[Asterisk-Users] What and When is the next version of Asterisk?

[Pavel Jezek]

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 <pavel.jezek at i.cz> 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.
>