[asterisk-dev] Jitter Buffer

[Kevin P. Fleming]

John Lange wrote:
> One simple example demonstrates why this makes sense; in the case where
> the sip channel is talking to an application inside Asterisk or
> otherwise connecting to a something which doesn't have jbenable option
> there is no way to activate the JB and therefore audio is jittered.

This is why we have already been discussing moving the jitter buffer
processing into ast_read() from ast_write(), so that it can be used for
non-channel destinations as well.