<p dir="ltr">Jitterbuffer was always enabled on the receiving side in all tests, like in the piece of  config in my email. I can try disabling it.</p>
<p dir="ltr">I can also try giving raw audio to playback, instead of 16-bit wav. </p>
<p dir="ltr">But basically, it&#39;s not a bug, but rather a feature allowing to check the clock precision. But it will be great if we understand its nature :)<br><br><br><br></p>
<div class="gmail_quote">On May 10, 2015 12:47 AM, &quot;Anthony Minessale&quot; &lt;<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>&gt; wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>Guess:</div><div><br></div>Did you try using it with the jitter buffer enabled?<div>The raw recording cannot pre-buffer the audio so its more likely jitter may be rendered into the file.</div><div>Sometimes the gaps that are experienced are just from late packets.</div><div><br></div><div>The WAV recordings have a more complicated process for buffering and mixing the file.</div><div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, May 8, 2015 at 5:12 PM, Stanislav Sinyagin <span dir="ltr">&lt;<a href="mailto:ssinyagin@gmail.com" target="_blank">ssinyagin@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I&#39;m observing an effect which needs explanation. Comments from the<br>
core developers will be appreciated. The effect was tested with<br>
versions 1.4.18 and today&#39;s master, on 64-bit Debian 7 and Ubuntu<br>
14.04. All test calls were in PCMU or PCMA.<br>
<br>
My customer requested me to build a server for call quality assurance<br>
for their telephony system. I installed FreeSWITCH and set the<br>
following in the public dialplan to record the incoming audio:<br>
<br>
    &lt;extension name=&quot;record&quot;&gt;<br>
      &lt;condition field=&quot;destination_number&quot; expression=&quot;^record_(.+)$&quot;&gt;<br>
        &lt;action application=&quot;jitterbuffer&quot; data=&quot;60:200:20&quot;/&gt;<br>
        &lt;action application=&quot;set&quot; data=&quot;RECORD_READ_ONLY=true&quot;/&gt;<br>
        &lt;action application=&quot;set&quot; data=&quot;send_silence_when_idle=400&quot;/&gt;<br>
        &lt;action application=&quot;set&quot; data=&quot;record_waste_resources=true&quot;/&gt;<br>
        &lt;action application=&quot;answer&quot;/&gt;<br>
        &lt;action application=&quot;record_session&quot; data=&quot;/var/tmp/record_$1&quot;/&gt;<br>
        &lt;action application=&quot;playback&quot; data=&quot;silence_stream://-1&quot;/&gt;<br>
      &lt;/condition&gt;<br>
    &lt;/extension&gt;<br>
<br>
The first try was with a DigitalOcean (KVM) virtual machine. I started<br>
recording *.wav files, and sometimes there were skipped frames: 1-2<br>
skipped frames in a 2-minute call, one in every 10-15 calls.<br>
<br>
Then I changed the configuration to write raw audio files (removed the<br>
.wav extension from the record_session argument). As a result, the<br>
recorded input audio was quite bad: lost frames every few seconds in<br>
every call.<br>
<br>
Then I made test calls within the server itself, by originating a call<br>
to its public profile and playing the test WAV audio:<br>
<br>
fs_cli  -x &#39;originate sofia/external/<a href="http://record_03@111.222.222.111:5080" target="_blank">record_03@111.222.222.111:5080</a><br>
&amp;playback(/var/tmp/ITU-T_P_50_BRITISH_ENGLISH.wav)&#39;<br>
<br>
The resulting input raw audio was also choppy. The same result was on<br>
another VM on a different physical server at DigitalOcean.<br>
<br>
Then I made the same self-call tests on a Xen VM and on a baremetal<br>
ARM server, and there the recorded audio was of perfect quality.<br>
<br>
Self-calls with recording into WAV files produced audio of perfect quality.<br>
<br>
Setting RECORD_USE_THREAD=false did not change the effect.<br>
<br>
Example of choppy received audio, converted from PCMU to WAV for convenience:<br>
<a href="http://www.k-open.com/s/record_04-in.wav" target="_blank">http://www.k-open.com/s/record_04-in.wav</a><br>
The source audio:<br>
<a href="http://murmur.voxserv.ch/media/ITU-T_P_50_BRITISH_ENGLISH.wav" target="_blank">http://murmur.voxserv.ch/media/ITU-T_P_50_BRITISH_ENGLISH.wav</a><br>
<br>
So, it looks like the clock that is available at KVM is not precise<br>
enough, but that&#39;s not my question.<br>
<br>
QUESTION: why is raw recording so much more sensitive to the clock precision?<br>
<br>
thanks,<br>
stanislav<br>
<br>
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</blockquote></div><br><br clear="all"><div><br></div>-- <br><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr">Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬<div><br><div>☞ <a href="http://freeswitch.org/" target="_blank">http://freeswitch.org/</a>  ☞ <a href="http://cluecon.com/" target="_blank">http://cluecon.com/</a>  ☞ <a href="http://twitter.com/FreeSWITCH" target="_blank">http://twitter.com/FreeSWITCH</a></div><div><div>☞ <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch ☞ <u><a href="http://freeswitch.org/g+" target="_blank">http://freeswitch.org/g+</a></u><br><br></div><div>ClueCon Weekly Development Call <br></div><div>☎ <a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a>  ☎ <a href="tel:%2B19193869900" value="+19193869900" target="_blank">+19193869900</a> </div><div><br></div></div></div><div><a href="https://www.youtube.com/watch?v=9XXgW34t40s" style="color:rgb(17,85,204);font-size:12.8000001907349px" target="_blank">https://www.youtube.com/watch?v=9XXgW34t40s</a></div><div><a href="https://www.youtube.com/watch?v=NLaDpGQuZDA" target="_blank">https://www.youtube.com/watch?v=NLaDpGQuZDA</a><br></div></div></div></div></div></div></div>
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