[asterisk-dev] [svn-commits] twilson: trunk r275998 - in /trunk/channels: chan_sip.c sip/include/dialog.h

[Terry Wilson]

> I don't agree fully with this change, Terry. There is still some RTCP activity that may happen after the BYE is sent or received. We should stay open for a few seconds, but possibly not 32 secs (or T1*64 as it really is). If we already have RTCP BYE, then it's fine to close the RTP. If not, we should propably hang around and wait for a few seconds of timeout, then close it. We will need those final RTCP reports for my code in pinefrog. If just close without receiving the final report from the other end (in combination with RTCP bye) vital statistics about QoS will be gone.

In fact, it also caused some crashing, so I reverted it. :-( Changes are never as easy as I hope they will be, apparently. I'll try to get an improved patch written and actually put up on reviewboard this time. Sorry!

Terry