Am Samstag, den 21.05.2005, 18:14 +0200 schrieb Damien Sandras: > But your > tests seem to indicate that the problem is in Asterisk. I will try to > see if the audio becomes bad with time or if it is just the info in the > rtp packets that make the jitter increase when it should not. Asterisk > has no RTCP channel. I may be entirely wrong, but the info in my call history seems to indicate, that sipgate does use Asterisk PBX. So I assumed I was calling via Asterisk... -- Andre Schaefer <a schaefer uni-duisburg de>
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