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

[Damien Sandras <dsandras seconix com>]

Le dimanche 22 mai 2005 à 11:55 +0200, Daniel Huhardeaux a écrit :
> Damien Sandras a écrit :
> 
> >Hello,
> 
> Hi Damien
> >
> >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?
> >  
> >
> I called through my asterisk box to:
> 
> . FWD echo test ~ 2mn20s. Jitter moved in ms like this 50-40-37-35-60.
> 
> . Telco provider, also echo test, 3mn: 40-37-35-50-60-40-37-60-120-95-80
> 
> Asterisk CVS-HEAD 05/07/05

That seems pretty normal. So the problem must be on my side. But please
continue watching the jitter buffer!

Thanks,

> 
-- 
 _      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