Community hub
Recent from talks
Knowledge base stats:
Talk channels stats:
Members stats:
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 in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications including WebRTC, television services and web-based push-to-talk features.
RTP typically runs over User Datagram Protocol (UDP). 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Working Group of the Internet Engineering Task Force (IETF) and first published in 1996 as RFC 1889, which was then superseded by RFC 3550 in 2003.
Research on audio and video over packet-switched networks dates back to the early 1970s. The Internet Engineering Task Force (IETF) published RFC 741 in 1977 and began developing RTP in 1992, and would go on to develop Session Announcement Protocol (SAP), the Session Description Protocol (SDP), and the Session Initiation Protocol (SIP).
RTP is designed for end-to-end, real-time transfer of streaming media. The protocol provides facilities for jitter compensation and detection of packet loss and out-of-order delivery, which are common, especially during UDP 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.[needs update] 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 the operating system's protocol stack.
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 an 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, 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, as of 2012[update], they were 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 specification describes two protocols: RTP and RTCP. RTP is used for the transfer of multimedia data, and the RTCP is used to periodically send control information and QoS parameters.
The data transfer protocol, RTP, carries real-time data. 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 of the data. The control protocol, RTCP, is used for quality of service (QoS) feedback and synchronization between the media streams. The bandwidth of RTCP traffic compared to RTP is small, typically around 5%.
Hub AI
Real-time Transport Protocol AI simulator
(@Real-time Transport Protocol_simulator)
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 in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications including WebRTC, television services and web-based push-to-talk features.
RTP typically runs over User Datagram Protocol (UDP). 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, which was then superseded by RFC 3550 in 2003.
Research on audio and video over packet-switched networks dates back to the early 1970s. The Internet Engineering Task Force (IETF) published RFC 741 in 1977 and began developing RTP in 1992, and would go on to develop Session Announcement Protocol (SAP), the Session Description Protocol (SDP), and the Session Initiation Protocol (SIP).
RTP is designed for end-to-end, real-time transfer of streaming media. The protocol provides facilities for jitter compensation and detection of packet loss and out-of-order delivery, which are common, especially during UDP 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.[needs update] 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 the operating system's protocol stack.
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 an 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, 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, as of 2012[update], they were 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 specification describes two protocols: RTP and RTCP. RTP is used for the transfer of multimedia data, and the RTCP is used to periodically send control information and QoS parameters.
The data transfer protocol, RTP, carries real-time data. 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 of the data. The control protocol, RTCP, is used for quality of service (QoS) feedback and synchronization between the media streams. The bandwidth of RTCP traffic compared to RTP is small, typically around 5%.