On Tue, 2009-06-02 at 15:49 +0100, Adrian Marsh wrote: > However - my question would still stand, how exactly would I be able to > debug whats going on in the RTP stream? And why its stuttering > (sometimes halfway through a call). > > Any tips or tricks for actually debugging within Asterisk ?
Wireshark has a lot of RTP tools for looking at the latency and jitter and dropped packets on the line, which are the most common problems I find when helping people diagnose poor audio connections. It won't tell you what is *causing* the problem, but it will help you know what the problem actually is. >From there, you can start to track down the source of the problem one network segment at a time. For example... is the poor audio being caused by network problems between the phone and Asterisk, or between Asterisk and your upstream provider. -- Jared Smith Training Manager Digium, Inc. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users