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



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,

* André
-- 
Andre Schaefer <a schaefer uni-duisburg de>

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