For years, internet voice used speech-optimized codecs that handled music poorly, while music codecs wasted bits on speech. The IETF's Internet Wideband Audio Codec working group set out to produce one codec for interactive audio, from 6 kbit/s narrowband speech to high-quality stereo music.

The result, Opus, merged technology from Skype's SILK codec and Xiph.Org's CELT codec into a single hybrid, defined as RFC 6716 on the IETF Standards Track in September 2012. The codec is totally open and royalty-free, with an up-to-date reference implementation available to everyone.

The decisive design choice was the hybrid itself: speech and music modes in one bitstream meant one codec could serve VoIP, videoconferencing, in-game chat and streaming. When the IETF made Opus a required codec for WebRTC, every browser-based call shipped with it, and the format became infrastructure.

Merge two codecs; make a hybrid both sides accept

The hybrid merged two existing communities instead of asking either to abandon its codec.

Royalty-free licensing removed the patent barriers that blocked other codecs.

IETF standardization gave it neutral governance and broad industry buy-in.

WebRTC's mandate put Opus in every browser by default.

When two constituencies need different properties, a hybrid that folds both codecs into one standard removes the need to choose: the merged spec inherits each side's supporters and becomes default.

Opus is now the mandatory audio codec for WebRTC endpoints, so browsers and VoIP services built on WebRTC ship with it by default, and the open reference implementation is available to all. The hybrid design that united SILK and CELT became the standard neither side could have won alone.

FOLLOW THE EVIDENCE

The sources

  1. RFC 6716 — Definition of the Opus Audio Codec rfc-editor.org
  2. RFC 7874 — WebRTC Audio Codec and Processing Requirements rfc-editor.org
  3. Opus Codec — Documentation opus-codec.org
  4. Opus (audio format) en.wikipedia.org