[Asterisk-Dev] Jitter Buffer

Lubomir Christov voip at minitelecom.org
Tue Jan 20 00:02:03 MST 2004

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

if (abs(rtp->lastts - pred) < 640)
                        rtp->lastts = pred;

Change the 640 to something larger, perhaps even as high as 8000 (one


Chris Wilson wrote:

>Hi all,
>I've done a bit of searching in the Asterisk source and on Google, and it 
>seems to me that either there is no jitter buffer for SIP, or I can't find 
>it. However, there is a jitter buffer for IAX 1 and 2.
>I'm running SIP (phones don't support IAX) over a wireless network shared
>with data traffic, which causes some increase in jitter when large volumes 
>of data are transferred. Would it be useful to have a SIP jitter buffer, 
>is it something I could add, and would Digium be interested?
>Could it be abstracted out of the IAX 1/2 jitter buffer code, which seems 
>to be roughly duplicated between the two modules? If so, does anyone have 
>any hints on what sort of interface I should provide to it, and where I 
>should put my source file?
>Is there an Asterisk hacker's guide to explain how Asterisk does memory 
>allocation, callbacks, etc. and standards for data types, or should I just 
>try to copy what's present as best I can?
>Is anyone willing to help me test and debug this?
>Cisco VoIP phones seem to have particularly good handling of jitter and 
>packet loss. For example, I deliberately created 10% packet loss, and the 
>perceived call quality (to me) was about 99%. At 20% packet loss it was 
>about 90%. Budgetone phones seem to handle this very badly in my 
>experience, with every packet lost (or late) generating an annoying click 
>and a gap in sound. Can anyone else confirm or deny?
>Can anyone point me to a standard for how reconstruction of lost packets
>should be done, so I can look at implementing it for the jitter buffer?
>Cheers, Chris.

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