[asterisk-dev] sip/rtp jitterbuffer in 1.4?

[Roy Sigurd Karlsbakk]

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
roy at karlsbakk.net
(+47) 98013356
---
"Ubuntu: Ancient African word for ''I'm sick of compiling Gentoo''"  
-- Jeff Waugh