<div dir="ltr">You are basically turning off the feature that allows the remote end to stop sending audio but have FS still process the stream.<div>The timer will allow the channel to read at the specified interval even when the other side stops sending audio.</div>
<div><br></div><div>If the other end has a clock sync issue with FS or FS does not have a reliable timing source, it could cause your problem.</div><div><br></div><div>If you happen to have a sipura or linksys, check the wiki for specific ways to mitigate some issues.</div>
<div><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Dec 2, 2013 at 6:06 AM, Ashwin Jain <span dir="ltr"><<a href="mailto:ashwinrkjain@gmail.com" target="_blank">ashwinrkjain@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><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" target="_blank">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".<span class="HOEnZb"><font color="#888888"><br clear="all">
<div><br></div>-- <br><div dir="ltr">Thanks and Regards,<br>
Ashwin Jain<br></div>
</font></span></div></div>
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