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

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