The response is missing the a=rtpmap line that would tell linphone what payload the gateway means when it says 112.
On Thursday 05 July 2012 17:46:23 Yatin Patil wrote: > 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:[email protected]:33000 > SIP/2.0 > Via: SIP/2.0/UDP 192.168.111.215:5060;rport;branch=z9hG4bK247713395 > From: <sip:[email protected]>;tag=1981465341 > To: <sip:[email protected]:33000> > Call-ID: 230194435 > CSeq: 20 INVITE > Contact: <sip:[email protected]> > 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:[email protected]>;tag=1981465341 > To: <sip:[email protected]:33000>;tag=random > Call-ID: 230194435 > Contact: <sip:[email protected]: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 -- ------------------------------------------------------- Alastair Johnson SolutionTrax Technologies - http://www.solutiontrax.com T: 01908 268902 F: 0709 2117048 _______________________________________________ Linphone-users mailing list [email protected] https://lists.nongnu.org/mailman/listinfo/linphone-users
