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 < fdelawa...@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:8...@conference.freeswitch.org<sip%3a...@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.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.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 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.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 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: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. > > > > > > > > > > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users@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@lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > 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_miness...@hotmail.com<msn%3aanthony_miness...@hotmail.com> > > > > > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com<paypal%3aanthony.miness...@gmail.com> > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > > > sip:8...@conference.freeswitch.org<sip%3a...@conference.freeswitch.org> > > > iax:gu...@conference.freeswitch.org/888 > > > > > googletalk:conf+...@conference.freeswitch.org<googletalk%3aconf%2b...@conference.freeswitch.org> > > > pstn:+19193869900 > > > > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users@lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > 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_miness...@hotmail.com<msn%3aanthony_miness...@hotmail.com> > > > > > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com<paypal%3aanthony.miness...@gmail.com> > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > > > sip:8...@conference.freeswitch.org<sip%3a...@conference.freeswitch.org> > > > iax:gu...@conference.freeswitch.org/888 > > > > > googletalk:conf+...@conference.freeswitch.org<googletalk%3aconf%2b...@conference.freeswitch.org> > > > pstn:+19193869900 > > > > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users@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@lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > 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_miness...@hotmail.com <msn%3aanthony_miness...@hotmail.com> > > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com<paypal%3aanthony.miness...@gmail.com> > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:8...@conference.freeswitch.org <sip%3a...@conference.freeswitch.org> > > iax:gu...@conference.freeswitch.org/888 > > googletalk:conf+...@conference.freeswitch.org<googletalk%3aconf%2b...@conference.freeswitch.org> > > pstn:+19193869900 > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users@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@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > 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_miness...@hotmail.com <msn%3aanthony_miness...@hotmail.com> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com<paypal%3aanthony.miness...@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org <sip%3a...@conference.freeswitch.org> iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org<googletalk%3aconf%2b...@conference.freeswitch.org> pstn:+19193869900
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