[Freeswitch-users] Voice quality monitoring via loopback on sip endpoint

[David Ponzone]

Well, not really :)
The signal processing involved is not obvious, mainly because what you want is to compare 2 signals, keeping in mind that the human ear is the target (MOSPESQ is normally computed using a group of real people, each of them scoring the quality of the audio).
I don't know any software doing that for free. I am pretty sure in the commercial world, there should be something available.

Check http://www.phoenixdatacom.com/voip.html

Another solution is to use an external quality insurance service to do that for you. We have one like that in France. You give them an entry point in your network where to inject the audio, an exit point where to record the audio they injected, and they give you your MOSPESQ score.
The main reason to go that way, I think, is that the score they give you may be used as a reference against competitors.

And finally, you may probably build your own benchmark reading this:

http://www.freeswitch.org/node/297

I guess Kristian Kielhofner could tell you more about it.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

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Le 02/02/2011 à 20:00, Juan Antonio Ibañez Santorum a écrit :

> Could you give any info about which tools to use?
> 
> Regards
> 
> 2011/2/2 David Ponzone <david.ponzone at ipeva.fr>
> A common way to test audio quality is to have a device (sipp for instance) calling your FS and playing a known audio file.
> FS would then record it and then, by comparing the recorded and the original one and using some signal processing, you can compute a MOSPESQ figure.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.
> 
> 
> 
> 
> Le 02/02/2011 à 19:25, Marc De Corny a écrit :
> 
>> Hi all,
>> I am thinking of a way of testing the quality of a voice call. 
>> 
>> Ideally i would like to create a call from FS with a loopback parameter that tells phone to answer the call automatically and loopback the RTP. This is a common functionality in most phoned (for example cisco)
>> The sent and received packets could be compared for jitter, latency and packet loss and a result extracted.
>> 
>> Does this already exist? I know there some specific tool that accomplish this bit they are expensive and awkward to use. Starting and retrieving the result from an api would be fantastic.
>> 
>> Any ideas are welcome.
>> Thanks
>> Marc
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