软件包：asterisk-opus（13.7+20171009-2）
asterisk-opus 的相关链接
Debian 的资源：
下载源码包 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]
维护小组：
外部的资源：
- 主页 [github.com]
相似软件包：
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).
其他与 asterisk-opus 有关的软件包
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- dep: asterisk
- Open Source Private Branch Exchange (PBX)
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- dep: asterisk-1fb7f5c06d7a2052e38d021b3d8ca151
- 本虚包由这些包填实： asterisk
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- dep: libc6 (>= 2.4)
- GNU C 语言运行库：共享库
同时作为一个虚包由这些包填实： libc6-udeb
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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