Package: asterisk-codec-opus
Version: 2021-11-01-20522fbc-2
Depends: libc, asterisk, libopus
Source: feeds/telephony/net/asterisk-opus
SourceName: asterisk-opus
License: GPL-2.0
LicenseFiles: LICENSE
Section: net
SourceDateEpoch: 1689712895
Maintainer: Jiri Slachta <jiri@slachta.eu>
Architecture: mips_mips32
Installed-Size: 5585
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.
