Jacques, Olivier (OpenCall Test Infra) wrote: >> Yes, Wireshark can re-construct the audio, but it's without the >> jitter-buffer of the client device in mind. It merely strings the RTP >> packets together and makes a WAV file. I learned this the hard way. >> > > This is not true anymore. The "VoIP Calls/RTP Player" feature (as > available in latest development releases of Wireshark 0.99.4) allows to > reconstruct the audio _with_ jitter buffer in mind. > > It works this way: > - You specify the jitter buffer size (in ms) > - You press "Decode" button: Wireshark re-construct the audio. > - RTP packets with an excessive jitter are dropped > - The number of RTP packets dropped are counted and displayed > - You can listen to resulting audio from within Wireshark > > See picture attached. > > Of course, this doesn't take into account other client-side parameters > like adaptive jitter buffer, bad clocking, bad RTP implementation, ... > > Last warning, RTP player supports G711 A/u law codecs at the moment. It > is possible to add your own codecs, the RTP player feature being well > designed for that, but codecs licensing issues will certainly prevent > many codecs to be included in Wireshark. > > Olivier. > Shouldn't this info be included in the wiki (it's documented nowhere else AFAIK)?
Regards, ULFL > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------ > > _______________________________________________ > Wireshark-users mailing list > [email protected] > http://www.wireshark.org/mailman/listinfo/wireshark-users > _______________________________________________ Wireshark-users mailing list [email protected] http://www.wireshark.org/mailman/listinfo/wireshark-users
