Simple Network Management Protocol
Simple Network Management Protocol is an Internet Standard protocol for collecting and organizing information about managed devices on IP networks and for modifying that information to change device behavior. Devices that support SNMP include cable modems, switches, workstations and more. SNMP is used in network 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 configuration; these variables can be remotely queried by managing applications. Three significant versions of SNMP have been deployed. SNMPv1 is the original version of the protocol. More recent versions, SNMPv2c and SNMPv3, feature improvements in performance and security. 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 layer protocol, a database schema, a set of data objects.
In typical uses of SNMP, one or more administrative computers called managers have the task of monitoring or managing a group of hosts or devices on a computer network. Each managed system executes a software component called an agent which reports information via SNMP to the manager. An SNMP-managed network consists of three key components: Managed devices Agent – software which runs on managed devices Network management station – software which runs on the managerA managed device is a network node that implements an SNMP interface that allows unidirectional or bidirectional access to node-specific information. Managed devices exchange node-specific information with the NMSs. Sometimes called network elements, the managed devices can be any type of device, but not limited to, access servers, cable modems, hubs, IP telephones, IP video cameras, computer hosts, printers. An agent is a network-management software module. 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 control managed devices. NMSs provide the bulk of the memory resources required for network management. One or more NMSs may exist on any managed network. SNMP agents expose management data on the managed systems as variables; the protocol permits active management tasks, such as configuration changes, through remote modification of these variables. The variables accessible via SNMP are organized in hierarchies. SNMP itself does not define which variables a managed system should offer. Rather, SNMP uses an extensible design; these hierarchies are described as a management information base. MIBs describe the structure of the management data of a device subsystem; 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. All SNMP messages are transported via User Datagram Protocol.
The SNMP agent receives requests on UDP port 161. The manager may send requests from any available source port to port 161 in the agent; the agent response is sent back to the source port on the manager. The manager receives notifications on port 162; the agent may generate notifications from any available port. When used with Transport Layer Security or Datagram Transport Layer Security, requests are received on port 10161 and notifications are sent to port 10162. SNMPv1 specifies five core protocol data units. Two other PDUs, GetBulkRequest and InformRequest were added in SNMPv2 and the Report PDU was added in SNMPv3. All SNMP PDUs are constructed as follows: The seven SNMP PDU types as identified by the PDU-type field are as follows: GetRequest A manager-to-agent request to retrieve the value of a variable or list of variables. Desired variables are specified in variable bindings. Retrieval of the specified variable values is to be done as an atomic operation by the agent. A Response with current values is returned.
SetRequest A manager-to-agent request to change the value of a variable or list of variables. Variable bindings are specified in the body of the request. Changes to all specified variables are to be made as an atomic operation by the agent. A Response with new values for the variables is returned. GetNextRequest A manager-to-agent request to discover available variables and their values. Returns a Response with variable binding for the lexicographically next variable in the MIB; the entire MIB of an agent can be walked by iterative application of GetNextRequest starting at OID 0. Rows of a table can be read by specifying column OIDs in the variable bindings of the request. GetBulkRequest A manager-to-agent request for multiple iterations of GetNextRequest. An optimized version of GetNextRequest. Returns a Response with multiple variable bindings walked from the variable binding or bindings in the request. PDU specific non-repeaters and max-repetitions fields are used to control response behavior.
GetBulkRequest was introduced in SNMPv2. Response Returns variable bindings and acknowledgement from agent to manager for GetRequest, SetRequest, GetNextRequest, GetBulkRequest and InformRequest. Error reporting is provided by error-index fields. Although it was used as a response to both gets and sets, this P
X.25 is an ITU-T standard protocol suite for packet-switched wide area network communication. An X.25 WAN consists of packet-switching exchange nodes as the networking hardware, leased lines, plain old telephone service connections, or ISDN connections as physical links. X.25 was defined by the International Telegraph and Telephone Consultative Committee in a series of drafts and finalized in a publication known as The Orange Book in 1976. X.25 networks were popular during the 1980s with telecommunications companies and in financial transaction systems such as automated teller machines. However, most uses have moved to Internet Protocol systems instead. X.25 is still used and available in niche applications such as Retronet that allows vintage computers to use the internet. X.25 is one of the oldest packet-switched services available. It was developed before the OSI Reference Model; the protocol suite is designed as three conceptual layers, which correspond to the lower three layers of the seven-layer OSI model.
It supports functionality not found in the OSI network layer. X.25 was developed in the ITU-T Study Group VII based upon a number of emerging data network projects. Various updates and additions were worked into the standard recorded in the ITU series of technical books describing the telecommunication systems; these books were published every fourth year with different-colored covers. The X.25 specification is only part of the larger set of X-Series specifications on public data networks. The public data network was the common name given to the international collection of X.25 providers. Their combined network had large global coverage into the 1990s. Publicly accessible X.25 networks were set up in most countries during the 1970s and 1980s, to lower the cost of accessing various online services. Beginning in the early 1990s, in North America, use of X.25 networks started to be replaced by Frame Relay services offered by national telephone companies. Most systems that required X.25 now use TCP/IP, however it is possible to transport X.25 over TCP/IP when necessary.
X.25 networks are still in use throughout the world. A variant called AX.25 is used by amateur packet radio. Racal Paknet, now known as Widanet, is still in operation in many regions of the world, running on an X.25 protocol base. In some countries, like the Netherlands or Germany, it is possible to use a stripped version of X.25 via the D-channel of an ISDN-2 connection for low-volume applications such as point-of-sale terminals. Additionally X.25 is still under heavy use in the aeronautical business though a transition to modern protocols like X.400 is without option as X.25 hardware becomes rare and costly. As as March 2006, the United States National Airspace Data Interchange Network has used X.25 to interconnect remote airfields with Air Route Traffic Control Centers. France was one of the last remaining countries where commercial end-user service based on X.25 operated. Known as Minitel it was based on Videotex, itself running on X.25. In 2002, Minitel had about 9 million users, in 2011, it still accounted for about 2 million users in France when France Télécom announced it would shut down the service by 30 June 2012.
As planned, service was terminated 30 June 2012. There were 800,000 terminals still in operation at the time; the general concept of the X. 25 was to create a global packet-switched network. Much of the X.25 system is a description of the rigorous error correction needed to achieve this, as well as more efficient sharing of capital-intensive physical resources. The X. 25 specification defines only the interface between an X. 25 network. X.75, a protocol similar to X.25, defines the interface between two X.25 networks to allow connections to traverse two or more networks. X.25 does not specify how the network operates internally – many X.25 network implementations used something similar to X.25 or X.75 internally, but others used quite different protocols internally. The ISO protocol equivalent to X.25, ISO 8208, is compatible with X.25, but additionally includes provision for two X.25 DTEs to be directly connected to each other with no network in between. By separating the Packet-Layer Protocol, ISO 8208 permits operation over additional networks such as ISO 8802 LLC2 and the OSI data link layer.
X.25 defined three basic protocol levels or architectural layers. In the original specifications these were referred to as levels and had a level number, whereas all ITU-T X.25 recommendations and ISO 8208 standards released after 1984 refer to them as layers. The layer numbers were dropped to avoid confusion with the OSI Model layers. Physical layer: This layer specifies the physical, electrical and procedural characteristics to control the physical link between a DTE and a DCE. Common implementations use X. 21, EIA-449 or other serial protocols. Data link layer: The data link layer consists of the link access procedure for data interchange on the link between a DTE and a DCE. In its implementation, the Link Access Procedure, Balanced is a data link protocol that manages a communication session and controls the packet framing, it is a bit-oriented protocol that provides orderly delivery. Packet layer: This layer defined a packet-layer protocol for exchanging control and user data packets to form a packet-switching network based on virtual calls, acco
Hypertext Transfer Protocol
The Hypertext Transfer Protocol is an application protocol for distributed, hypermedia information systems. HTTP is the foundation of data communication for the World Wide Web, where hypertext documents include hyperlinks to other resources that the user can access, for example by a mouse click or by tapping the screen in a web browser. HTTP was developed to facilitate the World Wide Web. Development of HTTP was initiated by Tim Berners-Lee at CERN in 1989. Development of HTTP standards 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 RFC 2068 in 1997, although this was made obsolete by RFC 2616 in 1999 and again by the RFC 7230 family of RFCs in 2014. A version, the successor HTTP/2, was standardized in 2015, is now supported by major web servers and browsers over Transport Layer Security using Application-Layer Protocol Negotiation extension where TLS 1.2 or newer is required.
HTTP functions as a request–response protocol in the client–server computing model. A web browser, for example, may be the client and an application running on a computer hosting a website may be the server; 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, returns a response message to the client; the response contains completion status information about the request and may 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, 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 benefit from web cache servers that deliver content on behalf of upstream servers to improve response time.
Web browsers cache accessed web resources and reuse them, when possible, to reduce network traffic. HTTP proxy servers at private network boundaries can facilitate communication for clients without a globally routable address, by relaying messages with external servers. 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, Transmission Control Protocol is used. 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 and https. URIs and hyperlinks in HTML documents form interlinked 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, stylesheets, etc after the page has been delivered.
HTTP/1.1 communications therefore experience less latency as the establishment of TCP connections presents considerable overhead. The term hypertext was coined by Ted Nelson in 1965 in the Xanadu Project, in turn inspired by Vannevar Bush's 1930s vision of the microfilm-based information retrieval and management "memex" system described in his 1945 essay "As We May Think". Tim Berners-Lee and his team at CERN are credited with inventing the original HTTP, along with HTML and the associated technology for a web server and a text-based web browser. 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. The response from the server was always an HTML page; the first documented version of HTTP was HTTP V0.9. Dave Raggett led the HTTP Working Group in 1995 and wanted to expand the protocol with extended operations, extended negotiation, richer meta-information, tied with a security protocol which became more efficient by adding additional methods and header fields.
RFC 1945 introduced and recognized HTTP V1.0 in 1996. The HTTP WG planned to publish new standards in December 1995 and the support for pre-standard HTTP/1.1 based on the developing RFC 2068 was adopted by the major browser developers in early 1996. By March that year, pre-standard HTTP/1.1 was supported in Arena, Netscape 2.0, Netscape Navigator Gold 2.01, Mosaic 2.7, Lynx 2.5, in Internet Explorer 2.0. 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 released in January 1997. Improvements and updates to the HTTP/1.1 standard were released under RFC 2616 in June 1999. In 2007, the HTTPbis Working Group was formed, in part, to revise and clarify the HTTP/1.1 specification. In June 2014, the WG released an updated six-part specification obsoleting RFC 2616: RFC 7230, HTTP/1.1: Message Syntax and Routing RFC 7231, HTTP/1.1: Semantics and Content RFC 7232, HTTP/1.1: Conditional Requests RFC 7233, HTTP/1.1: Range Requests RFC 7234, HTTP/1.1: Caching RFC 7235, HTTP/1
AppleTalk is a discontinued proprietary suite of networking protocols developed by Apple Inc. for their Macintosh computers. AppleTalk includes a number of features that allow local area networks to be connected with no prior setup or the need for a centralized router or server of any sort. Connected AppleTalk-equipped systems automatically assign addresses, update the distributed namespace, configure any required inter-networking routing. AppleTalk was released in 1985, was the primary protocol used by Apple devices through the 1980s and 1990s. Versions were released for the IBM PC and compatibles and the Apple IIGS. AppleTalk support was available in most networked printers, some file servers, a number of routers; the rise of TCP/IP during the 1990s led to a reimplementation of most of these types of support on that protocol, AppleTalk became unsupported as of the release of Mac OS X v10.6 in 2009. Many of AppleTalk's more advanced autoconfiguration features have since been introduced in Bonjour, while Universal Plug and Play serves similar needs.
After the release of the Apple Lisa computer in January 1983, Apple invested considerable effort in the development of a local area networking system for the machines. Known as AppleNet, it was based on the seminal Xerox XNS protocol stack but running on a custom 1 Mbit/s coaxial cable system rather than Xerox's 2.94 Mbit/s Ethernet. AppleNet was announced early in 1983 with a fall introduction at the target price of $500 for plug-in AppleNet cards for the Lisa and the Apple II. At that time, early LAN systems were just coming to market, including Ethernet, Token Ring and ARCNET; this was a topic of major commercial effort at the time, dominating shows like the National Computer Conference in Anaheim in May 1983. All of the systems were jockeying for position in the market, but at this time Ethernet's widespread acceptance suggested it was to become a de facto standard, it was at this show that Steve Jobs asked Gursharan Sidhu a innocuous question, "Why has networking not caught on?"Four months in October, AppleNet was cancelled.
At the time, they announced that "Apple realized that it's not in the business to create a networking system. We built and used AppleNet in-house, but we realized that if we had shipped it, we would have seen new standards coming up." In January, Jobs announced that they would instead be supporting IBM's Token Ring, which he expected to come out in a "few months". Through this period, Apple was deep in development of the Macintosh computer. During development, engineers had made the decision to use the Zilog 8530 serial controller chip instead of the lower-cost and more common UART to provide serial port connections; the SCC cost about $5 more than a UART, but offered much higher speeds of up to 250 kilobits per second and internally supported a number of basic networking-like protocols like IBM's Bisync. The SCC was chosen. Peripherals equipped with similar SCCs could communicate using the built-in protocols, interleaving their data with other peripherals on the same bus; this would eliminate the need for more ports on the back of the machine, allowed for the elimination of expansion slots for supporting more complex devices.
The initial concept was known as AppleBus, envisioning a system controlled by the host Macintosh polling "dumb" devices in a fashion similar to the modern Universal Serial Bus. The Macintosh team had begun work on what would become the LaserWriter, had considered a number of other options to answer the question of how to share these expensive machines and other resources. A series of memos from Bob Belleville clarified these concepts, outlining the Mac, LaserWriter and a file server system which would become the Macintosh Office. By late 1983 it was clear that IBM's Token Ring would not be ready in time for the launch of the Mac, might miss the launch of these other products as well. In the end, Token Ring would not ship until October 1985. Jobs' earlier question to Sidhu had sparked a number of ideas; when AppleNet was cancelled in October, Sidhu led an effort to develop a new networking system based on the AppleBus hardware. This new system would not have to conform to any existing preconceptions, was designed to be worthy of the Mac – a system, user-installable, had zero-configuration, no fixed network addresses – in short, a true plug-and-play network.
Considerable effort was needed, but by the time the Mac was released, the basic concepts had been outlined, some of the low-level protocols were on their way to completion. Sidhu mentioned the work to Belleville; the "new" AppleBus was announced in early 1984, allowing direct connection from the Mac or Lisa through a small box that plugged into the serial port and connected via cables to the next computer upstream and downstream. Adaptors for Apple II and Apple III were announced. Apple announced that AppleBus networks could be attached to, would appear to be a single node within, a Token Ring system. Details of how this would work were sketchy. Just prior to its release in early 1985, AppleBus was renamed AppleTalk; the system had a number of limitations, including a speed of only 230.4 kbit/s, a maximum distance of 1000 feet from end to end, only 32 nodes per LAN. But as the basic hardware was built into the Mac, adding nodes only cost about $50 for the adaptor box. In comparison, Ethernet or Token Ring cards cost thousands of dollars.
Additionally, the entire networking stack required only about 6 kB of RAM, allowing it to run on any Mac. The slow speed of AppleTalk allowed further reductions in cost. Instead of using RS
Stream Control Transmission Protocol
The Stream Control Transmission Protocol is a computer networking communications protocol which operates at the transport layer and serves a role similar to the popular protocols TCP and UDP. It is standardized by IETF in RFC 4960. SCTP provides some of the features of both UDP and TCP: it is message-oriented like UDP and ensures reliable, in-sequence transport of messages with congestion control like TCP, it differs from those protocols by providing multi-homing and redundant paths to increase resilience and reliability. In the absence of native SCTP support in operating systems, it is possible to tunnel SCTP over UDP, as well as to map TCP API calls to SCTP calls so existing applications can use SCTP without modification; the reference implementation was released as part of FreeBSD version 7. It has since been ported; the IETF Signaling Transport working group defined the protocol in the year 2000, the IETF Transport Area working group maintains it. RFC 4960 defines the protocol. RFC 3286 provides an introduction.
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, 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, rather than transporting an unbroken stream of bytes as does TCP; as in UDP, in SCTP a sender sends a message in one operation, that exact message is passed to the receiving application process in one operation. In contrast, TCP is a stream-oriented protocol; however TCP does not allow the receiver to know how many times the sender application called on the TCP transport passing it groups of bytes to be sent out.
At the sender, TCP appends more bytes to a queue of bytes waiting to go out over the network, rather than having to keep a queue of individual separate outbound messages which must be preserved as such. The term multi-streaming refers to the capability of SCTP to transmit several independent streams of chunks in parallel, for example transmitting web page images together with the web page text. 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 byte sequence number with each segment. SCTP, on the other hand, assigns a message-id to each message sent in a stream; this allows independent ordering of messages in different streams. However, message ordering is optional in SCTP. Features of SCTP include: Reliable transmission of both ordered and unordered data streams. Multihoming support in which 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 eliminates unnecessary head-of-line blocking, as opposed to TCP byte-stream delivery. Explicit partial reliability. 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 intended it for the transport of telephony over Internet Protocol, with the goal of duplicating some of the reliability attributes of the SS7 signaling network in IP. This IETF effort is known as SIGTRAN. In the meantime, other uses have been proposed, for example, the Diameter protocol and Reliable Server Pooling. TCP has provided the primary means to transfer data reliably across the Internet. However, TCP has imposed limitations on several applications. From RFC 4960: TCP provides both reliable data transfer and strict order-of-transmission delivery of data.
Some applications need reliable transfer without sequence maintenance, while others would be satisfied with partial ordering of the data. In both of these cases, the head-of-line blocking property of TCP causes unnecessary delay. For applications exchanging distinct records or messages, the stream-oriented nature of TCP requires the addition of explicit markers or other encoding to delineate the individual records. In order to avoid sending many small IP packets where one single larger packet would have sufficed, the TCP implementation may delay transmitting data while waiting for more data being queued by the application. If and when such a small delay is undesirable, the application must explicitly request undelayed transmission on a case-by-case basis using the push facility. SCTP on the other hand allows undelayed transmission to be configured as a default for an association, eliminating any undesired delays, but at the cost of higher transfer overhead; the limited scope of TCP sockets complicates the task of providing highly-available data transfer capability using multi-homed hosts.
TCP is vulnerable to denial-of-service attacks, such as SYN attacks. Adoption has been slowed by lack of awareness, lack of implementations (particularly in Microsoft Windows
Frame Relay is a standardized wide area network technology that specifies the physical and data link layers of digital telecommunications channels using a packet switching methodology. Designed for transport across Integrated Services Digital Network infrastructure, it may be used today in the context of many other network interfaces. Network providers implement Frame Relay for voice and data as an encapsulation technique used between local area networks over a wide area network; each end-user gets a private line to a Frame Relay node. The Frame Relay network handles the transmission over a changing path transparent to all end-user extensively used WAN protocols, it is less expensive than leased lines and, one reason for its popularity. The extreme simplicity of configuring user equipment in a Frame Relay network offers another reason for Frame Relay's popularity. With the advent of Ethernet over fiber optics, MPLS, VPN and dedicated broadband services such as cable modem and DSL, the end may loom for the Frame Relay protocol and encapsulation.
However many rural areas remain lacking cable modem services. In such cases, the least expensive type of non-dial-up connection remains a 64-kbit/s Frame Relay line, thus a retail chain, for instance, may use Frame Relay for connecting rural stores into their corporate WAN. The designers of Frame Relay aimed to provide a telecommunication service for cost-efficient data transmission for intermittent traffic between local area networks and between end-points in a wide area network. Frame Relay puts data in variable-size units called "frames" and leaves any necessary error-correction up to the end-points; this speeds up overall data transmission. For most services, the network provides a permanent virtual circuit, which means that the customer sees a continuous, dedicated connection without having to pay for a full-time leased line, while the service-provider figures out the route each frame travels to its destination and can charge based on usage. An enterprise can select a level of service quality, prioritizing some frames and making others less important.
Frame Relay can run on full T-carrier system carriers. Frame Relay complements and provides a mid-range service between basic rate ISDN, which offers bandwidth at 128 kbit/s, Asynchronous Transfer Mode, which operates in somewhat similar fashion to Frame Relay but at speeds from 155.520 Mbit/s to 622.080 Mbit/s. Frame Relay has its technical base in the older X.25 packet-switching technology, designed for transmitting data on analog voice lines. Unlike X.25, whose designers expected analog signals with a high chance of transmission errors, Frame Relay is a fast packet switching technology operating over links with a low chance of transmission errors, which means that the protocol does not attempt to correct errors. When a Frame Relay network detects an error in a frame, it drops that frame; the end points have the responsibility for detecting and retransmitting dropped frames. Frame Relay serves to connect local area networks with major backbones, as well as on public wide-area networks and in private network environments with leased lines over T-1 lines.
It requires a dedicated connection during the transmission period. Frame Relay does not provide an ideal path for voice or video transmission, both of which require a steady flow of transmissions. However, under certain circumstances and video transmission do use Frame Relay. Frame Relay originated as an extension of integrated services digital network, its designers aimed to enable a packet-switched network to transport over circuit-switched technology. The technology has become a stand-alone and cost-effective means of creating a WAN. Frame Relay switches create virtual circuits to connect remote LANs to a WAN; the Frame Relay network exists between a LAN border device a router, the carrier switch. The technology used by the carrier to transport data between the switches is variable and may differ among carriers; the sophistication of the technology requires a thorough understanding of the terms used to describe how Frame Relay works. Without a firm understanding of Frame Relay, it is difficult to troubleshoot its performance.
Frame-relay frame structure mirrors exactly that defined for LAP-D. Traffic analysis can distinguish Frame Relay format from LAP-D by its lack of a control field; each Frame Relay protocol data unit consists of the following fields: Flag Field. The flag is used to perform high-level data link synchronization which indicates the beginning and end of the frame with the unique pattern 01111110. To ensure that the 01111110 pattern does not appear somewhere inside the frame, bit stuffing and destuffing procedures are used. Address Field; each address field may occupy either octet 2 to 3, octet 2 to 4, or octet 2 to 5, depending on the range of the address in use. A two-octet address field comprises the EA=ADDRESS FIELD EXTENSION BITS and the C/R=COMMAND/RESPONSE BIT. DLCI-Data Link Connection Identifier Bits; the DLCI serves to identify the virtual connection so that the receiving end knows which information connection a frame belongs to. Note that this DLCI has only local significance. A single physical channel can multiplex several different virtual connections.
FECN, BECN, DE bits. These bits report congestion: FECN=Forward Explicit Congestion Notific
Session Initiation Protocol
The Session Initiation Protocol is a signaling protocol used for initiating and terminating real-time sessions that include voice and messaging applications. SIP is used for signaling and controlling multimedia communication sessions in applications of Internet telephony for voice and video calls, in private IP telephone systems, in instant messaging over Internet Protocol networks as well as mobile phone calling over LTE; the protocol defines the specific format of messages exchanged and the sequence of communications for cooperation of the participants. SIP is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol and the Simple Mail Transfer Protocol. A call established with SIP may consist of multiple media streams, but no separate streams are required for applications, such as text messaging, that exchange data as payload in the SIP message. SIP works in conjunction with several other protocols that carry the session media. Most media type and parameter negotiation and media setup is performed with the Session Description Protocol, carried as payload in SIP messages.
SIP is designed to be independent of the underlying transport layer protocol, can be used with the User Datagram Protocol, the Transmission Control Protocol, the Stream Control Transmission Protocol. For secure transmissions of SIP messages over insecure network links, the protocol may be encrypted with Transport Layer Security. For the transmission of media streams the SDP payload carried in SIP messages employs the Real-time Transport Protocol or the Secure Real-time Transport Protocol. SIP was designed by Mark Handley, Henning Schulzrinne, Eve Schooler and Jonathan Rosenberg in 1996; the protocol was standardized as RFC 2543 in 1999. In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IP Multimedia Subsystem architecture for IP-based streaming multimedia services in cellular networks. In June 2002 the specification was revised in RFC 3261 and various extensions and clarifications have been published since. SIP was designed to provide a signaling and call setup protocol for IP-based communications supporting the call processing functions and features present in the public switched telephone network with a vision of supporting new multimedia applications.
It has been extended for video conferencing, streaming media distribution, instant messaging, presence information, file transfer, Internet fax and online games. SIP is distinguished by its proponents for having roots in the Internet community rather than in the telecommunications industry. SIP has been standardized by the IETF, while other protocols, such as H.323, have traditionally been associated with the International Telecommunication Union. SIP is only involved for the signaling operations of a media communication session and is used to set up and terminate voice or video calls. SIP can be used to establish multiparty sessions, it allows modification of existing calls. The modification can involve changing addresses or ports, inviting more participants, adding or deleting media streams. SIP has found applications in messaging applications, such as instant messaging, event subscription and notification. SIP works in conjunction with several other protocols that specify the media format and coding and that carry the media once the call is set up.
For call setup, the body of a SIP message contains a Session Description Protocol data unit, which specifies the media format and media communication protocol. Voice and video media streams are carried between the terminals using the Real-time Transport Protocol or Secure Real-time Transport Protocol; every resource of a SIP network, such as user agents, call routers, voicemail boxes, are identified by a Uniform Resource Identifier. The syntax of the URI follows the general standard syntax used in Web services and e-mail; the URI scheme used for SIP is sip and a typical SIP URI has the form sip:username@domainname or sip:username@hostport, where domainname requires DNS SRV records to locate the servers for SIP domain while hostport can be an IP address or a qualified domain name of the host and port. If secure transmission is required, the scheme sips is used. SIP employs design elements similar to the HTTP request/response transaction model; each transaction consists of a client request that invokes a particular method or function on the server and at least one response.
SIP reuses most of the header fields, encoding rules and status codes of HTTP, providing a readable text-based format. SIP can be carried by several transport layer protocols including Transmission Control Protocol, User Datagram Protocol, Stream Control Transmission Protocol. SIP clients use TCP or UDP on port numbers 5060 or 5061 for SIP traffic to servers and other endpoints. Port 5060 is used for non-encrypted signaling traffic whereas port 5061 is used for traffic encrypted with Transport Layer Security. SIP-based telephony networks implement call processing features of Signaling System 7, for which special SIP protocol extensions exist, although the two protocols themselves are different. SS7 is a centralized protocol, characterized by a complex central network architecture and dumb endpoints. SIP is a client-server protocol of equipotent peers. SIP features are implemented in the communicating endpoints, while the traditional SS7 architecture is in use only between switching centers; the network elements that use the Session Initiation Protocol for commun