[Asterisk-Dev] Jitter Buffer

Brian West brian at bkw.org
Mon Jan 19 23:12:33 MST 2004

> 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.

The jitter buffer is SIP->ZAP otherwise it doesn't exist.

> 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?

If you allow reinvites the phones should do that for you.

> 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?

Sounds like a good idea but if you allow reinvites then the stream won't
be going thru asterisk thus it won't need to worry with jitter buffers.

> 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?

Anything grandstream needs to be thrown in the lake.


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