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[Linphone-users] help with 'No codec intersection' error

From: Yatin Patil
Subject: [Linphone-users] help with 'No codec intersection' error
Date: Thu, 5 Jul 2012 17:46:23 -0700
Hello,

I am trying to make Linphone work with a VLC server through the gateway which translates messages between SIP and RTSP. Eventually I want to listen/view video streamed by a VLC server on Linphone client. Linphone, gateway and VLC are running on the same machine.
But, I am getting error regarding 'No codec intersection'. Looking at the Linphone debug log, it details error as 'error: Incompatible SDP offer received in 200Ok, need to abort the call'.

This is an initial INVITE message from Linphone:
--------------------------------------------------------------------------------------------------------------
INVITE sip:address@hidden:33000 SIP/2.0
Via: SIP/2.0/UDP 192.168.111.215:5060;rport;branch=z9hG4bK247713395
From: <sip:address@hidden>;tag=1981465341
To: <sip:address@hidden:33000>
Call-ID: 230194435
CSeq: 20 INVITE
Contact: <sip:address@hidden>
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Max-Forwards: 70
User-Agent: Linphone/3.4.3 (eXosip2/3.3.0)
Subject: Phone call
Content-Length:   320

v=0
o=yatin 446 446 IN IP4 192.168.111.215
s=Talk
c=IN IP4 192.168.111.215
t=0 0
m=audio 7078 RTP/AVP 112 111 110 0 0 3 0 8 101
a=rtpmap:112 speex/32000
a=fmtp:112 vbr=on
a=rtpmap:111 speex/16000
a=fmtp:111 vbr=on
a=rtpmap:110 speex/8000
a=fmtp:110 vbr=on
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
--------------------------------------------------------------------------------------------------------------


This is a 200 OK response from the gateway
--------------------------------------------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.111.215:5060;rport;branch=z9hG4bK247713395
From: <sip:address@hidden>;tag=1981465341
To: <sip:address@hidden:33000>;tag=random
Call-ID: 230194435
Contact: <sip:address@hidden:33000>
CSeq: 20 INVITE
Max-Forward: 70
Content-Type: application/sdp
Content-Length: 111

v=0
o=yatin 446 446 IN IP4 192.168.111.215
s=Talk
c=IN IP4 192.168.111.215
t=0 0
m=audio 50140 RTP/AVP 112
--------------------------------------------------------------------------------------------------------------

Can anybody please help me with debugging of this issue.

Thanks !!
Yatin

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