I think, that sip/rtp jitterbuffer is one of the most wanted feature,
but because still not included in trunk too few peoples improving it...
what to try include this soon to trunk, and only if problems will be not
solved before 1.4 release candidate, remove out of asterisk 1.4 ...
also good candidate to 1.4 is new codec negotiation algorithm, seems be
actively maintained/finalized
http://bugs.digium.com/view.php?id=4825
PJ
BJ Weschke wrote:
On 5/25/06, Pavel Jezek <[EMAIL PROTECTED]> wrote:
... and because sip/rtp jitterbuffer implementation still isn't in
trunk, so will not be included in 1.4 release? :'(
They were working on it pretty actively on Tuesday, but they were
still having issues when someone tested it on the dev conf call. It
has until the end of the month to get in.
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