[Asterisk-Users] Load Testing
Tim Connolly
tim at timsnet.com
Sat Aug 13 11:27:41 MST 2005
I guess as long as music on hold was playing...
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tim Connolly
Sent: Saturday, August 13, 2005 1:25 PM
To: asterisk-biz at thevoipconnection.com; 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Load Testing
Almost.. A call on hold doesn't represent the true bandwidth and CPU that a
*real* call utilitizes. Short of producing an echo or feedback on each call
to make it look like a real call, I'm not sure how you could create a real
call test scenario.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of The VoIP
Connection
Sent: Friday, August 12, 2005 10:42 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Load Testing
Anton,
A great tool for "ghetto" call capacity testing is a single snom phone.
There is no limit to how many calls a snom phone can make, just put it on
hold and dial again. So, with a single snom phone and a little imagination
you can test any number of scenarios. You can approximate basic SIP
capacity by creating an extension that plays the asterisk test message and
dialing it repeatedly until quality starts to degrade or asterisk gives up.
To simulate actual call throughput you really need another (faster) machine
to connect to, but you can use the same technique.
You can run "top" on the console while you are doing your tests to see what
resources you are using. Check your logs when you are done to see what
errors were generated when it came unglued. CPU is not always the limiting
resource, especially with Digium card interfaces which tend to be bound by
FSB speed, but echo cancellation and codec conversion will burn a LOT of
cycles.
Michael Crown
Managing Partner
www.thevoipconnection.com
321.989.6728 ext. 611
sip:611 at voiceserver.thevoipconnection.com
> -----Original Message-----
> From: Anton Krall [mailto:akrall-lists at intruder.com.mx]
> Sent: Friday, August 12, 2005 9:56 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] Load Testing
>
> Guys.
>
> How and which tools to use to load test an asterisk install?
> Say for example, you need to see how many calls can be routed
> thru before losing quality and making the cpu jump to the roof?
>
>
>
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