<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=iso-2022-jp">
<META content="MSHTML 6.00.2900.3268" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff><FONT face="MS UI Gothic">
<DIV>Hello,<BR></DIV>
<DIV>I saw source code of Asterisk (1.4.3), but I wonder if VoIP RTP
packets are <BR>passed through jitterbuffer when SIP softphone participate
in MeetMe conference.</DIV>
<DIV> </DIV>
<DIV>I found out that jitterbuffer handling is executed at normal peer-to-peer
dialing<BR>(bridging two channels, ast_generic_bridge()).<BR>But jitterbuffer
handling procedure seems not to be executed, <BR>when SIP softphone participate
in MeetMe conference through SIP channel.</DIV>
<DIV> </DIV>
<DIV>I want to know where jitterbuffer handling is executed in MeetMe
conference, <BR>or is not executed.</DIV>
<DIV> </DIV>
<DIV>I use ztdummy for zaptel pseudo device at conference.<BR>Is zaptel(ztdummy)
handling jitterbuffer?<BR> <BR>Best
regards,<BR>Takashi.</FONT></DIV></BODY></HTML>