27 jan 2010 kl. 16.30 skrev Victor Pascual Avila: > 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.
...and some sends just a RTCP BYE without any report, which is against the RFC. Hello, SNOM! :-) I find it buggy to be able to place calls and have no RTCP reports sent at all, just because an internal timer doesn't go off. But it seems like most of the major hard phones behave this way. Well, Asterisk may lead the way then :-) /O _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
