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Working Group of the Internet Engineering Task Force (IETF) and first published in 1996 as RFC 1889 which was then superseded by RFC 3550 in 2003.
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. 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 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, 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 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%.
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). These protocols may use the Session Description Protocol to specify the parameters for the sessions.
An RTP session is established for each multimedia stream. Audio and video streams may use separate RTP sessions, enabling a receiver to selectively receive components of a particular stream. The specification recommends that RTP port numbers are chosen to be even and that each associated RTCP port be the next higher odd number.:68 However, a single port is chosen for RTP and RTCP in applications that multiplex the protocols. RTP and RTCP typically use unprivileged UDP ports (1024 to 65535), but may also use other transport protocols, most notably, SCTP and DCCP, as the protocol design is transport independent.
RTP is used by real-time multimedia applications such as voice over IP, audio over IP, WebRTC and Internet Protocol television
One of the design considerations for RTP is to carry a range of multimedia formats and allow new formats without revising the RTP standard. To this end, the information required by a specific application of the protocol is not included in the generic RTP header but is instead provided through separate RTP profiles and associated payload formats. For each class of application (e.g., audio, video), RTP defines a profile and one or more associated payload formats. A complete specification of RTP for a particular application usage requires profile and payload format specifications.:71
The profile defines the codecs used to encode the payload data and their mapping to payload format codes in the Payload Type (PT) field of the RTP header. Each profile is accompanied by several payload format specifications, each of which describes the transport of a particular encoded data. The audio payload formats include G.711, G.723, G.726, G.729, GSM, QCELP, MP3, and DTMF, and the video payload formats include H.261, H.263, H.264, H.265, and MPEG-1/MPEG-2. The mapping of MPEG-4 audio/video streams to RTP packets is specified in RFC 3016, and H.263 video payloads are described in RFC 2429.
Examples of RTP profiles include:
The experimental Control Data Profile for RTP (RTP/CDP) for machine-to-machine communications.