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

Package: asterisk-codec-opus
Submenu: Telephony
Version: 2021.11.01~20522fbcd3fdf6f0adb20602d096d14cd69055e8-r2
Depends: +libc asterisk +libopus
Conflicts: 
Menu-Depends: 
Provides: 
Section: net
Category: Network
Repository: base
Title: Opus codec support
Maintainer: Jiri Slachta <jiri@slachta.eu>
Source: asterisk-opus-2021.11.01~20522fbcd3fdf6f0adb20602d096d14cd69055e8.tar.zst
License: GPL-2.0
LicenseFiles: LICENSE
URL: https://github.com/traud/asterisk-opus
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.
@@

Package: asterisk-format-ogg-opus
Submenu: Telephony
Version: 2021.11.01~20522fbcd3fdf6f0adb20602d096d14cd69055e8-r2
Depends: +libc asterisk +libopus +libopusfile +libopusenc
Conflicts: 
Menu-Depends: 
Provides: 
Section: net
Category: Network
Repository: base
Title: OGG/Opus audio support
Maintainer: Jiri Slachta <jiri@slachta.eu>
Source: asterisk-opus-2021.11.01~20522fbcd3fdf6f0adb20602d096d14cd69055e8.tar.zst
License: GPL-2.0
LicenseFiles: LICENSE
URL: https://github.com/traud/asterisk-opus
Type: ipkg
Description:   Reading and writing audio files in the OGG/Opus format.
@@


