[Freeswitch-users] mod_conference scalability

Thu Dec 17 13:41:01 PST 2009

I did a test with the trunk version for the one conference case, and it is
the same results as for 1.0.4. The audio failed at around 300 listeners.
Oddly though, it consumed less %CPU (240% instead of 300%), and yet the
audio still failed at the same number of listeners.

Brian.

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Thursday, December 17, 2009 3:49 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_conference scalability

We didn't post it anywhere but we just get overwhelmed with them and many of
them are unfounded and take up a lot of time to track down.  That does not
mean you have not found a real problem but the first step is trying trunk.

On Thu, Dec 17, 2009 at 2:32 PM, Brian <brian at proximosystems.com> wrote:

I didn't realize there was a policy about load testing questions. What forum
should I have used for this?

I didn't get the chance to test on FS trunk yet, but when I do I will
provide you with the feedback when I do. Just let me know what forum to use
for this topic from now on.

Thanks,

Brian.

From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
Sent: Thursday, December 17, 2009 2:42 PM

To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_conference scalability

One man's stable release is another man's 6 month old release with hundreds
of known fixed bugs.
If one of the core developers tells you to try it, you may as well take the
time to try it now that you have opened a forum questioning the scalability.

When you tested asterisk did you actually use 600 phones and verify that
each one can hear the audio perfectly and in time with what the speaker was
saying?  Did you try same on FS? 

Did you optimize your dialplan on FS to deal with a load test or follow any
of the recommended performance tuning page.

All of the answers to these questions are really moot because we have a
policy against entertaining load testing questions but if you like asterisk,
by all means, use it, and good luck to you if those numbers you are testing
at are what you plan to put in real production.........

On Thu, Dec 17, 2009 at 1:29 PM, Brian <brian at proximosystems.com> wrote:

Hi Mike,

I didn't get around to testing on the FreeSWITCH trunk yet. Are there
substantial fixes to mod_conference in the FreeSWITCH trunk that might
increase capacity for my scenario of one speaker and many listeners? If I
want to put this into a production environment, I would need a stable
version, which as far as I know is the 1.0.4 version.

However, I did test on Asterisk 1.4 using app_conference, and doing the same
scenario was able to get 1 speaker and 600 listeners on a single conference
with no audio issues. The CPU at that point was just over 300%, same as
where the single conference scenario failed on FreeSWITCH with 300
listeners.  I was able to push it to over 700 listeners before I reached
400% CPU usage (I guess maxing out my quad-core processors), and asterisk
finally crashed. But up until that point, there were no audio problems. 

I've read a lot about how FreeSWITCH is supposed to be more scalable than
Asterisk, but unless there is something wrong with my FreeSWITCH setup,
Asterisk was clearly the winner in this test - more than doubling FreeSWITCH
capacity in this case. Again, maybe there is something on the FreeSWITCH
side that I'm doing wrong, but I don't see what it could be.

Brian.

From: Michael Jerris [mailto:mike at jerris.com] 
Sent: Thursday, December 17, 2009 10:18 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_conference scalability

I would be curious what the same tests produce with svn trunk of FreeSWITCH.

Mike

On Dec 16, 2009, at 4:49 PM, Brian wrote:

Hi,

I'm new to FreeSWITCH and I'm testing the scalability of mod_conference to
see if it will scale better that other solutions. My scenario is to have one
speaker, and many listeners (mute). Since I have only one speaker, I was
expecting this to scale well because there is no audio mixing required, just
send each frame of the single speaker to each listener. Unfortunately, my
testing was disappointing, and it didn't scale nearly as well as I'd hoped
(based on what I've read on how FreeSWITCH is supposed to be generally very
scalable).

Here's my server setup is this:

FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of
RAM. I've set file logging to "notice" level. My conference profile is
configured to suppress several events, hoping that it would improve
performance.

Here are a few scenarios I tested, and roughly where I reached the point of
audio failure on the conferences:

Scenario 1:

1 conference, 1 speaker, audio failed at approx 300 listeners (mute)

Scenario 2:

4 conferences, 1 speaker per conference, audio failed approx 110 listeners
per conference (so just over 400 total channels on the system).

Scenario 3:

16 conferences, 1 speaker per conference, audio failed at 32 listeners per
conference (so just over 500 total channels on the system).

Looking at the output from "top", it seems that in all 3 scenarios, the
audio quality failed when the % CPU for the FreeSWITCH process exceeded
300%.

I was hoping maybe someone else might have done similar testing, or maybe
has suggestions on how to improve the performance. Or perhaps an alternate
solution to the one speaker, many listener case?

Thanks,

Brian.

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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
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