[asterisk-users] server performance....
Sevana Oy
sales at sevana.fi
Fri Mar 4 20:01:12 CST 2011
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" <lathama at gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
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
> <viswavardhanreddy at gmail.com> 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 lathama at gmail.com ~~~
>
> --
> _____________________________________________________________________
> -- 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
More information about the asterisk-users
mailing list