<p dir="ltr">Of course I made such direct tests too. They usually don't show anything interesting, as both machines rely on Linux clock. In my tests, the calls were passing through TDM links where the clock should be much more precise.</p>
<div class="gmail_quote">On Apr 9, 2015 9:01 PM, "William King" <<a href="mailto:william.king@quentustech.com">william.king@quentustech.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I'm surprised nobody has suggested doing point to point(FS to FS) within<br>
the same data center testing to remove the provider from the mix, then<br>
doing a packet capture on the receiving box. Wireshark will be able to<br>
determine the network jitter. Once you can determine and compensate for<br>
the level of RTP network jitter, then you can start looking for<br>
potential audio issues.<br>
<br>
I've seen situations like you are describing, and they've all come down<br>
to the host machine was overloaded either at the CPU level, or the<br>
network stack.<br>
<br>
William King<br>
Senior Engineer<br>
Quentus Technologies, INC<br>
1037 NE 65th St Suite 273<br>
Seattle, WA 98115<br>
Main: (877) 211-9337<br>
Office: (206) 388-4772<br>
Cell: (253) 686-5518<br>
<a href="mailto:william.king@quentustech.com">william.king@quentustech.com</a><br>
<br>
On 04/03/2015 05:16 PM, Stanislav Sinyagin wrote:<br>
> hi,<br>
><br>
> Here are some test results which might be interesting:<br>
><br>
> I've set up a test FreeSWITCH server at digitalocean (they use KVM),<br>
> and originated test calls to some ITSP accounts. Each call played a<br>
> test sound file (voice samples from ITU:<br>
> <a href="http://www.itu.int/net/itu-t/sigdb/genaudio/AudioForm-g.aspx?val=1000050" target="_blank">http://www.itu.int/net/itu-t/sigdb/genaudio/AudioForm-g.aspx?val=1000050</a><br>
> ), and on the receiving end another FreeSWITCH server was recording<br>
> the received session. The receiving server was on a XEN VM. Both<br>
> servers talked G711 to their corresponding ITSP accounts.<br>
><br>
> Then I compared the source audio and received audio in Audacity editor.<br>
><br>
> The tests showed that the received voice wave shifts 20ms from the<br>
> original sound every few seconds, randomly back and forward. This<br>
> skewing was always present, regardless of ITSP and time of day.<br>
><br>
> Then I moved the sending server to a XEN VM, and the waves appeared<br>
> almost perfectly synchronized (there was a 5ms skew in one test, and I<br>
> guess it's related to transcoding buffer, as the sound was obviously<br>
> transcoded).<br>
><br>
> The effect was always visible with a 30-second sample, and I also made<br>
> additional tests with 2-minute samples.<br>
><br>
> It would also be interesting to test this from a VmWare VM. If someone<br>
> wants to run such a test, feel free to contact me directly.<br>
><br>
><br>
> cheers,<br>
> stan<br>
><br>
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</blockquote></div>