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

[Yatin Patil]

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 <address@hidden> 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: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

--
-------------------------------------------------------
Alastair Johnson
SolutionTrax Technologies - http://www.solutiontrax.com
T: 01908 268902   F: 0709 2117048

_______________________________________________
Linphone-users mailing list
address@hidden
https://lists.nongnu.org/mailman/listinfo/linphone-users