[Asterisk-Dev] Jitter Buffer

Chris Wilson chris at netservers.co.uk
Tue Jan 20 02:38:24 MST 2004


Hi Lubo,

> Yes, SIP don't have a jitter buffer, but if you increase this value in 
> rtp.c  it will help you :)

What does this value do? Is it a fixed delay between receiving the packet 
and writing it to another channel? If so, perhaps it is a jitter 
buffer, and it just needs to be auto-adaptive?

> if (abs(rtp->lastts - pred) < 640)
>                         rtp->lastts = pred;
> 
> Change the 640 to something larger, perhaps even as high as 8000 (one
> second).

Units are samples (8kHz)?

Cheers, Chris.
-- 
_  __ __     _
 / __/ / ,__(_)_  | Chris Wilson -- UNIX Firewall Lead Developer |
/ (_  ,\/ _/ /_ \ | NetServers.co.uk http://www.netservers.co.uk |
\__/_/_/_//_/___/ | 21 Signet Court, Cambridge, UK. 01223 576516 |




More information about the asterisk-dev mailing list