[Freeswitch-users] Call quality statistics from CDR reports?

[d.campbell]

Greetings freeswitch community,

Freeswitch is an amazing piece of work, and I have been able to use it 
with great success in large-scale scenarios.  Much thanks to all the 
contributors!

One feature I'm not finding is a report on call quality for SIP/RTP 
calls.  I see that the CDRs contain several potentially relevant values 
including:

     rtp_audio_in_jb_packet_count
     rtp_audio_in_flush_packet_count
     rtp_audio_in_largest_jb_size

Unfortunately, there is no documentation on exactly what these values 
mean.  Would it be possible to compute packet loss and jitter statistics 
on the inbound call from these numbers or others?

Much thanks,
Doug Campbell