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 > 38 - Norhtec MicroClient Jr DX > 130 - VIA EPIA EN12000EG > 176 - Asus Pundit R350 > 320 - Gigabyte 945GCM-S2L -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users