<html><body><div style="color:#000; background-color:#fff; 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 style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 12pt;"> <div dir="ltr"> "delay_echo" is even more convenient for testing: then you can clearly hear your own voice without distortions from echo cancellation.<br><br><br><hr size="1"> <font face="Arial" size="2"> <b><span style="font-weight:bold;">From:</span></b> Grant Bagdasarian <GB@cm.nl><br> <b><span style="font-weight: bold;">To:</span></b> FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> <br> <b><span style="font-weight: bold;">Sent:</span></b> Tuesday, October 22, 2013 4:03 PM<br> <b><span style="font-weight:
bold;">Subject:</span></b> Re: [Freeswitch-users] Audio quality issues<br> </font> </div> <div class="y_msg_container"><br><div id="yiv6350825511"><style>#yiv6350825511 #yiv6350825511 --
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#yiv6350825511 </style><div><div class="yiv6350825511WordSection1"><div class="yiv6350825511MsoNormal"><span style="font-size:11.0pt;">Yes, it does! </span></div><div class="yiv6350825511MsoNormal"><span style="font-size:11.0pt;"> </span></div><div class="yiv6350825511MsoNormal"><span style="font-size:11.0pt;">I also found this: </span><a rel="nofollow" shape="rect" target="_blank" href="https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo">https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo</a></div><div class="yiv6350825511MsoNormal"> </div><div class="yiv6350825511MsoNormal">Echo application for FS. </div><div class="yiv6350825511MsoNormal">Going to see if that works first. If not, I’ll setup a Sipp in server mode.<span style="font-size:11.0pt;"></span></div><div class="yiv6350825511MsoNormal"><span style="font-size:11.0pt;"> </span></div><div class="yiv6350825511yqt9881622502" id="yiv6350825511yqt90360"><div
class="yiv6350825511MsoNormal"><b><span style="font-size:10.0pt;">From:</span></b><span style="font-size:10.0pt;"> freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] <b>On Behalf Of </b>Ali Pey<br clear="none"><b>Sent:</b> Tuesday, October 22, 2013 3:50 PM<br clear="none"><b>To:</b> FreeSWITCH Users Help<br clear="none"><b>Subject:</b> Re: [Freeswitch-users] Audio quality issues</span></div><div class="yiv6350825511MsoNormal"> </div><div><div class="yiv6350825511MsoNormal">You can make calls from sipp that also terminates on sipp and then play a wave file in sipp.</div><div><div class="yiv6350825511MsoNormal"> </div></div><div><div class="yiv6350825511MsoNormal">Change your dial plan in FS to route the calls to an instant of sipp that can terminate the calls.</div></div><div><div class="yiv6350825511MsoNormal"> </div></div><div><div class="yiv6350825511MsoNormal">Does this make
sense?</div><div><div class="yiv6350825511MsoNormal" style="margin-bottom:12.0pt;"> </div><div><div class="yiv6350825511MsoNormal">On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian <<a rel="nofollow" shape="rect" ymailto="mailto:GB@cm.nl" target="_blank" href="mailto:GB@cm.nl">GB@cm.nl</a>> wrote:</div><div><div><div class="yiv6350825511MsoNormal" style=""><span style="font-size:11.0pt;">A while back I used iotop to measure the disk access, and FS was hardly using any io during tests.</span></div><div class="yiv6350825511MsoNormal" style=""><span style="font-size:11.0pt;"> </span></div><div class="yiv6350825511MsoNormal" style=""><span style="font-size:11.0pt;">How do I simulate two way audio? </span></div><div class="yiv6350825511MsoNormal" style=""><span style="font-size:11.0pt;">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></div><div class="yiv6350825511MsoNormal" style=""><span style="font-size:11.0pt;">Or is it enough for Sipp to send the media?</span></div><div class="yiv6350825511MsoNormal" style=""><span style="font-size:11.0pt;"> </span></div><div class="yiv6350825511MsoNormal" style=""><b><span style="font-size:10.0pt;">From:</span></b><span style="font-size:10.0pt;"> <a rel="nofollow" shape="rect" ymailto="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank" href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a rel="nofollow" shape="rect" ymailto="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank" href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>] <b>On Behalf Of </b>Ali Pey<br clear="none"><b>Sent:</b> Tuesday, October 22, 2013 2:25 PM</span></div><div><div><div
class="yiv6350825511MsoNormal"><br clear="none"><b>To:</b> FreeSWITCH Users Help<br clear="none"><b>Subject:</b> Re: [Freeswitch-users] Audio quality issues</div></div></div><div class="yiv6350825511MsoNormal" style=""> </div><div><div><div><div class="yiv6350825511MsoNormal" style="">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.</div><div><div class="yiv6350825511MsoNormal" style=""> </div></div></div></div><div><div><div><div><div class="yiv6350825511MsoNormal" style="margin-bottom:12.0pt;"> </div><div><div class="yiv6350825511MsoNormal" style="">On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian <<a
rel="nofollow" shape="rect" ymailto="mailto:GB@cm.nl" target="_blank" href="mailto:GB@cm.nl">GB@cm.nl</a>> wrote:</div><div><div><div class="yiv6350825511MsoNormal" style=""><span style="font-size:11.0pt;">The network shouldn’t be an issue, since we have at least 1Gbps lines. The tests stay within the network.</span></div><div class="yiv6350825511MsoNormal" style=""><span style="font-size:11.0pt;"> </span></div><div class="yiv6350825511MsoNormal" style=""><span style="font-size:11.0pt;">I forgot to mention the calls are being distributed across two machines by a Kamailio instance.</span></div><div class="yiv6350825511MsoNormal" style=""><span style="font-size:11.0pt;">So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls.</span></div><div class="yiv6350825511MsoNormal" style=""><span style="font-size:11.0pt;">CPU load reaches about 70% per machine.</span></div><div class="yiv6350825511MsoNormal"
style=""><span style="font-size:11.0pt;"> </span></div><div class="yiv6350825511MsoNormal" style=""><span style="font-size:11.0pt;">At this point both FS machines are virtualized, since the performance gain wasn’t that much compared to physical. </span></div><div class="yiv6350825511MsoNormal" style=""><span style="font-size:11.0pt;">The VM host shows it is using ~3/4 of its CPU resources.</span></div><div class="yiv6350825511MsoNormal" style=""><span style="font-size:11.0pt;"> </span></div><div class="yiv6350825511MsoNormal" style=""><span style="font-size:11.0pt;">Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length. </span></div><div class="yiv6350825511MsoNormal" style=""><span style="font-size:11.0pt;"> </span></div><div class="yiv6350825511MsoNormal" style=""><span style="font-size:11.0pt;">Also, FS is running on Ubuntu Server 12.04 x64.</span></div><div
class="yiv6350825511MsoNormal" style=""><span style="font-size:11.0pt;"> </span></div><div><div style="border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in;"><div class="yiv6350825511MsoNormal" style=""><b><span style="font-size:10.0pt;">From:</span></b><span style="font-size:10.0pt;"> <a rel="nofollow" shape="rect" ymailto="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank" href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a rel="nofollow" shape="rect" ymailto="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank" href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>] <b>On Behalf Of </b>Stanislav Sinyagin<br clear="none"><b>Sent:</b> Tuesday, October 22, 2013 11:11 AM<br clear="none"><b>To:</b> FreeSWITCH Users Help<br clear="none"><b>Subject:</b> Re: [Freeswitch-users]
Audio quality issues</span></div></div></div><div><div><div class="yiv6350825511MsoNormal" style=""> </div><div><div class="yiv6350825511MsoNormal" style="margin-bottom:12.0pt;background:white;"><span style="">800 calls at 64kbps is 51Mbps. <br clear="none">Could there be a network issue, like a 100Mbps line between the endpoints?<br clear="none"><br clear="none">How heavy is your CPU load? "htop" command would be helpful in this.</span></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style=""> </span></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style=""> </span></div></div><div><div><div><div class="yiv6350825511MsoNormal" style="text-align:center;background:white;" align="center"><span style=""></span><hr align="center" size="1" width="100%"></div><div class="yiv6350825511MsoNormal" style="background:white;"><b><span
style="font-size:10.0pt;">From:</span></b><span style="font-size:10.0pt;"> Grant Bagdasarian <<a rel="nofollow" shape="rect" ymailto="mailto:GB@cm.nl" target="_blank" href="mailto:GB@cm.nl">GB@cm.nl</a>><br clear="none"><b>To:</b> "FreeSWITCH Users Help (<a rel="nofollow" shape="rect" ymailto="mailto:freeswitch-users@lists.freeswitch.org" target="_blank" href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>)" <<a rel="nofollow" shape="rect" ymailto="mailto:freeswitch-users@lists.freeswitch.org" target="_blank" href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>> <br clear="none"><b>Sent:</b> Tuesday, October 22, 2013 10:14 AM<br clear="none"><b>Subject:</b> [Freeswitch-users] Audio quality issues</span></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style=""> </span></div><div><div><div><div><div
class="yiv6350825511MsoNormal" style="background:white;"><span style="">Hello,</span></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style=""> </span></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style="">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></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style="">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></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style="">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></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style="">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></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style=""> </span></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style="">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></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style=""> </span></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style="">I disabled all the modules we don’t need, like CDR’s, conferencing,
etc.</span></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style=""> </span></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style="">Are there any parameters(config files)/modules that can affect the quality of the audio stream?</span></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style=""> </span></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style="">Regards,</span></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style=""> </span></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style="">Grant</span></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span style=""> </span></div></div><div><div class="yiv6350825511MsoNormal" style="background:white;"><span
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