Source-Makefile: package/feeds/telephony/asterisk-opus/Makefile

Package: asterisk16-codec-opus
Submenu: Telephony
Version: 20171009-1
Depends: +libc +GCC_LIBSSP:libssp +USE_GLIBC:librt +USE_GLIBC:libpthread +libopus asterisk16
Conflicts: 
Menu-Depends: 
Provides: 
Build-Variant: asterisk16
Section: net
Category: Network
Repository: telephony
Title: Opus codec support
Maintainer: Jiri Slachta <jiri@slachta.eu>
Source: asterisk-opus-20171009.tar.xz
License: GPL-2.0
LicenseFiles: LICENSE
Type: ipkg
Description:   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.
https://github.com/traud/asterisk-opus
Jiri Slachta <jiri@slachta.eu>
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