RTP
RTP (Real-time Transport Protocol, RFC 3550) carries real-time audio and video over UDP, with sequence numbers, timestamps, and SSRC supporting ordering, jitter buffering, and stream mixing.
Purpose
It provides low-latency transport with ordering and synchronization for real-time media, serving as the media transport of VoIP and WebRTC.
Mechanism
The RTP header carries version, PT (payload type), sequence number (loss detection), timestamp (sync and jitter buffering), and SSRC (source identification). The payload is an encoded audio/video frame. RTCP accompanies it with quality statistics.
Use cases
- VoIP audio transport
- WebRTC media
- Video conferencing
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