[Freeswitch-users] Clock precision and raw audio recording

[Stanislav Sinyagin]

I'm observing an effect which needs explanation. Comments from the
core developers will be appreciated. The effect was tested with
versions 1.4.18 and today's master, on 64-bit Debian 7 and Ubuntu
14.04. All test calls were in PCMU or PCMA.

My customer requested me to build a server for call quality assurance
for their telephony system. I installed FreeSWITCH and set the
following in the public dialplan to record the incoming audio:

    <extension name="record">
      <condition field="destination_number" expression="^record_(.+)$">
        <action application="jitterbuffer" data="60:200:20"/>
        <action application="set" data="RECORD_READ_ONLY=true"/>
        <action application="set" data="send_silence_when_idle=400"/>
        <action application="set" data="record_waste_resources=true"/>
        <action application="answer"/>
        <action application="record_session" data="/var/tmp/record_$1"/>
        <action application="playback" data="silence_stream://-1"/>
      </condition>
    </extension>

The first try was with a DigitalOcean (KVM) virtual machine. I started
recording *.wav files, and sometimes there were skipped frames: 1-2
skipped frames in a 2-minute call, one in every 10-15 calls.

Then I changed the configuration to write raw audio files (removed the
.wav extension from the record_session argument). As a result, the
recorded input audio was quite bad: lost frames every few seconds in
every call.

Then I made test calls within the server itself, by originating a call
to its public profile and playing the test WAV audio:

fs_cli  -x 'originate sofia/external/record_03 at 111.222.222.111:5080
&playback(/var/tmp/ITU-T_P_50_BRITISH_ENGLISH.wav)'

The resulting input raw audio was also choppy. The same result was on
another VM on a different physical server at DigitalOcean.

Then I made the same self-call tests on a Xen VM and on a baremetal
ARM server, and there the recorded audio was of perfect quality.

Self-calls with recording into WAV files produced audio of perfect quality.

Setting RECORD_USE_THREAD=false did not change the effect.

Example of choppy received audio, converted from PCMU to WAV for convenience:
http://www.k-open.com/s/record_04-in.wav
The source audio:
http://murmur.voxserv.ch/media/ITU-T_P_50_BRITISH_ENGLISH.wav

So, it looks like the clock that is available at KVM is not precise
enough, but that's not my question.

QUESTION: why is raw recording so much more sensitive to the clock precision?

thanks,
stanislav