[Freeswitch-users] mod_conference scalability

Anthony Minessale anthony.minessale at gmail.com
Thu Dec 17 11:42:03 PST 2009


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.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
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 <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091217/4d726120/attachment-0002.html 


More information about the FreeSWITCH-users mailing list