Re: [GnomeMeeting-devel-list] To SIP users not using Asterisk

[Damien Sandras <dsandras seconix com>]

Le samedi 21 mai 2005 à 18:49 +0200, Andre Schaefer a écrit :
> 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...


Then it could be my Asterisk version that is buggy. Anyway, please all
check your jitter buffer and report back if you have weird results.

Thank you!


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 _      Damien Sandras
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