[Freeswitch-users] mod_conference scalability

Anthony Minessale anthony.minessale at gmail.com
Fri Dec 18 09:47:40 PST 2009


yes, I understand.
My reply was to the thread in general not directed at you =p


On Fri, Dec 18, 2009 at 11:41 AM, François Delawarde <
fdelawarde at wirelessmundi.com> wrote:

> It was of course just bad humor, I love both projects for what they are,
> and I agree that both have their own advantages and inconvenients.
>
> For example, accessing that same conference from a dahdi card could be
> another goal where Asterisk would be at an advantage, as chan_dahdi is
> still superior (in the more tested sense) than openzap+mod_openzap.
>
> I just use both projects separately or together depending on what's
> needed!
>
> I'm no banker nor do I understand the code, but many thanks for all
> those unpaid contributions providing an excellent alternative for free
> telephony. Your names really deserve being engraved in google's cache
> for eternity. :-)
>
> But still, I would like to see those numbers...
>
> François.
>
>
> On Fri, 2009-12-18 at 10:34 -0600, Anthony Minessale wrote:
> > 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<sip%3A888 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.
> >         >
> >         >
> >         >
> >         >
> >         >
> >         > _______________________________________________
> >         > FreeSWITCH-users mailing list
> >         > FreeSWITCH-users at lists.freeswitch.org
> >         >
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         >
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> >         >
> >         >
> >         >
> >         >
> >         >
> >         >
> >         > _______________________________________________
> >         > FreeSWITCH-users mailing list
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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<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
> >         >
> >         >
> >         >
> >         > _______________________________________________
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> >         > FreeSWITCH-users at lists.freeswitch.org
> >         >
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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<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
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> >         > iax:guest at conference.freeswitch.org/888
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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
> >
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> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
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> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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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 <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>
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