Analyzing RTP Streams

[Dustin Schuemann <dschuemann () gmail com>]

I am trying to diagnose a VOIP issue. When I play the call it sounds fine.
The call doesn't have any dropped packets. It does say payload changed to
PT=0. The mean jitter was 1.29 ms, Max jitter was 12.64 ms. and the max
delta was 22.93 ms. The internal caller hears the end users voice drop every
other word. The outside caller doesn't have any issues. I believe this is a
network issue but Im not able to pin point it. I guess I don't quite know
how to read the RTP analysis screen.

Any help you can provide would be grateful.

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