El 22/05/13 12:25, Paul Belanger escribió:
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 <mi...@enterux.in>
To: Tommy Cooper <tomcoope...@yahoo.com>; Asterisk Users Mailing List
- Non-Commercial Discussion <asterisk-users@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
Hi!
I haven't used it, but there is a quality test algorithm provided by
ITU.
http://stackoverflow.com/questions/2329403/how-to-start-a-voice-quality-pesq-test
http://en.wikipedia.org/wiki/PESQ
http://ieeexplore.ieee.org/xpl/articleDetails.jsp?tp=&arnumber=6043771&queryText%3DDevelopment+of+a+Speech+Quality+Monitoring+Tool+based+on+ITU-T+P.862
-----
CeSPI
Centro Superior para el Procesamiento de la Información
Universidad Nacional de La Plata
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