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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Transport Working Group of the Internet Engineering Task Force (IETF) and first published in 1996 as RFC 1889, superseded by RFC 3550 in 2003. ==Overview== RTP is designed for end-to-end, real-time, transfer of streaming media. The protocol provides facilities for jitter compensation and detection of out of sequence arrival in data, which are common during transmissions on an IP network. RTP allows data transfer to multiple destinations through IP 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.〔 Real-time multimedia streaming applications require timely delivery of information and often can tolerate some packet loss to achieve this goal. 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. The Transmission Control Protocol (TCP), although standardized for RTP use,〔RFC 4571〕 is not normally used in RTP applications because TCP favors reliability over timeliness. 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. 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 RTCP. RTP is used for transfer of multimedia data, and the RTCP is used to periodically send control information and QoS parameters.〔 抄文引用元・出典: フリー百科事典『 ウィキペディア（Wikipedia）』 ■ウィキペディアで「Real-time Transport Protocol」の詳細全文を読む スポンサード リンク
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