| Real-time Transport Protocol | |
| Is Stack: | No |
| Abbreviation: | RTP |
| Purpose: | Delivering audio and video |
| Developer: | Audio-Video Transport Working Group of the IETF |
| Based On: | Network Voice 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Transport Working Group of the Internet Engineering Task Force (IETF) and first published in 1996 as which was then superseded by in 2003.[1]
Research on audio and video over packet-switched networks dates back to the early 1970s. The Internet Engineering Task Force (IETF) published 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.[2] 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 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[3] and DCCP,[4] although,, they were not in widespread use.[5]
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.[6]
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.[7] The RTP and RTCP design is independent of the transport protocol. Applications most typically use UDP with port numbers in the unprivileged range (1024 to 65535).[8] The Stream Control Transmission Protocol (SCTP) and the Datagram Congestion Control Protocol (DCCP) may be used when a reliable transport protocol is desired. 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 can be used for RTP and RTCP in applications that multiplex the protocols.[9]
RTP is used by real-time multimedia applications such as voice over IP, audio over IP, WebRTC and Internet Protocol television.
See main article: RTP payload formats.
RTP is designed to carry a multitude of multimedia formats, which permits the development of 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. For each class of application (e.g., audio, video), RTP defines a profile and associated payload formats.[10] Every instantiation of RTP in a particular application requires a profile and payload format specifications.
The profile defines the codecs used to encode the payload data and their mapping to payload format codes in the protocol field Payload Type (PT) of the RTP header. Each profile is accompanied by several payload format specifications, each of which describes the transport of particular encoded data. Examples of audio payload formats are G.711, G.723, G.726, G.729, GSM, QCELP, MP3, and DTMF, and examples of video payloads are 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, and H.263 video payloads are described in .[11]
Examples of RTP profiles include:
RTP packets are created at the application layer and handed to the transport layer for delivery. Each unit of RTP media data created by an application begins with the RTP packet header.
| 1 | 2 | 3 | ||||||||||||||||||||||||||||||||||
|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|
| Octet | Bit | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | 9 | 10 | 11 | 12 | 13 | 14 | 15 | 16 | 17 | 18 | 19 | 20 | 21 | 22 | 23 | 24 | 25 | 26 | 27 | 28 | 29 | 30 | 31 | |||
| 0 | 0 | Version | P | X | CC | M | PT | Sequence number | ||||||||||||||||||||||||||||
| 4 | 32 | Timestamp | ||||||||||||||||||||||||||||||||||
| 8 | 64 | SSRC identifier | ||||||||||||||||||||||||||||||||||
| 12 | 96 | CSRC identifiers ... | ||||||||||||||||||||||||||||||||||
| 12+4×CC | 96+32×CC | Profile-specific extension header ID | Extension header length | |||||||||||||||||||||||||||||||||
| 16+4×CC | 128+32×CC | Extension header ... | ||||||||||||||||||||||||||||||||||
The RTP header has a minimum size of 12 bytes. After the header, optional header extensions may be present. This is followed by the RTP payload, the format of which is determined by the particular class of application. The fields in the header are as follows:
A functional multimedia application requires other protocols and standards used in conjunction with RTP. Protocols such as SIP, Jingle, RTSP, H.225 and H.245 are used for session initiation, control and termination. Other standards, such as H.264, MPEG and H.263, are used for encoding the payload data as specified by the applicable RTP profile.
An RTP sender captures the multimedia data, then encodes, frames and transmits it as RTP packets with appropriate timestamps and increasing timestamps and sequence numbers. The sender sets the payload type field in accordance with connection negotiation and the RTP profile in use. The RTP receiver detects missing packets and may reorder packets. It decodes the media data in the packets according to the payload type and presents the stream to its user.
SDP: Session Description Protocol, M. Handley, V. Jacobson, C. Perkins, IETF (July 2006)