I understand that if two endpoints have the reinvite=yes option, the RTP stream will not pass through the Asterisk server.
Is there any way to determine whether a given call is actually happening directly between the two end points or Asterisk is relaying the stream?
The closest I have seen is at connect time if I am watching the console, I see this:
Packet2Packet bridging SIP/6010-08498fb8 and SIP/6000-0849cf20
which seems to indicate that (1) the stream is passing through Asterisk, and (2) no transcoding is happening. Is this correct?
Anyway, I would like to figure out the RTP path of an ongoing call, even if I wasn’t around at connect time.
-Ramon F Herrera