[Freeswitch-users] Audio quality issues

Wed Oct 23 02:44:18 MSD 2013

"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 at cm.nl>
To: FreeSWITCH Users Help <freeswitch-users at 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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at 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 at 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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at 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 at 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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at 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 at cm.nl>
To: "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)" <freeswitch-users at 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

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