[Asterisk-Dev] Jitter Buffer

[Chris Wilson]

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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