[asterisk-users] Sound quality issue

[Andreas Sikkema]

> I am sure there are RTP packets losses somewhere, except RTP debug in
> the asterisk CLI, how can i determine where the problem come from ?

If it is possible to make a network trace in a Wireshark compatible
format, Wireshark can parse all the SIP and RTP messaging and give you
lots of statistics, including packet loss, jitter, etc. Check the
Wireshark site (http://www.wireshark.org/) for more information.

-- 
Andreas Sikkema