[Freeswitch-users] mod_conference scalability

Thu Dec 17 12:48:33 PST 2009

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
>
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-- 
Anthony Minessale II

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