that is because skype use propietary coding scheemes, which means
ethereal cant see whats the payload, If you make the test with a sip
based client you can.
I have my own server with asterisk, which uses UDP/RTP

Cheers

flo
On Jan 8, 6:12 pm, "Jeff Liebermann" <j...@comix.santa-cruz.ca.us>
wrote:
> 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 RTPThanks. 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).