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

[Damien Sandras <dsandras seconix com>]

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?
-- 
 _      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