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

[Steve Kann]

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
>>>
>>>Frontier Online - Opérateur Internet - http://www.frontier.fr
>>>Direct Centrex - Opérateur Télécom sur IP - http://www.directcentrex.com
>>>Direct Nom - Registrar de nom de domaine - http://www.directcnom.com
>>>
>>>_______________________________________________
>>>Asterisk-Dev mailing list
>>>Asterisk-Dev at lists.digium.com
>>>http://lists.digium.com/mailman/listinfo/asterisk-dev
>>>To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>>>
>>>      
>>>
>_______________________________________________
>Asterisk-Dev mailing list
>Asterisk-Dev at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-dev
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>  
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20050826/cda1ab5f/attachment.htm