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codec_opus: Codec opus module for Asterisk

This configuration documentation is for functionality provided by codec_opus.

Configuration File: codecs.conf

[opus]: Codec opus module for Asterisk options

Configuration Option Reference

Option Name Type Default Value Regular Expression Description
application Custom voip false Encoder's application type.
bitrate Custom auto false Encoder's bit rate.
cbr Boolean no false Encoder's constant bit rate value.
complexity Integer 10 false Encoder's computational complexity.
dtx Boolean no false Encoder's discontinuous transmission value.
fec Boolean yes false Encoder's forward error correction value.
max_bandwidth Custom full false Encoder's maximum bandwidth allowed.
max_playback_rate Custom 48000 false Encoder's maximum playback rate.
packet_loss Integer 0 false Encoder's packet loss percentage.
signal Custom auto false Encoder's signal type.
type None false Must be of type 'opus'.

Configuration Option Descriptions

application
  • voip

  • audio

  • low_delay

bitrate

Can be any number between 500 and 512000 as well as one of the following opus values:

  • auto

  • max

cbr

True/False value where 0/false/no represents a variable bit rate and 1/true/yes is constant bit rate.

complexity

Can be any number between 0 and 10, inclusive. Note, 10 equals the highest complexity.

dtx

True/False value where 0/false/no represents disabled and 1/true/yes is enabled.

fec

True/False value where 0/false/no represents disabled and 1/true/yes is enabled.

max_bandwidth

Sets an upper bandwidth bound on the encoder. Can be any of the following:

  • narrow

  • medium

  • wide

  • super_wide

  • full

max_playback_rate

Any value between 8000 and 48000, inclusive. Although typically it should match one of the usual Opus bandwidths.

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).

signal

Aids in mode selection on the encoder:

  • auto

  • voice

  • music

Generated Version

This documentation was generated from Asterisk branch 22 using version GIT