To this end, the information required by a specific application of the protocol is not included in the generic RTP is designed to carry a multitude of multimedia formats, which permits the development of new formats without revising the RTP standard. RTP is used by real-time multimedia applications such as voice over IP, audio over IP, WebRTC and Internet Protocol television The RTP specification recommends even port numbers for RTP, and the use of the next odd port number for the associated RTCP session. A single port be usedįor RTP and RTCP in applications that multiplex the protocols. May be used when a reliable transport protocol is desired. Applications most typically use UDP with port numbers in the unprivileged range (1024 to 65535). The Stream Control Transmission Protocol (SCTP) and the Datagram Congestion Control Protocol (DCCP) Audio and video streams may use separate RTP sessions, enabling a receiver to selectively receive components of a particular stream. The RTP and RTCP design is independent of These protocols may use the Session Description Protocol toĪn RTP session is established for each multimedia stream. RTP sessions are typically initiated between communicating peers using a signaling protocol, such as H.323, the Session Initiation Protocol (SIP), RTSP, or Jingle (XMPP). The bandwidth of RTCP traffic compared to RTP is small, typically around 5%. Of the data. The control protocol, RTCP, is used for quality of service (QoS) feedback and synchronization between the media streams. Information provided by this protocol includes timestamps (for synchronization), sequence numbers (for packet loss and reordering detection) and the payload format which indicates the encoded format The data transfer protocol, RTP, carries real-time data. RTP is used for the transfer of multimedia data, and the RTCP is used to periodically send control information and QoS parameters. RTP is used in conjunction with other protocols such as H.323 and RTSP. The RTP specification describes two protocols: RTPĪnd RTCP. RTP was developed by the Audio/Video Transport working group of the IETF standards organization. Sessions are SCTP and DCCP, although, as of 2012, they are not in widespread use. Instead the majority of the RTP implementations are built on the User Datagram Protocol (UDP). Other transport protocols specifically designed for multimedia Of audio data, which can be made unnoticeable with suitable error concealment algorithms. The Transmission Control Protocol (TCP), although standardized for RTP use, is not normally used in RTPĪpplications because TCP favors reliability over timeliness. For example, loss of a packet in audio application may result in loss of a fraction of a second Real-time multimedia streaming applications require timely delivery of information and often can tolerate some packet loss to achieve this goal. Profile and payload format. The design of RTP is based on the architectural principle known as application-layer framing where protocol functions are implemented in the application as opposed to in the operating system's protocol 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 The protocol provides facilities for jitter compensation and detection of packet loss and out-of-order delivery, which are common especiallyĭuring UDP transmissions on an IP network. RTP is designed for end-to-end, real-time transfer of streaming media. 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 which was then superseded by RFC 3550 in 2003. Initiation Protocol (SIP) which establishes connections across the network.

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 Of service (QoS) and aids synchronization of multiple streams. While RTP carries the media streams (e.g., audio and video), RTCP is used to monitor transmission statistics and quality RTP is used in conjunction with the RTP Control Protocol (RTCP). RTP typically runs over User Datagram Protocol (UDP). Media, such as telephony, video teleconference applications including WebRTC, television services, and web-based push-to-talk features. RTP is used in communication and entertainment systems that involve streaming The Real-time Transport Protocol ( RTP) is a network protocol for delivering audio and video over IP networks.

Ast rtp read rtp codec 96 received from