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

Tue Jan 22 11:18:58 UTC 2008

Thanks everybody,

I agree with your analysis.

Andrea

From: herve.jourdain at mstarsemi.com
To: androjoker at hotmail.com; mh at markholloway.com
CC: discussion at sipforum.org
Subject: RE: [SIPForum-discussion] How does the SDP offer and answer works
Date: Tue, 22 Jan 2008 10:55:19 +0100

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]
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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