Re: [Freeswitch-users] mod_conference scalability

2009-12-18 Thread Steven Ayre
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 anthony.miness...@gmail.com:
 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 br...@proximosystems.com 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 

Re: [Freeswitch-users] rootkit?

2008-12-08 Thread Steven Ayre
A few files which you can choose to install to let the Freeswitch
developers access your machine remotely if you ask them to look at a
problem you're having with Freeswitch.

It's not a real rootkit - just the SSH public key for their private
key (so they don't need a password) and configuration files for bash +
emacs (so they're working in the environment they're used to). It
doesn't burrow into your OS, hide itself or any of the other insidious
things a real rootkit does, and once you delete the files they can no
longer access your machine.

It isn't installed unless you choose to do so.

-Steve



2008/12/7 Martin Joseph [EMAIL PROTECTED]:
 What is the rootkit item that appears to be added to the SVN of trunk?

 Thanks,
 Marty


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