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

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