[Asterisk-Users] SIP jitter?

[Roy Sigurd Karlsbakk]

> 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.

Is there a reason why this isn't in HEAD?

roy