[Freeswitch-users] mod_conference scalability

David Knell dave at 3c.co.uk
Thu Dec 17 13:06:48 PST 2009 

Hi Brian,

I imagine that one of the issues is that you're using a complex
sledgehammer (mod_conference) to crack a simple nut - that of having
multiple listeners listening to a single speaker.

As far as I am aware, FreeSWITCH doesn't have anything built in which
will allow this kind of simple audio path switching - maybe someone more
knowledgeable than me will correct me if I'm wrong?

I presented some stuff at ClueCon which would address this kind of
simple application and ought to scale well beyond what you've seen with
FS or Asterisk.  It's still pretty basic [I'd do more with it if I
wasn't so busy joshing with the other Brian on Facebook], and has never
been deployed in anger but, if you're interested, drop me a note
off-list.

--Dave

> 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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>  
> 
> 
> 
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> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
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> 
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