<div dir="ltr"><div class="gmail_default" style="font-family:arial,helvetica,sans-serif">Hi</div><div class="gmail_default" style="font-family:arial,helvetica,sans-serif"><br></div><div class="gmail_default" style="font-family:arial,helvetica,sans-serif">
I have three questions about the jitterbuffer (with FreeSwithc 1.2.22).</div><div class="gmail_default" style="font-family:arial,helvetica,sans-serif"><br></div><div class="gmail_default" style="font-family:arial,helvetica,sans-serif">
1) I seem to fail configuring the jitterbuffer via the dialplan using the channel variable jitterbuffer_msec as described here: <a href="https://wiki.freeswitch.org/wiki/Jitterbuffer">https://wiki.freeswitch.org/wiki/Jitterbuffer</a></div>
<div class="gmail_default" style="font-family:arial,helvetica,sans-serif"><br></div><div class="gmail_default" style="font-family:arial,helvetica,sans-serif">Here's the part of my dialplan:</div><div class="gmail_default" style="font-family:arial,helvetica,sans-serif">
<br></div><div class="gmail_default"><div class="gmail_default"><font face="arial, helvetica, sans-serif"><span style="white-space:pre"><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>
<br></span></font></div><div>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.</div>
<div><br></div><div>I think this might have something to do that the variable <span style="font-family:arial,helvetica,sans-serif;white-space:pre">jitterbuffer_msec is not available yet when </span>sofia_glue_activate_rtp() gets executed. Correct? Maybe related to using inline="true"? But even if I use <span style="font-family:arial,helvetica,sans-serif;white-space:pre"><action inline="true" application="set" data="jitterbuffer_msec=60:200:20"/>, the jitterbuffer does not seem to get configured.</span></div>
<div><br></div><div>=> question 1: how can I configure the jitterbuffer via the <span style="font-family:arial,helvetica,sans-serif">dialplan using the channel variable jitterbuffer_msec?</span></div><div><span style="font-family:arial,helvetica,sans-serif"><br>
</span></div><div><span style="font-family:arial,helvetica,sans-serif">Btw: enabling </span><font face="arial, helvetica, sans-serif"><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.</font></div>
<div><span style="font-family:arial,helvetica,sans-serif"><br></span></div><div><span style="font-family:arial,helvetica,sans-serif">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?</span></div>
<div><span style="font-family:arial,helvetica,sans-serif"><br></span></div><div><span style="font-family:arial,helvetica,sans-serif">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.</span></div>
<div><span style="font-family:arial,helvetica,sans-serif"><br></span></div><div><span style="font-family:arial,helvetica,sans-serif">Thanks and best regards, Markus</span></div><div><span style="font-family:arial,helvetica,sans-serif"><br>
</span></div></div></div>