You'll have to do a git bisect if you propose another version works better.<div><br></div><div><br><span></span><div><br>On Sunday, May 10, 2015, Stanislav Sinyagin <<a href="mailto:ssinyagin@gmail.com">ssinyagin@gmail.com</a>> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">the problem disappears after rolling back to December code, so I<br>
opened a Jira ticket:<br>
<a href="https://freeswitch.org/jira/browse/FS-7541" target="_blank">https://freeswitch.org/jira/browse/FS-7541</a><br>
<br>
<br>
<br>
On Sun, May 10, 2015 at 4:42 PM, Stanislav Sinyagin <<a href="javascript:;" onclick="_e(event, 'cvml', 'ssinyagin@gmail.com')">ssinyagin@gmail.com</a>> wrote:<br>
> I've run it on a different Xen server which is more powerful and more<br>
> busy, and got also the same chopping audio in native recording.<br>
><br>
><br>
><br>
> On Sat, May 9, 2015 at 12:12 AM, Stanislav Sinyagin <<a href="javascript:;" onclick="_e(event, 'cvml', 'ssinyagin@gmail.com')">ssinyagin@gmail.com</a>> wrote:<br>
>> I'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'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>
>> <extension name="record"><br>
>> <condition field="destination_number" expression="^record_(.+)$"><br>
>> <action application="jitterbuffer" data="60:200:20"/><br>
>> <action application="set" data="RECORD_READ_ONLY=true"/><br>
>> <action application="set" data="send_silence_when_idle=400"/><br>
>> <action application="set" data="record_waste_resources=true"/><br>
>> <action application="answer"/><br>
>> <action application="record_session" data="/var/tmp/record_$1"/><br>
>> <action application="playback" data="silence_stream://-1"/><br>
>> </condition><br>
>> </extension><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 'originate sofia/external/<a href="http://record_03@111.222.222.111:5080" target="_blank">record_03@111.222.222.111:5080</a><br>
>> &playback(/var/tmp/ITU-T_P_50_BRITISH_ENGLISH.wav)'<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'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></div><br><br>-- <br><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> ☎ +19193869900 </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><br>