[Freeswitch-users] Clock precision and raw audio recording

[Anthony Minessale]

You'll have to do a git bisect if you propose another version works better.



On Sunday, May 10, 2015, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:

> the problem disappears after rolling back to December code, so I
> opened a Jira ticket:
> https://freeswitch.org/jira/browse/FS-7541
>
>
>
> On Sun, May 10, 2015 at 4:42 PM, Stanislav Sinyagin <ssinyagin at gmail.com
> <javascript:;>> wrote:
> > I've run it on a different Xen server which is more powerful and more
> > busy, and got also the same chopping audio in native recording.
> >
> >
> >
> > On Sat, May 9, 2015 at 12:12 AM, Stanislav Sinyagin <ssinyagin at gmail.com
> <javascript:;>> 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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-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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