Editing Real-time Transport Protocol (section)

==Overview==
Research on audio and video over packet-switched networks dates back to the early 1970s. The [[Internet Engineering Task Force]] (IETF) published {{IETF RFC|741}} in 1977 and began developing RTP in 1992,{{sfn|Perkins|2003|p=6}} 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 principle|end-to-end]], [[real-time computing|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]].<ref name="Hardy_298">{{Cite book| author=Daniel Hardy | title=Network | page= [https://books.google.com/books?id=Oq8SEUW1wdQC&pg=PT320 298] | publisher= De Boeck Université | year= 2002}}</ref> RTP is regarded as the primary standard for audio/video transport in IP networks and is used with an associated profile and payload format.<ref name="Perkins_55" />{{update inline|reason=Need a more recent source to verify continued importance|date=February 2025}} 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.<ref name="Perkins_46">{{harvnb|Perkins|2003|p=46}}</ref> The [[Transmission Control Protocol]] (TCP), although standardized for RTP use,{{Ref 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).<ref name="Perkins_46"/> Other transport protocols specifically designed for multimedia sessions are [[SCTP]]<ref>{{Cite book|last=Farrel|first=Adrian |title=The Internet and its protocols|publisher=Morgan Kaufmann|year=2004|page=363|url=https://books.google.com/books?id=LtBegQowqFsC&q=rtp+sctp&pg=PA363|isbn=978-1-55860-913-6}}</ref> and [[DCCP]],<ref>{{Cite book|last=Ozaktas|first=Haldun M.|author2=Levent Onural |title=THREE-DIMENSIONAL TELEVISION|publisher=Springer|year=2007|page=356|url=https://books.google.com/books?id=kQvCHpuXji8C&q=rtp+dccp&pg=PA356|isbn=978-3-540-72531-2}}</ref> although, {{As of|2012|lc=on}}, they were not in widespread use.<ref>{{Cite news|url=http://www.networkworld.com/article/2222277/cisco-subnet/what-about-stream-control-transmission-protocol--sctp--.html|archive-url=https://web.archive.org/web/20140830095541/http://www.networkworld.com/article/2222277/cisco-subnet/what-about-stream-control-transmission-protocol--sctp--.html|url-status=dead|archive-date=August 30, 2014|title=What About Stream Control Transmission Protocol (SCTP)?|last=Hogg|first=Scott|newspaper=Network World|access-date=2017-10-04}}</ref>

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]].<ref name="Perkins_55">{{harvnb|Perkins|2003|p=55}}</ref> 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.<ref name="Peterson_430"/>

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.<ref name="Colins_56">{{harvnb|Perkins|2003|p=56}}</ref> 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%.<ref name="Colins_56"/><ref>{{harvnb|Peterson|Davie|2007|p=435}}</ref>

RTP sessions are typically initiated between communicating peers using a signaling protocol, such as H.323, the [[Session Initiation Protocol]] (SIP), RTSP, or [[Jingle (protocol)|Jingle]] ([[XMPP]]). These protocols may use the [[Session Description Protocol]] to specify the parameters for the sessions.{{Ref RFC|8866}}

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.<ref name="Zurawski_28">{{Cite book|last=Zurawski|first=Richard|title=The industrial information technology handbook|publisher=CRC Press|year=2004|pages=[https://books.google.com/books?id=MwMDUBKZ3wwC&pg=PT225&dq=RTP+session 28–7]|chapter=RTP, RTCP and RTSP protocols|chapter-url=https://books.google.com/books?id=MwMDUBKZ3wwC|isbn=978-0-8493-1985-3}}</ref> 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).<ref name="Collins_47">{{Cite book|last=Collins|first=Daniel|title=Carrier grade voice over IP|publisher=McGraw-Hill Professional|year=2002|pages=[https://books.google.com/books?id=PVIuN9Y5FGMC&pg=PA47&dq=RTP+session 47]|chapter=Transporting Voice by using IP|isbn=978-0-07-136326-6}}</ref> 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.{{Ref RFC|3550|rp=68}} A single port can be used for RTP and RTCP in applications that multiplex the protocols.{{Ref RFC|5761}}

RTP is used by real-time multimedia applications such as [[voice over IP]], [[audio over IP]], [[WebRTC]],  [[Internet Protocol television]], and [[professional video over IP]] including [[SMPTE 2022]] and [[SMPTE 2110]].