<div dir="ltr">Hi,<div><br></div><div>I was facing lot of audio quality issues and turning off sip jitter buffer didn't solved anything. I set rtp-timer-name = none (it was set to soft earlier) and it seemed to solved the problem (<a href="http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075616.html">http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075616.html</a>). </div>
<div><br></div><div>Can someone explain what was the issue and how it got fixed? I wasn't able to find any documentation related to "rtp-timer-name".<br clear="all"><div><br></div>-- <br><div dir="ltr">Thanks and Regards,<br>
Ashwin Jain<br></div>
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