Have you verified that you are actually
sending sound over the RTP streams?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 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
< [EMAIL PROTECTED]> 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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