RtP

Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks in real time. It is designed to provide end-to-end delivery of time-sensitive data and is widely used in applications such as Voice over IP, video conferencing, and streaming media. RTP itself does not guarantee timely delivery or data integrity; it relies on higher-layer protocols and the RTP Control Protocol (RTCP) for monitoring and control.

RTP typically runs on top of UDP, though other transports can be used. Each RTP packet contains a fixed header followed by a payload. The header carries version, padding, extension, CSRC count, marker, payload type, sequence number, timestamp, and a 32-bit synchronization source identifier (SSRC). The payload type identifies the codec or format and can be static or dynamically assigned. The timestamp reflects the sampling instant of the first sample in the packet, enabling the receiver to reconstruct timing, while the sequence number allows detection of loss and reordering. The marker bit signals important events such as frame boundaries.

RTP is commonly used with RTCP, the RTP Control Protocol, which provides out-of-band control information used to monitor transmission statistics and quality of service. RTCP reports include reception quality metrics such as packet loss, jitter, and round-trip time, and they help synchronize multiple RTP streams.

Standards and profiles: RTP was first published in RFC 1889 and is now defined by RFC 3550. RFC 3551 specifies payload formats for common codecs. Profiles such as RTP/AVP define typical operation and payload type mapping. Security considerations are addressed by SRTP, which provides encryption, integrity, and replay protection for RTP streams.

Deployment notes: RTP is central to many telephony and streaming stacks and is often signaled by protocols such as SIP or RTSP. It is designed for real-time delivery but does not itself provide end-to-end reliability; application-level mechanisms may compensate for loss.