[asterisk-users] Article - a method on how to evaluate an Asteriskserver

[Jeff Brower]

Ioan-

Sounds like this would give a useful measurement regardless of server type, network config, and other variable issues.
 That should be a great tool.

Do you have any plans to test with Asterisk in 'native bridging' mode?  I.e. with RTP streams not touched in any way
by Asterisk?  I assume that would be the absolute max that Asterisk can handle.

-Jeff

> I would like to share with you an article [1] we have issued last week
> (sorry, currently only in Romanian language - we plan to provide an
> English version soon).
>
> This article is describing a method to be used for obtaining the
> maximum number of SIP simultaneous calls an Asterisk server could
> process safely (meaning no errors/maintain control of the machine and
> without RTP frame drops)
>
> We used SIPP (with modified uas and uac_pcap scenarios) + 2 scripts
> for controlling the test (one is running on the tested Asterisk server
> - start-test.sh, for data collection and load analysis and the other
> is running on the SIPP+Asterisk testing machine, for call quality
> control and SIPP instance control - sipp-controller.sh) + customized
> Asterisk dialplans and SIP configuration.
>
> The best part is that this method could be used for testing any type
> of Asterisk PBXs (from embedded to bigger servers), having
> capabilities to balance the load to several SIPP call
> generators/answer engines in case the tested server have more
> processing power than the testing machine. We have use this method to
> test 4 machines and the results are for the maximum number of G.711
> ulaw - ulaw SIP calls are summarized in [2].
>
> Also, this method is describing how to configure SIPP and Asterisk in
> order to test different transcoding scenarios (like ulaw to gsm).
>
> Basically the controller script increase the number of simultaneous
> calls (one SIPP call generator is calling an extension on the tested
> Asterisk server and the call is answered by anotther SIPP answer
> engine) till one of the load or quality tests failed.
>
> The tests are:
> 	- load evaluation -> how much time a `sleep 1` command take on the
> tested server
> 	- SIP RTT evaluation -> what is the average RTT of a SIP INVITE message
> 	- audio quality evaluation -> based on evaluating of the call
> "monitor" file size  (on the tested Asterisk server we use an echo
> application and the file is recorded on the testing machine)
>
> Even that the translation service provided free by Google is not the
> best way to read our article in English (or other languages) I
> encourage you to read it (the pictures and the results are very easy
> to understand) and send your feedback or comments here.
>
> Best regards,
> --
> Ioan Indreias
> www.modulo.ro
>
> Notes:
> [1] - http://www.modulo.ro/Modulo/ro/Articole/Determinarea_capacitatii_maxime_a_unei_centrale_Asterisk.html
>
> [2] Maximum number of G.711 ulaw - ulaw SIP calls
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