[Asterisk-Dev] Re: [Asterisk-Users] RTP timing issues

[steve at daviesfam.org]

On Thu, 7 Oct 2004, Bart Coppens wrote:

> Some time ago, I announced a bounty to solve the issues with regards to 
> silence compression (chopped voice) and one way voice. To get this solved, 
> Asterisk should get the clocking from an internal source in a way that an 
> ouput stream can be generated without getting any RTP input.
> 
> Now my company is exposing a payment of 1000USD for this bounty. This 
> payment have to justified through an official invoice.
> 
> Can someone give me an indication if this can be achieved?


It can be achieved.

Steve