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[ Source: asterisk-opus  ]

Package: asterisk-opus (13.7+20171009-2)

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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).

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Download for all available architectures
Architecture Package Size Installed Size Files
amd64 14.9 kB74.0 kB [list of files]
arm64 14.6 kB61.0 kB [list of files]
armel 14.6 kB58.0 kB [list of files]
armhf 14.0 kB54.0 kB [list of files]
i386 15.1 kB66.0 kB [list of files]
mips64el 14.0 kB64.0 kB [list of files]
mipsel 13.9 kB56.0 kB [list of files]
ppc64el 15.7 kB221.0 kB [list of files]
s390x 14.5 kB61.0 kB [list of files]