The captured RTP stream's analysis shows the following:
Do you think this is wrong? How can I verify the jitter by looking at the individual packet latencies? asked 27 Nov '10, 23:24 skypemesm |
3 Answers:
This looks wrong to me if you have valid RTP (delta would be non-zero). Care to put up a sample? answered 02 Dec '10, 23:18 martyvis |
Check if you have "Incorrect timestamp" messages in status field. For dynamic payload types Wireshark is unable to calculate jitter etc., because it does not know the sampling frequency of codec used. In order for Wireshark to work with dynamic payload types someone must implement either manual setting of sampling frequency or parsing SDP files. answered 03 Dec '10, 02:47 kamokr |
Have you just filtered down to the one particular RTP conversation in one direction? If so, you can add a frame delta time column and increase the precision. I would use this technique to validate you 0.00% jitter. It could be that the jitter is present, but below 0.005% and is rounded down. Just a thought. answered 06 Dec '10, 10:41 Paul Stewart |
Reading the sampling rate is implemented in trunk, i don't remember if it made it into 1.4.
Tried trunk, rev. 35106, does not work. Are you sure it is fully implemented?