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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. Automatic repeat request (ARQ) protocols have an impo ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Computer Networking
A computer network is a collection of communicating computers and other devices, such as printers and smart phones. In order to communicate, the computers and devices must be connected by wired media like copper cables, optical fibers, or by wireless communication. The devices may be connected in a variety of network topologies. In order to communicate over the network, computers use agreed-on rules, called communication protocols, over whatever medium is used. The computer network can include personal computers, Server (computing), servers, networking hardware, or other specialized or general-purpose Host (network), hosts. They are identified by network addresses and may have hostnames. Hostnames serve as memorable labels for the nodes and are 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 tr ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
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 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Forward Error Correction
In computing, telecommunication, information theory, and coding theory, forward error correction (FEC) or channel coding is a technique used for controlling errors in data transmission over unreliable or noisy communication channels. The central idea is that the sender encodes the message in a redundant way, most often by using an error correction code, or error correcting code (ECC). The redundancy allows the receiver not only to detect errors that may occur anywhere in the message, but often to correct a limited number of errors. Therefore a reverse channel to request re-transmission may not be needed. The cost is a fixed, higher forward channel bandwidth. 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. FEC can be applied in situations where re-transmissions are costly or impossible, such as one-way communication links or when transmitting to multiple receivers in m ... [...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 commonly used interchangeably with ping time, which can be determined with the ping command. However, ping time may differ from experienced RTT with other protocols since the payload and priority associated with ICMP messages used by ping may differ from that of other traffic. 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 RTT is a measure of the amount of time taken for an entire message to be sent to a destination and for a reply to be sent back to the sender. The time to send the mes ... [...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 1964, 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. Baran wrote: "The network user who has called up a ''virtual connection'' to an end station and has trans ... [...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 (networkin ... [...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 (officially introduced in 2021), a multiplexed transport protocol built on UDP. HTTP/3 uses similar semantics compared to earlier revisions of the protocol, including the same request methods, status codes, and message fields, but encodes them and maintains session state differently. However, partially due to the protocol's adoption of QUIC, HTTP/3 has lower latency and loads more quickly in real-world usage when compared with previous versions: in some cases over four times as fast than with HTTP/1.1 (which, for many websites, is the only HTTP version deployed). As of September 2024, HTTP/3 is supported by more than 95% of major web browsers in use and 34% of the top 10 million websites. It has ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Frame (networking)
A frame is a digital data transmission unit in computer networking and telecommunications. 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 follo ... [...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 and again in June 2022). The initial 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 201 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Encapsulation (networking)
Encapsulation is the computer-networking process of concatenating layer-specific headers or trailers with a service data unit (i.e. a payload) for transmitting information over computer networks. Deencapsulation (or de-encapsulation) is the reverse computer-networking process for receiving information; it removes from the protocol data unit (PDU) a previously concatenated header or trailer that an underlying communications layer transmitted. Encapsulation and deencapsulation allow the design of modular communication protocols so to logically separate the function of each communications layer, and abstract the structure of the communicated information over the other communications layers. These two processes are common features of the computer-networking models and protocol suites, like in the OSI model and internet protocol suite. However, encapsulation/deencapsulation processes can also serve as malicious features like in the tunneling protocols.Raman, D., Sutter, B. ... [...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 ran ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
