[asterisk-users] jitterbuffer for chan_sip on asterisk 1.2

[Olle E. Johansson]

1 sep 2009 kl. 08.17 skrev James Mutuku:

> Hello,
>
> From http://www.voip-info.org/wiki/view/Asterisk+new+jitterbuffer,  
> it says that there For Asterisk 1.2 there was no jitterbuffer in the  
> RTP-based channels (i.e. chan_sip).
>
> I am using 1.2 and Ind there is no reason to upgrade. Are there any  
> developments on this?

Well, the development ended up being named Asterisk 1.4 which included  
jitter buffers. That's a good reason to update!

/O