Logging Asterisk RTCP statistics

Most SIP devices are capable of sending statistics about the quality of voice calls in the RTCP packages. Audio quality statistics can be a valuable tool for tracking and alerting on potential problems. Asterisk has some rudementary support for capturing this data but it is not well documented.

note: this posting is about the SIP RTCP statistics only

The way it works: At the conclusion of a call, Asterisk (actually chan_sip) sets a channel the channel variable RTPQUDIOQOS which will contain a string that looks something like this:

ssrc=193193741;themssrc=1214799925;lp=0;rxjitter=0.001591;rxcount=175;txjitter=0.000107;txcount=184;rlp=0;rtt=0.068000

Unfortunately, at the moment it seems that they are nearly useless. Due to several bugs in the way the statistucs are gathered the numbers are not useful.