套件：asterisk-opus（13.7+20171009-2）
asterisk-opus 的相關連結
Debian 的資源：
下載原始碼套件 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]
維護小組：
外部的資源：
- 主頁 [github.com]
相似套件：
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).
其他與 asterisk-opus 有關的套件
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- dep: asterisk
- Open Source Private Branch Exchange (PBX)
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- dep: asterisk-1fb7f5c06d7a2052e38d021b3d8ca151
- 本虛擬套件由這些套件填實： asterisk
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- dep: libc6 (>= 2.14) [amd64]
- GNU C 函式庫：共用函式庫
同時作為一個虛擬套件由這些套件填實： 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