Hello, 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. 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. 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. 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. 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. I disabled all the modules we don't need, like CDR's, conferencing, etc. Are there any parameters(config files)/modules that can affect the quality of the audio stream? Regards, Grant
800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints? How heavy is your CPU load? "htop" command would be helpful in this. ________________________________ From: Grant Bagdasarian <GB@cm.nl> To: "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org)" <freeswitch-users@lists.freeswitch.org> Sent: Tuesday, October 22, 2013 10:14 AM Subject: [Freeswitch-users] Audio quality issues Hello, 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. 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. 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. 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. 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. I disabled all the modules we don’t need, like CDR’s, conferencing, etc. Are there any parameters(config files)/modules that can affect the quality of the audio stream? Regards, Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
The network shouldn’t be an issue, since we have at least 1Gbps lines. The tests stay within the network. I forgot to mention the calls are being distributed across two machines by a Kamailio instance. So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls. CPU load reaches about 70% per machine. At this point both FS machines are virtualized, since the performance gain wasn’t that much compared to physical. The VM host shows it is using ~3/4 of its CPU resources. Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length. Also, FS is running on Ubuntu Server 12.04 x64. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Stanislav Sinyagin Sent: Tuesday, October 22, 2013 11:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues 800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints? How heavy is your CPU load? "htop" command would be helpful in this. ________________________________ From: Grant Bagdasarian <GB@cm.nl<mailto:GB@cm.nl>> To: "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>)" <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Sent: Tuesday, October 22, 2013 10:14 AM Subject: [Freeswitch-users] Audio quality issues Hello, 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. 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. 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. 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. 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. I disabled all the modules we don’t need, like CDR’s, conferencing, etc. Are there any parameters(config files)/modules that can affect the quality of the audio stream? Regards, Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com<http://www.cudatel.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://wiki.freeswitch.org<http://wiki.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org<http://www.freeswitch.org/>
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. On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian <GB@cm.nl> wrote:
The network shouldn’t be an issue, since we have at least 1Gbps lines. The tests stay within the network.****
** **
I forgot to mention the calls are being distributed across two machines by a Kamailio instance.****
So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls.****
CPU load reaches about 70% per machine.****
** **
At this point both FS machines are virtualized, since the performance gain wasn’t that much compared to physical. ****
The VM host shows it is using ~3/4 of its CPU resources.****
** **
Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length. ****
** **
Also, FS is running on Ubuntu Server 12.04 x64.****
** **
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Stanislav Sinyagin *Sent:* Tuesday, October 22, 2013 11:11 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Audio quality issues****
** **
800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints?
How heavy is your CPU load? "htop" command would be helpful in this.
****
** **
** ** ------------------------------
*From:* Grant Bagdasarian <GB@cm.nl> *To:* "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org)" < freeswitch-users@lists.freeswitch.org> *Sent:* Tuesday, October 22, 2013 10:14 AM *Subject:* [Freeswitch-users] Audio quality issues****
** **
Hello,****
****
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.****
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. ****
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.****
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.****
****
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. ****
****
I disabled all the modules we don’t need, like CDR’s, conferencing, etc.** **
****
Are there any parameters(config files)/modules that can affect the quality of the audio stream?****
****
Regards,****
****
Grant****
****
****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
A while back I used iotop to measure the disk access, and FS was hardly using any io during tests. How do I simulate two way audio? 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? Or is it enough for Sipp to send the media? From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ali Pey Sent: Tuesday, October 22, 2013 2:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues 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. On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian <GB@cm.nl<mailto:GB@cm.nl>> wrote: The network shouldn't be an issue, since we have at least 1Gbps lines. The tests stay within the network. I forgot to mention the calls are being distributed across two machines by a Kamailio instance. So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls. CPU load reaches about 70% per machine. At this point both FS machines are virtualized, since the performance gain wasn't that much compared to physical. The VM host shows it is using ~3/4 of its CPU resources. Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length. Also, FS is running on Ubuntu Server 12.04 x64. From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Stanislav Sinyagin Sent: Tuesday, October 22, 2013 11:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues 800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints? How heavy is your CPU load? "htop" command would be helpful in this. ________________________________ From: Grant Bagdasarian <GB@cm.nl<mailto:GB@cm.nl>> To: "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>)" <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Sent: Tuesday, October 22, 2013 10:14 AM Subject: [Freeswitch-users] Audio quality issues Hello, 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. 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. 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. 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. 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. I disabled all the modules we don't need, like CDR's, conferencing, etc. Are there any parameters(config files)/modules that can affect the quality of the audio stream? Regards, Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com<http://www.cudatel.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://wiki.freeswitch.org<http://wiki.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org [cid:~WRD000.jpg]
You can make calls from sipp that also terminates on sipp and then play a wave file in sipp. Change your dial plan in FS to route the calls to an instant of sipp that can terminate the calls. Does this make sense? On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian <GB@cm.nl> wrote:
A while back I used iotop to measure the disk access, and FS was hardly using any io during tests.****
** **
How do I simulate two way audio? ****
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?****
Or is it enough for Sipp to send the media?****
** **
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Ali Pey *Sent:* Tuesday, October 22, 2013 2:25 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Audio quality issues****
** **
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.****
** **
** **
On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian <GB@cm.nl> wrote:****
The network shouldn’t be an issue, since we have at least 1Gbps lines. The tests stay within the network.****
****
I forgot to mention the calls are being distributed across two machines by a Kamailio instance.****
So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls.****
CPU load reaches about 70% per machine.****
****
At this point both FS machines are virtualized, since the performance gain wasn’t that much compared to physical. ****
The VM host shows it is using ~3/4 of its CPU resources.****
****
Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length. ****
****
Also, FS is running on Ubuntu Server 12.04 x64.****
****
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Stanislav Sinyagin *Sent:* Tuesday, October 22, 2013 11:11 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Audio quality issues****
****
800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints?
How heavy is your CPU load? "htop" command would be helpful in this.****
****
**** ------------------------------
*From:* Grant Bagdasarian <GB@cm.nl> *To:* "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org)" < freeswitch-users@lists.freeswitch.org> *Sent:* Tuesday, October 22, 2013 10:14 AM *Subject:* [Freeswitch-users] Audio quality issues****
****
Hello,****
****
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.****
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. ****
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.****
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.****
****
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. ****
****
I disabled all the modules we don’t need, like CDR’s, conferencing, etc.** **
****
Are there any parameters(config files)/modules that can affect the quality of the audio stream?****
****
Regards,****
****
Grant****
****
****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org****
** **
[image: Image removed by sender.]****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Yes, it does! I also found this: https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo Echo application for FS. Going to see if that works first. If not, I'll setup a Sipp in server mode. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ali Pey Sent: Tuesday, October 22, 2013 3:50 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues You can make calls from sipp that also terminates on sipp and then play a wave file in sipp. Change your dial plan in FS to route the calls to an instant of sipp that can terminate the calls. Does this make sense? On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian <GB@cm.nl<mailto:GB@cm.nl>> wrote: A while back I used iotop to measure the disk access, and FS was hardly using any io during tests. How do I simulate two way audio? 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? Or is it enough for Sipp to send the media? From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Ali Pey Sent: Tuesday, October 22, 2013 2:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues 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. On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian <GB@cm.nl<mailto:GB@cm.nl>> wrote: The network shouldn't be an issue, since we have at least 1Gbps lines. The tests stay within the network. I forgot to mention the calls are being distributed across two machines by a Kamailio instance. So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls. CPU load reaches about 70% per machine. At this point both FS machines are virtualized, since the performance gain wasn't that much compared to physical. The VM host shows it is using ~3/4 of its CPU resources. Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length. Also, FS is running on Ubuntu Server 12.04 x64. From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Stanislav Sinyagin Sent: Tuesday, October 22, 2013 11:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues 800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints? How heavy is your CPU load? "htop" command would be helpful in this. ________________________________ From: Grant Bagdasarian <GB@cm.nl<mailto:GB@cm.nl>> To: "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>)" <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Sent: Tuesday, October 22, 2013 10:14 AM Subject: [Freeswitch-users] Audio quality issues Hello, 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. 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. 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. 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. 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. I disabled all the modules we don't need, like CDR's, conferencing, etc. Are there any parameters(config files)/modules that can affect the quality of the audio stream? Regards, Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com<http://www.cudatel.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://wiki.freeswitch.org<http://wiki.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org [cid:~WRD000.jpg] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org [cid:~WRD000.jpg]
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 :) Please do post your results here. These would be some valuable info for the community. On Tue, Oct 22, 2013 at 10:03 AM, Grant Bagdasarian <GB@cm.nl> wrote:
Yes, it does! ****
** **
I also found this: https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo****
** **
Echo application for FS. ****
Going to see if that works first. If not, I’ll setup a Sipp in server mode. ****
** **
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Ali Pey *Sent:* Tuesday, October 22, 2013 3:50 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Audio quality issues****
** **
You can make calls from sipp that also terminates on sipp and then play a wave file in sipp.****
** **
Change your dial plan in FS to route the calls to an instant of sipp that can terminate the calls.****
** **
Does this make sense?****
** **
On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian <GB@cm.nl> wrote:****
A while back I used iotop to measure the disk access, and FS was hardly using any io during tests.****
****
How do I simulate two way audio? ****
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?****
Or is it enough for Sipp to send the media?****
****
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Ali Pey *Sent:* Tuesday, October 22, 2013 2:25 PM****
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Audio quality issues****
****
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.****
****
****
On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian <GB@cm.nl> wrote:****
The network shouldn’t be an issue, since we have at least 1Gbps lines. The tests stay within the network.****
****
I forgot to mention the calls are being distributed across two machines by a Kamailio instance.****
So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls.****
CPU load reaches about 70% per machine.****
****
At this point both FS machines are virtualized, since the performance gain wasn’t that much compared to physical. ****
The VM host shows it is using ~3/4 of its CPU resources.****
****
Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length. ****
****
Also, FS is running on Ubuntu Server 12.04 x64.****
****
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Stanislav Sinyagin *Sent:* Tuesday, October 22, 2013 11:11 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Audio quality issues****
****
800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints?
How heavy is your CPU load? "htop" command would be helpful in this.****
****
**** ------------------------------
*From:* Grant Bagdasarian <GB@cm.nl> *To:* "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org)" < freeswitch-users@lists.freeswitch.org> *Sent:* Tuesday, October 22, 2013 10:14 AM *Subject:* [Freeswitch-users] Audio quality issues****
****
Hello,****
****
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.****
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. ****
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.****
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.****
****
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. ****
****
I disabled all the modules we don’t need, like CDR’s, conferencing, etc.** **
****
Are there any parameters(config files)/modules that can affect the quality of the audio stream?****
****
Regards,****
****
Grant****
****
****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org****
****
[image: Image removed by sender.]****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org****
** **
[image: Image removed by sender.]****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Putting the sound file on a ramdisk usually counts out disk IO. You've made a mistake somewhere that you should look for if you saw no gain from using physical machines over virtuals. I'm always squeamish about helping on load test questions because we tend to get caught up in it and eventually use up a lot of time just steering people into things that fall outside the scope of the project. Baseline: I would try one of your servers on a modern kernel (Debian 7 or equiv) using latest HEAD build from master or stable branch and put your test extension high in your dialplan to avoid the extra stuff that goes on in the demo pbx config. Also disable presence on sip with manage-presence and manage-shared-appearance both commented out or set to false in the sofia profile. On Tue, Oct 22, 2013 at 10:28 AM, Ali Pey <alipey@gmail.com> wrote:
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 :)
Please do post your results here. These would be some valuable info for the community.
On Tue, Oct 22, 2013 at 10:03 AM, Grant Bagdasarian <GB@cm.nl> wrote:
Yes, it does! ****
** **
I also found this: https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo****
** **
Echo application for FS. ****
Going to see if that works first. If not, I’ll setup a Sipp in server mode.****
** **
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Ali Pey *Sent:* Tuesday, October 22, 2013 3:50 PM
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Audio quality issues****
** **
You can make calls from sipp that also terminates on sipp and then play a wave file in sipp.****
** **
Change your dial plan in FS to route the calls to an instant of sipp that can terminate the calls.****
** **
Does this make sense?****
** **
On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian <GB@cm.nl> wrote:****
A while back I used iotop to measure the disk access, and FS was hardly using any io during tests.****
****
How do I simulate two way audio? ****
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?****
Or is it enough for Sipp to send the media?****
****
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Ali Pey *Sent:* Tuesday, October 22, 2013 2:25 PM****
*To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Audio quality issues****
****
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.****
****
****
On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian <GB@cm.nl> wrote:****
The network shouldn’t be an issue, since we have at least 1Gbps lines. The tests stay within the network.****
****
I forgot to mention the calls are being distributed across two machines by a Kamailio instance.****
So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls.****
CPU load reaches about 70% per machine.****
****
At this point both FS machines are virtualized, since the performance gain wasn’t that much compared to physical. ****
The VM host shows it is using ~3/4 of its CPU resources.****
****
Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length. ****
****
Also, FS is running on Ubuntu Server 12.04 x64.****
****
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Stanislav Sinyagin *Sent:* Tuesday, October 22, 2013 11:11 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Audio quality issues****
****
800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints?
How heavy is your CPU load? "htop" command would be helpful in this.****
****
**** ------------------------------
*From:* Grant Bagdasarian <GB@cm.nl> *To:* "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org)" < freeswitch-users@lists.freeswitch.org> *Sent:* Tuesday, October 22, 2013 10:14 AM *Subject:* [Freeswitch-users] Audio quality issues****
****
Hello,****
****
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.****
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. ****
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. ****
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.****
****
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. ****
****
I disabled all the modules we don’t need, like CDR’s, conferencing, etc.* ***
****
Are there any parameters(config files)/modules that can affect the quality of the audio stream?****
****
Regards,****
****
Grant****
****
****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org****
****
[image: Image removed by sender.]****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org****
** **
[image: Image removed by sender.]****
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
Hi, I would be also interested in this as i am experiancing same issue, poor audio quality and media dropping. Br, Miha On Tue, 22 Oct 2013 16:03:29 +0200 Grant Bagdasarian <GB@cm.nl> wrote:
Yes, it does!
I also found this:
https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo
Echo application for FS. Going to see if that works first. If not, I'll setup a Sipp in server mode.
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ali Pey Sent: Tuesday, October 22, 2013 3:50 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues
You can make calls from sipp that also terminates on sipp and then play a wave file in sipp.
Change your dial plan in FS to route the calls to an instant of sipp that can terminate the calls.
Does this make sense?
On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian <GB@cm.nl<mailto:GB@cm.nl>> wrote: A while back I used iotop to measure the disk access, and FS was hardly using any io during tests.
How do I simulate two way audio? 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? Or is it enough for Sipp to send the media?
From:
freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>
[mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>]
On Behalf Of Ali Pey Sent: Tuesday, October 22, 2013 2:25 PM
To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues
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.
On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian <GB@cm.nl<mailto:GB@cm.nl>> wrote: The network shouldn't be an issue, since we have at least 1Gbps lines. The tests stay within the network.
I forgot to mention the calls are being distributed across two machines by a Kamailio instance. So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls. CPU load reaches about 70% per machine.
At this point both FS machines are virtualized, since the performance gain wasn't that much compared to physical. The VM host shows it is using ~3/4 of its CPU resources.
Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length.
Also, FS is running on Ubuntu Server 12.04 x64.
From:
freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>
[mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>]
On Behalf Of Stanislav Sinyagin Sent: Tuesday, October 22, 2013 11:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues
800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints?
How heavy is your CPU load? "htop" command would be helpful in this.
________________________________ From: Grant Bagdasarian <GB@cm.nl<mailto:GB@cm.nl>> To: "FreeSWITCH Users Help
(freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>)"
<freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>>
Sent: Tuesday, October 22, 2013 10:14 AM Subject: [Freeswitch-users] Audio quality issues
Hello,
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. 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. 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. 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.
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.
I disabled all the modules we don't need, like CDR's, conferencing, etc.
Are there any parameters(config files)/modules that can affect the quality of the audio stream?
Regards,
Grant
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org<mailto:consulting@freeswitch.org>
http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com<http://www.cudatel.com/>
Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://wiki.freeswitch.org<http://wiki.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/>
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org<mailto:consulting@freeswitch.org>
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[cid:~WRD000.jpg]
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org<mailto:consulting@freeswitch.org>
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[cid:~WRD000.jpg]
My 2cents is to move to bare metal. We ran a Guest on Commercial Licensed ESX and without real load (>50%) started seeing bad behavior. We moved to Bare Metal just over 3weeks ago and since then no more strange behavior. I've setup OpenVZ for LAB environment and so far no issues with almost the same load when using Commercial Licensed ESX. Kind Regards Sent from my iPhone
On Oct 22, 2013, at 19:28, "Miha" <miha@softnet.si> wrote:
Hi,
I would be also interested in this as i am experiancing same issue, poor audio quality and media dropping.
Br,
Miha
On Tue, 22 Oct 2013 16:03:29 +0200 Grant Bagdasarian <GB@cm.nl> wrote:
Yes, it does!
I also found this: https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo
Echo application for FS. Going to see if that works first. If not, I'll setup a Sipp in server mode.
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ali Pey Sent: Tuesday, October 22, 2013 3:50 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues
You can make calls from sipp that also terminates on sipp and then play a wave file in sipp.
Change your dial plan in FS to route the calls to an instant of sipp that can terminate the calls.
Does this make sense?
On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian <GB@cm.nl<mailto:GB@cm.nl>> wrote: A while back I used iotop to measure the disk access, and FS was hardly using any io during tests.
How do I simulate two way audio? 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? Or is it enough for Sipp to send the media?
From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Ali Pey Sent: Tuesday, October 22, 2013 2:25 PM
To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues
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.
On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian <GB@cm.nl<mailto:GB@cm.nl>> wrote: The network shouldn't be an issue, since we have at least 1Gbps lines. The tests stay within the network.
I forgot to mention the calls are being distributed across two machines by a Kamailio instance. So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls. CPU load reaches about 70% per machine.
At this point both FS machines are virtualized, since the performance gain wasn't that much compared to physical. The VM host shows it is using ~3/4 of its CPU resources.
Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length.
Also, FS is running on Ubuntu Server 12.04 x64.
From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Stanislav Sinyagin Sent: Tuesday, October 22, 2013 11:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues
800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints?
How heavy is your CPU load? "htop" command would be helpful in this.
________________________________ From: Grant Bagdasarian <GB@cm.nl<mailto:GB@cm.nl>> To: "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>)" <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Sent: Tuesday, October 22, 2013 10:14 AM Subject: [Freeswitch-users] Audio quality issues
Hello,
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. 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. 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. 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.
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.
I disabled all the modules we don't need, like CDR's, conferencing, etc.
Are there any parameters(config files)/modules that can affect the quality of the audio stream?
Regards,
Grant
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com<http://www.cudatel.com/>
Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://wiki.freeswitch.org<http://wiki.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org<http://www.freeswitch.org/>
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[cid:~WRD000.jpg]
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[cid:~WRD000.jpg]
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
You have turned off CDR, but what about logging? Logging at certain levels gets pretty taxing at these call rates. Have you implemented all the hints from https://wiki.freeswitch.org/wiki/Performance_Tuning and https://wiki.freeswitch.org/wiki/Performance_testing_and_configurations ? Guillermo On Tue, Oct 22, 2013 at 3:41 PM, Deon Vermeulen <vermeulen.deon@gmail.com>wrote:
My 2cents is to move to bare metal. We ran a Guest on Commercial Licensed ESX and without real load (>50%) started seeing bad behavior. We moved to Bare Metal just over 3weeks ago and since then no more strange behavior.
I've setup OpenVZ for LAB environment and so far no issues with almost the same load when using Commercial Licensed ESX.
Kind Regards
Sent from my iPhone
On Oct 22, 2013, at 19:28, "Miha" <miha@softnet.si> wrote:
Hi,
I would be also interested in this as i am experiancing same issue, poor audio quality and media dropping.
Br,
Miha
On Tue, 22 Oct 2013 16:03:29 +0200 Grant Bagdasarian <GB@cm.nl> wrote:
Yes, it does!
I also found this: https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo
Echo application for FS. Going to see if that works first. If not, I'll setup a Sipp in server mode.
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ali Pey Sent: Tuesday, October 22, 2013 3:50 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues
You can make calls from sipp that also terminates on sipp and then play a wave file in sipp.
Change your dial plan in FS to route the calls to an instant of sipp that can terminate the calls.
Does this make sense?
On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian <GB@cm.nl<mailto:GB@cm.nl>> wrote: A while back I used iotop to measure the disk access, and FS was hardly using any io during tests.
How do I simulate two way audio? 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? Or is it enough for Sipp to send the media?
From: freeswitch-users-bounces@lists.freeswitch.org<mailto: freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto: freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Ali Pey Sent: Tuesday, October 22, 2013 2:25 PM
To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues
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.
On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian <GB@cm.nl<mailto:GB@cm.nl>> wrote: The network shouldn't be an issue, since we have at least 1Gbps lines. The tests stay within the network.
I forgot to mention the calls are being distributed across two machines by a Kamailio instance. So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls. CPU load reaches about 70% per machine.
At this point both FS machines are virtualized, since the performance gain wasn't that much compared to physical. The VM host shows it is using ~3/4 of its CPU resources.
Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length.
Also, FS is running on Ubuntu Server 12.04 x64.
From: freeswitch-users-bounces@lists.freeswitch.org<mailto: freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto: freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Stanislav Sinyagin Sent: Tuesday, October 22, 2013 11:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues
800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints?
How heavy is your CPU load? "htop" command would be helpful in this.
________________________________ From: Grant Bagdasarian <GB@cm.nl<mailto:GB@cm.nl>> To: "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org<mailto: freeswitch-users@lists.freeswitch.org>)" <freeswitch-users@lists.freeswitch.org<mailto: freeswitch-users@lists.freeswitch.org>> Sent: Tuesday, October 22, 2013 10:14 AM Subject: [Freeswitch-users] Audio quality issues
Hello,
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. 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. 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. 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.
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.
I disabled all the modules we don't need, like CDR's, conferencing, etc.
Are there any parameters(config files)/modules that can affect the quality of the audio stream?
Regards,
Grant
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com<http://www.cudatel.com/>
Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://wiki.freeswitch.org<http://wiki.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org<http://www.freeswitch.org/>
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[cid:~WRD000.jpg]
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[cid:~WRD000.jpg]
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Guillermo Ruiz Camauer
"delay_echo" is even more convenient for testing: then you can clearly hear your own voice without distortions from echo cancellation. ________________________________ From: Grant Bagdasarian <GB@cm.nl> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Tuesday, October 22, 2013 4:03 PM Subject: Re: [Freeswitch-users] Audio quality issues Yes, it does! I also found this: https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_echo Echo application for FS. Going to see if that works first. If not, I’ll setup a Sipp in server mode. From:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ali Pey Sent: Tuesday, October 22, 2013 3:50 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues You can make calls from sipp that also terminates on sipp and then play a wave file in sipp. Change your dial plan in FS to route the calls to an instant of sipp that can terminate the calls. Does this make sense? On Tue, Oct 22, 2013 at 9:22 AM, Grant Bagdasarian <GB@cm.nl> wrote: A while back I used iotop to measure the disk access, and FS was hardly using any io during tests. How do I simulate two way audio? 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? Or is it enough for Sipp to send the media? From:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ali Pey Sent: Tuesday, October 22, 2013 2:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues 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. On Tue, Oct 22, 2013 at 6:01 AM, Grant Bagdasarian <GB@cm.nl> wrote: The network shouldn’t be an issue, since we have at least 1Gbps lines. The tests stay within the network. I forgot to mention the calls are being distributed across two machines by a Kamailio instance. So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls. CPU load reaches about 70% per machine. At this point both FS machines are virtualized, since the performance gain wasn’t that much compared to physical. The VM host shows it is using ~3/4 of its CPU resources. Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length. Also, FS is running on Ubuntu Server 12.04 x64. From:freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Stanislav Sinyagin Sent: Tuesday, October 22, 2013 11:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues 800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints? How heavy is your CPU load? "htop" command would be helpful in this. ________________________________ From:Grant Bagdasarian <GB@cm.nl> To: "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org)" <freeswitch-users@lists.freeswitch.org> Sent: Tuesday, October 22, 2013 10:14 AM Subject: [Freeswitch-users] Audio quality issues Hello, 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. 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. 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. 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. 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. I disabled all the modules we don’t need, like CDR’s, conferencing, etc. Are there any parameters(config files)/modules that can affect the quality of the audio stream? Regards, Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I wouldn't be shocked if the virtual nic's are bottlenecking on pps. I've seen this before. Also possible you have a crap physical nic. What kind of nic is it and what virtualization technology? On Oct 22, 2013, at 6:01 AM, Grant Bagdasarian <GB@cm.nl> wrote:
The network shouldn’t be an issue, since we have at least 1Gbps lines. The tests stay within the network.
I forgot to mention the calls are being distributed across two machines by a Kamailio instance. So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls. CPU load reaches about 70% per machine.
At this point both FS machines are virtualized, since the performance gain wasn’t that much compared to physical. The VM host shows it is using ~3/4 of its CPU resources.
Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length.
Also, FS is running on Ubuntu Server 12.04 x64.
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Stanislav Sinyagin Sent: Tuesday, October 22, 2013 11:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues
800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints?
How heavy is your CPU load? "htop" command would be helpful in this.
From: Grant Bagdasarian <GB@cm.nl> To: "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org)" <freeswitch-users@lists.freeswitch.org> Sent: Tuesday, October 22, 2013 10:14 AM Subject: [Freeswitch-users] Audio quality issues
Hello,
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. 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. 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. 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.
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.
I disabled all the modules we don’t need, like CDR’s, conferencing, etc.
Are there any parameters(config files)/modules that can affect the quality of the audio stream?
Regards,
Grant
We're using VMWare ESXi 5.1 (Free Version). NIC version: Intel I350 Gigabit Network. From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Tuesday, October 22, 2013 3:28 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues I wouldn't be shocked if the virtual nic's are bottlenecking on pps. I've seen this before. Also possible you have a crap physical nic. What kind of nic is it and what virtualization technology? On Oct 22, 2013, at 6:01 AM, Grant Bagdasarian <GB@cm.nl<mailto:GB@cm.nl>> wrote: The network shouldn't be an issue, since we have at least 1Gbps lines. The tests stay within the network. I forgot to mention the calls are being distributed across two machines by a Kamailio instance. So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls. CPU load reaches about 70% per machine. At this point both FS machines are virtualized, since the performance gain wasn't that much compared to physical. The VM host shows it is using ~3/4 of its CPU resources. Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length. Also, FS is running on Ubuntu Server 12.04 x64. From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:users-bounces@lists.freeswitch.org>] On Behalf Of Stanislav Sinyagin Sent: Tuesday, October 22, 2013 11:11 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Audio quality issues 800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints? How heavy is your CPU load? "htop" command would be helpful in this. ________________________________ From: Grant Bagdasarian <GB@cm.nl<mailto:GB@cm.nl>> To: "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>)" <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Sent: Tuesday, October 22, 2013 10:14 AM Subject: [Freeswitch-users] Audio quality issues Hello, 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. 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. 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. 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. 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. I disabled all the modules we don't need, like CDR's, conferencing, etc. Are there any parameters(config files)/modules that can affect the quality of the audio stream? Regards, Grant
Hi Grant, Two possibilities spring to mind: - If your audio is coming from a disk, can that disk keep up? - How good is your ethernet switch? Notice in the second point I said how good, not how expensive. Many switches choke on a large number of small media packets, including some expensive big name products. Regards, Steve On 10/22/2013 06:01 PM, Grant Bagdasarian wrote:
The network shouldn’t be an issue, since we have at least 1Gbps lines. The tests stay within the network.
I forgot to mention the calls are being distributed across two machines by a Kamailio instance.
So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls.
CPU load reaches about 70% per machine.
At this point both FS machines are virtualized, since the performance gain wasn’t that much compared to physical.
The VM host shows it is using ~3/4 of its CPU resources.
Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length.
Also, FS is running on Ubuntu Server 12.04 x64.
*From:*freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Stanislav Sinyagin *Sent:* Tuesday, October 22, 2013 11:11 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Audio quality issues
800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints?
How heavy is your CPU load? "htop" command would be helpful in this.
------------------------------------------------------------------------
*From:*Grant Bagdasarian <GB@cm.nl <mailto:GB@cm.nl>> *To:* "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>)" <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> *Sent:* Tuesday, October 22, 2013 10:14 AM *Subject:* [Freeswitch-users] Audio quality issues
Hello,
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.
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.
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.
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.
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.
I disabled all the modules we don’t need, like CDR’s, conferencing, etc.
Are there any parameters(config files)/modules that can affect the quality of the audio stream?
Regards,
Grant
Hi Steve, do you have any good preposal which switch has good performance for voip trafic? miha Dne 10/22/2013 3:32 PM, piše Steve Underwood:
Hi Grant,
Two possibilities spring to mind:
- If your audio is coming from a disk, can that disk keep up? - How good is your ethernet switch?
Notice in the second point I said how good, not how expensive. Many switches choke on a large number of small media packets, including some expensive big name products.
Regards, Steve
On 10/22/2013 06:01 PM, Grant Bagdasarian wrote:
The network shouldn’t be an issue, since we have at least 1Gbps lines. The tests stay within the network.
I forgot to mention the calls are being distributed across two machines by a Kamailio instance.
So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls.
CPU load reaches about 70% per machine.
At this point both FS machines are virtualized, since the performance gain wasn’t that much compared to physical.
The VM host shows it is using ~3/4 of its CPU resources.
Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length.
Also, FS is running on Ubuntu Server 12.04 x64.
*From:*freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Stanislav Sinyagin *Sent:* Tuesday, October 22, 2013 11:11 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Audio quality issues
800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints?
How heavy is your CPU load? "htop" command would be helpful in this.
------------------------------------------------------------------------
*From:*Grant Bagdasarian <GB@cm.nl <mailto:GB@cm.nl>> *To:* "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>)" <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> *Sent:* Tuesday, October 22, 2013 10:14 AM *Subject:* [Freeswitch-users] Audio quality issues
Hello,
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.
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.
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.
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.
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.
I disabled all the modules we don’t need, like CDR’s, conferencing, etc.
Are there any parameters(config files)/modules that can affect the quality of the audio stream?
Regards,
Grant
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Here is some good data on per call real bandwidth consumption for different codecs: Codec BR NEB G.711 64 Kbps 87.2 Kbps G.729 8 Kbps 31.2 Kbps G.723.1 6.4 Kbps 21.9 Kbps G.723.1 5.3 Kbps 20.8 Kbps G.726 32 Kbps 55.2 Kbps G.726 24 Kbps 47.2 Kbps G.728 16 Kbps 31.5 Kbps iLBC 15 Kbps 27.7 Kbps BR = Bit rate NEB = Nominal Ethernet Bandwidth (one direction) On Wed, Oct 23, 2013 at 5:33 AM, Miha <miha@softnet.si> wrote:
Hi Steve,
do you have any good preposal which switch has good performance for voip trafic?
miha
Dne 10/22/2013 3:32 PM, piše Steve Underwood:
Hi Grant,
Two possibilities spring to mind:
- If your audio is coming from a disk, can that disk keep up? - How good is your ethernet switch?
Notice in the second point I said how good, not how expensive. Many switches choke on a large number of small media packets, including some expensive big name products.
Regards, Steve
On 10/22/2013 06:01 PM, Grant Bagdasarian wrote:
The network shouldn’t be an issue, since we have at least 1Gbps lines. The tests stay within the network.
I forgot to mention the calls are being distributed across two machines by a Kamailio instance.
So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls.
CPU load reaches about 70% per machine.
At this point both FS machines are virtualized, since the performance gain wasn’t that much compared to physical.
The VM host shows it is using ~3/4 of its CPU resources.
Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length.
Also, FS is running on Ubuntu Server 12.04 x64.
*From:*freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Stanislav Sinyagin *Sent:* Tuesday, October 22, 2013 11:11 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Audio quality issues
800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints?
How heavy is your CPU load? "htop" command would be helpful in this.
------------------------------------------------------------------------
*From:*Grant Bagdasarian <GB@cm.nl <mailto:GB@cm.nl>> *To:* "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>)" <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> *Sent:* Tuesday, October 22, 2013 10:14 AM *Subject:* [Freeswitch-users] Audio quality issues
Hello,
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.
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.
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.
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.
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.
I disabled all the modules we don’t need, like CDR’s, conferencing, etc.
Are there any parameters(config files)/modules that can affect the quality of the audio stream?
Regards,
Grant
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Guillermo Ruiz Camauer
And a link with a great explanation of how to arrive at some of those numbers: http://voiceonbits.com/2010/08/15/codec-bandwidth-calculation-g711g729/ Guillermo On Wed, Oct 23, 2013 at 4:28 PM, Guillermo Ruiz Camauer <grcamauer@gmail.com
wrote:
Here is some good data on per call real bandwidth consumption for different codecs:
Codec BR NEB G.711 64 Kbps 87.2 Kbps G.729 8 Kbps 31.2 Kbps G.723.1 6.4 Kbps 21.9 Kbps G.723.1 5.3 Kbps 20.8 Kbps G.726 32 Kbps 55.2 Kbps G.726 24 Kbps 47.2 Kbps G.728 16 Kbps 31.5 Kbps iLBC 15 Kbps 27.7 Kbps
BR = Bit rate NEB = Nominal Ethernet Bandwidth (one direction)
On Wed, Oct 23, 2013 at 5:33 AM, Miha <miha@softnet.si> wrote:
Hi Steve,
do you have any good preposal which switch has good performance for voip trafic?
miha
Dne 10/22/2013 3:32 PM, piše Steve Underwood:
Hi Grant,
Two possibilities spring to mind:
- If your audio is coming from a disk, can that disk keep up? - How good is your ethernet switch?
Notice in the second point I said how good, not how expensive. Many switches choke on a large number of small media packets, including some expensive big name products.
Regards, Steve
On 10/22/2013 06:01 PM, Grant Bagdasarian wrote:
The network shouldn’t be an issue, since we have at least 1Gbps lines. The tests stay within the network.
I forgot to mention the calls are being distributed across two machines by a Kamailio instance.
So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls.
CPU load reaches about 70% per machine.
At this point both FS machines are virtualized, since the performance gain wasn’t that much compared to physical.
The VM host shows it is using ~3/4 of its CPU resources.
Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length.
Also, FS is running on Ubuntu Server 12.04 x64.
*From:*freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Stanislav Sinyagin *Sent:* Tuesday, October 22, 2013 11:11 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Audio quality issues
800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints?
How heavy is your CPU load? "htop" command would be helpful in this.
*From:*Grant Bagdasarian <GB@cm.nl <mailto:GB@cm.nl>> *To:* "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>)" <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> *Sent:* Tuesday, October 22, 2013 10:14 AM *Subject:* [Freeswitch-users] Audio quality issues
Hello,
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.
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.
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.
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.
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.
I disabled all the modules we don’t need, like CDR’s, conferencing,
etc.
Are there any parameters(config files)/modules that can affect the quality of the audio stream?
Regards,
Grant
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Guillermo Ruiz Camauer
-- Guillermo Ruiz Camauer
Your network may be rated for 1 gbps, but that may assume packets at or near the MTU size, and that MTU used in the rating might be around 9000 bytes. RTP packets are typically around 200 or less bytes each, sent every ptime (default is 20ms). You may saturate your network infrastructure long before you saturate your FreeSWITCH. -Yossi On 10/22/2013 05:01 AM, Grant Bagdasarian wrote:
The network shouldn't be an issue, since we have at least 1Gbps lines. The tests stay within the network.
I forgot to mention the calls are being distributed across two machines by a Kamailio instance.
So for a total of 800 concurrent calls generated by Sipp, each machine has 400 active calls.
CPU load reaches about 70% per machine.
At this point both FS machines are virtualized, since the performance gain wasn't that much compared to physical.
The VM host shows it is using ~3/4 of its CPU resources.
Htop shows that the normal priority threads(green) and the kernel threads(red) are about the same length.
Also, FS is running on Ubuntu Server 12.04 x64.
*From:*freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Stanislav Sinyagin *Sent:* Tuesday, October 22, 2013 11:11 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Audio quality issues
800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints?
How heavy is your CPU load? "htop" command would be helpful in this.
------------------------------------------------------------------------
*From:*Grant Bagdasarian <GB@cm.nl <mailto:GB@cm.nl>> *To:* "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>)" <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> *Sent:* Tuesday, October 22, 2013 10:14 AM *Subject:* [Freeswitch-users] Audio quality issues
Hello,
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.
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.
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.
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.
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.
I disabled all the modules we don't need, like CDR's, conferencing, etc.
Are there any parameters(config files)/modules that can affect the quality of the audio stream?
Regards,
Grant
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://wiki.freeswitch.org <http://wiki.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org <http://www.freeswitch.org/>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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. Donny ---------- Forwarded message ---------- From: "Stanislav Sinyagin" <ssinyagin@yahoo.com> Date: Oct 22, 2013 4:12 PM Subject: Re: [Freeswitch-users] Audio quality issues To: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org> Cc: 800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints? How heavy is your CPU load? "htop" command would be helpful in this. ------------------------------ *From:* Grant Bagdasarian <GB@cm.nl> *To:* "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org)" < freeswitch-users@lists.freeswitch.org> *Sent:* Tuesday, October 22, 2013 10:14 AM *Subject:* [Freeswitch-users] Audio quality issues Hello, 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. 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. 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. 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. 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. I disabled all the modules we don’t need, like CDR’s, conferencing, etc. Are there any parameters(config files)/modules that can affect the quality of the audio stream? Regards, Grant _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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..... On Tue, Oct 22, 2013 at 5:58 PM, Donny Hardyanto <hardyanto.donny@gmail.com>wrote:
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.
Donny
---------- Forwarded message ---------- From: "Stanislav Sinyagin" <ssinyagin@yahoo.com> Date: Oct 22, 2013 4:12 PM Subject: Re: [Freeswitch-users] Audio quality issues To: "FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org> Cc:
800 calls at 64kbps is 51Mbps. Could there be a network issue, like a 100Mbps line between the endpoints?
How heavy is your CPU load? "htop" command would be helpful in this.
------------------------------ *From:* Grant Bagdasarian <GB@cm.nl> *To:* "FreeSWITCH Users Help (freeswitch-users@lists.freeswitch.org)" < freeswitch-users@lists.freeswitch.org> *Sent:* Tuesday, October 22, 2013 10:14 AM *Subject:* [Freeswitch-users] Audio quality issues
Hello,
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. 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. 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. 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.
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.
I disabled all the modules we don’t need, like CDR’s, conferencing, etc.
Are there any parameters(config files)/modules that can affect the quality of the audio stream?
Regards,
Grant
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
participants (11)
-
Ali Pey
-
Anthony Minessale
-
Deon Vermeulen
-
Donny Hardyanto
-
Grant Bagdasarian
-
Guillermo Ruiz Camauer
-
I put the Who? in Mishehu
-
Michael Jerris
-
Miha
-
Stanislav Sinyagin
-
Steve Underwood