[Freeswitch-users] mod_conference scalability

[Michael Jerris]

What is your dialplan on the secondary box?

On Dec 18, 2009, at 9:08 AM, Brian <brian at proximosystems.com> wrote:

> I’ve got FS running on a 64 bit OS, and here is more info on the tes 
> t procedure.
>
>
>
> I’ve got one server (primary) that hosts the speaker call (this is m 
> eant to be a primary conference with a few speakers, but my test sim 
> plifies this to just one speaker). I’ve got a second server (seconda 
> ry) that hosts the conference that all the listeners go into, and I  
> have two other servers that I use automate the listener calls. The g 
> oal is to have several secondary servers to scale the listener side  
> of things, but for this initial test I’ve only got one secondary ser 
> ver.
>
>
>
> The primary server dials into the secondary conference server so  
> that the listeners can hear the speaker conference on the primary  
> server.
>
>
>
> The automated listener servers start dialing into the listener  
> conference at a combined rate of 5 calls per second (i.e. 2.5 calls  
> per second each). The play an audio loop that represents noise on  
> their end, which since they are listeners, should be ignored anyway.
>
>
>
> As I ramp up the automated listener calls, I manually call into the  
> conference from either my SIP phone, or from a land line using a DID  
> that I have directed to the conference.
>
>
>
> All calls are using SIP with uLaw 8000hz codec. Also, I’ve set up th 
> e profile for the listener conference to disable many of the events:
>
>
>
>     <profile name="listener">
>
>       <param name="domain" value="$${domain}"/>
>
>       <param name="rate" value="8000"/>
>
>       <param name="moh-sound" value="moh.wav"/>
>
>       <param name="suppress-events" value="start-talking,stop- 
> talking,energy-level,volume-level,gain-level,mute-detect,energy- 
> level-member,volume-in-member,volume-out-member,lock,unlock,floor- 
> change"/>
>
>       <param name="caller-controls" value="listener_controls"/>
>
>     </profile>
>
>
>
> I do have caller controls for the listener, since in my production I  
> will need to generate and handle events for listener DTMF.
>
>
>
> To compare FreeSWITCH vs Asterisk, I just swap out the secondary  
> conference server and everything else stays the same.
>
>
>
> Brian.
>
>
>
> From: Brian West [mailto:brian at freeswitch.org]
> Sent: Thursday, December 17, 2009 5:20 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_conference scalability
>
>
>
> What exactly are you doing I know it goes better than that.. are you  
> using 64bit?
>
>
>
> / b
>
>
>
> On Dec 17, 2009, at 3:41 PM, Brian 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.
>
>
>
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