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Re: [Sip] Re: [Sip-implementors] Early media query



The Replaces approach is a possibility. But please explain why you think all the possibilities are invalid.

	Paul

Xia, Zhi Feng (Bruce) wrote:
Given the restrictions in RFC, all four methods seem invalid.

Before we can change the restrictions in the RFC,  it looks to me using "replaces" is a better approach...

How about this flow:

       Caller                                     Callee
                  |                                            |
                  |                                            |
                  |(1) INVITE with offer        |
                  |---------------------------->    |
                  |                                           |
                  |                                           |
                  |(2) 183 with answer 1    |
                  |<-------------------------------|
                  |                                           |
                  |                                           |
                  |(3) PRACK                       |
                  |---------------------------->   |
                  |                                           |
                  |                                           |
                  |(4) 200 PRACK               |
                  |<------------------------------|
                  |(5) 200 OK                      |
                  |<----------------------------  |
                  |                                         |
                  |(6) ACK                           |
                  |---------------------------->|
                  |                                        |
                  |(7)Re-INVITE(replaces with new call id)
                  |<----------------------------|
                  |   NOTIFY                       |
                     |----------------------------->
                     |BYE(the original call id)
                     |----------------------------->
                  |                                        |
                  |(8) 200 OK with SDP   |
                  |---------------------------->|
                  |                                        |
                  |(9) ACK                          |
                  |---------------------------->|

-----Original Message----- From: Paul Kyzivat [mailto:pkyzivat at cisco.com] Sent: 2006?10?5? 22:52 To: Darshan Bildikar Cc: sip at ietf.org; sip-implementors at cs.columbia.edu; 'Retesh Chadha' Subject: [Sip] Re: [Sip-implementors] Early media query




Darshan Bildikar wrote:
I don't think that even Method 3 and 4 are OK because they will involve a
change in the origin line parameter of the SDP and that is not allowed as
per the RFC.

Which RFC?

I suport Crister's response.

	Paul

BTW, what was the rationale behind not allowing the origin line to change?
It makes implementing apps like prepaid and CRBT very difficult. Could
someone please throw some light on this?

-----Original Message-----
From: sip-implementors-bounces at cs.columbia.edu
[mailto:sip-implementors-bounces at cs.columbia.edu] On Behalf Of Retesh Chadha
Sent: Monday, October 02, 2006 5:46 PM
To: Udit_Goyal at 3com.com
Cc: sip at ietf.org; sip-implementors at cs.columbia.edu
Subject: Re: [Sip-implementors] Early media query

Hi Udit
Of all the methods you have specified, Method 1 and 2 are wrong as per
the RFC which states that the sdp of all the provisional and 2xx
response of INVITE should be same if they are present.

Both Method 3 and 4 look fine, with method 4 being better because in
that case 200 ok for INVITE is only sent when the final negotiation
(call media) is done.

Hope this helps.

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