[Asterisk-Users] SIP jitter?

[Peter Svensson]

On Tue, 8 Feb 2005, Roy Sigurd Karlsbakk wrote:

> >> how can I tune SIP jitter? is it possible today in asterisk?
> >
> > I assume you are asking for how to alleviate the effects of jitter on 
> > the
> > RTP audio streams initated by SIP? Asterisk currently only has a jitter
> > buffer for IAX, not for RTP streams. There are pland for the next
> > generation jitter buffer code to hook into RTP as well.
> >
> > There is an entry on the bug tracker that touches on this topic.
> 
> is this in HEAD yet?

See http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002532

There isn't even any code for SIP yet. However the iax integration works 
wonders for a link with just a bit of packet loss and jitter. Voice 
conversations are nice and crisp and without the pops associated with lost 
packets or growth of the jitter buffer.

Peter