Wireshark-users: [Wireshark-users] Analyzing RTP Streams
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.
