[Asterisk-Dev] jitterbuf timestamps range

[Kris Boutilier]

> -----Original Message-----
> From: Steve Kann [mailto:stevek at stevek.com]
> Sent: Tuesday, May 10, 2005 6:45 AM
> To: Asterisk Developers Mailing List
> Subject: Re: [Asterisk-Dev] jitterbuf timestamps range
> 
> 
> Slav Klenov wrote:
> 
> > Hi all,
> >
> > According to the RTP RFC the timestamp field of an rtp 
> frame is 32 bit.
{clip}
> 
> I'm not sure why I chose long for all the timestamping, etc, in the 
> jitterbuffer, and I haven't really examined what, exactly, 
> will happen when they overflow, however overflow on a 32-bit system will happen 
> after 596 hours (24.85 days).   It seems like something to think about 
> eventually, but calls that last longer than 24 days seem 
> pretty unusual (and, who knows how asterisk itself handles this).
> 

With the current implementation of newjb, there will essentially be permanent loss of audio when the timestamps jump backwards massively (ie. the TS wraps). This was the essence of bugs 3965/4221.

Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District