<div dir="ltr"><div><div>Hi,<br></div>Came across this thread wile having the exact same issue, do you mind sharing some info about your solution?<br></div>Thanks<br><div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Dec 17, 2009 at 4:06 PM, David Knell <span dir="ltr"><<a href="mailto:dave@3c.co.uk" target="_blank">dave@3c.co.uk</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi Brian,<br>
<br>
I imagine that one of the issues is that you're using a complex<br>
sledgehammer (mod_conference) to crack a simple nut - that of having<br>
multiple listeners listening to a single speaker.<br>
<br>
As far as I am aware, FreeSWITCH doesn't have anything built in which<br>
will allow this kind of simple audio path switching - maybe someone more<br>
knowledgeable than me will correct me if I'm wrong?<br>
<br>
I presented some stuff at ClueCon which would address this kind of<br>
simple application and ought to scale well beyond what you've seen with<br>
FS or Asterisk. It's still pretty basic [I'd do more with it if I<br>
wasn't so busy joshing with the other Brian on Facebook], and has never<br>
been deployed in anger but, if you're interested, drop me a note<br>
off-list.<br>
<br>
--Dave<br>
<div class="HOEnZb"><div class="h5"><br>
> I didn’t realize there was a policy about load testing 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 do I will<br>
> provide you with the feedback when I do. Just let me know 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@<wbr>gmail.com</a>] <br>
> Sent: Thursday, December 17, 2009 2:42 PM<br>
> To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.<wbr>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 release with<br>
> hundreds of known fixed bugs.<br>
> If one of the core developers tells you to try it, you may as well<br>
> take the time to try it now that you have opened a forum questioning<br>
> the scalability.<br>
> <br>
> When you tested asterisk did you actually use 600 phones and verify<br>
> that each one can hear the audio perfectly and in time with 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 test or<br>
> follow any of the recommended performance tuning page.<br>
> <br>
> All of the answers to these questions are really moot because we have<br>
> a policy against entertaining load testing questions but if you like<br>
> asterisk, by all means, use it, and good luck to you if those numbers<br>
> you are testing at are what you plan to put in real<br>
> production.........<br>
> <br>
> <br>
> <br>
> On Thu, Dec 17, 2009 at 1:29 PM, Brian <<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. Are there<br>
> substantial fixes to mod_conference in the FreeSWITCH trunk that might<br>
> increase capacity for my scenario of one speaker and many listeners?<br>
> If I want to put this into a production environment, I would 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, and doing<br>
> the same scenario was able to get 1 speaker and 600 listeners on a<br>
> single conference with no audio issues. The CPU at that point was just<br>
> over 300%, same as where the single conference scenario failed on<br>
> FreeSWITCH with 300 listeners. I was able to push it to over 700<br>
> listeners before I reached 400% CPU usage (I guess maxing out my<br>
> quad-core processors), and asterisk finally crashed. But up 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 scalable<br>
> than Asterisk, but unless there is something wrong with my FreeSWITCH<br>
> setup, Asterisk was clearly the winner in this test – more than<br>
> doubling FreeSWITCH capacity in this case. Again, maybe there is<br>
> something on the FreeSWITCH side that I’m doing wrong, but I 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.<wbr>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 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 solutions. My<br>
> scenario is to have one speaker, and many listeners (mute). Since I<br>
> have only one speaker, I was expecting this to scale well because<br>
> there is no audio mixing required, just send each frame of the single<br>
> speaker to each listener. Unfortunately, my testing was disappointing,<br>
> and it didn’t scale nearly as well as I’d hoped (based on what I’ve<br>
> read on how FreeSWITCH is supposed to be generally very 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 server, 4 Gig<br>
> of RAM. I’ve set file logging to “notice” level. My conference profile<br>
> is configured to suppress several events, hoping that it would improve<br>
> performance.<br>
> <br>
> <br>
> <br>
> <br>
> <br>
> Here are a few scenarios I tested, and roughly where I 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 listeners (mute)<br>
> <br>
> <br>
> <br>
> <br>
> <br>
> Scenario 2:<br>
> <br>
> <br>
> 4 conferences, 1 speaker per conference, audio failed approx 110<br>
> listeners per conference (so just over 400 total channels on the<br>
> system).<br>
> <br>
> <br>
> <br>
> <br>
> <br>
> Scenario 3:<br>
> <br>
> <br>
> 16 conferences, 1 speaker per conference, audio failed at 32 listeners<br>
> per conference (so just over 500 total channels on the system).<br>
> <br>
> <br>
> <br>
> <br>
> <br>
> <br>
> <br>
> <br>
> Looking at the output from “top”, it seems that in all 3 scenarios,<br>
> the audio quality failed when the % CPU for the FreeSWITCH process<br>
> exceeded 300%.<br>
> <br>
> <br>
> <br>
> <br>
> <br>
> I was hoping maybe someone else might have done similar testing, or<br>
> maybe has suggestions on how to improve the performance. Or 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>
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> -- <br>
> Anthony Minessale II<br>
> <br>
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</div></div></blockquote></div><br></div></div></div></div></div>