<div dir="ltr">It may be mad at the /0 which I assume is <name>/<speed> with blank name and 0 speed<div>It may be that it's supposed to ignore it but since it doesn't like the format it still gets mad.</div>
<div>Even better would be if they didn't send a pointless a= with questionable content in the first place.</div><div><br><div><br></div></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Sep 19, 2013 at 7:45 AM, Brian West <span dir="ltr"><<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Can you provide the entire sip dialog?<br>
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--<br>
Brian West<br>
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<div><div class="h5"><br>
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On Sep 18, 2013, at 7:11 PM, Sam Russell <<a href="mailto:sam.h.russell@gmail.com">sam.h.russell@gmail.com</a>> wrote:<br>
<br>
> Hi,<br>
><br>
> 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.<br>
><br>
> 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?<br>
><br>
> t=0 0<br>
> m=audio 31720 RTP/AVP 8 101<br>
> a=rtpmap:8 PCMA/8000<br>
> a=rtpmap:101 telephone-event/8000<br>
> a=fmtp:101 0-15<br>
> m=video 0 RTP/AVP 96<br>
> a=rtpmap:96 /0<br>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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