[Asterisk-Users] SIP jitter?

[Steve Kann]

Peter Svensson wrote:

>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.
>>>      
>>>
>>is this in HEAD yet?
>>    
>>
>
>See http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002532
>
>There isn't even any code for SIP yet. However the iax integration works 
>wonders for a link with just a bit of packet loss and jitter. Voice 
>conversations are nice and crisp and without the pops associated with lost 
>packets or growth of the jitter buffer.
>  
>

Glad it's working for you, Peter..

I don't want to mention names, but somebody has been working on 
integration into the RTP stack and chan_sip, based on what I wrote in 
the bug, and some IRC conversations..

-SteveK