[asterisk-dev] jitterbuffer on MeetMe conference

[Takashi Ohashi]

Hello,

I saw source code of Asterisk (1.4.3), but I wonder if VoIP RTP packets are
passed through jitterbuffer when SIP softphone participate in MeetMe 
conference.

I found out that jitterbuffer handling is executed at normal peer-to-peer 
dialing
(bridging two channels, ast_generic_bridge()).
But jitterbuffer handling procedure seems not to be executed,
when SIP softphone participate in MeetMe conference through SIP channel.

I want to know where jitterbuffer handling is executed in MeetMe conference,
or is not executed.

I use ztdummy for zaptel pseudo device at conference.
Is zaptel(ztdummy) handling jitterbuffer?

Best regards,
Takashi. 
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