Opus Interactive Audio Codec
Opus is the codec for real-time audio on the internet. WebRTC mandates it; Discord, Zoom, and most VoIP apps use it. Understanding bitrate, FEC, and frame sizing directly impacts audio quality and bandwidth.
What It Defines
Opus is a versatile audio codec covering 6–510 kbps, 8–48 kHz sample rates, 2.5–60 ms frame sizes, and both speech (SILK layer) and music (CELT layer) modes. Royalty-free, open-source, and mandated by WebRTC. Provides variable bitrate (VBR), discontinuous transmission (DTX), and forward error correction (FEC) for lossy networks.
Canonical (Normative)
Convenient (Practical)
The canonical publication point for finalized RFCs. If a protocol is standardized as an RFC, the RFC Editor text is the normative final reference. Published by the IETF, IRTF, IAB, and independent stream.
Related Specs
RTP carries all real-time media: VoIP, WebRTC, live video, and video conferencing. Understanding SSRC demultiplexing, jitter buffers, and RTCP feedback loops is essential for any media application.
Video calls (Meet, Zoom web), voice calls, peer-to-peer file transfer, and collaborative tools. The browser API surface for real-time A/V communication.
Any media you send over the internet should be encrypted. SRTP is the protocol. In WebRTC it's automatic via DTLS-SRTP; in SIP deployments you must configure it explicitly. Know it when auditing VoIP security.