULD NOT, or hopefully MUST NOT, be used.
> > >>
> > >>If we can codify a clear, and complete, set of rules for 
> > where offers 
> > >>and answers can appear, that provides full functionality, and it 
> > >>excludes these cases, then fine.
> > >>
> > >>In the case of 4.a.1, removing it causes no loss of 
> > functionality, so I 
> > >>am inclined to agree. It might ultimately result in exchanging more 
> > >>messages, but the number is probably insignificant.
> > >>
> > >>In the case of 3.c.1, removing it requires that both sides support 
> > >>UPDATE. So its removal may remove functionality in the case 
> > where one 
> > >>side or the other doesn't support UPDATE. This may or may not be 
> > >>significant.
> > >>
> > > 
> > > [TS] As you said, UPDATE can solve the problem.
> > > You can use extensions when it wants the functionality that 
> > the base 
> > > specifications can not provide. This is what the extensions are.
> > > 
> > > 
> > >>>Anyway for INVITE transaction, as explained in my previous mail,
> > >>>it is clear in RFC 3261 to me that one INVITE transaction can 
> > >>>include only one offer/answer exchange.
> > >>>
> > >>>Then 4.a.ii is not permitted.
> > >>
> > >>I think I identified cases above where this is needed.
> > >>
> > > 
> > > [TS] The reason I think it is not permitted is, as Christer 
> > pointed our,
> > > that RFC 3261 does not permit and any other RFCs do not override it.
> > > 
> > > 
> > >>Takuya Sawada wrote:
> > >> > Hi,
> > >> >
> > >> >
> > >> >>I do not think offer/answer exchange beyond the SIP transaction
> > >> >>is possible. I am not sure it is prohibited somewhere in RFC, but
> > >> >>even if it is not prohibited explicitly, one should not assumed
> > >> >>that is possible. One can achieve the same thing without such a
> > >> >>tweak.
> > >> >
> > >> >
> > >> > I am not exactly correct here.
> > >> > When an offer is in INVITE reliable provisonal response and
> > >> > an answer is in PRACK offer/answer exchange is beyond one
> > >> > transaction.
> > >> > What I want to say is that how offer/answer is used in a certain
> > >> > method is explicitly scoped in each RFC which defines the method.
> > >>
> > >>The trouble with these is that they were written in a 
> > certain order. As 
> > >>each was written, a broadening of thinking seems to have 
> > occurred. This 
> > >>is reflected in each, but not necessarily fully reflected 
> > in the earlier 
> > >>specs.
> > >>
> > > 
> > > [TS] I may agree with this. At least this makes it 
> > difficult to understand
> > > and share the same view among all. Actually, there seems to 
> > be a little 
> > > ambiguity with PRACK and UPDATE collision case as I pointed 
> > > out in this mail, etc.
> > > 
> > > 
> > >> > How PRACK is used for offer/answer is defined in RFC 3262.
> > >> > There seems to be only two cases:
> > >> >  - offer/answer is reliable 1xx(INVITE)/PRACK
> > >> >  - offer/answer is PRACK/2xx(PRACK)
> > >> >    only after INVITE's offer/answer exchange complete.
> > >>
> > >>I agree that this seems pretty well locked down as it 
> > relates to 3261. 
> > >>These lock downs of course do apply to the things that came 
> > later. But 
> > >>they seem at least awkward and arbitrary in some cases.
> > >>
> > >> > How UPDATE is used for offer/answer is defined in RFC 3311.
> > >> >  - offer/answer is UPDATE/2xx(UPDATE)
> > >>
> > >>Here I disagree. I looked pretty closely at this, and I 
> > don't see *any* 
> > >>normative language for how offer/answer is used with it. There is a 
> > >>bunch of non-normative language which appears to simply restate 
> > >>implications of 3261 and 3264 in the presence of update.
> > >>
> > > 
> > > [TS] Again, this is the point I can not agree. I agree with 
> > that there is
> > > no normative language to prohibit offer in the response of UPDATE
> > > like "2xx response to UPDATE MUST NOT contain offer".
> > > Actually, response can have an SDP even if UPDATE does not have
> > > an offer SDP. But an SDP can be neither an offer nor an answer 
> > > unless clear semantics are defined in any RFCs. I can not find 
> > > any usage or semantics to use offer/answer exchange beyond
> > > one UPDATE transaction.
> > > IMO, SDP in a 2xx of UPDATE when an UPDATE request w/o
> > > offer is neither an offer nor an answer like SDP in INFO is 
> > > neither an offer nor an answer.
> > > 
> > > 
> > >> > How INVITE is used for offer/answer is defined in RFC 3261.
> > >> >  - offer/answer is INVITE/non-failure reliable response
> > >> >  - offer/answer is the first non-failure first 
> > >>response/acknowlegement for the response
> > >> >    , which is 2xx(INVITE)/ACK in RFC 3261 and 
> > 1xx(INVITE)/PRACK in 
> > >>RFC 3262.
> > >> >  * Note that the second or later SDP in INVITE 
> > non-failure response
> > >> >    is not an offer. It is nothing.
> > >> >
> > >> > There is no other way than explicitly defined in any RFC, IMO.
> > >>
> > >>I believe you are right that nothing more is explicitly defined.
> > >>But 3264 is pretty general and broad. It is not prescriptive in how 
> > >>offers and answers are transmitted. 
> > > 
> > > 
> > > [TS] Yes. Therefore, RFC 3264 requires restrictions on how 
> > > to exchange offer/answer in higher layer protocol.
> > > 
> > > 
> > >>And while the other RFCs may not 
> > >>explicitly specify other ways, there are other ways that aren't 
> > >>explicitly prohibited. In general with sip, things that aren't 
> > >>explicitly prohibited are permitted.
> > >>
> > > 
> > > [TS] Yes. Any SIP messages can contain SDP at any moment. 
> > > But without semantics, it is neither an offer nor an answer.
> > > 
> > > 
> > >>I just want to get this nailed down, so I know how to build 
> > a stack that 
> > >>can both handle every valid case thrown at it, and that can 
> > initiate all 
> > >>the needed functionality.
> > >>
> > >>I am thinking that maybe somebody (probably me) needs to 
> > write a best 
> > >>practices document that pulls this all together, specifying 
> > all the ways 
> > >>that offer/answer can be used. Following that, I think it will be 
> > >>advisable to tighten up one or more of the RFCs in places 
> > where things 
> > >>haven't been prohibited but should be.
> > >>
> > > 
> > > [TS] The guideline document that you are proposing can be useful.
> > > Collecting all the possible collision cases, recommending UPDATE 
> > > offer rather than PRACK offer if UPDATE is supported and no need 
> > > to reduce the number of messages, enumerating Warning header code 
> > > examples to be included when offer is rejected with 488, etc.
> > > 
> > > Regards,
> > > Takuya
> > > 
> > > 
> > >>Paul
> > >>
> > >>_______________________________________________
> > >>Sip-implementors mailing list
> > >>Sip-implementors@cs.columbia.edu
> > >>http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> > > 
> > > 
> > > --------
> > > Takuya Sawada
> > > KDDI Corporation (KDDI)
> > > Garden Air Tower, 3-10-10  Iidabashi Chiyoda-ku
> > > Tokyo 102-8460, Japan
> > > Tel: +81-3-6678-2997
> > > Fax: +81-3-6678-0285
> > > tu-sawada@kddi.com
> > > 
> > 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP


--------
Takuya Sawada
KDDI Corporation (KDDI)
Garden Air Tower, 3-10-10  Iidabashi Chiyoda-ku
Tokyo 102-8460, Japan
Tel: +81-3-6678-2997
Fax: +81-3-6678-0285
tu-sawada@kddi.com

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



From exim@www1.ietf.org  Tue Jan 27 21:50:36 2004
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To: pkyzivat@cisco.com, tu-sawada@kddi.com
Cc: sipping@ietf.org
Subject: Re: [Sipping] Need to clarify rules for offer/answer usage in sip
From: Takuya Sawada <tu-sawada@kddi.com>
References: <4010591C.20906@cisco.com>
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Hi,

Though, it has already pointed out and corrected in sip-implementors mailing list,
I want to add one correction to my summary below here for sure.

I wrote;
"In the case that offer2 is in the first reliable provisional response or ACK,"
the correct words are
"In the case that offer2 is in the first reliable provisional response or 2xx,"

And also I wrote;
"There is no possibility that offer2 is in the first reliable response or ACK."
I wanted to say
"There is no possibility that offer2 is in the first reliable response or 2xx."

I think we can agree on the most parts, it may need some clarifications, thogh,
if we could read through the multiple documents carefully.
But some may think differently and implemente their interpretation on their
devices. It seems to be dangerous.
Paul's proposal, I think, would be formalizing this as a document like
call flow examples documents. Some people may consider this as a useful work
to desrve doing this. Some may not.

Regards,
Takuya

> Takuya,
> 
> You make good points. I think you have convinced me of how this all 
> needs to work. I believe we are also both in agreement that the 
> necessary behavior is currently underspecified in the RFCs.
> 
> I think a useful next step would be to hear from the authors of the RFCs 
> in question, and hopefully obtain their concurrence. Then we can discuss 
> the best way to formalize this. I am moving this to the sipping list and 
> changing the title in order to pick up those authors, some of which 
> don't normally watch sip-implementors.
> 
> 	Paul
> 
> Takuya Sawada wrote:
> > Hi, Paul,
> > 
> > Let me summarize my view of "restrictions" defined in the current SIP RFCs.
> > * Probably I think I may share the same view with Christer in most parts.
> > 
> > Offer/Answer exchange described in RFC 3264 is independent from 
> > higher layer protocol.
> > Higher layer protocol provides means to exchange offer/answer
> > and this needs to provide means for resolving "glare" situation
> > as described in the last paragraph in RFC 3264.
> > 
> > [How to provide means to exchange offer/answer in SIP] 
> > 
> > <RFC 3261>
> > RFC 3261 provides the following two ways to exchange offer/answer;
> > - INVITE/2xx
> > - 2xx/ACK
> > See the page 80 in RFC 3261.
> > 
> > <RFC 3262>
> > RFC 3262 provides the additional three ways to exchange offer/answer;
> > Note that here "1xx" is a reliable provisional response to INVITE.
> > - INVITE/1xx ... a first reliable 1xx with SDP. (may not be the first 1xx)
> > - 1xx/PRACK ... Only when INVITE w/o SDP, 1xx must be the first reliable response.
> > - PRACK/2xx(PRACK) ... Only after a completion of INVITE/1xx or 1xx/PRACK.
> > See chapter 5 in RFC 3262.
> > 
> > <RFC 3311>
> > RFC 3311 provides an additional way to exchange;
> > - UPDATE/2xx(UPDATE)
> > Only when after  any offer/answer exchange defined in RFC 3261 and RFC 3262
> > has completed and no offer/answer exchange in progress.
> > See section 5.1, 5.2 in RFC 3311.
> > 
> > Currently, as far as I understand, no other ways are not defined in any RFCs.
> > Assuming this is correct, let me clarify how to resolve "glare" cases.
> > * I can not imagine how to resolve "glare" if other offer/answer exchanges
> >   are possible beyond my assumption here.
> > 
> > [How to resolve "glare" condition]
> > One reason why the usage of a SIP method to exchange offer/answer 
> > needs to be carefully restricted in RFCs is that UA must detect mis-ordered
> > message delivery and must avoid confusion.
> > In RFC 3264, "it MUST NOT generate a new offer if it has
> > received an offer which it has not yet answered or rejected.
> >  it MUST NOT generate a new offer if it has generated a
> >  prior offer for which it has not yet received an answer or a rejection."
> > So now there seems to be two possible message crossing case
> > to be considered assuming that the above rules are guaranteed.
> > 
> > Case1
> > A                B
> > |offer1          |
> > |--------------->|
> > |         answer1|
> > |<-----\ /-------|
> > |       X  offer2|
> > |<-----/ \-------|
> > 
> > Case2
> > A                B
> > |offer1    offer2|
> > |------\ /-------|
> > |       X        |
> > |<-----/ \------>|
> > 
> > Case1:
> > Assuming that offer/answer exchange follows the restrictions I described
> > above, A can know an SDP in offer2 is not an answer to offer1.
> > Then offer2 must be rejected with a 500 with Retry-After, when 
> > offer2 is UPDATE. And I assume that it can apply to reINVITE,
> > which may not be stated clearly.
> > I am not sure that when this happens when offer2 is in PRACK.
> > But probably 500 with Retry-After will also apply, even it results in
> > delay for reliabile transfer mechanism for provisional response.
> > In the case that offer2 is in the first reliable provisional response
> > or ACK, it seems to me that it never happens, if there is a rule that
> > INVITE, even  w/o SDP, must not be sent if pending offer exists.
> > I am not sure whether it is clearly stated.
> > 
> > Case2:
> > Assuming that offer/answer exchange follows the restrictions I described
> > above, A can know an SDP in offer2 is not an answer to offer1.
> > Then offer2 must be rejected with a 491, if it is UPDATE or (re)INVITE.
> > In the case that offer2 is in PRACK, it may be accepted with 200 or
> > may be rejected with 491. 491 may be adequate for offer/answer but it may
> > delay the completion of a reliable response transfer mechanism or, in
> > worst case, may result in failure to complete SIP transaction because
> > there is no clear retry rule when PRACK is rejected with 491.
> > When RFC 3262 was defined, there were no UPDATE, therefore offer in PRACK
> > never collides with other offers from the other side. I guess this is why
> > there is no rule in RFC 3262 when PRACK offer collides with UPDATE offer.
> > (It is safe to use UPDATE instead of PRACK when UPDATE is supported)
> > Anyway, because there is no PRACK against PRACK collision in the current 
> > RFCs, the other side is always expected to send 491.
> > There is no possibility that offer2 is in the first reliable response or ACK.
> > Because A must not send new offer when it send INVITE w/o offer.
> > 
> > Is this a correct summary?
> > 
> > More comments inline with [TS].
> > 
> > 
> >>Takuya,
> >>
> >>RFC 3312 introduces situations in which multiple offer/answer exchanges 
> >>are required before a single INVITE transaction can complete. Of course 
> >>multiple transactions are involved because PRACK and/or UPDATE are used 
> >>to achieve part of the exchange, but the INVITE transaction can't finish 
> >>until they are done.
> >>
> >>This is at least one case in which offers and answers are spread across 
> >>transactions in complex ways.
> >>
> > 
> > [TS] In my reading so far, there is the only one case which offer/answer is
> > across transaction, that is offer in the first reliable provisional response 
> > of INVITE and answer is in PRACK. There is no other case that an offer is in
> > a transaction and an answer is in another transaction.
> > I agree with that any number of offer/answer exchanges can occur during
> > one INVITE transaction.
> > 
> > 
> >>There are also cases independent of 3312 where there is a need to send a 
> >>subsequent offer/answer before an invite transaction completes. For 
> >>instance, this is one of the techniques for dealing with port conflicts 
> >>introduced as a result of forking.
> >>
> > 
> > [TS] Yes. I did not disagree with that an offer in UPDATE request can
> > be used before the completion of INVITE transaction.
> > 
> > 
> >>Once an invite transaction has completed, including one or more 
> >>offer/answer pairs, additional offer/answer cycles can be initiated 
> >>using either reINVITE or UPDATE.
> >>
> >>In the case of reINVITE, I believe there is a requirement that one or 
> >>more offer/answer cycles be completed before it can succeed. And if it 
> >>fails, the changed tentatively imparted by all of those offer/answer 
> >>cycles are to be rolled back.
> >>
> >>The case of UPDATE is a little less clear. If there is no invite 
> >>transaction in progress, and an offer is sent in an UPDATE, and the 
> >>update is rejected, it must be the case that the dialog state remains as 
> >>it was before the update. If there is no invite in progress and an 
> >>UPDATE is sent without an offer, there seems to be no prohibition 
> >>against sending an offer in the update response. In that case, the 
> >>answerer would have no way to refuse the offer and retain the dialog as 
> >>it is. The only option the answerer would have is to send another UPDATE 
> >>containing the answer. I doubt this is intended behavior, but it isn't 
> >>prohibited.
> >>
> > 
> > [TS] This is the point I can not agree. I can not find any description that
> > UPDATE response can contain an offer in RFC 3311. IMO, this means UPDATE response
> > can not have any offer. SIP RFCs define restrictions to utilize RFC 3264 model.
> > UAs can not do any offer/answer exchange which is not defined clearly in any RFCs.
> > 
> > 
> >>More below.
> >>
> >>Paul
> >>
> >>Takuya Sawada wrote:
> >>
> >>>Hi,
> >>>
> >>>In my understanding,
> >>>
> >>>- Once previous offer/answer completes each side can issue new offer.
> >>>- INVITE request contains the offer, and the first reliable 1xx response
> >>>  which contains answer, offer/answer for the INVITE transaction completes.
> >>>- UPDATE transaction and PRACK transaction can exchange offer/answer.
> >>>- Therefore, once the reliable 1xx response which contains answer is received,
> >>>  caller can start offer/answer with PRACK or UPDATE or any new method.
> >>>  And once the reliable 1xx response which contains answer is sent,
> >>>  callee can start offer/answer with UPDATE.
> >>>
> >>>
> >>>
> >>>>P.S. For those with the patience, below I have tried to enumerate all 
> >>>>the offer/answer cases thru the 2nd offer. It is pretty painful.
> >>>>
> >>>
> >>><snip>
> >>>
> >>>>2) invite with 100rel
> >>>>   a) offer in invite
> >>>>      Possible places for answer:
> >>>>      i)   answer in first reliable 1xx
> >>>>      ii)  answer in subsequent reliable 1xx
> >>>>      iii) answer in 2xx
> >>>>   b) offer not in invite
> >>>>      offer must be in first reliable 1xx
> >>>>      answer must be in PRACK to 1xx w/offer
> >>>>
> >>>>Multiple offer/answer cycle prior to completion of invite.
> >>>>Can only happen in subcases of (2) above.
> >>>>It requires use of UPDATE in some cases.
> >>>>
> >>>>3) original offerer sends 2nd offer
> >>>>   a) following 2.a.i or 2.a.i.i:
> >>>>      Possible places for 2nd offer:
> >>>>      i)   offer2 in PRACK of answer1
> >>>>      ii)  offer2 in PRACK of some subsequent 1xx
> >>>>      iii) offer2 in UPDATE
> >>>>   b) following 2.a.iii:
> >>>>      No 2nd offer possible until after completion of invite.
> >>>>   c) following 2.b:
> >>>>      i)   offer2 in 2xx to PRACK w/answer1
> >>>>      ii)  offer2 in UPDATE
> >>>>
> >>>>4) original answerer sends 2nd offer:
> >>>>   Not clear if this is intended to be legal at all prior to
> >>>>   completion of invite. It appears existing text permits
> >>>>   as follows:
> >>>>   a) following 2.a.i or 2.a.ii:
> >>>>      Possible places for 2nd offer:
> >>>>      i)   offer2 in 2xx to prack of answer1.
> >>>>      ii)  offer2 in subsequent reliable 1xx
> >>>>           following the one carrying answer1.
> >>>>      iii) offer2 in UPDATE following answer1.
> >>>>           (No reason to prefer this to other possibilities.)
> >>>>   b) following 2.a.iii:
> >>>>      No 2nd offer possible until after completion of invite.
> >>>>   c) following 2b:
> >>>>      i)   if another reliable 1xx is received,
> >>>>           could send offer2 in the PRACK.
> >>>>           (but can't count on this happening.)
> >>>>      ii)  offer2 in UPDATE after sending answer1.
> >>>
> >>>
> >>>I do not think offer/answer exchange beyond the SIP transaction
> >>>is possible. I am not sure it is prohibited somewhere in RFC, but
> >>>even if it is not prohibited explicitly, one should not assumed
> >>>that is possible.
> >>
> >>Following the principle of being conservative in what you send and 
> >>liberal in what you receive, I want to assume that I might receive 
> >>anything that isn't prohibited, and do something reasonable with it.
> >>
> > 
> > [TS] I love this principle. IMO, if I receive an SDP which I do not expect
> > to receive in RFC, one possible solution is that I will accept SIP message
> > and ignore the SDP. Content-Disposition header's handling parameter
> > should be checked, though.
> > Anyway, here I do not know what is something reasonable to do with it.
> > 
> > 
> >>I would be willing to be conservative in initiating this stuff if I 
> >>could figure out what that is. Given the current state of 
> >>underspecification I don't know that I will draw the same conclusion 
> >>than anyone else will.
> >>
> > 
> > [TS] There seems to be some uderspecification in RFCs. But I want to 
> > confirm the following;
> >   RFCs only state where and when an offer and an answer can appear.
> >   If RFCs do not state "it can", then it MUST NOT contain offer/answer.
> >   Because RFCs define "restrictions" where and when offer/answer
> >  exchange can occur.
> > First I think we should reach a consensus with this point.
> > 
> > 
> >> > One can achieve the same thing without such a
> >>
> >>>tweak. 
> >>
> >>Not sure what you are referring to here, or whether it is negated in 
> >>your subsequent posting. See above for a number of things that must be 
> >>possible.
> >>
> >>
> >>>Then, 3.c.i and 4.a.1 SHOULD NOT, or hopefully MUST NOT, be used.
> >>
> >>If we can codify a clear, and complete, set of rules for where offers 
> >>and answers can appear, that provides full functionality, and it 
> >>excludes these cases, then fine.
> >>
> >>In the case of 4.a.1, removing it causes no loss of functionality, so I 
> >>am inclined to agree. It might ultimately result in exchanging more 
> >>messages, but the number is probably insignificant.
> >>
> >>In the case of 3.c.1, removing it requires that both sides support 
> >>UPDATE. So its removal may remove functionality in the case where one 
> >>side or the other doesn't support UPDATE. This may or may not be 
> >>significant.
> >>
> > 
> > [TS] As you said, UPDATE can solve the problem.
> > You can use extensions when it wants the functionality that the base 
> > specifications can not provide. This is what the extensions are.
> > 
> > 
> >>>Anyway for INVITE transaction, as explained in my previous mail,
> >>>it is clear in RFC 3261 to me that one INVITE transaction can 
> >>>include only one offer/answer exchange.
> >>>
> >>>Then 4.a.ii is not permitted.
> >>
> >>I think I identified cases above where this is needed.
> >>
> > 
> > [TS] The reason I think it is not permitted is, as Christer pointed our,
> > that RFC 3261 does not permit and any other RFCs do not override it.
> > 
> > 
> >>Takuya Sawada wrote:
> >> > Hi,
> >> >
> >> >
> >> >>I do not think offer/answer exchange beyond the SIP transaction
> >> >>is possible. I am not sure it is prohibited somewhere in RFC, but
> >> >>even if it is not prohibited explicitly, one should not assumed
> >> >>that is possible. One can achieve the same thing without such a
> >> >>tweak.
> >> >
> >> >
> >> > I am not exactly correct here.
> >> > When an offer is in INVITE reliable provisonal response and
> >> > an answer is in PRACK offer/answer exchange is beyond one
> >> > transaction.
> >> > What I want to say is that how offer/answer is used in a certain
> >> > method is explicitly scoped in each RFC which defines the method.
> >>
> >>The trouble with these is that they were written in a certain order. As 
> >>each was written, a broadening of thinking seems to have occurred. This 
> >>is reflected in each, but not necessarily fully reflected in the earlier 
> >>specs.
> >>
> > 
> > [TS] I may agree with this. At least this makes it difficult to understand
> > and share the same view among all. Actually, there seems to be a little 
> > ambiguity with PRACK and UPDATE collision case as I pointed 
> > out in this mail, etc.
> > 
> > 
> >> > How PRACK is used for offer/answer is defined in RFC 3262.
> >> > There seems to be only two cases:
> >> >  - offer/answer is reliable 1xx(INVITE)/PRACK
> >> >  - offer/answer is PRACK/2xx(PRACK)
> >> >    only after INVITE's offer/answer exchange complete.
> >>
> >>I agree that this seems pretty well locked down as it relates to 3261. 
> >>These lock downs of course do apply to the things that came later. But 
> >>they seem at least awkward and arbitrary in some cases.
> >>
> >> > How UPDATE is used for offer/answer is defined in RFC 3311.
> >> >  - offer/answer is UPDATE/2xx(UPDATE)
> >>
> >>Here I disagree. I looked pretty closely at this, and I don't see *any* 
> >>normative language for how offer/answer is used with it. There is a 
> >>bunch of non-normative language which appears to simply restate 
> >>implications of 3261 and 3264 in the presence of update.
> >>
> > 
> > [TS] Again, this is the point I can not agree. I agree with that there is
> > no normative language to prohibit offer in the response of UPDATE
> > like "2xx response to UPDATE MUST NOT contain offer".
> > Actually, response can have an SDP even if UPDATE does not have
> > an offer SDP. But an SDP can be neither an offer nor an answer 
> > unless clear semantics are defined in any RFCs. I can not find 
> > any usage or semantics to use offer/answer exchange beyond
> > one UPDATE transaction.
> > IMO, SDP in a 2xx of UPDATE when an UPDATE request w/o
> > offer is neither an offer nor an answer like SDP in INFO is 
> > neither an offer nor an answer.
> > 
> > 
> >> > How INVITE is used for offer/answer is defined in RFC 3261.
> >> >  - offer/answer is INVITE/non-failure reliable response
> >> >  - offer/answer is the first non-failure first 
> >>response/acknowlegement for the response
> >> >    , which is 2xx(INVITE)/ACK in RFC 3261 and 1xx(INVITE)/PRACK in 
> >>RFC 3262.
> >> >  * Note that the second or later SDP in INVITE non-failure response
> >> >    is not an offer. It is nothing.
> >> >
> >> > There is no other way than explicitly defined in any RFC, IMO.
> >>
> >>I believe you are right that nothing more is explicitly defined.
> >>But 3264 is pretty general and broad. It is not prescriptive in how 
> >>offers and answers are transmitted. 
> > 
> > 
> > [TS] Yes. Therefore, RFC 3264 requires restrictions on how 
> > to exchange offer/answer in higher layer protocol.
> > 
> > 
> >>And while the other RFCs may not 
> >>explicitly specify other ways, there are other ways that aren't 
> >>explicitly prohibited. In general with sip, things that aren't 
> >>explicitly prohibited are permitted.
> >>
> > 
> > [TS] Yes. Any SIP messages can contain SDP at any moment. 
> > But without semantics, it is neither an offer nor an answer.
> > 
> > 
> >>I just want to get this nailed down, so I know how to build a stack that 
> >>can both handle every valid case thrown at it, and that can initiate all 
> >>the needed functionality.
> >>
> >>I am thinking that maybe somebody (probably me) needs to write a best 
> >>practices document that pulls this all together, specifying all the ways 
> >>that offer/answer can be used. Following that, I think it will be 
> >>advisable to tighten up one or more of the RFCs in places where things 
> >>haven't been prohibited but should be.
> >>
> > 
> > [TS] The guideline document that you are proposing can be useful.
> > Collecting all the possible collision cases, recommending UPDATE 
> > offer rather than PRACK offer if UPDATE is supported and no need 
> > to reduce the number of messages, enumerating Warning header code 
> > examples to be included when offer is rejected with 488, etc.
> > 
> > Regards,
> > Takuya
> > 
> > 
> >>Paul
> >>
> >>_______________________________________________
> >>Sip-implementors mailing list
> >>Sip-implementors@cs.columbia.edu
> >>http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> > 
> > 
> > --------
> > Takuya Sawada
> > KDDI Corporation (KDDI)
> > Garden Air Tower, 3-10-10  Iidabashi Chiyoda-ku
> > Tokyo 102-8460, Japan
> > Tel: +81-3-6678-2997
> > Fax: +81-3-6678-0285
> > tu-sawada@kddi.com
> > 
> 
> 
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP


--------
Takuya Sawada
KDDI Corporation (KDDI)
Garden Air Tower, 3-10-10  Iidabashi Chiyoda-ku
Tokyo 102-8460, Japan
Tel: +81-3-6678-2997
Fax: +81-3-6678-0285
tu-sawada@kddi.com

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP



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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "'Takuya Sawada'" <tu-sawada@kddi.com>, tu-sawada@kddi.com
Cc: sipping@ietf.org
Subject: RE: [Sipping] RE: Need to clarify rules for offer/answer usage in
	 sip
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Hi,

>Drawbacks of PRACK are;
>- If PRACK is rejected due to the offer in PRACK, it would 
>delay the reliable mechanism.

[CHH] This has been discussed before, and my understanding is that a PRACK MUST NOT be rejected due to the offer. You MUST send a 200 OK response to PRACK (unless there is some other reason to reject it, of course), no matter if you accept the offer or not. And, that is the problem of using PRACK for sending offers: there is no way to reject them. And, that is one of the reasons why I am not a big fan of using PRACK for this purpose in the first place, but I guess that is out of the scope of this discussion :)

>- If it wants to send offer in PRACK, it ahs to wait for the 
>reliable response.
> 
>The advantage of PRACK over UPDATE is just saving one message cycle.
> 
>But I am not sure I can say it "must" use UPDATE.

[CHH] Again, my understanding is that you can send an offer/answer in PRACK once, and the reason for that was to save one message cycle in pre-condition (RFC3312) scenarios.

Regards,

Christer Holmberg
Ericsson Finland


> 
> Regards,
> Takuya
> 
> > Regards,
> > 
> > Christer Holmberg
> > Ericsson Finland
> > 
> > 
> > > -----Original Message-----
> > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com]
> > > Sent: 26. tammikuuta 2004 18:43
> > > To: Takuya Sawada
> > > Cc: Christer Holmberg (JO/LMF); sipping@ietf.org
> > > Subject: Need to clarify rules for offer/answer usage in sip
> > > 
> > > 
> > > Takuya,
> > > 
> > > You make good points. I think you have convinced me of 
> how this all 
> > > needs to work. I believe we are also both in agreement that the 
> > > necessary behavior is currently underspecified in the RFCs.
> > > 
> > > I think a useful next step would be to hear from the authors 
> > > of the RFCs 
> > > in question, and hopefully obtain their concurrence. Then we 
> > > can discuss 
> > > the best way to formalize this. I am moving this to the 
> > > sipping list and 
> > > changing the title in order to pick up those authors, 
> some of which 
> > > don't normally watch sip-implementors.
> > > 
> > > 	Paul
> > > 
> > > Takuya Sawada wrote:
> > > > Hi, Paul,
> > > > 
> > > > Let me summarize my view of "restrictions" defined in the 
> > > current SIP RFCs.
> > > > * Probably I think I may share the same view with Christer 
> > > in most parts.
> > > > 
> > > > Offer/Answer exchange described in RFC 3264 is independent from 
> > > > higher layer protocol.
> > > > Higher layer protocol provides means to exchange offer/answer
> > > > and this needs to provide means for resolving "glare" situation
> > > > as described in the last paragraph in RFC 3264.
> > > > 
> > > > [How to provide means to exchange offer/answer in SIP] 
> > > > 
> > > > <RFC 3261>
> > > > RFC 3261 provides the following two ways to exchange 
> offer/answer;
> > > > - INVITE/2xx
> > > > - 2xx/ACK
> > > > See the page 80 in RFC 3261.
> > > > 
> > > > <RFC 3262>
> > > > RFC 3262 provides the additional three ways to exchange 
> > > offer/answer;
> > > > Note that here "1xx" is a reliable provisional response 
> to INVITE.
> > > > - INVITE/1xx ... a first reliable 1xx with SDP. (may not be 
> > > the first 1xx)
> > > > - 1xx/PRACK ... Only when INVITE w/o SDP, 1xx must be the 
> > > first reliable response.
> > > > - PRACK/2xx(PRACK) ... Only after a completion of 
> > > INVITE/1xx or 1xx/PRACK.
> > > > See chapter 5 in RFC 3262.
> > > > 
> > > > <RFC 3311>
> > > > RFC 3311 provides an additional way to exchange;
> > > > - UPDATE/2xx(UPDATE)
> > > > Only when after  any offer/answer exchange defined in RFC 
> > > 3261 and RFC 3262
> > > > has completed and no offer/answer exchange in progress.
> > > > See section 5.1, 5.2 in RFC 3311.
> > > > 
> > > > Currently, as far as I understand, no other ways are not 
> > > defined in any RFCs.
> > > > Assuming this is correct, let me clarify how to resolve 
> > > "glare" cases.
> > > > * I can not imagine how to resolve "glare" if other 
> > > offer/answer exchanges
> > > >   are possible beyond my assumption here.
> > > > 
> > > > [How to resolve "glare" condition]
> > > > One reason why the usage of a SIP method to exchange 
> offer/answer 
> > > > needs to be carefully restricted in RFCs is that UA must 
> > > detect mis-ordered
> > > > message delivery and must avoid confusion.
> > > > In RFC 3264, "it MUST NOT generate a new offer if it has
> > > > received an offer which it has not yet answered or rejected.
> > > >  it MUST NOT generate a new offer if it has generated a
> > > >  prior offer for which it has not yet received an answer or 
> > > a rejection."
> > > > So now there seems to be two possible message crossing case
> > > > to be considered assuming that the above rules are guaranteed.
> > > > 
> > > > Case1
> > > > A                B
> > > > |offer1          |
> > > > |--------------->|
> > > > |         answer1|
> > > > |<-----\ /-------|
> > > > |       X  offer2|
> > > > |<-----/ \-------|
> > > > 
> > > > Case2
> > > > A                B
> > > > |offer1    offer2|
> > > > |------\ /-------|
> > > > |       X        |
> > > > |<-----/ \------>|
> > > > 
> > > > Case1:
> > > > Assuming that offer/answer exchange follows the 
> > > restrictions I described
> > > > above, A can know an SDP in offer2 is not an answer to offer1.
> > > > Then offer2 must be rejected with a 500 with Retry-After, when 
> > > > offer2 is UPDATE. And I assume that it can apply to reINVITE,
> > > > which may not be stated clearly.
> > > > I am not sure that when this happens when offer2 is in PRACK.
> > > > But probably 500 with Retry-After will also apply, even it 
> > > results in
> > > > delay for reliabile transfer mechanism for provisional response.
> > > > In the case that offer2 is in the first reliable 
> > > provisional response
> > > > or ACK, it seems to me that it never happens, if there is a 
> > > rule that
> > > > INVITE, even  w/o SDP, must not be sent if pending offer exists.
> > > > I am not sure whether it is clearly stated.
> > > > 
> > > > Case2:
> > > > Assuming that offer/answer exchange follows the 
> > > restrictions I described
> > > > above, A can know an SDP in offer2 is not an answer to offer1.
> > > > Then offer2 must be rejected with a 491, if it is UPDATE or 
> > > (re)INVITE.
> > > > In the case that offer2 is in PRACK, it may be accepted 
> with 200 or
> > > > may be rejected with 491. 491 may be adequate for 
> > > offer/answer but it may
> > > > delay the completion of a reliable response transfer 
> > > mechanism or, in
> > > > worst case, may result in failure to complete SIP 
> > > transaction because
> > > > there is no clear retry rule when PRACK is rejected with 491.
> > > > When RFC 3262 was defined, there were no UPDATE, therefore 
> > > offer in PRACK
> > > > never collides with other offers from the other side. I 
> > > guess this is why
> > > > there is no rule in RFC 3262 when PRACK offer collides with 
> > > UPDATE offer.
> > > > (It is safe to use UPDATE instead of PRACK when UPDATE 
> is supported)
> > > > Anyway, because there is no PRACK against PRACK collision 
> > > in the current 
> > > > RFCs, the other side is always expected to send 491.
> > > > There is no possibility that offer2 is in the first 
> > > reliable response or ACK.
> > > > Because A must not send new offer when it send INVITE w/o offer.
> > > > 
> > > > Is this a correct summary?
> > > > 
> > > > More comments inline with [TS].
> > > > 
> > > > 
> > > >>Takuya,
> > > >>
> > > >>RFC 3312 introduces situations in which multiple 
> > > offer/answer exchanges 
> > > >>are required before a single INVITE transaction can 
> > > complete. Of course 
> > > >>multiple transactions are involved because PRACK and/or 
> > > UPDATE are used 
> > > >>to achieve part of the exchange, but the INVITE transaction 
> > > can't finish 
> > > >>until they are done.
> > > >>
> > > >>This is at least one case in which offers and answers are 
> > > spread across 
> > > >>transactions in complex ways.
> > > >>
> > > > 
> > > > [TS] In my reading so far, there is the only one case which 
> > > offer/answer is
> > > > across transaction, that is offer in the first reliable 
> > > provisional response 
> > > > of INVITE and answer is in PRACK. There is no other case 
> > > that an offer is in
> > > > a transaction and an answer is in another transaction.
> > > > I agree with that any number of offer/answer exchanges can 
> > > occur during
> > > > one INVITE transaction.
> > > > 
> > > > 
> > > >>There are also cases independent of 3312 where there is a 
> > > need to send a 
> > > >>subsequent offer/answer before an invite transaction 
> completes. For 
> > > >>instance, this is one of the techniques for dealing with 
> > > port conflicts 
> > > >>introduced as a result of forking.
> > > >>
> > > > 
> > > > [TS] Yes. I did not disagree with that an offer in UPDATE 
> > > request can
> > > > be used before the completion of INVITE transaction.
> > > > 
> > > > 
> > > >>Once an invite transaction has completed, including one or more 
> > > >>offer/answer pairs, additional offer/answer cycles can 
> be initiated 
> > > >>using either reINVITE or UPDATE.
> > > >>
> > > >>In the case of reINVITE, I believe there is a requirement 
> > > that one or 
> > > >>more offer/answer cycles be completed before it can 
> > > succeed. And if it 
> > > >>fails, the changed tentatively imparted by all of those 
> > > offer/answer 
> > > >>cycles are to be rolled back.
> > > >>
> > > >>The case of UPDATE is a little less clear. If there is 
> no invite 
> > > >>transaction in progress, and an offer is sent in an 
> UPDATE, and the 
> > > >>update is rejected, it must be the case that the dialog 
> > > state remains as 
> > > >>it was before the update. If there is no invite in 
> progress and an 
> > > >>UPDATE is sent without an offer, there seems to be no 
> prohibition 
> > > >>against sending an offer in the update response. In 
> that case, the 
> > > >>answerer would have no way to refuse the offer and retain 
> > > the dialog as 
> > > >>it is. The only option the answerer would have is to send 
> > > another UPDATE 
> > > >>containing the answer. I doubt this is intended behavior, 
> > > but it isn't 
> > > >>prohibited.
> > > >>
> > > > 
> > > > [TS] This is the point I can not agree. I can not find any 
> > > description that
> > > > UPDATE response can contain an offer in RFC 3311. IMO, this 
> > > means UPDATE response
> > > > can not have any offer. SIP RFCs define restrictions to 
> > > utilize RFC 3264 model.
> > > > UAs can not do any offer/answer exchange which is not 
> > > defined clearly in any RFCs.
> > > > 
> > > > 
> > > >>More below.
> > > >>
> > > >>Paul
> > > >>
> > > >>Takuya Sawada wrote:
> > > >>
> > > >>>Hi,
> > > >>>
> > > >>>In my understanding,
> > > >>>
> > > >>>- Once previous offer/answer completes each side can issue 
> > > new offer.
> > > >>>- INVITE request contains the offer, and the first 
> > > reliable 1xx response
> > > >>>  which contains answer, offer/answer for the INVITE 
> > > transaction completes.
> > > >>>- UPDATE transaction and PRACK transaction can exchange 
> > > offer/answer.
> > > >>>- Therefore, once the reliable 1xx response which contains 
> > > answer is received,
> > > >>>  caller can start offer/answer with PRACK or UPDATE or 
> > > any new method.
> > > >>>  And once the reliable 1xx response which contains 
> answer is sent,
> > > >>>  callee can start offer/answer with UPDATE.
> > > >>>
> > > >>>
> > > >>>
> > > >>>>P.S. For those with the patience, below I have tried to 
> > > enumerate all 
> > > >>>>the offer/answer cases thru the 2nd offer. It is 
> pretty painful.
> > > >>>>
> > > >>>
> > > >>><snip>
> > > >>>
> > > >>>>2) invite with 100rel
> > > >>>>   a) offer in invite
> > > >>>>      Possible places for answer:
> > > >>>>      i)   answer in first reliable 1xx
> > > >>>>      ii)  answer in subsequent reliable 1xx
> > > >>>>      iii) answer in 2xx
> > > >>>>   b) offer not in invite
> > > >>>>      offer must be in first reliable 1xx
> > > >>>>      answer must be in PRACK to 1xx w/offer
> > > >>>>
> > > >>>>Multiple offer/answer cycle prior to completion of invite.
> > > >>>>Can only happen in subcases of (2) above.
> > > >>>>It requires use of UPDATE in some cases.
> > > >>>>
> > > >>>>3) original offerer sends 2nd offer
> > > >>>>   a) following 2.a.i or 2.a.i.i:
> > > >>>>      Possible places for 2nd offer:
> > > >>>>      i)   offer2 in PRACK of answer1
> > > >>>>      ii)  offer2 in PRACK of some subsequent 1xx
> > > >>>>      iii) offer2 in UPDATE
> > > >>>>   b) following 2.a.iii:
> > > >>>>      No 2nd offer possible until after completion of invite.
> > > >>>>   c) following 2.b:
> > > >>>>      i)   offer2 in 2xx to PRACK w/answer1
> > > >>>>      ii)  offer2 in UPDATE
> > > >>>>
> > > >>>>4) original answerer sends 2nd offer:
> > > >>>>   Not clear if this is intended to be legal at all prior to
> > > >>>>   completion of invite. It appears existing text permits
> > > >>>>   as follows:
> > > >>>>   a) following 2.a.i or 2.a.ii:
> > > >>>>      Possible places for 2nd offer:
> > > >>>>      i)   offer2 in 2xx to prack of answer1.
> > > >>>>      ii)  offer2 in subsequent reliable 1xx
> > > >>>>           following the one carrying answer1.
> > > >>>>      iii) offer2 in UPDATE following answer1.
> > > >>>>           (No reason to prefer this to other possibilities.)
> > > >>>>   b) following 2.a.iii:
> > > >>>>      No 2nd offer possible until after completion of invite.
> > > >>>>   c) following 2b:
> > > >>>>      i)   if another reliable 1xx is received,
> > > >>>>           could send offer2 in the PRACK.
> > > >>>>           (but can't count on this happening.)
> > > >>>>      ii)  offer2 in UPDATE after sending answer1.
> > > >>>
> > > >>>
> > > >>>I do not think offer/answer exchange beyond the SIP transaction
> > > >>>is possible. I am not sure it is prohibited somewhere 
> in RFC, but
> > > >>>even if it is not prohibited explicitly, one should not assumed
> > > >>>that is possible.
> > > >>
> > > >>Following the principle of being conservative in what 
> you send and 
> > > >>liberal in what you receive, I want to assume that I 
> might receive 
> > > >>anything that isn't prohibited, and do something 
> reasonable with it.
> > > >>
> > > > 
> > > > [TS] I love this principle. IMO, if I receive an SDP which 
> > > I do not expect
> > > > to receive in RFC, one possible solution is that I will 
> > > accept SIP message
> > > > and ignore the SDP. Content-Disposition header's 
> handling parameter
> > > > should be checked, though.
> > > > Anyway, here I do not know what is something reasonable to 
> > > do with it.
> > > > 
> > > > 
> > > >>I would be willing to be conservative in initiating 
> this stuff if I 
> > > >>could figure out what that is. Given the current state of 
> > > >>underspecification I don't know that I will draw the same 
> > > conclusion 
> > > >>than anyone else will.
> > > >>
> > > > 
> > > > [TS] There seems to be some uderspecification in RFCs. But 
> > > I want to 
> > > > confirm the following;
> > > >   RFCs only state where and when an offer and an answer 
> can appear.
> > > >   If RFCs do not state "it can", then it MUST NOT contain 
> > > offer/answer.
> > > >   Because RFCs define "restrictions" where and when offer/answer
> > > >  exchange can occur.
> > > > First I think we should reach a consensus with this point.
> > > > 
> > > > 
> > > >> > One can achieve the same thing without such a
> > > >>
> > > >>>tweak. 
> > > >>
> > > >>Not sure what you are referring to here, or whether it is 
> > > negated in 
> > > >>your subsequent posting. See above for a number of things 
> > > that must be 
> > > >>possible.
> > > >>
> > > >>
> > > >>>Then, 3.c.i and 4.a.1 SHOULD NOT, or hopefully MUST 
> NOT, be used.
> > > >>
> > > >>If we can codify a clear, and complete, set of rules for 
> > > where offers 
> > > >>and answers can appear, that provides full 
> functionality, and it 
> > > >>excludes these cases, then fine.
> > > >>
> > > >>In the case of 4.a.1, removing it causes no loss of 
> > > functionality, so I 
> > > >>am inclined to agree. It might ultimately result in 
> exchanging more 
> > > >>messages, but the number is probably insignificant.
> > > >>
> > > >>In the case of 3.c.1, removing it requires that both 
> sides support 
> > > >>UPDATE. So its removal may remove functionality in the case 
> > > where one 
> > > >>side or the other doesn't support UPDATE. This may or 
> may not be 
> > > >>significant.
> > > >>
> > > > 
> > > > [TS] As you said, UPDATE can solve the problem.
> > > > You can use extensions when it wants the functionality that 
> > > the base 
> > > > specifications can not provide. This is what the extensions are.
> > > > 
> > > > 
> > > >>>Anyway for INVITE transaction, as explained in my 
> previous mail,
> > > >>>it is clear in RFC 3261 to me that one INVITE transaction can 
> > > >>>include only one offer/answer exchange.
> > > >>>
> > > >>>Then 4.a.ii is not permitted.
> > > >>
> > > >>I think I identified cases above where this is needed.
> > > >>
> > > > 
> > > > [TS] The reason I think it is not permitted is, as Christer 
> > > pointed our,
> > > > that RFC 3261 does not permit and any other RFCs do not 
> override it.
> > > > 
> > > > 
> > > >>Takuya Sawada wrote:
> > > >> > Hi,
> > > >> >
> > > >> >
> > > >> >>I do not think offer/answer exchange beyond the SIP 
> transaction
> > > >> >>is possible. I am not sure it is prohibited 
> somewhere in RFC, but
> > > >> >>even if it is not prohibited explicitly, one should 
> not assumed
> > > >> >>that is possible. One can achieve the same thing 
> without such a
> > > >> >>tweak.
> > > >> >
> > > >> >
> > > >> > I am not exactly correct here.
> > > >> > When an offer is in INVITE reliable provisonal response and
> > > >> > an answer is in PRACK offer/answer exchange is beyond one
> > > >> > transaction.
> > > >> > What I want to say is that how offer/answer is used 
> in a certain
> > > >> > method is explicitly scoped in each RFC which 
> defines the method.
> > > >>
> > > >>The trouble with these is that they were written in a 
> > > certain order. As 
> > > >>each was written, a broadening of thinking seems to have 
> > > occurred. This 
> > > >>is reflected in each, but not necessarily fully reflected 
> > > in the earlier 
> > > >>specs.
> > > >>
> > > > 
> > > > [TS] I may agree with this. At least this makes it 
> > > difficult to understand
> > > > and share the same view among all. Actually, there seems to 
> > > be a little 
> > > > ambiguity with PRACK and UPDATE collision case as I pointed 
> > > > out in this mail, etc.
> > > > 
> > > > 
> > > >> > How PRACK is used for offer/answer is defined in RFC 3262.
> > > >> > There seems to be only two cases:
> > > >> >  - offer/answer is reliable 1xx(INVITE)/PRACK
> > > >> >  - offer/answer is PRACK/2xx(PRACK)
> > > >> >    only after INVITE's offer/answer exchange complete.
> > > >>
> > > >>I agree that this seems pretty well locked down as it 
> > > relates to 3261. 
> > > >>These lock downs of course do apply to the things that came 
> > > later. But 
> > > >>they seem at least awkward and arbitrary in some cases.
> > > >>
> > > >> > How UPDATE is used for offer/answer is defined in RFC 3311.
> > > >> >  - offer/answer is UPDATE/2xx(UPDATE)
> > > >>
> > > >>Here I disagree. I looked pretty closely at this, and I 
> > > don't see *any* 
> > > >>normative language for how offer/answer is used with 
> it. There is a 
> > > >>bunch of non-normative language which appears to simply restate 
> > > >>implications of 3261 and 3264 in the presence of update.
> > > >>
> > > > 
> > > > [TS] Again, this is the point I can not agree. I agree with 
> > > that there is
> > > > no normative language to prohibit offer in the response 
> of UPDATE
> > > > like "2xx response to UPDATE MUST NOT contain offer".
> > > > Actually, response can have an SDP even if UPDATE does not have
> > > > an offer SDP. But an SDP can be neither an offer nor an answer 
> > > > unless clear semantics are defined in any RFCs. I can not find 
> > > > any usage or semantics to use offer/answer exchange beyond
> > > > one UPDATE transaction.
> > > > IMO, SDP in a 2xx of UPDATE when an UPDATE request w/o
> > > > offer is neither an offer nor an answer like SDP in INFO is 
> > > > neither an offer nor an answer.
> > > > 
> > > > 
> > > >> > How INVITE is used for offer/answer is defined in RFC 3261.
> > > >> >  - offer/answer is INVITE/non-failure reliable response
> > > >> >  - offer/answer is the first non-failure first 
> > > >>response/acknowlegement for the response
> > > >> >    , which is 2xx(INVITE)/ACK in RFC 3261 and 
> > > 1xx(INVITE)/PRACK in 
> > > >>RFC 3262.
> > > >> >  * Note that the second or later SDP in INVITE 
> > > non-failure response
> > > >> >    is not an offer. It is nothing.
> > > >> >
> > > >> > There is no other way than explicitly defined in any 
> RFC, IMO.
> > > >>
> > > >>I believe you are right that nothing more is explicitly defined.
> > > >>But 3264 is pretty general and broad. It is not 
> prescriptive in how 
> > > >>offers and answers are transmitted. 
> > > > 
> > > > 
> > > > [TS] Yes. Therefore, RFC 3264 requires restrictions on how 
> > > > to exchange offer/answer in higher layer protocol.
> > > > 
> > > > 
> > > >>And while the other RFCs may not 
> > > >>explicitly specify other ways, there are other ways that aren't 
> > > >>explicitly prohibited. In general with sip, things that aren't 
> > > >>explicitly prohibited are permitted.
> > > >>
> > > > 
> > > > [TS] Yes. Any SIP messages can contain SDP at any moment. 
> > > > But without semantics, it is neither an offer nor an answer.
> > > > 
> > > > 
> > > >>I just want to get this nailed down, so I know how to build 
> > > a stack that 
> > > >>can both handle every valid case thrown at it, and that can 
> > > initiate all 
> > > >>the needed functionality.
> > > >>
> > > >>I am thinking that maybe somebody (probably me) needs to 
> > > write a best 
> > > >>practices document that pulls this all together, specifying 
> > > all the ways 
> > > >>that offer/answer can be used. Following that, I think 
> it will be 
> > > >>advisable to tighten up one or more of the RFCs in places 
> > > where things 
> > > >>haven't been prohibited but should be.
> > > >>
> > > > 
> > > > [TS] The guideline document that you are proposing can 
> be useful.
> > > > Collecting all the possible collision cases, 
> recommending UPDATE 
> > > > offer rather than PRACK offer if UPDATE is supported 
> and no need 
> > > > to reduce the number of messages, enumerating Warning 
> header code 
> > > > examples to be included when offer is rejected with 488, etc.
> > > > 
> > > > Regards,
> > > > Takuya
> > > > 
> > > > 
> > > >>Paul
> > > >>
> > > >>_______________________________________________
> > > >>Sip-implementors mailing list
> > > >>Sip-implementors@cs.columbia.edu
> > > >>http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
> > > > 
> > > > 
> > > > --------
> > > > Takuya Sawada
> > > > KDDI Corporation (KDDI)
> > > > Garden Air Tower, 3-10-10  Iidabashi Chiyoda-ku
> > > > Tokyo 102-8460, Japan
> > > > Tel: +81-3-6678-2997
> > > > Fax: +81-3-6678-0285
> > > > tu-sawada@kddi.com
> > > > 
> > > 
> > 
> > _______________________________________________
> > Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> > This list is for NEW development of the application of SIP
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sip@ietf.org for new developments of core SIP
> 
> 
> --------
> Takuya Sawada
> KDDI Corporation (KDDI)
> Garden Air Tower, 3-10-10  Iidabashi Chiyoda-ku
> Tokyo 102-8460, Japan
> Tel: +81-3-6678-2997
> Fax: +81-3-6678-0285
> tu-sawada@kddi.com
> 

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