[asterisk-dev] Interpreting RTPAUDIOQOS statistics

[Don Morrison]

I've starting logging RTPAUDIOQOS statistics to CDR records in Master.csv
and I wrote a program to parse and extract out those statistics (running
1.4). 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:

1) 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?

2) 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?

3) Are there standard thresholds for jitter and lost packets that indicate
that the audio is degrading or is unacceptable?

4) A couple of the values in the example above look bad (eg. rlp and
txjitter). Can I really trust this?

Thanks.

Don Morrison