What is your dialplan on the secondary box?
On Dec 18, 2009, at 9:08 AM, Brian <br...@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: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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