package: asterisk16-codec-opus

Name:
asterisk16-codec-opus
Version:
20171009-1
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. \\ \\
Installed size:
5kB
Dependencies:
libc, libssp, libopus, asterisk16
Categories:
network---telephony
Repositories:
telephony
Architectures:
aarch64_cortex-a53, aarch64_cortex-a72, aarch64_generic, arc_arc700, arc_archs, arm_arm1176jzf-s_vfp, arm_arm926ej-s, arm_cortex-a15_neon-vfpv4, arm_cortex-a7_neon-vfpv4, arm_cortex-a8_vfpv3, arm_cortex-a9, arm_cortex-a9_neon, arm_cortex-a9_vfpv3, arm_fa526, arm_mpcore, arm_mpcore_vfp, arm_xscale, i386_pentium4, mips_24kc, mips_mips32, mipsel_24kc, mipsel_24kc_24kf, mipsel_74kc, mipsel_mips32, powerpc_464fp, powerpc_8540, x86_64
OpenWrt release:
OpenWrt-19.07.0
File size:
6kB
License:
GPL-2.0
Maintainer:
Jiri Slachta
Bug report:
Bug reports
Source code:
Sources
This website uses cookies. By using the website, you agree with storing cookies on your computer. Also you acknowledge that you have read and understand our Privacy Policy. If you do not agree leave the website.More information about cookies
  • Last modified: 2021/09/15 21:51
  • by 127.0.0.1