<div dir="ltr">That was a memorable milestone in FS development history the first time we broke that barrier and got confused why it was not working.... =D  Before that it was unheard of to saturate 100mbps with existing voip software.....<div>
<br></div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Oct 22, 2013 at 5:58 PM, Donny Hardyanto <span dir="ltr">&lt;<a href="mailto:hardyanto.donny@gmail.com" target="_blank">hardyanto.donny@gmail.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><p dir="ltr"> </p>
<div class="gmail_quote">Actually G711 will consumes abaout 167kbps for a call (64 kbps for each side, and the other is IP header, RTP header, Ethernet header) so 800 calls of this will be 133.8 Mbps. If your line is 100 Mbps then you have saturated the line.</div>
<span class="HOEnZb"><font color="#888888">
<div class="gmail_quote"><br></div><div class="gmail_quote">Donny</div><div class="gmail_quote"><br></div></font></span><div class="gmail_quote"><div class="im">---------- Forwarded message ----------<br>From: &quot;Stanislav Sinyagin&quot; &lt;<a href="mailto:ssinyagin@yahoo.com" target="_blank">ssinyagin@yahoo.com</a>&gt;<br>

Date: Oct 22, 2013 4:12 PM<br>Subject: Re: [Freeswitch-users] Audio quality issues<br></div><div class="im">
To: &quot;FreeSWITCH Users Help&quot; &lt;<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>&gt;<br></div><div><div class="h5">Cc: <br><br type="attribution">
<div><div style="font-size:12pt;font-family:HelveticaNeue,Helvetica Neue,Helvetica,Arial,Lucida Grande,sans-serif">

800 calls at 64kbps is 51Mbps. <br>Could there be a network issue, like a 100Mbps line between the endpoints?<br><br>How heavy is your CPU load?  &quot;htop&quot; command would be helpful in this.<br><br><br><div><span><br>


</span></div><div><br></div>  <div style="font-family:HelveticaNeue,Helvetica Neue,Helvetica,Arial,Lucida Grande,sans-serif;font-size:12pt"> <div style="font-family:HelveticaNeue,Helvetica Neue,Helvetica,Arial,Lucida Grande,sans-serif;font-size:12pt">


 <div dir="ltr"> <hr size="1">  <font face="Arial"> <b><span style="font-weight:bold">From:</span></b> Grant Bagdasarian &lt;<a href="mailto:GB@cm.nl" target="_blank">GB@cm.nl</a>&gt;<br> <b><span style="font-weight:bold">To:</span></b> &quot;FreeSWITCH Users Help (<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>)&quot; &lt;<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>&gt; <br>



 <b><span style="font-weight:bold">Sent:</span></b> Tuesday, October 22, 2013 10:14 AM<br> <b><span style="font-weight:bold">Subject:</span></b> [Freeswitch-users] Audio quality issues<br> </font> </div> <div><br><div><div>


<div><div>Hello,</div><div>  </div><div>I was wondering what the maximum concurrent calls for FS before audio quality becomes an issue? I assume the specs of the machine would also affect this.</div><div>We are currently running FS on a Six Core (12 Threads) Intel E5-2430 CPU and get about 800 concurrent calls at 10-20 CPS. The audio quality at these rates is still fair, but we do notice some quality issue’s. </div>


<div>Going above these numbers screws up the audio quality: choppy sound, audio drops etc. We aren’t doing any heavy media processing, just simply playing a file (G711-Alaw) which lasts about 2 minutes during the load test.</div>


<div>These numbers are for one way audio, where Sipp doesn’t echo the RTP
 back. These numbers get lower once Sipp echo’s the RTP.</div><div>  </div><div>I’ve tried FS on a physical box and also on a virtual box (ESXi 5.1), but the performance gain on physical vs virtual isn’t that much. </div>


<div>  </div><div>I disabled all the modules we don’t need, like CDR’s, conferencing, etc.</div><div>  </div><div>Are there any parameters(config files)/modules that can affect the quality of the audio stream?</div><div>

  </div>
<div>Regards,</div><div>  </div><div>Grant</div><div>  </div><div>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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