[Asterisk-Dev] RTP timing issues
Bart Coppens
coppens_b at hotmail.com
Thu Oct 7 05:27:24 MST 2004
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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