[Freeswitch-users] mod_conference scalability
François Delawarde
fdelawarde at wirelessmundi.com
Fri Dec 18 09:41:44 PST 2009
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 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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> >
> >
> >
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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
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> > iax:guest at conference.freeswitch.org/888
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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
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> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
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