Bug#848560: ITP: asterisk-opus -- opus module for Asterisk

[Jonas Smedegaard]

Package: wnpp
Severity: wishlist
Owner: Jonas Smedegaard <dr at jones.dk>

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* Package name    : asterisk-opus
  Version         : 13.7+20161113
  Upstream Author : Lorenzo Miniero
* URL             : https://github.com/traud/asterisk-opus
* License         : GPL-2
  Programming Lang: C
  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).

Opus module is currently included as a patch to Asterisk itself, but
that complicates security tracking (see bug#847666), and same codebase
has recently grown support for building out-of-tree as well.

This package will be maintained in the VoIP team.

 - Jonas

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