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Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration

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These aren't the files you're looking for...

Because Digium has now released their own codec_opus for Asterisk 13 and Asterisk 14, I don't see a compelling reason for anyone to be using this one.

The code is still here, but if you are looking for continued development I suggest you take a look at Alexander Traud's fork.

If you're concerned that Digium's codec_opus "phones home," just set up dnsmasq with the following config:

local=/stats.asterisk.org/

And you're done. You could probably also do something with /etc/hosts, but using dnsmasq is cleaner.

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Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration

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