August 8, 2010

Internet Protocol Suite: Application Layer Protocols

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Internet protocol Suite- The Internet Protocol Suite (commonly known as TCP/IP) is the set of communications protocols used for the Internet and other similar networks. It is named from two of the most important protocols in it: the Transmission Control Protocol (TCP) and the Internet Protocol (IP), which were the first two networking protocols defined in this standard. Today's IP networking represents a synthesis of several developments that began to evolve in the 1960s and 1970s, namely the Internet and LANs (Local Area Networks), which emerged in the mid- to late-1980s, together with the advent of the World Wide Web in the early 1990s.

The Internet Protocol Suite, like many protocol suites, may be viewed as a set of layers. Each layer solves a set of problems involving the transmission of data, and provides a well-defined service to the upper layer protocols based on using services from some lower layers. Upper layers are logically closer to the user and deal with more abstract data, relying on lower layer protocols to translate data into forms that can eventually be physically transmitted.

The TCP/IP model consists of four layers (RFC 1122). From lowest to highest, these are the Link Layer, the Internet Layer, the Transport Layer, and the Application Layer.
 
Dynamic Host Configuration Protocol
In computer networking, the Dynamic Host Configuration Protocol (DHCP) is a network application protocol used by devices (DHCP clients) to obtain configuration information for operation in an Internet Protocol network. This protocol reduces system administration workload, allowing networks to add devices with little or no manual intervention.

RFC 1531 initially defined DHCP as a standard-track protocol in October 1993, succeeding the Bootstrap Protocol (BOOTP). The next update, RFC 2131 released in 1997 is the current DHCP definition for Internet Protocol version 4 (IPv4) networks. The extensions of DHCP for IPv6 (DHCPv6) were published as RFC 3315

Dynamic Host Configuration Protocol automates network-parameter assignment to network devices from one or multiple, fault-tolerant DHCP servers. Even in small networks, DHCP is useful because it can make it easy to add new machines to the network.

When a DHCP-configured client (a computer or any other network-aware device) connects to a network, the DHCP client sends a broadcast query requesting necessary information from a DHCP server. The DHCP server manages a pool of IP addresses and information about client configuration parameters such as default gateway, domain name, the DNS servers, other servers such as time servers, and so forth. On receiving a valid request, the server assigns the computer an IP address, a lease (length of time the allocation is valid), and other IP configuration parameters, such as the subnet mask and the default gateway. The query is typically initiated immediately after booting, and must complete before the client can initiate IP-based communication with other hosts.

Depending on implementation, the DHCP server may have three methods of allocating IP-addresses:

  • dynamic allocation: A network administrator assigns a range of IP addresses to DHCP, and each client computer on the LAN has its IP software configured to request an IP address from the DHCP server during network initialization. The request-and-grant process uses a lease concept with a controllable time period, allowing the DHCP server to reclaim (and then reallocate) IP addresses that are not renewed (dynamic re-use of IP addresses).'

  • automatic allocation: The DHCP server permanently assigns a free IP address to a requesting 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, so that it can preferentially assign to a client the same IP address that the client previously had.

  • static allocation: The DHCP server allocates an IP address based on a table with MAC address/IP address pairs, which are manually filled in (perhaps by a network administrator). Only requesting clients with a MAC address listed in this table will be allocated an IP address. This feature (which is not supported by all routers) is variously called Static DHCP Assignment (by DD-WRT), fixed-address (by the dhcpd documentation), DHCP reservation or Static DHCP (by Cisco/Linksys), and IP reservation or MAC/IP binding (by various other router manufacturers).
Domain Name System
The Domain Name System (DNS) is a hierarchical naming system for computers, services, or any resource connected to the Internet or a private network. It associates various information with domain names assigned to each of the participants. Most importantly, it translates domain names meaningful to humans into the numerical (binary) identifiers associated with networking equipment for the purpose of locating and addressing these devices worldwide. An often used analogy to explain the Domain Name System is that it serves as the "phone book" for the Internet by translating human-friendly computer hostnames into IP addresses. For example, www.example.com translates to 208.77.188.166.



The Domain Name System makes it possible to assign domain names to groups of Internet users in a meaningful way, independent of each user's physical location. Because of this, World-Wide Web (WWW) hyperlinks and Internet contact information can remain consistent and constant even if the current Internet routing arrangements change or the participant uses a mobile device. Internet domain names are easier to remember than IP addresses such as 208.77.188.166 (IPv4) or 2001:db8:1f70::999:de8:7648:6e8 (IPv6). People take advantage of this when they recite meaningful URLs and e-mail addresses without having to know how the machine will actually locate them.

File Transfer Protocol (FTP) is a standard network protocol used to exchange and manipulate files over an Internet Protocol computer network, such as the Internet. FTP is built on a client-server architecture and utilizes separate control and data connections between the client and server applications. Client applications were originally interactive command-line tools with a standardized command syntax, but graphical user interfaces have been developed for all desktop operating systems in use today. FTP is also often used as an application component to automatically transfer files for program internal functions. FTP can be used with user-based password authentication or with anonymous user access.

GPRS Tunnelling Protocol (GTP) is a group of IP-based communications protocols used to carry General Packet Radio Service (GPRS) within GSM and UMTS networks.

GTP can be decomposed into separate protocols, GTP-C, GTP-U and GTP'. GTP-C is used within the GPRS core network for signaling between Gateway GPRS Support Nodes (GGSN) and Serving GPRS Support Nodes (SGSN). This allows the SGSN to activate a session on a user's 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 has an independent function. It can be used for carrying charging data from the Charging Data Function (CDF) of the GSM or UMTS network to the Charging Gateway Function (CGF). In most cases, this should mean from many individual network elements such as the GGSNs to a centralized computer that delivers the charging data more conveniently to the network operator's billing center. The GTP protocol is implemented only by SGSNs and GGSNs. GPRS mobile stations (MSs) are connected to a SGSN without being aware of GTP. GTP can be used with UDP or TCP. GTP version one is used only on UDP.

Hypertext Transfer Protocol (HTTP) is an application-level protocol for distributed, collaborative, hypermedia information systems. Its use for retrieving inter-linked resources led to the establishment of the World Wide Web.

HTTP development was coordinated by the World Wide Web Consortium and the Internet Engineering Task Force (IETF), culminating in the publication of a series of Requests for Comments (RFCs), most notably RFC 2616 (June 1999), which defines HTTP/1.1, the version of HTTP in common use.

Support for pre-standard HTTP/1.1 based on the then developing RFC 2068 was rapidly adopted by the major browser developers in early 1996. By March 1996, pre-standard HTTP/1.1 was supported in Netscape 2.0, Netscape Navigator Gold 2.01, Mosaic 2.7, Lynx 2.5, and in Internet Explorer 3.0. End user adoption of the new browsers was rapid. In March 1996, one web hosting company reported that over 40% of browsers in use on the Internet were HTTP 1.1 compliant. That same web hosting company reported that by June 1996, 65% of all browsers accessing their servers were HTTP 1.1 Compliant. The HTTP 1.1 standard as defined in RFC 2068 was officially released in January 1997. Improvements and updates to the The HTTP/1.1 standard were released under RFC 2616 in June 1999.

HTTP is a request/response standard of a client and a server. A client is the end-user, the server is the web site. The client making an HTTP request—using a web browser, spider, or other end-user tool—is referred to as the user agent. The responding server—which stores or creates resources such as HTML files and images—is called the origin server. In between the user agent and origin server may be several intermediaries, such as proxies, gateways, and tunnels. HTTP is not constrained to using TCP/IP and its supporting layers, although this is its most popular application on the Internet. Indeed HTTP can be "implemented on top of any other protocol on the Internet, or on other networks." HTTP only presumes a reliable transport; any protocol that provides such guarantees can be used."

Typically, an HTTP client initiates a request. It establishes a Transmission Control Protocol (TCP) connection to a particular port on a host (port 80 by default; see List of TCP and UDP port numbers). An HTTP server listening on that port waits for the client to send a request message. Upon receiving the request, the server sends back a status line, such as "HTTP/1.1 200 OK", and a message of its own, the body of which is perhaps the requested resource, an error message, or some other information.

Resources to be accessed by HTTP are identified using Uniform Resource Identifiers (URIs)—or, more specifically, Uniform Resource Locators (URLs)—using the http: or https URI schemes.

[Header = IMAP]
The Internet Message Access Protocol (IMAP) is one of the two most prevalent Internet standard protocols for e-mail retrieval, the other being the Post Office Protocol. Virtually all modern e-mail clients and mail servers support both protocols as a means of transferring e-mail messages from a server, such as those used by Gmail, to a client, such as Mozilla Thunderbird, KMail, Apple Mail and Microsoft Outlook.

E-mail messages are usually sent to an e-mail server that stores received messages in the recipient's e-mail mailbox. The user retrieves messages with either a web browser or an e-mail client that uses one of a number of e-mail retrieval protocols. Some clients and servers preferentially use vendor-specific, proprietary protocols, but most support the Internet standard protocols, SMTP for sending e-mail and POP and IMAP for retrieving e-mail, allowing interoperability with other servers and clients. SMTP can also be used for retrieving email; it is more suitable for permanent Internet connection than, say, a dialup connection, and is supported by most e-mail client software. For example, Microsoft's Outlook client uses a proprietary protocol to communicate with an Exchange server as does IBM's Notes client when communicating with a Domino server, but all of these products also support POP, IMAP, and outgoing SMTP. Support for the Internet standard protocols allows many e-mail clients such as Pegasus Mail or Mozilla Thunderbird (see comparison of e-mail clients) to access these servers, and allows the clients to be used with other servers (see list of mail servers).

[Header = IRC]
Internet Relay Chat (IRC) is a form of real-time Internet text messaging (chat) or synchronous conferencing. It is mainly designed for group communication in discussion forums, called channels, but also allows one-to-one communication via private message as well as chat and data transfers via Direct Client-to-Client.

As of May 2009, the top 100 IRC networks served more than half a million users at a time, with hundreds of thousands of channels (the vast majority of which stand mostly vacant), operating on a total of roughly 1,500 servers worldwide.

IRC was created by Jarkko Oikarinen in 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 was used to report on the 1991 Soviet coup d'├ętat attempt throughout a media blackout. It was previously used in a similar fashion during the Gulf War. Logs of these and other events are kept in the ibiblio archive.

IRC client software is available for virtually every computer operating system that supports TCP/IP networking.

[Header = MEGACO]
Media Gateway Control Protocol (Megaco)
Megaco (officially H.248) is an implementation of the Media Gateway Control Protocol architecture for controlling Media Gateways on Internet Protocol (IP) networks and the public switched telephone network (PSTN). The general base architecture and programming interface was originally described in RFC 2805 and the current specific Megaco definition is ITU-T Recommendation H.248.1.

Megaco defines the protocol for Media Gateway Controllers to control Media Gateways for the support of multimedia streams across computer networks. It is typically used to provide Voice over Internet Protocol (VoIP) services (voice and fax) between IP networks and the PSTN, or entirely within IP networks.

The protocol was the result of collaboration of the MEGACO working group of the Internet Engineering Task Force (IETF) and International Telecommunication Union ITU-T Study Group 16. The IETF originally published the standard as RFC 3015, which was later replaced by RFC 3525. The term Megaco is the IETF designation. The ITU later took ownership of the protocol and IETF's version has been reclassified as historic. The ITU has published three versions of H.248.1, the most recent in September 2005.

H.248 encompasses not only the base protocol specification in H.248.1, but many extensions defined throughout the H.248 Sub-series.

Another implementation of the Media Gateway Control Protocol architecture exists in the similarly named MGCP protocol. This is used over the same interface and similar in application and service functionality, however, it is a different protocol and the underlying differences make them incompatible.

[Header = MGCP]
Media Gateway Control Protocol (MGCP)
MGCP is an implementation of the Media Gateway Control Protocol architecture for controlling Media Gateways on Internet Protocol (IP) networks and the public switched telephone network (PSTN). The general base architecture and programming interface is described in RFC 2805 and the current specific MGCP definition is RFC 3435 (obsoleted RFC 2705). It is a successor to the Simple Gateway Control Protocol (SGCP).

MGCP is a signaling and call control protocol used within Voice over IP (VoIP) systems that typically interoperate with the public switched telephone network (PSTN). As such it implements a PSTN-over-IP model with the power of the network residing in a call control center (softswitch, similar to the central office of the PSTN) and the endpoints being "low-intelligence" devices, mostly simply executing control commands. The protocol represents a decomposition of other VoIP models, such as H.323, in which the media gateways (e.g., H.323's gatekeeper) have higher levels of signalling intelligence.

MGCP uses the Session Description Protocol (SDP) for specifying and negotiating the media streams to be transmitted in a call session and the Real-time Transport Protocol (RTP) for framing of the media streams.

Another implementation of the Media Gateway Control Protocol architecture exists in the similarly named Megaco protocol, a collaboration of the Internet Engineering Task Force (RFC 3525) and International Telecommunication Union (Recommendation H.248.1). Both protocols follow the guidelines of the API Media Gateway Control Protocol Architecture and Requirements in RFC 2805. However, the protocols are incompatible due to differences in protocol syntax and underlying connection model.

[Header = 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.

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.

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.

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.


[Header = NTP]
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 on port 123 as its transport layer. It is designed particularly to resist the effects of variable latency by using a jitter buffer. NTP also refers to a reference software implementation that is distributed by the NTP Public Services Project.

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.

[Header = POP]
Post Office Protocol
In computing, the Post Office Protocol (POP) 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 and IMAP (Internet Message Access Protocol) are the two most prevalent Internet standard protocols for e-mail retrieval. Virtually all modern e-mail clients and servers support both. The POP protocol has been developed through several versions, with version 3 (POP3) being the current standard.

The design of POP and its procedures supports end-users with temporary Internet connections, such as dial-up access, allowing these users to retrieve e-mail when connected and then to view and manipulate the retrieved messages when offline. Although most clients have an option to leave mail on server, e-mail clients using POP generally connect, retrieve all messages, store them on the user's PC as new messages, delete them from the server, and then disconnect.

In contrast, the newer, more capable Internet Message Access Protocol (IMAP) supports both connected (online) and disconnected (offline) modes of operation. E-mail clients using IMAP generally leave messages on the server until the user explicitly deletes them. This and other aspects of IMAP operation allow multiple clients to access the same mailbox. Many e-mail clients support POP as well as IMAP to retrieve messages; however, fewer Internet Service Providers (ISPs) support IMAP.

[Header = RIP]
Routing Information Protocol
The Routing Information Protocol (RIP) is a dynamic routing protocol used in local and wide area networks. As such it is classified as an interior gateway protocol (IGP). It uses the distance-vector routing algorithm. It was first defined in RFC 1058 (1988). The protocol has since been extended several times, resulting in RIP Version 2 (RFC 2453). Both versions are still in use today, however, they are considered technically obsoleted by more advanced techniques, Open Shortest Path First (OSPF) and the OSI protocol IS-IS. RIP has also been adapted for use in IPv6 networks, a standard known as RIPng (RIP next generation), published in RFC 2080 (1997).

[Header = RPC]
Remote procedure call (RPC) is an Inter-process communication 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.


[Header = RTP]
The Real-time Transport Protocol (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, and superseded by RFC 3550 in 2003.

RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications and web-based push to talk features. For these it carries media streams controlled by H.323, MGCP, Megaco, SCCP, or Session Initiation Protocol (SIP) signaling protocols, making it one of the technical foundations of the Voice over IP industry.

RTP is usually used in conjunction with the RTP Control Protocol (RTCP). While RTP carries the media streams (e.g., audio and video) or out-of-band signaling (DTMF), RTCP is used to monitor transmission statistics and quality of service (QoS) information. When both protocols are used in conjunction, RTP is usually originated and received on even port numbers, whereas RTCP uses the next higher odd port number.

[Header = SDP]
The Session Description Protocol (SDP) is a format for describing streaming media initialization parameters in an ASCII string. The IETF published the original specification as an IETF Proposed Standard in April 1998, and subsequently published a revised specification as an IETF Proposed Standard as RFC 4566 in July 2006.

SDP is intended for describing multimedia communication sessions for the purposes of session announcement, session invitation, and parameter negotiation. SDP does not deliver media itself but is used for negotiation between end points of media type, format, and all associated properties. The set of properties and parameters are often called a session profile. SDP is designed to be extensible to support new media types and formats.

SDP started off as a component of the Session Announcement Protocol (SAP), but found other uses in conjunction with Real-time Transport Protocol (RTP), Real-time Streaming Protocol (RTSP), Session Initiation Protocol (SIP) and even as a standalone format for describing multicast sessions.

[Header = SIP]
The Session Initiation Protocol (SIP) is a signaling protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP). Other feasible application examples include video conferencing, streaming multimedia distribution, instant messaging, presence information and online games. 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.

SIP was originally designed by Henning Schulzrinne and Mark Handley starting in 1996. The latest version of the specification is RFC 3261 from the IETF Network Working Group. In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IP Multimedia Subsystem (IMS) architecture for IP-based streaming multimedia services in cellular systems.

[Header = SMTP]
Simple Mail Transfer Protocol (SMTP) is an Internet standard for electronic mail (e-mail) transmission across Internet Protocol (IP) networks. SMTP was first defined in RFC 821 (STD 15), and last updated by RFC 5321 (2008) which includes the extended SMTP (ESMTP) additions, and is the protocol in widespread use today.

While electronic mail servers and other mail transfer agents use SMTP to send and receive mail messages, user-level client mail applications typically only use SMTP for sending messages to a mail server for relaying. For receiving messages, client applications usually use either the Post Office Protocol (POP) or the Internet Message Access Protocol (IMAP) to access their mail box accounts on a mail server.

[Header = SOAP]
SOAP, originally defined as Simple Object Access Protocol, is a protocol specification for exchanging structured information in the implementation of Web Services in computer networks. It relies on Extensible Markup Language (XML) as its message format, and usually relies on other Application Layer protocols (most notably Remote Procedure Call (RPC) and HTTP) for message negotiation and transmission. SOAP can form the foundation layer of a web services protocol stack, providing a basic messaging framework upon which web services can be built.

As a layman's example of how SOAP procedures can be used, a SOAP message could be sent to a web service enabled web site (for example, a house price database) with the parameters needed for a search. The site would then return an XML-formatted document with the resulting data (prices, location, features, etc). Because the data is returned in a standardized machine-parseable format, it could then be integrated directly into a third-party site.

[Header = SSH]
Secure Shell or SSH is a network protocol that allows data to be exchanged using a secure channel between two networked devices.  Used primarily on Linux and Unix based systems to access shell accounts, SSH was designed as a replacement for Telnet and other insecure remote shells, which send information, notably passwords, in plaintext, leaving them open for interception. The encryption used by SSH provides confidentiality and integrity of data over an insecure network, such as the Internet.

[Header = TELNET]
Telnet (teletype network) is a network protocol used on the Internet or local area networks to provide a bidirectional interactive communications facility. Typically, telnet provides access to a command-line interface on a remote host via a virtual terminal connection which consists of an 8-bit byte oriented data connection over the Transmission Control Protocol (TCP). User data is interspersed in-band with TELNET control information.

Telnet was developed in 1969 beginning with RFC 15, extended in RFC 854, and standardized as Internet Engineering Task Force (IETF) Internet Standard STD 8, one of the first Internet standards.

The term telnet may also refer to the software that implements the client part of the protocol. Telnet client applications are available for virtually all computer platforms. Most network equipment and operating system with a TCP/IP stack support a Telnet service for remote configuration (including systems based on Windows NT). Because of security issues with Telnet, its use has waned in favor of SSH for remote access.

Transport Layer Security (TLS) and its predecessor, Secure Sockets Layer (SSL), are cryptographic protocols that provide security and data integrity for communications over networks such as the Internet. TLS and SSL encrypt the segments of network connections at the Transport Layer end-to-end.

Several versions of the protocols are in wide-spread use in applications like web browsing, electronic mail, Internet faxing, instant messaging and voice-over-IP (VoIP).

TLS is an IETF standards track protocol, last updated in RFC 5246, that was based on the earlier SSL specifications developed by Netscape Corporation.

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