On 5/25/06, Pavel Jezek <[EMAIL PROTECTED]> wrote:
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
PJ, I understand what you're saying, but playing devil's advocate, if this goes into /trunk it becomes part of the 1.4 release. The community is then tasked with supporting this feature, whether or not it is ready for prime time. If it's not ready, people then complain that features are in the release that "don't work" and "weren't tested". It's kind of a no-win situation. The best thing that can be done at this point to work towards trying to get this feature in is to have people test it in /trunk in their environments and report results back. If they are developers and can contribute to code improvements around it, that would be very much welcomed as well. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
