Hi Brian, I imagine that one of the issues is that you're using a complex sledgehammer (mod_conference) to crack a simple nut - that of having multiple listeners listening to a single speaker.
As far as I am aware, FreeSWITCH doesn't have anything built in which will allow this kind of simple audio path switching - maybe someone more knowledgeable than me will correct me if I'm wrong? I presented some stuff at ClueCon which would address this kind of simple application and ought to scale well beyond what you've seen with FS or Asterisk. It's still pretty basic [I'd do more with it if I wasn't so busy joshing with the other Brian on Facebook], and has never been deployed in anger but, if you're interested, drop me a note off-list. --Dave > 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 > 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 > > > _______________________________________________ > 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