Steve Kann has posted details of the latest patch added to the bugtracker:
I have a set of two patches for asterisk, at
Which implement a new jitterbuffer, and packet loss concealment, for IAX2 channels. I've also looked at how to implement this for SIP and other RTP-based VoIP channels; it shouldn't be hard..
What this means is much smoother jitterbuffer behavior, and loss concealment for asterisk. With this code, you can have a conversation over a link with 10% packet loss, and hardly notice..
Thanks to Steve Underwood, for his great Generic PLC algorithm, and Steve Davies, for moral support (where Moral support is loosely defined as "I'll work on this with you on day X", and then not appearing on day X :)
Anyway, there's still time to change the decision on how this is integrated into asterisk, jitterbuf.c/h need a bunch of cleanup, and there is still more "plumbing" and testing to do (i.e. ensuring we do the right thing when calls are transferred, handling trunk mode, disabling the jitterbuffer when we bridge to another VoIP channel, etc), but what's there seems to work for the simple case.
Help and comments appreciated..