[Asterisk-Dev] RTP timing issues

[Bart Coppens]

Hi,

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?

The WIKI will be updated accordingly

Best regards,
Bart Coppens
Email work: coppens.b at ccafrique.net
Email private: coppens_b at hotmail.com

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