Mitch Claborn wrote: > > Thank you for that most excellent post. I had guessed at most of the > SDP fields and meaning.
No problem. I actually like looking at SIP traces for some reason. > I have wireshark traces from the client and the RTP packets are not in > the trace, which I think means that the client software is simply not > producing them. I have opened a ticket with SFL phone support and will > post here if I find anything. That's a reasonable conclusion. Just make sure that you get some traces of good calls to verify that your tests are valid. > I did test the "muted microphone" theory. SFLphone continues to send > RTP packets even when the mic is muted, so that doesn't seem to be the > cause. It's always a good idea to rule out PEBKAC before spending a lot of time diagnosing a problem. > I've also compared the call initiation SIP and SDP packets between a > call that fails and one that works correctly. I can discern no > difference other than things like port numbers and call IDs. > > Tomorrow I'll be trying one of my agents on Bria instead of SFL - maybe > that will make a difference. It really seems like it may be a problem with the softphone. I'm sure the developers of SFLphone will appreciate your feedback, because not sending RTP is a pretty serious bug. I'll keep an eye on this thread and help out if I can. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users