[SIPForum-discussion] Is "fmtp" line needed in 200 OK answer?

[Fernando Lombardo]

Andrea,
 
Most likely your problem is with the a=fmtp line of the SDP. Many
implementations, in the absence of the fmtp line, assume that the
offer/answer is G729b (annexb=yes) because RFC 3555 says that this
should be the default behavior. So, in this case your answer will be
construed as G729b and so the device could potentially do two things:
pick the next codec it supports (g711 in this case) or terminate the
call.  Hope this helps,
 
-fernando

________________________________

From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of Andrea Puddu
Sent: Tuesday, December 11, 2007 9:28 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] Is "fmtp" line needed in 200 OK answer?


Hello guys,

I'm facing an issue when a mobile (common mobile 3G phone) tries to call
an internal SIP phone through its geographical number.
The issue is that when the SBC sends the INVITE to the IP phone,  in the
SDP  enclosure it  suggests these codecs:

m=audio 19570 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

The 200 OK of the phone (SDP part only) is:

m=audio 57956 RTP/AVP 18 8 0
a=rtpmap:18 g729/8000
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=sendrecv

So the IP phone start to talk with G729 and SBC replies with G711A!!! So
I can't hear the voice coming from mobile phone!!!!

I have two questions:

- Can the difference between the offer (G729) and the answer (g729) be
significant? I mean the difference because the "g" is not capital in the
answer
- Can the lack of the "fmtp" line in the answer cause troubles?


Thanks 1000,

Andrea





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