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

[Damien Sandras <dsandras seconix com>]

Le samedi 21 mai 2005 à 13:10 +0200, Andre Schaefer a écrit :
> 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.
> 

Yes, it is another bug in opal.

> One side had no statistical infomation each time, which was a bit
> strange. Is it only computed if you are calling?

No, it should always be there, I will investigate further. 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.
-- 
 _      Damien Sandras
(o-     GnomeMeeting: http://www.gnomemeeting.org/
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