Did you do any debugging info on the other side?<br>What client is it?<br><br>I can speak from experience that I use the conference with like 10 people in it at a minimum of 8 hours a day<br>and there are never any problems.<br>
<br>Try calling your endpoints into the FS public conference to compare<br><br><a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br><br>this is just a lowly VM<br><br><br><div class="gmail_quote">
On Thu, Mar 25, 2010 at 2:50 PM, Chris Fowler <span dir="ltr">&lt;chris@fowler.cc&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">









<div link="blue" vlink="purple" lang="EN-US">

<div>

<p class="MsoNormal"><span style="color: rgb(31, 73, 125);">Forgot to mention : Running
trunk – build 17104</span></p>

<p class="MsoNormal"><span style="color: rgb(31, 73, 125);"> </span></p>

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<p class="MsoNormal"><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;"> Chris Fowler <br>
<b>Sent:</b> Thursday, March 25, 2010 12:48 PM<br>
<b>To:</b> &#39;<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>&#39;<br>
<b>Subject:</b> Guidance on debugging audio glitches</span></p>

</div>

</div><div><div></div><div class="h5">

<p class="MsoNormal"> </p>

<p class="MsoNormal">Hi,</p>

<p class="MsoNormal"> </p>

<p class="MsoNormal">Here’s the problem I’m “hearing”:</p>

<p class="MsoNormal"> </p>

<p class="MsoNormal">Test Case #1</p>

<p class="MsoNormal">x1001 calls x1002</p>

<p class="MsoNormal">- Audio is rock solid</p>

<p class="MsoNormal"> </p>

<p class="MsoNormal">Test Case #2</p>

<p class="MsoNormal">x1001 calls x3001 (Conference Bridge)</p>

<p class="MsoNormal">x1002 calls x3001 (Conference Bridge)</p>

<p class="MsoNormal">- Audio is mostly perfect.  However, occasional
“glitches” in the audio can be heard</p>

<p class="MsoNormal"> </p>

<p class="MsoNormal">Notes:</p>

<p class="MsoNormal">- Bypass media is switched off.  FreeSWITCH routes
audio for both the bridged, and the conferenced call</p>

<p class="MsoNormal">- rtp<span style="color: black;">-</span>autoflush<span style="color: black;">-during-bridge is “true” on Sofia profile internal.xml</span></p>

<p class="MsoNormal"><span style="color: black;">-
I enabled the debug trace for auto-flush in switch_rtp.c (line 1895) and see
occasional auto flush of rtp packets for *<b>both</b>* test cases.  The
glitches don’t coincide with the flushing.  In fact with flushing you
can’t hear when data gets flushed (2x20ms packets).</span></p>

<p class="MsoNormal"><span style="color: black;"> </span></p>

<p class="MsoNormal">Environment</p>

<p class="MsoNormal">- CentOS 5.4</p>

<p class="MsoNormal">- Tested on AWS EC2 Small, Medium, Large instances.  On
Medium and Large CPU affinity for FreeSWITCH set to the 2<sup>nd</sup> core.</p>

<p class="MsoNormal"> </p>

<p class="MsoNormal">Questions:</p>

<p class="MsoNormal">- What’s different between bridging a call and a conference
call run through mod_conference in terms of audio processing?  </p>

<p class="MsoNormal">- Thoughts on how to debug this?</p>

<p class="MsoNormal"> </p>

<p class="MsoNormal">Thanks, Chris.</p>

</div></div></div>

</div>


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