[SIPForum-discussion] How does the SDP offer and answer works

Herve Jourdain herve.jourdain at mstarsemi.com
Tue Jan 22 09:55:19 UTC 2008


Hi,

 

As per RFC 3264 :

 

< The interpretation of fmtp parameters in an offer depends on the

   parameters.  In many cases, those parameters describe specific

   configurations of the media format, and should therefore be processed

   as the media format value itself would be.  This means that the same

   fmtp parameters with the same values MUST be present in the answer if

   the media format they describe is present in the answer.  Other fmtp

   parameters are more like parameters, for which it is perfectly

   acceptable for each agent to use different values.  In that case, the

   answer MAY contain fmtp parameters, and those MAY have the same

   values as those in the offer, or they MAY be different. >

 

In the case of G.729, the annexb parameter is OPTIONAL (as per RFC 3555), so
it doesn't need to be present in the answer.

But if not present in the answer, "yes" is implied (also as per RFC 3555).

 

So I think that's why you fallback to G.711, one is stating "no annexb", the
other is stating "annexb". and the one with "no annexb" decides it's not
compatible.

 

Regards,

 

Herve

 

  _____  

From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Andrea Puddu
Sent: lundi 21 janvier 2008 21:22
To: Mark Holloway
Cc: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] How does the SDP offer and answer works

 

 

Sorry,

I was wrong .... the SDP of the INVITE to B phone looks like:

......
m=audio 19386 RTP/AVP 18 8 0
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000

and the SDP of the 200 OK reply by B phone is:

..... 

m=audio 41000 RTP/AVP 18
a=rtpmap:18 G729/8000
a=sendrecv

As you can see the fmtp line is missing, even if B phone supports both G729
a and b. It seems that CPE "suffers" the lack of the fmtp line in the 200
OK. If the phone does not specify if it supports the annex in the 200 OK,
the CPE understand that the phone does not support that codec! So it
switches to the second codec in the list (g711).
But is it mandatory for the phone to specify the fmtp line in the 200 OK
reply, if the fmtp line is specified in the related INVITE?

Thanks.






  _____  


From: mh at markholloway.com
To: androjoker at hotmail.com
CC: discussion at sipforum.org
Subject: RE: [SIPForum-discussion] How does the SDP offer and answer works
Date: Fri, 18 Jan 2008 03:08:36 -0700

What is the make/model of SIP Endpoint B?    A proper format for G729b would
look like this:

 

Media Description, name and address (m): audio <PORT> RTP/AVP 18

Media Attribute (a): rtpmap:18 G729/8000

Media Attribute (a): fmtp:18 annexb=yes

 

I've never seen a SIP Endpoint list "g729 annex b" in the SDP.   It looks
like B is sending an invalid SDP to A for G729 which is why it falls back to
G711.

 

From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Andrea Puddu
Sent: Friday, January 18, 2008 1:47 AM
Cc: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] How does the SDP offer and answer works

 

 

 

Hi all,

I'm facing a similar issue. The scenario is:
A phone (A) calls another phone (B).
In the INVITE request, A specifies "g729 annex b" as the preferred codec.
CPE forwards the request to B phone with "g729 annex b" as the preferred
codec.
Phone B replies with a 200 OK specifing only "g729" as the preferred codec.
What does this mean as the phone supports both g729 a and b?
I think that phone B means that it accepts "g729 annex b", but it only
specifies "g729" in its 200 OK answer. I'm afraid that media gateway
expected a 200 OK with "g729 annex b" specified, if supported by phone.
Otherwise media gateway does not understand the 200 OK answer, maybe because
"g729" without annex specified is intended to be "g729a". So the media
gateway returns the 200 OK answer to phone A with the second codec in the
list (g711).
I don't know if it's mandatory that phone B specifies "g729 annex b" (if
supported) instead of "g729".

A                             CPE                                      B

 --------------------->        ---------------------------->
  INVITE (g729b)                  INVITE (g729b)


<---------------------        <----------------------------
   200 OK (g711)                    200 OK (g729)
 




Andrea

On 1/17/08, Mark Holloway <mh at markholloway.com> wrote: 

If A is a media gateway and B is a SIP subscriber, usually the gateway will
honor the preferred codec of the subscriber.    Thus meaning A is just
informing B of what he is capable of, but if B responds with rtpmap: 8
(G729) first and 0 (G711) second in the 200 OK, the call will be setup as
G729a (unless B also states annexb=yes, then it will be G729b).  
 

From: discussion-bounces at sipforum.org
<mailto:discussion-bounces at sipforum.org>
[mailto:discussion-bounces at sipforum.org] On Behalf Of arun kumar
Sent: Thursday, January 17, 2008 2:32 AM 
To: discussion at sipforum.org
Subject: [SIPForum-discussion] How does the SDP offer and answer works 

 

Hi can any one can guide me in this

 

 

       A---------INVITE/SDP----------> B

 

      The SDP offer  has the following 

                m= audio 49170 RTP/ AVP 0 8

              a= rtpmap: 0 PCMU/8000

             a= rtpmap: 8 PCMA/8000

                   m= video 49172 RTP/AVP 32

                   a= rtpmap: 32 MPV/90000

 

    A<-----------200OK-------B

 

                B also supports all the media and codec types provided in
the SDP offer of A

 

So what will be the SDP answer of B, will it send all the medai and codec
supported by itself ?

 

my question is who will select the media & codec for the Media session
establishment between  A & B, if more than  1 suitable codec matches between
the endpoints which codec is selected and on what basis is it done. 

 

 
 
 
v = 0
 
                      
 
 
                      

















-- 













With Regards



























Arun Kumar .P 




-- 
With Regards
Arun Kumar .P 

 


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