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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Have you verified that you are actually
sending sound over the RTP streams? <o:p></o:p></span></font></p>
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10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Rob Lith<br>
<b><span style='font-weight:bold'>Sent:</span></b> Friday, January 27, 2006
11:13 PM<br>
<b><span style='font-weight:bold'>To:</span></b> Asterisk Users Mailing List -
Non-Commercial Discussion<br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [Asterisk-Users]
5,000 concurrent calls system rollout question</span></font><o:p></o:p></p>
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<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>What's you mix of calls
going SIP/IAXand to PSTN?<br>
<br>
We've done some benchmark experiments on a 3GHz HT box with 1GB of ram,
mirrored traditional IDE disks. The box has a Digium quad-PRI, a TDM40B, a
TDM22B and a Sirrix quad-BRI board in it. This box can run 120 active calls
over 4 PRI spans. Its running <br>
MusicOnHold into 60 of the channels, playing various GSM prompts into the other
60. The "user" cpu usage is about 25%, the "system" cpu
about 25% also. We can add to that 5000 registered SIP peers and 5000 registered
IAX2 peers - total of about 100 registration refreshes per second. That adds
about 40% more user CPU and pretty much fills up CPU. Audio quality is still
perfectly fine, and PRI slips few and far between. Load average for the whole
mix sits between 5 and 10. About 550Kb/s out and 400Kb/s out on the ethernet
for the registration traffic. <br>
<br>
Also on <a href="http://www.voip-info.org">www.voip-info.org</a> - search for
dimensioning<br>
<br>
Rob<o:p></o:p></span></font></p>
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<p class=MsoNormal><span class=gmailquote><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>On 1/28/06, <b><span style='font-weight:bold'>Vic</span></b>
<<a href="mailto:svictor@yahoo.co.jp"> svictor@yahoo.co.jp</a>> wrote:</span></font></span><o:p></o:p></p>
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<p><font size=3 face="Times New Roman"><span style='font-size:12.0pt'>Hi,<o:p></o:p></span></font></p>
<p><font size=3 face="Times New Roman"><span style='font-size:12.0pt'>we are
currently considering different options for rolling out a large scale IP PBX
to handle around 3,000 + concurrent calls.<o:p></o:p></span></font></p>
<p><font size=3 face="Times New Roman"><span style='font-size:12.0pt'>Can
this be done with Asterisk? Has it been done before?<o:p></o:p></span></font></p>
<p><font size=3 face="Times New Roman"><span style='font-size:12.0pt'>I
really would like an input on this.<o:p></o:p></span></font></p>
<p><font size=3 face="Times New Roman"><span style='font-size:12.0pt'>Thanks!<o:p></o:p></span></font></p>
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