Thanks Alastair for your reply. I have already tried modifying 200OK response from gateway in following ways..but I still get the same error.
m=audio 50140 RTP/AVP 112 a=rtpmap:112 speex/32000 a=fmtp:112 vbr=on m=audio 50140 RTP/AVP 111 a=rtpmap:111 speex/16000 a=fmtp:111 vbr=on m=audio 50140 RTP/AVP 110 a=rtpmap:110 speex/8000 a=fmtp:110 vbr=on m=audio 50140 RTP/AVP 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 Do you see any problem in these responses or any other cause? Thanks !! Yatin On Fri, Jul 6, 2012 at 3:15 AM, Alastair Johnson <[email protected]>wrote: > 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 >
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