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

[Damien Sandras <dsandras seconix com>]

Le samedi 21 mai 2005 à 09:37 +0200, Andre Schaefer a écrit :
> Am Freitag, den 20.05.2005, 22:30 +0200 schrieb Damien Sandras:
> > Hello,
> > 
> > I've discovered a new problem in OPAL (ahah), when being connected in
> > SIP mode to Asterisk, the jitter buffer keeps increasing.
> > 
> > As the H.323 part doesn't have that problem and is using the same code,
> > it could be an Asterisk bug.
> > 
> > I know some of you are using GM with SIP providers. Can you make a call
> > and watch the jitter buffer value in the Statistics window and tell me
> > if it is also increasing?
> 
> As a quick test, I called the viocemailbox of sipgate. It changes
> dynamically between 40 and 80ms, back and forth. It does not increase
> beyound 80 ms.
> 
> Hope that helps,
> 

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?
-- 
 _      Damien Sandras
(o-     GnomeMeeting: http://www.gnomemeeting.org/
//\     FOSDEM 2005 : http://www.fosdem.org
v_/_    H.323 phone : callto:ils.seconix.com/dsandras seconix com