<div dir="ltr">I would try both and see if there is any difference. Echo should not create an overhead but you never konow until you test it :)<div><br></div><div>Please do post your results here. These would be some valuable info for the community.</div>
<div><br><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Oct 22, 2013 at 10:03 AM, Grant Bagdasarian <span dir="ltr"><<a href="mailto:GB@cm.nl" target="_blank">GB@cm.nl</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Yes, it does! <u></u><u></u></span></p><p class="MsoNormal">
<span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">I also found this: </span><a href="https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo" target="_blank">https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo</a><u></u><u></u></p>
<p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">Echo application for FS. <u></u><u></u></p><p class="MsoNormal">Going to see if that works first. If not, I’ll setup a Sipp in server mode.<span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>] <b>On Behalf Of </b>Ali Pey<br>
<b>Sent:</b> Tuesday, October 22, 2013 3:50 PM</span></p><div><div class="h5"><br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] Audio quality issues<u></u><u></u></div></div><p></p><p class="MsoNormal">
<u></u> <u></u></p><div><div><div class="h5"><p class="MsoNormal">You can make calls from sipp that also terminates on sipp and then play a wave file in sipp.<u></u><u></u></p><div><p class="MsoNormal"><u></u> <u></u></p>
</div><div><p class="MsoNormal">Change your dial plan in FS to route the calls to an instant of sipp that can terminate the calls.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div></div></div><div>
<div><div class="h5"><p class="MsoNormal">Does this make sense?<u></u><u></u></p><div><p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p><div><p class="MsoNormal">On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian <<a href="mailto:GB@cm.nl" target="_blank">GB@cm.nl</a>> wrote:<u></u><u></u></p>
<div><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">A while back I used iotop to measure the disk access, and FS was hardly using any io during tests.</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">How do I simulate two way audio? </span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">I know I can make Sipp send an RTP stream using a pcap file, but how do I make FS sent RTP back which is not read from disk? Does FS have an echo application?</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Or is it enough for Sipp to send the media?</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p>
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>] <b>On Behalf Of </b>Ali Pey<br>
<b>Sent:</b> Tuesday, October 22, 2013 2:25 PM</span><u></u><u></u></p><div><div><p class="MsoNormal"><br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] Audio quality issues<u></u><u></u></p></div>
</div><p class="MsoNormal"> <u></u><u></u></p><div><div><div><p class="MsoNormal">I think the problem here is that you are playing a file for every call for the duration of the call. The bottleneck seems to be the disk access. If there were to be two way audio path, FS would only proxy the media which would be quite faster as there would be no file reading and playing involved. Attempt a test case with fewer or no file play and only media proxy and test again.<u></u><u></u></p>
<div><p class="MsoNormal"> <u></u><u></u></p></div></div></div><div><div><div><div><p class="MsoNormal" style="margin-bottom:12.0pt"> <u></u><u></u></p><div><p class="MsoNormal">On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian <<a href="mailto:GB@cm.nl" target="_blank">GB@cm.nl</a>> wrote:<u></u><u></u></p>
<div><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">The network shouldn’t be an issue, since we have at least 1Gbps lines. The tests stay within the network.</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">I forgot to mention the calls are being distributed across two machines by a Kamailio instance.</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls.</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">CPU load reaches about 70% per machine.</span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">At this point both FS machines are virtualized, since the performance gain wasn’t that much compared to physical. </span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">The VM host shows it is using ~3/4 of its CPU resources.</span><u></u><u></u></p><p class="MsoNormal">
<span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length. </span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Also, FS is running on Ubuntu Server 12.04 x64.</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p><div><div style="border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>] <b>On Behalf Of </b>Stanislav Sinyagin<br>
<b>Sent:</b> Tuesday, October 22, 2013 11:11 AM<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] Audio quality issues</span><u></u><u></u></p></div></div><div><div><p class="MsoNormal"> <u></u><u></u></p>
<div><p class="MsoNormal" style="margin-bottom:12.0pt;background:white"><span style="font-family:"Helvetica","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? "htop" command would be helpful in this.</span><u></u><u></u></p><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif""> </span><u></u><u></u></p>
</div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif""> </span><u></u><u></u></p></div><div><div><div><div class="MsoNormal" align="center" style="text-align:center;background:white">
<span style="font-family:"Helvetica","sans-serif""><hr size="1" width="100%" align="center"></span></div><p class="MsoNormal" style="background:white"><b><span style="font-size:10.0pt;font-family:"Arial","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Arial","sans-serif""> Grant Bagdasarian <<a href="mailto:GB@cm.nl" target="_blank">GB@cm.nl</a>><br>
<b>To:</b> "FreeSWITCH Users Help (<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>)" <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>> <br>
<b>Sent:</b> Tuesday, October 22, 2013 10:14 AM<br><b>Subject:</b> [Freeswitch-users] Audio quality issues</span><u></u><u></u></p></div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif""> </span><u></u><u></u></p>
<div><div><div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif"">Hello,</span><u></u><u></u></p></div><div><p class="MsoNormal" style="background:white">
<span style="font-family:"Helvetica","sans-serif""> </span><u></u><u></u></p></div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif"">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.</span><u></u><u></u></p>
</div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif"">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. </span><u></u><u></u></p>
</div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif"">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.</span><u></u><u></u></p>
</div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif"">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.</span><u></u><u></u></p>
</div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif""> </span><u></u><u></u></p></div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif"">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. </span><u></u><u></u></p>
</div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif""> </span><u></u><u></u></p></div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif"">I disabled all the modules we don’t need, like CDR’s, conferencing, etc.</span><u></u><u></u></p>
</div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif""> </span><u></u><u></u></p></div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif"">Are there any parameters(config files)/modules that can affect the quality of the audio stream?</span><u></u><u></u></p>
</div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif""> </span><u></u><u></u></p></div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif"">Regards,</span><u></u><u></u></p>
</div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif""> </span><u></u><u></u></p></div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif"">Grant</span><u></u><u></u></p>
</div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif""> </span><u></u><u></u></p></div><div><p class="MsoNormal" style="background:white"><span style="font-family:"Helvetica","sans-serif""> </span><u></u><u></u></p>
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