[Asterisk-Users] 5,000 concurrent calls system rollout question

[Paul Mahler]

Have you verified that you are actually sending sound over the RTP streams? 

 

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From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Rob Lith
Sent: Friday, January 27, 2006 11:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question

 

What's you mix of calls going SIP/IAXand to PSTN?

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 
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. 

Also on www.voip-info.org - search for dimensioning

Rob

On 1/28/06, Vic < svictor at yahoo.co.jp <mailto:svictor at yahoo.co.jp> > wrote:


Hi,

we are currently considering different options for rolling out a large scale
IP PBX to handle around 3,000 + concurrent calls.

Can this be done with Asterisk? Has it been done before?

I really would like an input on this.

Thanks!


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