RTP Payload Format for Opus
WebRTC and modern SIP use RFC 7587 Opus RTP framing — not Ogg pages. FEC and DTX SDP attributes directly control call quality under packet loss.
What It Defines
Defines how Opus frames are carried in RTP: dynamic payload type, 48 kHz RTP clock rate (regardless of encoded bandwidth), stereo `sprop-stereo` SDP attribute, and `useinbandfec` for forward error correction. Distinct from Ogg Opus file encapsulation (RFC 7845).
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
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.
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.
Codec mismatch debugging starts here — SDP says `PCMU/8000` which is PT 0 per RFC 3551. Static vs dynamic payload types explain most SIP/WebRTC negotiation confusion.
Opus-in-Ogg is an alternative to WebM/MP4 for VoIP recordings and Icecast streams. Page/chunk structure is a classic telephony streaming envelope pattern.
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.