Where to get the statistics
- Asterisk 13+ (PJSIP), during a call:
asterisk -rx "pjsip show channelstats" - Legacy chan_sip:
asterisk -rx "sip show channelstats" - After the call: the RTPAUDIOQOS variable (store it in the CDR with Set(CDR(qos)=${RTPAUDIOQOS}) in a hangup handler).
- FreeSWITCH: during a call
fs_cli -x "uuid_dump <uuid>", afterwards the rtp_audio_in_* variables in the XML or JSON CDR.
Reading the direction
“Receive” is what reaches your server: loss here means trouble on the path from the other party or carrier to you. “Send” is what you sent; your server learns its loss from the other side’s RTCP reports. If no RTCP arrives (some carriers do not send it), the send side stays empty; that is not an error.
Test your own line: VoIP quality test. Full call analysis from a capture: pcap analyzer. Work out MOS by hand: MOS calculator.