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. \\ \\
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