While deployed DOCSIS RFI 1. 0 equipment generally only supports best efforts service, the DOCSIS RFI 1. 0 Interim-01 document discussed QoS extensions and mechanisms using IntServ, RSVP, RTP, and Synchronous Transfer Mode ( STM ) telephony ( as opposed to ATM ).

It supports streaming to any device or application that supports the HTTP, RTSP / RTP, TCP, UDP unicast and UDP multicast streaming protocols.

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

Interleaved binary data SHOULD only be used if RTSP is carried over TCP. Stream data such as RTP packets is encapsulated by an ASCII dollar sign ( 24 hexadecimal ), followed by a one-byte channel identifier, followed by the length of the encapsulated binary data as a binary, two-byte integer in network byte order.

Each $ block contains exactly one upper-layer protocol data unit, e. g., one RTP packet.

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

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

* The data transfer protocol, RTP, which deals with the transfer of real-time data.

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.

It partners RTP in the delivery and packaging of multimedia data, but does not transport any media streams itself.

Ephemeral Terminations represent Connections or data flows, such as RTP streams, or MP3 streams, and usually exist only for the duration of their use in a particular Context.

For example, an MRCP client may request to send some audio data for processing ( say, for speech recognition ), to which the server could respond with a message containing a suitable port number to send the data, since MRCP does not have support for audio data specifically as this would have to be handled by some other protocol, such as Real-time Transport Protocol ( RTP ).

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.

But the standard for encryption of RTP data is just a usual integer incremental counter.

Once RTP had started the connection, BSP took over and managed the data transfer.

* Real-time Transport Protocol ( RTP ) & Real Time Control Protocol ( RTCP ) required for media transfer

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.

Typical RTP set-ups allow multiple models to fly simultaneously on the same pole.

The information required by a specific application's needs is not included in the generic RTP header, but is instead provided through RTP profiles and payload formats.

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.

Until 1991, RTP owned its transmitter network, which was transferred to a state-owned enterprise which, through a series of mergers, became part of Portugal Telecom.

RTP aired the 2008 Olympic Games in HD through the ZON TV Cabo satellite and cable platform.

On August 3, 2012, it was announced that the Portuguese government would spun-off the broadcast license of RTP2 to another private company ( through a privatization process ), thus reducing the Free-to-air offer of the RTP to one channel.

The Internet Engineering Task Force's RTP MIDI open specification is gaining industry support, as proprietary MIDI / IP protocols require expensive licensing fees, or provide no advantage, apart from speed, over the original MIDI protocol.

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

RTP is one of the technical foundations of Voice over IP and in this context is often used in conjunction with a signaling protocol which assists in setting up connections across the network.

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

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

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.

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.

* RFC 2508, Compressing IP / UDP / RTP Headers for Low-Speed Serial Links

However, VoIP differs in that it uses RTP, UDP and IP, all of which are connectionless protocols.

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.

; 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.

These systems typically use highly asymmetric network design and rely on technologies such as multicast or RTP over UDP to achieve similar performance to high end-microwave.

In many applications, the Real-time Transport Protocol ( RTP ) is used for framing of multimedia content over multicast ; the Resource Reservation Protocol ( RSVP ) may be used for bandwidth reservation in a network supporting multicast distribution.

The codec was first designed to be utilized in H. 324 based systems ( PSTN and other circuit-switched network videoconferencing and videotelephony ), but has since also found use in H. 323 ( RTP / IP-based videoconferencing ), H. 320 ( ISDN-based videoconferencing ), RTSP ( streaming media ) and SIP ( Internet conferencing ) solutions.

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.

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.

RTP is originated and received on even port numbers and the associated RTCP communication uses the next higher odd port number.

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, superseded by RFC 3550 in 2003.

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

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.