How to Install and Uninstall asterisk-opus Package on Ubuntu 21.10 (Impish Indri)
Last updated: December 23,2024
1. Install "asterisk-opus" package
This is a short guide on how to install asterisk-opus on Ubuntu 21.10 (Impish Indri)
$
sudo apt update
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$
sudo apt install
asterisk-opus
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2. Uninstall "asterisk-opus" package
In this section, we are going to explain the necessary steps to uninstall asterisk-opus on Ubuntu 21.10 (Impish Indri):
$
sudo apt remove
asterisk-opus
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$
sudo apt autoclean && sudo apt autoremove
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3. Information about the asterisk-opus package on Ubuntu 21.10 (Impish Indri)
Package: asterisk-opus
Architecture: amd64
Version: 13.7+20171009-2
Priority: extra
Section: universe/comm
Origin: Ubuntu
Maintainer: Ubuntu Developers
Original-Maintainer: Debian VoIP Team
Bugs: https://bugs.launchpad.net/ubuntu/+filebug
Installed-Size: 74
Depends: asterisk, asterisk-1fb7f5c06d7a2052e38d021b3d8ca151, libc6 (>= 2.14)
Filename: pool/universe/a/asterisk-opus/asterisk-opus_13.7+20171009-2_amd64.deb
Size: 12632
MD5sum: 4a3d0c841e573c8d06f0f2b9357ba401
SHA1: d401d0cbf7f1189d37f378011d306972248be911
SHA256: 6dc68073b5bcfbe36242caaa382652ae33ee2e5243b4caa6c62a79b2b9fe8a08
SHA512: 7fb640353892481aca2c7e6df950123b50598b453d1e4c64bbd616641f4209cb5ac673e24ecf7412d7b84fdd5389cf077df046e1f231d6ab17b082f3abdad332
Homepage: https://github.com/traud/asterisk-opus/
Description-en: opus module for Asterisk
Module for the Asterisk open source PBX which allows you to use the
Opus audio codec.
.
Opus is the default audio codec in WebRTC. WebRTC is available in
Asterisk via SIP over WebSockets (WSS). Nevertheless, Opus can be used
for other transports (UDP, TCP, TLS) as well. Opus supersedes previous
codecs like CELT and SiLK. Furthermore in favor of Opus, other
open-source audio codecs are no longer developed, like Speex, iSAC,
iLBC, and Siren. If you use your Asterisk as a back-to-back user agent
(B2BUA) and you transcode between various audio codecs, one should
enable Opus for future compatibility.
.
Opus is not only supported for pass-through but can be transcoded as
well. This allows you to translate to/from other audio codecs like
those for landline telephones (ISDN: G.711; DECT: G.726-32; and HD:
G.722) or mobile phones (GSM, AMR, AMR-WB, 3GPP EVS).
Description-md5: f94ac1b4a8b93eb800a8eee913feef75
Architecture: amd64
Version: 13.7+20171009-2
Priority: extra
Section: universe/comm
Origin: Ubuntu
Maintainer: Ubuntu Developers
Original-Maintainer: Debian VoIP Team
Bugs: https://bugs.launchpad.net/ubuntu/+filebug
Installed-Size: 74
Depends: asterisk, asterisk-1fb7f5c06d7a2052e38d021b3d8ca151, libc6 (>= 2.14)
Filename: pool/universe/a/asterisk-opus/asterisk-opus_13.7+20171009-2_amd64.deb
Size: 12632
MD5sum: 4a3d0c841e573c8d06f0f2b9357ba401
SHA1: d401d0cbf7f1189d37f378011d306972248be911
SHA256: 6dc68073b5bcfbe36242caaa382652ae33ee2e5243b4caa6c62a79b2b9fe8a08
SHA512: 7fb640353892481aca2c7e6df950123b50598b453d1e4c64bbd616641f4209cb5ac673e24ecf7412d7b84fdd5389cf077df046e1f231d6ab17b082f3abdad332
Homepage: https://github.com/traud/asterisk-opus/
Description-en: opus module for Asterisk
Module for the Asterisk open source PBX which allows you to use the
Opus audio codec.
.
Opus is the default audio codec in WebRTC. WebRTC is available in
Asterisk via SIP over WebSockets (WSS). Nevertheless, Opus can be used
for other transports (UDP, TCP, TLS) as well. Opus supersedes previous
codecs like CELT and SiLK. Furthermore in favor of Opus, other
open-source audio codecs are no longer developed, like Speex, iSAC,
iLBC, and Siren. If you use your Asterisk as a back-to-back user agent
(B2BUA) and you transcode between various audio codecs, one should
enable Opus for future compatibility.
.
Opus is not only supported for pass-through but can be transcoded as
well. This allows you to translate to/from other audio codecs like
those for landline telephones (ISDN: G.711; DECT: G.726-32; and HD:
G.722) or mobile phones (GSM, AMR, AMR-WB, 3GPP EVS).
Description-md5: f94ac1b4a8b93eb800a8eee913feef75