Although most descriptions of WebRTC tout its rich media sharing capabilities, most discussion centers on video and data sharing. As it happens, WebRTC’s audio codec, Opus (at one time called “Harmony”), is among the most remarkable of its features. A free audio codec in development since 2007 by Octasic, Mozilla and the Xiph Foundation in collaboration with Google, Microsoft’s Skype unit and Broadcom, Opus was recently ratified in 2012 by the IETF as RFC 6716. Moreover, after considering ten codecs, the IETF also reached “strong consensus” to adopt Opus as a mandatory-to-implement (MTI) codec for WebRTC.

Update: Since then there has been a push to adopt more legacy codecs to facilitate interworking. See http://tools.ietf.org/html/draft-marjou-rtcweb-audio-codecs-for-interop-00

In any case, to handle the demands of WebRTC, Opus is not your run-of-the-mill codec locked into a specific bitrate and narrowly optimized for speech, high quality music or video. No, Opus can handle everything from low bitrate voice to high bitrate music coding thanks to the fact that it is an amalgam of Skype’s SILK codec that is optimized for low rate voice coding, and Xiph.org’s Constrained Energy Lapped Transform CELT codec, a very low delay, low CPU/memory requirement successor to Vorbis that handles higher bitrates and thus higher quality audio.