[Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec.

[Anthony Minessale]

you should have also turned in the sip trace
sofia profile internal siptrace on


On Thu, Jun 4, 2009 at 7:43 AM, Jim Burke <jim at evolutiontel.net> wrote:

> Hi All,
>
> Looking for some debugging tips and comments on what might be causing
> the media port in the 200OK ( Answer message) to be set to 0 by
> freeswitch.  Essentially it looks like data might be getting trampled
> somehow.
>
> Portion of 200OK going into Freeswitch
> m=audio 16416 RTP/AVP 18 100 101..a=rtpmap:18 G729a/8000..a=rtpmap:100
> NSE/8000..a=
>  fmtp:100 192-193..a=rtpmap:101 telephone-event/8000..a=fmtp:101
> 0-15..a=ptime:20..a=sendrecv..
>
> Portion of 200OK coming out of Freeswitch
> m=audio 0 RTP/AVP 96 100 101..a=rtpmap:96 G729a/8000..a=rtpmap:100
> NSE/8000..a=fmtp:100 192-193..a=rtpmap:101 telephone-event
>  /8000..a=fmtp:101 0-15..
>
> Note the media port has been set to 0 and the rtpmap for G729 is also
> not correct.  On receipt of this bad 200Ok the originator sends a BYE.
>
> We are using FS as a B2BUA with bypass_media set to true.  Thus IMHO
> Freeswitch should not be touching the SDP portion of the message and
> just passing it through.
>
> This can reproduce this at will, so I can collect whatever data is
> nessicary.  I have added the sofia debug from the console.
>
> Thanks,
> --
> Jim Burke
> Director Evolutiontel.
> http://www.evolutiontel.net
>
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>


-- 
Anthony Minessale II

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