[Freeswitch-users] mod_conference scalability

David Knell dave at 3c.co.uk
Fri Dec 18 12:56:41 PST 2009 

Hi Brian,

Have a look at this:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop
- I took a quick look through the code and couldn't see any reason why
you shouldn't have a bunch of eavesdroppers listening to a single
caller.  I'd be surprised if this didn't perform a lot better for your
application.

Cheers --

Dave

> I was evaluating the technologies available, and I thought you would
> be interested in my results. However, almost every other reply I get
> from you to my posts, rather than being helpful, has been hostile and
> insulting.
> 
>  
> 
> My scenario is not a hypothetical one of “having robots call the
> conference in a way that probably does not match reality”. In fact,
> this will very much reflect the reality of the application I’m
> building. Only instead of 300 listeners, I need to scale to over 2000
> listeners minimum – per event, with possibly more than one concurrent
> event. I want to pack as many listeners on one server as I can. I’m
> trying to find a real solution to a real problem.
> 
>  
> 
> I work with other open source projects and fund enhancements or fixes
> I need. FreeSWITCH would be no different. 
> 
>  
> 
> Brian.
> 
>  
> 
>  
> 
> From: Anthony Minessale [mailto:anthony.minessale at gmail.com] 
> Sent: Friday, December 18, 2009 11:34 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_conference scalability
> 
> 
>  
> 
> Conferencing is hardly the best place to judge performance.
> Quality is a far more important goal to me in conferencing.
> 
> Lets compare who can do 48khz conferences with several 32k siren
> callers on a polycom 6000, several more using G722 at 16khz and
> another handful of people on g711 ulaw all at different rates and
> ptimes talking in near-real time with low delay and low echo.  The
> fact that you can broadcast the conferences to icecast, control it
> from an external application and play files etc, and oh yeah, it can
> stream video.
> 
> Frankly, considering this is a free software project and so many
> people benefit, i would rather focus on quality than what numbers i
> can get from having robots call the conference in some way that
> probably does not match reality.  I would love for someone to sponsor
> the effort to add features to the conference module, but of course, I
> do not hold my breath, instead I continue to improve it for free when
> I find time.  This is one of many reasons I do not enjoy performance
> discussions unless I am talking to an engineer who understands the
> code or a banker ready to pay for improvements.  That is not my way of
> saying pay me or forget it as you can clearly see the conference
> module has made it to where it is today with no financial support at
> all.  Just the efforts of myself and several brave volunteers over the
> years who have contributed to it.
> 
> BTW,
> 
> We have a weekly call, there is one today in 30 minutes.
> Drop by sip:888 at conference.freeswitch.org This is just an openVZ
> instance mind you running at 48khz waiting for anyone to call in and
> say hi.
> 
> 
> 
> 
> 
> 
> On Fri, Dec 18, 2009 at 10:12 AM, François Delawarde
> <fdelawarde at wirelessmundi.com> wrote:
> 
> Hearing that Asterisk (1.4) scales 2x like FS is not common, sounds
> like
> a configuration error.
> 
> If not, I already see the title of the next Digium blog entry:
> "FreeSwitch scalability myth finally ends: The worst Asterisk version
> ever (1.4) beating the crap of the best and latest FS."
> 
> Anyway, you should compare FS trunk to Asterisk 1.6.2 to see who wins
> the final conference battle! :-)
> 
> François.
> 
> 
> 
> On Thu, 2009-12-17 at 16:41 -0500, Brian wrote:
> > 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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> > FreeSWITCH-users at lists.freeswitch.org
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> >
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> >
> >
> >
> >
> >
> >
> > _______________________________________________
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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
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:+19193869900
> >
> >
> >
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> >
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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
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:+19193869900
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
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> >
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> 
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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
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> 
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