asterisk users - Feb 2005 - SIP jitter?

hi

how can I tune SIP jitter? is it possible today in asterisk?

ryo

On Tue, 8 Feb 2005, Roy Sigurd Karlsbakk wrote:
> how can I tune SIP jitter? is it possible today in asterisk?
I assume you are asking for how to alleviate the effects of jitter on the 
RTP audio streams initated by SIP? Asterisk currently only has a jitter 
buffer for IAX, not for RTP streams. There are pland for the next 
generation jitter buffer code to hook into RTP as well. 

There is an entry on the bug tracker that touches on this topic.

Peter

>> how can I tune SIP jitter? is it possible today in asterisk?
>
> I assume you are asking for how to alleviate the effects of jitter on 
> the
> RTP audio streams initated by SIP? Asterisk currently only has a jitter
> buffer for IAX, not for RTP streams. There are pland for the next
> generation jitter buffer code to hook into RTP as well.
>
> There is an entry on the bug tracker that touches on this topic.
thanks

is this in HEAD yet?

roy

Well im using cvs stable in production, but i know several of the
carriers out there are using cvs head, because its the only one that has
realtime...

Anyway, cvs head is not testing. If you want to test the jitter buffer,
download the patches compile them and see what happens, then report the
results on mantis.
If everyone says it doesnt seem to break anything, it will make it to
cvs head, if its obviously broken it wont make it to cvs head.
I reported it to cause massive deadlocks when using it with several
simultaneous calls...

This is how the * development people seem to work, and unless someone
wants to start a 3rd branch, thats how its going to stay i think.... I
didnt say i wouldnt like the jitter buffer to be in some kind of
prepatched asterisk tree, but it should be in a 3rd  (all really
experimental things should be)

I dont think we disagree btw :)

Joachim.


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