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

[ Source: asterisk-opus  ]

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

• dep: asterisk

  Open Source Private Branch Exchange (PBX)

• dep: asterisk-1fb7f5c06d7a2052e38d021b3d8ca151

  virtual package provided by asterisk

• dep: libc6 (>= 2.4)

  GNU Biblioteca C: Bibliotecas compartilhadas
  also a virtual package provided by libc6-udeb

• dep: libopus0 (>= 1.1)

  biblioteca de execução do codec Opus

• dep: libopusfile0 (>= 0.5)

  High-level API for basic manipulation of Ogg Opus audio streams

Download asterisk-opus

Download for all available architectures

Architecture	Package Size	Installed Size	Files
armhf	14.0 kB	54.0 kB	[list of files]