Tuesday, October 20, 2015

Real-time Transport Protocol

The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and web-based push-to-talk features.
RTP is used in conjunction with the RTP Control Protocol (RTCP). While RTP carries the media streams (e.g., audio and video), RTCP is used to monitor transmission statistics and quality of service (QoS) and aids synchronization of multiple streams. RTP is one of the technical foundations of Voice over IP and in this context is often used in conjunction with a signaling protocol such as the Session Initiation Protocol (SIP) which establishes connections across the network.
RTP was developed by the Audio-Video Transport Working Group of the Internet Engineering Task Force (IETF) and first published in 1996 as RFC 1889, superseded by RFC 3550 in 2003.
RTP combines its data transport with a control protocol (RTCP), which makes it possible to monitor data delivery for large multicast networks. Typically, RTP runs on top of the UDP protocol, although the specification is general enough to support other transport protocols. An RTP session is established for each multimedia stream. A session consists of an IP address with a pair of ports for RTP and RTCP. For example, audio and video streams use separate RTP sessions, enabling a receiver to deselect a particular stream.

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