On Jan 8, 5:29 am, "vop...@gmail.com" <vop...@gmail.com> wrote:
> for jitter and traffic stats im using Ethereal. There is an option in
> stats -> RTP -> analize stream
> this jitter value is calculated according to the RFC 3550 for RTP

Thanks. I didn't know it would do that. I just tried it with both a
GizmoProject and a Skype call using WireShark 0.99.4. I couldn't
capture any RTP packets. I'm doing something wrong. These might be of
interest:
<http://wiki.wireshark.org/VoIP_calls>
<http://wiki.wireshark.org/RTP_statistics>
<http://wiki.wireshark.org/RTP>
I'll play with it some more when the phone stops ringing. (I hate
mondays).