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Reliable Byte Stream
A reliable byte stream is a common service paradigm in computer networking; it refers to a byte stream in which the bytes which emerge from the communication channel at the recipient are exactly the same, and in exactly the same order, as they were when the sender inserted them into the channel. The classic example of a reliable byte stream communication protocol is the Transmission Control Protocol, one of the major building blocks of the Internet. A reliable byte stream is not the only reliable service paradigm which computer network communication protocols provide, however; other protocols (e.g. SCTP) provide a reliable message stream, i.e. the data is divided up into distinct units, which are provided to the consumer of the data as discrete objects. Mechanism Communication protocols that implement reliable byte streams, generally over some unreliable lower level, use a number of mechanisms to provide that reliability. ARQ protocols have an important role for achieving rel ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Computer Networking
A computer network is a set of computers sharing resources located on or provided by network nodes. The computers use common communication protocols over digital interconnections to communicate with each other. These interconnections are made up of telecommunication network technologies, based on physically wired, optical, and wireless radio-frequency methods that may be arranged in a variety of network topologies. The nodes of a computer network can include personal computers, servers, networking hardware, or other specialised or general-purpose hosts. They are identified by network addresses, and may have hostnames. Hostnames serve as memorable labels for the nodes, rarely changed after initial assignment. Network addresses serve for locating and identifying the nodes by communication protocols such as the Internet Protocol. Computer networks may be classified by many criteria, including the transmission medium used to carry signals, bandwidth, communications protocols ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Network Latency
Network delay is a design and performance characteristic of a telecommunications network. It specifies the latency for a bit of data to travel across the network from one communication endpoint to another. It is typically measured in multiples or fractions of a second. Delay may differ slightly, depending on the location of the specific pair of communicating endpoints. Engineers usually report both the maximum and average delay, and they divide the delay into several parts: * Processing delay time it takes a router to process the packet header * Queuing delay time the packet spends in routing queues * Transmission delay time it takes to push the packet's bits onto the link * Propagation delay time for a signal to propagate through the media A certain minimum level of delay is experienced by signals due to the time it takes to transmit a packet serially through a link. This delay is extended by more variable levels of delay due to network congestion. IP network delays can range ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Automatic Repeat Request
Automatic repeat request (ARQ), also known as automatic repeat query, is an error-control method for data transmission that uses acknowledgements (messages sent by the receiver indicating that it has correctly received a packet) and timeouts (specified periods of time allowed to elapse before an acknowledgment is to be received) to achieve reliable data transmission over an unreliable communication channel. If the sender does not receive an acknowledgment before the timeout, it re-transmits the packet until it receives an acknowledgment or exceeds a predefined number of retransmissions. Variations of ARQ protocols include Stop-and-wait ARQ, Go-Back-N ARQ, and Selective Repeat ARQ. All three protocols usually use some form of sliding window protocol to help the sender determine which (if any) packets need to be retransmitted. These protocols reside in the data link or transport layers (layers 2 and 4) of the OSI model. Examples The Transmission Control Protocol uses a varia ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Forward Error Correction
In computing, telecommunication, information theory, and coding theory, an error correction code, sometimes error correcting code, (ECC) is used for controlling errors in data over unreliable or noisy communication channels. The central idea is the sender encodes the message with redundant information in the form of an ECC. The redundancy allows the receiver to detect a limited number of errors that may occur anywhere in the message, and often to correct these errors without retransmission. The American mathematician Richard Hamming pioneered this field in the 1940s and invented the first error-correcting code in 1950: the Hamming (7,4) code. ECC contrasts with error detection in that errors that are encountered can be corrected, not simply detected. The advantage is that a system using ECC does not require a reverse channel to request retransmission of data when an error occurs. The downside is that there is a fixed overhead that is added to the message, thereby requiring a h ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Round-trip Time
In telecommunications, round-trip delay (RTD) or round-trip time (RTT) is the amount of time it takes for a signal to be sent ''plus'' the amount of time it takes for acknowledgement of that signal having been received. This time delay includes propagation times for the paths between the two communication endpoints. In the context of computer networks, the signal is typically a data packet. RTT is also known as ping time, and can be determined with the ping command. End-to-end delay is the length of time it takes for a signal to travel in one direction and is often approximated as half the RTT. Protocol design Round-trip delay and bandwidth are independent of each other. As the available bandwidth of networks increases, the round trip time does not similarly decrease, as it depends primarily on constant factors such as physical distance and the speed of signal propagation. Networks with both high bandwidth and a high RTT (and thus high bandwidth-delay product) can have very larg ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Datagram
A datagram is a basic transfer unit associated with a packet-switched network. Datagrams are typically structured in header and payload sections. Datagrams provide a connectionless communication service across a packet-switched network. The delivery, arrival time, and order of arrival of datagrams need not be guaranteed by the network. History In the early 1970s, the term ''datagram'' was created by combining the words ''data'' and ''telegram'' by the CCITT rapporteur on packet switching, Halvor Bothner-By. While the word was new, the concept had already a long history. In 1962, Paul Baran described, in a RAND Corporation report, a hypothetical military network having to resist a nuclear attack. Small standardized "message blocks", bearing source and destination addresses, were stored and forwarded in computer nodes of a highly redundant meshed computer network. "The network user who has called up a "virtual connection" to an end station and has transmitted messages ... ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
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) ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
HTTP/3
HTTP/3 is the third major version of the Hypertext Transfer Protocol used to exchange information on the World Wide Web, complementing the widely-deployed HTTP/1.1 and HTTP/2. Unlike previous versions which relied on the well-established TCP (published in 1974), HTTP/3 uses QUIC, a multiplexed transport protocol built on UDP. On 6 June 2022, IETF published HTTP/3 as a Proposed Standard in . File:HTTP-1.1 vs. HTTP-2 vs. HTTP-3 Protocol Stack.svg, frame, Protocol Stack of HTTP/3 compared to HTTP/1.1 and HTTP/2 rect 0 44 85 94 HTTP/1 rect 0 95 85 137 Transport Layer Security rect 0 138 85 189 Transmission Control Protocol rect 126 44 210 94 HTTP/2 rect 126 95 210 137 TLS 1.2 rect 126 138 210 189 Transmission Control Protocol rect 251 44 336 74 HTTP/3 rect 255 90 328 121 TLS 1.3 rect 251 125 336 158 QUIC rect 251 159 336 189 User Datagram Protocol rect 0 189 336 231 Internet Protocol HTTP/3 uses similar semantics compared to earlier revisions of the protocol, including the ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Frame (networking)
A frame is a digital data transmission unit in computer networking and telecommunication. In packet switched systems, a frame is a simple container for a single network packet. In other telecommunications systems, a frame is a repeating structure supporting time-division multiplexing. A frame typically includes frame synchronization features consisting of a sequence of bits or symbols that indicate to the receiver the beginning and end of the payload data within the stream of symbols or bits it receives. If a receiver is connected to the system during frame transmission, it ignores the data until it detects a new frame synchronization sequence. Packet switching In the OSI model of computer networking, a frame is the protocol data unit at the data link layer. Frames are the result of the final layer of encapsulation before the data is transmitted over the physical layer. A frame is "the unit of transmission in a link layer protocol, and consists of a link layer header followed by ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
HTTP/2
HTTP/2 (originally named HTTP/2.0) is a major revision of the HTTP network protocol used by the World Wide Web. It was derived from the earlier experimental SPDY protocol, originally developed by Google. HTTP/2 was developed by the HTTP Working Group (also called httpbis, where "" means "twice") of the Internet Engineering Task Force (IETF). HTTP/2 is the first new version of HTTP since HTTP/1.1, which was standardized in in 1997. The Working Group presented HTTP/2 to the Internet Engineering Steering Group (IESG) for consideration as a Proposed Standard in December 2014, and IESG approved it to publish as Proposed Standard on February 17, 2015 (and was updated in February 2020 in regard to TLS 1.3). The HTTP/2 specification was published as on May 14, 2015. The standardization effort was supported by Chrome, Opera, Firefox, Internet Explorer 11, Safari, Amazon Silk, and Edge browsers. Most major browsers had added HTTP/2 support by the end of 2015. About 97% of web browsers ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Time-division Multiplexing
Time-division multiplexing (TDM) is a method of transmitting and receiving independent signals over a common signal path by means of synchronized switches at each end of the transmission line so that each signal appears on the line only a fraction of time in an alternating pattern. This method transmits two or more digital signals or analog signals over a common channel. It can be used when the bit rate of the transmission medium exceeds that of the signal to be transmitted. This form of signal multiplexing was developed in telecommunications for telegraphy systems in the late 19th century, but found its most common application in digital telephony in the second half of the 20th century. History Time-division multiplexing was first developed for applications in telegraphy to route multiple transmissions simultaneously over a single transmission line. In the 1870s, Émile Baudot developed a time-multiplexing system of multiple Hughes telegraph machines. In 1944, the Britis ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Encapsulation (networking)
In computer networking, encapsulation is a method of designing modular communication protocols in which logically separate functions in the network are abstracted from their underlying structures by inclusion or information hiding within higher-level objects. In other words, encapsulation "takes information from a higher layer and adds a header to it, treating the higher layer information as data". The physical layer is responsible for physical transmission of the data, link encapsulation allows local area networking, IP provides global addressing of individual computers, and TCP selects the process or application (i.e., the TCP or UDP port) that specifies the service such as a Web or TFTP server. During encapsulation, each layer builds a protocol data unit (PDU) by adding a header and optionally a trailer, both of which contain control information to the PDU from the layer above. For example, in the IP suite, the contents of a web page are encapsulated with an HTTP hea ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
