[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
> > >
> > 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
> > >
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> > > 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
> > >
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
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> > > 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
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > >
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > UNSUBSCRIBE:
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> > > http://www.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>
> > pstn:+19193869900
> > _______________________________________________
> > 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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