Real-time Transport Protocol
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.
What It Defines
RTP provides end-to-end delivery of real-time audio and video over UDP. Defines the 12-byte header (version, padding, extension, CC, marker, payload type, sequence number, timestamp, SSRC, CSRC), payload format negotiation, and the companion RTCP protocol for sender/receiver reports, NACK, and jitter/loss statistics.
Canonical (Normative)
Related References
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
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