[asterisk-users] Call quality - how to debug

[Jared Smith]

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.