I’ve starting logging RTPAUDIOQOS statistics to CDR records in Master.csv and I wrote a program to parse and extract out those statistics. I think it will be very useful, assuming the data is accurate. But I’m not sure I fully understand the data. Here is an example:
“ssrc=358281685;themssrc=1432487772;lp=0;rxjitter=0.000482;rxcount=386;txjitter=12593.274200;txcount=0;rlp=6424412;rtt=6683.786029”
Questions:
-
rxjitter is “our calculated jitter” and txjitter is “reported jitter of the other end”. I assume RTP has some way to do this reporting back an forth. But which end is the other end? If I have a call that originates from Sip device 1 and calls Sip device 2, which one is it?
-
My guess is that these statistics only reflect the connection to the call originator. Is that correct? If so, are there statistics on the connection to the callee? Can I get at them?
-
Are there standard thresholds for jitter and lost packets that indicate that the audio is degrading or is unacceptable?
-
A couple of the values in the example above look bad (eg. rlp and txjitter). Can I really trust this?
Thanks.