1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
|
Source: asterisk-opus
Section: comm
Priority: extra
Maintainer: Debian VoIP Team <pkg-voip-maintainers@lists.alioth.debian.org>
Uploaders: Jonas Smedegaard <dr@jones.dk>
Build-Depends: cdbs,
licensecheck,
debhelper,
dh-buildinfo,
asterisk-dev,
libopus-dev,
libopusfile-dev
Standards-Version: 3.9.8
Vcs-Git: https://anonscm.debian.org/git/pkg-voip/asterisk-opus.git
Vcs-Browser: https://anonscm.debian.org/cgit/pkg-voip/asterisk-opus.git
Homepage: http://github.com/traud/asterisk-opus/
Package: asterisk-opus
Architecture: any
Depends: ${cdbs:Depends},
${misc:Depends},
${shlibs:Depends}
Description: 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).
|
