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

[ Kaynak: 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).

asterisk-opus ile İlgili Diğer Paketler

• dep: asterisk

  Open Source Private Branch Exchange (PBX)

• dep: asterisk-1fb7f5c06d7a2052e38d021b3d8ca151

  sanal paketi sağlayan asterisk

• dep: libc6 (>= 2.4)

  GNU C Library: Shared libraries
  ayrıca şunun tarafından sağlanan bir sanal paket libc6-udeb

• dep: libopus0 (>= 1.1)

  Opus codec runtime library

• dep: libopusfile0 (>= 0.5)

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

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Tüm mevcut mimariler için indir

Mimari	Paket Boyutu	Kurulu Boyut	Dosyalar
armel	14,6 kB	58,0 kB	[dosya listesi]