Package: asterisk-opus (13.7+20171009-2)
Links for asterisk-opus
Debian Resources:
Download Source Package asterisk-opus:
- [asterisk-opus_13.7+20171009-2.dsc]
- [asterisk-opus_13.7+20171009.orig.tar.gz]
- [asterisk-opus_13.7+20171009-2.debian.tar.xz]
Maintainers:
External Resources:
- Homepage [github.com]
Similar packages:
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).
Other Packages Related to asterisk-opus
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- dep: asterisk
- Open Source Private Branch Exchange (PBX)
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- dep: asterisk-1fb7f5c06d7a2052e38d021b3d8ca151
- virtual package provided by asterisk
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- dep: libc6 (>= 2.14) [amd64]
- GNU C Library: Shared libraries
also a virtual package provided by libc6-udeb
- dep: libc6 (>= 2.17) [arm64]
- dep: libc6 (>= 2.4) [armhf, i386]
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- dep: libopus0 (>= 1.1)
- Opus codec runtime library
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- dep: libopusfile0 (>= 0.5)
- High-level API for basic manipulation of Ogg Opus audio streams
Download asterisk-opus
| Architecture | Package Size | Installed Size | Files |
|---|---|---|---|
| amd64 | 14.9 kB | 74.0 kB | [list of files] |
| arm64 | 14.6 kB | 61.0 kB | [list of files] |
| armhf | 14.0 kB | 54.0 kB | [list of files] |
| i386 | 15.1 kB | 66.0 kB | [list of files] |