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

Juan Antonio Ibañez Santorum juanito1982 at gmail.com
Wed Feb 2 22:00:23 MSK 2011


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
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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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