[Asterisk-Dev] Jitter Buffer

[Chris Wilson]

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.
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