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

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Are there any plans to support fec and packet loss percentage in asterisk 11 using this patch? Thanks.

Digium now [provides codec_opus for both Asterisk 13 and 14](http://downloads.digium.com/pub/telephony/codec_opus/). Given that, I don't see a compelling reason to continue with this fork. If anyone feels differently, let me know.

Testing the Opus codec for Asterisk, working fine with default sampling 48khz. The default sampling is fixed at 48khz and is there any option to use 16khz > Asterisk :...

I successfully installed patch and checks it work. IT works fine but asterisk cli sends this messages when rtp sended. [2015-02-05 00:53:01] WARNING[58036][C-00000005]: codec.c:363 ast_codec_samples_count: Unable to calculate samples for...