Hi Olle, On Wed, Jan 27, 2010 at 4:23 PM, Olle E. Johansson <[email protected]> wrote: > 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?
Most of the implementations I've worked with don't send a final RTCP report, just a goodbye packet. Cheers, -- Victor Pascual Ávila _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
