[Asterisk-Dev] SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev

[Olle E. Johansson]

Tonight, we've been having some discussion about the Jitter buffer. The
developer feels that this is far from ready and we have no reason to
include untested code in 1.2 stable.

We've decided to move this patch for inclusion in the new development
tree after the 1.2 release.

This does not mean that you are free - we still need help in testing
this and finding the current memory leaks. Valgrind pro's - come to rescue!

I am sorry that we could not reach the target with this patch. At the
same time, I'm proud that Asterisk.org as a project can take this
decision and move forward with quality in mind for the stable release.

/Olle