Can you provide the entire sip dialog?


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On Sep 18, 2013, at 7:11 PM, Sam Russell <sam.h.russell@gmail.com> wrote:

> Hi,
>
> I've been using Freeswitch as a PBX, and I'm having problems when making video calls when the video stream is rejected. The other endpoint (happens on 4 distinct endpoints) sends back a m= line with port 0, and an a= line with codec type as /0. When this happens, I get the error "sofia.c:1431 nua_i_media_error: unknown event 22: 988 Incomplete offer/answer" and the call is rejected.
>
> I think Freeswitch is getting upset with the a= line after the "m=video 0" line because there's no codec name. The RFC says that any a= lines after an m= line with port=0 must be ignored, is this a bug?
>
> t=0 0
> m=audio 31720 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> m=video 0 RTP/AVP 96
> a=rtpmap:96 /0

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