On a lab setup, I can see an Asterisk 11 system is correctly receiving and displaying (sip show channelstats) incoming RTCP reports but not any report to the other end.
Searching through *.sample files does show much. This highlighted I still have a lot to learn on RTCP.
My setup is:
Asterisk 13 <-----------> Asterisk 11
In my tests, Asterisk 13 box calls Asterisk 11 box.
1. In Asterisk 11 box, there is no RTCP trace in incoming INVITE’s SDP as if RTCP is implied. RFC3605 mentions rtcp attributes but also implicit communication. Is it common practice for SIP devices to send RTCP reports on implicit ports (ie port computed from RTP port) without explicit mention in INVITE message or more generally without negotiating with the other end ?
2. Is there a way to toggle on or off RTCP sending in Asterisk for all SIP
peers or PJSIP endpoints ?
3. Is there a way to toggle on or off RTCP sending in Asterisk for a given SIP peer or PJSIP endpoint ?
4. Same as 2 and 3 but with RTCP receiving (ie dumping incoming RTCP
5. Am I correct in thinking RTCP stats are meaningful with previous or next hop ? In other words, if a communication chain such as A <--> B <--> C <-->
D, packet lost between A and B are not reported to C.
6. Does RTPProxy support RTCP XR stats ?