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 con
ference in a way that probably does not match reality”. In fact, thi
s will very much reflect the reality of the application I’m building
. Only instead of 300 listeners, I need to scale to over 2000 listen
ers minimum – per event, with possibly more than one concurrent even
t. I want to pack as many listeners on one server as I can. I’m tryi
ng 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.miness...@gmail.com]
Sent: Friday, December 18, 2009 11:34 AM
To: freeswitch-users@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:8...@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 <fdelawa...@wirelessmundi.co
m> 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.miness...@gmail.com]
> Sent: Thursday, December 17, 2009 3:49 PM
> To: freeswitch-users@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 <br...@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 w
ill
> 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.miness...@gmail.com]
> Sent: Thursday, December 17, 2009 2:42 PM
>
>
> To: freeswitch-users@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 <br...@proximosystems.com>
> wrote:
>
> Hi Mike,
>
>
>
> I didn’t get around to testing on the FreeSWITCH trunk yet. Are th
ere
> 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 scalab
le
> 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:m...@jerris.com]
> Sent: Thursday, December 17, 2009 10:18 AM
> To: freeswitch-users@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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>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_miness...@hotmail.com
> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
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> iax:gu...@conference.freeswitch.org/888
> googletalk:conf+...@conference.freeswitch.org
> pstn:+19193869900
>
>
>
> _______________________________________________
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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_miness...@hotmail.com
> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:8...@conference.freeswitch.org
> iax:gu...@conference.freeswitch.org/888
> googletalk:conf+...@conference.freeswitch.org
> pstn:+19193869900
>
>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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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_miness...@hotmail.com
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