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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I’ve got FS running on a 64 bit OS, and here is more info
on the test procedure.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>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.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>The primary server dials into the secondary conference server so
that the listeners can hear the speaker conference on the primary server.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>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.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>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.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>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:<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> <profile name="listener"><o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> <param name="domain"
value="$${domain}"/><o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> <param name="rate"
value="8000"/><o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> <param
name="moh-sound" value="moh.wav"/><o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> <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"/><o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> <param
name="caller-controls" value="listener_controls"/><o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> </profile><o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I do have caller controls for the listener, since in my
production I will need to generate and handle events for listener DTMF.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>To compare FreeSWITCH vs Asterisk, I just swap out the secondary
conference server and everything else stays the same.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Brian.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> Brian West [mailto:brian@freeswitch.org] <br>
<b>Sent:</b> Thursday, December 17, 2009 5:20 PM<br>
<b>To:</b> freeswitch-users@lists.freeswitch.org<br>
<b>Subject:</b> Re: [Freeswitch-users] mod_conference scalability<o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>What exactly are you doing I know it goes better than that..
are you using 64bit?<o:p></o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<p class=MsoNormal>/ b<o:p></o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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<div>
<p class=MsoNormal>On Dec 17, 2009, at 3:41 PM, Brian wrote:<o:p></o:p></p>
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<p class=MsoNormal><br>
<br>
<o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>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.</span><o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> </span><o:p></o:p></p>
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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Brian.</span><o:p></o:p></p>
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<p class=MsoNormal><o:p> </o:p></p>
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