> A colleague of mine said that Ethereal can play back an RTP
> stream, a VoIP call to be specific, and that it will
> duplicate the network characteristics in the audio file. By
> that I mean high jitter rates and delay issues. Is that true?
> I can understand choppy audio play back due to lost packets
> because they did not get captured, but I don't think the
> audio file created will represent, and produce during
> playback, the poor audio quality due to jitter and delay.
Those effects are "absorbed" at receiver side by using a jitter buffer.
So you only need to emulate this jitter buffer while decoding the
packets to audio or video to emulate what the user experience is.
Provided, of course, that the receiver implements jitter buffer the same
way as your emulation does (fixed jitter buffer, variable jitter buffer,
...).
> I have been looking on the Web for a few days, but cannot
> seem to find any specific discussion of this question. I see
> numerous examples of many tools providing a mechanism to play
> back the audio captured in RTP streams, but none of them go
> into any detail on whether the play back accurately
> represents the original user's call quality experience.
Currently, Ethereal does not allow you to listen to RTP audio from
within Ethereal. Search "RTP" on Ethereal's wiki to see what is possible
using RTP analysis and other tools.
But stay tuned! Alejandro Vaquero is working on a plugin that provides
this ability, as well as taking into account jitter buffer size while
decoding.
Olivier.
