Brian, You haven't said what codecs are being used yet. Are the listeners using a different codec to the speaker? If so, you're potentially doing transcoding on every single channel, which would make CPU usage skyrocket.
-Steve 2009/12/17 Anthony Minessale <[email protected]>: > 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] >> GTALK/JABBER/PAYPAL:[email protected] >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:[email protected] >> iax:[email protected]/888 >> googletalk:[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] >> GTALK/JABBER/PAYPAL:[email protected] >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:[email protected] >> iax:[email protected]/888 >> googletalk:[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] > GTALK/JABBER/PAYPAL:[email protected] > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:[email protected] > iax:[email protected]/888 > googletalk:[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 > > _______________________________________________ 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
