Hi,

We have worked out another approach for load testing:

- generate using sipp certain number of test calls and that go to PBX echo server playing and receiving back pre-defined audio
- generate +1 test call, which also plays and receives back an audio file

Then we test the audio we received from the +1 test call using AQuA (Audio Quality Analyzer) and obtain a MOS score (AQuA is doing perceptual audio quality assessment, it's not calculating MOS as in G.107, but more likely in P.862, although the algorithms are absolutely different).

In this way we can always know how many calls can the PBX under test handle before actual call quality goes down. The whole test suit is put together with other testing (loop back call testing, conference bridge testing) capabilities into what we call Asterisk VQM. If my previous message goes through moderation you will be able to see screenshots as well :)

Best Regards,
Sevana Oy
http://www.sevana.fi


----- Original Message ----- From: "Andrew Latham" <[email protected]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]>
Sent: Friday, March 04, 2011 8:20 PM
Subject: Re: [asterisk-users] server performance....


On Fri, Mar 4, 2011 at 2:13 PM, viswavardhanreddy karna
<[email protected]> wrote:
Hi every one,
I am doing some experiments on asterisk server
performance...... How can we know server performance? can any one explain me
plz....
I have 2 doubts regarding the asterisk server performance...

1. When can we know asterisk server performance?
1. when server is in idle state ?
2. when the server is in busy state?

can any one please tell me when can the server performance is known i mean
when server is busy or in idle state?????????

Best Regards,
viswavardhanreddy


Many people test their servers with call-setups and call tear-downs.
Using another tool like sipp you can send 100-1000s of call-setups and
then do call tear-downs.  You can also use transcoding loops to test
the load.  If you have 1 call that is sent to a context where it dials
exten+1 and continues the loop until a target number, you can then set
the codec for each dialed number.  I know that there are many methods
of testing and this is just a common one.

~~~ Andrew "lathama" Latham [email protected] ~~~

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