<br><font size=2><tt>> I have a set of two patches for asterisk, at<br>
> <br>
> http://bugs.digium.com/bug_view_page.php?bug_id=0002532<br>
> <br>
> Which implement a new jitterbuffer, and packet loss concealment, for
<br>
> IAX2 channels. I've also looked at how to implement this for
SIP and <br>
> other RTP-based VoIP channels; it shouldn't be hard..<br>
> <br>
> What this means is much smoother jitterbuffer behavior, and loss <br>
> concealment for asterisk. With this code, you can have a conversation
<br>
> over a link with 10% packet loss, and hardly notice..<br>
> <br>
> Thanks to Steve Underwood, for his great Generic PLC algorithm, and
<br>
> Steve Davies, for moral support (where Moral support is loosely defined
<br>
> as "I'll work on this with you on day X", and then not appearing
on day <br>
> X :) <just kidding, Steve>).<br>
> <br>
> Anyway, there's still time to change the decision on how this is <br>
> integrated into asterisk, jitterbuf.c/h need a bunch of cleanup, and
<br>
> there is still more "plumbing" and testing to do (i.e. ensuring
we do <br>
> the right thing when calls are transferred, handling trunk mode, <br>
> disabling the jitterbuffer when we bridge to another VoIP channel,
etc), <br>
> but what's there seems to work for the simple case.<br>
> <br>
> Help and comments appreciated..<br>
> <br>
> <br>
> -SteveK<br>
</tt></font>
<br>
<br><font size=2><tt>SteveK,</tt></font>
<br>
<br><font size=2><tt>This is very exciting. To get this implemented into
stable would be awesome. Thanks for the hard work!</tt></font>
<br>
<br><font size=2><tt>-Ron</tt></font>