1.
Internet protocol suite
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The Internet protocol suite is the conceptual model and set of communications protocols used on the Internet and similar computer networks. It is commonly known as TCP/IP because the protocols in the suite are the Transmission Control Protocol. It is occasionally known as the Department of Defense model, because the development of the model was funded by DARPA. The Internet protocol suite provides end-to-end data communication specifying how data should be packetized, addressed, transmitted, routed and received and this functionality is organized into four abstraction layers which are used to sort all related protocols according to the scope of networking involved. Technical standards specifying the Internet protocol suite and many of its constituent protocols are maintained by the Internet Engineering Task Force, the Internet protocol suite model is a simpler model developed prior to the OSI model. The Internet protocol suite resulted from research and development conducted by the Defense Advanced Research Projects Agency in the late 1960s, after initiating the pioneering ARPANET in 1969, DARPA started work on a number of other data transmission technologies. In 1972, Robert E. Cerf credits Hubert Zimmermann and Louis Pouzin, designer of the CYCLADES network, the protocol was implemented as the Transmission Control Program, first published in 1974. Initially, the TCP managed both datagram transmissions and routing, but as the protocol grew, other researchers recommended a division of functionality into protocol layers, postel stated, “we are screwing up in our design of Internet protocols by violating the principle of layering”. Encapsulation of different mechanisms was intended to create an environment where the layers could access only what was needed from the lower layers. A monolithic design would be inflexible and lead to scalability issues, the Transmission Control Program was split into two distinct protocols, the Transmission Control Protocol and the Internet Protocol. The new suite replaced all protocols used previously and this design is known as the end-to-end principle. Using this design, it possible to connect almost any network to the ARPANET, irrespective of the local characteristics. One popular expression is that TCP/IP, the product of Cerf and Kahns work. A computer called a router is provided with an interface to each network and it forwards packets back and forth between them. Originally a router was called gateway, but the term was changed to avoid confusion with other types of gateways, from 1973 to 1974, Cerfs networking research group at Stanford worked out details of the idea, resulting in the first TCP specification. A significant technical influence was the early networking work at Xerox PARC, DARPA then contracted with BBN Technologies, Stanford University, and the University College London to develop operational versions of the protocol on different hardware platforms. Four versions were developed, TCP v1, TCP v2, TCP v3 and IP v3, the last protocol is still in use today. In 1975, a two-network TCP/IP communications test was performed between Stanford and University College London, in November,1977, a three-network TCP/IP test was conducted between sites in the US, the UK, and Norway
2.
Border Gateway Protocol
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Border Gateway Protocol is a standardized exterior gateway protocol designed to exchange routing and reachability information among autonomous systems on the Internet. The protocol is often classified as a path vector protocol but is also classed as a distance-vector routing protocol. The Border Gateway Protocol makes routing decisions based on paths, network policies, BGP may be used for routing within an autonomous system. In this application it is referred to as Interior Border Gateway Protocol, Internal BGP, in contrast, the Internet application of the protocol may be referred to as Exterior Border Gateway Protocol, External BGP, or eBGP. The current version of BGP is version 4, which was published as RFC4271 in 2006, RFC4271 corrected errors, clarified ambiguities, and updated the specification with common industry practices. The major enhancement was the support for Classless Inter-Domain Routing and use of route aggregation to decrease the size of routing tables, bGP4 has been in use on the Internet since 1994. BGP4 is standard for Internet routing, required of most Internet service providers to establish routing between one another, very large private IP networks use BGP internally. An example is the joining of a number of large Open Shortest Path First networks, another reason to use BGP is multihoming a network for better redundancy, either to multiple access points of a single ISP or to multiple ISPs. BGP neighbors, called peers, are established by manual configuration between routers to create a TCP session on port 179, a BGP speaker sends 19-byte keep-alive messages every 60 seconds to maintain the connection. Among routing protocols, BGP is unique in using TCP as its transport protocol, when BGP runs between two peers in the same autonomous system, it is referred to as Internal BGP. When it runs between different autonomous systems, it is called External BGP, other deployment topologies are also possible, such as running eBGP peering inside a VPN tunnel, allowing two remote sites to exchange routing information in a secure and isolated manner. The main difference between iBGP and eBGP peering is in the way routes that were received from one peer are propagated to other peers, for instance, new routes learned from an eBGP peer are typically redistributed to all iBGP peers as well as all other eBGP peers. However, if new routes are learned on an iBGP peering and these route-propagation rules effectively require that all iBGP peers inside an AS are interconnected in a full mesh. How routes are propagated can be controlled in detail via the route-maps mechanism and this mechanism consists of a set of rules. Each rule describes, for routes matching some given criteria, what action should be taken, the action could be to drop the route, or it could be to modify some attributes of the route before inserting it in the routing table. During the peering handshake, when OPEN messages are exchanged, BGP speakers can negotiate optional capabilities of the session, including multiprotocol extensions and various recovery modes. If the multiprotocol extensions to BGP are negotiated at the time of creation and these families include the IPv4, IPv6, IPv4/IPv6 Virtual Private Networks and multicast BGP. Increasingly, BGP is used as a generalized signaling protocol to carry information about routes that may not be part of the global Internet, such as VPNs
3.
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
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.
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
7.
Media Gateway Control Protocol
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The Media Gateway Control Protocol is a signaling and call control communications protocol used in voice over IP telecommunication systems. It implements the media gateway control protocol architecture for controlling media gateways on Internet Protocol networks connected to the switched telephone network. The protocol is a successor to the Simple Gateway Control Protocol, which was developed by Bellcore and Cisco, the endpoints are low-intelligence devices, mostly executing control commands and providing result indications in response. The protocol represents a decomposition of other VoIP models, such as H.323, MGCP is a text-based protocol consisting of commands and responses. It uses the Session Description Protocol for specifying and negotiating the media streams to be transmitted in a call session, the media gateway control protocol architecture and its methodologies and programming interfaces are described in RFC2805. MGCP is a protocol that allows a call control device such as a Call Agent to take control of a specific port on a media gateway. In MGCP context media gateway controller is referred to as call agent and this has the advantage of centralized gateway administration and provides for largely scalable IP Telephony solutions. MGCP assumes a call control architecture where there is limited intelligence at the edge, the MGCP assumes that Call Agents, will synchronize with each other to send coherent commands and responses to the gateways under their control. The Call Agent uses MGCP to tell the media gateway which events should be reported to the Call Agent, how endpoints should be inter-connected, MGCP also allows the Call Agent to audit the current state of endpoints on a media gateway. The media gateway uses MGCP to report events, such as off-hook or dialed digits, while any signaling gateway is usually on the same physical switch as a media gateway, there is no such need. The Call Agent does not use MGCP to control the Signaling Gateway, rather, typically, a media gateway is configured with a list of Call Agents from which it may accept programming. In principle, event notifications may be sent to different Call Agents for each endpoint on the gateway, in the event of such a failure it is the backup Call Agents responsibility to reprogram the MG so that the gateway comes under the control of the backup Call Agent. Care is needed in cases, two Call Agents may know that they have lost contact with one another, but this does not guarantee that they are not both attempting to control the same gateway. The ability to audit the gateway to determine which Call Agent is currently controlling can be used to such conflicts. MGCP assumes that the multiple Call Agents will maintain knowledge of state among themselves or rebuild it if necessary. Its failover features take into account both planned and unplanned outages, MGCP recognizes three essential elements of communication, the media gateway controller, the media gateway endpoint, and connections between these entities. A media gateway may host multiple endpoints and each endpoint should be able to engage in multiple connections, multiple connections on the endpoints support calling features such as call waiting and three-way calling. MGCP is a text-based protocol using a command and response model, commands and responses are encoded in messages that are structured and formatted with the whitespace characters space, horizontal tab, carriage return, and linefeed, and the colon and the full stop
8.
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
9.
Post Office Protocol
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In computing, the Post Office Protocol is an application-layer Internet standard protocol used by local e-mail clients to retrieve e-mail from a remote server over a TCP/IP connection. POP has been developed several versions, with version 3 being the last standard in common use before largely being made obsolete by the more advanced IMAP as well as webmail. POP supports download-and-delete requirements for access to remote mailboxes, other protocols, notably IMAP, provide more complete and complex remote access to typical mailbox operations. In the late 1990s and early 2000s, fewer Internet Service Providers supported IMAP due to the space that was required on the ISPs hardware. Contemporary e-mail clients supported POP, then over time popular mail client software added IMAP support, a POP3 server listens on well-known port 110. This unique identifier is permanent and unique to the maildrop and allows a client to access the message in different POP sessions. Mail is retrieved and marked for deletion by message-number, when the client exits the session, the mail marked for deletion is removed from the maildrop. POP1 was specified in RFC918, POP2 by RFC937 and its current specification is RFC1939, updated with an extension mechanism, RFC2449 and an authentication mechanism in RFC1734. The original POP3 specification supported only an unencrypted USER/PASS login mechanism or Berkeley. rhosts access control, POP3 currently supports several authentication methods to provide varying levels of protection against illegitimate access to a users e-mail. Most are provided by the POP3 extension mechanisms, POP3 clients support SASL authentication methods via the AUTH extension. MIT Project Athena also produced a Kerberized version, RFC1460 introduced APOP into the core protocol. APOP is a protocol which uses the MD5 hash function in an attempt to avoid replay attacks. Clients implementing APOP include Mozilla Thunderbird, Opera Mail, Eudora, KMail, Novell Evolution, windows Live Mail, PowerMail, Apple Mail, and Mutt. RFC1460 was obsoleted by RFC1725, which was in turn obsoleted by RFC1939, pOP4 exists only as an informal proposal adding basic folder management, multipart message support, as well as message flag management to compete with IMAP, but has not progressed since 2003. An extension mechanism was proposed in RFC2449 to accommodate general extensions as well as announce in an organized manner support for commands, such as TOP. The RFC did not intend to encourage extensions, and reaffirmed that the role of POP3 is to provide support for mainly download-and-delete requirements of mailbox handling. The extensions are termed capabilities and are listed by the CAPA command, except for APOP, the optional commands were included in the initial set of capabilities. Following the lead of ESMTP, capabilities beginning with an X signify local capabilities, the STARTTLS extension allows the use of Transport Layer Security or Secure Sockets Layer to be negotiated using the STLS command, on the standard POP3 port, rather than an alternate
10.
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
11.
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
12.
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
13.
Secure Shell
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Secure Shell is a cryptographic network protocol for operating network services securely over an unsecured network. The best known application is for remote login to computer systems by users. SSH provides a channel over an unsecured network in a client-server architecture. Common applications include remote command-line login and remote command execution, the protocol specification distinguishes between two major versions, referred to as SSH-1 and SSH-2. The most visible application of the protocol is for access to accounts on Unix-like operating systems. In 2015, Microsoft announced that they would include support for SSH in a future release. SSH was designed as a replacement for Telnet and for unsecured remote shell protocols such as the Berkeley rlogin, rsh and those protocols send information, notably passwords, in plaintext, rendering them susceptible to interception and disclosure using packet analysis. SSH uses public-key cryptography to authenticate the remote computer and allow it to authenticate the user, there are several ways to use SSH, one is to use automatically generated public-private key pairs to simply encrypt a network connection, and then use password authentication to log on. Another is to use a manually generated public-private key pair to perform the authentication, in this scenario, anyone can produce a matching pair of different keys. The public key is placed on all computers that must allow access to the owner of the private key. While authentication is based on the key, the key itself is never transferred through the network during authentication. SSH only verifies whether the person offering the public key also owns the matching private key. In all versions of SSH it is important to verify unknown public keys, i. e. associate the public keys with identities, accepting an attackers public key without validation will authorize an unauthorized attacker as a valid user. On Unix-like systems, the list of authorized public keys is typically stored in the directory of the user that is allowed to log in remotely. This file is respected by SSH only if it is not writable by anything apart from the owner, when the public key is present on the remote end and the matching private key is present on the local end, typing in the password is no longer required. However, for security the private key itself can be locked with a passphrase. The private key can also be looked for in standard places, the ssh-keygen utility produces the public and private keys, always in pairs. SSH also supports password-based authentication that is encrypted by automatically generated keys, in this case the attacker could imitate the legitimate server side, ask for the password, and obtain it
14.
XMPP
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Extensible Messaging and Presence Protocol is a communications protocol for message-oriented middleware based on XML. It enables the exchange of structured yet extensible data between any two or more network entities. Originally named Jabber, the protocol was developed by the Jabber open-source community in 1999 for near real-time instant messaging, presence information, the Internet Engineering Task Force formed an XMPP working group in 2002 to formalize the core protocols as an IETF instant messaging and presence technology. The XMPP Working group produced four specifications, which were approved as Proposed Standards in 2004, in 2011, RFC3920 and RFC3921 were superseded by RFC6120 and RFC6121 respectively, with RFC6122 specifying the XMPP address format. In 2015, RFC6122 was superseded by RFC7622, in addition to these core protocols standardized at the IETF, the XMPP Standards Foundation is active in developing open XMPP extensions. XMPP-based software is deployed widely across the Internet, and by 2003, was used by ten million people worldwide. Jeremie Miller began working on the Jabber technology in 1998 and released the first version of the server on January 4,1999. The early Jabber community focused on software, mainly the jabberd server. The early Jabber protocol, as developed in 1999 and 2000, formed the basis for XMPP as published in RFC3920, note that RFC3920 and RFC3921 have been superseded by RFC6120 and RFC6121, published in 2011. The first IM service based on XMPP was Jabber. org, from 1999 until February 2006, the service used jabberd as its server software, at which time it migrated to ejabberd. In January 2010, the migrated to the proprietary M-Link server software produced by Isode Ltd. In August 2005, Google introduced Google Talk, a combination VoIP and IM system that uses XMPP for instant messaging and as a base for a voice, the initial launch did not include server-to-server communications, Google enabled that feature on January 17,2006. Google has since added video functionality to Google Talk, also using the Jingle protocol for signaling, in May 2013, Google announced Jabber compatibility would be dropped from Google Talk for server-to-server federation, although it would retain client-to-server support. In January 2008, AOL introduced experimental XMPP support for its AOL Instant Messenger service, however, in March 2008, this service was discontinued. As of May 2011, AOL offers limited XMPP support, in September 2008, Cisco Systems acquired Jabber, Inc. the creators of the commercial product Jabber XCP. In February 2010, the social-networking site Facebook opened up its chat feature to third-party applications via XMPP, some functionality was unavailable through XMPP, and support was dropped in April 2014. Similarly, in December 2011, Microsoft released an XMPP interface to its Microsoft Messenger service, skype, its de facto successor, also provides limited XMPP support. However, these are not native XMPP implementations, decentralization The architecture of the XMPP network is similar to email, anyone can run their own XMPP server and there is no central master server
15.
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
16.
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
17.
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
18.
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
19.
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
20.
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
21.
Address Resolution Protocol
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The Address Resolution Protocol is a telecommunication protocol used for resolution of Internet layer addresses into link layer addresses, a critical function in computer networks. ARP was defined by RFC826 in 1982, is Internet Standard STD37, ARP is used for mapping a network address to a physical address like an Ethernet address. IPv4 over IEEE802.3 and IEEE802.11 is the most common usage, in Internet Protocol Version 6 networks, the functionality of ARP is provided by the Neighbor Discovery Protocol. The Address Resolution Protocol is a request and response protocol whose messages are encapsulated by a link layer protocol and it is communicated within the boundaries of a single network, never routed across internetwork nodes. However, ARP was not developed in the OSI networking framework, the Address Resolution Protocol uses a simple message format containing one address resolution request or response. The message header specifies these types, as well as the size of addresses of each, the message header is completed with the operation code for request and reply. The payload of the consists of four addresses, the hardware. The principal packet structure of ARP packets is shown in the table which illustrates the case of IPv4 networks running on Ethernet. In this scenario, the packet has 48-bit fields for the hardware address and target hardware address. Thus, the ARP packet size in this case is 28 bytes, the EtherType for ARP is 0x0806. Hardware type This field specifies the network protocol type, Protocol type This field specifies the internetwork protocol for which the ARP request is intended. For IPv4, this has the value 0x0800, the permitted PTYPE values share a numbering space with those for EtherType. Hardware length Length of a hardware address, Protocol length Length of addresses used in the upper layer protocol. Operation Specifies the operation that the sender is performing,1 for request,2 for reply, sender hardware address Media address of the sender. In an ARP request this field is used to indicate the address of the host sending the request, in an ARP reply this field is used to indicate the address of the host that the request was looking for. Note that switches do not pay attention to field, particularly in learning MAC addresses. The ARP PDU is encapsulated in Ethernet frame, and that is what Layer 2 devices examine, sender protocol address Internetwork address of the sender. Target hardware address Media address of the intended receiver, in an ARP request this field is ignored
22.
Tunneling protocol
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In computer networks, a tunneling protocol allows a network user to access or provide a network service that the underlying network does not support or provide directly. One important use of a protocol is to allow a foreign protocol to run over a network that does not support that particular protocol, for example. Because tunneling involves repackaging the traffic data into a different form, perhaps with encryption as standard, the tunneling protocol works by using the data portion of a packet to carry the packets that actually provide the service. Typically, the delivery protocol operates at an equal or higher level in the model than the payload protocol. To understand a particular protocol stack imposed by tunneling, network engineers must understand both the payload and delivery protocol sets, in this case, the delivery and payload protocols are the same, but the payload addresses are incompatible with those of the delivery network. It is also possible to establish a connection using the link layer. The Layer 2 Tunneling Protocol allows the transmission of frames between two nodes, a tunnel is not encrypted by default, it relies on the TCP/IP protocol chosen to determine the level of security. SSH uses port 22 to enable data encryption of payloads being transmitted over a network connection. IPsec has an end-to-end Transport Mode, but can operate in a tunneling mode through a trusted security gateway. A Secure Shell tunnel consists of a tunnel created through an SSH protocol connection. Users may set up SSH tunnels to transfer unencrypted traffic over a network through an encrypted channel, for example, Microsoft Windows machines can share files using the Server Message Block protocol, a non-encrypted protocol. If one were to mount a Microsoft Windows file-system remotely through the Internet, to mount the Windows file-system securely, one can establish a SSH tunnel that routes all SMB traffic to the remote fileserver through an encrypted channel. Even though the SMB protocol itself contains no encryption, the encrypted SSH channel through which it travels offers security, to set up a local SSH tunnel, one configures an SSH client to forward a specified local port to a port on the remote machine. Once the SSH tunnel has been established, the user can connect to the local port to access the network service. The local port does not have to be the same as the remote port, SSH tunnels provide a means to bypass firewalls that prohibit certain Internet services – so long as a site allows outgoing connections. For example, an organization may prohibit a user from accessing Internet web pages directly without passing through the proxy filter. But users may not wish to have their web traffic monitored or blocked by the proxy filter. If users can connect to an external SSH server, they can create an SSH tunnel to forward a given port on their machine to port 80 on a remote web server
23.
Layer 2 Tunneling Protocol
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In computer networking, Layer 2 Tunneling Protocol is a tunneling protocol used to support virtual private networks or as part of the delivery of services by ISPs. It does not provide any encryption or confidentiality by itself, rather, it relies on an encryption protocol that it passes within the tunnel to provide privacy. A new version of this protocol, L2TPv3, appeared as proposed standard RFC3931 in 2005, L2TPv3 provides additional security features, improved encapsulation, and the ability to carry data links other than simply Point-to-Point Protocol over an IP network. The entire L2TP packet, including payload and L2TP header, is sent within a User Datagram Protocol datagram and it is common to carry PPP sessions within an L2TP tunnel. L2TP does not provide confidentiality or strong authentication by itself, IPsec is often used to secure L2TP packets by providing confidentiality, authentication and integrity. The combination of two protocols is generally known as L2TP/IPsec. The two endpoints of an L2TP tunnel are called the LAC and the LNS, the L2TP LNS waits for new tunnels. Once a tunnel is established, the traffic between the peers is bidirectional. To be useful for networking, higher-level protocols are then run through the L2TP tunnel, to facilitate this, an L2TP session is established within the tunnel for each higher-level protocol such as PPP. Either the LAC or LNS may initiate sessions, the traffic for each session is isolated by L2TP, so it is possible to set up multiple virtual networks across a single tunnel. MTU should be considered when implementing L2TP, the packets exchanged within an L2TP tunnel are categorized as either control packets or data packets. L2TP provides reliability features for the packets, but no reliability for data packets. Reliability, if desired, must be provided by the nested protocols running within each session of the L2TP tunnel, a L2TP tunnel can extend across an entire PPP session or only across one segment of a two-segment session. Length Total length of the message in bytes, present only when length flag is set, tunnel ID Indicates the identifier for the control connection. Session ID Indicates the identifier for a session within a tunnel, Ns sequence number for this data or control message, beginning at zero and incrementing by one for each message sent. Present only when sequence flag set, Nr sequence number for expected message to be received. Nr is set to the Ns of the last in-order message received plus one, in data messages, Nr is reserved and, if present, MUST be ignored upon receipt. Offset Size Specifies where payload data is located past the L2TP header, if the offset field is present, the L2TP header ends after the last byte of the offset padding
24.
Point-to-Point Protocol
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In computer networking, Point-to-Point Protocol is a data link protocol used to establish a direct connection between two nodes. It connects two routers directly without any host or any other networking device in between and it can provide connection authentication, transmission encryption, and compression. PPP is used many types of physical networks including serial cable, phone line, trunk line, cellular telephone, specialized radio links. PPP is also used over Internet access connections, Internet service providers have used PPP for customer dial-up access to the Internet, since IP packets cannot be transmitted over a modem line on their own, without some data link protocol. The only requirement for PPP is that the circuit provided be duplex, PPP was designed to work with numerous network layer protocols, including Internet Protocol, TRILL, Novells Internetwork Packet Exchange, NBF, DECnet and AppleTalk. Like SLIP, this is a full Internet connection over telephone lines via modem and it is more reliable than SLIP because it double checks to make sure that Internet packets arrive intact. PPP was designed somewhat after the original HDLC specifications, the designers of PPP included many additional features that had been seen only in proprietary data-link protocols up to that time. RFC2516 describes Point-to-Point Protocol over Ethernet as a method for transmitting PPP over Ethernet that is used with DSL. RFC2364 describes Point-to-Point Protocol over ATM as a method for transmitting PPP over ATM Adaptation Layer 5, PPP is a layered protocol that has three components, An encapsulation component that is used to transmit datagrams over the specified physical layer. A Link Control Protocol to establish, configure, and test the link as well as negotiate capabilities, one or more Network Control Protocols used to negotiate optional configuration parameters and facilities for the network layer. There is one NCP for each higher-layer protocol supported by PPP, PPP is specified in RFC1661. Link Control Protocol initiates and terminates connections gracefully, allowing hosts to negotiate connection options and it is an integral part of PPP, and is defined in the same standard specification. LCP provides automatic configuration of the interfaces at each end and for selecting optional authentication, the LCP protocol runs on top of PPP and therefore a basic PPP connection has to be established before LCP is able to configure it. RFC1994 describes Challenge-handshake authentication protocol, which is preferred for establishing connections with ISPs. Although deprecated, Password authentication protocol is still sometimes used, another option for authentication over PPP is Extensible Authentication Protocol described in RFC2284. After the link has been established, additional network configuration may take place, most commonly, the Internet Protocol Control Protocol is used, although Internetwork Packet Exchange Control Protocol and AppleTalk Control Protocol were once very popular. Internet Protocol Version 6 Control Protocol will see extended use in the future, PPP permits multiple network layer protocols to operate on the same communication link. For every network layer protocol used, a separate Network Control Protocol is provided in order to encapsulate and negotiate options for the network layer protocols
25.
Ethernet
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Ethernet /ˈiːθərnɛt/ is a family of computer networking technologies commonly used in local area networks, metropolitan area networks and wide area networks. It was commercially introduced in 1980 and first standardized in 1983 as IEEE802.3, over time, Ethernet has largely replaced competing wired LAN technologies such as token ring, FDDI and ARCNET. The original 10BASE5 Ethernet uses coaxial cable as a medium, while the newer Ethernet variants use twisted pair. Over the course of its history, Ethernet data transfer rates have increased from the original 2.94 megabits per second to the latest 100 gigabits per second. The Ethernet standards comprise several wiring and signaling variants of the OSI physical layer in use with Ethernet, systems communicating over Ethernet divide a stream of data into shorter pieces called frames. As per the OSI model, Ethernet provides services up to, since its commercial release, Ethernet has retained a good degree of backward compatibility. Features such as the 48-bit MAC address and Ethernet frame format have influenced other networking protocols, the primary alternative for some uses of contemporary LANs is Wi-Fi, a wireless protocol standardized as IEEE802.11. Ethernet was developed at Xerox PARC between 1973 and 1974 and it was inspired by ALOHAnet, which Robert Metcalfe had studied as part of his PhD dissertation. In 1975, Xerox filed a patent application listing Metcalfe, David Boggs, Chuck Thacker, in 1976, after the system was deployed at PARC, Metcalfe and Boggs published a seminal paper. Metcalfe left Xerox in June 1979 to form 3Com and he convinced Digital Equipment Corporation, Intel, and Xerox to work together to promote Ethernet as a standard. The so-called DIX standard, for Digital/Intel/Xerox, specified 10 Mbit/s Ethernet, with 48-bit destination and source addresses and it was published on September 30,1980 as The Ethernet, A Local Area Network. Data Link Layer and Physical Layer Specifications, version 2 was published in November,1982 and defines what has become known as Ethernet II. Formal standardization efforts proceeded at the time and resulted in the publication of IEEE802.3 on June 23,1983. Ethernet initially competed with two largely proprietary systems, Token Ring and Token Bus, in the process, 3Com became a major company. 3Com shipped its first 10 Mbit/s Ethernet 3C100 NIC in March 1981, an Ethernet adapter card for the IBM PC was released in 1982, and, by 1985, 3Com had sold 100,000. Parallel port based Ethernet adapters were produced for a time, with drivers for DOS, by the early 1990s, Ethernet became so prevalent that it was a must-have feature for modern computers, and Ethernet ports began to appear on some PCs and most workstations. This process was sped up with the introduction of 10BASE-T and its relatively small modular connector. Since then, Ethernet technology has evolved to meet new bandwidth, in addition to computers, Ethernet is now used to interconnect appliances and other personal devices
26.
Digital subscriber line
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Digital subscriber line is a family of technologies that are used to transmit digital data over telephone lines. In telecommunications marketing, the term DSL is widely understood to mean asymmetric digital subscriber line, DSL service can be delivered simultaneously with wired telephone service on the same telephone line. This is possible because DSL uses higher frequency bands for data, on the customer premises, a DSL filter on each non-DSL outlet blocks any high-frequency interference to enable simultaneous use of the voice and DSL services. Bit rates of 1 Gbit/s have been reached in trials, in ADSL, the data throughput in the upstream direction is lower, hence the designation of asymmetric service. In symmetric digital subscriber line services, the downstream and upstream data rates are equal, researchers at Bell Labs have reached speeds of 10 Gbit/s, while delivering 1 Gbit/s symmetrical broadband access services using traditional copper telephone lines. These higher speeds are lab results, however, a 2012 survey found that DSL continues to be the dominant technology for broadband access with 364.1 million subscribers worldwide. For a long time it was thought that it was not possible to operate a conventional phone-line beyond low-speed limits, in the 1950s, ordinary twisted-pair telephone-cable often carried four megahertz television signals between studios, suggesting that such lines would allow transmitting many megabits per second. One such circuit in the UK ran some ten miles between the BBC studios in Newcastle-upon-Tyne and the Pontop Pike transmitting station and it was able to give the studios a low quality cue feed but not one suitable for transmission. However, these cables had other impairments besides Gaussian noise, preventing such rates from becoming practical in the field, the 1980s saw the development of techniques for broadband communications that allowed the limit to be greatly extended. A patent was filed in 1979 for the use of existing telephone wires for both telephones and data terminals that were connected to a computer via a digital data carrier system. Joseph W. Lechleiders contribution to DSL was his insight that an asymmetric arrangement offered more than double the capacity of symmetric DSL. ADSL supports two modes of transport—fast channel and interleaved channel, fast channel is preferred for streaming multimedia, where an occasional dropped bit is acceptable, but lags are less so. Interleaved channel works better for file transfers, where the data must be error-free. Older ADSL standards delivered 8 Mbit/s to the customer over about 2 km of unshielded twisted-pair copper wire, distances greater than 2 km significantly reduce the bandwidth usable on the wires, thus reducing the data rate. But ADSL loop extenders increase these distances by repeating the signal, until the late 1990s, the cost of digital signal processors for DSL was prohibitive. All types of DSL employ highly complex digital signal processing algorithms to overcome the inherent limitations of the twisted pair wires. Due to the advancements of very-large-scale integration technology, the cost of the equipment associated with a DSL deployment lowered significantly, the two main pieces of equipment are a digital subscriber line access multiplexer at one end and a DSL modem at the other end. A DSL connection can be deployed over existing cable, such deployment, even including equipment, is much cheaper than installing a new, high-bandwidth fiber-optic cable over the same route and distance
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