I've got FS running on a 64 bit OS, and here is more info on the test
procedure.
I've got one server (primary) that hosts the speaker call (this is meant to
be a primary conference with a few speakers, but my test simplifies this to
just one speaker). I've got a second server (secondary) that hosts the
conference that all the listeners go into, and I have two other servers that
I use automate the listener calls. The goal is to have several secondary
servers to scale the listener side of things, but for this initial test I've
only got one secondary server.
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 the
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,fl
oor-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:br...@freeswitch.org]
Sent: Thursday, December 17, 2009 5:20 PM
To: freeswitch-users@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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