I made additional tests by sending calls to sipgate.co.uk, and they confirm my results: when sending a call from my Xen server in Switzerland, the received audio is always perfect. But when sending from a Digitalocean host in Amsterdam, I get the slipping frames.

I will re-test again during a business day, when the servers are likely to be more busy.



On Sun, Apr 5, 2015 at 12:47 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:

Yrs, jitter buffer was enabled. Here I'm pretty sure that we deal with the clock skew.

On Apr 5, 2015 12:07 PM, "Brian West" <brian@freeswitch.org> wrote:
Correct me if I'm wrong but is sounds like you've described jitter.  Did you test with the jitter buffer enabled?

On Sunday, April 5, 2015, Regis M <regis.freeswitch.org@tornad.net> wrote:
nice work !!!  Thanks for sharing.



2015-04-05 2:37 GMT+02:00 Stanislav Sinyagin <ssinyagin@gmail.com>:
here we go:
https://txlab.wordpress.com/2015/04/05/testing-rtp-clock-precision-on-a-virtual-machine/



On Sat, Apr 4, 2015 at 4:26 PM, Stanislav Sinyagin <ssinyagin@gmail.com> wrote:
> On Sat, Apr 4, 2015 at 1:03 PM, Regis M <regis.freeswitch.org@tornad.net> wrote:
>> How do you check wavs together in audacity ?
>
> you just open two WAV files, align their starting position, and
> visually check the waves. If there's a problem, you would see that
> after few seconds, the waves are not aligned any more.
>
>> Can you send me 2 samples wav (send... and recorded) so I could see the 20ms
>> diff that you exposed...
>>
>> Is there a way to automate the result checking, so it could help to monitor
>> itsp quality
>
> Here
> https://github.com/openpreserve/scape-xcorrsound/issues/17
> this is a link to a tool which compares two sound files at the wave
> level. You can also see the sound samples in the ticket.
> The problem is, that currently this tool does not detect such slipping frames.
>
> I will also write a detailed article with examples, during the weekend.
>
>
>
>
>
>>
>>
>> 2015-04-04 12:53 GMT+02:00 Stanislav Sinyagin <ssinyagin@gmail.com>:
>>>
>>> Important is to send the media through two independent carriers - then you
>>> can be sure that your clock is compared against a high quality one.
>>>
>>> On Apr 4, 2015 12:46 PM, "Regis M" <regis.freeswitch.org@tornad.net>
>>> wrote:
>>>>
>>>> Hi,
>>>>
>>>> It's a very interresting test and conclusion.
>>>>
>>>> I will try to build a similar environnement on my freeswitch to check
>>>> some system :)
>>>>
>>>> thanks
>>>>
>>>>
>>>>
>>>> 2015-04-04 2:16 GMT+02:00 Stanislav Sinyagin <ssinyagin@gmail.com>:
>>>>>
>>>>> hi,
>>>>>
>>>>> Here are some test results which might be interesting:
>>>>>
>>>>> I've set up a test FreeSWITCH server at digitalocean (they use KVM),
>>>>> and originated test calls to some ITSP accounts. Each call played a
>>>>> test sound file (voice samples from ITU:
>>>>> http://www.itu.int/net/itu-t/sigdb/genaudio/AudioForm-g.aspx?val=1000050
>>>>> ), and on the receiving end another FreeSWITCH server was recording
>>>>> the received session. The receiving server was on a XEN VM. Both
>>>>> servers talked G711 to their corresponding ITSP accounts.
>>>>>
>>>>> Then I compared the source audio and received audio in Audacity editor.
>>>>>
>>>>> The tests showed that the received voice wave shifts 20ms from the
>>>>> original sound every few seconds, randomly back and forward. This
>>>>> skewing was always present, regardless of ITSP and time of day.
>>>>>
>>>>> Then I moved the sending server to a XEN VM, and the waves appeared
>>>>> almost perfectly synchronized (there was a 5ms skew in one test, and I
>>>>> guess it's related to transcoding buffer, as the sound was obviously
>>>>> transcoded).
>>>>>
>>>>> The effect was always visible with a 30-second sample, and I also made
>>>>> additional tests with 2-minute samples.
>>>>>
>>>>> It would also be interesting to test this from a VmWare VM. If someone
>>>>> wants to run such a test, feel free to contact me directly.
>>>>>
>>>>>
>>>>> cheers,
>>>>> stan
>>>>>
>>>>>
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