Hi, I captured the traffic during a skype conversation in the two end points. Then I have decoded the UDP's packets with an RTP protocol, in this way i'm able to see the timestamps of each packet. But, when I go to the option Telephony-->RTP-->show all the streams.., in the fields : loss packets and jitter appears always a 0, and I don't know why. Somebody can help me?
My objective is to do an study of the packets loss and jitter between a skype conversation.
asked 24 Oct '12, 04:47
Skype doesn't use RTP. It uses its own proprietary protocol so trying to decode its traffic as RTP isn't likely going to work out very well.
For some more info about the protocol (and the beginnings of a Skype dissector in Wireshark) see Wiki's Skype page.
answered 24 Oct '12, 07:21