What exactly is your test process? you should try increasing the interval in the conference profile to a bigger time slice maybe 30 40 or 60ms you could also increase the ptime to match as well.
like brian said you could use mod_shout to broadcast the single speaker to icecast and let people listen with itunes/winamp On Thu, Dec 17, 2009 at 3:41 PM, Brian <[email protected]> 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:[email protected]] > *Sent:* Thursday, December 17, 2009 3:49 PM > > *To:* [email protected] > *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 <[email protected]> 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:[email protected]] > *Sent:* Thursday, December 17, 2009 2:42 PM > > > *To:* [email protected] > *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 <[email protected]> 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:[email protected]] > *Sent:* Thursday, December 17, 2009 10:18 AM > *To:* [email protected] > *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 > [email protected] > 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 > [email protected] > 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:[email protected] <msn%[email protected]> > GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:[email protected] <sip%[email protected]> > iax:[email protected]/888 > googletalk:[email protected]<googletalk%3aconf%[email protected]> > pstn:+19193869900 > > > _______________________________________________ > FreeSWITCH-users mailing list > [email protected] > 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:[email protected] <msn%[email protected]> > GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:[email protected] <sip%[email protected]> > iax:[email protected]/888 > googletalk:[email protected]<googletalk%3aconf%[email protected]> > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-users mailing list > [email protected] > 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:[email protected] <msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <sip%[email protected]> iax:[email protected]/888 googletalk:[email protected]<googletalk%3aconf%[email protected]> pstn:+19193869900
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