RTP (Real-time Transport Protocol, RFC 3550) is the lightweight packet format that carries live media — every audio and video frame in WebRTC, SIP, and broadcast contribution. It runs over UDP because real-time media would rather lose a packet than wait for a retransmission, and its compact header adds just what the receiver needs: a payload type identifying the codec, a sequence number to detect loss and reordering, and a timestamp marking the media's sampling instant. RTP itself is deliberately minimal; the surrounding machinery — RTCP for control and statistics, SRTP for encryption, jitter buffers for timing — does the heavy lifting. It is the common substrate beneath essentially all interactive audio.
