hi all
the sip/rtp jitterbuffer from http://bugs.digium.com/view.php?id=3854
has been in the works for more than a year, and has been in testing
from the last patch since 1.2.1 or so. My testing shows it makes G.
711A work well with crystal-clear audio even on an overloaded
704/128kbps link. since asterisk is quite unusable for large-scale
itsp rollouts without a jitterbuffer for RTP-based protocols, SIP in
particular, it would be really nice to get slav's code into 1.4.
any idea if this'll happen? it really, really should......
thanks in advance
roy
--
Roy Sigurd Karlsbakk
[EMAIL PROTECTED]
(+47) 98013356
---
"Ubuntu: Ancient African word for ''I'm sick of compiling Gentoo''"
-- Jeff Waugh
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