[asterisk-users] Fw: Stress testing Asterisk

Paul Belanger paul.belanger at polybeacon.com
Wed May 22 10:25:05 CDT 2013


On 13-05-22 10:02 AM, Tommy Cooper wrote:
>  From the little experience I have I do not think that that is a good way of testing the quality of voice. SIP only initiates and eventually terminates the call, once that the call is connected, SIP and therefore Asterisk are no longer involved. Once the call is connected it is assigned to a trapsport layer protocol such as RTP. RTP is the actual protocol that delivers the voice call between endpoints. I  believe that the setup of your network, QoS, codecs etc... determine the voice quality of your system.
>
>
> ----- Forwarded Message -----
> From: Mitul Limbani <mitul at enterux.in>
> To: Tommy Cooper <tomcooper83 at yahoo.com>; Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
> Sent: Wednesday, May 22, 2013 3:23 PM
> Subject: Re: [asterisk-users] Stress testing Asterisk
>
>
>
> I have a question here.
>
> How can we test the quality of voice upon increasing the call load?
>
> Can we try passing a voice file using sipp and record the same in dial plan record application ? Is this reliable enough to simulate near real world scenario?
>

Once upon a time, we set out to create exactly this for testing 
asterisk.  Our goal would have been to run the test every week, 
comparing the results from the previous week, to make sure asterisk's 
performance was not getting worse as new commits happened.

We came up with the idea of loading testing asterisk using SIPp or some 
other dialer, then determining at what point asterisk would start 
failing (performance).  We decided the point of failure was quality of 
audio, since it is usually the first thing to go (even though call 
control still works).

It took a while, but with the help of Leif, we found a tool to analyse 
audio streams (using MOS score[1]).  Basically, you take the original 
audio file, play it across the network, then record the other side. 
Then, comparing the two files via Aqua, you get your MOS score.

If the score was less then x, you knew asterisk was hitting a 
performance limit.  Track that over time and concurrent calls, you have 
your metrics.

[1] http://www.sevana.fi/aqua.php

-- 
Paul Belanger | PolyBeacon, Inc.
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