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

[David Ponzone]

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
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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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