Real-time Transport Protocol
The Real-time Transport Protocol (RTP) defines a standardized packet format for delivering audio and video over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and first published in 1996 as RFC 1889, and superseded by RFC 3550 in 2003.
RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications and web-based push to talk features. For these it carries media streams controlled by H.323, MGCP, Megaco, SCCP, or Session Initiation Protocol (SIP) signaling protocols, making it one of the technical foundations of the Voice over IP industry.
RTP is usually used in conjunction with the RTP Control Protocol (RTCP). While RTP carries the media streams (e.g., audio and video) or out-of-band signaling (DTMF), RTCP is used to monitor transmission statistics and quality of service (QoS) information. When both protocols are used in conjunction, RTP is usually originated and received on even port numbers, whereas RTCP uses the next higher odd port number.
RTP was developed by the Audio/Video Transport working group of the IETF standards organization. RTP is used in conjunction with other protocols such as H.323 and RTSP. The RTP standard defines a pair of protocols, RTP and the Real-time Transport Control Protocol (RTCP). RTP is used for transfer of multimedia data, and the RTCP is used to periodically send control information and QOS parameters.
RTP is designed for end-to-end, real-time, transfer of multimedia data. The protocol provides facility for jitter compensation and detection of out of sequence arrival in data, that are common during transmissions on an IP network. RTP supports data transfer to multiple destinations through multicast. RTP is regarded as the primary standard for audio/video transport in IP networks and is used with an associated profile and payload format.
Multimedia applications need timely delivery and can tolerate some loss in packets. For example, loss of a packet in audio application may result in loss of a fraction of a second of audio data, which can be made unnoticeable with suitable error concealment algorithms. Multimedia applications require timeliness over reliability. The Transmission Control Protocol (TCP), although standardized for RTP use (RFC 4571), is not often used by RTP because of inherent latency introduced by connection establishment and error correction, instead the majority of the RTP implementations are built on the User Datagram Protocol (UDP). Other transport protocols specifically designed for multimedia sessions are SCTP and DCCP, although they are not in widespread use yet.