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

[Daniel Huhardeaux <devel tootai net>]

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

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