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Asterisk 13 transcoding module: Opus

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Several SIP ↔︎ DECT bases exist on the market, and the latest ones allow the Opus Codec: - Grandstream DP750 (tested) - Grandstream DP752 - Panasonic KX-TGP700 (tested) - Poly...

When Asterisk started the call but the remote party sends a re-INVITE on the SIP layer, the call might face one-way audio. One such a call scenario is [call hold](https://tools.ietf.org/html/rfc5359#section-2.1),...

[Snom](https://www.snom.com/) added the Opus Codec for their Desk Phones in the year [2014](http://forum.snom.com/?showtopic=10431&p=43487). At that time [RFC 7587](https://tools.ietf.org/html/rfc7587) was not final and the Snom implementation based on a draft from...

Hi, I'm using vp8 format to write frames into a file. It works great but I'm having issues to read this file or encode it into a playable format. Do...

for 100 calls it using 3 cores. can we optimize it for the same?

As promised, here's a simple PR that adds a call to the `OPUS_SET_COMPLEXITY` control in order to set up the complexity of the encoder when creating it. The default complexity...

lin(mono) opus(mono) lin(stereo) -> opus(stereo) lin(stereo) opus(stereo) Removed res_format_attr_opus.c -> default from Asterisk Master

Hi, the patch is working, but is crashing while compiling [CC] format_ogg_opus_open_source.c -> format_ogg_opus_open_source.o In file included from format_ogg_opus_open_source.c:36: /root/here1/asterisk-22.5.2/include/asterisk/format_cache.h:321:63: warning: ‘struct ast_codec’ declared inside parameter list will not be...