Am Samstag, den 21.05.2005, 12:52 +0200 schrieb Damien Sandras: > > That's a good news. However, I need more reports like this one, doing > calls during 1 to 5 minutes. If after 5 minutes, the jitter is not at > the maximum, then the chance is high that it is another Asterisk bug. > > Any other SIP users? Me again: I called another sipgate user. He also did use gnomemeeting. we talked for 5:00 and for 9:16 min repectively. The jitter buffer behaved well. It scaled between 57 and 65 ms in the one case and around 120 in the other case. The chosen codec was speex, if that matters. One call was not succesfull because the chosen codel iLBC did only transfer noise, somehow. One side had no statistical infomation each time, which was a bit strange. Is it only computed if you are calling? -- Andre Schaefer <a schaefer uni-duisburg de>
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