Very useful tool is Wireshark! (http://www.wireshark.org/).
During capture (or playback of a capture file) navigate to “Telephony -> VoIP Calls” and then select a VoIP session:
You can then click on “Graph” to get a SIP and RTP call flow chart:
Even nicer you can select “Player -> Decode -> Play” and hear the audio stream:
To analyse jitter select “Tools -> RTP -> All Streams“:
You can then select a stream and click on “Analyse“:
In the example above you can see a large delta of nearly 6 seconds during which a GVP / IPCS page fetch occurs. More on that in a later post ….