Glossary
Computer DCCP DHCP DNS ECN Fax FTP Gopher GTP HTTP IMAP Internet IP IP Address IPv4 IPv6 IRC Network NNTP NTP POP3 Printer Router RPC RSVP RTCP RTP RTSP SDP SIGTRAN SIP SMTP SNMP SOAP SSH STUN Switch TCP Telephone TELNET TLS UDP XMPP
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A computer is a machine that manipulates data according to a list of instructions.
The first devices that resemble modern computers date to the mid-20th century (around 1940 - 1945), although the computer concept and various machines similar to computers existed earlier. Early electronic computers were the size of a large room, consuming as much power as several hundred modern personal computers.[1] Modern computers are based on tiny integrated circuits and are millions to billions of times more capable while occupying a fraction of the space.[2] Today, simple computers may be made small enough to fit into a wristwatch and be powered from a watch battery. Personal computers in various forms are icons of the Information Age and are what most people think of as "a computer"; however, the most common form of computer in use today is the embedded computer. Embedded computers are small, simple devices that are used to control other devices — for example, they may be found in machines ranging from fighter aircraft to industrial robots, digital cameras, and children's toys.
The ability to store and execute lists of instructions called programs makes computers extremely versatile and distinguishes them from calculators. The Church–Turing thesis is a mathematical statement of this versatility: any computer with a certain minimum capability is, in principle, capable of performing the same tasks that any other computer can perform. Therefore, computers with capability and complexity ranging from that of a personal digital assistant to a supercomputer are all able to perform the same computational tasks given enough time and storage capacity.
DCCP - Datagram Congestion Control Protocol
The Datagram Congestion Control Protocol (DCCP) is a message-oriented transport layer protocol. DCCP implements reliable connection setup, teardown, ECN, congestion control, and feature negotiation. DCCP was published as RFC 4340, a proposed standard, by the IETF in March, 2006. Linux had an implementation of DCCP first released in Linux kernel version 2.6.14 (released 28-Oct-2005) and this continues to improve with each release.
DCCP provides a way to gain access to congestion control mechanisms without having to implement them at the application layer. It allows for flow-based semantics like in TCP, but does not require reliable in-order delivery. DCCP also does not require features of SCTP such as sequenced delivery within multiple streams.
DCCP is useful for applications with timing constraints on the delivery of data that may become useless to the receiver if reliable in-order delivery combined with congestion avoidance is used. Such applications might include streaming media and Internet telephony. Such applications have either settled for TCP or used UDP and implemented their own congestion control mechanisms (or no congestion control at all).
A DCCP connection contains acknowledgment traffic as well as data traffic. Acknowledgments inform a sender whether its packets arrived, and whether they were ECN marked. Acks are transmitted as reliably as the congestion control mechanism in use requires, possibly completely reliably.
DCCP has the option to have very long (48-bit) sequence numbers corresponding to a packet ID (rather than a byte ID as in TCP). The long length of the sequence numbers is intended to guard against "some blind attacks, such as the injection of DCCP-Resets into the connection."
DHCP - Dynamic Host Configuration Protocol
Dynamic Host Configuration Protocol (DHCP) is a protocol used by networked devices (clients) to obtain various parameters necessary for the clients to operate in an Internet Protocol (IP) network. By using this protocol, system administration workload greatly decreases, and devices can be added to the network with minimal or no manual configurations.
The Domain Name System (DNS) associates various sorts of information with domain names; most importantly, it serves as the "phone book" for the Internet by translating human-readable computer hostnames, e.g. www.example.com, into the IP addresses, e.g. 208.77.188.166, that networking equipment needs to deliver information. It also stores other information such as the list of mail exchange servers that accept email for a given domain. In providing a worldwide keyword-based redirection service, the Domain Name System is an essential component of contemporary Internet use.
ECN - Explicit Congestion Notification
ECN is an extension to the Internet Protocol which allows end-to-end notification of network congestion without dropping packets. ECN is an optional feature, and is only used when both endpoints signal that they want to use it.
Traditionally, TCP/IP networks signal congestion by dropping packets. When ECN is successfully negotiated, an ECN-aware router may set a bit in the IP header instead of dropping a packet in order to signal the beginning of congestion. The receiver of the packet echoes the congestion indication to the sender, which must react as though a packet drop were detected.
ECN uses two bits in the Differentiated Services field in the IP header, in the IPv4 TOS Byte or the IPv6 Traffic Class Octet. These two bits can be used to encode one of the values ECN-unaware transport, ECN-aware transport or congestion experienced.
Some outdated or buggy network equipment drops packets with the ECN bit set, rather than ignoring the bit
Fax (short for facsimile, from Latin fac simile, "make similar", i.e. "make a copy") is a telecommunications technology used to transfer copies (facsimiles) of documents, especially using affordable devices operating over the telephone network. The word telefax, short for telefacsimile, for "make a copy at a distance", is also used as a synonym. The device is also known as a telecopier in certain industries. When sending documents to people at large distances, faxes have a distinct advantage over postal mail in that the delivery is nearly instantenous, yet its disadvantages in quality and its proprietary format have relegated it to a position beneath email as the prevailing form of electronic document transferral.
In computing, the File Transfer Protocol (FTP) (Port 21) is a network protocol used to transfer data from one computer to another through a network, such as over the Internet.
FTP is a protocol for exchanging files over any TCP/IP based network to manipulate files on another computer on that network regardless of which operating systems are involved (if the computers permit FTP access). There are many existing FTP client and server programs. FTP servers can be set up anywhere between game servers, voice servers, internet hosts, and other physical servers.
Gopher is a distributed document search and retrieval network protocol designed for the Internet. Its goal is to function as an improved form of Anonymous FTP, enhanced with hyperlinking features similar to that of the World Wide Web.
The Gopher protocol offers some features not natively supported by the Web and imposes a much stronger hierarchy on information stored on it. Its text menu interface is well-suited to computing environments that rely heavily on remote computer terminals, common in universities at the time of its creation.
GTP - GPRS Tunnelling Protocol
GPRS Tunneling Protocol (or GTP) is an IP based protocol used within GSM and UMTS networks. It can be used with UDP or TCP. GTP version one is used only on UDP. There are in fact three separate protocols, GTP-C, GTP-U and GTP'.
GTP-C is used within the GPRS core network for signalling between GPRS Support Nodes (GGSNs and SGSNs). This allows the SGSN to activate a session on the users behalf (PDP context activation), to deactivate the same session, to adjust quality of service parameters or to update a session for a subscriber who has just arrived from another SGSN.
GTP-U is used for carrying user data within the GPRS core network and between the Radio Access Network and the core network. The user data transported can be packets in any of IPv4, IPv6 or PPP formats.
GTP' (GTP prime) uses the same message structure as GTP-C and GTP-U, but it is largely a completely separate protocol. It can be used for carrying charging data from the "Charging Data Function" of the GSM or UMTS network to the "Charging Gateway Function". In most cases, this should mean from many individual network elements such as the GGSNs to a centralised computer which then delivers the charging data more conveniently to the network operator's billing center.
HTTP - Hypertext Transfer Protocol
Hypertext Transfer Protocol (HTTP) is a communications protocol for the transfer of information on intranets and the World Wide Web. Its original purpose was to provide a way to publish and retrieve hypertext pages over the Internet.
IMAP - Internet Message Access Protocol
The Internet Message Access Protocol (commonly known as IMAP or IMAP4, and previously called Internet Mail Access Protocol, Interactive Mail Access Protocol (RFC 1064), and Interim Mail Access Protocol[1]) is an application layer Internet protocol operating on port 143 that allows a local client to access e-mail on a remote server. The current version, IMAP version 4 revision 1 (IMAP4rev1), is defined by RFC 3501. IMAP4 and POP3 (Post Office Protocol version 3) are the two most prevalent Internet standard protocols for e-mail retrieval. Virtually all modern e-mail clients and servers support both.
The Internet is a worldwide, publicly accessible series of interconnected computer networks that transmit data by packet switching using the standard Internet Protocol (IP). It is a "network of networks" that consists of millions of smaller domestic, academic, business, and government networks, which together carry various information and services, such as electronic mail, online chat, file transfer, and the interlinked web pages and other resources of the World Wide Web (WWW).
The Internet Protocol (IP) is a data-oriented protocol used for communicating data across a packet-switched internetwork.
IP is a network layer protocol in the Internet protocol suite and is encapsulated in a data link layer protocol (e.g., Ethernet). As a lower layer protocol, IP provides the service of communicable unique global addressing amongst computers.
An IP address (or Internet Protocol address) is a unique address that certain electronic devices use in order to identify and communicate with each other on a computer network utilizing the Internet Protocol standard (IP)—in simpler terms, a computer address. Any participating network device—including routers, switches, computers, infrastructure servers (e.g., NTP, DNS, DHCP, SNMP, etc.), printers, Internet fax machines, and some telephones—can have its own address that is unique within the scope of the specific network. Some IP addresses are intended to be unique within the scope of the global Internet, while others need to be unique only within the scope of an enterprise.
The IP address acts as a locator for one IP device to find another and interact with it. It is not intended, however, to act as an identifier that always uniquely identifies a particular device. In current practice, an IP address is not always a unique identifier, due to technologies such as dynamic assignment and network address translation.
IP addresses are managed and created by the Internet Assigned Numbers Authority (IANA). The IANA generally allocates super-blocks to Regional Internet Registries, who in turn allocate smaller blocks to Internet service providers and enterprises.
IPv4 only uses 32-bit (4-byte) addresses, which limits the address space to 4,294,967,296 (232) possible unique addresses. However, many are reserved for special purposes, such as private networks (~18 million addresses) or multicast addresses (~270 million addresses). This reduces the number of addresses that can be allocated as public Internet addresses, and as the number of addresses available is consumed, an IPv4 address shortage appears to be inevitable in the long run. This limitation has helped stimulate the push towards IPv6, which is currently in the early stages of deployment and is currently the only contender to replace IPv4.
IPv4 addresses are usually represented in dotted-decimal notation (four numbers, each ranging from 0 to 255, separated by dots, e.g. 147.132.42.18). Each range from 0 to 255 can be represented by 8 bits, and is therefore called an octet. It is possible, although less common, to write IPv4 addresses in binary or hexadecimal. When converting, each octet is treated as a separate number. (So 255.255.0.0 in dot-decimal would be FF.FF.00.00 in hexadecimal.)
IPv6 is a new standard protocol intended to replace IPv4 for the Internet. Addresses are 128 bits (16 bytes) wide, which, even with a generous assignment of netblocks, will more than suffice for the foreseeable future. In theory, there would be exactly 2128, or about 3.403 × 1038 unique host interface addresses. Further, this large address space will be sparsely populated, which makes it possible to again encode more routing information into the addresses themselves.
Example: 2001:0db8:85a3:08d3:1319:8a2e:0370:7334
Writing for Technology Review in 2004, Simson Garfinkel wrote notes that there will exist "roughly 5,000 addresses for every square micrometer of the Earth's surface".[1] This enormous magnitude of available IP addresses will be sufficiently large for the indefinite future, even though mobile phones, cars and all types of personal devices are coming to rely on the Internet for everyday purposes.
The above source, however, involves a common misperception about the IPv6 architecture. Its large address space is not intended to provide unique addresses for every possible point. Rather, the addressing architecture is such that it allows large blocks to be assigned for specific purposes and, where appropriate, aggregated for providing routing. With a large address space, there is not the need to have complex address conservation methods as used in classless inter-domain routing (CIDR).
Windows Vista, Apple Computer's Mac OS X, and an increasing range of Linux distributions include native support for the protocol, but it is not yet widely deployed elsewhere.
Internet Relay Chat (IRC) is a form of real-time Internet chat or synchronous conferencing. It is mainly designed for group (many-to-many) communication in discussion forums called channels, but also allows one-to-one communication via private message and data transfers via Direct Client-to-Client.
IRC was created by Jarkko Oikarinen in late August 1988 to replace a program called MUT (MultiUser talk) on a BBS called OuluBox in Finland. Oikarinen found inspiration in a chat system known as Bitnet Relay, which operated on the BITNET.
IRC gained prominence when it was used to report on the Soviet coup attempt of 1991 throughout a media blackout. It was previously used in a similar fashion during the Iraqi invasion. Relevant logs are available from ibiblio archive[1].
IRC client software is available for virtually every computer operating system.
A computer network is an interconnected group of computers. Networks may be classified by the network layer at which they operate according to basic reference models considered as standards in the industry, such as the four-layer Internet Protocol Suite model. While the seven-layer Open Systems Interconnection (OSI) reference model is better known in academia, the majority of networks use the Internet Protocol Suite (IP).
NNTP - Network News Transfer Protocol
The Network News Transfer Protocol or NNTP is an Internet application protocol used primarily for reading and posting Usenet articles (aka netnews), as well as transferring news among news servers. Brian Kantor of the University of California, San Diego and Phil Lapsley of the University of California, Berkeley completed RFC 977, the specification for the Network News Transfer Protocol, in March 1986. Other contributors included Stan Barber from the Baylor College of Medicine and Erik Fair of Apple Computer.
Usenet was originally designed around the UUCP network, with most article transfers taking place over direct computer-to-computer telephone links. Readers and posters would log into the same computers that hosted the servers, reading the articles directly from the local disk.
As local area networks and the Internet became more commonly used, it became desirable to allow newsreaders to be run on personal computers, and a means of employing the Internet to handle article transfers was desired. A newsreader, also known as a news client, is an application software that reads articles on Usenet (generally known as newsgroup), either directly from the news server's disks or via the NNTP.
Because networked Internet-compatible filesystems were not yet widely available, it was decided to develop a new protocol that resembled SMTP, but was tailored for reading newsgroups.
The well-known TCP port 119 is reserved for NNTP. When clients connect to a news server with SSL, TCP port 563 is used. This is sometimes referred to as NNTPS.
In October 2006, the IETF released RFC 3977 which updates the NNTP protocol and codifies many of the additions made over the years since RFC 977. The IMAP protocol can also be used for reading newsgroups.
The Network Time Protocol (NTP) is a protocol for synchronizing the clocks of computer systems over packet-switched, variable-latency data networks. NTP uses UDP port 123 as its transport layer. It is designed particularly to resist the effects of variable latency (jitter buffer).
NTP is one of the oldest Internet protocols still in use (since before 1985). NTP was originally designed by Dave Mills of the University of Delaware, who still maintains it, along with a team of volunteers.
NTP is not related to the much simpler DAYTIME (RFC 867) and TIME (RFC 868) protocols.
In computing, local e-mail clients use the Post Office Protocol version 3 (POP3), an application-layer Internet standard protocol, to retrieve e-mail from a remote server over a TCP/IP connection. Many subscribers to individual Internet service provider e-mail accounts access their e-mail with client software that uses POP3.
A computer printer, or more commonly a printer, produces a hard copy (permanent human-readable text and/or graphics) of documents stored in electronic form, usually on physical print media such as paper or transparencies. Many printers are primarily used as local computer peripherals, and are attached by a printer cable to a computer which serves as a document source. Some printers, commonly known as network printers, have built-in network interfaces (typically wireless or Ethernet), and can serve as a hardcopy device for any user on the network. Individual printers are often designed to support both local and network connected users at the same time.
In addition, many modern printers can directly interface to electronic media such as memory sticks or memory cards, or to image capture devices such as digital cameras, scanners; some printers are combined with a scanners and/or fax machines in a single unit. Printers that include non-printing features are sometimes called Multifunction Printers (MFP) or Multi-Function Devices (MFD).
A printer which is combined with a scanner can function as a kind of photocopier if so designed. Most MFPs include printing, scanning, and copying among their features. Printers are designed for low-volume, short-turnaround print jobs; requiring virtually no setup time to achieve a hard copy of a given document. However, printers are generally slow devices (30 pages per minute is considered fast; and many consumer printers are far slower than that), and the cost-per-page is relatively high. In contrast, the printing press (which serves much the same function), is designed and optimized for high-volume print jobs such as newspaper print runs--printing presses are capable of hundreds of pages per minute or more, and have an incremental cost-per-page which is a fraction of that of printers.
The printing press remains the machine of choice for high-volume, professional publishing. However, as printers have improved in quality and performance, many jobs which used to be done by professional print shops are now done by users on local printers; see desktop publishing. The world's first computer printer was a 19th century mechanically driven apparatus invented by Charles Babbage for his Difference Engine.[1]
A router (pronounced /rou't?r/) is a computer whose software and hardware are usually tailored to the tasks of routing and forwarding, generally containing a specialized operating system (e.g. Cisco's IOS or Juniper Networks JUNOS and JUNOSe or Extreme Networks XOS), RAM, NVRAM, flash memory, and one or more processors. High-end routers contain many processors and specialized Application-specific integrated circuits (ASIC) and do a great deal of parallel processing. Chassis based systems like the Nortel MERS-8600 or ERS-8600 routing switch, (pictured right) have multiple ASICs on every module and allow for a wide variety of LAN, MAN, METRO, and WAN port technologies or other connections that are customizable. However, with the proper software (such as SmoothWall, XORP or Quagga), even commodity PCs can act as routers.
Routers connect two or more logical subnets, which do not necessarily map one-to-one to the physical interfaces of the router.[1] The term layer 3 switch often is used interchangeably with router, but switch is really a marketing term without a rigorous technical definition. In marketing usage, it is generally optimized for Ethernet LAN interfaces and may not have other physical interface types.
Routers operate in two different planes [2]:
Control Plane, in which the router learns the outgoing interface that is most appropriate for forwarding specific packets to specific destinations,
Forwarding Plane, which is responsible for the actual process of sending a packet received on a logical interface to an outbound logical interface.
Remote procedure call (RPC) is a technology that allows a computer program to cause a subroutine or procedure to execute in another address space (commonly on another computer on a shared network) without the programmer explicitly coding the details for this remote interaction. That is, the programmer would write essentially the same code whether the subroutine is local to the executing program, or remote. When the software in question is written using object-oriented principles, RPC may be referred to as remote invocation or remote method invocation.
Note that there are many different technologies commonly used to accomplish this which are often incompatible, such as ONC RPC and DCE/RPC.
RSVP - Resource reservation protocol
The Resource ReSerVation Protocol (RSVP), described in RFC 2205, is a Transport layer protocol designed to reserve resources across a network for an integrated services Internet. "RSVP does not transport application data but is rather an Internet control protocol, like ICMP, IGMP, or routing protocols" - RFC 2205. RSVP provides receiver-initiated setup of resource reservations for multicast or unicast data flows with scaling and robustness.
RSVP can be used by either hosts or routers to request or deliver specific levels of quality of service (QoS) for application data streams or flows. RSVP defines how applications place reservations and how they can relinquish the reserved resources once the need for them has ended. RSVP operation will generally result in resources being reserved in each node along a path.
RSVP is not itself a routing protocol and was designed to interoperate with current and future routing protocols.
RSVP by itself is rarely deployed in telecommunications networks today[citation needed] but the traffic engineering extension of RSVP, or RSVP-TE, is becoming more widely accepted nowadays in many QoS-oriented networks.
RTCP - Real time control protocol
Real-time Transport Control Protocol (RTCP) is a sister protocol of the Real-time Transport Protocol (RTP). It is defined in RFC 3550 (which obsoletes RFC 1889).
RTCP provides out-of-band control information for an RTP flow. It partners RTP in the delivery and packaging of multimedia data, but does not transport any data itself. It is used periodically to transmit control packets to participants in a streaming multimedia session. The primary function of RTCP is to provide feedback on the quality of service being provided by RTP.
RTCP gathers statistics on a media connection and information such as bytes sent, packets sent, lost packets, jitter, feedback and round trip delay. An application may use this information to increase the quality of service, perhaps by limiting flow or using a different codec.
There are several type of RTCP packets: Sender report packet, Receiver report packet, Source Description RTCP Packet, Goodbye RTCP Packet and Application Specific RTCP packets.
RTCP itself does not provide any flow encryption or authentication means. SRTCP protocol can be used for that purpose.
RTP - Real-time Transport Protocol
The Real-time Transport Protocol (or RTP) defines a standardized packet format for delivering audio and video over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and first published in 1996 as RFC 1889 which was made obsolete in 2003 by RFC 3550. Real time transport protocol can also be used in conjunction with RTSP protocol which enhances the field of multimedia applications.
RTP does not have a standard TCP or UDP port on which it communicates. The only standard that it obeys is that UDP communications are done via an even port and the next higher odd port is used for RTP Control Protocol (RTCP) communications. Although there are no standards assigned, RTP is generally configured to use ports 16384-32767. RTP can carry any data with real-time characteristics, such as interactive audio and video. Call setup and tear-down for VoIP applications is usually performed by either SIP or H.323 protocols. The fact that RTP uses a dynamic port range makes it difficult for it to traverse firewalls. In order to get around this problem, it is often necessary to set up a STUN server.
It was originally designed as a multicast protocol, but has since been applied in many unicast applications. It is frequently used in streaming media systems (in conjunction with RTSP) as well as videoconferencing and push to talk systems (in conjunction with H.323 or SIP), making it the technical foundation of the Voice over IP industry. It goes along with the RTCP and is built on top of the User Datagram Protocol (UDP). Applications using RTP are less sensitive to packet loss, but typically very sensitive to delays, so UDP is a better choice than TCP for such applications.
According to RFC 1889, the services provided by RTP include:
Payload-type identification - Indication of what kind of content is being carried
Sequence numbering - PDU sequence number
Time stamping - allow synchronization and jitter calculations
Delivery monitoring
The protocols themselves do not provide mechanisms to ensure timely delivery. They also do not give any Quality of Service (QoS) guarantees. These things have to be provided by some other mechanism.
Also, out of order delivery is still possible, and flow and congestion control are not supported directly. However, the protocols do deliver the necessary data to the application to make sure it can put the received packets in the correct order. Also, RTCP provides information about reception quality which the application can use to make local adjustments. For example if a congestion is forming, the application could decide to lower the data rate.
RTP was also published by the ITU-T as H.225.0, but later removed once the IETF had a stable standards-track RFC published. It exists as an Internet Standard (STD 64) defined in RFC 3550 (which obsoletes RFC 1889). RFC 3551 (STD 65) (which obsoletes RFC 1890) defines a specific profile for Audio and Video Conferences with Minimal Control. RFC 3711 defines the Secure Real-time Transport Protocol (SRTP) profile (actually an extension to RTP Profile for Audio and Video Conferences) which can be used (optionally) to provide confidentiality, message authentication, and replay protection for audio and video streams being delivered.
RTSP - Real Time Streaming Protocol
The Real Time Streaming Protocol (RTSP), developed by the IETF and created in 1998 as RFC 2326, is a protocol for use in streaming media systems which allows a client to remotely control a streaming media server, issuing VCR-like commands such as "play" and "pause", and allowing time-based access to files on a server.
The sending of streaming data itself is not part of the RTSP protocol. Most RTSP servers use the standards-based RTP as the transport protocol for the actual audio/video data, acting somewhat as a metadata channel. The RTSP server from RealNetworks also features Real's proprietary RDT as the transport protocol.
SDP - Session Description Protocol
Session Description Protocol (SDP) is a format for describing streaming media initialization parameters. It has been published by the IETF as RFC 4566.
SDP is intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation. SDP does not provide the content of the media form itself but simply provides a negotiation between two end points to allow them to agree on a media type and format. This allows SDP to support upcoming media types and formats, enabling systems based on this technology to be forward compatible.
SDP started off as a component of the Session Announcement Protocol (SAP), but found other uses in conjunction with RTP, RTSP, SIP and just as a standalone format for describing multicast sessions.
There are five terms related to SDP:
Conference: It is a set of two or more communicating users along with the software they are using.
Session : Session is the multimedia sender and receiver and the flowing stream of data.
Session Announcement: A session announcement is a mechanism by which a session description is conveyed to users in a proactive fashion, i.e., the session description was not explicitly requested by the user.
Session Advertisement : same as session announcement
Session Description : A well defined format for conveying sufficient information to discover and participate in a multimedia session.
Below is a list of the syntax of SDP, please note that this list may not be exhaustive and new attributes are added to the RFC regularly
SIGTRAN - Stream Control Transmission Protocol
In the field of computer networking, the IETF Signaling Transport (SIGTRAN) working group defined the Stream Control Transmission Protocol (SCTP) as a transport layer protocol in 2000. RFC 4960 defines the protocol, with RFC 3286 providing an introductory text.
As a transport protocol, SCTP operates analogously to TCP or UDP. Indeed it provides some similar services as TCP—ensuring reliable, in-sequence transport of messages with congestion control. (In the absence of native SCTP support, it may sometimes be desirable to tunnel SCTP over UDP.)
SIP - Session Initiation Protocol
The Session Initiation Protocol (SIP) is a signalling protocol, widely used for setting up and tearing down multimedia communication sessions such as voice and video calls over the Internet. Other feasible application examples include video conferencing, streaming multimedia distribution, instant messaging, presence information and online games. In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IMS architecture for IP based streaming multimedia services in cellular systems.
The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams. The modification can involve changing addresses or ports, inviting more participants, adding or deleting media streams, etc.
The SIP protocol is situated at the session layer in the OSI model, and at the application layer in the TCP/IP model. SIP is designed to be independent of the underlying transport layer; it can run on TCP, UDP, or SCTP. It was originally designed by Henning Schulzrinne (Columbia University) and Mark Handley (UCL) starting in 1996. The latest version of the specification is RFC 3261 [1] from the IETF SIP Working Group.[2]
SIP has the following characteristics:
Transport-independent, because SIP can be used with UDP, TCP, SCTP, etc.
Text-based, allowing for humans to read and analyze SIP messages.
SMTP - Simple Mail Transfer Protocol
Simple Mail Transfer Protocol (SMTP) is the de facto standard for e-mail transmissions across the Internet. Formally SMTP is defined in RFC 821 (STD 10) as amended by RFC 1123 (STD 3) chapter 5. The protocol used today is also known as ESMTP and defined in RFC 2821.
SNMP - Simple Network Management Protocol
The Simple Network Management Protocol (SNMP) forms part of the internet protocol suite as defined by the Internet Engineering Task Force (IETF). SNMP is used in network management systems to monitor network-attached devices for conditions that warrant administrative attention. It consists of a set of standards for network management, including an Application Layer protocol, a database schema, and a set of data objects.[1]
SNMP exposes management data in the form of variables on the managed systems, which describe the system configuration. These variables can then be queried (and sometimes set) by managing applications.
SOAP (see below for name and origins) is a protocol for exchanging XML-based messages over computer networks, normally using HTTP/HTTPS. SOAP forms the foundation layer of the web services protocol stack providing a basic messaging framework upon which abstract layers can be built.
There are several different types of messaging patterns in SOAP, but by far the most common is the Remote Procedure Call (RPC) pattern, in which one network node (the client) sends a request message to another node (the server) and the server immediately sends a response message to the client. SOAP is the successor of XML-RPC, though it borrows its transport and interaction neutrality and the envelope/header/body from elsewhere, probably from WDDX.[
Secure Shell or SSH is a network protocol that allows data to be exchanged over a secure channel between two computers. Encryption provides confidentiality and integrity of data. SSH uses public-key cryptography to authenticate the remote computer and allow the remote computer to authenticate the user, if necessary.
SSH is typically used to log into a remote machine and execute commands, but it also supports tunneling, forwarding arbitrary TCP ports and X11 connections; it can transfer files using the associated SFTP or SCP protocols.
An SSH server, by default, listens on the standard TCP port 22.
An SSH client program is typically used for establishing connections to an SSHD daemon accepting remote connections. Both are commonly present on most modern operating systems, including Mac OS X, Linux, FreeBSD, Solaris and OpenVMS. Proprietary, freeware and open source versions of various levels of complexity and completeness exist.
STUN - Simple traversal of UDP over NATs
Simple Traversal of UDP through NATs (STUN), is a network protocol allowing a client behind a NAT (Network Address Translator) to find out its public address, the type of NAT it is behind and the internet-side port associated by the NAT with a particular local port. This information is used to set up UDP (User Datagram Protocol) communication between two hosts that are both behind NAT routers. The protocol is defined in RFC 3489.
A network switch is a computer networking device that connects network segments.
Low-end network switches appear nearly identical to network hubs, but a switch contains more "intelligence" (and comes with a correspondingly slightly higher price tag) than a network hub. Network switches are capable of inspecting data packets as they are received, determining the source and destination device of that packet, and forwarding it appropriately. By delivering each message only to the connected device it was intended for, a network switch conserves network bandwidth and offers generally better performance than a hub.
In the past, it was faster to use Layer 2 techniques to switch, when only MAC addresses could be looked up in content addressable memory (CAM). With the advent of ternary CAM (TCAM), it was equally fast to look up an IP address or a MAC address. TCAM is expensive, but very appropriate for enterprise switches that use default routes plus a moderate number of other routes. For routers that need a full Internet routing table, TCAM may not be cost-effective.
The first Ethernet switch was introduced by Kalpana in 1989.
TCP - Transmission Control Protocol
The Transmission Control Protocol (TCP) is one of the core protocols of the Internet protocol suite. TCP provides reliable, in-order delivery of a stream of bytes, making it suitable for applications like file transfer and e-mail. It is so important in the Internet protocol suite that sometimes the entire suite is referred to as "the TCP/IP protocol suite." TCP is the transport protocol that manages the individual conversations between web servers and web clients. TCP divides the HTTP messages into smaller pieces, called segments, to be sent to the destination client. It is also responsible for controlling the size and rate at which messages are exchanged between the server and the client.
The telephone (from the Greek words tele (????) = far and phone (????) = voice) is a telecommunications device that is used to transmit and receive sound (most commonly speech), usually two people conversing but occasionally three or more. It is one of the most common household appliances in the world today. Most telephones operate through transmission of electric signals over a complex telephone network which allows almost any phone user to communicate with almost anyone.
TELNET (TELecommunication NETwork) is a network protocol used on the Internet or local area network (LAN) connections. It was developed in 1969 beginning with RFC 15 and standardized as IETF STD 8, one of the first Internet standards.
The term telnet also refers to software which implements the client part of the protocol. TELNET clients have been available on most Unix systems for many years and are available for virtually all platforms. Most network equipment and OSs with a TCP/IP stack support some kind of TELNET service server for their remote configuration (including ones based on Windows NT). Because of security issues with TELNET, its use has waned as it is replaced by the use of SSH for remote access.
"To telnet" is also used as a verb meaning to establish or use a TELNET or other interactive TCP connection, as in, "To change your password, telnet to the server and run the passwd command".
Most often, a user will be telneting to a Unix-like server system or a simple network device such as a switch. For example, a user might "telnet in from home to check his mail at school". In doing so, he would be using a telnet client to connect from his computer to one of his servers. Once the connection is established, he would then log in with his account information and execute operating system commands remotely on that computer, such as ls or cd.
On many systems, the client may also be used to make interactive raw-TCP sessions. It is commonly believed that a telnet session which does not include an IAC (character 255) is functionally identical. This is not the case however due to special NVT (Network Virtual Terminal) rules such as the requirement for a bare CR (ASCII 13) to be followed by a NULL (ASCII 0).
TLS - Transport Layer Security
Transport Layer Security (TLS) and its predecessor, Secure Sockets Layer (SSL), are cryptographic protocols that provide secure communications on the Internet for such things as web browsing, e-mail, Internet faxing, instant messaging and other data transfers. There are slight differences between SSL and TLS, but they are substantially the same.
User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. Using UDP, programs on networked computers can send short messages sometimes known as datagrams (using Datagram Sockets) to one another. UDP is sometimes called the Universal Datagram Protocol. The protocol was designed by David P. Reed in 1980.
UDP does not guarantee reliability or ordering in the way that TCP does. Datagrams may arrive out of order, appear duplicated, or go missing without notice. Avoiding the overhead of checking whether every packet actually arrived makes UDP faster and more efficient, for applications that do not need guaranteed delivery. Time-sensitive applications often use UDP because dropped packets are preferable to delayed packets. UDP's stateless nature is also useful for servers that answer small queries from huge numbers of clients. Unlike TCP, UDP is compatible with packet broadcast (sending to all on local network) and multicasting (send to all subscribers).
Common network applications that use UDP include: the Domain Name System (DNS), streaming media applications such as IPTV, Voice over IP (VoIP), Trivial File Transfer Protocol (TFTP) and online games.
XMPP - Extensible Messaging and Presence Protocol
eXtensible Messaging and Presence Protocol (XMPP) (formerly known as Jabber) is an open, XML-inspired protocol for near real time, extensible instant messaging (IM) and presence information (a.k.a. buddy lists). The protocol is built to be extensible and other features such as Voice over IP and file transfer signaling have been added.
Unlike most instant messaging protocols, XMPP is based on open standards [1]. Like e-mail, it is an open system where anyone who has a domain name and a suitable Internet connection can run their own Jabber server and talk to users on other servers. The standard server implementations and many clients are also free and open source software.
The Internet Engineering Task Force (IETF) formed an XMPP Working Group in 2002 [2] to formalize the core protocols as an IETF Instant Messaging and presence technology. The four specifications produced by the XMPP WG were approved by the IESG as Proposed Standards in 2004. RFC 3920 and RFC 3921 are currently undergoing revisions (see Development) in preparation for advancing them to Draft Standard within the Internet Standards Process. The XMPP Standards Foundation (formerly the Jabber Software Foundation) is active in developing open XMPP extensions[3]. Unfortunately no Jabber technology correctly implements the RFCs in full.[citation needed]
XMPP-based software is deployed on thousands of servers across the Internet and by 2003 was used by over ten million people worldwide, according to the XMPP Standards Foundation.[1] Popular commercial servers include the Gizmo Project and Google Talk. Popular client applications include the freeware clients offered by Google and the Gizmo Project, multi-protocol instant messengers such as iChat and Pidgin (formerly Gaim), and free dedicated clients such as Psi.
