Hi

I have three questions about the jitterbuffer (with FreeSwithc 1.2.22).

1) I seem to fail configuring the jitterbuffer via the dialplan using the channel variable jitterbuffer_msec as described here: https://wiki.freeswitch.org/wiki/Jitterbuffer

Here's the part of my dialplan:

<extension name="conference"> <condition field="destination_number" expression="^18(.+)$"> <action application="set" data="jitterbuffer_msec=60:200:20"/> <action application="answer"/> <action application="conference" data="$1@audioswitch_conference"/> </condition> </extension>
When I open a channel using this dialplan, the jitterbuffer does not seem to get configured. I assume this because I cannot find a log message "Setting Jitterbuffer to..." from sofia_glue_activate_rtp() in my log file, so I assume that the part of the code setting the jitterbuffer does not execute.

I think this might have something to do that the variable jitterbuffer_msec is not available yet when sofia_glue_activate_rtp() gets executed. Correct? Maybe related to using inline="true"? But even if I use <action inline="true" application="set" data="jitterbuffer_msec=60:200:20"/>, the jitterbuffer does not seem to get configured.

=> question 1: how can I configure the jitterbuffer via the dialplan using the channel variable jitterbuffer_msec?

Btw: enabling <param name="auto-jitterbuffer-msec" value="20"/> in the SIP profile works fine and produces the expected log statement. But I'd rather use the channel variable as it also allows configuring the max buffer length.

2) How can I get the current status (size) of the jitterbuffer at runtime, e.g. from a console (mod_cli) command or via mod_event_socket?

3) Do I need a jitterbuffer at all for the setup with many SIP channels meeting in a FreeSwitch audio conference? My tests with a network with a big jitter indicate yes, because audio quality markedly improves when enabling the jitterbuffer.

Thanks and best regards, Markus