Audio quality issues

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Audio quality issues

Grant Bagdasarian

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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Re: Audio quality issues

Stanislav Sinyagin
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 <[hidden email]>
To: "FreeSWITCH Users Help ([hidden email])" <[hidden email]>
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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Re: Audio quality issues

Grant Bagdasarian

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: [hidden email] [mailto:[hidden email]] 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 <[hidden email]>
To: "FreeSWITCH Users Help ([hidden email])" <[hidden email]>
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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Re: Audio quality issues

Ali Pey
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 <[hidden email]> 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: [hidden email] [mailto:[hidden email]] 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 <[hidden email]>
To: "FreeSWITCH Users Help ([hidden email])" <[hidden email]>
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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Re: Audio quality issues

Grant Bagdasarian

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: [hidden email] [mailto:[hidden email]] 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 <[hidden email]> 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: [hidden email] [mailto:[hidden email]] 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 <[hidden email]>
To: "FreeSWITCH Users Help ([hidden email])" <[hidden email]>
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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Re: Audio quality issues

Michael Jerris
Administrator
In reply to this post by Grant Bagdasarian
<base href="x-msg://55704/">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 <[hidden email]> 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: [hidden email] [mailto:freeswitch-[hidden email]] 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 <[hidden email]>
To: "FreeSWITCH Users Help ([hidden email])" <[hidden email]> 
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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Re: Audio quality issues

Steve Underwood
In reply to this post by Grant Bagdasarian
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:*[hidden email]
> [mailto:[hidden email]] *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 <[hidden email] <mailto:[hidden email]>>
> *To:* "FreeSWITCH Users Help ([hidden email]
> <mailto:[hidden email]>)"
> <[hidden email]
> <mailto:[hidden email]>>
> *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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Re: Audio quality issues

Grant Bagdasarian
In reply to this post by Michael Jerris
<base href="x-msg://55704/">

We’re using VMWare ESXi 5.1 (Free Version).

 

NIC version: Intel I350 Gigabit Network.

 

From: [hidden email] [mailto:[hidden email]] 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 <[hidden email]> 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: [hidden email] [mailto:freeswitch-[hidden email]] 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 <[hidden email]>
To: "FreeSWITCH Users Help ([hidden email])" <[hidden email]> 
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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Re: Audio quality issues

Ali Pey
In reply to this post by Grant Bagdasarian
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 <[hidden email]> 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: [hidden email] [mailto:[hidden email]] 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 <[hidden email]> 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: [hidden email] [mailto:[hidden email]] 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 <[hidden email]>
To: "FreeSWITCH Users Help ([hidden email])" <[hidden email]>
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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FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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http://www.cluecon.com

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Re: Audio quality issues

Grant Bagdasarian

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: [hidden email] [mailto:[hidden email]] 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 <[hidden email]> 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: [hidden email] [mailto:[hidden email]] 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 <[hidden email]> 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: [hidden email] [mailto:[hidden email]] 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 <[hidden email]>
To: "FreeSWITCH Users Help ([hidden email])" <[hidden email]>
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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[hidden email]
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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http://wiki.freeswitch.org
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Official FreeSWITCH Sites
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http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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Official FreeSWITCH Sites
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http://wiki.freeswitch.org
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http://wiki.freeswitch.org
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Re: Audio quality issues

Ali Pey
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 <[hidden email]> 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: [hidden email] [mailto:[hidden email]] 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 <[hidden email]> 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: [hidden email] [mailto:[hidden email]] 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 <[hidden email]> 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: [hidden email] [mailto:[hidden email]] 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 <[hidden email]>
To: "FreeSWITCH Users Help ([hidden email])" <[hidden email]>
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:
[hidden email]
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
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_________________________________________________________________________
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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Official FreeSWITCH Sites
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http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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http://wiki.freeswitch.org
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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Official FreeSWITCH Sites
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http://wiki.freeswitch.org
http://www.cluecon.com

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Re: Audio quality issues

Anthony Minessale
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 <[hidden email]> 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 <[hidden email]> 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: [hidden email] [mailto:[hidden email]] 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 <[hidden email]> 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: [hidden email] [mailto:[hidden email]] 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 <[hidden email]> 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: [hidden email] [mailto:[hidden email]] 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 <[hidden email]>
To: "FreeSWITCH Users Help ([hidden email])" <[hidden email]>
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:
[hidden email]
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
[hidden email]
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:
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Re: Audio quality issues

Miha Zoubek
In reply to this post by Grant Bagdasarian
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 <[hidden email]> 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: [hidden email]
> [mailto:[hidden email]] 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
> <[hidden email]<mailto:[hidden email]>> 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:
>
[hidden email]<mailto:[hidden email]>
>
[mailto:[hidden email]<mailto:[hidden email]>]

> 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
> <[hidden email]<mailto:[hidden email]>> 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:
>
[hidden email]<mailto:[hidden email]>
>
[mailto:[hidden email]<mailto:[hidden email]>]

> 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 <[hidden email]<mailto:[hidden email]>>
> To: "FreeSWITCH Users Help
>
([hidden email]<mailto:[hidden email]>)"
>
<[hidden email]<mailto:[hidden email]>>

> 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:
>
[hidden email]<mailto:[hidden email]>
>
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
>
[hidden email]<mailto:[hidden email]>
>
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:
>
[hidden email]<mailto:[hidden email]>

> 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
>
[hidden email]<mailto:[hidden email]>
>
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:
>
[hidden email]<mailto:[hidden email]>

> 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
>
[hidden email]<mailto:[hidden email]>
>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
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> http://www.freeswitch.org
>
> [cid:~WRD000.jpg]


_________________________________________________________________________
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[hidden email]
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
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http://wiki.freeswitch.org
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Re: Audio quality issues

Deon Vermeulen
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" <[hidden email]> 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 <[hidden email]> 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: [hidden email]
>> [mailto:[hidden email]] 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
>> <[hidden email]<mailto:[hidden email]>> 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:
> [hidden email]<mailto:[hidden email]>
> [mailto:[hidden email]<mailto:[hidden email]>]
>> 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
>> <[hidden email]<mailto:[hidden email]>> 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:
> [hidden email]<mailto:[hidden email]>
> [mailto:[hidden email]<mailto:[hidden email]>]
>> 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 <[hidden email]<mailto:[hidden email]>>
>> To: "FreeSWITCH Users Help
> ([hidden email]<mailto:[hidden email]>)"
> <[hidden email]<mailto:[hidden email]>>
>> 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:
> [hidden email]<mailto:[hidden email]>
> 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
> [hidden email]<mailto:[hidden email]>
> 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:
> [hidden email]<mailto:[hidden email]>
>> 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
> [hidden email]<mailto:[hidden email]>
> 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:
> [hidden email]<mailto:[hidden email]>
>> 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
> [hidden email]<mailto:[hidden email]>
> 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:
> [hidden email]
> 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
> [hidden email]
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[hidden email]
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
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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Re: Audio quality issues

Guillermo Ruiz Camauer
You have turned off CDR, but what about logging?  Logging at certain levels gets pretty taxing at these call rates.
 
Guillermo


On Tue, Oct 22, 2013 at 3:41 PM, Deon Vermeulen <[hidden email]> 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" <[hidden email]> 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 <[hidden email]> 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: [hidden email]
>> [mailto:[hidden email]] 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
>> <[hidden email]<mailto:[hidden email]>> 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:
> [hidden email]<mailto:[hidden email]>
> [mailto:[hidden email]<mailto:[hidden email]>]
>> 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
>> <[hidden email]<mailto:[hidden email]>> 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:
> [hidden email]<mailto:[hidden email]>
> [mailto:[hidden email]<mailto:[hidden email]>]
>> 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 <[hidden email]<mailto:[hidden email]>>
>> To: "FreeSWITCH Users Help
> ([hidden email]<mailto:[hidden email]>)"
> <[hidden email]<mailto:[hidden email]>>
>> 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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Re: Audio quality issues

Stanislav Sinyagin
In reply to this post by Grant Bagdasarian
"delay_echo" is even more convenient for testing: then you can clearly hear your own voice without distortions from echo cancellation.



From: Grant Bagdasarian <[hidden email]>
To: FreeSWITCH Users Help <[hidden email]>
Sent: Tuesday, October 22, 2013 4:03 PM
Subject: Re: [Freeswitch-users] Audio quality issues

Yes, it does!
 
 
Echo application for FS.
Going to see if that works first. If not, I’ll setup a Sipp in server mode.
 
From: [hidden email] [mailto:[hidden email]] 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 <[hidden email]> 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: [hidden email] [mailto:[hidden email]] 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 <[hidden email]> 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: [hidden email] [mailto:[hidden email]] 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 <[hidden email]>
To: "FreeSWITCH Users Help ([hidden email])" <[hidden email]>
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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Re: Audio quality issues

Donny Hardyanto
In reply to this post by Stanislav Sinyagin

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" <[hidden email]>
Date: Oct 22, 2013 4:12 PM
Subject: Re: [Freeswitch-users] Audio quality issues
To: "FreeSWITCH Users Help" <[hidden email]>
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 <[hidden email]>
To: "FreeSWITCH Users Help ([hidden email])" <[hidden email]>
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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Re: Audio quality issues

Anthony Minessale
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 <[hidden email]> 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" <[hidden email]>
Date: Oct 22, 2013 4:12 PM
Subject: Re: [Freeswitch-users] Audio quality issues
To: "FreeSWITCH Users Help" <[hidden email]>
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 <[hidden email]>
To: "FreeSWITCH Users Help ([hidden email])" <[hidden email]>
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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Re: Audio quality issues

I put the Who? in Mishehu
In reply to this post by Grant Bagdasarian
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: [hidden email] [[hidden email]] 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 <[hidden email]>
To: "FreeSWITCH Users Help ([hidden email])" <[hidden email]>
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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Re: Audio quality issues

Miha Zoubek
In reply to this post by Steve Underwood
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:*[hidden email]
>> [mailto:[hidden email]] *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 <[hidden email] <mailto:[hidden email]>>
>> *To:* "FreeSWITCH Users Help ([hidden email]
>> <mailto:[hidden email]>)"
>> <[hidden email]
>> <mailto:[hidden email]>>
>> *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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