While doing work to improve Asterisk's support of RTCP I've gotten into a lot of RTCP issues. Some phones send just zeroes as NTP time stamps, thus giving bad input to RTT calculations and there are variations on how often the phones send RTCP reports - if at all.
Now to my question: I get reports from the other end in regards to packet loss in the stream I send over there. With some phones only sending reports every 30th second, I might get no report or have an up to 29 seconds old report at hangup. For Asterisk, I'm making sure we're sending a final report at hangup, in combination with RTCP BYE. Is that good practise, to force an extra report at the end of the call or is it just confusing for the receiver? /O _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
