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

jesse chat2jesse at gmail.com
Thu Feb 3 02:19:03 MSK 2011

You can have SIPP UAC plays Rtp , SIPP UAS Echo back , then compare number
of packets.
On Feb 2, 2011 10:34 AM, "David Ponzone" <david.ponzone at ipeva.fr> wrote:
> 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
> 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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