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Real-time Transport 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 a ...
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Network Protocol
A communication protocol is a system of rules that allows two or more entities of a communications system to transmit information via any kind of variation of a physical quantity. The protocol defines the rules, syntax, semantics and synchronization of communication and possible error recovery methods. Protocols may be implemented by hardware, software, or a combination of both. Communicating systems use well-defined formats for exchanging various messages. Each message has an exact meaning intended to elicit a response from a range of possible responses pre-determined for that particular situation. The specified behavior is typically independent of how it is to be implemented. Communication protocols have to be agreed upon by the parties involved. To reach an agreement, a protocol may be developed into a technical standard. A programming language describes the same for computations, so there is a close analogy between protocols and programming languages: ''protocols are to co ...
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Session Description Protocol
The Session Description Protocol (SDP) is a format for describing multimedia communication sessions for the purposes of announcement and invitation. Its predominant use is in support of streaming media applications, such as voice over IP (VoIP) and video conferencing. SDP does not deliver any media streams itself but is used between endpoints for negotiation of network metrics, media types, and other associated properties. The set of properties and parameters is called a ''session profile''. SDP is extensible for the support of new media types and formats. SDP was originally a component of the Session Announcement Protocol (SAP), but found other uses in conjunction with the Real-time Transport Protocol (RTP), the Real-time Streaming Protocol (RTSP), Session Initiation Protocol (SIP), and as a standalone protocol for describing multicast sessions. The IETF published the original specification as a Proposed Standard in April 1998. Revised specifications were released in 2006 (RFC ...
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DCCP
In computer networking, the Datagram Congestion Control Protocol (DCCP) is a message-oriented transport layer Communication protocol, protocol. DCCP implements reliable connection setup, teardown, Explicit Congestion Notification (ECN), congestion control, and feature negotiation. The IETF published DCCP as , a proposed standard, in March 2006. provides an introduction. Operation DCCP provides a way to gain access to congestion-control mechanisms without having to implement them at the application layer. It allows for flow-based semantics like in Transmission Control Protocol (TCP), but does not provide reliable in-order delivery. Sequenced delivery within multiple streams as in the Stream Control Transmission Protocol (SCTP) is not available in DCCP. A DCCP connection contains Acknowledgement (data networks), acknowledgment traffic as well as data traffic. Acknowledgments inform a sender whether its packets have arrived, and whether they were marked by Explicit Congestion Notif ...
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SCTP
The Stream Control Transmission Protocol (SCTP) is a computer networking communications protocol in the transport layer of the Internet protocol suite. Originally intended for Signaling System 7 (SS7) message transport in telecommunication, the protocol provides the message-oriented feature of the User Datagram Protocol (UDP), while ensuring reliable, in-sequence transport of messages with congestion control like the Transmission Control Protocol (TCP). Unlike UDP and TCP, the protocol supports multihoming and redundant paths to increase resilience and reliability. SCTP is standardized by the Internet Engineering Task Force (IETF) in . The SCTP reference implementation was released as part of FreeBSD version 7, and has since been widely ported to other platforms. Formal oversight The IETF Signaling Transport (SIGTRAN) working group defined the protocol (number 132) in October 2000, and the IETF Transport Area (TSVWG) working group maintains it. defines the protocol. provides an ...
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Transmission Control Protocol
The Transmission Control Protocol (TCP) is one of the main protocols of the Internet protocol suite. It originated in the initial network implementation in which it complemented the Internet Protocol (IP). Therefore, the entire suite is commonly referred to as TCP/IP. TCP provides reliable, ordered, and error-checked delivery of a stream of octets (bytes) between applications running on hosts communicating via an IP network. Major internet applications such as the World Wide Web, email, remote administration, and file transfer rely on TCP, which is part of the Transport Layer of the TCP/IP suite. SSL/TLS often runs on top of TCP. TCP is connection-oriented, and a connection between client and server is established before data can be sent. The server must be listening (passive open) for connection requests from clients before a connection is established. Three-way handshake (active open), retransmission, and error detection adds to reliability but lengthens latency. Applica ...
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Error Concealment
Error concealment is a technique used in signal processing that aims to minimize the deterioration of signals caused by missing data, called packet loss. A signal is a message sent from a transmitter to a receiver in multiple small packets. Packet loss occurs when these packets are misdirected, delayed, resequenced, or corrupted. Receiver-Based Techniques When error recovery occurs at the receiving end of the signal, it is receiver-based. These techniques focus on correcting corrupted or missing data. Waveform substitution Preliminary attempts at receiver-based error concealment involved packet repetition, replacing lost packets with copies of previously received packets. This function is computationally simple and is performed by a device on the receiver end called a "drop-out compensator". Zero Insertion When this technique is used, if a packet is lost, its entries are replaced with 0s. Interpolation Interpolation involves making educated guesses about the nature of a missi ...
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Multimedia
Multimedia is a form of communication that uses a combination of different content forms such as text, audio, images, animations, or video into a single interactive presentation, in contrast to traditional mass media, such as printed material or audio recordings, which features little to no interaction between users. Popular examples of multimedia include video podcasts, audio slideshows and animated videos. Multimedia also contains the principles and application of effective interactive communication such as the building blocks of software, hardware, and other technologies. Multimedia can be recorded for playback on computers, laptops, smartphones, and other electronic devices, either on demand or in real time (streaming). In the early years of multimedia, the term "rich media" was synonymous with interactive multimedia. Over time, hypermedia extensions brought multimedia to the World Wide Web. Terminology The term ''multimedia'' was ...
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Protocol Stack
The protocol stack or network stack is an implementation of a computer networking protocol suite or protocol family. Some of these terms are used interchangeably but strictly speaking, the ''suite'' is the definition of the communication protocols, and the ''stack'' is the software implementation of them. Individual protocols within a suite are often designed with a single purpose in mind. This modularization simplifies design and evaluation. Because each protocol module usually communicates with two others, they are commonly imagined as layers in a stack of protocols. The lowest protocol always deals with low-level interaction with the communications hardware. Each higher layer adds additional capabilities. User applications usually deal only with the topmost layers. General protocol suite description T ~ ~ ~ T ____ Imagine three computers: ''A'', ''B'', and ''C''. ''A'' and ''B'' both have radio equipment and can communicate via the airwaves using a suitable networ ...
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Application-layer Framing
Application-layer framing or application-level framing (ALF) is a method of allowing an application to use its semantics for the design of its network protocols. This procedure was first proposed by D. D. Clark and David L. Tennenhouse.Clark, D. D. and Tennenhouse, D. L. (1990). Architectural considerations for a new generation of protocols. In: ''ACM SIGCOMM Computer Communication Review archive'' Volume 20, Issue 4 (September 1990), Pages 200 - 208, ISSN 0146-483/ref> It works as follows: * The application splits the data into useful segments. ** These segments are called ADUs (application data units). * The ADUs can be processed in any order. * The lower layers keep the ADU borders. This procedure simplifies the quality of service negotiation and provides a simpler method of error checking. The Real-time Transport Protocol (RTP) is an example of where the semantics of the real-time application are used to segment the data. References See also * Frame (networking) ...
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IP Multicast
IP multicast is a method of sending Internet Protocol (IP) datagrams to a group of interested receivers in a single transmission. It is the IP-specific form of multicast and is used for streaming media and other network applications. It uses specially reserved multicast address blocks in IPv4 and IPv6. Protocols associated with IP multicast include Internet Group Management Protocol, Protocol Independent Multicast and Multicast VLAN Registration. IGMP snooping is used to manage IP multicast traffic on layer-2 networks. IP multicast is described in . IP multicast was first standardized in 1986. Its specifications have been augmented in to include group management and in to include administratively scoped addresses. Technical description IP multicast is a technique for one-to-many and many-to-many real-time communication over an IP infrastructure in a network. It scales to a larger receiver population by requiring neither prior knowledge of a receiver's identity nor prior knowl ...
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Out-of-order Delivery
In computer networking, out-of-order delivery is the delivery of data packets in a different order from which they were sent. Out-of-order delivery can be caused by packets following multiple paths through a network, by lower-layer retransmission procedures (such as automatic repeat request), or via parallel processing paths within network equipment that are not designed to ensure that packet ordering is preserved. One of the functions of TCP is to prevent the out-of-order delivery of data, either by reassembling packets in order or requesting retransmission of out-of-order packets. See also * Packet loss * Selective ACK * IP fragmentation * Head-of-line blocking Head-of-line blocking (HOL blocking) in computer networking is a performance-limiting phenomenon that occurs when a line of packets is held up in a queue by a first packet. Examples include input buffered network switches, out-of-order delivery a ... External links * RFC 4737, ''Packet Reordering Metrics'', A. ...
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Packet Loss
Packet loss occurs when one or more packets of data travelling across a computer network fail to reach their destination. Packet loss is either caused by errors in data transmission, typically across wireless networks, or network congestion.Kurose, J.F. & Ross, K.W. (2010). ''Computer Networking: A Top-Down Approach''. New York: Addison-Wesley. Packet loss is measured as a percentage of packets lost with respect to packets sent. The Transmission Control Protocol (TCP) detects packet loss and performs retransmissions to ensure reliable messaging. Packet loss in a TCP connection is also used to avoid congestion and thus produces an intentionally reduced throughput for the connection. In real-time applications like streaming media or online games, packet loss can affect a user's quality of experience (QoE). Causes The Internet Protocol (IP) is designed according to the end-to-end principle as a best-effort delivery service, with the intention of keeping the logic routers mus ...
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