I have recently announced on asterisk-users list about a new article we (Modulo Consulting) have issued almost 2 weeks ago. In this article (sorry, currently only in Romanian language - we plan to provide an English version soon) we described a method to be used in order to evaluate an Asterisk server in terms of maximum number of (SIP) calls it can handle.
In order to answer and to track more easy the questions and comments received on this article we decided to post on this forum as well.
Bellow is the announce we have made on asterisk-users list. Feel free to post about this article here - we will monitor this topic for new posts.
[quote] 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 .
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.
 - modulo.ro/Modulo/ro/Articole … erisk.html
 Maximum number of G.711 ulaw - ulaw SIP calls
[ul][li] 38 - Asterisk 1.4.29 (Astlinux 0.7.0) on Norhtec MicroClient Jr DX[/li]
[li] 130 - Asterisk 1.4.29 (Astlinux 0.7.0) on VIA EPIA EN12000EG[/li]
[li] 176 - Asterisk 1.4.22-4 (Trixbox CE 18.104.22.168) on Asus Pundit R350[/li]
[li] 320 - Asterisk 22.214.171.124 (Vicidialnow CE 1.3) on Gigabyte 945GCM-S2L[/li][/ul]