RTP is regarded as the primary standard for audio / video transport in IP networks and is used with an associated profile and payload format.

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 will require a profile and payload format specification ( s ).

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 ( see below ).

* The Secure Real-time Transport Protocol ( SRTP ) ( RFC 3711 ) defines a profile of RTP that provides cryptographic services for the transfer of payload data.

* RTP audio video profile

* RTP audio video profile

* RTP audio video profile

As part of the RTP audio video profile, RFC 3389 defines a standard for distributing comfort noise information in VoIP systems.

The Secure Real-time Transport Protocol ( or SRTP ) defines a profile of RTP ( Real-time Transport Protocol ), intended to provide encryption, message authentication and integrity, and replay protection to the RTP data in both unicast and multicast applications.

* Using RTP as a wrapper for video packets

Most RTSP servers use the Real-time Transport Protocol ( RTP ) in conjunction with Real-time Control Protocol ( RTCP ) for media stream delivery, however some vendors implement proprietary transport protocols.

RTSP using RTP and RTCP allows for the implementation of rate adaption.

This specifier typically includes a local port for receiving RTP data ( audio or video ), and another for RTCP data ( meta information ).

The Real-time Transport Protocol ( RTP ) defines a standardized packet format for delivering audio and video over IP networks.

RTP is designed for end-to-end, real-time, transfer of stream data.

The Transmission Control Protocol ( TCP ), although standardized for RTP use, is not normally used in RTP application because TCP favors reliability over timeliness.

RTP is used for transfer of multimedia data, and the RTCP is used to periodically send control information and QoS parameters.

An RTP Session is established for each multimedia stream.

A session consists of an IP address with a pair of ports for RTP and RTCP.

* The experimental Control Data Profile for RTP ( RTP / CDP ) for machine-to-machine communications.

SDP started off as a component of the Session Announcement Protocol ( SAP ), but found other uses in conjunction with Real-time Transport Protocol ( RTP ), Real-time Streaming Protocol ( RTSP ), Session Initiation Protocol ( SIP ) and even as a standalone format for describing multicast sessions.

However, in May 2004 he told Portuguese state-run television RTP, that he would not allow his name to be put up for nomination.

The media type for Speex is audio / ogg while contained by Ogg, and audio / speex ( previously audio / x-speex ) when transported through RTP or without container.

* RFC 5574 – RTP Payload Format for the Speex Codec

RTP was developed by the Audio / Video Transport working group of the IETF standards organization.

; Input: UDP / RTP unicast or multicast, HTTP, FTP, MMS, RTSP, RTMP, DVDs, VCD, SVCD, CD Audio, DVB, Video acquisition ( via V4l and DirectShow ), RSS / Atom Feeds, and from files stored on the user's computer.

* RFC 4424-Real-Time Transport Protocol ( RTP ) Payload Format for the Variable-Rate Multimode Wideband ( VMR-WB ) Extension Audio Codec

* RFC 4348-Real-Time Transport Protocol ( RTP ) Payload Format for the Variable-Rate Multimode Wideband ( VMR-WB ) Audio Codec

* RFC 4856 – Media Type Registration of Payload Formats in the RTP Profile for Audio and Video Conferences – audio / L8 and audio / L16 ( March 2007 )

* RFC 3190 – RTP Payload Format for 12-bit DAT Audio and 20-and 24-bit Linear Sampled Audio ( January 2002 )

* RFC 3551 – RTP Profile for Audio and Video Conferences with Minimal Control – L8 and L16 ( July 2003 )

** RFC 3551, Standard 65, RTP Profile for Audio and Video Conferences with Minimal Control

m = video 0 RTP / AVP 96

m = video 2232 RTP / AVP 31

RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and web-based push-to-talk features.

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.

For example, audio and video streams will have separate RTP sessions, enabling a receiver to deselect a particular stream.

An additional goal was to provide enough flexibility to allow the standard to be applied to a wide variety of applications on a wide variety of networks and systems, including low and high bit rates, low and high resolution video, broadcast, DVD storage, RTP / IP packet networks, and ITU-T multimedia telephony systems.

The RTP and UDP normally carry information such the payload type which is the type of codec, frame rate, video size and many others.

IP / TV was an MBONE compatible Windows and Unix-based application that transmitted single and multi-source audio and video traffic, ranging from low to DVD quality, using both unicast and IP multicast Real-time Transport Protocol ( RTP ) and Real time control protocol ( RTCP ).

While standards already exist for audio and video codecs ( e. g. MPEG ) and for real time streaming transport over IP networks ( e. g. RTP ) putting these together requires selecting profiles, describing payload formats, and resolving various options.

* NetMeeting audio and video codecs use RTP above UDP / IP connections.

* Real-time Transport Protocol ( RTP ), a standardised protocol for streaming audio and video over computer networks