1.
OSI model
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Its goal is the interoperability of diverse communication systems with standard protocols. The model partitions a communication system into abstraction layers, the original version of the model defined seven layers. A layer serves the layer above it and is served by the layer below it, two instances at the same layer are visualized as connected by a horizontal connection in that layer. The model is a product of the Open Systems Interconnection project at the International Organization for Standardization and these two international standards bodies each developed a document that defined similar networking models. In 1983, these two documents were merged to form a standard called The Basic Reference Model for Open Systems Interconnection, the standard is usually referred to as the Open Systems Interconnection Reference Model, the OSI Reference Model, or simply the OSI model. It was published in 1984 by both the ISO, as standard ISO7498, and the renamed CCITT as standard X.200. OSI had two components, an abstract model of networking, called the Basic Reference Model or seven-layer model. The concept of a model was provided by the work of Charles Bachman at Honeywell Information Services. Various aspects of OSI design evolved from experiences with the ARPANET, NPLNET, EIN, CYCLADES network, the new design was documented in ISO7498 and its various addenda. In this model, a system was divided into layers. Within each layer, one or more entities implement its functionality, each entity interacted directly only with the layer immediately beneath it, and provided facilities for use by the layer above it. Protocols enable an entity in one host to interact with an entity at the same layer in another host. Service definitions abstractly described the functionality provided to an -layer by an layer, the OSI standards documents are available from the ITU-T as the X. 200-series of recommendations. Some of the specifications were also available as part of the ITU-T X series. The equivalent ISO and ISO/IEC standards for the OSI model were available from ISO, the recommendation X.200 describes seven layers, labeled 1 to 7. Layer 1 is the lowest layer in this model, at each level N, two entities at the communicating devices exchange protocol data units by means of a layer N protocol. Each PDU contains a payload, called the service data unit, data processing by two communicating OSI-compatible devices is done as such, The data to be transmitted is composed at the topmost layer of the transmitting device into a protocol data unit. The PDU is passed to layer N-1, where it is known as the service data unit, at layer N-1 the SDU is concatenated with a header, a footer, or both, producing a layer N-1 PDU
2.
Abstraction layer
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In the Unix operating system, most types of input and output operations are considered to be streams of bytes read from a device or written to a device. This stream of model is used for file I/O, socket I/O. The devices physical characteristics are mediated by the system which in turn presents an abstract interface that allows the programmer to read. The operating system performs the actual transformation needed to read. Most graphics libraries such as OpenGL provide a graphical device model as an interface. The library is responsible for translating the commands provided by the programmer into the specific device commands needed to draw the graphical elements and objects, in computer science, an abstraction level is a generalization of a model or algorithm, away from any specific implementation. These generalizations arise from broad similarities that are best encapsulated by models that express similarities present in various specific implementations, a layer is on top of another, because it depends on it. Every layer can exist without the layers above it, and requires the layers below it to function, frequently abstraction layers can be composed into a hierarchy of abstraction levels. The ISO-OSI networking model comprises seven abstraction layers, a famous aphorism of David Wheeler is All problems in computer science can be solved by another level of indirection. This is often deliberately misquoted with abstraction substituted for indirection and it is also sometimes misattributed to Butler Lampson. Kevlin Henneys corollary to this is. except for the problem of too many layers of indirection, firmware may include only low-level software, but can also include all software, including an operating system and applications. A distinction can also be made from low-level programming languages like VHDL, machine language, assembly language to compiled languages and interpreter, hardware abstraction Application programming interface Application binary interface Database Information hiding Layer – in object-oriented design Protection ring Software engineering
3.
Session Initiation Protocol
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The Session Initiation Protocol is a communications protocol for signaling, for the purpose of controlling multimedia communication sessions. Internet telephony, business IP telephone systems, service providers and carriers use SIP, SIP can be used to set up and control voice and video calls, as well as instant messaging. The most common application of SIP is the setup and termination of Voice over IP telephone calls, SIP implements the functions of the session layer In the OSI 7-Layer Reference Model. In the 4-layer DoD Internet Model, SIP is an application layer protocol, SIP was designed to be independent of the underlying transport layer. The protocol defines the messages that are sent between endpoints, which govern establishment, termination and other elements of a call. SIP can be used for creating, modifying and terminating sessions consisting of one or several media streams and it is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol and the Simple Mail Transfer Protocol. SIP works in conjunction with other protocols that specify the media format, RTP Protocol data units may be encrypted byTransport Layer Security for secure transmission. SIP was originally designed by Mark Handley, Henning Schulzrinne, Eve Schooler, the protocol was standardized as RFC2543 in 1999. In November 2000, SIP was accepted as a 3GPP signaling protocol, in June 2002 the specification was revised in RFC3261 and various extensions and clarifications have been published since. The protocol was designed with the vision to new multimedia applications. It has been extended for video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP, SIP is distinguished by its proponents for having roots in the Internet community rather than in the telecommunications industry. SIP has been standardized primarily by the IETF, while other protocols, SIP by itself does not define these features, rather, its focus is call setup and signaling. The features that permit familiar telephone-like operations are performed by proxy servers, implementation and terminology are different in the SIP world compared to the PSTN but, to the end-user, the behavior is similar. SIP is only involved in the portion of a media communication session, primarily used to set up. SIP can be used to establish two-party or multiparty sessions and it also allows modification of existing calls. The modification can involve changing addresses or ports, inviting more participants, SIP has also found applications in messaging applications, such as instant messaging, and event subscription and notification. SIP works in concert with other protocols to specify the media format and coding. For call setup, the body of a SIP message contains a Session Description Protocol data unit, Voice and video media is typically specified to be communicated between the terminals using the Real-time Transport Protocol or Secure Real-Time Transport Protocol
4.
Domain Name System
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The Domain Name System is a hierarchical decentralized naming system for computers, services, or other resources connected to the Internet or a private network. It associates various information with domain names assigned to each of the participating entities, by providing a worldwide, distributed directory service, the Domain Name System is an essential component of the functionality of the Internet, that has been in use since 1985. The Domain Name System delegates the responsibility of assigning domain names, Network administrators may delegate authority over sub-domains of their allocated name space to other name servers. This mechanism provides distributed and fault tolerant service and was designed to avoid a large central database. The Domain Name System also specifies the technical functionality of the service that is at its core. It defines the DNS protocol, a specification of the data structures and data communication exchanges used in the DNS. Historically, other directory services preceding DNS were not scalable to large or global directories as they were based on text files. The Internet maintains two principal namespaces, the domain name hierarchy and the Internet Protocol address spaces, the Domain Name System maintains the domain name hierarchy and provides translation services between it and the address spaces. Internet name servers and a communication protocol implement the Domain Name System, a DNS name server is a server that stores the DNS records for a domain, a DNS name server responds with answers to queries against its database. The most common types of stored in the DNS database are for Start of Authority, IP addresses, SMTP mail exchangers, name servers, pointers for reverse DNS lookups. As a general purpose database, the DNS has also used in combating unsolicited email by storing a real-time blackhole list. The DNS database is stored in a structured zone file. An often-used analogy to explain the Domain Name System is that it serves as the book for the Internet by translating human-friendly computer hostnames into IP addresses. For example, the name www. example. com translates to the addresses 93.184.216.119 and 2606,2800,220, 6d, 26bf,1447,1097. Unlike a phone book, DNS can be updated, allowing a services location on the network to change without affecting the end users. Users take advantage of this when they use meaningful Uniform Resource Locators, an important and ubiquitous function of DNS is its central role in distributed Internet services such as cloud services and content delivery networks. When a user accesses a distributed Internet service using a URL and this process of using the DNS to assign proximal servers to users is key to providing faster and more reliable responses on the Internet and is widely used by most major Internet services. The DNS reflects the structure of administrative responsibility in the Internet, each subdomain is a zone of administrative autonomy delegated to a manager
5.
File Transfer Protocol
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The File Transfer Protocol is a standard network protocol used for the transfer of computer files from a server to a client using the Client–server model on a computer network. FTP is built on a model architecture and uses separate control. FTP users may authenticate themselves with a clear-text sign-in protocol, normally in the form of a username and password, for secure transmission that protects the username and password, and encrypts the content, FTP is often secured with SSL/TLS. SSH File Transfer Protocol is sometimes used instead, but is technologically different. The original specification for the File Transfer Protocol was written by Abhay Bhushan, until 1980, FTP ran on NCP, the predecessor of TCP/IP. The protocol was replaced by a TCP/IP version, RFC765 and RFC959. FTP may run in active or passive mode, which determines how the connection is established. In both cases, the client creates a TCP control connection from a random, usually an unprivileged, in active mode, the client starts listening for incoming data connections from the server on port M. It sends the FTP command PORT M to inform the server on which port it is listening, the server then initiates a data channel to the client from its port 20, the FTP server data port. In situations where the client is behind a firewall and unable to accept incoming TCP connections, both modes were updated in September 1998 to support IPv6. Further changes were introduced to the mode at that time. The server responds over the connection with three-digit status codes in ASCII with an optional text message. For example,200 means that the last command was successful, the numbers represent the code for the response and the optional text represents a human-readable explanation or request. An ongoing transfer of data over the data connection can be aborted using an interrupt message sent over the control connection. While transferring data over the network, four data representations can be used, ASCII mode, Data is converted, if needed, from the sending hosts character representation to 8-bit ASCII before transmission, and to the receiving hosts character representation. As a consequence, this mode is inappropriate for files that contain other than plain text. Image mode, The sending machine sends each file byte by byte, EBCDIC mode, Used for plain text between hosts using the EBCDIC character set. Local mode, Allows two computers with identical setups to send data in a format without the need to convert it to ASCII
6.
Gopher (protocol)
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The Gopher protocol /ˈɡoʊfər/ is a TCP/IP application layer protocol designed for distributing, searching, and retrieving documents over the Internet. The Gopher ecosystem is often regarded as the predecessor of the World Wide Web. The protocol was invented by a led by Mark P. McCahill at the University of Minnesota. It offers some features not natively supported by the Web and imposes a much stronger hierarchy on information stored on it, more recent Gopher revisions and graphical clients added support for multimedia. Gopher was preferred by many network administrators for using fewer resources than Web services. Gophers hierarchical structure provided a platform for the first large-scale electronic library connections, Gopher has been described by some enthusiasts as faster and more efficient and so much more organised than todays Web services. The Gopher protocol is still in use by enthusiasts, and although it has been almost entirely supplanted by the Web, a small population of actively maintained servers remains. Gopher system was released in mid-1991 by Mark P. McCahill, Farhad Anklesaria, Paul Lindner, Daniel Torrey and its central goals were, as stated in RFC1436, A file-like hierarchical arrangement that would be familiar to users. A system that can be created quickly and inexpensively, extending the file system metaphor, such as searches. By 1992, the method of locating someones e-mail address was to find their organizations CCSO nameserver entry in Gopher. The name was coined by Anklesaria as a play on several meanings of the word gopher, the University of Minnesota mascot is the gopher, a gofer is an assistant who goes for things, and a gopher burrows through the ground to reach a desired location. The World Wide Web was in its infancy in 1991, by the late 1990s, Gopher had largely ceased expanding. Several factors contributed to Gophers stagnation, In February 1993, the University of Minnesota announced that it would charge licensing fees for the use of its implementation of the Gopher server. As a consequence of this, some users were concerned that a fee would also be charged for independent implementations. Users were scared away from Gopher technology, to the advantage of the Web, in September 2000, the University of Minnesota re-licensed its Gopher software under the GNU General Public License. Gopher client functionality was duplicated by early Web browsers, such as Mosaic. Gopher has a rigid structure compared to the free-form HTML of the Web. With Gopher, every document has a format and type
7.
Hypertext Transfer Protocol
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The Hypertext Transfer Protocol is an application protocol for distributed, collaborative, and hypermedia information systems. HTTP is the foundation of data communication for the World Wide Web, Hypertext is structured text that uses logical links between nodes containing text. HTTP is the protocol to exchange or transfer hypertext, development of HTTP was initiated by Tim Berners-Lee at CERN in 1989. Standards development of HTTP was coordinated by the Internet Engineering Task Force and the World Wide Web Consortium, culminating in the publication of a series of Requests for Comments. The first definition of HTTP/1.1, the version of HTTP in common use, occurred in RFC2068 in 1997, although this was obsoleted by RFC2616 in 1999 and then again by RFC7230 and family in 2014. A later version, the successor HTTP/2, was standardized in 2015, HTTP functions as a request–response protocol in the client–server computing model. A web browser, for example, may be the client, the client submits an HTTP request message to the server. The server, which provides resources such as HTML files and other content, or performs other functions on behalf of the client, the response contains completion status information about the request and may also contain requested content in its message body. A web browser is an example of a user agent, other types of user agent include the indexing software used by search providers, voice browsers, mobile apps, and other software that accesses, consumes, or displays web content. HTTP is designed to permit intermediate network elements to improve or enable communications between clients and servers, high-traffic websites often benefit from web cache servers that deliver content on behalf of upstream servers to improve response time. Web browsers cache previously accessed web resources and reuse them when possible to network traffic. HTTP proxy servers at private network boundaries can facilitate communication for clients without a globally routable address, HTTP is an application layer protocol designed within the framework of the Internet protocol suite. Its definition presumes an underlying and reliable transport layer protocol, however HTTP can be adapted to use unreliable protocols such as the User Datagram Protocol, for example in HTTPU and Simple Service Discovery Protocol. HTTP resources are identified and located on the network by Uniform Resource Locators, using the Uniform Resource Identifiers schemes http, uRIs and hyperlinks in HTML documents form inter-linked hypertext documents. HTTP/1.1 is a revision of the original HTTP, in HTTP/1.0 a separate connection to the same server is made for every resource request. HTTP/1.1 can reuse a connection multiple times to download images, scripts, stylesheets, HTTP/1.1 communications therefore experience less latency as the establishment of TCP connections presents considerable overhead. Tim Berners-Lee and his team at CERN are credited with inventing the original HTTP along with HTML and the technology for a web server. Berners-Lee first proposed the WorldWideWeb project in 1989—now known as the World Wide Web, the first version of the protocol had only one method, namely GET, which would request a page from a server
8.
Network File System (protocol)
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NFS, like many other protocols, builds on the Open Network Computing Remote Procedure Call system. The NFS is a standard defined in Request for Comments. Sun used version 1 only for in-house experimental purposes, Version 2 of the protocol originally operated only over User Datagram Protocol. Its designers meant to keep the server side stateless, with locking implemented outside of the core protocol, people involved in the creation of NFS version 2 include Russel Sandberg, Bob Lyon, Bill Joy, Steve Kleiman, and others. The Virtual File System interface allows an implementation, reflected in a simple protocol. By February 1986, implementations were demonstrated for operating systems such as System V release 2, DOS, nFSv2 only allows the first 2 GB of a file to be read due to 32-bit limitations. The first NFS Version 3 proposal within Sun Microsystems was created not long after the release of NFS Version 2, the principal motivation was an attempt to mitigate the performance issue of the synchronous write operation in NFS Version 2. By July 1992, implementation practice had solved many shortcomings of NFS Version 2 and this became an acute pain point for Digital Equipment Corporation with the introduction of a 64-bit version of Ultrix to support their newly released 64-bit RISC processor, the Alpha 21064. At the time of introduction of Version 3, vendor support for TCP as a transport-layer protocol began increasing. Using TCP as a transport made using NFS over a WAN more feasible, Version 4, influenced by Andrew File System and Server Message Block, includes performance improvements, mandates strong security, and introduces a stateful protocol. Version 4 became the first version developed with the Internet Engineering Task Force after Sun Microsystems handed over the development of the NFS protocols, NFS version 4.2 is currently being developed. WebNFS, an extension to Version 2 and Version 3, allows NFS to integrate easily into Web-browsers. In 2007 Sun Microsystems open-sourced their client-side WebNFS implementation, various side-band protocols have become associated with NFS. It is also available to operating systems such as Acorn RISC OS, the classic Mac OS, OpenVMS, MS-DOS, Microsoft Windows, Novell NetWare, and IBM AS/400. Alternative remote file access protocols include the Server Message Block, Apple Filing Protocol, NetWare Core Protocol, haiku recently added NFSv4 support as part of a Google Summer of Code project. The server administrator determines what to make available, exporting the names and parameters of directories, typically using the configuration file. The server security-administration ensures that it can recognize and approve validated clients, the server network configuration ensures that appropriate clients can negotiate with it through any firewall system. The client machine requests access to exported data, typically by issuing a mount command, if all goes well, users on the client machine can then view and interact with mounted filesystems on the server within the parameters permitted
9.
Network Time Protocol
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Network Time Protocol is a networking protocol for clock synchronization between computer systems over packet-switched, variable-latency data networks. In operation since before 1985, NTP is one of the oldest Internet protocols in current use, NTP was designed by David L. Mills of the University of Delaware. NTP is intended to synchronize all participating computers to within a few milliseconds of Coordinated Universal Time and it uses a modified version of Marzullos algorithm to select accurate time servers and is designed to mitigate the effects of variable network latency. NTP can usually maintain time to within tens of milliseconds over the public Internet, asymmetric routes and network congestion can cause errors of 100 ms or more. The protocol is described in terms of a client-server model. Implementations send and receive timestamps using the User Datagram Protocol on port number 123 and they can also use broadcasting or multicasting, where clients passively listen to time updates after an initial round-trip calibrating exchange. NTP supplies a warning of any impending leap second adjustment, the current protocol is version 4, which is a proposed standard as documented in RFC5905. It is backward compatible with version 3, specified in RFC1305, the technology was later described in the 1981 Internet Engineering Note 173 and a public protocol was developed from it that was documented in RFC778. Other related network tools were available both then and now and they include the Daytime and Time protocols for recording the time of events, as well as the ICMP Timestamp and IP Timestamp option. In 1985, NTPv0 was implemented in both Fuzzball and Unix, and the NTP packet header and round-trip delay and offset calculations, in 1988, a much more complete specification of the NTPv1 protocol, with associated algorithms, was published in RFC1059. It drew on the results and clock filter algorithm documented in RFC956 and was the first version to describe the client-server and peer-to-peer modes. In 1989, RFC1119 was published defining NTPv2 by means of a state machine and it introduced a management protocol and cryptographic authentication scheme which have both survived into NTPv4. The design of NTP was criticized for lacking formal correctness principles by the DTSS community and their alternative design included Marzullos algorithm, a modified version of which was promptly added to NTP. The bulk of the algorithms from this era have also survived into NTPv4. In 1992, RFC1305 defined NTPv3, in subsequent years, as new features were added and algorithm improvements were made, it became apparent that a new protocol version was required. In 2010, RFC5905 was published containing a proposed specification for NTPv4, but the protocol has significantly moved on since then, and as of 2014, an updated RFC has yet to be published. Following the retirement of Mills from the University of Delaware, the implementation is currently maintained as an open source project led by Harlan Stenn. NTP uses a hierarchical, semi-layered system of time sources, each level of this hierarchy is termed a stratum and is assigned a number starting with zero at the top
10.
Simple Mail Transfer Protocol
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Simple Mail Transfer Protocol is an Internet standard for electronic mail transmission. First defined by RFC821 in 1982, it was last updated in 2008 with Extended SMTP additions by RFC5321, for retrieving messages, client applications usually use either IMAP or POP3. SMTP communication between mail servers uses TCP port 25, Mail clients on the other hand, often submit the outgoing emails to a mail server on port 587. Despite being deprecated, mail providers sometimes still permit the use of nonstandard port 465 for this purpose, SMTP connections secured by SSL, known as SMTPS, can be made using STARTTLS. Various forms of electronic messaging were used in the 1960s. People communicated with one another using systems developed for specific mainframe computers, as more computers were interconnected, especially in the US Governments ARPANET, standards were developed to allow users of different systems to email one another. SMTP grew out of these standards developed during the 1970s, fewer than 50 hosts were connected to the ARPANET at this time. Further implementations include FTP Mail and Mail Protocol, both from 1973, development work continued throughout the 1970s, until the ARPANET transitioned into the modern Internet around 1980. Jon Postel then proposed a Mail Transfer Protocol in 1980 that began to remove the reliance on FTP. SMTP was published as RFC788 in November 1981, also by Postel, the SMTP standard was developed around the same time as Usenet, a one-to-many communication network with some similarities. SMTP became widely used in the early 1980s, at the time, it was a complement to Unix to Unix Copy Program mail, which was better suited for handling email transfers between machines that were intermittently connected. SMTP, on the hand, works best when both the sending and receiving machines are connected to the network all the time. Both use a store and forward mechanism and are examples of push technology, though Usenets newsgroups are still propagated with UUCP between servers, UUCP as a mail transport has virtually disappeared along with the bang paths it used as message routing headers. Sendmail, released with 4. 1cBSD, right after RFC788, was one of the first mail transfer agents to implement SMTP, over time, as BSD Unix became the most popular operating system on the Internet, sendmail became the most common MTA. Some other popular SMTP server programs include Postfix, qmail, Novell GroupWise, Exim, Novell NetMail, Microsoft Exchange Server, Message submission and SMTP-AUTH were introduced in 1998 and 1999, both describing new trends in email delivery. Originally, SMTP servers were typically internal to an organization, receiving mail for the organization from the outside and this behavior is helpful when the message being fixed is an initial submission, but dangerous and harmful when the message originated elsewhere and is being relayed. Cleanly separating mail into submission and relay was seen as a way to permit, as spam became more prevalent, it was also seen as a way to provide authorization for mail being sent out from an organization, as well as traceability. This separation of relay and submission quickly became a foundation for modern email security practices, as this protocol started out purely ASCII text-based, it did not deal well with binary files, or characters in many non-English languages
11.
Simple Network Management Protocol
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Devices that typically support SNMP include cable modems, routers, switches, servers, workstations, printers, and more. SNMP is widely used in management for network monitoring. SNMP exposes management data in the form of variables on the managed systems organized in a management information base which describe the system status and these variables can then be remotely queried by managing applications. Three significant versions of SNMP have been developed and deployed, SNMPv1 is the original version of the protocol. More recent versions, SNMPv2c and SNMPv3, feature improvements in performance, SNMP is a component of the Internet Protocol Suite as defined by the Internet Engineering Task Force. It consists of a set of standards for network management, including an application protocol, a database schema. In typical uses of SNMP one or more computers, called managers, have the task of monitoring or managing a group of hosts or devices on a computer network. Each managed system executes, at all times, a component called an agent which reports information via SNMP to the manager. Managed devices exchange node-specific information with the NMSs, an agent is a network-management software module that resides on a managed device. An agent has local knowledge of management information and translates that information to or from an SNMP-specific form, a network management station executes applications that monitor and control managed devices. NMSs provide the bulk of the processing and memory resources required for network management, one or more NMSs may exist on any managed network. SNMP agents expose management data on the systems as variables. The protocol also permits active management tasks, such as modifying and applying a new configuration through remote modification of these variables, the variables accessible via SNMP are organized in hierarchies. SNMP itself does not define which information a managed system should offer, rather, SNMP uses an extensible design which allows applications to define their own hierarchies and metadata. These hierarchies, and other metadata, are described by management information bases, MIBs describe the structure of the management data of a device subsystem, they use a hierarchical namespace containing object identifiers. Each OID identifies a variable that can be read or set via SNMP, MIBs use the notation defined by Structure of Management Information Version 2.0, a subset of ASN.1. SNMP operates in the Application Layer of the Internet Protocol Suite, the SNMP agent receives requests on UDP port 161. The manager may send requests from any available port to port 161 in the agent
12.
Dynamic Host Configuration Protocol
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The Dynamic Host Configuration Protocol is a standardized network protocol used on Internet Protocol networks. The DHCP is controlled by a DHCP server that dynamically distributes network configuration parameters, such as IP addresses, a router or a residential gateway can be enabled to act as a DHCP server. A DHCP server enables computers to request IP addresses and networking parameters automatically, in the absence of a DHCP server, each computer or other device on the network needs to be statically assigned to an IP address. TCP/IP defines how devices on one network communicate with devices on another network, a DHCP server can manage TCP/IP settings for devices on a network, by automatically or dynamically assigning Internet Protocol addresses to the devices. As of 2011, networks ranging in size from home networks to large campus networks, most residential network routers receive a globally unique IP address within the provider network. Within a local network, a DHCP server assigns a local IP address to each device connected to the network, the DHCP operates based on the client–server model. When a computer or other device connects to a network, the DHCP client software sends a broadcast query requesting the necessary information, any DHCP server on the network may service the request. The DHCP server manages a pool of IP addresses and information about client configuration parameters such as default gateway, domain name, the name servers, a client typically queries for this information immediately after booting, and periodically thereafter before the expiration of the information. When a DHCP client refreshes an assignment, it requests the same parameter values. On large networks that consist of links, a single DHCP server may service the entire network when aided by DHCP relay agents located on the interconnecting routers. Such agents relay messages between DHCP clients and DHCP servers located on different subnets, the request-and-grant process uses a lease concept with a controllable time period, allowing the DHCP server to reclaim IP addresses that are not renewed. Automatic allocation The DHCP server permanently assigns an IP address to a client from the range defined by the administrator. This is like dynamic allocation, but the DHCP server keeps a table of past IP address assignments, manual allocation The DHCP server issues a private IP address dependent upon each clients MAC address, based on a predefined mapping by the administrator. If no match for the clients MAC address is found, the server may or may not optionally fall back to either Dynamic or Automatic allocation, DHCP is used for Internet Protocol version 4, as well as for IPv6. While both versions serve the purpose, the details of the protocol for IPv4 and IPv6 differ sufficiently that they may be considered separate protocols. For the IPv6 operation, devices may alternatively use stateless address autoconfiguration, IPv6 hosts may also use link-local addressing to achieve operations restricted to the local network link. In 1984, the Reverse Address Resolution Protocol, defined in RFC903, was introduced to allow simple devices such as diskless workstations to dynamically obtain a suitable IP address. However, because it acted at the link layer it made implementation difficult on many server platforms
13.
NETCONF
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The Network Configuration Protocol is a network management protocol developed and standardized by the IETF. It was developed in the NETCONF working group and published in December 2006 as RFC4741 and later revised in June 2011, the NETCONF protocol specification is an Internet Standards Track document. NETCONF provides mechanisms to install, manipulate, and delete the configuration of network devices and its operations are realized on top of a simple Remote Procedure Call layer. The NETCONF protocol uses an Extensible Markup Language based data encoding for the data as well as the protocol messages. The protocol messages are exchanged on top of a transport protocol. The NETCONF protocol can be partitioned into four layers, The Content layer consists of configuration data. The Operations layer defines a set of base operations to retrieve. The Messages layer provides a mechanism for encoding remote procedure calls, the Secure Transport layer provides a secure and reliable transport of messages between a client and a server. The NETCONF protocol has been implemented in network devices such as routers, one particular strength of NETCONF is its support for robust configuration change using transactions involving a number of devices. The IETF developed the Simple Network Management Protocol in the late 1980s, in June 2002, the Internet Architecture Board and key members of the IETFs network management community got together with network operators to discuss the situation. The results of this meeting are documented in RFC3535 and it turned out that operators were primarily using proprietary Command Line Interfaces to configure their devices. This had a number of features that the operators liked, including the fact that it was text-based, in addition, many equipment vendors did not provide the option to completely configure their devices via SNMP. As operators generally liked to write scripts to manage their boxes. Most notably was the nature of the output. The content and formatting of output was prone to change in unpredictable ways, around this same time, Juniper Networks had been using an XML-based network management approach. This was brought to the IETF and shared with the broader community, collectively, these two events led the IETF in May 2003 to the creation of the NETCONF working group. This working group was chartered to work on a network configuration protocol, the first version of the base NETCONF protocol was published as RFC4741 in December 2006. Several extensions were published in subsequent years, a revised version of the base NETCONF protocol was published as RFC6241 in June 2011
14.
Transmission Control Protocol
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The Transmission Control Protocol is one of the main protocols of the Internet protocol suite. It originated in the initial implementation in which it complemented the Internet Protocol. Therefore, the suite is commonly referred to as TCP/IP. TCP provides reliable, ordered, and error-checked delivery of a stream of octets between applications running on hosts communicating by an IP network, major Internet applications such as the World Wide Web, email, remote administration, and file transfer rely on TCP. Applications that do not require reliable data stream service may use the User Datagram Protocol, during May 1974, the Institute of Electrical and Electronic Engineers published a paper titled A Protocol for Packet Network Intercommunication. The papers authors, Vint Cerf and Bob Kahn, described a protocol for sharing resources using packet-switching among the nodes. A central control component of this model was the Transmission Control Program that incorporated both connection-oriented links and datagram services between hosts, the model became known informally as TCP/IP, although formally it was henceforth termed the Internet Protocol Suite. The Transmission Control Protocol provides a service at an intermediate level between an application program and the Internet Protocol. It provides host-to-host connectivity at the Transport Layer of the Internet model, an application does not need to know the particular mechanisms for sending data via a link to another host, such as the required packet fragmentation on the transmission medium. At the transport layer, the protocol handles all handshaking and transmission details, at the lower levels of the protocol stack, due to network congestion, traffic load balancing, or other unpredictable network behaviour, IP packets may be lost, duplicated, or delivered out of order. TCP detects these problems, requests re-transmission of lost data, rearranges out-of-order data, If the data still remains undelivered, the source is notified of this failure. Once the TCP receiver has reassembled the sequence of octets originally transmitted, thus, TCP abstracts the applications communication from the underlying networking details. TCP is optimized for accurate delivery rather than timely delivery and can incur relatively long delays while waiting for messages or re-transmissions of lost messages. Therefore, it is not particularly suitable for applications such as Voice over IP. For such applications, protocols like the Real-time Transport Protocol operating over the User Datagram Protocol are usually recommended instead, TCP is a reliable stream delivery service which guarantees that all bytes received will be identical with bytes sent and in the correct order. Since packet transfer by many networks is not reliable, a known as positive acknowledgement with re-transmission is used to guarantee reliability. This fundamental technique requires the receiver to respond with an acknowledgement message as it receives the data, the sender keeps a record of each packet it sends and maintains a timer from when the packet was sent. The sender re-transmits a packet if the timer expires before receiving the message acknowledgement, the timer is needed in case a packet gets lost or corrupted
15.
Stream Control Transmission Protocol
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In computer networking, the Stream Control Transmission Protocol is a transport-layer protocol, serving in a similar role to the popular protocols TCP and UDP. It is standardized by IETF in RFC4960, in the absence of native SCTP support in operating systems it is possible to tunnel SCTP over UDP, as well as mapping TCP API calls to SCTP ones. The reference implementation was released as part of FreeBSD version 7 and it has subsequently been widely ported. The IETF Signaling Transport working group defined the protocol in 2000, SCTP applications submit their data to be transmitted in messages to the SCTP transport layer. SCTP places messages and control information into separate chunks, each identified by a chunk header, the protocol can fragment a message into a number of data chunks, but each data chunk contains data from only one user message. SCTP bundles the chunks into SCTP packets, the SCTP packet, which is submitted to the Internet Protocol, consists of a packet header, SCTP control chunks, followed by SCTP data chunks. One can characterize SCTP as message-oriented, meaning it transports a sequence of messages, as in UDP, in SCTP a sender sends a message in one operation, and that exact message is passed to the receiving application process in one operation. In contrast, TCP is a protocol, transporting streams of bytes reliably. However TCP does not allow the receiver to know how many times the application called on the TCP transport passing it groups of bytes to be sent out. The term multi-streaming refers to the capability of SCTP to transmit several independent streams of chunks in parallel, in essence, it involves bundling several connections into a single SCTP association, operating on messages rather than bytes. TCP preserves byte order in the stream by including a number with each segment. SCTP, on the hand, assigns a sequence number to each message sent in a stream. This allows independent ordering of messages in different streams, however, message ordering is optional in SCTP, a receiving application may choose to process messages in the order of receipt instead of in the order of sending. Features of SCTP include, Multihoming support in one or both endpoints of a connection can consist of more than one IP address, enabling transparent fail-over between redundant network paths. Delivery of chunks within independent streams eliminate unnecessary head-of-line blocking, as opposed to TCP byte-stream delivery, path selection and monitoring to select a primary data transmission path and test the connectivity of the transmission path. Validation and acknowledgment mechanisms protect against flooding attacks and provide notification of duplicated or missing data chunks, improved error detection suitable for Ethernet jumbo frames. The designers of SCTP originally intended it for the transport of telephony over Internet Protocol and this IETF effort is known as SIGTRAN. In the meantime, other uses have been proposed, for example, TCP has provided the primary means to transfer data reliably across the Internet
16.
Internet Protocol
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The Internet Protocol is the principal communications protocol in the Internet protocol suite for relaying datagrams across network boundaries. Its routing function enables internetworking, and essentially establishes the Internet, IP has the task of delivering packets from the source host to the destination host solely based on the IP addresses in the packet headers. For this purpose, IP defines packet structures that encapsulate the data to be delivered and it also defines addressing methods that are used to label the datagram with source and destination information. The Internet protocol suite is often referred to as TCP/IP. The first major version of IP, Internet Protocol Version 4, is the dominant protocol of the Internet and its successor is Internet Protocol Version 6. The Internet Protocol is responsible for addressing hosts and for routing datagrams from a source host to a destination host across one or more IP networks. For this purpose, the Internet Protocol defines the format of packets, each datagram has two components, a header and a payload. The IP header is tagged with the source IP address, the destination IP address, the payload is the data that is transported. This method of nesting the data payload in a packet with a header is called encapsulation, IP addressing entails the assignment of IP addresses and associated parameters to host interfaces. The address space is divided into networks and subnetworks, involving the designation of network or routing prefixes, IP routing is performed by all hosts, as well as routers, whose main function is to transport packets across network boundaries. Routers communicate with one another via specially designed routing protocols, either interior gateway protocols or exterior gateway protocols, IP routing is also common in local networks. For example, many Ethernet switches support IP multicast operations and these switches use IP addresses and Internet Group Management Protocol to control multicast routing but use MAC addresses for the actual routing. The versions currently relevant are IPv4 and IPv6, in May 1974, the Institute of Electrical and Electronic Engineers published a paper entitled A Protocol for Packet Network Intercommunication. The papers authors, Vint Cerf and Bob Kahn, described a protocol for sharing resources using packet switching among network nodes. A central control component of this model was the Transmission Control Program that incorporated both connection-oriented links and datagram services between hosts, the model became known as the Department of Defense Internet Model and Internet protocol suite, and informally as TCP/IP. IP versions 0 to 3 were experimental versions, used between 1977 and 1979, IEN26, dated February 1978 describes a version of the IP header that uses a 1-bit version field. IEN28, dated February 1978 describes IPv2, IEN41, dated June 1978 describes the first protocol to be called IPv4. The IP header is different from the modern IPv4 header, IEN44, dated June 1978 describes another version of IPv4, also with a header different from the modern IPv4 header
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IPv4
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Internet Protocol version 4 is the fourth version of the Internet Protocol. It is one of the protocols of standards-based internetworking methods in the Internet. It still routes most Internet traffic today, despite the deployment of a successor protocol. IPv4 is described in IETF publication RFC791, replacing an earlier definition, IPv4 is a connectionless protocol for use on packet-switched networks. It operates on a best effort delivery model, in that it does not guarantee delivery and these aspects, including data integrity, are addressed by an upper layer transport protocol, such as the Transmission Control Protocol. IPv4 uses 32-bit addresses which limits the space to 4294967296 addresses. IPv4 reserves special address blocks for private networks and multicast addresses, IPv4 addresses may be represented in any notation expressing a 32-bit integer value. They are most often written in the notation, which consists of four octets of the address expressed individually in decimal numbers. For example, the quad-dotted IP address 192.0.2.235 represents the 32-bit decimal number 3221226219, which in hexadecimal format is 0xC00002EB. This may also be expressed in dotted hex format as 0xC0. 0x00. 0x02. 0xEB, or with octal byte values as 0300.0000.0002.0353. In the original design of IPv4, an IP address was divided into two parts, the identifier was the most significant octet of the address, and the host identifier was the rest of the address. The latter was called the rest field. This structure permitted a maximum of 256 network identifiers, which was found to be inadequate. To overcome this limit, the most-significant address octet was redefined in 1981 to create network classes, the revised system defined five classes. Classes A, B, and C had different bit lengths for network identification, the rest of the address was used as previously to identify a host within a network, which meant that each network class had a different capacity for addressing hosts. Class D was defined for multicast addressing and Class E was reserved for future applications, starting around 1985, methods were devised to subdivide IP networks. One method that has proved flexible is the use of the subnet mask. CIDR was designed to permit repartitioning of any address space so that smaller or larger blocks of addresses could be allocated to users, the hierarchical structure created by CIDR is managed by the Internet Assigned Numbers Authority and the regional Internet registries
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IPv6
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IPv6 was developed by the Internet Engineering Task Force to deal with the long-anticipated problem of IPv4 address exhaustion. IPv6 is intended to replace IPv4, every device on the Internet is assigned a unique IP address for identification and location definition. With the rapid growth of the Internet after commercialization in the 1990s, by 1998, the Internet Engineering Task Force had formalized the successor protocol. IPv6 uses a 128-bit address, theoretically allowing 2128, or approximately 3. 4×1038 addresses, the actual number is slightly smaller, as multiple ranges are reserved for special use or completely excluded from use. The total number of possible IPv6 addresses is more than 7. 9×1028 times as many as IPv4, the two protocols are not designed to be interoperable, complicating the transition to IPv6. However, several IPv6 transition mechanisms have been devised to permit communication between IPv4 and IPv6 hosts, IPv6 provides other technical benefits in addition to a larger addressing space. In particular, it permits hierarchical address allocation methods that facilitate route aggregation across the Internet, the use of multicast addressing is expanded and simplified, and provides additional optimization for the delivery of services. Device mobility, security, and configuration aspects have been considered in the design of the protocol, IPv6 was first formally described in Internet standard document RFC2460, published in December 1998. In addition to offering more addresses, IPv6 also implements features not present in IPv4 and it simplifies aspects of address assignment, network renumbering, and router announcements when changing network connectivity providers. It simplifies processing of packets in routers by placing the responsibility for packet fragmentation into the end points, Network security was a design requirement of the IPv6 architecture, and included the original specification of IPsec. IPv6 does not specify interoperability features with IPv4, but essentially creates a parallel, exchanging traffic between the two networks requires translator gateways employing one of several transition mechanisms, such as NAT64, or a tunneling protocol like 6to4, 6in4, or Teredo. Internet Protocol Version 4 was the first publicly used version of the Internet Protocol and it is currently described by IETF publication RFC791, which replaced an earlier definition. IPv4 included a system that used numerical identifiers consisting of 32 bits. These addresses are typically displayed in quad-dotted notation as decimal values of four octets, each in the range 0 to 255, thus, IPv4 provides an addressing capability of 232 or approximately 4.3 billion addresses. Address exhaustion was not initially a concern in IPv4 as this version was originally presumed to be a test of DARPAs networking concepts, during the first decade of operation of the Internet, it became apparent that methods had to be developed to conserve address space. The last unassigned top-level address blocks of 16 million IPv4 addresses were allocated in February 2011 by the Internet Assigned Numbers Authority to the five regional Internet registries. However, each RIR still has available address pools and is expected to continue with standard address allocation policies until one /8 Classless Inter-Domain Routing block remains, after that, only blocks of 1024 addresses will be provided from the RIRs to a local Internet registry. This leaves African Network Information Center as the sole regional internet registry that is using the normal protocol for distributing IPv4 addresses
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Internet Group Management Protocol
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The Internet Group Management Protocol is a communications protocol used by hosts and adjacent routers on IPv4 networks to establish multicast group memberships. IGMP is an part of IP multicast. IGMP can be used for one-to-many networking applications such as streaming video and gaming. IGMP is used on IPv4 networks, Multicast management on IPv6 networks is handled by Multicast Listener Discovery which uses ICMPv6 messaging in contrast to IGMPs bare IP encapsulation. A network designed to deliver a multicast service using IGMP might use this basic architecture, switches featuring IGMP snooping derive useful information by observing these IGMP transactions. Protocol Independent Multicast is then used between the local and remote multicast routers, to direct multicast traffic from the multicast server to many multicast clients, IGMP operates on the network layer, just the same as other network management protocols like ICMP. The IGMP protocol is implemented on a particular host and within a router, a host requests membership to a group through its local router while a router listens for these requests and periodically sends out subscription queries. A single router per subnet is elected to perform this querying function, some multilayer switches include an IGMP querier capability to allow their IGMP snooping features to work in the absence of an IP multicast capability in the larger network. IGMP is vulnerable to attacks, and firewalls commonly allow the user to disable it if not needed. There are three versions of IGMP, as defined by Request for Comments documents of the Internet Engineering Task Force. IGMPv1 is defined by RFC1112, IGMPv2 is defined by RFC2236, IGMPv2 improves over IGMPv1 by adding the ability for a host to signal desire to leave a multicast group. IGMPv3 improves over IGMPv2 mainly by supporting source-specific multicast and Membership Report aggregation, IGMP messages are carried in bare IP packets with IP protocol number 2. There is no transport layer used with IGMP messaging, similar to the Internet Control Message Protocol, there are several types of IGMP messages, Membership Queries, Membership Reports, and Leave Group messages. Membership Queries are sent by multicast routers to determine which multicast addresses are of interest to systems attached to its network, routers periodically send General Queries to refresh the group membership state for all systems on its network. Group-Specific Queries are used for determining the state for a particular multicast address. Group-and-Source-Specific Queries allow the router to determine if any systems desire reception of messages sent to a multicast group from a source address specified in a list of unicast addresses. Where, Type Indicates the message type as follows, Membership Query, Membership Report, the field has a resolution of 100 milliseconds, the value is taken directly. This field is only in Membership Query, in other messages it is set to 0
20.
AppleTalk
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AppleTalk was a proprietary suite of networking protocols developed by Apple Inc. for their Macintosh computers. AppleTalk includes a number of features that allow local area networks to be connected with no setup or the need for a centralized router or server of any sort. Connected AppleTalk-equipped systems automatically assign addresses, update the distributed namespace, AppleTalk was released in 1985, and was the primary protocol used by Apple devices through the 1980s and 1990s. Versions were also released for the IBM PC and compatibles and the Apple IIGS, AppleTalk support was also available in most networked printers, some file servers, and a number of routers. The rise of TCP/IP during the 1990s led to a reimplementation of most of these types of support on that protocol, many of AppleTalks more advanced autoconfiguration features have since been introduced in Bonjour, while Universal Plug and Play serves similar needs. After the release of the Apple Lisa computer in January 1983, known as AppleNet, it was based on the seminal Xerox XNS protocol stack but running on a custom 1 Mbit/s coaxial cable system rather than Xeroxs 2.94 Mbit/s Ethernet. AppleNet was announced early in 1983 with an introduction at the target price of $500 for plug-in AppleNet cards for the Lisa. At that time, early LAN systems were just coming to market, including Ethernet, Token Ring and this was a topic of major commercial effort at the time, dominating shows like the National Computer Conference in Anaheim in May 1983. All of the systems were jockeying for position in the market and it was at this show that Steve Jobs asked Gursharan Sidhu a seemingly innocuous question, Why has networking not caught on. Four months later, in October, AppleNet was cancelled, at the time, they announced that Apple realized that its not in the business to create a networking system. We built and used AppleNet in-house, but we realized if we had shipped it. In January, Jobs announced that they would instead be supporting IBMs Token Ring, through this period, Apple was deep in development of the Macintosh computer. During development, engineers had made the decision to use the Zilog 8530 serial controller chip instead of the lower cost and more common UART to provide serial port connections. The SCC cost about $5 more than a UART, but offered much higher speeds up to 250 kilobits per second, the SCC was chosen because it would allow multiple devices to be attached to the port. Peripherals equipped with similar SCCs could communicate using the built-in protocols and this would eliminate the need for more ports on the back of the machine, and allowed for the elimination of expansion slots for supporting more complex devices. The initial concept was known as AppleBus, envisioning a system controlled by the host Macintosh polling dumb devices in a similar to the modern Universal Serial Bus. The Macintosh team had begun work on what would become the LaserWriter. A series of memos from Bob Belleville clarified these concepts, outlining the Mac, LaserWriter, by late 1983 it was clear that IBMs Token Ring would not be ready in time for the launch of the Mac, and might miss the launch of these other products as well
21.
X.25
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X.25 is an ITU-T standard protocol suite for packet switched wide area network communication. An X.25 WAN consists of packet-switching exchange nodes as the networking hardware, X.25 is a family of protocols that was popular during the 1980s with telecommunications companies and in financial transaction systems such as automated teller machines. X.25 was originally defined by the International Telegraph and Telephone Consultative Committee in a series of drafts and finalized in a publication known as The Orange Book in 1976. While X.25 has, to an extent, been replaced by less complex protocols, especially the Internet protocol. X.25 is one of the oldest packet-switched services available and it was developed before the OSI Reference Model. The protocol suite is designed as three conceptual layers, which correspond closely to the three layers of the seven-layer OSI model. It also supports functionality not found in the OSI network layer, X.25 was developed in the ITU-T Study Group VII based upon a number of emerging data network projects. Various updates and additions were worked into the standard, eventually recorded in the ITU series of books describing the telecommunication systems. These books were published every year with different-colored covers. The X.25 specification is only part of the set of X-Series specifications on public data networks. The public data network was the name given to the international collection of X.25 providers. Their combined network had large global coverage during the 1980s and into the 1990s, publicly accessible X.25 networks were set up in most countries during the 1970s and 1980s, to lower the cost of accessing various online services. Beginning in the early 1990s, in North America, use of X.25 networks started to be replaced by Frame Relay, most systems that required X.25 now use TCP/IP, however it is possible to transport X.25 over TCP/IP when necessary. X.25 networks are still in use throughout the world, a variant called AX.25 is also used widely by amateur packet radio. Racal Paknet, now known as Widanet, is still in operation in many regions of the world, running on an X.25 protocol base. Additionally X.25 is still under heavy use in the business even though a transition to modern protocols like X.400 is without option as X.25 hardware becomes increasingly rare. As recently as March 2006, the United States National Airspace Data Interchange Network has used X.25 to interconnect remote airfields with Air Route Traffic Control Centers, france was one of the last remaining countries where commercial end-user service based on X.25 operated. Known as Minitel it was based on Videotex, itself running on X.25, as planned, service was terminated 30 June 2012
22.
Asynchronous Transfer Mode
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ATM was developed to meet the needs of the Broadband Integrated Services Digital Network, as defined in the late 1980s, and designed to unify telecommunication and computer networks. It was designed for a network that must handle both traditional high-throughput data traffic, and real-time, low-latency content such as voice and video. The reference model for ATM approximately maps to the three lowest layers of the ISO-OSI reference model, network layer, data link layer, and physical layer. ATM is a protocol used over the SONET/SDH backbone of the public switched telephone network and Integrated Services Digital Network. This differs from approaches such as the Internet Protocol or Ethernet that use variable sized packets, ATM uses a connection-oriented model in which a virtual circuit must be established between two endpoints before the actual data exchange begins. ATM eventually became dominated by Internet Protocol only technology, in the ISO-OSI reference model data link layer, the basic transfer units are generically called frames. In ATM these frames are of a length and specifically called cells. Under normal queuing conditions the cells might experience maximum queuing delays, to avoid this issue, all ATM packets, or cells, are the same small size. In addition, the cell structure means that ATM can be readily switched by hardware without the inherent delays introduced by software switched and routed frames. Thus, the designers of ATM utilized small data cells to reduce jitter in the multiplexing of data streams.544 to 45 Mbit/s in the USA, at this rate, a typical full-length 1500 byte data packet would take 77.42 µs to transmit. In a lower-speed link, such as a 1.544 Mbit/s T1 line, a queuing delay induced by several such data packets might exceed the figure of 7.8 ms several times over, in addition to any packet generation delay in the shorter speech packet. This was clearly unacceptable for speech traffic, which needs to have low jitter in the stream being fed into the codec if it is to produce good-quality sound. This allows smoothing out the jitter, but the delay introduced by passage through the buffer would require echo cancellers even in local networks, also, it would have increased the delay across the channel, and conversation is difficult over high-delay channels. Build a system that can provide low jitter to traffic that needs it. Operate on a 1,1 user basis, the design of ATM aimed for a low-jitter network interface. However, cells were introduced into the design to provide short queuing delays while continuing to support datagram traffic, ATM broke up all packets, data, and voice streams into 48-byte chunks, adding a 5-byte routing header to each one so that they could be reassembled later. The choice of 48 bytes was political rather than technical, most of the European parties eventually came around to the arguments made by the Americans, but France and a few others held out for a shorter cell length. With 32 bytes, France would have been able to implement an ATM-based voice network with calls from one end of France to the other requiring no echo cancellation,48 bytes was chosen as a compromise between the two sides
