[Asterisk-Dev] How to trigger PLC in a codec in SIP channels

[Gouri Johannsen]

I know that Jitter buffer and PLC implementation is not yet done in the 
SIP/RTP channels.  But I was wondering is there an easy way to indicate to 
the codec that a frame did not arrive on time with the current 
implementation (i.e. without JB?)

Along the same lines, anybody has an idea when the new jitterbuffer 
implementation will be ready for SIP?

Thanks,
Gouri.