yes, I understand.<br>My reply was to the thread in general not directed at you =p<br><br><br><div class="gmail_quote">On Fri, Dec 18, 2009 at 11:41 AM, François Delawarde <span dir="ltr"><<a href="mailto:fdelawarde@wirelessmundi.com">fdelawarde@wirelessmundi.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">It was of course just bad humor, I love both projects for what they are,<br>
and I agree that both have their own advantages and inconvenients.<br>
<br>
For example, accessing that same conference from a dahdi card could be<br>
another goal where Asterisk would be at an advantage, as chan_dahdi is<br>
still superior (in the more tested sense) than openzap+mod_openzap.<br>
<br>
I just use both projects separately or together depending on what's<br>
needed!<br>
<br>
I'm no banker nor do I understand the code, but many thanks for all<br>
those unpaid contributions providing an excellent alternative for free<br>
telephony. Your names really deserve being engraved in google's cache<br>
for eternity. :-)<br>
<br>
But still, I would like to see those numbers...<br>
<font color="#888888"><br>
François.<br>
</font><div><div></div><div class="h5"><br>
<br>
On Fri, 2009-12-18 at 10:34 -0600, Anthony Minessale wrote:<br>
> Conferencing is hardly the best place to judge performance.<br>
> Quality is a far more important goal to me in conferencing.<br>
><br>
> Lets compare who can do 48khz conferences with several 32k siren<br>
> callers on a polycom 6000, several more using G722 at 16khz and<br>
> another handful of people on g711 ulaw all at different rates and<br>
> ptimes talking in near-real time with low delay and low echo. The<br>
> fact that you can broadcast the conferences to icecast, control it<br>
> from an external application and play files etc, and oh yeah, it can<br>
> stream video.<br>
><br>
> Frankly, considering this is a free software project and so many<br>
> people benefit, i would rather focus on quality than what numbers i<br>
> can get from having robots call the conference in some way that<br>
> probably does not match reality. I would love for someone to sponsor<br>
> the effort to add features to the conference module, but of course, I<br>
> do not hold my breath, instead I continue to improve it for free when<br>
> I find time. This is one of many reasons I do not enjoy performance<br>
> discussions unless I am talking to an engineer who understands the<br>
> code or a banker ready to pay for improvements. That is not my way of<br>
> saying pay me or forget it as you can clearly see the conference<br>
> module has made it to where it is today with no financial support at<br>
> all. Just the efforts of myself and several brave volunteers over the<br>
> years who have contributed to it.<br>
><br>
> BTW,<br>
><br>
> We have a weekly call, there is one today in 30 minutes.<br>
> Drop by <a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a> This is just an openVZ<br>
> instance mind you running at 48khz waiting for anyone to call in and<br>
> say hi.<br>
><br>
><br>
><br>
><br>
><br>
> On Fri, Dec 18, 2009 at 10:12 AM, François Delawarde<br>
> <<a href="mailto:fdelawarde@wirelessmundi.com">fdelawarde@wirelessmundi.com</a>> wrote:<br>
> Hearing that Asterisk (1.4) scales 2x like FS is not common,<br>
> sounds like<br>
> a configuration error.<br>
><br>
> If not, I already see the title of the next Digium blog entry:<br>
> "FreeSwitch scalability myth finally ends: The worst Asterisk<br>
> version<br>
> ever (1.4) beating the crap of the best and latest FS."<br>
><br>
> Anyway, you should compare FS trunk to Asterisk 1.6.2 to see<br>
> who wins<br>
> the final conference battle! :-)<br>
><br>
> François.<br>
><br>
><br>
><br>
> On Thu, 2009-12-17 at 16:41 -0500, Brian wrote:<br>
> > I did a test with the trunk version for the one conference<br>
> case, and<br>
> > it is the same results as for 1.0.4. The audio failed at<br>
> around 300<br>
> > listeners. Oddly though, it consumed less %CPU (240% instead<br>
> of 300%),<br>
> > and yet the audio still failed at the same number of<br>
> listeners.<br>
> ><br>
> ><br>
> ><br>
> > Brian.<br>
> ><br>
> ><br>
> ><br>
> > From: Anthony Minessale [mailto:<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>]<br>
> > Sent: Thursday, December 17, 2009 3:49 PM<br>
> > To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
> > Subject: Re: [Freeswitch-users] mod_conference scalability<br>
> ><br>
> ><br>
> ><br>
> ><br>
> > We didn't post it anywhere but we just get overwhelmed with<br>
> them and<br>
> > many of them are unfounded and take up a lot of time to<br>
> track down.<br>
> > That does not mean you have not found a real problem but the<br>
> first<br>
> > step is trying trunk.<br>
> ><br>
> ><br>
> ><br>
> ><br>
> > On Thu, Dec 17, 2009 at 2:32 PM, Brian<br>
> <<a href="mailto:brian@proximosystems.com">brian@proximosystems.com</a>><br>
> > wrote:<br>
> ><br>
> > I didn’t realize there was a policy about load testing<br>
> questions. What<br>
> > forum should I have used for this?<br>
> ><br>
> ><br>
> ><br>
> > I didn’t get the chance to test on FS trunk yet, but when I<br>
> do I will<br>
> > provide you with the feedback when I do. Just let me know<br>
> what forum<br>
> > to use for this topic from now on.<br>
> ><br>
> ><br>
> ><br>
> > Thanks,<br>
> ><br>
> ><br>
> ><br>
> > Brian.<br>
> ><br>
> ><br>
> ><br>
> > From: Anthony Minessale [mailto:<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>]<br>
> > Sent: Thursday, December 17, 2009 2:42 PM<br>
> ><br>
> ><br>
> > To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
> > Subject: Re: [Freeswitch-users] mod_conference scalability<br>
> ><br>
> ><br>
> ><br>
> ><br>
> > One man's stable release is another man's 6 month old<br>
> release with<br>
> > hundreds of known fixed bugs.<br>
> > If one of the core developers tells you to try it, you may<br>
> as well<br>
> > take the time to try it now that you have opened a forum<br>
> questioning<br>
> > the scalability.<br>
> ><br>
> > When you tested asterisk did you actually use 600 phones and<br>
> verify<br>
> > that each one can hear the audio perfectly and in time with<br>
> what the<br>
> > speaker was saying? Did you try same on FS?<br>
> ><br>
> > Did you optimize your dialplan on FS to deal with a load<br>
> test or<br>
> > follow any of the recommended performance tuning page.<br>
> ><br>
> > All of the answers to these questions are really moot<br>
> because we have<br>
> > a policy against entertaining load testing questions but if<br>
> you like<br>
> > asterisk, by all means, use it, and good luck to you if<br>
> those numbers<br>
> > you are testing at are what you plan to put in real<br>
> > production.........<br>
> ><br>
> > On Thu, Dec 17, 2009 at 1:29 PM, Brian<br>
> <<a href="mailto:brian@proximosystems.com">brian@proximosystems.com</a>><br>
> > wrote:<br>
> ><br>
> > Hi Mike,<br>
> ><br>
> ><br>
> ><br>
> > I didn’t get around to testing on the FreeSWITCH trunk yet.<br>
> Are there<br>
> > substantial fixes to mod_conference in the FreeSWITCH trunk<br>
> that might<br>
> > increase capacity for my scenario of one speaker and many<br>
> listeners?<br>
> > If I want to put this into a production environment, I would<br>
> need a<br>
> > stable version, which as far as I know is the 1.0.4 version.<br>
> ><br>
> ><br>
> ><br>
> > However, I did test on Asterisk 1.4 using app_conference,<br>
> and doing<br>
> > the same scenario was able to get 1 speaker and 600<br>
> listeners on a<br>
> > single conference with no audio issues. The CPU at that<br>
> point was just<br>
> > over 300%, same as where the single conference scenario<br>
> failed on<br>
> > FreeSWITCH with 300 listeners. I was able to push it to<br>
> over 700<br>
> > listeners before I reached 400% CPU usage (I guess maxing<br>
> out my<br>
> > quad-core processors), and asterisk finally crashed. But up<br>
> until that<br>
> > point, there were no audio problems.<br>
> ><br>
> ><br>
> ><br>
> > I’ve read a lot about how FreeSWITCH is supposed to be more<br>
> scalable<br>
> > than Asterisk, but unless there is something wrong with my<br>
> FreeSWITCH<br>
> > setup, Asterisk was clearly the winner in this test – more<br>
> than<br>
> > doubling FreeSWITCH capacity in this case. Again, maybe<br>
> there is<br>
> > something on the FreeSWITCH side that I’m doing wrong, but I<br>
> don’t see<br>
> > what it could be.<br>
> ><br>
> ><br>
> ><br>
> > Brian.<br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> > From: Michael Jerris [mailto:<a href="mailto:mike@jerris.com">mike@jerris.com</a>]<br>
> > Sent: Thursday, December 17, 2009 10:18 AM<br>
> > To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
> > Subject: Re: [Freeswitch-users] mod_conference scalability<br>
> ><br>
> ><br>
> ><br>
> ><br>
> > I would be curious what the same tests produce with svn<br>
> trunk of<br>
> > FreeSWITCH.<br>
> ><br>
> ><br>
> ><br>
> ><br>
> > Mike<br>
> ><br>
> ><br>
> ><br>
> ><br>
> > On Dec 16, 2009, at 4:49 PM, Brian wrote:<br>
> ><br>
> ><br>
> ><br>
> ><br>
> > Hi,<br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> > I’m new to FreeSWITCH and I’m testing the scalability of<br>
> > mod_conference to see if it will scale better that other<br>
> solutions. My<br>
> > scenario is to have one speaker, and many listeners (mute).<br>
> Since I<br>
> > have only one speaker, I was expecting this to scale well<br>
> because<br>
> > there is no audio mixing required, just send each frame of<br>
> the single<br>
> > speaker to each listener. Unfortunately, my testing was<br>
> disappointing,<br>
> > and it didn’t scale nearly as well as I’d hoped (based on<br>
> what I’ve<br>
> > read on how FreeSWITCH is supposed to be generally very<br>
> scalable).<br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> > Here’s my server setup is this:<br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> > FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon<br>
> server, 4 Gig<br>
> > of RAM. I’ve set file logging to “notice” level. My<br>
> conference profile<br>
> > is configured to suppress several events, hoping that it<br>
> would improve<br>
> > performance.<br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> > Here are a few scenarios I tested, and roughly where I<br>
> reached the<br>
> > point of audio failure on the conferences:<br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> > Scenario 1:<br>
> ><br>
> ><br>
> > 1 conference, 1 speaker, audio failed at approx 300<br>
> listeners (mute)<br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> > Scenario 2:<br>
> ><br>
> ><br>
> > 4 conferences, 1 speaker per conference, audio failed approx<br>
> 110<br>
> > listeners per conference (so just over 400 total channels on<br>
> the<br>
> > system).<br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> > Scenario 3:<br>
> ><br>
> ><br>
> > 16 conferences, 1 speaker per conference, audio failed at 32<br>
> listeners<br>
> > per conference (so just over 500 total channels on the<br>
> system).<br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> > Looking at the output from “top”, it seems that in all 3<br>
> scenarios,<br>
> > the audio quality failed when the % CPU for the FreeSWITCH<br>
> process<br>
> > exceeded 300%.<br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> > I was hoping maybe someone else might have done similar<br>
> testing, or<br>
> > maybe has suggestions on how to improve the performance. Or<br>
> perhaps an<br>
> > alternate solution to the one speaker, many listener case?<br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> > Thanks,<br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> > Brian.<br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> > _______________________________________________<br>
> > FreeSWITCH-users mailing list<br>
> > <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
> ><br>
> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
> ><br>
> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> > _______________________________________________<br>
> > FreeSWITCH-users mailing list<br>
> > <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
> ><br>
> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
> ><br>
> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> > --<br>
> > Anthony Minessale II<br>
> ><br>
> > FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>
> > ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
> > Twitter: <a href="http://twitter.com/FreeSWITCH_wire" target="_blank">http://twitter.com/FreeSWITCH_wire</a><br>
> ><br>
> > AIM: anthm<br>
> > <a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>
> > GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
> > IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br>
> ><br>
> > FreeSWITCH Developer Conference<br>
> > <a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br>
> > <a href="http://iax:guest@conference.freeswitch.org/888" target="_blank">iax:guest@conference.freeswitch.org/888</a><br>
> > <a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>
> > pstn:+19193869900<br>
> ><br>
> ><br>
> ><br>
> > _______________________________________________<br>
> > FreeSWITCH-users mailing list<br>
> > <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
> ><br>
> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
> ><br>
> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
> ><br>
> ><br>
> ><br>
> ><br>
> ><br>
> > --<br>
> > Anthony Minessale II<br>
> ><br>
> > FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>
> > ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
> > Twitter: <a href="http://twitter.com/FreeSWITCH_wire" target="_blank">http://twitter.com/FreeSWITCH_wire</a><br>
> ><br>
> > AIM: anthm<br>
> > <a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>
> > GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
> > IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br>
> ><br>
> > FreeSWITCH Developer Conference<br>
> > <a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br>
> > <a href="http://iax:guest@conference.freeswitch.org/888" target="_blank">iax:guest@conference.freeswitch.org/888</a><br>
> > <a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>
> > pstn:+19193869900<br>
> ><br>
> ><br>
> > _______________________________________________<br>
> > FreeSWITCH-users mailing list<br>
> > <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
> ><br>
> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
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> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
><br>
><br>
> _______________________________________________<br>
> FreeSWITCH-users mailing list<br>
> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
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> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
><br>
><br>
><br>
><br>
> --<br>
> Anthony Minessale II<br>
><br>
> FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>
> ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
> Twitter: <a href="http://twitter.com/FreeSWITCH_wire" target="_blank">http://twitter.com/FreeSWITCH_wire</a><br>
><br>
> AIM: anthm<br>
> <a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>
> GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
> IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br>
><br>
> FreeSWITCH Developer Conference<br>
> <a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br>
> <a href="http://iax:guest@conference.freeswitch.org/888" target="_blank">iax:guest@conference.freeswitch.org/888</a><br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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