<p dir="ltr">+1<br></p>
<p dir="ltr">sent from my mobile,<br>
Giovanni Maruzzelli<br>
cell: +39 347 266 56 18</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: <a href="tel:%28877%29%20211-9337" value="+18772119337">(877) 211-9337</a><br>
Office: <a href="tel:%28206%29%20388-4772" value="+12063884772">(206) 388-4772</a><br>
Cell: <a href="tel:%28253%29%20686-5518" value="+12536865518">(253) 686-5518</a><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>
> _________________________________________________________________________<br>
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><br>
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><br>
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><br>
<br>
_________________________________________________________________________<br>
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<br>
Official FreeSWITCH Sites<br>
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<a href="http://confluence.freeswitch.org" target="_blank">http://confluence.freeswitch.org</a><br>
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</blockquote></div>