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35 lines
1.2 KiB
35 lines
1.2 KiB
Source-Makefile: feeds/telephony/net/asterisk-opus/Makefile
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Package: asterisk16-codec-opus
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Submenu: Telephony
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Version: 20171009-1
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Depends: +libc +GCC_LIBSSP:libssp +USE_GLIBC:librt +USE_GLIBC:libpthread +libopus asterisk16
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Conflicts:
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Menu-Depends:
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Provides:
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Build-Variant: asterisk16
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Section: net
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Category: Network
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Repository: base
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Title: Opus codec support
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Maintainer: Jiri Slachta <jiri@slachta.eu>
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Source: asterisk-opus-20171009.tar.xz
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License: GPL-2.0
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LicenseFiles: LICENSE
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Type: ipkg
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Description: Opus is the default audio codec in WebRTC. WebRTC is available in
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Asterisk via SIP over WebSockets (WSS). Nevertheless, Opus can be used
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for other transports (UDP, TCP, TLS) as well. Opus supersedes previous
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codecs like CELT and SiLK. Furthermore, in favor of Opus, other
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open-source audio codecs are no longer developed, like Speex, iSAC,
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iLBC, and Siren. If you use your Asterisk as a back-to-back user agent
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(B2BUA) and you transcode between various audio codecs, one should
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enable Opus for future compatibility.
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Opus is not only supported for pass-through but can be transcoded as
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well.
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https://github.com/traud/asterisk-opus
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Jiri Slachta <jiri@slachta.eu>
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