[Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev

[Zoa]

We did a lot of testing on the jitterbuffers, an i can only confirm that
(both the adaptive version and the fixed version) use up a lot of cpu
resources, (memory is not so bad).
When you go to 99% cpu utilization, there is some memory leak (as long
as you stay under 95% there is no memory leak). We have no clue if this
is in our code or in the main asterisk code, we do know its not in the
jitter buffer code itself as we have it with both the stevek version as
the slav version. (The slav version was created to make sure the problem
was not in the stevek version).

So you definately need to be able to turn it off (you can already, but
not per user/peer).

Zoa.
-----

http://www.asteriskguru.com



Steve Kann wrote:

> Matt wrote:
>
>>I'm going to assume it's still a patch based on what I've seen elsewhere.
>>As far as responding to arnaud I have to say "uhh why" as well.  Why
>>would you not want the jitter buffer on all the time for all users?
>>It really can only help (if it works correctly) which so far it does
>>indeed seem to.
>>
>>
> The present patch implementes a fixed length buffer of some sort.  So,
> even if it's working right, it will add some delay and resource usage
> (CPU, memory).  So, some people might not want to use it.
>
> The original adaptive buffer could theoretically be a lot better about
> not adding excessive delay when no jitter is present, but it still
> adds CPU and some memory overhead.
>
> Both implementations add complexity to the media path, and can
> theoretically cause issues, especially if the sender is sending
> incorrect timestamps, or has other bugs.  Ideally, we'd be able to
> diagnose and work around these bugs in the remote clients, but that
> takes time and effort, and the quickest short-term solution might be
> to just turn it off.
>
> So, I could see why people would want the ability to turn it off.
>
> -SteveK
>
>>On 8/26/05, Matt <mhoppes at gmail.com> wrote:
>>
>>
>>>I'm confused now.. is the sip jitter buffer currently available in the
>>>CVS-HEAD or does it still need to be patched in?
>>>
>>>On 8/26/05, Arnaud <arno at directcentrex.com> wrote:
>>>
>>>
>>>>Eric Wieling aka ManxPower wrote:
>>>>
>>>>
>>>>
>>>>>Kevin P. Fleming wrote:
>>>>>
>>>>>
>>>>>
>>>>>>AEL (pbx_ael.c) will be included in the 1.2 release, but will be
>>>>>>clearly marked in the UPGRADE.txt file as experimental. If we don't
>>>>>>include it in 1.2, we won't get very many testers other than the
>>>>>>brave souls who will continue to run the development branch :-)
>>>>>>
>>>>>>
>>>>>The same could be said about the SIP jitter buffer.
>>>>>
>>>>>
>>>>>
>>>>I agree, and also it will be better to be able to enable SIP jitter per user
>>>>--
>>>>
>>>>Arnaud Pignard (apignard at frontier.fr)
>>>>Standard : + 33 1 70 71 50 00 - Fax : +33 1 70 71 50 60
>>>>MSN : stormfr at hotrmail.com - ICQ : 20946060
>>>>
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