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Codec Opus - Asterisk Project - Asterisk Project Wiki
https://wiki.asterisk.org/wiki/display/AST/Codec+Opus
The Opus codec for Asterisk exposes a few configuration options that allow adjustments to be made to the encoder. The following options can be used to define custom format types within the codecs.conf file. These custom format types can then be specified in the "allow" line of an endpoint. Must be of type "opus".
Opus Codec for Asterisk - GitHub
https://github.com/wazo-platform/wazo-codec-opus-open-source
Opus is the default audio codec in WebRTC. Therefore, you can use Mozilla Firefox or Google Chrome via SIP over WebSockets in Asterisk. However, many traditional apps (SIP over UDP) added Opus as well. Simply add allow=opus in your configuration file sip.conf and the SIP channel driver chan_sip is able to negotiate Opus via SDP.
Configuring the Opus Encoder for Asterisk ⋆ Asterisk
https://www.asterisk.org/configuring-opus-encoder-asterisk/
The recently announced Opus codec for Asterisk exposes a few configuration options that allow you to manipulate the encoder for your particular setup. These options can be set within codecs.conf. They are useful for customizing a format type that can then be specified on the “allow” line of an endpoint.
Asterisk 16 Configuration_codec_opus - Asterisk Project ...
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_codec_opus
Codec opus module for Asterisk options Configuration Option Reference Configuration Option Descriptions packet_loss Can be any number between 0 and 100 (inclusive). Higher values result in a loss resistant behavior, however this has a cost on the quality (dependent upon a given bitrate). complexity Can be any number between 0 and 10, inclusive.
Asterisk 13 Opus Codec - Asterisk Support - Asterisk Community
https://community.asterisk.org/t/asterisk-13-opus-codec/68179
Support for Opus was added as of Asterisk 13.13.0. There were no changes to the codec implementation itself.
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