[Freeswitch-users] Clock precision and raw audio recording

[Stanislav Sinyagin]

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.

I can also try giving raw audio to playback, instead of 16-bit wav.

But basically, it's not a bug, but rather a feature allowing to check the
clock precision. But it will be great if we understand its nature :)



On May 10, 2015 12:47 AM, "Anthony Minessale" <anthony.minessale at gmail.com>
wrote:

> Guess:
>
> Did you try using it with the jitter buffer enabled?
> The raw recording cannot pre-buffer the audio so its more likely jitter
> may be rendered into the file.
> Sometimes the gaps that are experienced are just from late packets.
>
> The WAV recordings have a more complicated process for buffering and
> mixing the file.
>
>
>
> On Fri, May 8, 2015 at 5:12 PM, Stanislav Sinyagin <ssinyagin at gmail.com>
> wrote:
>
>> 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
>>
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>
>
>
> --
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