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Communications protocol Internet Protocol Internet protocol suite Signalling System No. 7 Public switched telephone network Voice over IP Session Initiation Protocol User Datagram Protocol User agent Hypertext Transfer Protocol H.323 H.225.0 Skinny Call Control Protocol Real Time Streaming Protocol Real-time Transport Protocol RTP Control Protocol Telephone number mapping ITU-T European Telecommunications Standards Institute Internet Engineering Task Force Request for Comments Institute of Electrical and Electronics Engineers Session Description Protocol 1 16 20 27 31 34 47 52 58 61 67 79 80 81 83 88 90 96 104 106 109 113 121

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Communications protocol

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Communications protocol
A communications protocol is a system of digital message formats and rules for exchanging those messages in or between computing systems and in telecommunications. A protocol may have a formal description. Protocols may include signaling, authentication and error detection and correction capabilities. A protocol definition defines the syntax, semantics, and synchronization of communication; the specified behavior is typically independent of how it is to be implemented. A protocol can therefore be implemented as hardware or software or both.

Introduction
While there is no generally accepted formal definition of "protocol" in computer science, an informal definition, based on the previous, could be "a set of procedures to be followed when communicating". In computer science the word algorithm is a synonym for the word procedure so a protocol is to communications what an algorithm is to mathematics. Communicating systems use well-defined formats for exchanging messages. Each message has an exact meaning intended to provoke a defined response of the receiver. A protocol therefore describes the syntax, semantics, and synchronization of communication. A programming language describes the same for computations, so there is a close analogy between protocols and programming languages: protocols are to communications what programming languages are to computations.[1] The communications protocols in use on the Internet are designed to function in very complex and diverse settings. To ease design, communications protocols are structured using a layering scheme as a basis. Instead of using a single universal protocol to handle all transmission tasks, a set of cooperating protocols fitting the layering scheme is used.[2] The layering scheme in use on the Internet is called the TCP/IP model. The actual protocols are collectively called the Internet protocol suite. The group responsible for this design is called the Internet Engineering Task Force (IETF). Obviously the number of layers of a layering scheme and the way the layers are defined can have a drastic impact on the protocols involved. This is where the analogies come into play for the TCP/IP model, because the designers of TCP/IP employed the same Figure 2. The TCP/IP model or Internet layering scheme and its relation to some common protocols. techniques used to conquer the complexity of programming language compilers (design by analogy) in the implementation of its protocols and its layering scheme.[3] Communications protocols have to be agreed upon by the parties involved. To reach agreement a protocol is developed into a technical standard.

The rules describing the meanings of the address value are collectively called an addressing scheme. transport-. because there is no common memory. The bitstrings are divided in fields and each field carries information relevant to the protocol. the rules can be expressed by algorithms and data structures. layering had proven to be a successful design approach for both compiler and operating system design and.[8] Systems typically do not use a single protocol to handle a transmission. Usually some address values have special meanings. internet. AX. Digital message bitstrings are exchanged.[14] • Address mapping. makes the protocol software operating system independent. Instead they use a set of cooperating protocols.Communications protocol 2 Communicating systems The information exchanged between devices on a network or other communications medium is governed by rules or conventions that can be set out in a technical specification called a communication protocol standard. Conceptually the bitstring is divided into two parts called the header area and the data area.[6] The best known frameworks are the TCP/IP model and the OSI model. much of the following should be addressed:[12] • Data formats for data exchange. This communication is governed by well-understood protocols. X.25. layering was applied to the protocols as well. At the time the Internet was developed. the protocol software modules are interfaced with a framework implemented on the machine's operating system. receiver address). The protocols can be arranged based on functionality in groups. To implement a networking protocol. In general. For instance to translate a logical IP address specified by the application to an Ethernet hardware address. so the header area contains the fields with more relevance to the protocol. The functionalities are mapped onto the layers. the actual data exchanged and any state-dependent behaviors are defined by the specification. The nature of the communication. The actual message is stored in the data area. Expressing the algorithms in a portable programming language. so sending to this address would result in a broadcast on the local network. AppleTalk and TCP/IP. so some sort of multiplexing/demultiplexing takes place. This is . Bitstrings longer than the maximum transmission unit (MTU) are divided in pieces of appropriate size. Sometimes protocols need to map addresses of one scheme on addresses of another scheme.and network interface-functions.[9] Some of the best known protocol suites include: IPX/SPX.25. for instance there is a group of transport protocols. Transmission is not necessarily reliable and can involve different hardware and operating systems on different systems.[7] This gave rise to the concept of layered protocols which nowadays forms the basis of protocol design. The selection of the next protocol is accomplished by extending the message with a protocolselector for each layer. Protocols should therefore specify rules governing the transmission. each layer solving a distinct class of problems relating to. These protocols can be embedded in the process code itself as small additional code fragments. communicating systems have to communicate with each other using shared transmission media. Operating systems are usually conceived of as consisting of a set of cooperating processes that manipulate a shared store (on the system itself) to communicate with each other.[10] To transmit a message. A connection between a sender and a receiver can be identified using an address pair (sender address.[13] • Address formats for data exchange. This framework implements the networking functionality of the operating system. Addresses are used to identify both the sender and the intended receiver(s). An all-1s address could be taken to mean an addressing of all stations on the network.[4] [5] In contrast. for instance: application-. sometimes called a protocol family or protocol suite. a protocol has to be selected from each layer. given the similarities between programming languages and communication protocols. In digital computing systems. The addresses are stored in the header area of the bitstrings.[11] Basic requirements of protocols Messages are sent and received on communicating systems to establish communications. allowing the receivers to determine whether the bitstrings are intended for themselves and should be processed or should be ignored.

We have seen that long bitstrings are divided in pieces. One has to consider the fact that algorithms. and then sent on the network individually. which is used as a record of the effects of assignments. the retransmission has no effect so the number of retransmissions is limited. Detection of transmission errors is necessary on networks which cannot guarantee error-free operation. Arrangements have to be made to accommodate the case when two parties want to gain control at the same time. Exceeding the retry limit is considered an error. Retransmissions can result duplicate pieces. Both intuitive descriptions as well as more formal specifications in the form of finite state machine models are used to describe the expected interactions of the protocol.Communications protocol referred to as address mapping. used as a record of identifier bindings.[21] Getting the data across a network is only part of the problem for a protocol. On timeouts.[22] Formal ways for describing the syntax of the communications are Abstract Syntax Notation One (a ISO standard) or Augmented Backus-Naur form (a IETF standard). making it possible for the receiver to detect differences caused by errors. By marking the pieces with sequence information at the sender.[19] • Sequence control.[16] Acknowledgements of correct reception of packets is required for connection oriented communication.[25] . Protocols and programming languages Protocols are to communications what algorithms or programming languages are to computations. programs and protocols are just different ways of describing expected behaviour of interacting objects. Acknowledgements are sent from receivers back to their respective senders. As a result pieces may arrive out of sequence. CRCs of the data area are added to the end of packets. The localized association of an identifier to a value established by a definition is termed a binding and the region of program text in which a binding is effective is known as its scope. intermediary systems along the route to the intended receiver(s) need to forward messages on behalf of the sender. Packets may be lost on the network or suffer from long delays. On the Internet.[23] [1] This analogy has important consequences for both the design and the development of protocols. so a protocol has to specify rules describing the context. ask for necessary retransmissions and reassemble the original message.[18] 3 • • • • • Direction of information flow needs to be addressed if transmissions can only occur in one direction at a time as on half-duplex links. Program text is structured using block constructs and definitions can be local to a block. These kind of rules are said to express the syntax of the communications. To cope with this. When systems are not directly connected. This is known as Media Access Control. Other rules determine whether the data is meaningful for the context in which the exchange takes place. This way of connecting networks is called internetworking.[24] The computational state is kept using two components: the environment. In programming languages the association of identifiers to a value is termed a definition. A familiar example of a protocolling language is the HTML language used to describe web pages which are the actual web protocols. a sender may expect an acknowledgement of correct reception from the receiver within a certain amount of time.[17] Loss of information .[15] Routing. These kind of rules are said to express the semantics of the communications.timeouts and retries. the receiver can determine what was lost or duplicated. In a common approach. The data received has to be evaluated in the context of the progress of the conversation. The receiver rejects the packets on CRC differences and arranges somehow for retransmission. the sender must assume the packet was not received and retransmit it. Flow control can be implemented by messaging from receiver to sender. and the store.[20] • Flow control is needed when the sender transmits faster than the receiver or intermediate network equipment can process the transmissions. under some protocols. In case of a permanently broken link. the networks are connected using routers. The pieces may get lost or delayed or take different routes to their destination on some types of networks.

email and web browsing). By analogy. to standardize protocols. allowing other network architectures.Communications protocol In communications.[29] Formal verification . Networking protocols operate in very heterogeneous environments consisting of very different network technologies and a (possibly) very rich set of applications. the use of a general purpose programming language would yield a large number of applications only differing in the details. Concurrent programming only deals with the synchronization of communication. a protocolling language could easily be imagined.[28] A suitably defined (dedicated) protocolling language would therefore have little syntax. because its virtual machine would have incorporated all possible principles and concepts making the virtual machine itself a universal protocol. hardly differing from this universal networking protocol. High-level protocols with relatively large complexity could however merit the implementation of language interpreters. Instead. so they can be coded with relative ease. the assignment ethernet[mac-address]:=message value could therefore make sense. TCP and IP. the IETF decided to reduce complexity by assuming a relatively simple network architecture allowing decomposition of the single universal networking protocol into two generic protocols. because all networking protocols use the same underlying principles and concepts. the equivalent of a store would be a collection of transmission media. A valid assignment in a protocol (as an analog of programming language) could be Ethernet:='message' . ISO choose a similar but more general path. For instance a mac-address identifies an ether network card on the transmission medium (the 'ether'). development of protocol standards using a consensus model (the agreement of a group of experts) might be a viable way to coordinate protocol design efforts. The notion of a universal networking protocol provides a rationale for standardization of networking protocols. in the same way.[26] By extending the assignment statement of an existing programming language with the semantics described. assuming the existence of a universal networking protocol. In this (networking) context a protocol is a language. and two classes of specific protocols. Protocol design Communicating systems operate in parallel. So. 4 Universal protocols The nice thing about standards is that you have so many to choose from. meaning a message is to be broadcast on the local ethernet. perhaps just enough to specify some parameters or optional modes of operation. The programming tools and techniques for dealing with parallel processes are collectively called concurrent programming. Tanenbaum in Computer Networks [27] Despite their numbers. An example of the latter case is the HTML language. —Andrew S. In our imaginary protocol. Operating systems provide reliable communication and synchronization facilities for communicating objects confined to the same system by means of system libraries. file transfer. instead of using a dedicated protocolling language. Concurrent programming has traditionally been a topic in operating systems theorie texts. message values are transferred using transmission media. The protocolling language would have some syntax and a lot of semantics describing this universal protocol and would therefore in effect be a protocol. A programmer using a general purpose programming language (like C or ADA) can use the routines in the libraries to implement a protocol. one dealing with the low-level network details and one dealing with the high-level details of common network applications (remote login. so a single universal protocol would be very hard to design and implement correctly. On a transmission medium there can be many receivers. instead of a collection of memory locations. networking protocols show little variety. The syntax and semantics of the communication governed by a low-level protocol usually have modest complexity.

Communications protocol seems indispensable. tuple space) can be defined using semaphores. Although both problems only involve two processes.[4] Conceptually. The semaphore is sufficiently elementary to be successfully studied by formal methods. On distributed systems there is no common central memory so the communications are always by means of message passing. The readers-writers problem is a generalization of the mutual exclusion problem.[32] This kind of design can be a bit of a challenge to say the least. The execution sequences are divided into sections. Specifications of concurrent programs can be formulated using formal logics (like CSP) which make it possible to prove properties of the programs. The interleaving scheme makes no timing assumptions other than that no process halts in its critical section and that ready processes are eventually scheduled for execution. Mealy. A concurrent program is correct if it does not violate some safety property such as mutual exclusion or rendezvous of critical sections and does not suffer of liveness properties such as deadlock or lockout. monitors. The basic primitive for synchronization is the semaphore. their solutions require rather complex algorithms (Dekker's algorithm. Incorrectness can be shown using execution scenarios. in particular and in retrospect. because concurrent programs are notorious for the hidden and sophisticated bugs they contain. In this case the processes simply have to wait for each other (synchronization by rendezvous) before exchanging data.and Moore machines are in use as design tools in digital electronics systems. Correctness of the concurrent program can only be shown using a mathematical argument. reentrant mutexes. The goal of the abstraction is to prove correctness of the program assuming the existence of some basic synchronization or data exchange mechanisms provided by the operating system (or other software) or hardware. 5 Concurrent programming A concurrent program is an abstraction of cooperating processes suitable for formal treatment and study. For the Internet protocols. the concurrent program consists of several sequential processes whose execution sequences are interleaved. or the sending/receiving of signals (message passing) using a shared transmission medium.[30] A mathematical approach to the study of concurrency and communication is referred to as Communicating Sequential Processes (CSP). The rendezvous is studied in the producer-consumer problem in which a producer process only produces data if and only if the consumer process is ready to consume the data. semaphores. All other primitives (locks. so it is important to keep things simple. Lamport's bakery algorithm). For correct operation of the program. This is achieved using small code fragments (protocols) at the start and the end of the critical sections. The code fragments determine whether the critical sections of two communicating processes should execute in parallel (rendezvous of processes) or should be executed sequentially (mutual exclusion of processes). The mechanisms are complex. A section manipulating shared resources is called a critical section. so more convenient higher level primitives are implemented with these mechanisms. this meant a basis for protocol design was needed to allow decomposition of protocols into much simpler.[5] Mutual exclusion is extensively studied in the mutual exclusion problem. Most third generation operating systems implement separate processes that use special instructions to ensure only one process can execute the restricted procedures.and Moore machines.[31] Concurrency can also be modelled using finite state machines like Mealy. The dining philosophers problem is a classical problem sufficiently difficult to expose many of the potential pitfalls of newly proposed primitives. used to store data or access-restricted procedures.[34] . cooperating protocols. which we encounter in the form of hardware used in telecommunications or electronic devices in general. message passing.[33] In order to synchronize or exchange data the processes must communicate by means of either a shared memory. The primitives are used to construct the concurrent program. the critical sections of the processes need to be properly sequenced and synchronized.

Also note that software is needed to implement both the 'xfer-mechanism' and a protocol (no protocol. The horizontal protocols are layered protocols and all belong to the protocol suite. A and B. a 'xfer-mechanism' performs communications and a cpu performs computations and the 'framework' introduces something that allows the protocols to be designed independent of one and another by providing separate execution environments for the protocols. complex protocols into simpler. The message flows are governed by rules. transport-.[37] Protocol layering Protocol layering now forms the basis of protocol design. so the analog of a data flow diagram Figure 3. and in communications. a (best-effort) hardware delivery mechanism layer is used to build a connectionless packet delivery system on top of which a reliable transport layer is built. By analogy we could say that the aforementioned 'xfer-mechanism' is comparable to a cpu. The blue lines therefore mark the boundaries of the (horizontal) protocol layers. In computations. The systems both make use of the same protocol suite. and protocols are very often stacked to give tunnelling. a diagram of the message flows in and between two systems.Communications protocol 6 A basis for protocol design Systems do not use a single protocol to handle a transmission. Instead they use a set of cooperating protocols. on top of which is the application software.[10] Together. In literature there are numerous references to the analogies between computer communication and programming. we have protocols and messages. The vertical protocols are not layered because they don't obey the protocol layering principle which states that a layered protocol is designed so that layer n at the destination receives exactly the same object sent by layer n at the source. called a protocol layer. sometimes called a protocol family or protocol suite. the layers make up a layering scheme or model. The Internet protocol suite consists of the following layers: application-. but it is also a functional decomposition.[23] To visualize protocol layering and protocol suites. internet. without worrying about how other layers perform. The delivery system is defined by the IP protocol and the transport system by the TCP protocol. because each protocol belongs to a functional class. is shown in figure 3.and network interface-functions. no communication).[2] The protocol layers each solve a distinct class of communications problems. The vertical flows (and protocols) are in system and the horizontal message flows (and protocols) are between systems. and dataformats specified by protocols. but they have to satisfy some minimal . so some kind of conceptual framework is needed to make this communication possible. we have algorithms and data. Message flows using a protocol suite. Typically.[8] It allows the decomposition of single.[35] [36] Layering Many communications protocols can be viewed as layered protocols where each layer gives a service to the protocol of the layers above and requires service from layers below. Layered protocols allow the protocol designer to concentrate on one layer at a time. Layers below and above these can be defined. Furthermore. would be some kind of message flow diagram. cooperating protocols.[9] To cooperate the protocols have to communicate with each other. it is repeatedly stated that protocols are to computer communication what programming languages are to computation. for example the internet protocol can be tunnelled across an ATM network protocol to provide connectivity by layering the internet protocol on top of the ATM protocol transport layer.[36] The vertical protocols neednot be the same protocols on both systems.

The pieces contain a header area and a data area.data'.Communications protocol assumptions to ensure the protocol layering principle holds for the layered protocols. so one cannot be designed without the other. Datagrams are passed from router to router until a router is reached that can deliver the datagram on a physically attached network (called direct delivery). The path can be an indication that the datagram should be delivered directly or it can be the address of a router known to be closer to the destination. packets.[41] Some of the more important features in this respect of the Internet architecture and the network services it provides are described next. To ensure both sides use the same protocol. The data in the header area identifies the source and the destination on the network of the packet. the protocol. Each computer is identified by an address on the Internet. not an individual computer. The netid identifies a network and the hostid identifies a host. An Internet Address identifies a connection to the network. This rule therefore ensures that the protocol layering principle holds and effectively virtualizes all but the lowest transmission lines. The mapping is called address resolution. data length. This way physical addresses are only used by the protocols of the network interface layer. IP datagrams or network frames depending on the layer in which the pieces are to be transmitted.[42] • Conceptually.[46] Physical networks are interconnected by routers.[48] A special entry can specify that a default router is chosen when there are no known paths. and other data meaningful to the protocol like CRC's of the data to be send. The result is that at the lowest level the piece looks like this: 'Header1.[44] • Network technology independence is achieved using the low-level address resolution protocol (ARP) which is used to map Internet addresses to physical addresses. Routers forward packets between interconnected networks making it possible for hosts to reach hosts on other physical networks.data' and in the layer directly above it: 'Header2. This can be achieved using a technique called Encapsulation. called messages or streams.[43] The netid is used by routers to decide where to send a packet.[38] Usually. • The Internet offers universal interconnection. an Internet addresses consists of a netid and a hostid. All the interconnected physical networks appear to the user as a single large network.Header3.[45] The TCP/IP protocols can make use of almost any underlying communication technology. a table called the IP routing table is consulted.Header2.Header3. a message or a stream of data is divided into small pieces.[49] 7 .[47] To decide whether a datagram is to be delivered directly or is to be send to a router closer to the destination. Message flows in the presence of a router networks. so for this reason some message flows are coloured red in figure 3. and a timestamp. This interconnection scheme is called an internetwork or internet. The term host is misleading in that an individual computer can have multiple network interfaces each having its own Internet address. both on the sending and receiving side. The message flows between two communicating system A and B in the presence of a router R are illustrated in figure 4. which means that any pair of computers connected to the Internet is allowed to communicate.data' and in the top layer: 'Header3.[39] [40] The rule enforced by the vertical protocols is that the pieces for transmission are to be encapsulated in the data area of all lower protocols on the sending side and the reverse is to happen on the receiving side. The design of the protocol layering and the network (or Internet) architecture are interrelated. the pieces also carry data identifying the protocol in their header. The table consists of pairs of networkids and the paths to be taken to reach known Figure 4.

Connectionless delivery means that the messages or streams are divided in pieces that are multiplexed separately on the high speed intermachine connections allowing the connections to be used concurrently. delayed or delivered out of order without notice to the sender or receiver. can make use of TCP.e.25 networks and ATM networks.[58] TCP/IP software is organized in four layers. The delivery of packets is said to be unreliable.[55] Software layering Having established the protocol layering and the protocols. This module reacts by encapsulating the message in its own data area and filling in its header data in accordance with the protocol it implements and interacts with the module below it by handing over this newly formed Figure 5: Protocol and software layering message whenever appropriate. Connection oriented systems build up virtual circuits (paths for exclusive use) between senders and receivers. Once build up the IP datagrams are send as if they were data through the virtual circuits and forwarded (as data) to the IP protocol modules.[50] • A Connectionless packet delivery (or packet-switched) system (or service) is offered by the Internet.[53] • A reliable stream transport service using the unreliable connectionless packet delivery service is defined by the transmission control protocol (TCP). A LAN. can be used on X.[59] 8 . so the message is send across. duplicated. The (top two horizontal) red arrows are virtual. The software modules implementing the protocols are represented by cubes. Each piece carries information identifying the destination. the protocol designer can now resume with the software design. The services are layered as well and the application programs residing in the layer above it. On the receiving system B the reverse happens. To send a message on system A. called tunneling. including best-effort delivery mechanisms like the ethernet. the top module interacts with the module directly below it and hands over the message to be encapsulated. called the application services. a WAN or a point-to-point link between two computers are all considered as one network. a receiving module discards the piece it has received and reports back the error condition to the original source of the piece on the same layer by handing the error message down or in case of the bottom module sending it across. This technique. not on any intermediate routers).[52] It is also possible to use TCP/IP protocols on connection oriented systems.[56] On protocol errors. The blue lines mark the layer boundaries. The bottom module directly interacts with the bottom module of system B. so ultimately (and assuming there were no transmission errors or protocol violations etc.[51] The unreliable connectionless delivery system is defined by the Internet Protocol (IP).Communications protocol • All networks are treated equal. The software has a layered organization and its relationship with protocol layering is visualized in figure 5.[54] Programs wishing to interact with the packet delivery system itself can do so using the user datagram protocol (UDP). The reassembly is done at the destination (i.[57] The division of the message or stream of data into pieces and the subsequent reassembly are handled in the layer that introduced the division/reassembly. The protocol also specifies the routing function.) the message gets delivered in its original form to the topmodule of system B. Unreliability arises only when resources are exhausted or underlying networks fail. which chooses a path over which data will be send. because packets may be lost. The information flow between the modules is represented by arrows. because it adapts well to different hardware.

Furthermore. • Internet layer.[61] 9 . producing relocatable machine code (datagram) that is passed to the loader which fills in the memory locations (ethernet addresses) to produce executeable code (network frame) to be loaded (function of the network interface layer) into physical memory (transmission medium). As a result. the translation software is layered as well. the translation process involves multiple data representations. ICMP error and control messages are handled as well in this layer. Packets to be send are accepted from the transport layer along with an identification of the receiving machine. The application program passes data to the transport layer for delivery. Incoming datagrams are checked for validity and the routing algorithm is used to decide whether the datagram should be processed locally or forwarded. This can be seen in the TCP/IP layering by considering the translation of a pascal program (message) that is compiled (function of the application layer) into an assembler program that is assembled (function of the transport layer) to object code (pieces) that is linked (function of the Internet layer) together with library object code (routing table) by the link editor. The boundary between application layer and transport layer is called the operating system boundary. allowing the software layers to be designed independently. • Network interface layer.[60] The modules below the application layer are generally considered part of the operating system. because each layer's output is passed as input to the next layer. At the highest layer. link editor. Program translation forms a linear sequence. The boundary between network interface layer and Internet layer is called the high-level protocol address boundary. If the datagram is addressed to the local machine. A network interface may consist of a device driver or a complex subsystem that uses its own data link protocol. the datagram header is deleted and the appropriate transport protocol for the packet is chosen. To do so. The stream of data is divided into packets by the module and each packet is passed along with a destination address to the next layer for transmission. the receiving side sends back acknowledgments and the sending side retransmits lost pieces called packets. and loader.Communications protocol • Application layer. The transport layer provides communication from one application to another. the services available across a TCP/IP internet are accessed by application programs. The Internet layer handles the communication between machines. The application chooses the style of transport to be used which can be a sequence of individual messages or a continuous stream of bytes. the terms between parentheses in the previous sentence denote the relevant analogs and the terms written cursively denote data representations. assembler.[3] The network interface layer uses physical addresses and all the other layers only use IP addresses. Program translation has been divided into four subproblems: compiler. To show just how closely the analogy fits. The packets are encapsulated in IP datagrams and the datagram headers are filled. The layer must accept data from many applications concurrently and therefore also includes codes in the packet header to identify the sending and receiving application program. A routing algorithm is used to determine if the datagram should be delivered directly or send to a router. The transport layer may regulate flow of information and provide reliable transport. The designers of the TCP/IP protocol suite were keen on imposing the same layering on the software framework. The datagram is passed to the appropriate network interface for transmission. • Transport layer. We see the same thing happening in protocol software where multiple protocols define the datarepresentations of the data passed between the software modules. Noting that the ways to conquer the complexity of program translation could readily be applied to protocols because of the analogy between programming languages and protocols. Passing data between these modules is much less expensive than passing data between an application program and the transport layer. The network interface layer is responsible for accepting IP datagrams and transmitting them over a specific network. ensuring that data arrives without error and in sequence.

Communications protocol Strict layering Strictly adhering to a layered model. the rules can be expressed by algorithms and datastructures. By describing the software interfaces of the modules on paper and agreeing on the interfaces.[65] 10 Protocol development For communication to take place. This activity is referred to as protocol development. perhaps more important point can be shown by considering the fact that some of the protocols in the Internet Protocol Suite cannot be expressed using the TCP/IP model. Such protocols are referred to as de facto standards. protocols gain market dominance without going through a standardization process. From a historical perspective.e.[64] To improve on the model. Recall that in digital computing systems. In the absence of standardization. at the cost of one or two extra layers. perhaps be split up into two protocols. especially when used to scare away competition. this was deliberately done to discourage users from using equipment from other manufacturers. makes the protocolsoftware operating system independent. There are more than 50 variants of the original bi-sync protocol. a practice known as strict layering. is not always the best approach to networking. proprietary software producers would not accept this. but there is a hidden caveat. It should be noted though that in some cases protocol standards are not sufficient to gain widespread acceptance i. The sourcecode could be considered a protocol specification. By specifying the algorithms on paper and detailing hardware dependencies in an unambiguous way. implementers are free to do it their way. because they are thought to serve an important public interest. creating incompatible versions on their networks. that when adhered to and published. thus inviting competition. In some cases. manufacturers and organizations felt free to 'enhance' the protocol. protocols have to be agreed upon.[6] In some cases. sometimes the sourcecode needs to be disclosed enforced by law or the government in the interest of the public. so getting approval can be very important for the protocol. so there is at least a trade-off between simplicity and performance. an offending protocol could. raising the opportunity of hardware independency. a paper draft is created. Often the members are in control of large market-shares relevant to the protocol and in many cases. this would enforce a source on all parties and for another. It was originally not intended to be used in a multinode network. Such a paper draft can be developed into a protocol standard by getting the approval of a standards organization. This is referred to as source independency. Positive exceptions exist. They can hold a market in a very negative grip. The need for protocol standards The need for protocol standards can be shown by looking at what happened to the bi-sync protocol (BSC) invented by IBM. because the sources are published and maintained in an open way. For one thing. BSC is an early link-level protocol used to connect two separate nodes. can have a serious impact on the performance of the implementation. ensures interoperability between software and hardware. The members of the standards organization agree to adhere to the standard on a voluntary basis. Expressing the algorithms in a portable programming language.[62] Strict layering.[63] Another. standards are enforced by law or the government. however is not suitable for the parties involved. This form of specification. To get the approval the paper draft needs to enter and successfully complete the standardization process. that a standard would have prevented at least some of this from happening. because the model is also used to provide a conceptual view on the suite for the intended users. a 'de facto standard' operating system like GNU/Linux does not have this negative grip on its market. in other words some of the protocols behave in ways not described by the model. One can assume. . De facto standards are common on emerging markets. niche markets. Standardization is therefore not the only solution for open systems interconnection. There is a trade-off to be made here between preciseness for the designer and clarity for the intended user. but doing so revealed several deficiencies of the protocol. or markets that are monopolized (or oligopolized). standardization should be seen as a measure to counteract the ill-effects of de facto standards.

. All conflicting views should be taken into account. reflecting that the members agreed the facility should be provided. The draft proposal is discussed by the member countries' standard bodies and other organizations within each country. Comments and suggestions are collated and national views will be formulated. the International Telecommunications Union (ITU).[66] The standardization process The standardization process starts off with ISO commissioning a sub-committee workgroup. The layers above it are numbered (from one to seven). The World Wide Web Consortium (W3C) produces protocols and standards for Web technologies. This often reflects conflicting views of some of the members. • Various inconsistencies and ambiguities will inevitably be found when implementing the standard. International standards are reissued periodically to handle the deficiencies and reflect changing views on the subject. and the Internet Engineering Task Force (IETF). The ITU is an umbrella organization of telecommunications engineers designing the public switched telephone network (PSTN). to progress to a draft proposal of the working group. and will contain some of the following 'features': • Various optional modes of operation. modification. future-proof framework suitable for structured protocols (such as layered protocols) and their standardization. because protocols also need a framework to operate. Standards organizations also do research and development for standards of the future. communicating systems are assumed to be connected by an underlying physical medium providing a basic (and unspecified) transmission mechanism. The original paper draft created by the designer will differ substantially from the standard. and compromise the proposal reaches the status of a draft international standard. for example to allow for setup of different packet sizes at startup time. The workgroup issues working drafts and discussion documents to interested parties (including other standards bodies) in order to provoke discussion and comments. The IETF maintains the protocols in use on the Internet.[67] Future of standardization (OSI) A lesson learned from ARPANET (the predecessor of the Internet) is that standardization of protocols is not enough. The process normally takes several years to complete. International standards organizations are supposed to be more impartial than local organizations with a national or commercial self-interest to consider. The IEEE controls many software and hardware protocols in the electronics industry for commercial and consumer devices. the standards organizations mentioned. much discussion and usually some disagreement on what the standard should provide and if it can satisfy all needs (usually not). • Parameters reserved for future use. In practice. because the parties could not reach consensus on the optimum packet size. cooperate closely with each other.[68] This gave rise to the ISO Open Systems Interconnection reference model (RM/OSI). This will generate a lot of questions. but could not reach agreement on how this should be done in the available time. • Parameters that are left undefined or allowed to take on values of a defined set at the discretion of the implementor. before the members of ISO vote on the proposal. For marine electronics the NMEA standards are used. the draft proposal has to consider the objections and counter-proposals to create a new draft proposal for another vote. the Institute of Electrical and Electronics Engineers (IEEE). After a lot of feedback. If rejected. and ultimately an international standard. which is used as a framework for the design of standard protocols and services conforming to the various layer specifications.Communications protocol 11 Standards organizations Some of the standards organizations of relevance for communications protocols are the International Organization for Standardization (ISO).[69] In the OSI model. This would prevent protocol standards with overlapping functionality and would allow clear definition of the responsibilities of a protocol at the different levels (layers). as well as many radio communication systems. often by way of compromise. It is therefore important to develop a general-purpose.

any necessary syntax transformations. interaction management so presentation entities can control whose turn it is to perform certain control functions. This layer is also responsible for (network) congestion control. so it is possible to use just one naming domain for all layers.[73] • The transport layer provides reliable and transparent data transfer in a cost effective way as required by the selected quality of service. The exchange of data link units (including flow control) is defined by this layer. the layers and their functionality are (from highest to lowest layer): • The application layer may provide the following services to the application processes: identification of the intended communication partners. which is implemented by using services of the (n-1)-layer. The address naming domains need not be restricted to one layer. the transmission techniques used. Errors occurring in the physical layer are detected and may be corrected. Connection-oriented networks are more suitable for wide area networks and connectionless networks are more suitable for local area networks.[72] • The session layer may provide the following services to the presentation layer: establishment and release of session connections. and service standards defining how a given layer communicates with the layer above it.[74] • The network layer does the setup. When systems are not directly connected. determination of cost and acceptable quality of service. two peer entities at a given layer use a (n)-protocol. agreement on privacy mechanisms for the communication. Using connections to communicate implies some form of session and (virtual) circuits. As a result the IETF developed its own standardization process based on "rough consensus and running code". the IETF had to cope with this and the fact that the Internet needed protocols which simple were not there.[70] For each layer there are two types of standards: protocol standards defining how peer entities at a given layer communicate. agreement on responsibility for error recovery and procedures for ensuring data integrity. It may support the multiplexing of several transport connections on to one network connection or split one transport connection into several network connections.[71] • The presentation layer may provide the following services to the application layer: a request for the establishment of a session. hence the (in the TCP/IP model lacking) session layer. and the setup. Errors are reported to the network layer.g. In the original version of RM/OSI. synchronization between cooperating application processes. including required logon and logoff procedures.[76] • The physical layer describes details like the electrical characteristics of the physical connection. establishment of the necessary authority to communicate. determination of availability and authentication of the partners. which assumes a connectionless network.[79] 12 .[78] At the time. An address uniquely identifies a service access point. formatting and special purpose transformations (e. a quarantine service which allows the sending presentation entity to instruct the receiving session entity not to release data to its presentation entity without permission. data compression and data encryption). character sets and data structures).g.[77] In contrast to the TCP/IP layering scheme. normal and expedited data exchange. To communicate. intermediate peer entities (called relays) are used. routing and relay functions are provided by this layer. maintenance and release of network paths between transport peer entities. Each layer provides service to the layer above it (or at the top to the application process) using the services of the layer immediately below it. The quality of service is negotiated between network and transport entities at the time the connection is set up. resynchronization of a session connection. The layers communicate with each other by means of an interface. The constituent members of ISO were mostly concerned with wide area networks. Corresponding layers at each system are called peer entities. data transfer. selection of the dialogue discipline. maintenance and release of data link connections. negotiation of the syntax to be used between the application layers. maintenance and clearing of physical connections. RM/OSI assumed a connection-oriented network. When relays are needed.[75] • The data link layer does the setup. reporting of unrecoverable exceptions to the presentation entity. so development of RM/OSI concentrated on connection oriented networks and connectionless networks were only mentioned in an addendum to RM/OSI. called a service access point. identification of any constraints on syntax (e.Communications protocol the nth layer is referred to as (n)-layer.

Despite the fact that the underlying assumptions of the layering schemes are different enough to warrant distinguishing the two. connection-oriented protocols and connectionless protocols are used on connection-oriented networks and connectionless networks respectively. A layering scheme combines both function and domain of use. 13 Taxonomies Classification schemes for protocols usually focus on domain of use and function. The dominant layering schemes are the ones proposed by the IETF and by ISO.[81] For an example of this practice see: List of network protocols. it is a common practice to compare the two by relating common protocols to the layers of the two schemes. As an example of domain of use. so the packets can be passed across a transport system using the high-level protocol. RM/OSI has extended its model to include connectionless services and because of this. Common types of protocols The Internet Protocol is used in concert with other protocols within the Internet Protocol Suite. The functionality of the layers has been described in the section on the future of standardization and an overview of protocols using this scheme is given in the article on OSI protocols. The functionality of the layers has been described in the section on software layering and an overview of protocols using this scheme is given in the article on Internet protocols.Communications protocol The standardization process is described by RFC2026 [80]. The layering scheme from ISO is called the OSI model or ISO layering. which is used to encapsulate packets in a high-level protocol. both TCP and IP could be developed into international standards. For an example of function consider a tunneling protocol. Prominent members of which include: • • • • • • • Transmission Control Protocol (TCP) User Datagram Protocol (UDP) Internet Control Message Protocol (ICMP) Hypertext Transfer Protocol (HTTP) Post Office Protocol (POP3) File Transfer Protocol (FTP) Internet Message Access Protocol (IMAP) Other instances of high level interaction protocols are: • • • • • IIOP RMI DCOM DDE SOAP . Nowadays. the IETF has become a standards organization for the protocols in use on the Internet. The layering scheme from the IETF is called Internet layering or TCP/IP layering.

different in detail.Definitions.3. p.Classful Internet Addresses.15. Section 1.Layering concepts and general definitions. Section 2.Sequence control. encapsulation. available online: http:/ / nptel. 11. p. p. p. [6] Marsden 1986. p. p. 1. [3] Comer 2000. 33-34.10 .The need for multiple protocols. p.The Protocol Layering Principle.Internet Architecture And Philosophy. "Each layer takes responsibility for handling one part of the problem. p. explains the advantages of backward error correction. p. 133. [19] Marsden 1986. 187. p. p. Section 3. p. [21] Marsden 1986. "They (protocols) are to communication what programming languages are to computation" [2] Comer 2000. 7. Section 7.The producer-consumer problem.Flow control. [26] Hoare (1985). Sect. [39] Comer 2000. 104. (2003). 177. 11.The Need For Multiple Protocols.10 . explains master/slave and the negotiations to gain control. [4] Ben-Ari 1982. explains similarities protocol software and compiler. p. xiv. com/ books?id=Pd-z64SJRBAC& pg=PA235).The concurrent programming abstraction.7. explains much of the following.Summary.5. deals with communication.1 . 11. [22] Comer 2000.3. [17] Marsden 1986. . introduces the decomposition in layers.3 .Loss of information . Section 3. [41] Comer 2000. 11. states: layering forms the basis for protocol design.3 . 11.timeouts and retries. explains how packets get lost and how sequencing solves this. Srinivasan. chapter 2 . 3.Direction of information flow.Network-Level Interconnection.2 Environments and stores. 27. p. summarizes the concurrent programming abstraction. though of course. 184. [12] Marsden 1986. 11. in/ video. 704. 34-35. Retrieved 22 June 2011. Section 14.The Disadvantage Of Layering. [38] Comer 2000. 181. Computer networks (http:/ / books. xxv. [34] Ben-Ari 1982. using other analogies. Section 3. 55. Sect. 178. [9] Comer 2000. Sect. Describes the transformations of messages or streams that can be observed in the protocol layers.v where c is the name of the communication channel and v is the value of the message. Sect. 96. [29] Ben-Ari 1982.2 .2 . 64-65.3 .The Need For Multiple Protocols. Glossary of Internetworking terms.The Need For Multiple Protocols. .4 . 235. ISBN 9780130661029. In the introduction: a communication is an event described by a pair c. p. Sect. p.Datagram Size. Sect. 192. Glossary of Internetworking Terms and Abbreviations. Network MTU. explains universal interconnection and internetworking.".The Need For Multiple Protocols. p. [31] Hoare 1985.3 .2 . 56. term protocol. p.The Conceptual Layers Of Protocol Software. xiii. 133. Section 4. multiplexing.Ethernet Frame Format. 177. php?courseId=1005& p=3 [33] Ben-Ari 1982. Section 3. p.Internet Services.The TCP/IP 5-Layer Reference Model. iitm. and Fragmentation. explains this by drawing analogies between computer communication and programming languages.4 . states the same about the software organization. p. [25] Tennent 1981.Communication. p.6 .7 . explains layered protocols. states the same. p.1 .The Need For Multiple Protocols. p. 177. Sect. p. states the same. p. states more or less the same. 11. uses BSC as an example to show the need for both standard protocols and a standard framework. [27] Tanenbaum.Foreword To The First Edition By The Late Jon Postel. states the same.4. explains address mapping. in his preface. [5] Ben-Ari 1982.686: term encapsulation. 4 . [14] Comer 2000.11 . explains the advantages of positive only acknowledgement and mentions datagram protocols as exceptions. ac.Acknowledgement. routing. 2.Why are standards necessary?. the semantics of blocks and definitions are described using environments and stores. explains the rendezvous variant in the absence of common memory.1 . Section 6. 35-36." [11] Comer 2000. p. linker. p. google. and naming are quite similar in any protocol suite. [35] Comer 2000. Section 3. Sect. Section 2. Sect. [13] Comer 2000. 192. Sect.7 . p.2 . p. describes service layering. p. NPTEL courses:::: Electronics & Communication Engineering :: Digital Circuits and Systems. Sect. 187. Chapter 4 . Ch.7 .16. Andrew S.3 . in his preface. 3. 18-19. [36] Comer 2000. layering. Chapter 3 . Sect.Functionality Of The Layers. 11. Ethernet frames are used as an example for administrative data for the protocol itself. Prentice Hall Professional. 28-33. 11. addressing and address mapping.The Basic Idea Behind Multiplexing And Demultiplexing. [40] Comer 2000. [23] Comer 2000.2 . Section 3. p. 11.7 . [15] Marsden 1986. [18] Marsden 1986. Sect. 27. [28] Comer 2000. p. p. [32] S. Sect. p. [20] Marsden 1986. 12. 11. [16] Marsden 1986.Fundamental protocol concepts and problem areas. "Protocols are to communication what algorithms are to computation" [24] Tennent 1981. p. defines scope and binding. [42] Comer 2000.3 .4 . Section 2. 30. Sect. p. 36-38. 177. states the same. p. p. Explains fragmentation and the effect on the header of the fragments. [7] Comer 2000.Communications protocol 14 Notes [1] Comer 2000.Detection and transmission errors. 11. 26-42.71. 64-67. Chapter 4 .3 . 178. loader.2 .5 .An overview of the book. p. [8] Sect. [10] Comer 2000. [30] Ben-Ari 1982.Communication. assembler. 177.2 . p. [37] Comer 2000. "The principles of architecture.

p. p.5. 131. follows HDLC to illustrate the process.11 Protocol layering.The session layer. states the same. p. 65. explaining why "strict layering is an imperfect model" [65] IETF 1989. [48] Comer 2000. [49] Comer 2000.Braden (ed.Layering concepts and general definitions. Section 1.9. [54] Comer 2000. explains this.5 . 189. p.3 .Advantages of standardisation. p. ISBN 0-13-018380-6.Error Reporting vs. [63] Comer 2000. p. 8. 2. p. the first two paragraphs describe the sending of a message through successive layers. Routers.7.Operating System Boundary. explains tunneling X. Section 14.3 .9 . 104.Organization. 15. Section 14.Internet Protocols And Standardization. 190.html • Douglas E. Sect.1 .com/articles/article. Sect 1. 7. p. p. Sect. 12. explains this. Sect. [79] Comer 2000. Available online at http://tools. [59] Comer 2000. 8. [58] Comer 2000. p. p. [57] Comer 2000.11. 192. Addison-Wesley 1999. [50] Comer 2000. gives details on the routing table. [53] Comer 2000.Organization.5. Section 6. Sect. Section 14.8 . [70] Marsden 1986. states the same. Sect. [60] Comer 2000. explains functionality of the layers. p.4 .ietf. p.).1 . . 120.5 . p. p. 15. [46] Comer 2000. 191.Some problems with standardisation. Sect 1. 179. Section 14.9. p. explains this. 206. states the same. In particular the introduction gives an overview of the design goals of the suite.Describes TCP/IP to the implementors of protocolsoftware. p. 121.4 .The Original Classful Addressing Scheme.2 .1 . 11.1 . 183-185.The TCP/IP 5-Layer Reference Model. explains the address scheme. [80] http:/ / tools. explains this. 11. Section 14. org/ html/ rfc2026 [81] Comer 2000. Prentice Hall. 194. 11. 195. Sect. 192. Sect. 4. Sect.com/spin/Doc/Book91.1 . [77] Marsden 1986. Sect. IETF (1989): RFC1122.Other Technologies Over Which TCP/IP Has Been Used. explains terminology. 86. 209.Summary. Sect.Indirect Delivery. explains why "strict layering can be extremely inefficient" giving examples of optimizations.8 .5 .High-Level Protocol Boundary. p. Section 6. explains this.Why are standards necessary?. Section 14.9 . p.aspx?p=20482 • Gerard J. Sect. explains ARP.Reassembly Of Fragments. R. [73] Marsden 1986. Sect. p.The TCP/IP 5-Layer Reference Model. ietf. describes the operating system boundary. 12. [61] Comer 2000. p. Error Correction. 46-47. In particular Ch. states the same.3. 15 References • Radia Perlman: Interconnections: Bridges. and Internetworking Protocols. 191. p.11 . Sect. 2nd Edition. 2nd paragraph: many design choices involve creative "breaking" of strict layering. describes the boundary. p.6 . Prentice Hall. p. [76] Marsden 1986. [67] Marsden 1986.). Sect. [55] Comer 2000.1 . [45] Comer 2000.1 . 9. p. 67. Also has a RFC guide and a Glossary of Internetworking Terms and Abbreviations. introduces TCP.Introduction. Sect. netid and routing. p. explains this. [47] Comer 2000.1 .3 .5 . Comer (2000). [44] Comer 2000. 11.Default Routes. 192. explains this. 64. p.Connectionless mode and RM/OSI. states the same. 188. describes the ICMP protocol that is used to handle datagram errors. ISBN 0-13-539925-4. [68] Marsden 1986. Sect. 8. p. Internetworking with TCP/IP . 97.Introduction.The network layer. [69] Marsden 1986.3.10 . Also available online at http://spinroot. p. Section 14.Connectionless Delivery System.Purposes Of The Internet Protocol.org/html/rfc1122. explains lessons learned from ARPANET. Sect.6 . describes reassembly of datagrams.1 . 46. ISBN 0-201-63448-1.10 .The transport layer. explains UDP.11 . Section 6. Protocols and Architecture (4th ed.10 .7 .1 .3 . explains default routing and its use. [78] Marsden 1986.X25NET And Tunnels. 86. 11. Section 14.3 . 2nd paragraph. [71] Marsden 1986. [75] Marsden 1986. 4. 97. 3.13 . Sect. 66-67. Switches. p.The Conceptual Layers Of Protocol Software. explains this. Sect.6 . 13.The presentation layer. explains why the IETF did not use existing protocols.Summary.informit. Sect.Communications protocol [43] Comer 2000. Sect 1. p.The application layer.3. • Internet Engineering Task Force abbr. 1991. Sect. explains the delivery system.Organization. 184. In particular Ch. p. mentions this.3. introduces OSI.All Networks Are Equal. [74] Marsden 1986.4 . states: This layerist organization was chosen for simplicity and clarity. p. 181. [56] Comer 2000. 2. p. Requirements for Internet Hosts -Communication Layers. 7. [52] Comer 2000. p. 5.1 .Addresses Specify Network Connections. 195. 118. 11. which is also available online at http://www. 15.Principles. [62] IETF 1989. [64] IETF 1989. 183. [72] Marsden 1986. Holzmann: Design and Validation of Computer Protocols.Next-Hop Routing. p. 7.The Disadvantage Of Layering.The physical layer. [66] Marsden 1986. p. Section 14.The data link layer. p. 18 on "network design folklore". states the same. 59.25 and mentions ATM. Section 14. [51] Comer 2000.

D.javvin.Communications protocol • M. The Internet Protocol Suite is therefore often referred to as TCP/IP. ISBN 0-86238-106-1. Prentice Hall International. • Brian W Marsden (1986): Communication network protocols 2nd Edition. Prentice Hall International. IP was the connectionless datagram service in the original Transmission Control Program introduced by Vint Cerf and Bob Kahn in 1974.wildpackets. Tanenbaum (1984): Structured computer organization 10th Print. For this purpose. • C.ipcomm. now referred to as Internet Protocol Version 4 (IPv4) is the dominant protocol of the Internet.pdf) Internet Protocol The Internet Protocol (IP) is the principal communications protocol used for relaying datagrams (packets) across an internetwork using the Internet Protocol Suite. Responsible for routing packets across network boundaries. although the successor. Available online via http://www.html) • PDF-Chart showing the Protocols and the OSI reference layer (http://www. Prentice Hall International. ISBN 0-13-709873-1. ISBN 0-13-854605-3.com/protocols/) • Overview of protocols in telecontrol field with OSI Reference Model (http://www. Chartwell Bratt.zframez. ISBN 0-13-153271-5.usingcsp.networkdictionary. • Andrew S. IP defines addressing methods and structures for datagram encapsulation. IP is the primary protocol in the Internet Layer of the Internet Protocol Suite and has the task of delivering datagrams from the source host to the destination host solely based on their addresses. the other being the connection-oriented Transmission Control Protocol (TCP).com/protocolsuite.com/elements/misc/ WP_encapsulation_chart.com/protocolsuite. it is the primary protocol that establishes the Internet.A. growing deployment worldwide. Internet Protocol Version 6 (IPv6) is in active. Tennent (1981): Principles of programming languages 10th Print. Internet Protocol Suite in operation between two hosts connected via two routers and the corresponding layers used at each hop . Historically. ISBN 0-13-701078-8.com • R. 16 External links • Javvin's Protocol Dictionary (http://www. html) • List of Data Communication Protocols (http://www.html) • Network Protocol Index (http://www. Hoare (1985): Communicating sequential processes 10th Print. Prentice Hall International.R.de/protocols_en. Ben-Ari (1982): Principles of concurrent programming 10th Print. The first major version of IP.

in contrast to so-called connection-oriented modes of transmission. it is possible that some packets are routed on a longer path to their destination. resulting in improper sequencing at the receiver. In network architectural language it is a connection-less protocol. IP does provide re-ordering of fragments delivered out-of-order. packet loss and duplication. One of the technical constraints is the size of data packets allowed on a given link.[1] Transmission Control Protocol (TCP) is an example of a protocol that will adjust its segment size to be smaller than the MTU. to ensure in-order delivery the upper layer may have to cache data until it can be passed to the application. For the benefit of reducing network complexity. has abandoned the use of IP header checksums for the benefit of rapid forwarding through routing elements in the network. although a facility exists in the Internet Control Message Protocol (ICMP) to do so. Since routing is dynamic for every packet and the network maintains no state of the path of prior packets. The lack of reliability permits various error conditions. cf. end-to-end principle. As a consequence of this design. The IPv4 internetworking layer has the capability to automatically fragment the original datagram into smaller units for transmission. For this purpose the Internet Protocol defines an addressing system that has two functions. For example. No central monitoring or performance measurement facility exists that tracks or maintains the state of the network. Each packet is tagged with a header that contains the meta-data for the purpose of delivery. User Datagram Protocol (UDP) and Internet Control Message Protocol (ICMP) disregard MTU size thereby forcing IP to fragment oversized datagrams. the intelligence in the network is purposely mostly located in the end nodes of each data transmission. Some of this responsibility lies also in the upper layer protocols between application and IP. Reliability The design principles of the Internet protocols assume that the network infrastructure is inherently unreliable at any single network element or transmission medium and that it is dynamic in terms of availability of links and nodes. IPv6. The only assistance that the Internet Protocol provides in Version 4 (IPv4) is to ensure that the IP packet header is error-free through computation of a checksum at the routing nodes. even if the path is available and reliable. or suitable for. In this case no notification is required to be sent to either end node. performing the data transmission requested. Routers in the transmission path simply forward packets to the next known local gateway matching the routing prefix for the destination address.Internet Protocol 17 Function The Internet Protocol is responsible for addressing hosts and routing datagrams (packets) from a source host to the destination host across one or more IP networks. The resolution or correction of any of these reliability issues is the responsibility of an upper layer protocol. as well as out-of-order packet delivery. on the other hand. In addition to issues of reliability. An application must assure that it uses proper transmission characteristics. In this case. Addresses identify hosts and provide a logical location service. This has the side-effect of discarding packets with bad headers on the spot. this dynamic nature and the diversity of the Internet and its components provide no guarantee that any particular path is actually capable of.[2] . Facilities exist to examine the maximum transmission unit (MTU) size of the local link. This process of tagging is also called encapsulation. as well as for the entire projected path to the destination when using IPv6. the Internet Protocol only provides best effort delivery and its service is characterized as unreliable. such data corruption.

4 billion. Version number 5 was used by the Internet Stream Protocol.4 × 1038 addresses). Version history In May 1974. Vint Cerf and Bob Kahn. described an internetworking protocol for sharing resources using packet-switching among the nodes.[5] Version numbers 0 through 3 were development versions of IPv4 used between 1977 and 1979. all United States government systems have demonstrated basic infrastructure support for IPv6 (if only at the backbone level). A humorous Request for Comments that made an IPv9 protocol center of its storyline was published on April 1. RFC 1347).[6] It was intended as an April Fool's Day joke. Addressing refers to how end hosts become assigned IP addresses and how subnetworks of IP host addresses are divided and grouped together.[3] These switches primarily use IP addresses but also support MAC addresses for compatibility with older layer 2 Ethernet switches. 340 undecillion. For example. IPv4 is described in RFC 791 (1981). an experimental streaming protocol. subsequently standardized as IPv6. Version numbers 6 through 9 were proposed for various protocol models designed to replace IPv4: SIPP (Simple Internet Protocol Plus. or 3.3 × 109. The Internet Protocol is one of the determining elements that define the Internet. which typically use either interior gateway protocols (IGPs) or external gateway protocols (EGPs) to help make IP datagram forwarding decisions across IP connected networks. The monolithic Transmission Control Program was later divided into a modular architecture consisting of the Transmission Control Protocol at the connection-oriented layer and the Internet Protocol at the internetworking (datagram) layer. The model became known informally as TCP/IP.Internet Protocol 18 IP addressing and routing Perhaps the most complex aspects of IP are IP addressing and routing. as of June 2008. though these came with little or no support from the wider industry and academia. The successor to IPv4 is IPv6. Ethernet switches sold today support IP multicast. with number 4 assigned as the formal protocol version number carried in every IP datagram. Version number 6 was eventually chosen as the official assignment for the successor Internet protocol. IPv4 uses 32-bit addresses (c. addresses) while IPv6 uses 128-bit addresses (c."[4] The paper's authors. IP routing is performed by all hosts. A central control component of this model was the "Transmission Control Program" (TCP) that incorporated both connection-oriented links and datagram services between hosts. known now as IPv6). but most importantly by internetwork routers. Other protocol proposals named "IPv9" and "IPv8" have also briefly surfaced. 1994 by the IETF. or 4. Although adoption of IPv6 has been slow.[7] . The dominant internetworking protocol in the Internet Layer in use today is IPv4. IP routing is also common in local networks. PIP (RFC 1621) and TUBA (TCP and UDP with Bigger Addresses. Its most prominent modification from version 4 is the addressing system. TP/IX (RFC 1475). the Institute of Electrical and Electronic Engineers (IEEE) published a paper entitled "A Protocol for Packet Network Interconnection. although formally referenced as the Internet Protocol Suite.

com RFC 1606: A Historical Perspective On The Usage Of IP Version 9.pdf) • IPv6. gcn. uk/ 2004/ 07/ 06/ ipv9_hype_dismissed/ ) Security Assessment of the Internet Protocol (IP) (http:/ / www.tuwien. cpni.org/Computers/Internet/Protocols/) at the Open Directory Project • RFC 791 • Data Communication Lectures of Manfred Lindner . Theregister. April 1. pdf) Security Assessment of the Internet Protocol version 4 (IPv4) (http:/ / tools.ac. Vol.Part IPv6 (http://www. 5. gov.com (http:/ / www. New Riders Publishing. com/ print/ 25_16/ 41051-1.dmoz. Cerf. Karanjit. html). 1997. uk/ Docs/ InternetProtocol. 637-648 CIO council adds to IPv6 transition primer (http:/ / www. gcn.[8] and is currently being pursued within the IETF. 22.081/infobase/L30-IP_Technology_Basics_v4-6. "A Protocol for Packet Network Intercommunication". May 1974 pp.pdf) • Data Communication Lectures of Manfred Lindner . No. ISBN 1-56205-714-6 Basic Journey of a Packet (http:/ / www.at/lva/384.com .tuwien.ict. com/ infocus/ 1870) Netgear ProSafe XSM7224S reference manual Vinton G. org/ html/ draft-ietf-opsec-ip-security) External links • Internet Protocol (http://www.081/ infobase/L80-IPv6_v4-6.Internet Protocol 19 Reference diagrams Sample encapsulation of application data from UDP to a Link protocol frame Vulnerabilities The Internet Protocol is vulnerable to a variety of attacks.tuwien. Kahn. co.Part IP Technology Basics (http://www.ict.ict. was published in 2008.081/infobase/L31-IP_Technology_Details_v4-7.at/ lva/384. Inside TCP/IP. ietf. along with proposed mitigations.Knowledge Center for Next Generation Internet IPv6 (http://www. securityfocus. A thorough vulnerability assessment. IEEE Transactions on Communications. theregister.at/ lva/384.pdf) • Data Communication Lectures of Manfred Lindner .ac. Robert E.com) .Part IP Technology Details (http://www.ac.[9] References [1] [2] [3] [4] [5] [6] [7] [8] [9] Siyan.ipv6. 1994.

History The Internet protocol suite resulted from research and development conducted by the Defense Advanced Research Projects Agency (DARPA) in the early 1970s. In the spring of 1973. with important influences on this design. which were the first networking protocols defined in this standard. Host-to-host communication tasks are handled in the transport layer. Vinton Cerf. From the lowest to the highest communication layer. In 1972. joined Kahn to work on open-architecture interconnection models with the goal of designing the next protocol generation for the ARPANET. This layer handles application-based interaction on a process-to-process level between communicating Internet hosts. the hosts became responsible. and recognized the value of being able to communicate across both. together with the advent of the World Wide Web in the early 1990s. the developer of the existing ARPANET Network Control Program (NCP) protocol. as in the ARPANET. . Modern IP networking represents a synthesis of several developments that began to evolve in the 1960s and 1970s. site of the first three-way internetworked transmission. the internet layer. instead of the network being responsible for reliability. reflected loosely in the layer names. By the summer of 1973. these are the link layer. As such. which provides a general application-agnostic framework to transmit data between hosts using protocols like the Transmission Control Protocol and the User Datagram Protocol (UDP). After initiating the pioneering ARPANET in 1969. the highest-level application layer contains all protocols that are defined each specifically for the functioning of the vast array of data communications services. and. Kahn joined the DARPA Information Processing Technology Office. Cerf credits Hubert Zimmerman and Louis Pouzin. the transport layer. DARPA started work on a number of other data transmission technologies. where he worked on both satellite packet networks and ground-based radio packet networks. Finally. The internet layer facilitates the interconnection of local networks. this layer establishes the Internet. Diagram of the first internetworked connection The network's design included the recognition it should provide only the functions of efficiently transmitting and routing traffic between end nodes and that all other intelligence should be located at the edge of the network. where the differences between network protocols were hidden by using a common internetwork protocol. The link layer contains communication technologies for the local network to which the host is connected directly by hardware components.Internet protocol suite 20 Internet protocol suite The Internet protocol suite is the set of communications protocols used for the Internet and other similar networks. Robert E. and the application layer. Kahn and Cerf had worked out a fundamental reformulation. designer of the CYCLADES network. Using a simple A Stanford Research Institute packet radio van. in the end nodes. which emerged during the 1980s.[1] [2] The layers define the operational scope or reach of the protocols in each layer. It is commonly known as TCP/IP from its most important protocols: Transmission Control Protocol (TCP) and Internet Protocol (IP). Each layer has functionality that solves a set of problems in its scope. The Internet protocol suite classifies its methods and protocols into four hierarchical abstraction layers. namely the precursors of the Internet and local area networks.

It forwards packets back and forth between them. It was attended from the beginning by large corporations such as IBM and DEC. Known as BSD. Xerox' XNS and all other protocols disappeared from the commercial landscape. For example. Jay Elinsky and Oleg Vishnepolsky of IBM Research wrote VM/CMS and OS/2 TCP/IP stacks. In November. Back then most of these TCP/IP stacks were written single-handedly by very few talented programmers. resulting in the first TCP specification. OSI. 1977. many companies marketed and sold TCP/IP software for Windows. but the term was changed to avoid confusion with other types of gateways. thereby solving Kahn's initial problem. Four versions were developed: TCP v1.[5] In March 1982. DARPA then contracted with BBN Technologies. One popular expression is that TCP/IP.Internet protocol suite design. The migration of the ARPANET to TCP/IP was officially completed on flag day January 1. UK. the Internet Architecture Board held a three day workshop on TCP/IP for the computer industry. a two-network TCP/IP communications test was performed between Stanford and University College London (UCL). Several other TCP/IP prototypes were developed at multiple research centers between 1978 and 1983. TCP v2. In IBM the TCP/IP development was undertaken from 1984 onward in the group of Barry Appelman who later moved to AOL to be the head of all AOL's development efforts. [citation needed] IBM. and Norway. respectively[10] . ATT and DEC were the first major corporations to adopt TCP/IP despite having competing internal protocols (SNA. which produced the PARC Universal Packet protocol suite. XNS. The first VM/CMS TCP/IP stack came from university of Wisconsin[8] . and every year from 1985 through 1993 the number of attendees tripled. Wollongong [7]. Until Microsoft released TCP/IP stack in 1995 with its Windows 95.[6] In 1985. The idea was worked out in more detailed form by Cerf's networking research group at Stanford in the 1973–74 period. At IBM Barry Appelman with a handful of developers was able to maneuver the corporate politics to get a stream of TCP/IP products for various IBM systems .MVS. It was founded by Dan Lynch. At the same time several smaller companies began offering TCP/IP stacks for DOS and MS Windows. IBM included. this code base was adopted by a variety of vendors. is provided with an interface to each network. a three-network TCP/IP test was conducted between sites in the US. when the new protocols were permanently activated. focused on network interoperability via further adoption of TCP/IP. etc) and politics. IBM's SNA." A computer. and then stability with TCP/IP v4 — the standard protocol still in use on the Internet today. The spread of TCP/IP was fueled further in June 1989 when AT&T agreed to put the TCP/IP code developed for UNIX system at University of Berkeley into public domain. OS/2 among others. the eventual product of Cerf and Kahn's work. 1983. VM. will run over "two tin cans and a string. [citations needed] 21 . the dominance of TCP/IP over other protocols became insurmountable. the US Department of Defense declared TCP/IP as the standard for all military computer networking. irrespective of their local characteristics. it became possible to connect almost any network to the ARPANET. for their own TCP/IP stacks. John Romkey's PC/IP implementation was the first IBM PC TCP/IP stack. was also a significant technical influence. promoting the protocol and leading to its increasing commercial use. John Romkey of FTP Software was the author of the MIT PC/IP package as well as at a later point of FTP Software[9] . a split into TCP v3 and IP v3 in the spring of 1978. but when it finally came.[4] The early networking work at Xerox PARC. Microsoft's native NetBIOS. In 1985 the first Interop conference was held. Windows 95 support for TCP/IP came a little late in the Internet evolution. one of early Internet activists. In 1975. Stanford University.[3] Originally a router was called gateway. called a router. attended by 250 vendor representatives. and the University College London to develop operational versions of the protocol on different hardware platforms. much of which existed around the same period of time. such as FTP Software. Interoperability conferences have been held since then every year.

is Internet Protocol version 4 (IPv4). and it essentially establishes the Internet. the internet layer.Internet protocol suite 22 Layers in the Internet protocol suite The concept of layers The Internet protocol suite uses encapsulation to provide abstraction of protocols and services. an application (the highest level of the model) uses a set of protocols to send its data down the layers. The role of layering in TCP/IP may be illustrated by an example network scenario (right-hand diagram). Encapsulation is usually aligned with the division of the protocol suite into layers of general functionality. the interworking of different IP networks. The transport layer operates on top of the internet layer. The Internet Protocol is the principal component of the internet layer. This limitation was eliminated by the standardization of Internet Protocol version 6 (IPv6) in 1998. every other detail of the communication is hidden from each process. The flow of user data through the layers used at each hop is shown. were directly connected to each other by some kind of data pipe. With this functionality. This model was not intended to be a rigid reference model into which new protocols have to fit in order to be accepted as a standard. and beginning production implementations in approximately 2006. The layer simply establishes a basic data channel that an application uses in its task-specific data exchange. meaning it handles the details of data transmission that are independent of the structure of user data and the logistics of exchanging information for any particular specific purpose. The underlying mechanisms that transmit data between the host computers are located in the lower protocol layers. and it defines two addressing systems to identify network hosts computers. The application on each host Instantiations of the TCP/IP stack operating on two hosts each executes read and write operations as if the processes connected to its router on the Internet. the internet layer makes possible internetworking. these port numbers have been standardized so that client computers may address specific services of a server computer without the involvement of service announcements or directory services. the transport layer. and the link layer. For this purpose the layer establishes the concept of the port. . being further encapsulated at each level. The original address system of the ARPANET and its successor. the Internet. a numbered logical construct allocated specifically for each of the communication channels an application needs. The transport layer establishes host-to-host connectivity. In general. For many types of services. The internet layer is not only agnostic of application data structures as the transport layer. According to RFC 1122. in which two Internet host computers communicate across local network boundaries constituted by their internetworking routers. and to locate them on the network. the Internet protocol suite organizes the functional groups of protocols and methods into four layers. It only provides an unreliable datagram transmission facility between hosts located on potentially different IP networks by forwarding the transport layer datagrams to an appropriate next-hop router for further relaying to its destination. the application layer. but it also does not distinguish between operation of the various transport layer protocols. It uses a 32-bit IP address and is therefore capable of identifying approximately four billion hosts.

such as network interface layer. in view to avoid confusion with the similarly named OSI model layer 2 (data link layer) of the seven layer OSI model. As the user data. or if a hardware layer is assumed below the link layer.[22] Different authors have interpreted the RFCs differently regarding the question whether the link layer (and the TCP/IP model) covers OSI model layer 1 (physical layer) issues. It comprises the tasks of specific networking requirements on the local link.[14] [15] [17] in which hardware issues are emphasized.Internet protocol suite The lowest layer in the Internet protocol suite is the link layer. those are often discussed in terms of OSI model layer 1 (physical layer) and OSI model layer 2 (data link layer). A receiving host reverses the encapsulation at each layer by extracting the higher level data and passing it up the stack to the receiving process. RFC 1122 [11] Four layers [18] "Internet model" Tanenbaum Cisco [12] Academy Four layers Kurose [13] Forouzan [14] [15] Comer [16] Kozierok Four+one layers Stallings [17] Arpanet Reference Model 1982 (RFC 871) Three layers Four layers [19] Five layers Five layers "TCP/IP reference [20] model" Application "Internet model" "Five-layer Internet model" or "TCP/IP protocol suite" Application "TCP/IP 5-layer reference model" "TCP/IP model" "Arpanet reference model" Application [18] [21] Transport [18] Internet Link [18] Application Application Application Application/Process Transport Transport Transport Transport Host-to-host or transport Internet Host-to-host Internet Host-to-network Internetwork Network interface Network Data link Internet [18] Data link (Network Network access interface) (Hardware) Physical Network interface Physical These textbooks are secondary sources that may contravene the intent of RFC 1122 and other IETF primary sources. In literature with a bottom-up approach to Internet communication. The mapping often results in a model with five layers where the link layer is split into an OSI model layer 2 (data link layer) on top of an OSI model layer 1 (physical layer). is passed through the descending layers of the protocol stack each layer adds encapsulation information as illustrated in the diagram (right). Some authors have tried to use other names for the link layer. the network segment that a hosts network interface is connected to. first manipulated and structured in the application layer. Others have attempted to map the Internet Protocol model onto the OSI model. . 23 Layer names and number of layers in the literature Successive encapsulation of application data descending through the protocol stack The following table shows the layer names before transmission on the local network link and the number of layers of networking models presented in RFCs and textbooks in widespread use in today's university computer networking courses. This involves interacting with the hardware-specific functions of network interfaces and specific transmission technologies.

including all consumer-targeted systems. html) [14] Behrouz A. is mapped to OSI layer 4 (transport layer). The IETF makes no effort to follow the OSI model although RFCs sometimes refer to it. "To communicate using the Internet system. isti. cnr.Internet protocol suite The internet layer is usually directly mapped into OSI model layer 3 (network layer). F.[22] 24 Implementations Most computer operating systems in use today. Braden (ed. UDP. weblab. it/ education/ ssfs/ lezioni/ slides/ archives/ cern. microsoft. Other protocols could be easily added later (often they can be implemented entirely in the userspace). edu/ ~rh120/ other/ tcpdigest_paper. a host must implement the layered set of protocols comprising the Internet protocol suite. sometimes also including aspects of OSI layer 5 (session layer) functionality.). Data Communications and Networking (http:/ / books. Network Fundamentals: CCNA Exploration Companion Guide. The transport layer of the TCP/IP model. romkey. [7] http:/ / support. (December 1974) Internet History (http:/ / www. as it limits usage of the whole implementation. October 1989 RFC 1123. References [1] [2] [3] [4] [5] [6] RFC 1122. Dye. OSI layer 7 (application layer). htm) [9] (http:/ / www. Specification of Internet Transmission Control Protocol. include a TCP/IP implementation. p. Most of the IP implementations are accessible to the programmers using socket abstraction (usable also with other protocols) and proper API for most of the operations. Computer Networking: A Top-Down Approach. Unique implementations include Lightweight TCP/IP. com/ kb/ 108007 [8] (http:/ / www. and the remaining functionality of OSI layer 5 (session layer) are collapsed into TCP/IP's application layer. "From the ARPANET to the Internet" (http:/ / www. for example DNS for resolving domain names to IP addresses or DHCP client for automatic configuration of network interfaces. Braden (ed. sometimes also described as the host-to-host layer. but it is rarely done. a more general concept of network functionality. This interface is known as BSD sockets and was used initially in C. google. Requirements for Internet Hosts – Communication Layers. IPv6.). V.0321497708. R. TCP and sometimes IGMP. Ross.. ISBN 1-58713-208-7 [13] James F. txt). The IETF has repeatedly stated that Internet protocol and architecture development is not intended to be compliant to the OSI model. R. It is in principle possible to support only one of transport protocols (i. ICMP. The argument is that these OSI layers do not usually exist as separate processes and protocols in Internet applications. 2007. com/ about/ ) [10] Barry Appelman [11] IETF. Requirements for Internet Hosts – Application and Support. com/ educator/ academic/ product/ 0. Requirements for IP Version 4 Routers. addressing Internet architecture. Keith W. Mark A. com/ books?id=U3Gcf65Pu9IC& printsec=frontcover& dq=forouzan+ "computer+ networks"#PPA29. htm) Ronda Hauben. the Internet protocol stack has never been altered by the Internet Engineering Task Force from the four layers defined in RFC 1122. a stack and associated protocols for amateur packet radio systems and personal computers connected via serial lines. October 1989 RFC 1812. livinginternet.7." [12] Mark Dye. Forouzan. com/ i/ ii. A host typically must implement at least one protocol from each layer. Retrieved 2007-07-05. 2008. contains a section entitled: "Layering Considered Harmful". and IGMP (IGMPv6) have some additional required functions. Minimally acceptable implementation includes implementation for (from most essential to the less essential) IP. Baker (June 1995) RFC 675. However.00+ en-USS_01DBC. RFC 3439. pearsonhighered. Kurose. and often is accompanied with integrated IPSec security layer.e. Cerf et al. an open source stack designed for embedded systems and KA9Q NOS. simple UDP). and ICMP (ICMPv6).M1) . TCP Digest (UUCP). ARP. columbia. RFC 1122. ISBN 0-321-49770-8 (http:/ / www. Wendell. . OSI layer 6 (presentation layer). beyond own version of ARP (NBP).

doi:10.pdf). com/ books?id=Pd-z64SJRBAC& pg=PA42& vq=internet+ layer& dq=networks#PPA42.. "The protocol layers [. Computer Networks.M1) [17] William Stallings. ISBN 1-59059-345-6 W. Lee.edu/~rh120/other/tcpdigest_paper. Richard Stevens.]:[. Microsoft Windows Server 2003 TCP/IP Protocols and Services. NNTP.3.org/rfc/rfc0675.eventhelix. 41. Retrieved 2011-10-16. p. . Protocols and Architecture.] Link Layer" [19] Andrew Tanenbaum. Volume 1: The Protocols.. ISBN 0-201-63495-3 • Andrew S.com/network/tcpip. TCP/IP Illustrated. RFC 1122. TCP/IP Illustrated. Meyer (December 2002). (2003).html) . Pearson Prentice Hall 2005. ISBN 0-13-975483-0." [22] R.tcpipguide. Windows(R) 2000 TCP/IP Black Book. ISBN 0-13-187671-6 (http:/ / books.txt) • TCP/IP Sequence Diagrams (http://www. Computer Networks. ISBN 0130661023.1 Network Programming.1145/52324. Internet Engineering Task Force. p. • • • • • • Craig Hunt TCP/IP Network Administration.8..] Transport Layer [. Vinton Cerf.columbia. Volume 3: TCP for Transactions. google. google. (2002).livinginternet. com/ books?id=Pm4RgYV2w4YC& pg=PA131& dq="TCP/ IP+ model+ layers"#PPA129. TCP/IP Protocol Suite (2nd ed.4. com/ books?id=jonyuTASbWAC& pg=PA155& cad=0_0#PPA161. ISBN 86-7991-142-9 • Joseph G. SIGCOMM '88 Symposium proceedings on Communications architectures and protocols (ACM): 106–114. ISBN 0-201-63354-X • W. and the UNIX Domain Protocols. Tanenbaum.. D. Davies and Thomas F.." [20] Tanenbaum. google. (1988).52336.com/articles/internet/internet-practice-4. RFC 1122. Comer.cs. ISBN 0-07-246060-1. Ian McLean. Data and Computer Communications. TCP/IP Illustrated. Wright. Protocols and Architecture. D. Richard Stevens and Gary R. Comer. Computer Networks. McGraw-Hill..NET 1.. Volume 2: The Implementation. Prentice Hall.. "[. Prentice Hall 2006.html) • The TCP/IP Guide (http://www.] the OSI model has seven layers and the TCP/IP has four layers...itprc.com/RealtimeMantra/Networking/) • The Internet in Practice (http://www.M1) [16] Charles M.com/i/ii.Specification of Internet Transmission Control Program. (1999).princeton.com/tcpipfaq/) • TCP/IP Resources List (http://www.7-8.private.] are as follows [.com/TCPIP_State_Transition_Diagram.2 The TCP/IP Reference Model" Excerpt at Google Books (http:/ / books. Internetworking with TCP/IP: Principles. ISBN 1-57610-687-X Ajit Mungale Pro ... Richard Stevens. December 1974 Version • TCP/IP State Transition Diagram (http://www. Andrew S..searchandgo.il/tcpip_rl. ISBN 0-13-066102-3 • Clark. • RFC 675 (http://www. "The application layer is the top layer of the Internet protocol suite. External links • Internet History (http://www. and TCP/IP (reviewed by Cerf and Kahn). Prentice Hall. ISBN 0-13-243310-9 (http:/ / books. Bush.txt) . "The TCP/IP Guide". edu/ in-notes/ rfc3439. google. IP Fundamentals.night-ray. Internetworking with TCP/IP .php) • TCP/IP .Directory & Informational Resource (http://softtechinfo. "The Design Philosophy of the DARPA Internet Protocols" (http://www.] Application Layer [. Behrouz A. O'Reilly (1998) ISBN 1-56592-322-7 Maufer.. "1. ISBN 0-7356-1291-9 • Forouzan. Kozierok.).M1) [21] IETF.Pages on Robert Kahn. isi.Internet protocol suite [15] Douglas E.A comprehensive look at the protocols and the procedures/processes involved • A Study of the ARPANET TCP/IP Digest (http://www.edu/ ~jrex/teaching/spring2005/reading/clark88. HTTP. com/ books?id=c_AWmhkovR0C& pg=PA35& dq="internet+ layer"+ "network+ access+ layer") [18] IETF. Thomas A. p.ietf.pdf) (PDF) • RFC 1180 A TCP/IP Tutorial .Principles.com/free/) .org.htm) -. Some Internet Architectural Guidelines and Philosophy (http:/ / www.] Internet Layer [. No Starch Press 2005 (http:/ / books.4.from the Internet Engineering Task Force (January 1991) • TCP/IP FAQ (http://www. section 1. txt). Retrieved 2007-11-20 25 Further reading • Douglas E. ISBN 0-201-63346-9 W.

ipprimer.com/2007/10/ dns-tcpip-commands-from-command-prompt. conversational style • Introduction to TCP/IP (http://www.net/projects/cipsuite/) — Robust TCP/IP stack for embedded devices without an Operating System 26 .Internet protocol suite • Daryl's TCP/IP Primer (http://www.linux-tutorial.com) .html) • cIPS (http://sourceforge.Intro to TCP/IP LAN administration.webgk.info/MContent-142) • TCP/IP commands from command prompt (http://blog.

. specifically the United Kingdom. The Internet Engineering Task Force (IETF) has also defined level 2. 7 SS7 protocol suite OSI layer SS7 protocols Application INAP. SS6 and SS7 implement out-of-band signaling protocols. Network Data link Physical MTP Level 3 + SCCP MTP Level 2 MTP Level 1 Signalling System No. This led to security problems with blue boxes. 5 (SS5). There is only one international SS7 protocol defined by ITU-T in its Q. Each national variant has its own unique characteristics.. Most national variants are based on two widely deployed national variants as standardized by ANSI and ETSI. an abbreviation for Common Channel Signalling System 7. in which the call-setup information was sent by playing special multi-frequency tones into the telephone lines. In North America it is often referred to as CCSS7. 7 was defined as an international standard by ITU-T in its 1980 (Yellow Book) Q. known as bearer channels in the parlance of the telecom industry. . The main purpose is to set up and tear down telephone calls.[2] Signalling System No. CAP. This suite of protocols is called SIGTRAN. which are in turn based on the international protocol defined by ITU-T. many national variants of the SS7 protocols.[2] SS7 has substantially replaced SS6. 7 or Signalling System #7.[1] There are however. it is sometimes called C7 (CCITT number 7) and is also known as number 7 and CCIS7 (Common Channel Interoffice Signaling 7). 7 (SS7) is a set of telephony signaling protocols which are used to set up most of the world's public switched telephone network telephone calls. 7 27 Signalling System No.[3] explicitly keep the end-user's audio path—the so-called speech path—separate from . with the exception that R1 and R2 variants are still used in numerous nations. ISUP. Signalling System No. prepaid billing mechanisms.. short message service (SMS). 6 (SS6) in 1977. 3. In some European countries. MAP. local number portability. Some national variants with rather striking characteristics are the China (PRC) and Japan (TTC) national variants..Signalling System No. Other uses include number translation. and a variety of other mass market services. It is usually referenced as Signalling System No. IS-41. and 4 protocols that are compatible with SS7: • MTP2 (M2UA and M2PA) • MTP3 (M3UA) • Signalling Connection Control Part (SCCP) (SUA) but use a Stream Control Transmission Protocol (SCTP) transport mechanism. TCAP. SS5 and earlier systems used in-band signaling. History Common Channel Signaling protocols have been developed by major telephone companies and the ITU-T since 1975 and the first international Common Channel Signaling protocol was defined by the ITU-T as Signalling System No.[1] SS7 was designed to replace SS6.7XX-series recommendations. which had a restricted 28-bit signal unit that was both limited in function and not amenable to digital systems. carried in a separate signaling channel.700-series recommendations. or simply abbreviated to SS7. R1 and R2.

can communicate significant amounts of information when setting up a call. and service logic. call screening. This permits rich call-related services to be developed. A.and B-bit signaling). When the signaling is performed on the same circuit that will ultimately carry the conversation of the call. Usually only the dialed digits were signaled. Some of the first such services were call management related services that many take for granted today: call forwarding (busy and no answer). SS7 signaling is termed Common Channel Signaling (CCS) in that the path and facility used by the signaling is separate and distinct from the telecommunications channels that will ultimately carry the telephone conversation.[5] An example of this control information is the digits dialed by the caller. (North America never accomplished full upgrade to the ISDN and the predominant telephone service is still the older POTS. conference calling. Because of the mechanisms used by signaling methods prior to SS7 (battery reversal. it becomes possible to exchange signaling without first seizing a facility. being a high-speed and high-performance packet-based communications protocol. when used in telephony. Because SS7 signaling does not require seizure of a channel for a conversation prior to the exchange of control information. the caller's billing number. While running on a transport based upon IP. This is the case for earlier analogue trunks. but simply transport existing national and international variants of SS7. This required a separate channel dedicated solely to signaling. these older methods could not communicate much signaling information. Another characteristic of ISUP made .[7] The Telephone User Part (TUP) was adopted in Europe and the Integrated Services Digital Network (ISDN) User Part (ISUP) adapted for public switched telephone network (PSTN) calls was adopted in North America. It also permits the subscriber increased mobility due to the decoupling of service logic from the subscription switch. MF and R2 digital trunks. The common channel signaling paradigm was translated to IP via the SIGTRAN protocols as defined by the IETF. dialed digits and charge number digits were outpulsed. multi-frequency digit outpulsing. refers to the exchange of control information associated with the establishment of a telephone call on a telecommunications circuit. during the call.) Due to its richness and the need for an out-of-band channel for its operation. maintenance. and the number of available channels was rapidly increasing anyway at the time SS7 was implemented. busy callback. but permits service logic to be distributed throughout the telephone network and executed more expediently at originating switches far in advance of call routing. and DSS1/DASS PBX trunks.[6] The earliest deployed upper layer protocols in the SS7 signaling suite were dedicated to the setup. it is termed channel associated signaling (CAS). SS7 signaling is mostly used for signaling between telephone switches and not for signaling between local exchanges and customer-premises equipment (CPE). but the greater speed of signaling decreased the holding time of the bearer channels. 7 the signaling phase to eliminate the possibility that end users may introduce tones that would be mistaken for those used for signaling. In contrast.Signalling System No.[4] 28 Functionality The term signaling. This makes possible a set of network-based services that do not rely upon the call being routed to a particular subscription switch at which service logic would be executed. feature activation. ISUP was later used in Europe when the European networks upgraded to the ISDN. calling name and number display. See falsing. voice mail. and only during call setup. call waiting. With CCS. NFAS is signaling that is not directly associated with the path that a conversation will traverse and may concern other information located at a centralized database such as service subscription. non-facility associated signalling (NFAS) became possible. leading to significant savings and performance increases in both signaling and facility usage. For charged calls. and at the end of the call. malicious caller identification. SS6 and SS7 are referred to as so-called Common Channel Interoffice Signalling Systems (CCIS) or Common Channel Signaling (CCS) due to their hard separation of signaling and bearer channels. SS7. the SIGTRAN protocols are not an SS7 variant. and other call-related information. and release of telephone calls.

[5] Also possible with SS7 is Non-Call-Associated Signaling.536 Mbit/s) or E1 (1.536. There are differences between the specifications for the 1. or 1. 1. Quasi-associated signaling reduces the number of SS7 links necessary to interconnect all switching exchanges and SCPs in an SS7 signaling network.[11] When operating in the quasi-associated mode.536 and 1. i.5 Mbit/s rate. .[16] The protocols for SIGTRAN are M2PA. and service control point (SCPs). a point code.[8] An example of this is the exchange of the registration information used between a mobile telephone and a home location register (HLR) database: a database that tracks the location of the mobile. The associated mode of signaling is not the predominant choice of modes in North America. 64. SS7 links normally are directly connected between switching exchanges using F-links.[12] Physical network SS7 is an out-of-band signaling protocol.3 for the 1. and F) and three signaling nodes .[10] When operating in the associated mode. SS7 signaling progresses from the originating switch to the terminating switch. An SS7 network has to be made up of SS7-capable equipment from end to end in order to provide its full functionality. in North America one (56 or 64 kbit/s) or all (1. 7 possible by SS7 with NFAS is the exchange of signaling information during the middle of a call. The links between nodes are full-duplex 56. SS7 links are normally indirectly connected between switching exchanges using an intervening network of STPs. M3UA and SUA. One or more signaling links can be connected to the same two endpoints that together form a signaling link set. This mode is more economical for small networks.703 [14] for the 1. E. This separation extends onto the physical network by having circuits that are solely dedicated to carrying the SS7 links.536 Mbit/s rate.0 Mbit/s rates) are called high speed links (HSL) in contrast to the low speed (56 and 64 kbit/s) links. High speed links are specified in ITU-T Recommendation Q.111. simply referred to as the 1. D. In Europe they are usually one (64 kbit/s) or all (1.984 kbit/s) timeslots (DS0s) within an E1 facility.5 Mbit/s and 2. which is signaling that is not directly related to the establishment of a telephone call. B. M2UA. Other examples include Intelligent Network and local number portability databases. separate from the bearer channels that carry the voice or data. The quasi-associated mode of signaling is the predominant choice of modes in North America. This mode is more economical for large networks with lightly loaded signaling links.5 Mbit/s and 2. Clearly splitting the signaling plane and voice circuits. SS7 signaling progresses from switch to switch through the PSTN following the same path as the associated facilities that carry the telephone call.984 kbit/s graded communications channels.[15] SIGTRAN provides signaling using SCTP associations over the Internet Protocol.984 Mbit/s) transmission facility for the transport of SS7 signaling messages. following a path through a separate SS7 signaling network composed of signal transfer points.0 Mbit/s rates. Extended services are provided by a database interface at the SCP level using the SS7 network. SS7 is designed to operate in two modes: associated mode and quasi-associated mode. This direct connection is called associated signaling. C. and ANSI Standard T1.Signalling System No.536 kbit/s) timeslots (DS0As or DS0s) within a T1 facility. High speed links utilize the entire bandwidth of a T1 (1. This indirect connection is called quasi-associated signaling. signal transfer point (STPs). In North America. Each node is identified on the network by a number.[9] 29 Signaling modes As well as providing for signaling with these various degrees of association with call set up and the facilities used to carry calls.Service switching point (SSPs). The network is made up of several link types (A.e. Signaling links are added to link sets to increase the signaling capacity of the link set. In Europe.[13] SS7 links at higher signaling capacity (1.984 Mbit/s.

4. message handling and routing to the higher levels. itu. MAP..[1] The Transaction Capabilities Application Part (TCAP) is the primary SCCP User in the Core Network. 319).700 (http:/ / www.[17] Telephone User Part (TUP) is a link-by-link signaling system used to connect calls. the Intelligent Network Application Part and the CAMEL Application Part. TCAP. Section 3. 7. int/ rec/ T-REC-Q. pp. pp. [3] (Ronayne 1986. Network Data link Physical MTP Level 3 + SCCP MTP Level 2 MTP Level 1 The SS7 protocol stack borrows partially from the OSI Model of a packetized digital protocol stack. [11] (Dryburgh 2004. 456). 141). SCCP completes the functions of the OSI network layer: end-to-end addressing and routing. pp 81-86. 700-199303-I/ en/ ). [10] ITU-T Recommendation Q. IS-41. [12] (Dryburgh 2004. 703-199607-I/ en/ [15] ITU-T Recommendation Q. xx). [9] (Russell 2002. or links to Intelligent Network Application Part (INAP) for intelligent networks.700 (http:/ / www. "Scope and field of application". 700-199303-I/ en/ ). OSI layers 1 to 3 are provided by the Message Transfer Part (MTP) and the Signalling Connection Control Part (SCCP) of the SS7 protocol (together referred to as the Network Service Part (NSP)). "signaling modes". p.700 (http:/ / www. providing a circuit-based protocol to establish. ISDN User Part (ISUP) is the key user part. itu. int/ rec/ T-REC-Q. p.. txt) . TCAP provides transaction capabilities to its Users (TC-Users). 700-199303-I/ en/ ).2.. using SCCP in connectionless mode. Together with MTP Level 3 it is called the Network Service Part (NSP). 700/ en) [2] (Ronayne 1986.1. information transfer. ietf. [14] http:/ / www. 145). 22–23). [17] ITU-T Recommendation Q. maintain. 433). References [1] ITU-T Recommendation Q.2. p.3. such as the Mobile Application Part. ISUP. 318). p. itu. itu. 703-199607-I/ en/ ). [7] ITU-T Recommendation Q. [6] (Russell 2002. [13] ITU-T Recommendation Q. p.703 (http:/ / www.Signalling System No. p. the User Part provides layer 7. int/ rec/ T-REC-Q. SCCP in connection oriented mode provides the transport layer for air interface protocols such as BSSAP and RANAP. 7 30 SS7 protocol suite SS7 protocol suite OSI layer SS7 protocols Application INAP. int/ rec/ T-REC-Q. 23). Currently there are no protocol components that provide OSI layers 4 through 6.700 (http:/ / www. p. CAP. p. 711-200103-I/ en/ ). itu. . org/ rfc/ rfc2719. and end the connections for calls. itu. int/ rec/ T-REC-Q. 4-5. [4] RFC 2719 (http:/ / www. int/ rec/ T-REC-Q. [8] (Russell 2002. p. [16] (Russell 2002. "Additions for a national option for high speed signaling links".711 (http:/ / www. . itu. for circuit related signaling. Annex A. connectionless messages (UDTs). Transaction Capabilities Application Part (TCAP) is used to create database queries and invoke advanced network functionality. Signalling Connection Control Part (SCCP) is at functional Level 4.Framework Architecture for Signaling Transport [5] (Russell 2002.. Section 1. and management services for users of the Network Service Part (NSP). Section 2. The Message Transfer Part (MTP) covers a portion of the functions of the OSI network layer including: network interface. pp 1-2. int/ rec/ T-REC-Q. such as the Telephone User Part (TUP) or the ISDN User Part (ISUP). or Mobile Application Part (MAP) for mobile services. p.

com/page/tutorials/ss7-tutorial) • SeveNTest online SS7 decoder (http://decoder. fiber optic cables.pdf) • SS7 open source project (http://www.pt. 7 (SS7/C7): Protocol. Each telephone was wired to a local telephone exchange. Networks were connected together in a hierarchical manner until they spanned cities. Inc.com/ss7. ISBN 1-587-05040-4. all inter-connected by switching centers. though the term was unknown for many decades. The technical operation of the PSTN utilizes standards created by the ITU-T. Link to online version of text below. John P.com/2007/06/05/ ss7-backbone-of-mobile-networks) • www. Automation introduced pulse dialing between the phone and the exchange. cellular networks. and Services. Originally a network of fixed-line analog telephone systems.A Brief Comparison with TCP/IP (http://www.).html) • Introduction to signaling (http://www..com: practical overview (http://www. continents and oceans.org) • Mobicents SS7 open source project (http://www. A tutorial about signaling. A user who wished to speak whistled into the transmitter until the other party heard.openss7. Lee. The combination of the interconnected networks and the single numbering plan make it possible for any phone in the world to dial any other phone. Users who wanted to talk to different people had as many telephones as necessary for the purpose.org/ss7/intro.163 and E. • Ronayne. It consists of telephone lines. countries. and the exchanges were wired together with trunks.com/services-tutorials.net). a bell was added for signalling..164 standards. • SS7 Protocol layer architecture overview. tutorials and discussion forum (http://www. Indianapolis: Cisco Press. followed by more sophisticated address signaling including multi-frequency.protocols. Sams & Co. culminating in the SS7 network that connected most exchanges by the end of the 20th century. 31 External links • Signaling System No.buzzinbees. and telephones took advantage of the exchange principle already employed in telegraph networks. Architecture. thus allowing any telephone in the world to communicate with any other. These standards allow different networks in different countries to interconnect seamlessly. however.). and then among exchanges. html) • SS7 . and then a switch hook.mobicents. and Services (http://www. the PSTN is now almost entirely digital in its core and includes mobile as well as fixed telephones. Soon. 7 (SS7/C7): Protocol.seventest. 7 • Dryburgh. (1986).php). Introduction to Digital Communications Switching (1st edition ed.Signalling System No. wired together in pairs. Signalling System No. microwave transmission links. Architecture.kenneyjacob.ss7-training. Jeff Hewitt (2004). "The Digital Network".telecomspace. and undersea telephone cables. communications satellites. • Russell.com/cgi-bin/decoder) Public switched telephone network The public switched telephone network (PSTN) is the network of the world's public circuit-switched telephone networks. Travis (2002). . • Excellent SS7 Tutorial (http://www. New York: McGraw-Hill. This was the beginning of the PSTN. Signaling System #7 (4th Edition ed.protocols. Full HTML version of the 2004 edition of the Dryburgh/Hewitt book (above). ISBN 978-0071387729. There is also a single global address space for telephone numbers based on the E. History The first telephones had no network but were in private use. Indianapolis: Howard W. ISBN 0-672-22498-4.com/pbook/pdf/ss7.

There are also private networks run by large companies which are linked to the PSTN only through limited gateways. only the oldest parts of the telephone network still use analog technology for anything other than the last mile loop to the end user. with perhaps only two layers. the central government has a regulator dedicated to monitoring the provision of PSTN services in that country. . The model developed by the US and Canada was adopted by other nations. connections to other countries. Their tasks may be for example to ensure that end customers are not over-charged for services where monopolies may exist.K. and in recent years digital services have been increasingly rolled out to end users using services such as DSL. They may also regulate the prices charged between the operators to carry each others traffic. so that if a call cannot be handled in a local cluster. Regulation of the PSTN In most countries. the General Post Office in the United Kingdom brought together a number of private companies to form a single nationalised company. in modern networks the cost of transmission and equipment is lower and. with adaptations for local markets. The work of A. Technology in the PSTN Network topology The PSTN network architecture had to evolve over the years to support increasing numbers of subscribers.Public switched telephone network The growth of the PSTN meant that traffic engineering techniques needed to be deployed to deliver quality of service (QoS) guarantees for the users. FTTx and cable modem systems. and conceived a vision of end-to-end circuit switched services. The original concept was that the telephone exchanges are arranged into hierarchies. usually for military purposes. This reduced the number of connecting trunks required between operators over long distances and also kept local traffic separate. it is passed to one higher up for onward routing. 32 PSTN operators The task of building the networks and selling services to customers fell to the network operators. In the 1970s the telecommunications industry began implementing packet switched network data services using the X. they are much flatter. like a large private branch exchange (PBX). Erlang established the mathematical foundations of methods required to determine the capacity requirements and configuration of equipment and the number of personnel required to deliver a specific level of service. In the 1980s the industry began planning for digital services assuming they would follow much the same pattern as voice services. ISDN. However. The B-ISDN vision has been overtaken by the disruptive technology of the Internet.25 protocol transported over much of the end-to-end equipment as was already in use in the PSTN. calls. In recent decades however. Today. although hierarchies still exist. There are a number of large private telephone networks which are not linked to the PSTN. known as the Broadband Integrated Services Digital Network (B-ISDN). The first company to be incorporated to provide PSTN services was the Bell Telephone Company in the United States. direct dialling and so on. For example. these state monopolies were broken up or sold off through privatization. the job of providing telephone networks fell to government as the investment required was very large and the provision of telephone service was increasingly becoming an essential public utility. In some countries however.

The call is carried over the PSTN using a 64 kbit/s channel. Within the access network. com/ doc_detail. most automated telephone exchanges now use digital switching rather than mechanical or analog switching. or into large business premises. although some parts still use the older PDH technology. A Digital Signal 1 (DS1) circuit carries 24 DS0s on a North American or Japanese T-carrier (T1) line.711. To carry a typical phone call from a calling party to a called party.ihs. . The call is switched using a call set up protocol (usually ISUP) between the telephone exchanges under an overall routing strategy. Retrieved 2011-11-20. usually into cabinets at the roadside in residential areas. However analog two-wire circuits are still used to connect the last mile from the exchange to the telephone in the home (also called the local loop). Impact on IP standards Voice quality over PSTN networks was used as the benchmark for the development of the Telecommunications Industry Association's TIA-TSB-116 standard on voice-quality recommendations for IP telephony. originally designed by Bell Labs.Document Details" (http:/ / global. These aggregated circuits are conveyed from the initial multiplexer to the exchange over a set of equipment collectively known as the access network.Public switched telephone network 33 Digital channels As described above. called circuits or channels. or 32 DS0s (30 for calls plus two for framing and signaling) on an E-carrier (E1) line used in most other countries. ihs.[1] References [1] "IHS . SONET and Synchronous Digital Hierarchy (SDH) technologies. The call is then transmitted from one end to another via telephone exchanges. the multiplexing function is moved as close to the end user as possible. the analog audio signal is digitized at an 8 kHz sample rate with 8-bit resolution using a special type of nonlinear pulse code modulation known as G. there are a number of reference points defined. The name given to this channel is Digital Signal 0 (DS0). The protocols at this reference point were standardized in ETSI areas as the V5 interface. for example. to determine acceptable levels of audio delay and echo. The DS0 circuit is the basic granularity of circuit switching in a telephone exchange. Global. The access network and inter-exchange transport use synchronous optical transmission. This is the reference point between a primary multiplexer and an exchange. A DS0 is also known as a timeslot because DS0s are aggregated in time-division multiplexing (TDM) equipment to form higher capacity communication links. cfm?currency_code=USD& customer_id=21254C2A5E0A& oshid=21254C2A5C0A& shopping_cart_id=292558332E4B4048405B2D33220A& country_code=US& lang_code=ENGL& item_s_key=00381377& item_key_date=880731& input_doc_number=TSB-116& input_doc_title=). Most of these are of interest mainly to ISDN but one – the V reference point – is of more general interest. The trunks connecting the exchanges are also digital. . In modern networks.com.

A notable proprietary implementation is the Skype protocol.[3] Protocols Voice over IP has been implemented in various ways using both proprietary and open protocols and standards. However. they are actually different. similar steps (usually in the reverse order) such as reception of the IP packets. digitization of the analog voice signal. packetization. . and broadband phone. and/or voice-messaging applications—that are transported via the Internet. such as MGCP and SIP.323 IP Multimedia Subsystem (IMS) Media Gateway Control Protocol (MGCP) Session Initiation Protocol (SIP) Real-time Transport Protocol (RTP) Session Description Protocol (SDP) Inter-Asterisk eXchange (IAX) The H.Voice over IP 34 Voice over IP Voice over Internet Protocol (Voice over IP. In particular. communication protocols. Examples of technologies used to implement Voice over IP include: • • • • • • • H. while others support high fidelity stereo codecs. which is in part based on the principles of peer-to-peer (P2P) networking. less complex protocols. methodologies.323 protocol was one of the first VoIP protocols that found widespread implementation for long-distance traffic. the Session Initiation Protocol (SIP) has gained widespread VoIP market penetration. some implementations rely on narrowband and compressed speech. H.[1] Even though IP Telephony and VoIP are terms that are used interchangeably. IP telephony has to do with digital telephony systems that use IP protocols for voice communication while VoIP is actually a subset of IP Telephony. VoIP is a technology used by IP telephony as a means of transporting phone calls. Other terms frequently encountered and often used synonymously with VoIP are IP telephony.[2] VoIP systems employ session control protocols to control the set-up and tear-down of calls as well as audio codecs which encode speech allowing transmission over an IP network as digital audio via an audio stream. since the development of newer. decoding of the packets and digital-to-analog conversion reproduce the original voice stream. VoIP) is a family of technologies. The codec used is varied between different implementations of VoIP (and often a range of codecs are used).323 deployments are increasingly limited to carrying existing long-haul network traffic. Internet telephony. and transmission as Internet Protocol (IP) packets over a packet-switched network. and transmission techniques for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks. VoIP is available on many smartphones and internet devices so even the users of portable devices that are not phones can still make calls or send SMS text messages over 3G or Wi-Fi. such as the Internet. encoding. voice over broadband (VoBB). broadband telephony. SMS. The steps involved in originating a VoIP telephone call are signaling and media channel setup. rather than the public switched telephone network (PSTN). On the receiving side. Internet telephony refers to communications services—Voice. as well as local area network services. fax.

Some residential Internet gateways and cablemodems have this function built in. or headset. by which subscribers place and receive telephone calls in much the same manner as they would via the public switched telephone network (PSTN). The application typically presents a dial pad and display field to the user to operate the application by mouse clicks or keyboard input. They are typically designed in the style of traditional digital business telephones. VoIP switches may run on commodity hardware. Phone calls between subscribers of the same provider are usually free when flat-fee service is not available A VoIP phone is necessary to connect to a VoIP service provider. This sometimes includes international calls to certain countries.[7] VoIP allows both voice and data communications to be run over a single network.[8] The prices of extensions on VoIP are lower than for PBX and key systems. • A softphone is application software installed on a networked computer that is equipped with a microphone and speaker. including cellphones.[8] . Web conferences and more—as discrete units that can all be delivered via any means and to any handset. Full-service VoIP phone companies provide inbound and outbound service with Direct Inbound Dialing.Voice over IP 35 Adoption Consumer market A major development that started in 2004 was the introduction of mass-market VoIP services that utilize existing broadband Internet access. e-mail.[6] VoIP solutions aimed at businesses have evolved into "unified communications" services that treat all communications—phone calls. voice mail. businesses are migrating from traditional copper-wire telephone systems to VoIP systems to reduce their monthly phone costs. this is often referred to as "IP backhaul". 80% of all new PBX lines installed internationally were VoIP. In 2008. Corporate use Because of the bandwidth efficiency and low costs that VoIP technology can provide. This can be implemented in several ways: • Dedicated VoIP phones connect directly to the IP network using technologies such as wired Ethernet or wireless Wi-Fi. Example of residential network including VoIP • An analog telephone adapter is a device that connects to the network and implements the electronics and firmware to operate a conventional analog telephone attached through a modular phone jack. PSTN and mobile network providers It is becoming increasingly common for telecommunications providers to use VoIP telephony over dedicated and public IP networks to connect switching centres and to interconnect with other telephony network providers. Two kinds of competitors are competing in this space: one set is focused on VoIP for medium to large enterprises. which can significantly reduce infrastructure costs. while another is targeting the small-to-medium business (SMB) market. these devices rely on standard interfaces. such as PCs or Linux systems. faxes.[4] [5] Smartphones and Wi-Fi enabled mobile phones may have SIP clients built into the firmware or available as an application download. Many offer unlimited domestic calling for a flat monthly subscription fee. Rather than closed architectures.

however. although this is not commonly done. and are delivered in sequential order.[8] Skype. has begun to cater to businesses. However. Cells from the same VC are always sent sequentially. Therefore.[13] [14] By default. They use ATM Adaptation Layer 5 (AAL5) to segment each Ethernet packet into a series of 53-byte ATM cells for transmission and reassemble them back into Ethernet packets at the receiver. so this increases relative header overhead on every link along the user's Internet paths.[15] An alternative to preemption on slower links. but inside they are actually Asynchronous Transfer Mode (ATM) modems. are already in place with VoIP. Every Ethernet packet must be completely transmitted before another can begin. latency can be minimized by marking voice packets as being delay-sensitive with methods such as DiffServ. VoIP implementations may face problems mitigating latency and jitter. If a second PVC were . But every packet must contain protocol headers.[13] A VoIP packet usually has to wait for the current packet to finish transmission. Dual-mode phones enable users to continue their conversations as they move between an outside cellular service and an internal Wi-Fi network. as they are caused by the physical distance the packets travel. such as dialup and DSL. especially on high-speed links where transmission times are short even for maximum-sized packets. Most of the difficulties of creating a secure telephone connection over traditional phone lines. the great majority of DSL providers use only one VC for each customer.[15] ADSL modems provide Ethernet (or Ethernet over USB) connections to local equipment.[9] In the United States the Social Security Administration (SSA) is converting its field offices of 63. so the transmitter can multiplex the active virtual circuits (VCs) in any arbitrary order. Examples include: • Routing phone calls over existing data networks to avoid the need for separate voice and data networks. Fixed delays cannot be controlled. intuitive user interfaces. so users can often make simple system configuration changes. • Secure calls using standardized protocols (such as Secure Real-time Transport Protocol).Voice over IP VoIP devices have simple.000 workers from traditional phone installations to a VoIP infrastructure carried over its existing data network.[10] [11] 36 Advantages Operational cost VoIP can be a benefit for reducing communication and infrastructure costs. It is a best-effort network without fundamental Quality of Service (QoS) guarantees. network routers handle traffic on a first-come. which originally marketed itself as a service among friends. A virtual circuit identifier (VCI) is part of the 5-byte header on every ATM cell. It is only necessary to encrypt and authenticate the existing data stream. such as digitizing and digital transmission. not just the bottleneck (usually Internet access) link.[12] • The ability to transmit more than one telephone call over a single broadband connection. Maintenance becomes simpler as there are fewer devices to oversee. is to reduce the maximum transmission time by reducing the maximum transmission unit. Challenges Quality of service Communication on the IP network is inherently less reliable in contrast to the circuit-switched public telephone network. even those with bundled VoIP service. providing free-of-charge connections between any users on the Skype network and connecting to and from ordinary PSTN telephones for a charge. first-served basis. although it is possible to preempt (abort) a less important packet in mid-transmission. as it does not provide a network-based mechanism to ensure that data packets are not lost. Network routers on high volume traffic links may introduce latency that exceeds permissible thresholds for VoIP. so that it is no longer necessary to carry both a desktop phone and a cellphone.

. Jitter results from the rapid and random (i. A full-size (1500 byte) Ethernet frame takes 94 ms to transmit at 128 kbit/s but only 8 ms at 1.[17] In such a way.[13] So QoS mechanisms can avoid the undesirable loss of VoIP packets by immediately transmitting them ahead of any queued bulk traffic on the same link. It has been suggested to rely on the packetized nature of media in VoIP communications and transmit the stream of packets from the source phone to the destination phone simultaneously across different routes (multi-path routing). If this is the bottleneck link. The receiver must resequence IP packets that arrive out of order and recover gracefully when packets arrive too late or not at all. deliberately increasing latency to improve the chance that each packet will be on hand when it is time for the voice engine to play it. given high priority and reserved for VoIP. a circuit switched system of insufficient capacity will refuse new connections while carrying the remainder without impairment. we can model jitter as a gaussian random variable. delays of 400–600 ms are typical. Because ATM links are multiplexed on a cell-by-cell basis. however. Thus according to the central limit theorem.[13] ATM's potential for latency reduction is greatest on slow links..g.1p priority tagging so that VoIP can be queued ahead of less time-critical traffic. because worst-case latency decreases with increasing link speed.4%. But VoIP usually uses UDP not TCP because recovering from congestion through retransmission usually entails too much latency. In practice. When the load on a link grows so quickly that its switches experience queue overflows. This "ATM tax" is incurred by every DSL user whether or not he takes advantage of multiple virtual circuits - and few can. roughly twice the total header overhead of a 1500 byte TCP/IP Ethernet packet (with TCP timestamps). optical fiber) and the routers driving them do not have enough buffering for queuing delays to be significant.5 Mbit/s. a high priority packet would have to wait at most 53 byte times to begin transmission. temporary failures have less impact on the communication quality. They are especially problematic when satellite circuits are involved because of the long distance to a geostationary satellite and back.Voice over IP established. This suggests continually estimating the mean delay and its standard deviation and setting the playout delay so that only packets delayed more than several standard deviations above the mean will arrive too late to be useful. and they generally support IEEE 802. this latency is probably small enough to ensure good VoIP performance without MTU reductions or multiple ATM PVCs. the variance in latency of many Internet paths is dominated by a small number (often one) of relatively slow and congested "bottleneck" links. This system may be more prone to congestion and DoS attacks[16] than traditional circuit switched systems. even when that bulk traffic queue is overflowing. it is the sum of several other random variables that are at least somewhat independent: the individual queuing delays of the routers along the Internet path in question. carry Ethernet without intermediate ATM/AAL5 layers. i. momentary audio interruptions. Although jitter is a random variable. Most Internet backbone links are now so fast (e. travels in packets over IP networks with fixed maximum capacity.e. The added delay is thus a compromise between excessive latency and excessive dropout. ATM has substantial header overhead: 5/53 = 9.g. then a low priority data packet could be suspended in mid-transmission and a VoIP packet sent right away on the high priority VC.e. VoIP receivers counter jitter by storing incoming packets briefly in a "de-jitter" or "playout" buffer. and no need to abort a low priority packet and resend it later. In capillary routing it has been suggested to use at the packet level Fountain codes or particularly raptor codes for transmitting extra 37 . unpredictable) changes in queue lengths along a given Internet path due to competition from other users for the same transmission links. congestion results and data packets are lost. while the quality of real-time data such as telephone conversations on packet-switched networks degrades dramatically. and all other data. 10 Gbit/s) that their delays are dominated by the transmission medium (e. Then the link would pick up the low priority VC where it left off.[13] Fixed delays cannot be controlled as they are caused by the physical distance the packets travel.[13] Voice. There would be no need to reduce the interface MTU and accept the resulting increase in higher layer protocol overhead. VDSL and VDSL2. This signals a transport protocol like TCP to reduce its transmission rate to alleviate the congestion. The latest generations of DSL.

11e is an approved amendment to the IEEE 802. the exchange of information between the endpoints for improved call quality calculation and a variety of other applications. 38 Susceptibility to power failure Telephones for traditional residential analog service are usually connected directly to telephone company phone lines which provide direct current to power most basic analog handsets independently of locally available power. such as Voice over Wireless IP. which provides a way to create a high-speed (up to 1 gigabit per second) Local area network using existing home wiring (power lines.[18] Some VoIP service providers use customer premise equipment (e. SIP RTCP Summary Reports. signal / noise / echo level. IP Phones and VoIP telephone adapters connect to routers or cable modems which typically depend on the availability of mains electricity or locally generated power. RFC 3611 VoIP metrics reports are exchanged between IP endpoints on an occasional basis during a call. .g. RFC 3611 VoIP metrics reports are intended to support real time feedback related to QoS problems. • IEEE 802. gap length/density). These include RTCP Extended Report (RFC 3611).11 standard that defines a set of quality-of-service enhancements for wireless LAN applications through modifications to the Media Access Control (MAC) layer.248. packet loss/discard burst metrics (burst length/density. • The ITU-T G. packet discard rate (because of jitter). H. The standard is considered of critical importance for delay-sensitive applications.30 and MGCP extensions.hn provides QoS by means of "Contention-Free Transmission Opportunities" (CFTXOPs) which are allocated to flows (such as a VoIP call) which require QoS and which have negotiated a "contract" with the network controllers. phone lines and coaxial cables). Some examples include: • IEEE 802. cablemodems) with battery-backed power supplies to assure uninterrupted service for up to several hours in case of local power failures.460. Some VoIP service providers implement services to route calls to other telephone services of the subscriber. The RFC 3611 VoIP Metrics block is generated by an IP phone or gateway during a live call and contains information on packet loss rate.1p defines 8 different classes of service (including one dedicated to voice) for traffic on layer-2 wired Ethernet. Mean Opinion Scores (MOS) and R factors and configuration information related to the jitter buffer. Layer-2 quality of service A number of protocols that deal with the data link layer and physical layer include quality-of-service mechanisms that can be used to ensure that applications like VoIP work well even in congested scenarios. and an end of call message sent via SIP RTCP Summary Report or one of the other signaling protocol extensions. in the event that the customer's network device is inaccessible to terminate the call. such as built-in voicemail or phone book features. G.9 Annex B (for H. such a cellular phone. network delay..323).Voice over IP redundant packets making the communication more reliable. A number of protocols have been defined to support the reporting of QoS/QoE for VoIP calls. end system delay. or that have other modern phone features. The susceptibility of phone service to power failures is a common problem even with traditional analog service in areas where many customers purchase modern telephone units that operate with wireless handsets to a base station. H.hn standard. Such battery-backed devices typically are designed for use with analog handsets.

However. Unlike in cellular phones. in the United States.[20] There are more complications since IP allows a great deal of mobility. even roaming via another cellular company. Since the IP address itself does not necessarily provide location information presently. Emergency calls. so the address itself reveals no useful information for the emergency services.. a single-point failure can isolate communities from all communication. The broadband service provider knows the physical location. rather than the address of the ISP. but is not necessarily tracking the IP addresses in use. At the VoIP level. Even during the 9/11 attacks on the World Trade Centers the internet routed data around the failed nodes that were housed . cannot easily be routed to a nearby call center.Voice over IP 39 Emergency calls The nature of IP makes it difficult to locate network users geographically.[20] VoIP Enhanced 911 (E911) is a method by which VoIP providers in the United States support emergency services. it is not so simple. IP addresses are often dynamically assigned. but this does not necessarily allow the mapping of an IP address to that location. including Enhanced 911 and equivalent services in other locales.[20] One shortcoming of VoIP E911 is that the emergency system is based on a static table lookup. In the IP world. So in this case. it does not recognize how that IP traffic was engaged. allowing individuals to call for emergency services and many businesses to continue to operate normally. An Internet outage does not necessarily mean that a voice communication outage will occur simultaneously. Sometimes. since a mobile user could be anywhere that there is network coverage. or at the time a broadband router is engaged. there is no relationship between IP address and physical location. so this could be many kilometres away or even in another country. If an emergency call comes from that number. so the ISP may allocate an address for online access. In short. and can relate a specific telephone number to the user. All VoIP providers that provide access to the public switched telephone network are required to implement E911. However. a 3G mobile handset or USB wireless broadband adapter. A broadband provider may know the location where the wires terminate. a phone or gateway may identify itself with a SIP registrar by using a username and password. the Wireless Communications and Public Safety Act of 1999 leaves the burden of responsibility upon the subscribers and not the service providers to keep their emergency information up to date. then the IP address has no relationship with any physical location.[19] [20] A fixed line phone has a direct relationship between a telephone number and a physical location. where the location of an E911 call can be traced using Assisted GPS or other methods. the Internet Telephony Service Provider (ITSP) knows that a particular user is online.[20] Lack of redundancy With the current separation of the Internet and the PSTN. a certain amount of redundancy is provided. VoIP systems may route emergency calls to a non-emergency phone line at the intended department. but does not necessarily know to which physical location it corresponds. In situations where telephone services become completely reliant on the Internet infrastructure. therefore.[20] a service for which the subscriber may be charged. For example. the internet as designed by DARPA in the early 1980s was specifically designed to be fault tolerant under adverse conditions.g. today a "best efforts" approach is to use an available database to find that user and the physical address the user chose to associate with that telephone number—clearly an imperfect solution. To provide another example: if mobile data is used. a broadband connection can be used to dial a virtual private network that is employer-owned. The ISP recognizes individual IP addresses. the IP address being used will belong to the range of the employer. e. The VoIP E911 emergency-calling system associates a physical address with the calling party's telephone number as required by the Wireless Communications and Public Safety Act of 1999. Participation in E911 is not required and customers may opt-out of E911 service. then the physical location is known. the VoIP E911 information is only accurate so long as subscribers are diligent in keeping their emergency address information up-to-date. When this is done. at least one major police department has strongly objected to this practice as potentially endangering the public. In the United States.

it is the responsibility of the former carrier to "map" the old number to the undisclosed number assigned by the new carrier. Typically. LCR providers can no longer rely on using the network root prefix to determine how to route a call.164 is a global FGFnumbering standard for both the PSTN and PLMN. .164 to allow calls to be routed to and from VoIP subscribers and the PSTN/PLMN. VoIP has been identified in the past as a Least Cost Routing (LCR) system. A dialed number is initially received by the original carrier and quickly rerouted to the new carrier.[23] VoIP implementations can also allow other identification techniques to be used. This is achieved by maintaining a database of numbers. In November 2007. which is based on checking the destination of each telephone call as it is made.[21] Number portability is a service that allows a subscriber to select a new telephone carrier without requiring a new number to be issued. The FCC mandates carrier compliance with these consumer-protection stipulations.Voice over IP in or near the towers. MNP checks are important to assure that this quality of service is met. Often VoIP implementations employ methods of translating non-E. it needs to provide a certain level of reliability when handling calls.164 identifiers to E. So single point failures while possible in some geographic areas are not the norm for the internet as a whole. VoIP service providers are able to offer business subscribers the level of reliability they require. it might be necessary to query the GSM network about which home network a mobile phone number belongs to. For example. Multiple porting references must be maintained even if the subscriber returns to the original carrier. such as the Skype-In service provided by Skype[26] and the ENUM service in IMS and SIP. Instead. A voice call originating in the VoIP environment also faces challenges to reach its destination if the number is routed to a mobile phone number on a traditional mobile carrier. like the UK. and then sending the call via the network that will cost the customer the least. 40 Number portability Local number portability (LNP) and Mobile number portability (MNP) also impact VoIP business.[27] Echo can also be an issue for PSTN integration. they must now determine the actual network of every number before routing the call. Skype allows subscribers to choose "Skype names"[24] (usernames) whereas SIP implementations can use URIs[25] similar to email addresses. VoIP solutions also need to handle MNP when routing a voice call. With GSM number portability now in place. the Federal Communications Commission in the United States released an order extending number portability obligations to interconnected VoIP providers and carriers that support VoIP providers. As the popularity of VoIP increases in the enterprise markets because of least cost routing options. Therefore. In countries without a central database.[28] Common causes of echo include impedance mismatches in analog circuitry and acoustic coupling of the transmit and receive signal at the receiving end. Most VoIP implementations support E. By handling MNP lookups before routing a call and by assuring that the voice call will actually work.164 numbers and vice-versa. PSTN integration E.[22] This rating is subject to some debate given the complexity of call routing created by number portability.

record conversations and break into voice mailboxes. although having a secure phone is much easier to implement with VoIP than traditional phone lines. Dr Tom Berson. Some vendors also use compression."[42] . Further research has shown that tapping into a fiber optic network without detection is difficult if not impossible. but is provided by the majority of VoIP providers. providing many businesses great flexibility. it is relatively easy to eavesdrop on VoIP calls and even change their content. thus permitting spoofing attacks. Many VoIP carriers allow callers to configure arbitrary Caller ID information.Voice over IP 41 Security VoIP telephone systems are susceptible to attacks as are any internet-connected devices.[40] Caller ID Caller ID support among VoIP providers varies. however. For example. or wrongfully obtain anything of value . to assess the security of the Skype software.[32] An attacker with a packet sniffer could intercept your VoIP calls if you are not on a secure VLAN. This means that once a voice packet is within the internet backbone it is relatively safe from interception. A modicum of security is afforded by patented audio codecs in proprietary implementations that are not easily available for open source applications.[34] Secure Voice over IP is accomplished by encrypting VoIP with Type 1 encryption. Private Session Border Controllers are used along with firewalls to enable VoIP calls to and from protected networks. which may make eavesdropping more difficult. As a result. Skype uses a proprietary protocol to route calls through other Skype peers on the network.. such as Wireshark. real security requires encryption and cryptographic authentication which are not widely supported at a consumer level.[33] Securing VoIP To prevent the above security concerns government and military organizations are using Voice over Secure IP (VoSIP). In 2005. harvest customer data. but as of January 2009 still has not been enacted.[35] [36] [37] [38] [39] Public Secure VoIP is also available with free GNU programs. The Truth in Caller ID Act has been in preparation in the US Congress since 2006. Secure Voice over Secure IP is accomplished by using Type 1 encryption on a classified network. Skype does not use SRTP. such security through obscurity has not proven effective in other fields. and his conclusions are available in a published report. like SIPRNet. allowing it to traverse symmetric NATs and firewalls.[29] [30] [31] Another challenge is routing VoIP traffic through firewalls and network address translators. It is possible to use IPsec to secure P2P VoIP by using opportunistic encryption. and Secure Voice over Secure IP (SVoSIP) to protect confidential and classified VoIP communications.. This means that hackers who know about these vulnerabilities (such as insecure passwords) can institute denial-of-service attacks. that facilitate sniffing of VoIP conversations. The existing security standard Secure Real-time Transport Protocol (SRTP) and the new ZRTP protocol are available on Analog Telephone Adapters (ATAs) as well as various softphones. There are open source solutions. Other methods to traverse NATs involve using protocols such as STUN or Interactive Connectivity Establishment (ICE). Many consumer VoIP solutions do not support encryption. Skype invited a researcher. but uses encryption which is transparent to the Skype provider. physical security of the switches within an enterprise and the facility security provided by ISPs make packet capture less of a problem than originally foreseen. However. Secure Voice over IP (SVoIP).[41] Business grade VoIP equipment and software often makes it easy to modify caller ID information. However. This bill proposes to make it a crime in the United States to "knowingly transmit misleading or inaccurate caller identification information with the intent to defraud. cause harm.

connected to an analog system. In other words. [44] The T. It could be a fax machine with an RJ-45 connector plugged straight into an IP network. or it could be a computer pretending to be a fax machine.38.38 built-in capabilities which allow the user to plug right into the network and transmit/receive faxes in native T. a T. With T. but in other cases they fail. Some newer high end fax machines have T. but with the right settings and error correction mode. The data you lose will only be a small piece. TCP is better suited for use between two IP devices. there is an increased likelihood that you will receive enough of the transmission to satisfy the requirements of the fax machine for output of the sent document. if needed.38 like the Ricoh 4410NF Fax Machine. because it would no longer be possible to assume a conventional PSTN telephone line would be available in consumer's homes.38 packet transmission includes a majority of the data sent in the prior packet. or an ATA box (or similar). T. two successive lost packets are needed to actually lose any data. .[43] Fax handling Support for sending faxes over VoIP implementations is still limited.38 is available. The existing voice codecs are not designed for fax transmission. conventional modems and other similar devices that depend on access to a PSTN telephone line for some or all of their functionality. They may only work with push-button telephones using the touch-tone system.[48] A unique feature of T.[46] UDP transmissions are preferred as they do not require testing for dropped packets and as such since each T.38 was designed to use UDP and TCP transmission methods across an IP network. the inefficiency of digitizing an analog representation (modem signal) of a digital representation (a document image) of analog data (an original document) more than negates any bandwidth advantage of VoIP. older fax machines. This in an attempt to overcome the obstacles of simulating real time transmissions using packet based protocol. An alternative IP-based solution for delivering fax-over-IP called T.38 protocol is designed to compensate for the differences between traditional packet-less communications over analog lines and packet based transmissions which are the basis for IP communications. The VoIP user may use a pulse-to-tone converter. or Fax over IP. alarm systems. Sending faxes using VoIP is sometimes referred to as FoIP. they are designed to digitize an analog representation of a human voice efficiently.[45] Originally. However. which is the core fax protocol.38 is that each packet contains a portion of the main data sent in the previous packet.[47] There have been updated versions of T.30 to resolve the fax over IP issues. The fax machine could be a traditional fax machine connected to the PSTN. These types of calls sometimes complete without any problems. some ancillary equipment makers may be forced to redesign equipment.38 termination point has a higher degree of success in re-assembling the fax transmission back into its original form for interpretation by the end device. satellite television receivers. If VoIP and cellular substitution becomes very popular. benefit from UDP near real-time characteristics due to the "no recovery rule" when a UDP packet is lost or an error occurs during transmission. However. Support for other telephony devices Another challenge for VoIP implementations is the proper handling of outgoing calls from other telephony devices such as digital video recorders. the fax "sounds" simply do not fit in the VoIP channel.Voice over IP 42 Compatibility with traditional analog telephone sets Some analog telephone adapters do not decode pulse dialing from older phones.

and specifically VoIP does not draw as clear a line to the location of a caller or a call's recipient as the traditional phone system does. The relevant EU Directive is not clearly drafted concerning obligations which can exist independently of market power (e. the treatment of VoIP service providers is a decision for each Member State's national telecoms regulator. In the European Union. disclose any limitations on their E-911 functionality to their consumers. VoIP services that function over managed networks are often considered to be a viable substitute for PSTN telephone services (despite the problems of power outages and lack of geographical information). the obligation to offer access to emergency calls). VoIP operators in the US are required to support local number portability. The issue in question is calls between Americans and foreigners. and that creates a whole set of new legal challenges. and enable law enforcement authorities to conduct surveillance pursuant to the Communications Assistance for Law Enforcement Act (CALEA).[51] VoIP operators also receive the benefit of certain US telecommunications regulations.g. even if a service provider has "significant market power". where the government is monopolizing telecommunication service. universal service contributions. As VoIP's low cost and flexibility convinces more and more organizations to adopt the technology. A review of the EU Directive is under way and should be complete by 2007. make service accessible to people with disabilities. VoIP services that function over unmanaged networks are often considered to be too poor in quality to be a viable substitute for PSTN services. "Interconnected" VoIP operators also must provide Enhanced 911 service. The country has installed firewalls to prevent international calls being made using VoIP.[52] Throughout the developing world. the surveillance for law enforcement agencies becomes more difficult.[53] In Ethiopia. restrictions on the use of VoIP are imposed. as a result. including in Panama where VoIP is taxed. major operators that provide these services (in practice. governments are becoming more interested in regulating VoIP in a manner similar to PSTN services.[50] In the US. .Voice over IP 43 Legal issues As the popularity of VoIP grows. The National Security Agency (NSA) is not authorized to tap Americans' conversations without a warrant—but the Internet. as a result. countries where regulation is weak or captured by the dominant operator.. Guyana where VoIP is prohibited and India where its retail commercial sales is allowed but only for long distance service. they may be provided without any specific obligations. the Internet). and other mandated payments. VoIP technology has also increased security concerns because VoIP and similar technologies have made it more difficult for the government to determine where a target is physically located when communications are being intercepted. it is a criminal offense to offer services using VoIP. including an entitlement to interconnection and exchange of traffic with incumbent local exchange carriers via wholesale carriers. which must use competition law to define relevant national markets and then determine whether any service provider on those national markets has "significant market power" (and so should be subject to certain obligations). These measures were taken after the popularity of VoIP reduced the income generated by the state owned telecommunication company. pay regulatory fees. incumbent operators) may find themselves bound by obligations of price control or accounting separation. Providers of "nomadic" VoIP service—those who are unable to determine the location of their users—are exempt from state telecommunications regulation. and obtain affirmative acknowledgements of these disclosures from all consumers. and it is impossible to say definitively whether VoIP service providers of either type are bound by them. A general distinction is usually made between VoIP services that function over managed networks (via broadband connections) and VoIP services that function over unmanaged networks (essentially.[49] Another legal issue that the US Congress is debating concerns changes to the Foreign Intelligence Surveillance Act. the Federal Communications Commission now requires all interconnected VoIP service providers to comply with requirements comparable to those for traditional telecommunications service providers.

2007. Those who violate the law stand to be fined 50.317 US dollars) or spend two years in jail or both.323 standard. on the grounds that the service members' VoIP providers were not registered.[60] • 1996 – • ITU-T begins development of standards for the transmission and signaling of voice communications over Internet Protocol networks with the H.[54] 44 Historical milestones • 1974 – The Institute of Electrical and Electronic Engineers (IEEE) published a paper titled "A Protocol for Packet Network Interconnection. Foreign VoIP providers encounter high barriers to government registration.000 Omani Rial (about 130.[65] • 2004 – Commercial VoIP service providers proliferate. . Microsoft and Radvision initiated standardization activities for VoIP communications system. police in Oman have raided 121 internet cafes throughout the country and arrested 212 people for using/providing VoIP services.[56] • 1994 . most of whom offer flat rates.[58] [59] • Beginning in 1995. the gateway that converts the VoIP call to a POTS call should not be inside India. Intel.[61] • US telecommunication companies petition the US Congress to ban Internet phone technology. but if the remote side is a normal phone. In the UAE and Oman it is illegal to use any form of VoIP.[64] • Mark Spencer of Digium develops the first open source private branch exchange (PBX) software (Asterisk).Alon Cohen and Lior Haramaty invent Voice over IP. and subscribing to the ISP services provided on base may continue to use their US-based VoIP subscription. Unlike many VoIP providers."[55] • 1981 – IPv4 is described in RFC 791. • 1985 – The National Science Foundation commissions the creation of NSFNET.[57] • 1995 – VocalTec releases the first commercial Internet phone software. This effectively means that people who have PCs can use them to make a VoIP call to any number. wherein USFK service members arriving in Korea before June 1. it is legal to use VoIP. In the Republic of Korea. which by contract will offer pricing similar to the flat rates offered by US VoIP providers. Korean VoIP services are generally metered and charged at rates similar to terrestrial calling.[62] • 1997 – Level 3 began development of its first softswitch. In 2009. but later arrivals must use a Korean-based VoIP provider. a term they coined in 1998. Providing or using VoIP services is illegal in Oman. to the extent that Web sites of Skype and Gizmo5 are blocked. but it is illegal to have VoIP gateways inside India. only providers registered with the government are authorized to offer VoIP services.Voice over IP In India. This issue came to a head in 2006 when Internet service providers providing personal Internet services by contract to United States Forces Korea members residing on USFK bases threatened to block off access to VoIP services used by USFK members as an economical way to keep in contact with their families in the United States. A compromise was reached between USFK and Korean telecommunications officials in January 2007.[63] • 1999 – • The Session Initiation Protocol (SIP) specification RFC 2543 is released.

commsdesign. Retrieved 2009-01-21. pdf). ictregulationtoolkit. gov/ prod/ ecfs/ retrieve. htm). C (2010). axiaaudio. com/ en/ US/ docs/ ios/ solutions_docs/ qos_solutions/ QoSVoIP/ QoSVoIP. totaltele. What are some advantages of VOIP? [13] "Quality of Service for Voice over IP" (http:/ / www. Retrieved May 3. [6] Michael Dosch and Steve Church. iec. . M (2005). Retrieved 27 April 2011. 2011. govtech. Retrieved 2011-06-21. Library Technology Reports 46 (5): 11–19. [4] "Carriers look to IP for backhaul" (http:/ / www. continuitycentral. Government Computer News. . . com/ Articles/ 2009/ 06/ 01/ SSA-VOIP-implementation. . 3083. vē'ō'ī'pē (vee-oh-eye-pee). Software VoIP. skype. May 24. forbes. com/ hot-topics/ MCP/ articles/ 1311-application-level-network-interoperability-the-evolution-ims. Retrieved 2009-01-21. 2006. TMCnet. fcc. gov/ cgb/ consumerfacts/ numbport. [15] "Quality of Service for Voice over IP" (http:/ / www. com/ 2008/ 12/ 09/ skype-vonage-ringcentral_leadership_clayton_in_rc_1209claytonchristensen_inl. Skype. htm).jsessionid=YBQXDJILLQX0IQSNDLPSKH0CJUNN2JVN?articleID=159907138). com/ sip-trunking) [23] "RFC 3824 – Using E. com/ feature074. Retrieved 2009-04-27. Retrieved 2009-01-21. [25] "RFC 3969 – The Internet Assigned Number Authority (IANA) Uniform Resource Identifier (URI) Parameter Registry for the Session Initiation Protocol (SIP)" (http:/ / www. forbes. Retrieved 2009-01-21. com/ View. 2011. 2007. com/ help/ guides/ createskypename_windows/ ). Retrieved 2009-03-16. R. skype. com/ tech/ voip/ default. html) [19] Letter from the City of New York to the Federal Communications Commission (http:/ / gullfoss2. . Axia Audio. 2009. [21] Keeping your telephone number when you change your service provider – FCC (http:/ / www. asp). "OSS Architecture & Requirements for VoIP Networks". . gov/ voip/ ). as in voice) may be the most common within the industry. Bell Labs Technical Journal 10 (1): 31–45. html#wp1029054). . . Renee (December 9. Retrieved May 2. com/ allfeatures/ onlinenumber/ ). "VoIP In The Broadcast Studio" (http:/ / www. packetizer. [9] "Skype For Business" (http:/ / www. . [11] "Social Security to Build "World's Largest VOIP"" (http:/ / www. fcc. . jhtml. html). and Integrated and Mobile VoIP". [8] Korzeniowski.Voice over IP 45 Pronunciation The acronym VoIP has been pronounced variably since the inception of the term. [26] "Your personal online number" (http:/ / www. [14] Prabhakar. [27] "Application-level Network Interoperability and the Evolution of IMS" (http:/ / ipcommunications. Retrieved 2009-01-21. packetizer. com/ en/ US/ docs/ ios/ solutions_docs/ qos_solutions/ QoSVoIP/ QoSVoIP. [24] "Create a Skype Name" (http:/ / www. com). html) [22] VoIpservice. "Three Technologies You Need In 2009" (http:/ / www. The Internet Society.& Thotton. 2005. Retrieved 2009-05-28. tmcnet. google. networkstraining. but generally.164 numbers with the Session Initiation Protocol (SIP)" (http:/ / www. "SSA goes big on VOIP" (http:/ / gcn. Skype. Peter (January 8. Telecommunications Online. . Rastogi.4 VOIP – Regulatory Issues – Universal Service (http:/ / www. com/ gt/ 275677). Government Technology.gov (http:/ / www. aspx?s=gcndaily_280509& Page=1). Retrieved 2009-01-21. December 8. Forbes. [16] VOIP – Vulnerability over Internet Protocol (http:/ / www. skype. ieee. Retrieved 2009-05-29. [12] FCC. have been used. com/ showArticle. Apart from spelling out the acronym letter by letter. G. forbes. 2004. [5] "Mobile's IP challenge" (http:/ / www. com/ books?id=8CNBbrxytcAC& pg=PA557& lpg=PA557& dq=PSTN+ gateway+ VOIP+ impedance+ mismatch) .com (http:/ / www. 2011. [3] Booth. . com/ rfc/ rfc3824/ ). fcc. "Chapter 2: IP Phones. .com. Retrieved 2009-03-03. htm) [17] IEEE Multipath routing with adaptive playback scheduling for Voice over IP in Service Overlay Networks (Abstract) (http:/ / ieeexplore. Retrieved 2009-03-02. Total Telecom. skype. html). org/ online/ tutorials/ int_tele/ index. html/ ). [28] Packetcable Implementation P557 – Jeff Riddel – ISBN 1587051818 Google Books Preview (http:/ / books.. The Internet Society. com/ 2009/ 01/ 08/ small-business-voip-ent-tech-cx_bm_0108bmightytech09. com/ business/ allfeatures/ 3skypephone/ ). . December 1. the single syllable vŏy'p (voyp.com. 2004. . cgi?native_or_pdf=pdf& id_document=6517587651) [20] "FCC Consumer Advisory VoIP and 911 Service" (http:/ / www. org/ en/ Section. voipservice.[66] References [1] "Voice over Internet Protocol.. fcc. cisco. Retrieved May 3. [7] Callahan. gov/ cgb/ consumerfacts/ voip911. January 21. cisco.com. [10] William Jackson (2009-05-27). org/ xpl/ freeabs_all. there are three likely possible pronunciations: vō'ī'pē (vo-eye-pee) and vō'ip (vo-ipp). Retrieved 2009-01-21. aspx?ID=77588& t=4). "Businesses Move To Voice-Over-IP" (http:/ / www. .com. 2009). June 1. . . International Engineering Consortium. 2008). com/ rfc/ rfc3969/ ). jsp?arnumber=4520089) [18] ICT Regulation Tool Kit – 4. [2] "IP Telephony Vs VoIP" (http:/ / www. Definition and Overview" (http:/ / www.

int/ rec/ T-REC-H. Retrieved 2011-07-15. voip-info. com/ ricoh-4410nf. Tom. Packet Loss Concealment in a Secure Voice over IP Environment [38] GDC4S. cs. gov/ voip/ ) [53] Proenza. .Schulzrinne.Schooler. "VOIP System Security: Time to Worry. ipall. doi:10. UMass Discussion on UDP transmission Characteristics.com. . Cellcrypt secure VOIP heading to BlackBerry. [40] Freesorftwaremagazine. SIP: Session Initiation Protocol" (http:/ / www. . "FoIP". 46 . [57] US 5. 22. [60] The free Library RADVision and Intel Target Compatibility Between RADVision's H. [33] Skype. The National Science Foundation.org (http:/ / paper. "Secure VoIP. Ray (Secure VoIP.F. Retrieved 2006-04-05. Secure Voice-over-IP [36] Sans. com/ documents/ D-VIPER-14-1007_p11. R. Global IP Alliance (http:/ / www. Francisco J. 2007. com/ article. 637–648.us (http:/ / www. [50] Greenberg. html). com/ 2008/ 04/ 02/ voip-system-security-time-to-worry-or-maybe-not/ ?scp=4& sq=voip& st=cse/ ). freesoftwaremagazine. reference. pp. Robert E. Kahn. [31] Stanton. I'm a Fake!" (Telespoof and Fakecaller). asp?section=104& article=41826& archive=true).gov (http:/ / www. Zakon. 2008). skype. [54] Stripes. [59] "The 10 that Established VOIP (Part 1: VocalTec)" (http:/ / www. New York Times. org). pdf). itu. disa. ITU-T. 47 C.771). ilocus. html). . Faxing over IP networks [46] Umass. html). com/ Developer-Solutions/ ) [61] "H.825. mil/ stigs/ stig/ VoIP-STIG-V2R2. edu/ kurose/ transport/ UDP. Retrieved 2007-11-07. blogs. umass. txt). Computer Fraud & Security 4: 11–14. thefreelibrary. Retrieved 2009-03-02. . Stars and Stripes: USFK deal keeps VoIP access for troops [55] Vinton G. "Truth in Caller ID Act".. html). govtrack. html). [63] "The 10 that Established VOIP (Part 2: Level 3)" (http:/ / www. pdf). gov/ about/ history/ nsf0050/ internet/ launch. "A Protocol for Packet Network Intercommunication". Retrieved 2009-01-21. html). 2008). Retrieved 2009-03-02. . ipall. com/ blog/ tom-keating/ docs/ cti-buyers-guide-1996.38 [48] FaxSuperstore. [47] Faqs.Rosenberg. or Maybe Not" (http:/ / bits. Computer Telephony Interaction Magazine. com/ browse/ voip). . . Secure VOIP calling. ietf. html/ ). org/ about).com (http:/ / www. Retrieved 2009-01-21.org (http:/ / www. Retrieved 2009-01-21. org/ wiki/ view/ FoIP).org.323 Visual telephone systems and equipment for local area networks which provide a non-guaranteed quality of service" (http:/ / www. org/ blog/ 2006/ 09/ 29/ hello-mom-im-a-fake/ ). Cerf. stripes. State-of-the-art voice over IP encryptor [39] Networkworld. pdf) (PDF). SANS Institute InfoSec Reading Room [37] JHU. . html). (2006). 9 (2007) [52] FCC.R.com (http:/ / www.com (http:/ / www.mil (http:/ / iase. html). com/ news/ 2009/ 041609-cellcrypt-secure-voip-heading-to. org/ rfc/ rfc2543.gov (http:/ / www. "The State Of Cybersecurity Wiretapping's Fuzzy Future" (http:/ / www. org/ foip. org/ 07_book/ 200706/ 20070610. pdf). clsp. iLocus. free software. Computer Fraud & Security 4: 11–14. IEEE Transactions on Communications. org/ reading_room/ whitepapers/ voip/ secure_voice_over_ip_322). Ricoh 4410NF [49] "Global VOIP Policy Status Matrix" (http:/ / www. [56] "The Launch of NSFNET" (http:/ / www. "The Road to Broadband Development in Developing Countries is through Competition Driven by Wireless and VOIP" (http:/ / www. Dictionary.org (http:/ / soft-switch. fcc. [32] "Examining Two Well-Known Attacks on VOIP" (http:/ / www. [45] Soft-Switch. gdc4s. espacenet. us/ congress/ billtext.com. faqs.forbes. 323-199611-S/ en). . 1997 VoiP Developer Solutions (http:/ / www. org/ rfcs/ rfc3362. 1994 [58] Keating. . Handley.Voice over IP [29] Taub. Vol. radvision.1016/S1361-3723(06)70333-5. sans. Retrieved 2009-01-21. [64] "RFC 2543. tmcnet.org (http:/ / www.com (http:/ / download. Andy (May 15. org/ pdf/ Wireless& VOIP_10July2006. com/ columns/ secure_voip_calling_free_software_right_to_privacy).com (http:/ / www.org (http:/ / voipsa.com (http:/ / dictionary. [43] Oldphoneworks. htm). com/ textdoc?DB=EPODOC& IDX=US5. gov/ nara/ cfr/ waisidx_07/ 47cfr9_07. doi:10. R. Eric (April 2. RFC 3362-T. com/ RADVision+ and+ Intel+ Target+ Compatibility+ Between+ RADVision's+ . com/ 2007/ 07/ the_10_that_established_voip_p_2.323/320 Videoconferencing Gateway And Intel's Business Video Conferencing And TeamStation Products. Blog: "Hello Mom. 5. circleid.com (http:/ / www. pdf). access. jhu. -a019467970) June 2. gpo. pdf). iLocus.edu (http:/ / www. com/ share/ security/ 2005-031 security evaluation. com/ 2007/ 07/ the_10_that_established_voip_p_1. faqs. Tom Berson/Anagram Laboratories [34] Disa.org (http:/ / www. [65] "What is Asterisk" (http:/ / www. [30] Stanton. CircleID. "Skype Security Evaluation". Internet Protocol Telephony & Voice over Internet Protocol Security Technical Implementation Guide [35] IJCSNS. oldphoneworks. (http:/ / www. filed on November 10. Retrieved 2008-04-07. pt. No. "Internet Phone Release 4" (http:/ / blog. org/ matrix/ ). nsf. July 13. ijcsns. edu/ ~cwhite/ papers/ asilo_04_LossConceal_final. asterisk. . and the right to privacy [41] VOIPSA. .An Achievable Goal). [42] Govtrack. Retrieved 2009-01-21. faxsuperstore.1016/S1361-3723(06)70333-5. ilocus. nytimes. networkworld.an achievable goal". html). Retrieved 2007-11-07. [51] GPO. xpd?bill=s111-30). pdf) (PDF). [66] Voip | Define Voip at Dictionary. com/ 2008/ 05/ 15/ wiretapping-voip-lichtblau-tech-security08-cx_ag_0515wiretap.com (http:/ / www.edu (http:/ / www-net. Asterisk.825. . May 1974. forbes. www. com/ antique-phone-parts/ parts-and-pieces/ pulse-to-tone-converters/ ) [44] Voip-Info. . Retrieved 2006-11-23. com/ posts/ examining_two_well_known_attacks_on_voip1/ ).771 (http:/ / worldwide. e-forall. org/ rfcs/ rfc2235. [62] "RFC 2235" (http:/ / www. Retrieved 2009-01-21.

and adding or deleting media streams.org/Business/Telecommunications/Services/VoIP/) at the Open Directory Project Session Initiation Protocol The Session Initiation Protocol (SIP) is an IETF-defined signaling protocol widely used for controlling communication sessions such as voice and video calls over Internet Protocol (IP).[3] The free software community started to provide more and more of the SIP technology required to build both end points as well as proxy and registrar servers leading to a commodification of the technology. The SIP protocol is an Application Layer protocol designed to be independent of the underlying Transport Layer. In November 2000. It also allows modification of existing calls. SIP reuses most of the header fields. SIP has also found applications in messaging applications. in addition to entire private branch exchange (IP PBX) solutions that compete in the market against mostly proprietary IP PBX implementations from established vendors. Other SIP applications include video conferencing. instant messaging. encoding rules and status codes of HTTP.[1] It is a text-based protocol.[5] Each transaction consists of a client request that invokes a particular method or function on the server and at least one response. and event subscription and notification. The stack can work in proxy server or user agent scenarios and has been used in numerous commercial and research projects. such as instant messaging. The latest version of the specification is RFC 3261 from the IETF Network Working Group published in June 2002. Protocol design SIP employs design elements similar to the HTTP request/response transaction model. The modification can involve changing addresses or ports. or Stream Control Transmission Protocol (SCTP). Port 5060 is commonly used for non-encrypted signaling traffic whereas port 5061 is typically used for traffic encrypted with Transport Layer Security (TLS). SIP works in concert with several other protocols and is only involved in the signaling portion of a communication session. The National Institute of Standards and Technology (NIST). inviting more participants. providing a readable text-based format. incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP). it can run on Transmission Control Protocol (TCP). modifying and terminating two-party (unicast) or multiparty (multicast) sessions. file transfer and online games. As an example. SIP clients typically use TCP or UDP on port numbers 5060 and/or 5061 to connect to SIP servers and other SIP endpoints. client applications and SDKs. Advanced Networking Technologies Division provides a public domain implementation of the JAVA Standard for SIP[4] which serves as a reference implementation for the standard. Sessions may consist of one or several media streams. which has accelerated global adoption. There are a large number of . SIP is primarily used in setting up and tearing down voice or video calls. streaming multimedia distribution. It supports RFC 3261 in full and a number of extension RFCs including RFC 3265 (Subscribe / Notify) and RFC 3262 (Provisional Reliable Responses) etc. presence information.Voice over IP 47 External links • Voice over IP (http://www.dmoz. User Datagram Protocol (UDP).[2] History SIP was originally designed by Henning Schulzrinne and Mark Handley in 1996. the open source community at SIPfoundry actively develops a variety of SIP stacks. The protocol can be used for creating. 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.

SIP by itself does not define these features. A motivating goal for SIP was to provide a signaling and call setup protocol for IP-based communications that can support a superset of the call processing functions and features present in the public switched telephone network (PSTN). A typical SIP URI is of the form: sip:username:password@host:port. . though the two protocols themselves are very different. The URI scheme used for SIP is sip:. rather. SIP has been standardized and governed primarily by the IETF. characterized by a complex central network architecture and dumb endpoints (traditional telephone handsets).[7] [8] SIP phones may be implemented as a hardware device or as a softphone. The first proposed standard version (SIP 1. SIP features are implemented in the communicating endpoints (i. Parameters (port numbers. which receives the requests and returns a SIP response. As vendors increasingly implement SIP as a standard telephony platform.[9] Each resource of a SIP network.[11] and it can be useful in diagnosing SIP compatibility problems.Session Initiation Protocol SIP-related Internet Engineering Task Force (IETF) documents that define behavior for such applications. and the User Agent Server (UAS).323. the scheme sips: is used and SIP messages must be transported over Transport Layer Security (TLS).0 and clarified in RFC 3261. The voice and video stream communications in SIP applications are carried over another application protocol.323. SIP-enabled telephony networks can also implement many of the more advanced call processing features present in Signaling System 7 (SS7). causing a phone to ring. SIP network elements sometimes store this information. The features that permit familiar telephone-like operations: dialing a number. embedded in endpoints (terminating devices built in either hardware or software).0) was defined by RFC 2543. which means that the receiving SIP server can see this information. protocols. H. Although several other VoIP signaling protocols exist (such as BICC. hold/unhold. hearing ringback tones or a busy signal . containing a text description of the software/hardware/product involved. and call transfer. which are implemented in the network. SIP is a peer-to-peer protocol. The User-Agent field is sent in request messages. reject. 48 Network elements A SIP user agent (UA) is a logical network end-point used to create or receive SIP messages and thereby manage a SIP session. although some implementations are still relying on the older definitions. have traditionally been associated with the International Telecommunication Union (ITU). which sends SIP requests. while other protocols. such as a User Agent or a voicemail box. A SIP UA can perform the role of a User Agent Client (UAC). Examples are devices from Nokia and Research in Motion. SS7 is a centralized protocol. such as dial. the user agent may identify itself using a message header field 'User-Agent'. at the edge of the network) contrary to traditional SS7 features. MGCP. the distinction between hardware-based and software-based SIP phones is being blurred and SIP elements are implemented in the basic firmware functions of many IP-capable devices. codecs) for these media streams are defined and negotiated using the Session Description Protocol (SDP) which is transported in the SIP packet body. MEGACO). This version of the protocol was further refined to version 2.e. as in HTTP.[6] A SIP phone is a SIP user agent that provides the traditional call functions of a telephone. thus it requires only a simple (and thus scalable) core network with intelligence distributed to the network edge. If secure transmission is required. the behavior is similar. answer. often driven by 4G efforts. is identified by a Uniform Resource Identifier (URI). These roles of UAC and UAS only last for the duration of a SIP transaction.are performed by proxy servers and user agents. the Real-time Transport Protocol (RTP). its focus is call-setup and signaling. such as H. based on the general standard syntax[10] also used in Web services and e-mail. Implementation and terminology are different in the SIP world but to the end-user. SIP is distinguished by its proponents for having roots in the IP community rather than the telecommunications industry.[6] In SIP.

RFC 3261 defines the following methods:[13] • REGISTER: Used by a UA to indicate its current IP address and the URLs for which it would like to receive calls. which is why the protocol is described as peer-to-peer. although other protocol schemes are possible (such as tel:). rewrites specific parts of a request message before forwarding it. and assistance in NAT traversal. Other SIP-related network elements include. directing the client to contact an alternate set of URIs. Session border controllers (SBC) Serve as middle boxes between UA and SIP server for various types of functions. • OPTIONS: Requests information about the capabilities of a caller.[12] The first line of a response has a response code. RFC 3261 defines these server elements. gateways Devices that exist at the edge between a SIP network and other networks (as a phone network). making sure a user is allowed to make a call). More than one user agent can register at the same URI. which means its job is to ensure that a request is sent to another entity "closer" to the targeted user. indicating where the request should be sent. • BYE: Terminates a session between two users in a conference. and. without setting up a call. 49 SIP messages SIP is a text-based protocol with syntax similar to that of HTTP. But it is also possible and often good for network scalability to place this location service with a redirect server. indicated by the sip: scheme. • INVITE: Used to establish a media session between user agents. Registrar A server that accepts REGISTER requests and places the information it receives in those requests into the location service for the domain it handles which registers one or more IP addresses to a certain SIP URI. PRACK is sent in response to provisional response (1xx). For SIP requests. Proxies are also useful for enforcing policy (for example. SIP registrars are logical elements. • PRACK (Provisional Response Acknowledgement): PRACK improves network reliability by adding an acknowledgement system to the provisional Responses (1xx). The SIP response types defined in RFC 3261 fall in one of the following categories:[14] . The first line of a request has a method. with the result that all registered user agents will receive a call to the SIP URI. including network topology hiding. A proxy server primarily plays the role of routing. Although two SIP endpoints can communicate without any intervening SIP infrastructure. Proxy server An intermediary entity that acts as both a server and a client for the purpose of making requests on behalf of other clients. Redirect server A user agent server that generates 3xx (Redirection) responses to requests it receives. and a Request-URI. The redirect server allows proxy servers to direct SIP session invitations to external domains. if necessary. • CANCEL: Terminates a pending request. defining the nature of the request. • ACK: Confirms reliable message exchanges. and are commonly co-located with SIP proxies.Session Initiation Protocol SIP also defines server network elements. There are two different types of SIP messages: requests and responses. A proxy interprets. this approach is often impractical for a public service.

Redirection (3xx): Further action needs to be taken (typically by sender) to complete the request. On delivery of the ACK (2) both sides of the transaction are complete.[15] . If we take the above example. and so include an acknowledgment (ACK) of any non-failing final response (e. If no response is received after a timer controlled wait period the UAC may have chosen to terminate the transaction or retransmit the INVITE.g. referred to as a Dialog in SIP. User1’s UAC uses an Invite Client Transaction to send the initial INVITE (1) message. understood. User1 was confident the INVITE was delivered reliably.Session Initiation Protocol • • • • • • Provisional (1xx): Request received and being processed. Because of these transactional mechanisms. Success (2xx): The action was successfully received. The responses may include zero-or-more Provisional (1xx) responses and one-or-more final (2xx-6xx) responses. Client Transactions send requests and Server Transactions respond to those requests with one-or-more responses. SIP can make use of un-reliable transports such as User Datagram Protocol (UDP). 200 OK). Invite transactions differ in that they can establish a long-running conversation. Server Error (5xx): The server failed to fulfill an apparently valid request. once a response was received. Global Failure (6xx): The request cannot be fulfilled at any server. And in this case. The transactions maintain an internal state and make use of timers. However. Client Error (4xx): The request contains bad syntax or cannot be fulfilled at the server. 50 Transactions SIP makes use of transactions to control the exchanges between participants and deliver messages reliably. and accepted. User1’s UAC then must acknowledge the response. a Dialog may have been established. Transactions are further categorized as either Invite or Non-Invite.

[4] "JAIN SIP project" (http:/ / java. which is important as there are many country-specific variants of ISUP that have been implemented over the last 30 years. 774. notably to allow ISUP messages to be transported over SIP networks.[16] The SIP developer community meets regularly at the SIP Forum SIPit [17] events to test interoperability and test implementations of new RFCs. 2010-12-07. Many VoIP phone companies allow customers to use their own SIP devices. [9] ""BlackBerry MVS Software"" (http:/ / na. video telephony. IETF. MSRP (Message Session Relay Protocol) allows instant message sessions and file transfer. and terminate communication sessions based on ISUP using SIP and IP networks. [10] RFC 3986. voipuser. The Internet Society (2005) [2] Johnston. SIP-I was defined by the ITU-T. fax and data. [5] William Stallings. Uniform Resource Identifiers (URI): Generic Syntax. p. for example to notify that motion has been detected out-of-hours in a protected area. SIP-I and SIP-T[19] are two protocols with similar features. Second Edition. SIP Gateways. html)VoIP User. SIP-enabled video surveillance cameras can make calls to alert the owner or operator that an event has occurred. CRC Press. . ISBN 1580531687.[20] References [1] RFC 4168. Ietf. [3] "SIP core working group charter" (http:/ / www. where SIP-T was defined via the IETF RFC route. google. SIP test suite is developed by a Specialist Task Force at ETSI (STF 196). IETF. SIP is used in audio over IP for broadcasting applications where it provides an interoperable means for audio interfaces from different manufacturers to make connections with one another. and SIP Trunking services providing replacements for ISDN telephone lines. net/ projects/ jsip). p. com/ books?id=b8oisvv6fDAC& pg=PT774). This preserves all of the detail available in the ISUP header. (2004). . html). The Internet Society (2005) [11] "User-Agents We Have Known " (http:/ / www. org/ dyn/ wg/ charter/ sipcore-charter. com/ books?id=BYxdyekyRlwC& pg=PA76). com/ eng/ services/ business/ blackberry_mvs/ ). Conformance testing TTCN-3 test specification language is used for the purposes of specifying conformance tests for SIP implementations. Alan B. SIP: Understanding the Session Initiation Protocol.209 [6] RFC 3261.com. Retrieved 2011-01-11. Syngress. [8] Porter. .org. SIP: Session Initiation Protocol [7] Azzedine (2006). Antonio Rosela (2006). Thomas. . is a protocol used to create. Na. ISBN 9781584884651. org/ forum_topic_14998. or softphones. . or the Session Initiation Protocol with encapsulated ISUP. pp.blackberry. Retrieved 2011-01-11. ietf. Andy Zmolek. Services using SIP-I include voice. Practical VoIP Security (http:/ / books. Retrieved 2011-07-26. 76–77.[18] SIP-ISUP interworking SIP-I. such as SIP-capable telephone sets. Handbook of algorithms for wireless networking and mobile computing (http:/ / books.org .Session Initiation Protocol 51 Instant messaging and presence The Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions (SIMPLE) is the SIP-based suite of standards for instant messaging and presence information. The Stream Control Transmission Protocol (SCTP) as a Transport for the Session Initiation Protocol (SIP). Applications The market for consumer SIP devices continues to expand. modify. Jan Kanclirz. ISBN 9781597490603. and it is not always possible to express all of the same detail using a native SIP message. google. blackberry. Artech House. there are many devices such as SIP Terminal Adapters.

net [18] Jonsson. UDP's stateless nature is also useful for servers answering small queries from huge numbers of clients.[2] Common network applications that use UDP include: the Domain Name System (DNS). [19] "RFC3372: SIP-T Context and Architectures" (http:/ / www. "Streaming audio contributions over IP" (http:/ / tech.columbia.com/sip. com/ Documents/ KonneticSIPIntroduction. EBU Technical Review.edu/sip/) hosted by Columbia University • SIP Latest specifications (http://www. which may not be an option in a real-time system.htm) hosted by SIPKnowledge • Some examples of SIP Use Cases (features) (http://www. etsi. .xhtml) User Datagram Protocol The User Datagram Protocol (UDP) is one of the core members of the Internet Protocol Suite. UDP supports packet broadcast (sending to all on local network) and multicasting (send to all subscribers).Session Initiation Protocol [12] Stallings.214-215 [14] Stallings. konnetic. UDP provides an unreliable service and datagrams may arrive out of order.216-217 [15] James Wright. [20] White Paper: "Why SIP-I? A Switching Core Protocol Recommendation" (http:/ / www. IP tunneling protocols and many online games. ietf. Voice over IP (VoIP). pdf) 52 External links • Computers/Internet/Protocols/SIP/ (http://www.[1] If error correction facilities are needed at the network interface level. Retrieved 2011-01-11. avoiding the overhead of such processing at the network interface level. Reed in 1980 and formally defined in RFC 768. Mathias Coinchon (2008). pdf) [17] http:/ / www. computer applications can send messages. ebu. pdf) (PDF).org/assignments/sip-parameters) • IANA: SIP Event Types Namespace (http://www. .An Introduction" (http:/ / www.dmoz. org/ rfc/ rfc3372.iana. txt). .iana. Trivial File Transfer Protocol (TFTP). Lars. p. Retrieved 2011-01-11. [16] Experiences of Using TTCN-3 for Testing SIP and also OSP (http:/ / portal. Konnetic. Thus. UDP assumes that error checking and correction is either not necessary or performed in the application. or go missing without notice. org/ ptcc/ downloads/ TTCN3SIPOSP.com/eBooks. in this case referred to as datagrams. the set of network protocols used for the Internet.org/assignments/sip-events/sip-events. streaming media applications such as IPTV. sipit. Time-sensitive applications often use UDP because dropping packets is preferable to waiting for delayed packets. . 4gamericas.org/Computers/Internet/Protocols/SIP//) at the Open Directory Project • Henning Schulzrinne's SIP homepage (http://www. With UDP. pp. "SIP . to other hosts on an Internet Protocol (IP) network without requiring prior communications to set up special transmission channels or data paths.sipknowledge. an application may use the Transmission Control Protocol (TCP) or Stream Control Transmission Protocol (SCTP) which are designed for this purpose. 2002-09.cs.sipcenter. pdf) (PDF). pp. ordering. UDP uses a simple transmission model without implicit handshaking dialogues for providing reliability.214 [13] Stallings. org/ documents/ 3G_Americas_SIP-I_White_Paper_August_2007-FINAL. ch/ webdav/ site/ tech/ shared/ techreview/ trev_2008-Q1_coinchon. Retrieved 2010-12-27.nsf/html/Personal+Selective+ Presence) • IANA: SIP Parameters (http://www. or data integrity. The protocol was designed by David P. appear duplicated. Unlike TCP.

On Unix-like operating systems. If the source host is the client.[2] Length A field that specifies the length in bytes of the entire datagram: header and data. offset (bits) 0 32 64+ 0 – 15 16 – 31 Source Port Number Destination Port Number Length Data Checksum The UDP header consists of 4 fields.[4] If transmission reliability is desired. and can be used for any purpose. they are used as temporary ports primarily by clients when communicating with servers. UDP is sometimes referred to as Unreliable Datagram Protocol.527 bytes of data) for a UDP datagram.[2] Checksum . Port numbers 1024 through 49151 are the registered ports used for IANA-registered services. it must be implemented in the user's application. If the source host is the server.535 − 8 byte UDP header − 20 byte IP header). from which software running on the host may randomly choose a port in order to define itself. For this reason. using one of these ports requires superuser operating permission. then it should be zero. The field size sets a theoretical limit of 65. if the client is the destination host then the port number will likely be an ephemeral port number and if the destination host is the server then the port number will likely be a well-known port number. a 16 bit integer value. A port is a software structure that is identified by the port number.[2] Port numbers 0 through 1023 are used for common.[2] In effect.[2] Destination port number This field identifies the receiver's port and is required.User Datagram Protocol 53 Service ports UDP applications use datagram sockets to establish host-to-host communications. Packet structure UDP is a minimal message-oriented Transport Layer protocol that is documented in IETF RFC 768. but is a permissible source port value if the sending process does not expect messages in response. The practical limit for the data length which is imposed by the underlying IPv4 protocol is 65. The minimum length is 8 bytes since that's the length of the header. allowing for port numbers between 0 and 65535. which is a combination of an IP address and a service port. Port 0 is reserved.[3] UDP provides application multiplexing (via port numbers) and integrity verification (via checksum) of the header and payload.507 bytes (65.[1] The use of two of those is optional in IPv4 (pink background in table). They are also used as ephemeral ports. well-known services. UDP provides no guarantees to the upper layer protocol for message delivery and the UDP protocol layer retains no state of UDP messages once sent. In IPv6 only the source port is optional (see below). the port number is likely to be an ephemeral port number. the port number is likely to be a well-known port number. Similar to source port number. Ports 49152 through 65535 are dynamic ports that are not officially designated for any specific service. If not used. Source port number This field identifies the sender's port when meaningful and should be assumed to be the port to reply to if needed. An application binds a socket to its endpoint of data transmission. The Internet Assigned Numbers Authority has divided port numbers into three ranges.535 bytes (8 byte header + 65. each of which is 2 bytes (16 bits).

the checksum is computed using a PSEUDO-HEADER that contains some of the same information from the real IPv4 header. UDP checksum computation is optional for IPv4. the field uses the value all-zeros. If the checksum calculation results in the value zero (all 16 bits 0) it should be sent as the one's complement (all 1s). If no checksum is generated by the transmitter. and the data. The protocol is that for UDP (see List of IP protocol numbers): 17 (0x11). The method used to compute it is changed as documented in RFC 2460: Any transport or other upper-layer protocol that includes the addresses from the IP header in its checksum computation must be modified for use over IPv6 to include the 128-bit IPv6 addresses. the UDP header. again a PSEUDO-HEADER is used that mimics the real IPv6 header: . If a checksum is not used it should be set to the value zero. padded with zero octets at the end (if necessary) to make a multiple of two octets.[5] This field is not optional for IPv6. The sum is then one's complemented to yield the value of the UDP checksum field. The difference between IPv4 and IPv6 is in the data used to compute the checksum.[6] 54 Checksum computation The method used to compute the checksum is defined in RFC 768: Checksum is the 16-bit one's complement of the one's complement sum of a pseudo header of information from the IP header. IPv4 PSEUDO-HEADER When UDP runs over IPv4.User Datagram Protocol The checksum field is used for error-checking of the header and data. The UDP length field is the length of the UDP header and data. all 16-bit words are summed using one's complement arithmetic. The following table defines the PSEUDO-HEADER used only for the checksum calculation. IPv6 PSEUDO-HEADER When UDP runs over IPv6.[6] When computing the checksum.[5] In other words. The PSEUDO-HEADER is not the real IPv4 header used to send an IP packet. the checksum is mandatory. bits 0 32 64 96 128 160+ 0–7 8 – 15 16 – 23 24 – 31 Source address Destination address Zeros Protocol Source Port Length Data UDP length Destination Port Checksum The source and destination addresses are those in the IPv4 header.

[2] Most often. Potentially more seriously. it will be the address in the last element of the Routing header. unlike TCP. high rate UDP traffic loads. Reliability and congestion control solutions Lacking reliability. if the IPv6 packet does not contain a Routing header. The value of the Next Header field is the protocol value for UDP: 17. The UDP length field is the length of the UDP header and data. as they frequently give a bandwidth load that is inelastic. UDP-based applications don't necessarily have good congestion avoidance and control mechanisms. at the originating node. The Datagram Congestion Control Protocol (DCCP) is being designed as a partial solution to this potential problem by adding end host TCP-friendly congestion control behavior to high-rate UDP streams such as streaming media. it will be the destination address in the IPv6 header. UDP applications must generally be willing to accept some loss. Congestion insensitive UDP applications that consume a large fraction of available bandwidth could endanger the stability of the internet. otherwise. errors or duplication. and. UDP applications do not employ reliability mechanisms and may even be hindered by them. The destination address is the final destination. Some applications such as TFTP may add rudimentary reliability mechanisms into the application layer as needed. Streaming media. If an application requires a high degree of reliability. a protocol such as the Transmission Control Protocol or erasure codes may be used instead. Network-based mechanisms have been proposed to minimize potential congestion collapse effects of uncontrolled. loss of packets is not usually a fatal problem. real-time multiplayer games and voice over IP (VoIP) are examples of applications that often use UDP. . Network-based elements such as routers using packet queuing and dropping techniques are often the only tool available to slow down excessive UDP traffic. that will be the destination address in the IPv6 header.User Datagram Protocol 55 bits 0 32 64 96 128 160 192 224 256 288 320 352 384+ 0 – 7 8 – 15 16 – 23 Source address 24 – 31 Destination address UDP length Zeros Source Port Length Data Next Header Destination Port Checksum The source address is the one in the IPv6 header. at the receiving node. In these particular applications.

which means that it requires handshaking to set up end-to-end communications. unless they implement congestion control measures at the application level. • Lightweight – There is no ordering of messages. • Datagrams – Packets are sent individually and are checked for integrity only if they arrive. • Streaming – Data is read as a byte stream. there's either no missing data. It is assumed in VoIP UDP that the end users provide any necessary real time confirmation that the message has been received. accounting. When data segments arrive in the wrong order. Communication is achieved by transmitting information in one direction from source to destination without verifying the readiness or state of the receiver. the order in which they arrive cannot be predicted. the first message will reach the receiving application first. Real-time video and audio streaming protocols are designed to handle occasional lost packets. When TCP detects packet loss. it cannot be known if it will reach its destination. the Routing Information Protocol (RIP)[1] and the Dynamic Host Configuration Protocol (DHCP). • Unreliable – When a message is sent. Multiple attempts to deliver the message are made. UDP is a simpler message-based connectionless protocol. one primary benefit of UDP over TCP is the application to voice over internet protocol (VoIP) where any handshaking would hinder clear voice communication. • No congestion control – UDP itself does not avoid congestion. so only slight degradation in quality occurs. • Reliable – TCP manages message acknowledgment. However. Connectionless protocols do not set up a dedicated end-to-end connection. If it gets lost along the way. and database systems. developing quality of service solutions is seen as crucial by some. It is a small transport layer designed on top of IP. • Heavyweight – TCP requires three packets to set up a socket connection. Because both TCP and UDP run over the same network. it will throttle back its data rate usage. where queries must be fast and only consist of a single request followed by a single reply packet.[7] Comparison of UDP and TCP Transmission Control Protocol is a connection-oriented protocol. Packets have definite boundaries which are honored upon receipt. retransmission or timeout. Once a connection is set up user data may be sent bi-directionally over the connection.User Datagram Protocol 56 Applications Numerous key Internet applications use UDP. retransmission and timeout. TCP buffers the out-of-order data until all data can be properly re-ordered and delivered to the application. many businesses are finding that a recent increase in UDP traffic from these real-time applications is hindering the performance of applications using TCP. Since both real-time and business applications are important to businesses. before any user data can be sent. There is no concept of acknowledgment. it could get lost along the way. meaning a read operation at the receiver socket will yield an entire message as it was originally sent. in case of multiple timeouts. the server will re-request the lost part. TCP handles reliability and congestion control. no distinguishing indications are transmitted to signal message (segment) boundaries. and it's possible for high bandwidth applications to trigger congestion collapse. such as point of sale. the Simple Network Management Protocol (SNMP). rather than large delays if lost packets were retransmitted. • Ordered – if two messages are sent over a connection in sequence. no tracking connections. or. In TCP. Voice and video traffic is generally transmitted using UDP. etc. . the connection is dropped. including: the Domain Name System (DNS). • Not ordered – If two messages are sent to the same recipient.

TCP/IP: Protocol Suite. RFC references • • • • RFC 768 – User Datagram Protocol RFC 2460 – Internet Protocol. Retrieved 2011-08-17. Version 6 (IPv6) Specification RFC 4113 – Management Information Base for the UDP RFC 5405 – Unicast UDP Usage Guidelines for Application Designers External links • • • • • IANA Port Assignments (http://www. [4] Clark. [3] content@ipv6.A. K. 1st ed. htm). org/ html/ rfc2460 [7] "The impact of UDP on Data Applications" (http:/ / www. RFC 768: User Datagram Protocol.P. New Delhi. J.org/assignments/port-numbers) The Trouble with UDP Scanning (PDF) (http://condor.com.edu/~jkristof/papers/udpscanning.faqs.User Datagram Protocol 57 References [1] Kurose. (2000). West Sussex. ISBN 9780131365483. W.. (2010).com/en-us/magazine/cc163648.microsoft.com. Computer Networking: A Top-Down Approach (5th ed. [5] Postel. Boston.). [2] Forouzan. com/ 2007/ 08/ whiteboard_series_nice_guys_fi. Data Networks IP and the Internet. . J. MA: Pearson Education. India: Tata McGraw-Hill Publishing Company Limited. Internet Engineering Task Force. F. Retrieved from http:/ / tools. com/ articles/ general/ User-Datagram-Protocol.pdf) Breakdown of UDP frame (http://www.com/enp/protocol/udp.com.depaul. . org/ html/ rfc768 [6] Deering S. html).networksorcery.html) . Ross. & Hinden R. (August 1980). RFC 2460: Internet Protocol.htm) UDP on MSDN Magazine Sockets and WCF (http://msdn. B. Retrieved from http:/ / tools.org/docs/iptables/udpconnections. (December 1998).aspx) UDP connections (http://www. networkperformancedaily. Internet Engineering Task Force. Ipv6. (2003). Networkperformancedaily. Retrieved 2011-08-17. 1st ed.iana. "UDP Protocol Overview" (http:/ / ipv6. Version 6 (IPv6) Specification. ietf. England: John Wiley & Sons Ltd. M. ietf.

The term most notably refers to applications that access the World Wide Web.0: Previously used to indicate compatibility with the Mozilla rendering engine (iPad. by submitting a characteristic identification string to its operating peer. In HTTP and SIP. Safari on the iPad has used the following: Mozilla/5. en-us): Details of the system in which the browser is running AppleWebKit/531. this is transmitted in a header field User-Agent. your user agent string may be similar to WikiBrowser/1. which means it can provide a streamlined .10: The platform the browser uses (KHTML. Possible privacy issue The information could help distinguish Internet users from one another because it differs. like Gecko) Mobile/7B405 The components of this string are as follows: • • • • Mozilla/5. but other systems. User agent identification Some user agents identify their software as part of the client–server conversation. When a user agent operates. Bots.5 bits of identifying information.0). or software revision. often considerably. it typically identifies itself. For example.10 (KHTML.0. For example if your product were called WikiBrowser. screen readers and braille browsers used by people with disabilities.txt file). its application type. often also include a URL and/or e-mail address so that the Webmaster can contact the operator of the bot. CPU OS 3_2_1 like Mac OS X. An unofficial format.21. used by Web browsers is as follows: Mozilla/[version] ([system and browser information]) [platform] ([platform details]) [extensions]. like Gecko): Browser platform details • Mobile/7B405: This is used by the browser to indicate specific enhancements that are available directly in the browser or through third parties. such as the Session Initiation Protocol (SIP). en-us) AppleWebKit/531. In the HTTP. This string is then used by the communications partner to characterize the client and optionally select suitable content or operating parameters for the session. The user agent string is one of the criteria by which Web crawlers may be excluded from accessing certain parts of a Web site using the Robots Exclusion Standard (robots. as well as mobile phones.[4] Format RFC 1945 merely requires the format of a user agent to be a string made up of the product and optional comments.0 (iPad. CPU OS 3_2_1 like Mac OS X. SIP. from user to user and on average browser versions carry 10. U. use the term user agent to refer to both end points of a communications session.21. based on the above.[1] Web user agents range from Web browsers to search engine crawlers (spiders). U. software vendor.0 Gecko/1. In this case. this indicates the Layout engine and version. a user agent is a client application implementing a network protocol used in communications within a client–server distributed computing system. the identity is transmitted via the User-Agent field in the request header.0) • Layout engine and version(Gecko/1.User agent 58 User agent In computing. such as Web crawlers. For example. this may be used to provide properly formatted content for desktop computers and for smartphones. An example of this is Microsoft Live Meeting which registers an extension so that the Live Meeting service knows if the software is already installed. as described by RFC 1945 [3]. and SMTP/NNTP[2] protocols. The parts of this string are as follows: • Product name and version (WikiBrowser/1. operating system.

[9] Web sites specifically targeted towards mobile phones. On the Internet.[7] encouraging developers to design Web pages that work equally well with any browser. this will result in a different site being shown when browsing the page with a specific browser. "U" stands for "USA" (for the version with 128-bit encryption). e. because older browsers could not render the same content. User agent sniffing The term user agent sniffing refers to the practice of Web sites showing different content when viewed with a certain user agent.User agent experience to joining meetings.[8] allowing correct rendering in as many browsers as possible. and N to specify the encryption strength in the user agent string.[10] Following the lifting of export restrictions. most vendors supported 256-bit encryption. This may mean that less-popular browsers are not sent complex content.g. also have the ability to change the user agent string. I. since mobile browsers often differ greatly from each other. and to test for specific browser features rather than particular browser versions or brands. such as Netscape Navigator and Internet Explorer. e. even though they might be able to deal with it correctly or. like NTT DoCoMo's I-Mode or Vodafone's Vodafone Live! portals. 59 User agent spoofing The popularity of various Web browser products has varied throughout the Web's history. Encryption strength notations Web browsers created in the United States. the W3C recommends creating HTML markup that is standard. These differences can be small.g. Until the United States government allowed encryption with keys longer than 40 bits to be exported. Therefore. Other HTTP client programs. mobile Web portals will often generate completely different markup code depending on the mobile phone used to browse them. Spam bots and Web scrapers often use fake user agents. since it encourages browser-specific design and penalizes new browsers with unrecognized user agent identifications. or quite extensive. Instead. and this has influenced the design of Web sites in such a way that Web sites are sometimes designed to work well only with particular browsers. User agent sniffing is mostly considered poor practice. A result of user agent spoofing may be that collected statistics of Web browser usage are inaccurate. in extreme cases. A useful example of this is Microsoft Exchange Server 2003's Outlook Web Access feature.. "I" stands for "International" — the browser has 40-bit encryption and can be used anywhere in the world — and "N" stands (de facto) for "None" (no encryption).. When viewed with Internet Explorer 6 (or newer).[5] Thus. rather than according to uniform standards by the World Wide Web Consortium (W3C) or the Internet Engineering Task Force (IETF). use the letters U. At times it has been popular among Web developers to initiate Viewable With Any Browser campaigns. more functionality is displayed compared to the same page in any older browsers. often rely heavily on user agent sniffing. For example. resizing of certain images to fit smaller screens. rendering of the page in WML instead of XHTML. vendors shipped various browser versions with different encryption strengths. in 1996. Web sites often include code to detect browser version to adjust the page design sent according to the user agent string received. Many developments in mobile browsing have been made in the last few years. . various browsers have a feature to cloak or spoof their identification to force certain server-side content. refused all content. like download managers and offline browsers. the Android browser identifies itself as Safari in order to aid compatibility[6] . while many older phones that do not possess these new technologies are still heavily used.

Stephen. IETF.5 Bits of Identifying Information on Average (https:/ / www. 2. Electronic Frontier Foundation. [9] Clary. The Internet Society (2002) Netnews Article Format (http:/ / tools. SIP: Session Initiation Protocol. org/ campaign/ ) [8] Pemberton.useragentstring. . org/ build/ user-agent-strings. Mozilla Developer Center. 15 Peter Eckersley. External links • User-agents. "Browser Detection and Cross Browser Support" (https:/ / developer. Bob (10 February 2003). database of known user agent strings (http://www. org/ campaign/ ) [6] http:/ / androidcommunity. mozilla.. IETF.com/) . sec. com/ forums/ f8/ android-browser-reports-itself-as-apple-safari-4701/ [7] "Viewable with Any Browser" campaign (http:/ / www. mozilla. org/ MarkUp/ #guidelines).13.org.org) • Analyze any user agent string (http://www. org/ html/ rfc5536& #035. 3. RFC 5536. [5] Burstein complaining ". W3C. http:/ / tools. anybrowser. [10] Zawinski.2. . November 2009.User agent 60 References [1] [2] [3] [4] RFC 3261. org/ html/ rfc1945#section-10.org. html). Retrieved 25 August 2011. ietf. . mozilla.section-3. eff. anybrowser. org/ deeplinks/ 2010/ 01/ tracking-by-user-agent)". Retrieved 10/18/2011. "W3C Markup Validation Service" (http:/ / www. Mozilla. Jamie (1998-03-28). Retrieved 2009-05-30.user-agents. I've been rejected until I come back with Netscape" (http:/ / www. org/ en/ Browser_Detection_and_Cross_Browser_Support). " Browser Versions Carry 10. Retrieved 2010-01-08. "user-agent strings (obsolete)" (http:/ / www-archive. ietf. 27 January 2010.. 13). w3. .

such as the User Datagram Protocol (UDP) in methods such as the Simple Service Discovery Protocol (SSDP). High-traffic websites often benefit from web cache servers that deliver content on behalf of the original. which stores content. most notably RFC 2616 (June 1999). . The server. HTTP is an Application Layer protocol designed within the framework of the Internet Protocol Suite. known as web crawlers. collaborative. However. culminating in the publication of a series of Requests for Comments (RFCs). returns a response message to the client.1. while an application running on a computer hosting a web site functions as a server. the version of HTTP in common use. or variations of the web browser such as voice browsers. such as HTML files. The protocol definitions presume a reliable Transport Layer protocol for host-to-host data transfer.[2] The Transmission Control Protocol (TCP) is the dominant protocol in use for this purpose. The standards development of HTTP has been coordinated by the Internet Engineering Task Force (IETF) and the World Wide Web Consortium (W3C). which defines HTTP/1. so-called origin server. or performs other functions on behalf of the client. A response contains completion status information about the request and may contain any content requested by the client in its message body. or provides resources. HTTP proxy servers at network boundaries facilitate communication when clients without a globally routable address are located in private networks by relaying the requests and responses between clients and servers. The client submits an HTTP request message to the server. hypermedia information systems. acts as a client. Other user agents can include the indexing software used by search providers.[1] HTTP is the foundation of data communication for the World Wide Web. In HTTP. which present an interactive voice user interface. A web browser (or client) is often referred to as a user agent (UA). The HTTP protocol is designed to permit intermediate network elements to improve or enable communications between clients and servers. Technical overview HTTP functions as a request-response protocol in the client-server computing model.Hypertext Transfer Protocol 61 Hypertext Transfer Protocol HTTP Persistence · Compression · HTTPS Request methods OPTIONS · GET · HEAD · POST · PUT · DELETE · TRACE · CONNECT Header fields Cookie · ETag · Location · Referer DNT · X-Forwarded-For Status codes 301 Moved permanently 302 Found 303 See Other 403 Forbidden 404 Not Found The Hypertext Transfer Protocol (HTTP) is a networking protocol for distributed. a web browser. for example. HTTP has found application even with unreliable protocols. to improve response time.

an error message. That same web hosting company reported that by June 1996.1 was supported in Arena.[9] Netscape Navigator Gold 2. Berners-Lee and his team are credited with inventing the original HTTP protocol along with the HTML and the associated technology for a web server and a text-based web browser. End user adoption of the new browsers was rapid.1 compliant. Uniform Resource Locators (URLs)—using the http or https URI schemes.[9] Mosaic 2.1 communications experience less latency as the establishment of TCP connections presents considerable overhead. richer meta-information.1 compliant. and in Internet Explorer 3. The first version of the protocol had only one method.[4] The first documented version of HTTP was HTTP V0. The HTTP WG planned to publish new standards in December 1995[8] and the support for pre-standard HTTP/1.1 can reuse a connection multiple times. which would request a page from a server. on the Internet. namely GET.[3] The response from the server was always an HTML page. extended negotiation. Lynx 2. By March 1996. or some other information.1 based on the then developing RFC 2068 (called HTTP-NG) was rapidly adopted by the major browser developers in early 1996. to download.0.[9] Netscape 2. In March 1996. HTTP/1. An HTTP server listening on that port waits for a client's request message. more specifically. pre-standard HTTP/1.Hypertext Transfer Protocol HTTP Resources are identified and located on the network by Uniform Resource Identifiers (URIs)—or.7. such as "HTTP/1. called hypertext documents.1 200 OK".0 uses a separate connection to the same server for every request-response transaction. Dave Raggett led the HTTP Working Group (HTTP WG) in 1995 and wanted to expand the protocol extended operations. the body of which is perhaps the requested resource.1 standard were released under RFC 2616 in June 1999. one web hosting company reported that over 40% of browsers in use on the Internet were HTTP 1. Upon receiving the request. The original version of HTTP (HTTP/1. for instance.[6] [7] RFC 1945 officially introduced and recognized HTTP V1. the server sends back a status line. Improvements and updates to the HTTP/1. form a system of inter-linked resources.0. that led to the establishment of the World Wide Web in 1990 by English computer scientist and innovator Tim Berners-Lee. see List of TCP and UDP port numbers). 62 History The term HyperText was coined by Ted Nelson who in turn was inspired by Vannevar Bush's microfilm-based "memex". images for a just delivered page. Tim Berners-Lee first proposed the "WorldWideWeb" project — now known as the World Wide Web.[10] The HTTP/1.1.5.9 [5] (1991).0) was revised in HTTP/1.01. An HTTP client initiates a request by establishing a Transmission Control Protocol (TCP) connection to a particular port on a server (typically port 80.1 standard as defined in RFC 2068 was officially released in January 1997.0 in 1996. 65% of all browsers accessing their servers were HTTP/1. while HTTP/1. HTTP session An HTTP session is a sequence of network request-response transactions. tied with a security protocol and got more efficient by adding additional methods and header fields. Hence HTTP/1. and a message of its own.[1] . URIs and the Hypertext Markup Language (HTML).

such as GET /images/logo.0 specification in RFC1945 [12].g.[11] Although <CR><LF> is required <LF> alone is also accepted by most servers. depends on the implementation of the server.1. A request line containing only the path name is accepted by servers to maintain compatibility with HTTP clients before the HTTP/1. In the HTTP/1. OPTIONS An HTTP request made using telnet. such as Accept-Language: en • An empty line. but without the response body. so that a client can see what (if any) changes or additions have been made by intermediate servers. Requests using GET (and a few other HTTP methods) "SHOULD NOT have the significance of taking an action other than retrieval". The request line and headers must all end with <CR><LF> (that is. a carriage return followed by a line feed). but also by the relevant limitations. GET Requests a representation of the specified resource.1 protocol.png from server • Headers. The request. "Web application design should be informed by the above principles. response headers and response body are highlighted. This may result in the creation of a new resource or the updates of existing resources or both. saying.Hypertext Transfer Protocol 63 Request message The request message consists of the following: • Request line. from an HTML form) to the identified resource. TRACE Echoes back the received request.. The data is included in the body of the request. all headers except Host are optional. This is useful for retrieving meta-information written in response headers. without having to transport the entire content. the resource corresponds to a file or the output of an executable residing on the server.[1] The W3C has published guidance principles on this distinction. which requests a resource called /images/logo.png HTTP/1. Often. whether pre-existing data or data that is generated dynamically. • An optional message body."[14] See safe methods below. The empty line must consist of only <CR><LF> and no other whitespace. .[13] Request methods HTTP defines nine methods (sometimes referred to as "verbs") indicating the desired action to be performed on the identified resource. DELETE Deletes the specified resource. What this resource represents. POST Submits data to be processed (e. HEAD Asks for the response identical to the one that would correspond to a GET request. PUT Uploads a representation of the specified resource.

or external side effects such as financial transactions or transmission of email. caching. meaning that multiple identical requests should have the same effect as a single request.Hypertext Transfer Protocol Returns the HTTP methods that the server supports for specified URL. Security software tools such as Tenable Nessus and Microsoft URLScan report on the presence of these methods as being security issues. if a user agent assumes that repeating the same request is safe when it isn't. This is discouraged. usually to facilitate SSL-encrypted communication (HTTPS) through an unencrypted HTTP proxy. Note that whether a method is idempotent is not enforced by the protocol or web server. whenever possible. search engines and other automated agents. being prescribed as safe. which can make unintended changes on the server. beyond relatively harmless effects such as logging. because they can be used by attackers to gather information or bypass security controls during attacks. 64 Safe methods Some methods (for example. because it can cause problems for Web caching. as HTTP is a stateless protocol. . may result in undesirable consequences. also the OPTIONS method. which means they are intended only for information retrieval and should not change the state of the server. This can be used to check the functionality of a web server by requesting '*' instead of a specific resource. Furthermore. careless or deliberate programming can cause non-trivial changes on the server. HEAD. the POST method is not necessarily idempotent. and therefore sending an identical POST request multiple times may further affect state or cause further side effects (such as financial transactions). the serving of banner advertisements or incrementing a web counter. however.[1] In contrast. HEAD. TRACK and DEBUG are considered potentially 'unsafe' by some security professionals. It is perfectly possible to write a web application in which (for example) a database insert or other non-idempotent action is triggered by a GET or other request. but in other cases this could be due to an accident. such as when a user does not realize that their action will result in sending another request. should also be idempotent. In some cases this may be desirable. or they did not receive adequate feedback that their first request was successful. Ignoring this recommendation. Idempotent methods and web applications Methods PUT and DELETE are defined to be idempotent. GET. In other words.[15] PATCH Is used to apply partial modifications to a resource. some that do not conform tend to make requests without regard to context or consequences. Therefore. While web browsers may show alert dialog boxes to warn users in some cases where reloading a page may re-submit a POST request. OPTIONS and TRACE) are defined as safe. Despite the prescribed safety of GET requests. OPTIONS and TRACE. PUT and DELETE are intended for actions that may cause side effects either on the server. they should not have side effects. Making arbitrary GET requests without regard to the context of the application's state should therefore be considered safe. Such methods are therefore not usually used by conforming web robots or web crawlers. in practice their handling by the server is not technically limited in any way. By contrast. CONNECT Converts the request connection to a transparent TCP/IP tunnel. methods such as TRACE. it is generally up to the web application to handle cases where a POST request should not be submitted more than once.[16] HTTP servers are required to implement at least the GET and HEAD methods[17] and. methods such as POST. Methods GET.

/index. it can use the first digit of the code to determine the general class of the response. Example session Below is a sample conversation between an HTTP client and an HTTP server running on www.com␍␊ ␍␊ .0. the standard reason phrases are only recommendations and can be replaced with "local equivalents" at the web developer's discretion. which is when a server transmits just the portion of a resource explicitly requested by a client. The way the user agent handles the response primarily depends on the code and secondarily on the response headers. A common solution is the use of HTTP cookies. where a connection could be reused for more than one request.example. so HTTP Secure is the dominant method of establishing a secure HTTP connection. Version 1.com. the web application may have to track the user's progress from page to page. In HTTP/1.[18] Also. Custom status codes can be used since.Hypertext Transfer Protocol 65 Status codes In HTTP/1. because the client does not need to re-negotiate the TCP connection after the first request has been sent. A stateless protocol does not require the server to retain information or status about each user for the duration of multiple requests. HTTP pipelining further reduces lag time. e. Other methods include server side sessions. For example. though this might be unwise since the standard explicitly specifies that status codes are machine-readable and reason phrases are human-readable. however.1 of the protocol made bandwidth optimization improvements to HTTP/1.php?session_id=some_unique_session_code. the user agent might display the reason phrase to the user to provide further information about the nature of the problem.9 and 1.g. Another improvement to the protocol was byte serving. hidden variables (when the current page is a form). nearly non-existent.1 introduced chunked transfer encoding to allow content on persistent connections to be streamed.0.. allowing clients to send multiple requests before a previous response has been received to the first one. HTTP/1. and URL-rewriting using URI-encoded parameters.1␍␊ Host: www.1 a keep-alive-mechanism was introduced. The standard also allows the user agent to attempt to interpret the reason phrase. For example. Persistent connections In HTTP/0.0 and since. the connection is closed after a single request/response pair.html HTTP/1. port 80. Such persistent connections reduce request latency perceptibly. rather than buffered.example. Browser support for the latter two is. if the user agent encounters a code it does not recognize. Client request GET /index. when a web server is required to customize the content of a web page for a user. Secure HTTP There are three methods of establishing a secure HTTP connection: HTTP Secure.1 Upgrade header. HTTP session state HTTP is a stateless protocol. the first line of the HTTP response is called the status line and includes a numeric status code (such as "404") and a textual reason phrase (such as "Not Found"). If the status code indicated a problem. Secure Hypertext Transfer Protocol and the HTTP/1.

The HTTP/1. Roy T. shtml). it means that the web server will close the TCP connection immediately after the transfer of this response. This is useful. Dave. World Wide Web Consortium.4. org/ pub/ WWW/ Protocols/ HTTP/ AsImplemented. html). [8] Raggett. html#sec1. .0. which is called byte serving. . Berners-Lee.1" (http:/ / www. org/ People/ Raggett/ profile.. Masinter. "Progress on HTTP-NG" (http:/ / www. each in the form of a carriage return followed by a line feed.3. org/ Arena/ webworld/ httpwgplans." (http:/ / www. org/ html/ rfc2616). w3. Mogul. Jeffrey C. Chunked transfer encoding uses a chunk size of 0 to mark the end of the content. "Hypertext Transfer Protocol Working Group" (http:/ / www. w3. "HTTP WG Plans" (http:/ / www.. Identity encoding without Content-Length reads content until the socket is closed. "RFC 2616: Hypertext Transfer Protocol -. . [7] Raggett. Internet RFC 2616. html). w3. org/ Arena/ webworld/ httpwgcharter. while Content-Length indicates its length in bytes. org/ History/ 19921103-hypertext/ hypertext/ WWW/ Protocols/ HTTP. A Content-Encoding like gzip can be used to compress the transmitted data. allowing name-based virtual hosting. html). 66 Server response HTTP/1. World Wide Web Consortium..1. org/ Protocols/ HTTP-NG/ http-ng-status. 4). . com/ glossary/ http1. Retrieved 24 July 2010. [11] Cailliau. Retrieved 29 September 2010. org/ rfc/ rfc1945. org/ Protocols/ rfc2616/ rfc2616-sec1. Most of the header lines are optional. James. Dave.com Glossary entry. html). Retrieved 1 September 2010. Dave. Paul J. World Wide Web Consortium. webcom. When Content-Length is missing the length is determined in other ways. [2] Fielding. org/ History/ 19921103-hypertext/ hypertext/ WWW/ Protocols/ HTTP/ HTTP2. Tim. charset=UTF-8 A server response is followed by a blank line and text of the requested page.Hypertext Transfer Protocol A client request (consisting in this case of the request line and only one header) is followed by a blank line.7 (Unix) (Red-Hat/Linux) Last-Modified: Wed. Retrieved 29 September 2010. Robert (1 July 1992). Content-Type specifies the Internet media type of the data conveyed by the http message. . 2009. Retrieved on January 21. 23 May 2005 22:38:34 GMT Server: Apache/1. html).1 webserver publishes its ability to respond to requests for certain byte ranges of the document by setting the header Accept-Ranges: bytes. section 1. "Dave Raggett's Bio" (http:/ / www. html [6] Raggett. Henrik Frystyk. World Wide Web Consortium. "Updates To HTTP" (http:/ / www. so that the request ends with a double newline. Retrieved 31 August 2010.1" (http:/ / tools. While optional in HTTP/1. w3. html). . it is mandatory in HTTP/1.HTTP/1. When Connection: close is sent in a header. html). [9] Simon Spero. if the client needs to have only certain portions[19] of a resource sent by the server. w3. Nielsen. [4] Tim Berners-Lee. 1. World Wide Web Consortium. "The Original HTTP as defined in 1991" (http:/ / www. [10] "HTTP/1. w3. World Wide Web Consortium. Gettys. . The "Host" header distinguishes between various DNS names sharing a single IP address. "HyperText Transfer Protocol" (http:/ / www.3. [5] http:/ / www. org/ Protocols/ HTTP/ AsImplemented. Leach. ietf. References [1] Fielding. Retrieved 11 June 2010. Webcom. Retrieved 2009-05-29. 08 Jan 2003 23:11:55 GMT Etag: "3f80f-1b6-3e1cb03b" Accept-Ranges: bytes Content-Length: 438 Connection: close Content-Type: text/html. Tim. . et al. [12] http:/ / www. w3. World Wide Web Consortium. The ETag (entity tag) header is used to determine if a cached version of the requested resource is identical to the current version of the resource on the server. ietf. Retrieved 11 June 2010. Berners-Lee (June 1999). Larry. [3] Berners-Lee. w3. w3. txt . .1 200 OK Date: Mon.

US-CERT.html). • "Design Issues for HTTP" (http://www. It is a part of the ITU-T H.w3. 1. Retrieved 2010-08-01.1 Section 5. .1" (http:/ / tools. Byte Range Retrieval Extension to HTTP 67 Further reading • HTTP 0. Technical Architecture Group finding. org/ vuls/ id/ 150227). Retrieved 2007-05-10.org/Protocols/DesignIssues. similar to the ISDN call model. James M.org. Within the context of H. Design Issues by Berners-Lee when he was designing the protocol. Ian (2004). ietf. and 3G mobile networks. w3.html). 1998-05-14. Retrieved 2010-08-01.html) External links • "Change History for HTTP" (http://www. W3.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. .w3. Addressability. com/ features/ http11). A detailed technical history of HTTP.323’s strength lies in multimedia communication functionality designed specifically for IP networks. html#checklist).323 standard addresses call signaling and control.323 call signaling is based on the ITU-T Recommendation Q. HTTP/1.. [15] "Vulnerability Note VU#150227: HTTP proxy default configurations allow arbitrary TCP connections" (http:/ / www.org. ietf. "RFC 5789: PATCH Method for HTTP" (http:/ / tools. an IP-based PBX might be a gatekeeper or other call control element which provides service to telephones or videophones.ietf.org/Protocols/History. . W3C.org. org/ 2001/ tag/ doc/ whenToUseGet. 2002-05-17.323. [18] "6. W3. Retrieved 2010-08-01. org/ html/ rfc5789). Retrieved 2010-08-01. W3. H. [19] Tools.ietf. 1).org/Protocols/Classic. org/ html/ draft-ietf-http-range-retrieval-00). H.1 Status-Line" (http:/ / www. "URIs.As Implemented in 1991 (http://www. W3. list of other classic documents recounting the early protocol history H.1. cert.Hypertext Transfer Protocol [13] "Apache Week.org.9 . 1). and QSIG over ISDN. [17] "HTTP 1. Snell. park.[1] It is widely implemented by voice and videoconferencing equipment manufacturers. such as call transfer. w3. html#sec6.931 protocol and is suited for transmitting calls across networks using a mixture of IP.1" (http:/ / www. 090502 apacheweek. and bandwidth control for point-to-point and multi-point conferences. is used within various Internet real-time applications such as GnuGK and NetMeeting and is widely deployed worldwide by service providers and enterprises for both voice and video services over IP networks.323 H. [16] Dusseault. • "Classic HTTP Documents" (http://www.323 excels at providing basic telephony functionality and interoperability.w3. apacheweek.com [14] Jacobs. Retrieved 2010-08-01. A call model. ietf. Lisa. Tools.html). pick-up. While H.org/Protocols/HTTP/AsImplemented. org/ html/ rfc2616#section-5.w3. eases the introduction of IP telephony into existing networks of ISDN-based PBX systems. . which also address multimedia communications over ISDN.org. PSTN.org (http:/ / tools. and hold. org/ Protocols/ rfc2616/ rfc2616-sec6. The H. including transitions to IP-based PBXs. . ISDN. kb. multimedia transport and control. and the use of HTTP GET and POST" (http:/ / www.32x series of protocols. Retrieved 26 September 2010. the PSTN or SS7. . . Such a device may provide or facilitate both basic services and supplementary services.

235 series describes security within H. In addition to those ITU-T Recommendations. but also the supplementary services needed to address business communication expectations. or text) between any two entities.323 was approved in 2009. H.323 systems also implement other protocols that are defined in various ITU-T Recommendations to provide supplementary services support or deliver other functionality to the user. H.323 is a system specification that describes the use of several ITU-T and IETF protocols.0 Registration.323 was being used for communication.225. is "Packet-Based Multimedia Communications Systems. but over WANs and within large carrier networks.460.323 was changed when published in 1998. H. including the Real-time Transport Protocol (RTP). but was quickly adopted by the industry as a means of transmitting voice communication over a variety of IP networks. H. and H.18. H.239 describes dual stream use in videoconferencing.[3] Recognizing that H.460.225. The protocols that comprise the core of almost any H. . Protocols H.19 for Network address translation (NAT) / Firewall (FW) traversal. which describes the messages and procedures used for capability exchange.323 entities in order to establish communication.323 was published by the ITU in November 1996[2] with an emphasis of enabling videoconferencing capabilities over a Local Area Network (LAN). • H. with each version being backward-compatible with the previous version. including ITU-T Recommendations H.245 control protocol for multimedia communication. not only defining the basic call model. which is used between any two H.323 68 History The first version of H.[4] The title. which has since remained unchanged. • Real-time Transport Protocol (RTP).323 system are:[6] • H.323 implements various IETF Request for Comments (RFCs) for media transport and media packetization. • H. including security for both signaling and media.323 was the first VoIP standard to adopt the Internet Engineering Task Force (IETF) standard Real-time Transport Protocol (RTP) to transport audio and video over IP networks. which is used between an H.323 endpoint and a Gatekeeper to provide address resolution and admission control services.17. Over the years. video and data. usually one for live video.323 has been revised and re-published with enhancements necessary to better-enable both voice and video functionality over Packet-switched networks. H.323.450 series describes various supplementary services. not only on LANs. the title of H. video.460 series defines optional extensions that might be implemented by an endpoint or a Gatekeeper." The current version of H.460. opening and closing logical channels for audio.323 was the relatively early availability of a set of standards. Admission and Status (RAS).0 Call Signaling. H.[5] One strength of H. including WANs and the Internet (see VoIP).H. which is used for sending or receiving multimedia information (voice. control and indications. Some of those Recommendations are: • • • • H. Many H. the other for still images.

H.323 Network Elements Terminals Terminals in an H. G.323 deployments. All terminals shall be capable of transmitting and receiving A-law and μ-law.711.723.323 equipment include: • Audio codecs: G.323 terminal is something referred to as a Protocol stack.225.323 terminals shall have an audio codec and shall be capable of encoding and decoding speech according to ITU-T Rec. In most H.323 system.264 All H.263. most H. as well as RTP or other protocols described above.726.261 QCIF.323 terminals providing video communications shall be capable of encoding and decoding video according to H.[5] H.711. video. and Border Elements. at least two terminals are required in order to enable communication between two people. Gatekeepers.323 utilizes both ITU-defined codecs and codecs defined outside the ITU. which implements the functionality defined by the H.722. Support for other audio and video codecs is optional.323 69 Codecs H.0 and H. figure 1. terminals.140 • Video codecs: H. While not all elements are required. G. depicts a complete. Those elements are Terminals. H. Inside an H. a gatekeeper is employed in order to.A complete. facilitate address resolution.323 system. G.1. sophisticated stack that provides support for voice.323 system defines several network elements that work together in order to deliver rich multimedia communication capabilities. In reality. They might exist in the form of a simple IP phone or a powerful high-definition videoconferencing system.729 (including G. H. and various forms of data communication. but the logical arrangement is useful in understanding the relationships.323 network are the most fundamental elements in any H. Speex • Text codecs: T. The diagram. Codecs that are widely implemented by H. Gateways. Multipoint Control Units (MCUs). The protocol Figure 1 . Collectively. G.729a).323 systems do not implement such a wide array of capabilities.728. All H. . G. H. as those are the devices that users would normally encounter. sophisticated protocol stack stack would include an implementation of the basic protocol defined in ITU-T Recommendation H. multipoint control units and gateways are often referred to as endpoints.245.261. among other things. G.323 Architecture The H.

323 network that provides a number of services to terminals. Most of the third generation (3G) mobile networks deployed today utilize the H. This collection of devices does not necessarily have to have an associated physical topology. Some MCUs also provide multipoint data collaboration capabilities.323-based terminals in corporate networks through such gateway devices. and so forth. the user might be able to see all of the other participants in the conference. endpoints utilize the RAS protocol in order to learn the IP address of the remote endpoint and a call is established directly with the remote device. In the gatekeeper routed mode. What this means to the end user is that. is that H. Gateways are also used within the enterprise in order to enable enterprise IP phones to communicate through the service provider to users on the PSTN. 70 . Gatekeepers A Gatekeeper is an optional component in the H.H. Those services include endpoint registration. Rather. A collection of endpoints that are registered to a single Gatekeeper in H. by placing a video call into an H. namely "direct routed" and "gatekeeper routed" mode. gatekeepers use RAS to communicate with other gatekeepers. Direct routed mode is the most efficient and most widely deployed mode. Of the various functions performed by the gatekeeper. Gatekeepers may be designed to operate in one of two signaling modes. an MCU is a conference bridge not unlike the conference bridges used in the PSTN today.323 Multipoint Control Units A Multipoint Control Unit (MCU) is responsible for managing multipoint conferences and is composed of two logical entities referred to as the Multipoint Controller (MC) and the Multipoint Processor (MP). international H. it also gives the gatekeeper complete control over the call and the ability to provide supplementary services on behalf of the endpoints. The most significant difference. Likewise. a zone may be entirely logical and is arbitrarily defined by the network administrator. In this mode.323 is referred to as a “zone”. user authentication.320 and H.324 to communicate with H. Gateways are widely used today in order to enable the legacy PSTN phones to interconnect with the large.323 systems.324 protocol and are able to communicate with H. address resolution.323 endpoints use the RAS protocol to communicate with a gatekeeper. and MCU devices.323 MCU.323 MCUs might be capable of mixing or switching video. address resolution is the most important as it enables two endpoints to contact each other without either endpoint having to know the IP address of the other endpoint. Gateways Gateways are devices that enable communication between H. however. admission control. in addition to the normal audio mixing done by a traditional conference bridge. not only hear their voices. If one party in a conversation is utilizing a terminal that is not an H. While the latter requires the gatekeeper to have more processing power. Gatekeepers have the ability to neighbor together so that call resolution can happen between zones. Dial plans facilitate “inter-zone” dialing so that two endpoints in separate zones can still communicate with each other. call signaling always passes through the gatekeeper.323 networks that are presently deployed by services providers.323 networks and other networks. such as PSTN or ISDN networks. Neighboring facilitates the use of dial plans such as the Global Dialing Scheme.323 terminal. H. gateways. In more practical terms. Gateways are also used in order to enable videoconferencing devices based on H. then the call must pass through a gateway in order to enable both parties to communicate.

telephones. and gatekeepers. and gatekeepers network there may be hundreds or thousands of gateway devices. Within a service provider elements.0 Call Signaling Once the address of the remote endpoint is resolved. H.225. The diagram. such as logical arrangement by city. the endpoint will use H. The border element is a signaling entity that generally sits at the edge of the administrative domain and communicates with another administrative domain.323 is defined as a binary protocol. peer elements. An administrative domain is the collection of all zones that are under the control of a single person or organization.An illustration of an administrative domain with border elements.0 messages are: • • • • Setup and Setup acknowledge Call Proceeding Connect Alerting • Information • Release Complete . Taken together. Such architecture is intended to enable large-scale deployments within carrier networks and to enable services such as clearinghouses. The role of a border or peer element is understood via the definition of an "administrative domain". call pricing information. but that do not manage endpoints directly and provide some services that are not described in the RAS protocol. Peer elements are entities within the administrative domain that. Below is an overview of the various communication flows in H. The syntax of the protocol is defined in ASN.0 Call Signaling in order to establish communication with the remote entity.323 Border Elements and Peer Elements Border Elements and Peer Elements are optional entities similar to a Gatekeeper.225. or other H.323 Network Signaling H.323 network elements. such as a service Figure 2 .1 and uses the Packed Encoding Rules (PER) form of message encoding for efficient message encoding on the wire. 71 H.225. provides an illustration of an administrative domain with border elements.H. H. help to propagate information learned from the border elements throughout the administrative domain. peer provider.323 systems. or other important data necessary to enable communication between the two administrative domains. all of the zones within the service provider network would appear to another service provider as an "administrative domain". figure 2. The service provider might arrange devices into "zones" that enable the service provider to best manage all of the devices under its control. which allows for efficient message processing in network elements. more or less. video terminals. This communication might include such things as access authorization information.

gatekeepers use RAS to communicate with peer gatekeepers. reject. RAS Signaling Endpoints use the RAS protocol in order to communicate with a gatekeeper. reject. Endpoints are then required to notify their gatekeeper (if gatekeepers are used) that the call has ended. the endpoint (EP) on the left initiated communication with the gateway on the right and the gateway connected the call with the called party. but most calls that utilize the Fast Connect procedures defined within H. reject. reject. and confirm messages (RRx) Unregister request.323 call may be established as follows (figure 3): In this example. and confirm messages (ARx) Bandwidth request. and confirm messages (LRx) Info request. reject. call flows are often more complex Figure 3 . nack. Once a call has concluded. a device will send a Release Complete message. reject.323 call than the one shown.323 can be established with as few as 2 or 3 messages. and response (IRx) Nonstandard message Unknown message response Request in progress (RIP) Resource availability indication and confirm (RAx) Service control indication and response (SCx) Admission confirm sequence (ACS) . In reality. reject. and confirm messages (URx) Admission request.H.323 • • • • Facility Progress Status and Status Inquiry Notify 72 In the simplest form. and confirm messages (GRx) Registration request. an H. ack.Establishment of an H. Endpoints must notify their gatekeeper (if gatekeepers are used) that they are in a call. RAS is a fairly simple protocol composed of just a few messages. Likewise. and confirm (DRx) Location request. Namely: • • • • • • • • • • • • • • Gatekeeper request. and confirm message (BRx) Disengage request.

323 73 When an endpoint is powered on. Upon receiving a call. H.245 is a rather voluminous specification with many procedures that fully enable multipoint communication. a remote endpoint will also send an ARQ and receive an ACF in order to get permission to accept the incoming call. to which the gatekeeper responds with a registration confirm (RCF).245 also enables real-time data conferencing capability through protocols like T. including audio. by consulting another gatekeeper. H. Figure 4 depicts a high-level communication exchange between two endpoints (EP) and two gatekeepers (GK). For transmission of audio. but are integrated to provide the user with a seamless multimedia experience. Capability Negotiation Of the functionality provided by H.323 system.711. .245 provides capabilities such as capability negotiation. H. audio and video flows). It has support for both unicast and multicast communication. H. opening and closing of "logical channels" (i. Codecs that are widely implemented by H. master/slave determination. At this point. for example. it will send an admission request (ARQ) to the gatekeeper.126.1. H.723.245 Call Control Once a call has initiated (but not necessarily fully connected) endpoints may initiate H. to authenticate the calling device or to ensure that there is available bandwidth for the call. The endpoint can then place the call.264 • Audio codecs: G.e. G.245. This is necessary.726 • Text codecs: T. text. capability negotiation is arguably the most important. H. H. and conference control. If a GRQ has been Figure 4 .261. or text. though in practice most implementations only implement the minimum necessary in order to enable point-to-point voice and video communication. and data communication.120. When an endpoint wishes to place a call. G. The gatekeeper will then resolve the address (either locally. T. Gatekeepers will then respond with a gatekeeper confirm (GCF). video.729.120-based applications generally operate in parallel with the H.323 devices utilize both ITU-defined codecs and codecs defined outside the ITU.245 call control signaling in order to provide more extensive control over the conference. flow control. video. it will generally send either a gatekeeper request (GRQ) message to "discover" gatekeepers that are willing to provide service or will send a registration request (RRQ) to a gatekeeper that is predefined in the system’s administrative setup.A high-level communication exchange between two endpoints (EP) and two sent the endpoint will then select a gatekeepers (GK) gatekeeper with which to register by sending a registration request (RRQ). allowing the size of a conference to theoretically grow without bound.263.323 equipment include: • Video codecs: H.128.245 enables rich multimedia capabilities.729a.120 provides such capabilities as application sharing T. as it enables devices to communicate without having prior knowledge of the capabilities of the remote entity.. T. G.140 H. the endpoint is known to the network and can make and place calls. G. or by querying some other network service) and return the address of the remote endpoint in the admission confirm message (ACF). electronic whiteboard T.H.

134 within the context of the conference. The number of message exchanges is numerous. Fast Connect is widely supported in the industry. it is possible to establish a call with bi-directional media flowing with no more than two messages. Upon receipt of the acknowledgement message. and text chat T. VoIP Phone or "soft phone"). When an H. Master/Slave Determination After sending a TCS message.A typical H. Using VoIP between two enterprise locations would not necessarily require a VoIP service provider.323 entities (through H. For this reason. devices may then open "logical channels" or media flows.245 communication is established between the two entities." This process. each has an important purpose. Logical Channel Signaling Once capabilities are exchanged and master/slave determination steps have completed. Fast Connect A typical H. Most service providers charge a monthly fee. then additional costs when calls are made. the two endpoints (EP) in this figure would be transmitting audio in each direction.245 exchanges) will attempt to determine which device is the "master" and which is the "slave. 74 Use cases H. the Terminal Capability Set (TCS) message is the first message transmitted to the other side. Even so.323 device and when H. referred to as Master/Slave Determination (MSD). so there is no noticeable delay to the calling or called party.0 call establishment procedures. it is the master who takes the action to reject the incompatible flow. most devices still implement the complete H. For example.H. H. an endpoint may then transmit audio or video to the remote endpoint.245 exchange bi-directional media flows as part of the H. H. as the master in a call settles all "disputes" between the two devices. .225.323 file transfer T. VoIP requires a connection to the Internet or another packet switched network. like in figure 3. which enables a device to establish Figure 5 .245 exchange as shown above and performs that message exchange in parallel to other activities. With Fast Connect.323 and Voice over IP services Voice over Internet Protocol (VoIP) describes the transmission of voice using the Internet or other packet switched networks. a subscription to a VoIP service provider and a client (an analogue telephone adapter (ATA).323 version 2 (published in 1998) introduced a concept called Fast Connect. ITU-T Recommendation H. but nonetheless takes time. for example. The service provider offers the connection to other VoIP services or to the PSTN. This is done by simply sending an Open Logical Channel (OLC) message and receiving an acknowledgement message.245 exchange looks similar to figure 5: After this exchange of messages. if both endpoints attempt to open incompatible media flows.323 is one of the standards used in VoIP. is important.323 device initiates communication with a remote H.127.

int/ rec/ T-REC-H. [3] ITU-T Recommendation H. • ITU-T Recommendation H.225. int/ rec/ T-REC-H. H. 75 H. "H. proprietary formats.245/en/).323 System for a detailed list. 229–230. .246 (http://www. in force. distance education.[7] References [1] Davidson. there has been a strong spurt of growth and use of H.itu.323 and Videoconference services A videoconference. int/ rec/ T-REC-H. com/ books?id=S5P7-Xtq7W8C& pg=PA229). and includes H.H. itu. Call signalling protocols and media stream packetization for packet-based multimedia communication systems. [4] ITU-T Recommendation H.323 has been used in the industry to enable large-scale international video conferences that are significantly larger than the typical video conference. or videoteleconference (VTC). pp. itu. and broadband in particular. ITU-T Recommendations of the H. ISBN 9781578701681.itu. Interworking of H-series multimedia terminals with H-series multimedia terminals and voice/voiceband terminals on GSTN and ISDN. 323/ en/ ). [6] See ITU-T Recommendations of the H.323. [2] ITU-T Recommendation H.0/en/). [7] "EVO The Collaboration Network" (http:/ / evo. Control protocol for multimedia communication.323 (02/1998) (http:/ / www. telemedicine and business.323" (http:/ / books. There are MCU bridges for IP and ISDN-based videoconferencing. Due to the price point and proliferation of the Internet. . dedicated VTC systems have all required components packaged into a single piece of equipment while desktop VTC systems are add-ons to normal PC's.int/rec/T-REC-H.323 System ITU-T H. There are basically two types of videoconferencing. Access Grid provides broadly similar functionality.323 has been widely deployed by companies who wish to interconnect remote locations over IP using a number of various wired and wireless technologies. Packet-based multimedia communications systems.323 (12/2009) (http:/ / www. [5] ITU-T Recommendation H. Brian Gracely.323/en/).itu.323 support. first version of H. Packet-based multimedia communications systems. James Peters. 323-199802-S/ en/ ). edu/ evoGate). such as DSL. Jim Peters. Simultaneous videoconferencing among three or more remote points is possible by means of a Multipoint Control Unit (MCU).225.323 (11/1996) (http:/ / www.0 (http://www. Cisco Press.int/rec/T-REC-H. itu. 323). caltech. Packet-based multimedia communications systems. International Conferences H. is a set of telecommunication technologies allowing two or more locations to interact via two-way video and audio transmissions simultaneously. • ITU-T Recommendation H. Alternatives The IETF produced a standard called the Session Initiation Protocol (SIP) that also enables voice and video communication over IP.246/en/). • ITU-T Recommendation H.H. superseded and withdrawn component. Some providers (such as Skype) also use their own closed.323 (http://www. itu.int/rec/T-REC-H.int/rec/T-REC-H. with more emphasis on open-source and utilizing multicast. 323-199611-S/ en/ ).324 (using regular analog phone lines and 3G mobile phones). for example.323 (http:/ / www.323 Core Recommendations • ITU-T Recommendation H. Videoconferencing is utilized in various situations. Retrieved 2010-03-08.323-based IP videoconferencing. transforming them into VTC devices. int/ rec/ T-REC-H. One of the most widely attended is an annual event called Megaconference. Voice over IP fundamentals. Jonathan. There are also other ITU-T recommendations used for videoconferencing and videophone services .320 (using ISDN) and H.245 (http://www. . EVO also provides relatively open functionality via Java. google.itu.323 H.323 is accessible to anyone with a high speed Internet connection.

450.itu. H.450.323 security: Hybrid security profile.int/rec/T-REC-H.450.323.235. H. H. • ITU-T Recommendation H.450.int/rec/T-REC-H.6/en/).323 security framework: Voice encryption profile with native H.4/en/).7/en/).450.itu.int/rec/T-REC-H.5/en/).1 (http://www.itu.9/en/).235. H. • ITU-T Recommendation H.itu.450.235.235 Series Recommendations • ITU-T Recommendation H.323 security framework: Usage of the MIKEY key management protocol for the Secure Real Time Transport Protocol (SRTP) within H.6 (http://www.235.2 (http://www.itu.itu.int/rec/T-REC-H.itu. • ITU-T Recommendation H.450.235.int/rec/T-REC-H. • ITU-T Recommendation H.10 (http://www. • ITU-T Recommendation H.int/rec/T-REC-H. • ITU-T Recommendation H.323.450.450. • ITU-T Recommendation H.283/en/).235.5 (http://www.9/en/). 76 .323.323.235. • ITU-T Recommendation H.9 (http://www.235.itu. • ITU-T Recommendation H.4 (http://www.itu.245 key management.3/en/).450.450 Series Recommendations • ITU-T Recommendation H.int/rec/T-REC-H.450. Call hold supplementary service for H.450.323.int/rec/T-REC-H.7 (http://www.int/rec/T-REC-H. • ITU-T Recommendation H.6/en/).int/rec/T-REC-H.235.2/en/).323 security: Security gateway support for H. Multimedia management information base.itu. Call waiting supplementary service for H. • ITU-T Recommendation H.itu.H.itu. Generic functional protocol for the support of supplementary services in H. • ITU-T Recommendation H.itu.450. Call completion supplementary service for H.4 (http://www. Remote device control logical channel transport. • ITU-T Recommendation H.323.235.235.int/rec/T-REC-H.int/rec/T-REC-H.323 security: Framework for secure authentication in RAS using weak shared secrets.itu.8/en/). H.450.3 (http://www.itu.450.itu.int/rec/T-REC-H. Call park and call pickup supplementary service for H.323 security framework: Baseline security profile.1 (http://www.2 (http://www.int/rec/T-REC-H.323. ITU-T H.450.3/en/).323.int/rec/T-REC-H.5 (http://www.235.450. H.235.235.2/en/).235.323.323 • ITU-T Recommendation H. H.235.235. H.4/en/). Call diversion supplementary service for H.235.itu. Call offering supplementary service for H.int/rec/T-REC-H.450.1/en/).7/en/).283 (http://www.9 (http://www.int/rec/T-REC-H. Call transfer supplementary service for H. Name identification supplementary service for H.450.itu.323.6 (http://www.itu.323 security: Direct and selective routed call security.1/en/).235/H.323 security framework: Signature security profile. ITU-T H.int/rec/T-REC-H.341/en/). • ITU-T Recommendation H.235.7 (http://www.8/en/). Message waiting indication supplementary service for H.itu. H.323.323 security: Key exchange for SRTP using secure signalling channels.itu. • ITU-T Recommendation H.int/rec/T-REC-H.int/rec/T-REC-H.5/en/).int/rec/T-REC-H. • ITU-T Recommendation H. • ITU-T Recommendation H.8 (http://www.8 (http://www.10/en/).341 (http://www. • ITU-T Recommendation H.3 (http://www.450.

14/en/). H.1 (http://www.12/en/).int/rec/T-REC-H.323.itu. • ITU-T Recommendation H.323 systems. Multiple message release sequence capability. • ITU-T Recommendation H.9/en/).460.11 (http://www. Call priority designation and country/international network of call origination identification for H. • ITU-T Recommendation H.323 systems. Traversal of H.11 (http://www.460.19 (http://www.460.5/en/).460.int/rec/T-REC-H.itu.1/en/).2 (http://www.19/en/).460.20 (http://www.itu. Using H.itu.4/en/).460.460.323 and SCN networks.2/en/).int/rec/T-REC-H. • ITU-T Recommendation H. • ITU-T Recommendation H. Circuit maps within H.itu. Call intrusion supplementary service for H.int/rec/T-REC-H. 77 .460.12 (http://www.460.int/rec/T-REC-H.460. Called user release control.itu.int/rec/T-REC-H.460.itu.460.18/en/).450.int/rec/T-REC-H.21/en/).460.460.int/rec/T-REC-H.225.18 (http://www.16 (http://www.323 systems. Glare control indicator within H.8 (http://www. ITU-T H.itu.460.460.460.itu.itu.931 information elements of the same type.int/rec/T-REC-H. Guidelines for the use of the generic extensible framework.itu.int/rec/T-REC-H.323 systems.460.460.10/en/).itu.int/rec/T-REC-H. Location number within H.3/en/).323.323 signalling across network address translators and firewalls.8/en/).itu.0 transport of multiple Q. • ITU-T Recommendation H.12/en/).9 (http://www. Support for online QoS-monitoring reporting within H.460.323 systems.int/rec/T-REC-H.450.20/en/).int/rec/T-REC-H. Call signalling transport channel suspension and redirection within H. Traversal of H.15 (http://www.itu.460.int/rec/T-REC-H.7 (http://www.11/en/).H.323 priority calls.itu.12 (http://www.itu.int/rec/T-REC-H.itu.int/rec/T-REC-H.460.323 • ITU-T Recommendation H.460.int/rec/T-REC-H.460. Number Portability interworking between H. • ITU-T Recommendation H.int/rec/T-REC-H.10 (http://www. Querying for alternate routes within H. • ITU-T Recommendation H.225.itu.4 (http://www.int/rec/T-REC-H.323 systems. • ITU-T Recommendation H. Delayed call establishment within H.460. • ITU-T Recommendation H.16/en/).323 systems.15/en/).int/rec/T-REC-H.17/en/). Call party category within H.460. Digit maps within H.itu.6 (http://www.460.13/en/).460.460.460. • ITU-T Recommendation H.460 Series Recommendations • ITU-T Recommendation H.7/en/).itu.460.460.450.int/rec/T-REC-H.6/en/). • ITU-T Recommendation H. • ITU-T Recommendation H.11/en/).itu.3 (http://www.460.itu. • ITU-T Recommendation H.460. Message broadcast for H.460.323 systems.0 call signalling connection as transport for H. Common Information Additional Network Feature for H.323 systems. • ITU-T Recommendation H. • ITU-T Recommendation H.itu.323 systems.460. • ITU-T Recommendation H.323 media across network address translators and firewalls.int/rec/T-REC-H.460. Support for Multi-Level Precedence and Preemption (MLPP) within H. • ITU-T Recommendation H.323 RAS messages. Extended Fast Connect feature.5 (http://www.450.460.int/rec/T-REC-H.460. • ITU-T Recommendation H.13 (http://www.17 (http://www.460. • ITU-T Recommendation H.14 (http://www.21 (http://www. • ITU-T Recommendation H.460.323 systems.460.

H.323 • ITU-T Recommendation H.460.22 (http://www.itu.int/rec/T-REC-H.460.22/en/), Negotiation of security protocols to protect H.225.0 Call Signalling Messages. ITU-T H.500 Series Recommendations • ITU-T Recommendation H.501 (http://www.itu.int/rec/T-REC-H.501/en/), Protocol for mobility management and intra/inter-domain communication in multimedia systems. • ITU-T Recommendation H.510 (http://www.itu.int/rec/T-REC-H.510/en/), Mobility for H.323 multimedia systems and services. • ITU-T Recommendation H.530 (http://www.itu.int/rec/T-REC-H.530/en/), Symmetric security procedures for H.323 mobility in H.510.

78

External links
General
• • • • H.323 Definition and overview (http://www.iec.org/online/tutorials/h323/index.html) H.323 Forum (http://www.h323forum.org/) H.323 Information site (http://www.packetizer.com/ipmc/h323/) H.323 Tutorial and resources (http://www.telecomspace.com/vop-h323.html)

• Implementing H.323 (Zip) (http://hive.packetizer.com/users/h323forum/papers/basic_h323_von_99.zip)

Papers
• H.323 Protocol Overview (technical) (http://hive.packetizer.com/users/packetizer/papers/h323/ h323_protocol_overview.pdf) • H.323 Overview (less technical) (http://hive.packetizer.com/users/packetizer/papers/h323/ overview_of_h323.pdf) • H.323 Call flow covering H.225, Q.931, H.245, RTP and RTCP protocols (PDF) (http://www.eventhelix.com/ RealtimeMantra/Telecom/h323_call_flow.pdf) • H.323 Call flow (communication example) (http://www.en.voipforo.com/H323/H323_example.php) • H.323 List of papers and presentations (http://www.h323forum.org/papers/)

Projects
• • • • H.323 Plus open source H.323 project (http://www.h323plus.org/) Xmeeting for Mac OS X (http://xmeeting.sourceforge.net/) GNU (OpenSource) Gatekeeper (http://www.gnugk.org/) Ekiga: open source VoIP and video conferencing application for GNOME (http://www.ekiga.org/)

H.225.0

79

H.225.0
H.225.0 is part of the H.323 family of telecommunication protocols. The main objective of H.225.0 is the definition of messages and procedures for: • Call Signaling: establish, control and end an H.323 call. The call signaling of H.225.0 is based on the call setup procedures for ISDN, Recommendation Q.931. • RAS Signaling Function: perform registration, admission, bandwidth changes, status and disengage procedures between endpoints and an H.323 Gatekeeper. The RAS signaling function uses a separate channel (RAS channel). This set of messages is called Registration, Admission and Status (RAS). • Additionally, H.225.0 defines media packetization using RTP for traditional codecs like G.711, H.261 and H.263. Newer ITU recommendations for media compression/signaling refer to RFCs for packetization (e.g. H.264 utilizes the single NAL unit mode of RFC 3984 in H.323 as defined in H.241). The Message Encoding is performed by utilizing Q.931 User-to-User information elements to convey H.225 messages. The messages themselves are encoded according to the Packed Encoding Rules (PER) of ASN.1. The structure of H.225 follows the Q.931 standard as seen in the table below.
8 7 6 5 4 3 2 1 Octet 1 2 3 (-4)

Protocol Discriminator 0 0 0 0 Length of call reference bits Call reference value 0 Message type Information Elements H.225 structure

External links
• ITU-T H.225.0 Recommendation [1]

References
[1] http:/ / www. itu. int/ rec/ T-REC-H. 225. 0/ en

Skinny Call Control Protocol

80

Skinny Call Control Protocol
The Skinny Call Control Protocol (SCCP, or short Skinny) is a proprietary network terminal control protocol originally developed by Selsius Systems. The SCCP technology is now owned and defined by Cisco Systems, Inc.. SCCP is a lightweight protocol for session signaling with Cisco CallManager.[1] Examples of SCCP clients include the Cisco 7900 series of IP phones, Cisco IP Communicator softphone and the 802.11b wireless Cisco 7920, along with Cisco Unity voicemail server. CallManager acts as a signaling proxy for call events initiated over other common protocols such as H.323, Session Initiation Protocol (SIP), ISDN and/or MGCP.

Protocol components
A SCCP client uses TCP/IP to communicate with one or more Call Manager applications in a cluster. It uses the Real-time Transport Protocol (RTP) over UDP-transport for the bearer traffic (real-time audio stream) with other Skinny clients or an H.323 terminal. SCCP is a stimulus-based protocol and is designed as a communications protocol for hardware endpoints and other embedded systems, with significant CPU and memory constraints.

Origin
Cisco acquired SCCP technology when it acquired Selsius Corporation in 1998.[2] As a remnant of the Selsius origin of the current Cisco IP phones, the default device name format for registered Cisco phones with CallManager is SEP -- as in Selsius Ethernet Phone -- followed by the MAC address. Cisco also has marketed a Skinny-based softphone called Cisco IP Communicator.

Other implementations
Other companies like Symbol Technologies, SocketIP, and Digium have implemented this protocol in VoIP Terminals (phones), Media Gateway Controllers, and Softswitches. An open source implementation is available in the Asterisk (PBX) & FreeSWITCH[3] systems. IPBlue[4] markets a software phone (soft phone) which uses SCCP for signaling and emulates a Cisco 7960 hardware phone. The Cisco Unified Application Environment, the product acquired by Cisco when they purchased Metreos supports using SCCP to emulate Cisco 7960 hardware phones allowing applications to access all Cisco line-side features.

References
[1] Call states sent to SCCP endpoints by Cisco CallManager (http:/ / www. cisco. com/ en/ US/ products/ sw/ voicesw/ ps556/ products_tech_note09186a0080624977. shtml) [2] Cisco Systems acquisition of Selsius Systems (http:/ / newsroom. cisco. com/ dlls/ fspnisapi6d14. html) [3] http:/ / wiki. freeswitch. org/ wiki/ Mod_skinny [4] http:/ / www. ipblue. com/ IPBlue

External links
• Skinny Call Control Protocol (SCCP) - Cisco Systems (http://www.cisco.com/en/US/tech/tk652/tk701/ tk589/tsd_technology_support_sub-protocol_home.html) • Skinny Call Control Protocol (SCCP) - Fragment of a VoIP course dealing with SCCP, with sample captures (http://hisown.com/Talks/VoIP - Excerpt (SCCP).pdf) • Alternative SCCP channel driver for Asterisk. Sourceforge Project called chan-sccp-b (http://chan-sccp-b. sourceforge.net)

an identifier is used when needed to track concurrent sessions. and another for RTCP data (meta information). In the typical case. and the type of reply data that can be handled. some commands travel in the other direction (i. The protocol is used for establishing and controlling media sessions between end points. are also available. A range parameter on a PAUSE request specifies when to pause. or a single media stream URL (to play only that stream). Among other things. The transmission of streaming data itself is not a task of the RTSP protocol. such as play and pause. The server reply usually confirms the chosen parameters. the stream is played from the beginning and plays to the end. This specifier typically includes a local port for receiving RTP data (audio or video). If no range is specified. typically in Session Description Protocol (SDP) format. RTSP has state. Clients of media servers issue VCR-like commands. there is one media stream each for audio and video. like the OPTIONS request. The request contains an aggregate or media stream URL. SETUP A SETUP request specifies how a single media stream must be transported.). The default transport layer port number is 554. PLAY A PLAY request will cause one or all media streams to be played.e. DESCRIBE A DESCRIBE request includes an RTSP URL (rtsp://. . for example. RTSP was developed by the Multiparty Multimedia Session Control Working Group (MMUSIC WG) of the Internet Engineering Task Force (IETF) and published as RFC 2326 in 1998. OPTIONS An OPTIONS request returns the request types the server will accept. The default port for the RTSP protocol is 554 for both UDP (deprecated and very rarely used) and TCP transports. RTSP uses TCP to maintain an end-to-end connection and. also features RealNetworks' proprietary RDT stream transport. A range can be specified. the pause occurs immediately and indefinitely. from server to client).Real Time Streaming Protocol 81 Real Time Streaming Protocol The Real Time Streaming Protocol (RTSP) is a network control protocol designed for use in entertainment and communications systems to control streaming media servers.[1] Protocol directives While similar in some ways to HTTP.. however some vendors implement proprietary transport protocols. Presented here are the basic RTSP requests. the presentation description lists the media streams controlled with the aggregate URL. Play requests can be stacked by sending multiple PLAY requests. such as the server's chosen ports.. The URL may be the aggregate URL (to play all media streams). Most RTSP servers use the Real-time Transport Protocol (RTP) for media stream delivery. to facilitate real-time control of playback of media files from the server. so it can later be resumed with a PLAY request. RTSP defines control sequences useful in controlling multimedia playback. PAUSE A PAUSE request temporarily halts one or all media streams. When the range parameter is omitted. Like HTTP. This reply includes the presentation description. Each media stream must be configured using SETUP before an aggregate play request may be sent. it is resumed at the point it was paused. This must be done before a PLAY request is sent. The RTSP server from RealNetworks. and fills in the missing parts. The request contains the media stream URL and a transport specifier. While HTTP is stateless. Some typical HTTP requests. while most RTSP control messages are sent by the client to the server. or. if the stream is paused.

TEARDOWN A TEARDOWN request is used to terminate the session. Comes in both open-source and proprietary flavors.20. Windows Media Services: Microsoft's streaming server included with Windows Server.3gp GStreamer Media Player Classic MPlayer MythTV via Freebox QuickTime RealPlayer Skype Spotify VLC media player • • • • • • • • • • Winamp • Windows Media Player • xine .youtube. FFmpeg: includes ffserver a GPL or LGPL RTSP streaming server. Helix Universal Server: RealNetworks commercial streaming server for RTSP. VX30: Streaming video server and embedded JAVA client from Maui X-Stream. It stops all media streams and frees all session related data on the server. Darwin Streaming Server: Open-sourced version of QuickTime Streaming Server maintained by Apple. • • • • • • • • • • Feng: Lean and mean streaming server with focus with rfc compliance. VideoLAN: Open source media player and streaming server. iPhone OS.com/some-media. Gstreamer based RTSP Server and client.3gp 2. ViaMotion : integrated RTSP server for Video On Demand by Anevia Client • cURL (beginning with version 7. Live playback: ffplay rtsp://some-server.com/some-media. Xenon Streaming Server: Mobile streaming server from Vidiator Technology (US) Inc. RtpRtspStack: Streaming server which is designed for low footprint and high performance applications. pvServer: Formerly called PacketVideo Streaming Server.youtube. this is Alcatel-Lucent's streaming server product.3gp -acodec copy -vcodec copy filename. 82 Implementations Server • • • • QuickTime Streaming Server: Apple's closed-source streaming server that ships with Mac OS X Server. Silverlight and HTTP streaming media clients • Helix DNA Server: RealNetworks' streaming server.0—9 February 2010[3] ) • FFmpeg (undocumented[4] [5] [6] [7] ) 1. • LIVE555: Open source C++ server and client libraries used in well known clients like VLC and mplayer. Erlyvideo[2] has RTSP client and can restream video to other protocols. RTMP.Real Time Streaming Protocol RECORD The RECORD request can be used to send a stream to the server for storage. Saving live broadcast to a file: ffmpeg -i rtsp://some-server.

html#SEC6) (as of 30 May 2010) that FFplay understands RTSP—found in one obscure mention: "`-rtp_tcp' Force RTP/TCP protocol usage instead of RTP/UDP.org/). RTP is used in conjunction with the RTP Control Protocol (RTCP). Archived from the original (http://www. It is only meaningful if you are streaming with the RTSP protocol.html). RTP was developed by the Audio-Video Transport Working Group of the Internet Engineering Task Force (IETF) and first published in 1996 as RFC 1889. "built on Dec 3 2007 17:42:57"." External links • "Real Time Streaming Protocol Information and Updates" (http://web. org/ ffmpeg-doc. html No mention of RTSP upon executing the command ffmpeg -h.rtsp. org& num=100). se/ changes.org) on 2007-03-06. RTP is one of the technical foundations of Voice over IP and in this context is often used in conjunction with a signaling protocol which assists in setting up connections across the network.com/quicktime/icefloe/dispatch028. ffmpeg. a statically linked Windows executable. html) as of 30 May 2010. (FFmpeg release used: "FFmpeg version SVN-r11143".Real Time Streaming Protocol • JetAudio 83 References [1] [2] [3] [4] RFC 2326.apple. org& num=100& start=100) (147 results total) [6] No mention of RTSP in FFmpeg documentation (http:/ / www. 1998 erlyvideo website (http:/ / erlyvideo. television services and web-based push-to-talk features. While RTP carries the media streams (e. superseded by RFC 3550 in 2003. RTCP is used to monitor transmission statistics and quality of service (QoS) and aids synchronization of multiple streams.) [5] No RTSP usage documented on FFmpeg. page 2 (http:/ / www. IETF. com/ search?q=rtsp+ site:ffmpeg. RTP is used extensively in communication and entertainment systems that involve streaming media. [7] An indirect hint in FFplay documentation (http:/ / www. • Tunnelling RTSP and RTP through HTTP (http://developer.. Real Time Streaming Protocol (RTSP). haxx. RTP is originated and received on even port numbers and the associated RTCP communication uses the next higher odd port number.g. video teleconference applications.rtsp.. Section "Supported file protocols" of this help screen only lists "file: http: pipe: rtp: tcp: udp:" and skips over "rtsp:". google. . google. A standard solution to help RTSP work through firewalls and web proxies Real-time Transport Protocol The Real-time Transport Protocol (RTP) defines a standardized packet format for delivering audio and video over IP networks. ffmpeg. Executing ffmpeg -format yields a multi-page list of items with one inconspicuous mention: "RTSP input format". org/ ) http:/ / curl. com/ search?q=rtsp+ site:ffmpeg. org/ ffplay-doc.org website as of 30 May 2010: Google search page 1 (http:/ / www. a central information repository about RTSP.archive. audio and video).org/web/20070306002838/ http://www. such as telephony.

[2] For each class of application (e. real-time. RTP defines a profile and one or more associated payload formats. sequence numbers (for packet loss and reordering detection) and the payload format which indicates the encoded format of the data.) and allow new formats to be added without revising the RTP standard. RTCP.g. and the RTCP is used to periodically send control information and QoS parameters. that are common during transmissions on an IP network. RTP and RTCP typically use unprivileged UDP ports (1024 to 65535). typically around 5%. but is instead provided through RTP profiles and payload formats.[2] A complete specification of RTP for a particular application usage will require a profile and payload format specification(s).[4] The Transmission Control Protocol (TCP).Real-time Transport Protocol 84 Overview RTP was developed by the Audio/Video Transport working group of the IETF standards organization. The information required by a specific application's needs is not included in the generic RTP header. as the protocol design is transport independent.[3] RTP is regarded as the primary standard for audio/video transport in IP networks and is used with an associated profile and payload format.[12] but may use other transport protocols (most notably.[2] RTP is designed for end-to-end. RTP.[5] is not normally used in RTP application because TCP favors reliability over timeliness. video).[10] The ports which form a session are negotiated using other protocols such as RTSP (using SDP in the setup method)[11] and SIP. used to specify quality of service (QoS) feedback and synchronization between the media streams. transfer of stream data. MPEG-4.[13] .[7] Protocol components The RTP specification describes two sub-protocols: • The data transfer protocol.323 and RTSP. as of 2010. which deals with the transfer of real-time data. A session consists of an IP address with a pair of ports for RTP and RTCP. For example. an RTP port should be even and the RTCP port is the next higher odd port number. which can be made unnoticeable with suitable error concealment algorithms. MPEG. According to the specification. SCTP and DCCP) as well. audio and video streams will have separate RTP sessions. The design of RTP is based on the architectural principle known as application level framing (ALF). they are not in widespread use. although standardized for RTP use.[4] Other transport protocols specifically designed for multimedia sessions are SCTP[6] and DCCP. Information provided by this protocol include timestamps (for synchronization).323 or Session Initiation Protocol (SIP) • An optional media description protocol such as Session Description Protocol Sessions An RTP Session is established for each multimedia stream. Profiles and Payload formats One of the design considerations of RTP was to support a range of multimedia formats (such as H. For example. although. enabling a receiver to deselect a particular stream.264. RTP and RTCP. RTP is used in conjunction with other protocols such as H.[1] Real-time multimedia streaming applications require timely delivery of information and can tolerate some packet loss to achieve this goal. MJPEG. The bandwidth of RTCP traffic compared to RTP is small. The protocol provides facility for jitter compensation and detection of out of sequence arrival in data. loss of a packet in audio application may result in loss of a fraction of a second of audio data. RTP supports data transfer to multiple destinations through IP multicast. audio. RTP is used for transfer of multimedia data. Instead the majority of the RTP implementations are built on the User Datagram Protocol (UDP).[8] • A control protocol.[1] The RTP standard defines a pair of protocols.[8] [9] • An optional signaling protocol such as H. etc..

A padding might be used to fill up a block of certain size.[18] The fields in the header are as follows: • Version: (2 bits) Indicates the version of the protocol.. After the header. see RTP Profile for audio and video conferences with minimal control (RFC 3551). G. The RTP does not specify any action on packet loss. Current version is 2. the format of which is determined by the particular class of application. video applications may play the last known frame in place of the missing frame. MP3. it means that the current data has some special relevance for the application. 96+32×CC 128+32×CC The RTP header has a minimum size of 12 bytes.[14] H.. H.[19] • P (Padding): (1 bit) Used to indicate if there are extra padding bytes at the end of the RTP packet.263. Profile-specific extension header ID Extension header length Extension header . For example. This is specified by an RTP profile. GSM. the initial value of the sequence . 85 Packet header RTP packet header bit offset 0 32 64 96 0-1 2 3 X 4-7 CC 8 M 9-15 PT 16-31 Sequence Number Version P Timestamp SSRC identifier CSRC identifiers .[22] According to RFC 3550.[19] • CC (CSRC Count): (4 bits) Contains the number of CSRC identifiers (defined below) that follow the fixed header. For example.723. • The Secure Real-time Transport Protocol (SRTP) (RFC 3711) defines a profile of RTP that provides cryptographic services for the transfer of payload data.[21] • Sequence Number: (16 bits) The sequence number is incremented by one for each RTP data packet sent and is to be used by the receiver to detect packet loss and to restore packet sequence. and a payload type identifier (in header) using Session Description Protocol (SDP). If it is set.261.726.[15] Examples of RTP Profiles include: • The RTP profile for Audio and video conferences with minimal control (RFC 3551) defines a set of static payload type assignments.[1] Some of the audio payload formats include: G.[16] • The experimental Control Data Profile for RTP (RTP/CDP[17] ) for machine-to-machine communications.711.[20] • PT (Payload Type): (7 bits) Indicates the format of the payload and determines its interpretation by the application. QCELP.. MPEG-4[14] etc. for example as required by an encryption algorithm. and a mechanism for mapping between a payload format.729. This is application or profile specific. G. and some of the video payload formats include: H.[19] • X (Extension): (1 bit) Indicates presence of an Extension header between standard header and payload data.. it is left to the application to take appropriate action.264.[20] • M (Marker): (1 bit) Used at the application level and defined by a profile.. Each profile is accompanied by several payload format specifications. This is followed by the RTP payload. each of which describes the transport of a particular encoded data.Real-time Transport Protocol The profile defines the codecs used to encode the payload data and their mapping to payload format codes in the Payload Type (PT) field of the RTP header (see below). optional header extensions may be present. G. DTMF etc.

a common sample rate in digital telephony) could use that value as its clock resolution. control and termination..264.[23] An RTP sender captures the multimedia data. Proposed Standard. an audio application that samples data once every 125 µs (8 kHz.264 Video RFC 4103. The granularity of the timing is application specific. detects missing packets and may perform reordering of packets.[20] CSRC: Contributing source IDs enumerate contributing sources to a stream which has been generated from multiple sources. Proposed Standard. For example. RTP Payload Format for Transport of MPEG-4 Elementary Streams RFC 3016. The clock granularity is one of the details that is specified in the RTP profile for an application.[3] Timestamp: (32 bits) Used to enable the receiver to play back the received samples at appropriate intervals. Obsolete. RTP Payload Format for H. RTSP. The RTP receiver. which is then encoded. are used to encode the payload data as specified via RTP Profile. Protocols like SIP. RTP: A Transport Protocol for Real-Time Applications RFC 3551.263 etc. H.245 are used for session initiation. framed and transmitted as RTP packets with appropriate timestamps and increasing sequence numbers.[20] 86 • • • • RTP-based systems A complete network based system will include other protocols and standards in conjunction with RTP. When several media streams are present. H. MPEG. captures the RTP packets. RTP Payload Format for H. Depending on the RTP Profile in use.[23] RFC references • • • • • • • • RFC 3550. The frames are decoded depending on the payload format and presented to the end user. Other standards like H. the timestamps are independent in each stream. RTP Payload Format for Text Conversation RFC 3640.[20] Extension header: (optional) The first 32-bit word contains a profile-specific identifier (16 bits) and a length specifier (16 bits) that indicates the length of the extension (EHL=extension header length) in 32-bit units. RTP Profile for Audio and Video Conferences with Minimal Control RFC 6184. the Payload Type field is set. RTP Payload Format for MPEG-4 Audio/Visual Streams RFC 2250. but the presence of sequence numbers makes it possible to detect missing packets.[22] SSRC: (32 bits) Synchronization source identifier uniquely identifies the source of a stream.225 and H.Real-time Transport Protocol number should be random to make known-plaintext attacks on encryption more difficult. Standard 64. The synchronization sources within the same RTP session will be unique. Standard 65. RTP Payload Format for MPEG1/MPEG2 Video .264 Video RFC 3984.[20] RTP provides no guarantee of delivery. and may not be relied upon for media synchronization. excluding the 32 bits of the extension header.

Finley (2010). Philip A. Morgan Kaufmann. ISBN 0071363262.. C. [13] RFC 3550. De Boeck Université. "Transporting Voice by using IP". google. RTP (http://books. com/ books?id=PVIuN9Y5FGMC& pg=PA47& dq=RTP+ session). p. V. pp. ISBN 9780974094526. p. p. ISBN 9783540725312. p.  28–7 (http:/ / books. BoD . p. pp. ISBN 9781558609136. Colin (2003). p. Network Protocols Handbook.56 [9] Peterson.435 [10] Zurawski.Books on Demand. RTCP and RTSP protocols" (http:/ / books. [11] RFC 4566: SDP: Session Description Protocol. google.google. Larry L. [8] Colin Perkins. Computer Networks. de/ books?id=t18ehd_vM6wC& lpg=PP1& pg=PA9). p. Springer. Computer Networks (http://books. 366. Mihaela van der Schaar (2007). Carrier grade voice over IP. 431 (http:/ / books. ISBN 9783839184608. 363. [7] Ozaktas. com/ books?id=zGVVuO-6w3IC& pg=PA430).com/books?id=D_GrQa2ZcLwC&pg=PA144). Serial Communication over RTP/CDP. p. google. . M. com/ books?id=zGVVuO-6w3IC& pg=PA431) [20] "RTP Data Transfer Protocol" (http:/ / rfc-ref. p. Morgan Kaufmann.  (http:/ / books. [15] Perkins. . pp. ISBN 155860832X.com/ ?id=zGVVuO-6w3IC) (4 ed. Handley. The industrial information technology handbook. p. pp. html). p. Retrieved 2009-03-18. ISBN 9780849319853. THREE-DIMENSIONAL TELEVISION (http:/ / books. . Davie (2007). p. org/ RFC-TEXTS/ 3550/ chapter4. [4] Colin Perkins. pp. ISBN 9780123740137. Peterson (2007). com/ books?id=Oq8SEUW1wdQC& pg=PT320).google. Richard (2004). com/ ?id=LtBegQowqFsC& pg=PA363& dq=rtp+ sctp).60 [16] Collin Perkins.367 [17] Breese. • "RTP" (http://books. ISBN 0120884801. Haldun M. Multimedia over IP and wireless networks.  47 (http:/ / books. 414. google. RFC-Ref. 806. Morgan Kaufmann. pp.59 [22] Peterson. com/ books?id=MwMDUBKZ3wwC& pg=PT225& dq=RTP+ session). 430 [19] Peterson.46 [5] RFC 4571 [6] Farrel. [3] Daniel Hardy (2002).  430 (http:/ / books. [21] Colin Perkins.Real-time Transport Protocol 87 Notes [1] Colin Perkins. 432 (http:/ / books.  298 (http:/ / books. pp. google. Jacobson. "RTP. IETF (July 2006) [12] Collins. Network. The Internet and its protocols (http:/ / books. Javvin Technologies. India. • Peterson.youtube. 2005. p.com/?id=OM7YJAy9_m8C). com/ ?id=kQvCHpuXji8C& pg=PA366& dq=rtp+ dccp). Adrian (2004). • "RTP" (http://www. 2008. Broadband Networks. google. google. Ministry of Human resources. google. google. .71 [14] Chou. Levent Onural (2007). Daniel (2002). CRC Press. [18] Peterson.. com/ books?id=zeLFs3GD0QQC& pg=PA514). Academic Press.11-13 References • Perkins.google. google. Perkins.). ISBN 9780672322495. Addison-Wesley.com/watch?v=OaL2vVFbCG4&feature=channel_page).  514 (http:/ / books. com/ ?id=MwMDUBKZ3wwC).55 [2] Larry L. Bruce S. p. com/ books?id=zGVVuO-6w3IC& pg=PA432) [23] Perkins. google. . McGraw-Hill Professional. p..

lost packet counts.cs. columbia.Jrtplib) • RTPMobile . RTCP gathers statistics for a media connection and information such as transmitted octet and packet counts.linphone.org/eng/documentation/dev/ortp. this permits non-intrusive session quality monitoring. whereas RTP itself is not.edm. a C++ RTP library (http://research.[2] The primary function of RTCP is to provide feedback on the quality of service (QoS) in media distribution by periodically sending statistics information to participants in a streaming multimedia session.uhasselt.edu/~hgs/rtp/faq.[1] superseding its original standardization in 1996 (RFC 1889).gnu. A fourth.cs. Although a source identifier (SSRC) of an RTP stream is expected to be unique. Protocol functions RTCP provides three basic functions expected to be implemented in all RTP sessions: • The primary function of RTCP is to gather statistics on quality aspects of the media distribution during a session and transmit this data to the session media source and other session participants. If the session is carried over a multicast network.columbia. .org) RTP Control Protocol The Real-Time Transport Control Protocol (RTCP) is a sister protocol of the Real-time Transport Protocol (RTP). the protocol must include session bandwidth management. with the Secure Real-time Transport Protocol (SRTP) defined in RFC 3711.Real-time Transport Protocol 88 External links • oRTP. Such information may be used by the source for adaptive media encoding (codec) and detection of transmission faults. the instantaneous binding of source identifiers to end-points may change during a session. even in a multicast session which may involve thousands of recipients. perhaps by limiting flow. Furthermore. or using a different codec.php?n=CS. RTCP bandwidth usage should generally not exceed 5% of total session bandwidth. Such mechanisms may be implemented. • RTCP reports are expected to be sent by all participants. for example. because RTCP is a convenient means to reach all session participants. with RTCP messages being sent over the next highest odd-numbered port. is the provisioning of session control functions. Its basic functionality and packet structure is defined in the RTP specification RFC 3550. It partners RTP in the delivery and packaging of multimedia data.org/software/ccrtp/) • JRTPLIB. This is achieved by dynamically controlling the frequency of report transmissions. providing a full streaming suite including experimental SCTP support (http://lscube. Thus.edu/~hgs/rtp) (including FAQ (http://www. jitter. The CNAME establishes unique identification of end-points across an application instance (multiple use of media tools) and for third-party monitoring.html)) • GNU ccRTP (http://www.sitesled.be/~jori/page/index.com) • LScube project. RTCP itself does not provide any flow encryption or authentication methods. optional feature. • RTCP provides canonical end-point identifiers (CNAME) to all session participants. RTCP provides out-of-band statistics and control information for an RTP flow.html) • Henning Schulzrinne's RTP page (http://www. 25% of the RTCP bandwidth should be reserved to media sources at all times. an open source . so that in large conferences new participants can receive the CNAME identifiers of the senders without excessive delay. RTP library from Linphone written in C (http://www. to avoid network congestion. RTP is only transmitted by a media source.NET. An application may use this information to control quality of service parameters. Such traffic will increase proportionally with the number of participants. and round-trip delay time. but does not transport any media streams itself.NET RTP library (http://rtpmobile. Typically RTP will be sent on an even-numbered UDP port.

[5] Hierarchical aggregation Hierarchical aggregation (also known as RTCP feedback hierarchy) is an optimization of the RTCP feedback model and its aim is to shift the maximum number of users limit further together with QoS measurement. A standards-based extension of RTCP is the Extended Report packet type introduced by RFC 3611. very long delays (minutes to hours) between RTCP reports may occur. receiver report.RTP Control Protocol 89 Message types RTCP distinguishes several types of packets: sender report. those that do not send RTP packets.[3] Sender report (SR) The sender report is sent periodically by the active senders in a conference to report transmission and reception statistics for all RTP packets sent during the interval. It allows an end-point to announce that it is leaving the conference. Application-specific message (APP) The application-specific message provides a mechanism to design application-specific extensions to the RTCP protocol. only the most modern IPTV systems use hierarchical aggregation. and bye. but it is not so suitable for large-scale applications with huge number of users. This affords the potential of inappropriate reporting of the relevant statistics by the receiver or cause evaluation by the media sender to be inaccurate relative to the current state of the session. RTCP biasing and hierarchical aggregation. such as in IPTV. this message is a direct announcement. Source description (SDES) The Source Description message is used to send the CNAME item to session participants. Receiver report (RR) The receiver report is for passive participants. Methods have been introduced to alleviate the problems:[4] RTCP filtering. Scalability in large deployments In large-scale applications. because of the RTCP bandwidth control mechanism required to control congestion (see #Protocol functions). which is the number of seconds elapsed since midnight on January 1. Although other sources can detect the absence of a source. 1900. End of participation (BYE) A source sends a BYE message to shut down a stream. It is particularly important when both audio and video are transmitted simultaneously. such as in Internet Protocol Television (IPTV). the protocol is extensible and allows application-specific RTCP packets. In addition. Another type of multicast could be Any-Source Multicast. Acceptable frequencies are usually less than one minute. The sender report includes an absolute timestamp. e-mail address. As of 2007. because audio and video streams use independent relative timestamps. It may also be used to provide additional information such as the name. It is also useful to a media mixer. The report informs the sender and other receivers about the quality of service. source description. . The absolute timestamp allows the receiver to synchronize RTP messages.[6] It is used with Source-Specific Multicast where only a single source is allowed. and address of the owner or controller of the source. telephone number.

Caceres. using existing E. R. ISBN 9780974094526. V. academypublisher. Dan Komosný. T. ENUM also permits connecting the IP world to the telephone system in a seamless manner. pdf).com/books?id=D_GrQa2ZcLwC&pg=PA145). The Internet Society (July 2003) [2] RFC 3605. and for the development of new IP multimedia services. 806.com/books?id=OM7YJAy9_m8C). Clark (Ed. com/ jnw/ vol03/ no03/ jnw03030110. Rationale Being able to dial telephone calls the way customers have come to expect is considered crucial for the convergence of classic telephone service (PSTN) and Internet telephony (VoIP). 2005. Javvin Technologies. The problem of a single universal personal identifier for multiple communication services can be solved with different approaches. S. One simple approach is the Electronic Number Mapping System ENUM (also known as Telephone Number Mapping). vutbr. March 2008 [5] Realtime control protocol and its improvements for Internet Protocol Television (http:/ / www. cz/ multicastssm/ pdf/ TreeStructureMulticastFeedback.164 standard.). R. Large-Scale RTCP Feedback Optimization. Network Protocols Handbook.com/ books?id=zGVVuO-6w3IC) (4 ed.164 NUmber Mapping (ENUM) standard.The Sixth International Conference on Networking. Bruce S.google..3 (3). Martinique. • "RTCP" (http://books. • Peterson. Computer Networks (http://books.google. RTP (http://books. utko. ICN07 . developed by the IETF. feec. Frederick. Colin (2003).RTP Control Protocol 90 References [1] RFC 3550. The Internet Society (November 2003) [4] Vít Novotný.). while the Internet uses the Domain Name System for linking domain names to IP addresses and other resource information. protocols and infrastructure to indirectly access different services available under a single personal identifier. A. The most prominent facility for telephone number mapping is the E. RTP Control Protocol Extended Reports (RTCP XR). Schulzrinne. Friedman (Ed. Telephone numbers are systematically organized in the E. Tree Structure for Specific-Source Multicast with Feedback Aggregation (http:/ / adela. Morgan Kaufmann. Microsoft (October 2003) [3] RFC 3611. ISBN 9780672322495. It uses special DNS record types to translate a telephone number into a Uniform Resource Identifier or IP address that can be used in Internet communications. 2007. .google. Dan Komosný. Telephone number mapping systems provide facilities to determine applicable Internet communications servers responsible for servicing a given telephone number by simple lookups in the Domain Name System. Casner. Journal of Networks. RTP: A Transport Protocol for Real-Time Applications. pp. Jacobson. Addison-Wesley. 414.164 telephone numbers. Davie (2007). ISBN 0-7695-2805-8 Additional reading • Perkins. Larry L. H. Vol. Real Time Control Protocol (RTCP) attribute in Session Description Protocol (SDP). ISBN 9780123740137.). Huitema. Telephone number mapping Telephone number mapping is the process of unifying the telephone number system of the public switched telephone network with the Internet addressing and identification name spaces. pp. pdf) [6] Vít Novotný. C.

namely "e164. when the calling party is at the IP side.4.5. IN NAPTR 100 10 "u" "E2U+sip" "!^.164 called party number.2.5. Therefore. the User Agent (UA) piece of software of the dialler will allow to dial a E. the number will be translated at the ENUM gateway into the corresponding URI. a VoIP number.164 regular personal telephone number associated with the personal URI.4. "Naming Authority Pointer Resource Records" as defined in RFC 3403. This URI will be used for looking-up and fetching the NAPTR record obtaining the called party wishes about how the call should be forwarded or terminated (either on IP or on PSTN terminations) – the so-called access information – which the registrant (the called party) has specified by writing his/her choice at the ‘NAPTR record’. IP-telephony addresses.com!" . but the dialler UA will convert it into a URI. by reversing the numbers.e164. web pages. Calling by using a new personal E. a fax number.net.net!" . This example specifies that if you want to use the "E2U+sip" service.arpa suffix thus: 2.164 numbers on the IP side of the network. NAPTR records are used for setting the subscriber call forwarding/termination preferences.2. mobile telephone numbers. your SIP address is sip:1234@555telco. For example. The following Figure illustrates how ENUM works by giving an example: Subscriber A sets out to call Subscriber B. RFC 6116 defines how any E. GPS coordinates. to be used on the PSTN part of the network • Authority to write his call forwarding/termination preferences in the NAPTR record accessible via the personal URI This works as follows: (1) the Registrar provides the Subscriber (or Registrant) with a domain name. The ITU ENUM allocates a specific zone. such as e-mail addresses.164 addresses to SIP addresses. as explained below • One E. (2) a personal E.164 number. if your number is +15551234. your SIP address is sip:4242@555telco.example.e164.example. The URI domain name of (1) is biunivocally associated (one-to-one mapped) to the subscriber E.Telephone number mapping 91 System details For an ENUM subscriber to be able to activate and use the ENUM service it needs to obtain three elements from a Registrar: • A personal Uniform Resource Identifier (URI) to be used on the IP part of the network. if your number is +15554242.net. you should use sip:phoneme@example. In the Domain Name System. IN NAPTR 102 10 "u" "E2U+mailto" "!^. The regular expression can be used by a telephone company to easily assign addresses to all of its clients. separating them with dots and adding the e164.164 number. a personal website.*$!mailto:myemail@example. call diversions or instant messaging.*$!sip:phoneme@example. to be used for looking-up at the ENUM gateway DNS and fetch the NAPTR record obtaining the called party wishes about how the call should be forwarded or terminated (again.4. Alternatively. the URI.5.5.arpa The URI can then be used for obtaining the Internet Protocol addresses for services such as the Session Initiation Protocol (SIP) VoIP telephony. that will be used for accessing a DNS server to fetch a NAPTR record. such as +1 555 42 42 can be transformed into a Uniform Resource Identifier (URI).164 number (the ENUM number) to look-up at a database is therefore an indirect calling support service. Therefore the whole system can 'translate' E.net as the address. voice mail systems.1.4.1.arpa" for use with ENUM E.164 ENUM number of (2).5. if a calling party being at the PSTN network dials a called party ENUM number by touch typing the E. An example NAPTR record is: $ORIGIN 2. .164 telephone number (the ENUM number). either on IP or on PSTN terminations).5.arpa. Finally (3) the NAPTR record corresponding to the subscriber URI contains the subscriber call forwarding/termination preferences.

8. Application scenario for Voice over IP (VoIP) with ENUM.4.3. or at the PABX or Gateway. The User Agent of an ENUM-enabled subscriber terminal device.e164. The user gets the requested address back on his PC or terminal. The terminal application now sets up a communication link.) in the PC or terminal the number is converted to a domain name. In the example above. when the user types the telephone number in his web browser ENUM enabled agent and indicates what item of information he is looking for (email address.3. 2. or a PBX. 4. ENUM therefore in fact functions as a mechanism for translating a telephone number into a domain name with the requested address or number associated with it.Telephone number mapping 92 1. but without the user viewing how this is done.e164.7. translates the request for the number +34 98 765 4321 in accordance with the rule described in RFC 6116 into the ENUM domain 1.3. as this is done automatically behind the scene using a user agent software in his PC or terminal. For instance. SIP per RFC 3261.arpa. etc.8. 3. or a Gateway. which send back the NAPTR records associated with the name. just as he is currently unaware that he is using the DNS when he makes a connection with the Internet or what is going on at the telephone switch when he makes a call.2. as per RFC 3403. telephone number. This is sent to ENUM servers on the Internet.5.9.4. Uses Call forwarding with ENUM .9. A request is sent to the Domain Name System (DNS) asking it to look up the ENUM domain 1.4. The query returns a result in the form of so called Naming Authority Pointer Resource NAPTR records.arpa.6. The ENUM user does not notice anything of this reversal and the DNS database look-up. The access information and any priority indicated for them are stored in these.4. and the call is routed via the Internet.3.7.6.5. the response is an address that can be reached in the Internet using the VoIP protocol. web address.2.

It is also possible that if an E. either at the IP or at the PSTN side of the network. this leads to confusion and complexity. What happens at a country level is a National Matter. he has to begin deciding how to establish the call with the other party: via VoIP. In general the conventional DNS registry-registrar model is used. the registrants.arpa are delegated on a country-code basis by ITU. Alternative ENUM-like services such as E164.164 numbers are registered in which of these alternate ENUM-like trees. Furthermore. cellphone.164 numbers: billions of people knowing how to dial using numbers. but most likely it will be a list of how the call should be forwarded to various fixed-line. It is the called party choice to opt-in ENUM and also to decide to let the calling party know her/his wishes. However. With ENUM indirect dialling it is . It takes registration requests from registrars who are agents of the end users. Next. Each delegation is normally made to a regulatory body designated by the national government for the country code concerned. When they call someone who does not use ENUM. cellphones. secretarial or voice mail services.arpa. The national ENUM registry manages and operates the DNS infrastructure and related systems for country-code.arpa. Multiple DNS lookups may be needed and it is far from simple to know which E. to establish a connection with the subscriber.164 number is registered in several of these trees. the subscriber "owning" a particular E. it is an indirect dialling service designed to work seamlessly on PSTN and VoIP that builds on the great value of the e.e164. If the called person has opted to use ENUM she/he will have published the ENUM number and have entered (via ENUM NAPTR) his/her wishes for how the call should be terminated. entering a URI or dialling a number.org [1] have also emerged. a voice message will be sent to the subscriber via a listed email address. This might be a single VoIP identifier. an attempt will be made. the next record selected will be that for a connection to Call forwarding with ENUM a PSTN or mobile telephone. Registrars are typically VoIP providers and telcos who bundle an ENUM registration as part of a VoIP service package. Called party facility ENUM can also be viewed as a called party facility.164 number may not be aware that their number has been entered into one or more of these alternate ENUM-like trees or what information these alternate trees are returning for their E. Today when a user places a regular phone call. using the VoIP record from this list. If the subscriber is not online. People using an ENUM-enabled VoIP service can dial the registrant's existing number and be connected to the registrant's VoIP telephone over the Internet instead of using the PSTN. If this attempt fails too. there can be inconsistencies in the information that is returned. telephone numbers and email addresses. calls complete over the Public Switched Telephone Network or PSTN in the usual manner. Fixed-line PSTN. Subdomains of e164. which leads to an ENUM lookup (such as is provided by SIP Broker). The caller uses the telephone to dial the number of another subscriber. The DNS responds to the caller by returning a list with NAPTR records for VoIP communication. Basically.Telephone number mapping 93 One way of doing call forwarding with ENUM is illustrated in the next figure.164 number. These services might verify PSTN numbers and can be used in addition to e164.

The registry The registry is the manager of a national ENUM zone. Examples are the numbers for fixed telephony and mobile telephony. The registrant or subscriber is thus the person whose information has been included in ENUM and must not be confused with the person who uses the Internet to find an address through ENUM. The number holder operator will be interconnected to other operators and will receive from them calls to his assigned range of numbers.. and is being the subject of new IETF recommendations to support VoIP peering. as it were. in the same way DNS is used internally to networks. In addition every registrant must receive equal and open access. with subscriber opt-in capabilities and delegation at the country code level in the e164. the top of the national ENUM hierarchy and ensures that reference is made to the registrars’ servers on which the access information is located.arpa domain. Quite often. A presence enhanced ENUM facility having various profiles could automatically change the called party wishes as a function of where he/she is available.. Because of the hierarchical structure of the DNS. • Private ENUM: A carrier. These include: The registrant or subscriber The registrant is the person or subscriber that makes his access information available to others through ENUM. Parties having a direct interest in ENUM Various parties are involved with ENUM. 94 ENUM varieties One potential source of confusion. when talking about ENUM. The government or the regulator Usually a governmental entity or a regulatory authority has control over the National zone of ENUM and will play a role in the appointment of the registry. there can only be one registry for every national ENUM zone. To prevent abuse of this position. The number holder operator Telephony services or telecommunication services operators have being assigned blocks of numbers by the regulator. people speaking of ENUM are really referring to only one of the following: • Public ENUM: The original vision of ENUM as a global. The registry forms. This is also referred to as user ENUM. not the individuals. for the . email or whatever telecom service he uses without having to tell all his contacts about that. public directory-like database. VoIP operator or ISP may use ENUM techniques within its own networks. IMS. • Carrier ENUM: Groups of carriers or communication service providers agree to share subscriber information via ENUM in private peering relationships.Telephone number mapping the called party wishes that matter and solve that decision.. This could be a mechanism to automatically switch between cellphone and VoIP to the most convenient (or the less costing) termination. They subsequently enable their users to use individual telephone numbers from those number blocks. Carrier ENUM is also referred to as infrastructure ENUM. webpage. The carriers themselves control subscriber information. Another benefit of indirect dialling is to free the user to change his phone telco. The registrar The registrar is the party who manages the registrant’s access information and ensures that it is publicly available on the Internet.. requirements are strict with respect to the impartiality of the registry and the costs and quality of the service. is the variety of ENUM implementations in place today.

Consequently in ENUM once the operator number holder assigns a number to a registrant. to whom he will transit the calls.org . the number belongs to that registrant during his/her entire life.164 Number Space into the DNS [17] Voice Peering Forum [18] ENUM Convergence News [19] SIP Broker's multi-root ENUM lookup [20] lookup under e.PH Global ENUM Registry [22] Enum 2 go [23] ENUM en español [24] . secure multilateral Carrier/Infrastructure ENUM services [15] CircleID: ENUM Convergence [16] ENUM: Mapping the E.arpa and UPT ENUM services [21] E164. will have an arrangement with a gateway operator.eu . if the registrant wishes to change his initial number holder operator (that might also coincide being his gateway operator) there have to be provisions for the ENUM number to be ported from the initial operator to other number holder operators.charters/enum-charter.The E.int/osg/spu/enum/ • http://www.Free public ENUM database for Europeans [12] Voice Peering Fabric ENUM Registry [13] Association deploying Global Services using UPT and ENUM under +87810 country code [14] XConnect Global Networks Ltd.html External links • • • • • • • • • • • • • • • • • • • • • • Enum base [4] Country Code 1 ENUM LLC [5] TNS Carrier ENUM Services [6] Carrier Enum Registry and Tandem Transit [7] GSMA PathFinder Carrier ENUM Technology [8] NeuStar Global SIP Inter-working [9] UK ENUM database run by Nominet [10] Distributed Universal Number Discovery [11] e164.164 to Uniform Resource Identifiers (URI) Dynamic Delegation Discovery System (DDDS) Application (ENUM) • RFC 3403 .It's All in the Numbers [3] • http://www.164.The bridge between telephony and internet [2] • ENUM . meant to be valid for the registrant life.org/html. alternatively. But ENUM is a personal number. In ENUM the number holder operator will typically be the gateway operator or.Dynamic Delegation Discovery System (DDDS) Part Three: The Domain Name System (DNS) Database • ENUM . Hence.Free public ENUM database [1] openENUM.ietf.itu. 95 References • RFC 6116 .Telephone number mapping calls to be terminated.

Telephone number mapping

96

References
[1] http:/ / www. e164. org [2] http:/ / www. eurescom. de/ message/ messageSep2004/ ENUM_The_bridge_between_telephony_and_Internet. asp [3] http:/ / www. tmcnet. com/ it/ 0403/ 0403inim. htm [4] http:/ / www. enum. org/ [5] http:/ / www. enumllc. com/ [6] http:/ / www. tnsi. com/ products-services/ telecom/ telecom-services/ carrier-enum-solutions [7] http:/ / www. carrierenum. us/ [8] http:/ / gsmworld. com/ our-work/ programmes-and-initiatives/ pathfinder/ [9] http:/ / neustar. biz/ interoperability/ sipix. cfm [10] http:/ / www. nic. uk/ tech/ enum/ [11] http:/ / www. dundi. info/ [12] http:/ / www. openenum. eu/ [13] http:/ / www. thevpf. com/ feature/ enum [14] http:/ / www. visionng. com/ [15] http:/ / www. xconnect. net/ [16] http:/ / www. circleid. com/ tags/ enum [17] http:/ / www. circleid. com/ posts/ enum_mapping_e164_into_dns/ [18] http:/ / www. thevpf. com/ [19] http:/ / www. enumregistry. com/ enum/ enumnews. aspx [20] http:/ / www. sipbroker. com/ sipbroker/ action/ enumLookup [21] [22] [23] [24] http:/ / www. enum2go. com http:/ / www. e164. ph http:/ / www. enum2go. com/ http:/ / es. wikitel. info/ wiki/ ENUM

ITU-T
The ITU Telecommunication Standardization Sector (ITU-T) is one of the three sectors (divisions or units) of the International Telecommunication Union (ITU); it coordinates standards for telecommunications. The standardization work of ITU dates back to 1865, with the birth of the International Telegraph Union. It became a United Nations specialized agency in 1947, and the International Telegraph and Telephone Consultative Committee (CCITT, from French: Comité Consultatif International Téléphonique et Télégraphique) was created in 1956. It was renamed ITU-T in 1993.[1] ITU has been an intergovernmental public-private partnership organization since its inception and now has a membership of 191 countries (Member States) and over 700 public and private sector companies as well as international and regional telecommunication entities, known as Sector Members and Associates, which undertake most of the work of the Sector.[2] ITU-T has a permanent secretariat, the Telecommunication Standardization Bureau (TSB), based at the ITU HQ in Geneva, Switzerland. The elected Director of the Bureau is Malcolm Johnson of the UK. Johnson was elected by the ITU Membership to the directorship for a 4-year term in November 2006 and was reelected for a second term starting January 2011.

ITU-T

97

Primary function
The ITU-T mission is to ensure the efficient and timely production of standards covering all fields of telecommunications on a worldwide basis, as well as defining tariff and accounting principles for international telecommunication services.[3] The international standards that are produced by the ITU-T are referred to as "Recommendations" (with the word ordinarily capitalized to distinguish its meaning from the ordinary sense of the word "recommendation"), as they become mandatory only when adopted as part of a national law. Since the ITU-T is part of the ITU, which is a United Nations specialized agency, its standards carry more formal international weight than those of most other standards development organizations that publish technical specifications of a similar form.[4]

History
Although the ITU itself dates back to 1865,[1] the formal standardization processes are more recent. Two consultative committees were created by the ITU’s 1925 Paris conference to deal with the complexities of the international telephone services (known as CCIF, as the French acronym) and long-distance telegraphy (CCIT).[5] In view of the basic similarity of many of the technical problems faced by the CCIF and CCIT, a decision was taken in 1956 to merge them to become the single International Telegraph and Telephone Consultative Committee (CCITT, in the French acronym).[5] In 1992, the Plenipotentiary Conference (the top policy-making conference of ITU) saw a reform of ITU, giving the Union greater flexibility to adapt to an increasingly complex, interactive and competitive environment. It was at this time that CCITT was renamed the Telecommunication Standardization Sector (ITU-T), as one of three Sectors of the Union alongside the Radiocommunication Sector (ITU-R) and the Telecommunication Development Sector (ITU-D).[6] Historically, the Recommendations of the CCITT were presented to four-yearly plenary assemblies for endorsement, and the full set of Recommendations were published after each plenary assembly. However, the delays in producing texts, and translating them into other working languages, did not suit the fast pace of change in the telecommunications industry.[7]

"Real time" standardization
The rise of the personal computer industry in the early 1980s created a new common practice among both consumers and businesses of adopting "bleeding edge" communications technology even if it was not yet standardized. Thus, standards organizations had to put forth standards much faster, or find themselves ratifying de facto standards after the fact. The ITU-T now operates under much more streamlined processes. The time between an initial proposal of a draft document by a member company and the final approval of a full-status ITU-T Recommendation can now be as short as a few months (or less in some cases). This makes the standardization approval process in the ITU-T much more responsive to the needs of rapid technology development than in the ITU's historical past.[8] New and updated Recommendations are published on an almost daily basis, and much of the library of over 3,270 Recommendations is now free of charge online.[9] [10] [11] (Specifications jointly maintained by the ITU-T and ISO/IEC are not free.[12] ) ITU-T has moreover tried to facilitate cooperation between the various forums and standard-developing organizations (SDOs). This collaboration is necessary to avoid duplication of work and the consequent risk of conflicting standards in the market place.[13]

ITU-T In the work of standardization, ITU-T cooperates with other SDOs, e.g., the International Organization for Standardization (ISO) and the Internet Engineering Task Force (IETF).[14]

98

Development of Recommendations
Most of the work of ITU-T is carried out by its Sector Members and Associates, while the Telecommunication Standardization Bureau (TSB) is the executive arm of ITU-T and coordinator for a number of workshops and seminars to progress existing work areas and explore new ones. The events cover a wide array of topics in the field of information and communication technologies (ICT) and attract high-ranking experts as speakers, and attendees from engineers to high-level management from all industry sectors.[15] The technical work, the development of Recommendations, of ITU-T is managed by Study Groups (SGs). The people involved in these SGs are experts in telecommunications from all over the world. There are currently 13 SGs. Study groups meet face to face according to a calendar issued by the TSB.[16] SGs are augmented by Focus Groups (FGs), an instrument created by ITU-T, providing a way to quickly react to ICT standardization needs and allowing great flexibility in terms of participation and working methods. The key difference between SGs and FGs is that the latter have greater freedom to organize and finance themselves, and to involve non-members in their work. Focus Groups can be created very quickly, are usually short-lived and can choose their own working methods, leadership, financing, and types of deliverables.[17] Recent examples include work on Next Generation Networking, Internet Protocol Television (IPTV) and digital identity management.[18]

Approval of Recommendations
The “Alternative Approval Process” (AAP) is a fast-track approval procedure that was developed to allow standards to be brought to market in the timeframe that industry now demands. This dramatic overhaul of standards-making by streamlining approval procedures was implemented in 2001 and is estimated to have cut the time involved in this critical aspect of the standardization process by 80 to 90 per cent. This means that an average standard which took around four years to approve and publish until the mid nineties, and two years until 1997, can now be approved in an average of two months, or as little as five weeks. Besides streamlining the underlying procedures involved in the approval process, an important contributory factor to the use of AAP is electronic document handling. Once the approval process has begun the rest of the process can be completed electronically, in the vast majority of cases, with no further physical meetings. The introduction of AAP also formalizes public/private partnership in the approval process by providing equal opportunities for both Sector Members and Member States in the approval of technical standards. Once the text of a draft Recommendation prepared by SG experts is considered mature, it is submitted for review to an SG meeting. If agreed by the meeting it is given Consent. This means that the SG has given its consent that the text is sufficiently mature to initiate a final review process leading to approval of the draft Recommendation. After this Consent has been achieved, TSB announces the start of the AAP procedure by posting the draft text to the ITU-T web site and calling for comments. This gives the opportunity for all members to review the text. This phase, called Last Call, is a four-week period in which comments can be submitted by Member States and Sector Members. If no comments other than editorial corrections are received, the Recommendation is considered approved since no issues were identified that might need any further work. However, if there are any comments, the SG chairman, in consultation with TSB, sets up a comment resolution process by the concerned experts. The revised text is then posted on the web for an Additional Review period of three weeks. Similar to the Last Call phase, in Additional Review the Recommendation is considered as approved if no comments are received. If comments are received, it is apparent that there are some issues that still need more work, and the draft text and all comments are sent to the next Study Group meeting for further discussion and possible approval.[19]

Many of the Recommendations that define OSI are also ISO standards. sound programme and other multimedia signals Protection against interference Construction. French. Series A B C D E F G H I J K L M N O P Q R S T U V Organization of the work of ITU-T Means of expression: definitions. classification General telecommunication statistics General tariff principles Overall network operation. Russian and Spanish). symbols.ITU-T Those Recommendations considered as having policy or regulatory implications are approved through what is known as the “Traditional Approval Process” (TAP). local line networks Switching and signalling Telegraph transmission Telegraph services terminal equipment Terminals for telematic services Telegraph switching Data communication over the telephone network Description . so the year of issue may be necessary to identify a specific version of a Recommendation.5xx.500 Recommendation forms the introduction and overview to the set.500 Recommendation. service operation and human factors Non-telephone telecommunication services Transmission systems and media. and standards for mobile telephone systems are developed in ETSI and other forums.500" is used both to refer to the specific X. English. it will (mostly) keep the same number. TAP Recommendations are also translated into the six working languages of ITU (Arabic. including TMN and network maintenance Maintenance: international sound programme and television transmission circuits Specifications of measuring equipment Telephone transmission quality.500. digital systems and networks Audiovisual and multimedia systems Integrated services digital network Cable networks and transmission of television. installation and protection of cables and other elements of outside plant Telecommunication management. where the specific X. Chinese. where X is the series and 500 is an identifying number. Series of ITU Recommendations ITU-T issues Recommendations that have names like X. and to the entire family of Recommendations named X. Standards for Internet protocols are typically developed in the IETF. which allows a longer period for reflection and commenting by Member States.[20] 99 Series and Recommendations ITU-T Recommendations are the names given to telecommunications and computer protocol specification documents published by ITU-T. telephone service. telephone installations. When a Recommendation is updated. The term "X.

addressing and naming Operation.299 G.499 G.G.699 G.G.299 Y.G.G.799 Y.899 G.8100 .G.G.399 Y.300 .6999 G.499 Y.8299 G.8000 .400 .599 Y.G.699 Y.600 .G.800 .G.G.199 Y.8699 G.199 G.900 .800–Y.100 .1999 G.G.500–Y. Internet protocol aspects and next-generation networks Languages and general software aspects for telecommunication systems G Series Recommendations Transmission systems and media.999 G.799 G.6000 .9000 .7999 G.ITU-T 100 X Y Z Data networks.7000 .G. applications and middleware Network aspects Interfaces and protocols Numbering.449 Specification Type INTERNATIONAL TELEPHONE CONNECTIONS AND CIRCUITS GENERAL CHARACTERISTICS COMMON TO ALL ANALOGUE CARRIERTRANSMISSION SYSTEMS INDIVIDUAL CHARACTERISTICS OF INTERNATIONAL CARRIER TELEPHONE SYSTEMS ON METALLIC LINES GENERAL CHARACTERISTICS OF INTERNATIONAL CARRIER TELEPHONE SYSTEMS ON RADIO-RELAY OR SATELLITE LINKS AND INTERCONNECTION WITH METALLIC LINES COORDINATION OF RADIOTELEPHONY AND LINE TELEPHONY TRANSMISSION MEDIA AND OPTICAL SYSTEMS CHARACTERISTICS DIGITAL TERMINAL EQUIPMENTS DIGITAL NETWORKS DIGITAL SECTIONS AND DIGITAL LINE SYSTEM QUALITY OF SERVICE AND PERFORMANCE – GENERIC AND USER-RELATED ASPECTS TRANSMISSION MEDIA CHARACTERISTICS DATA OVER TRANSPORT – GENERIC ASPECTS PACKET OVER TRANSPORT ASPECTS Ethernet over Transport aspects MPLS over Transport aspects Quality and availability targets Service Management ACCESS NETWORKS G.399 G. digital systems and networks Specification Range G.200–Y.G.300–Y.G.G. administration and maintenance Security Performances Specification Type .8200 .G.100–Y.200 .9999 Y Series Recommendations Global Information Infrastructure.G.600–Y.700 .8999 G.700–Y.8600 . Internet Protocol Aspects And Nextgeneration Networks Specification Range GLOBAL INFORMATION INFRASTRUCTURE Y.899 INTERNET PROTOCOL ASPECTS General Services.1000 .G. open system communications and security Global information infrastructure.450 .8099 G.8000 .400–Y.8199 G.

1300–Y.2299 Y. dealing with telegraph and telephone.2500–Y.2400–Y. at the World Administrative Telegraphy and Telephone Conference held in Melbourne.1799 Y.2499 Y.[21] In line with the current Constitution and Convention of ITU.1000–Y. safety of life and priority of telecommunications and charging and accounting principles.[22] The ITRs comprise ten articles which deal.1400–Y. though a few countries. Before then a process of review of the ITRs.1100–Y.1800–Y. cooperation between countries and national administrations.2100–Y.2300–Y. The ITRs go back to the earliest days of the ITU when there were two separate treaties. will continue.2200–Y.2700–Y.1700–Y. the International Telecommunication Regulations.[23] . including United States and United Kingdom.1200–Y. The ITRs were adopted.1699 Y.1499 Y.1399 Y.2399 Y.2199 Y. 1988 (WATTC-88). with the definition of international telecommunication services. The adoption of the ITRs in 1988 is often taken as the start of the wider liberalization process in international telecommunications. access.1899 NEXT GENERATION NETWORKS Y.2800–Y. which began in 1998.2249 Y. had made steps to liberalize their markets before 1988.2799 Y.2899 Frameworks and functional architecture models Quality of Service and performance Service aspects: Service capabilities and service architecture Service aspects: Interoperability of services and networks in NGN Numbering. as a single treaty. inter alia.1299 Y. network capabilities and resource management Transport Interworking Quality of service and network performance Signalling Operation.ITU-T 101 Y. the ITRs can be amended through a World Conference on International Telecommunications (WCIT).1600–Y.1199 Y.1599 Y. administration and maintenance Charging International Telecommunication Regulations (ITRs) In addition to the ITU-T Recommendations. ITU-T is also the custodian of a binding international treaty.2250–Y. which have non-mandatory status until they are adopted in national laws.2000–Y.2099 Y.2599 Y. naming and addressing Network management Network control architectures and protocols Security Generalized mobility General Services and applications Architecture.1099 Y.1500–Y. and the next is scheduled for 2012.

798 Passive optical networks (PON) G.hn (Next generation wired home networking over powerlines. installation and protection of cables and other elements of outside plant. G.323 family of standards for multimedia and VoIP Interconnection rate harmonization.7xx series Standards relating to Quality of Service (QoS) Specification and Description Language Synchronous Digital Hierarchy (SDH) G. the need to converge and optimize the operating networks and the extraordinary expansion of digital traffic (i.25 (x)DSL (Digital Subscriber Line) series of standards for broadband telecoms Hot topics • ITU-T is committed to “bridging the standardization gap” – disparities in the ability of developing countries.e. H. .320 and H. T.707 – G.[24] • The ICT Security Standards Roadmap[25] has been developed to assist in the development of security standards by bringing together information about existing standards and current standards work in key standards development organizations. etc..1 (Abstract Syntax Notation One) Coding of audio G. T.37. L-series Data communicationover the telephone network. • The Next Generation Networks (NGN) concept takes into consideration new realities in the telecommunication industry characterized by factors such as.72x series Coding of still images JPEG T.4.264/MPEG-4 AVC Construction. implement. mobility.324 ISDN (Integrated Services Digital Network) Q.2 – T.931 Open Systems Interconnection Optical Transport Network (OTN) G.212 ISDN and PSTN/3G videoconferencing systems.509 Public telecommunication numbering plan.80 and JPEG 2000 T. D-series International Emergency Preference Scheme E.709. G.711 and G. contribute to and influence international ICT standards.38 G.). T.262/MPEG2-Video and H.803 Wavelength-division multiplexing (WDM) X. E. increasing demand for new multimedia services.106 IMSI codes used in SIM cards E. phonelines and coaxial cable) H. to access.30.983.984.987 Public Key Infrastructure (PKI) X. relative to developed ones. G.164 Security framework X.ITU-T 102 Key standards published by ITU • • • • • • • • • • • • • • • • • • • • • • • • • • • ASN.805 Signalling System 7 Q. V-series Fax standards T.800 series Coding of video coding H.

int/ITU-D/) ITU-T Study Groups (http://www. pdf) (s10) [15] apdip.int/ITU-T/) ITU Radiocommunication Sector (ITU-R) (http://www.int/ITU-T/newslog/) ITU-T Guide for beginners (PDF) (http://www. int/ ITU-T/ newslog/ Free+ Access+ For+ All+ To+ ITUT+ Standards. itu. html) [21] itu. pdf).itu. html) [19] itu. stanford.net (http:/ / www. int/ ITU-T/ focusgroups/ index.int (http:/ / www. int/ ITU-T/ itr/ files/ ITR-e. [8] "CCITT . int/ ITU-T/ 50/ docs/ ITU-T_50. (s13-16).ITU-T 103 References [1] "ITU's History" (http:/ / www. itu. Retrieved 2011-03-20. itu. html) [17] itu. September 1 2007.50 YEARS OF EXCELLENCE . 22 (http:/ / www. aspx) [13] "CCITT . itu.int (http:/ / www. int/ ITU-T/ info/ index. html) [18] ITU Telecommunication Standardization Sector (ITU-T) .1956-2006" (http:/ / www. html) [20] itu.50 YEARS OF EXCELLENCE . pdf). itu. pdf). itu. p. int/ osg/ spu/ stratpol/ ITRs/ auth/ itr-03. int/ en/ history/ overview/ Pages/ history. html) [16] apdip. General information on TSB (http:/ / www. int/ ITU-T/ focusgroups/ index. Retrieved 2011-03-20. .int/) ITU Standardization Sector (ITU-T) (http://www. Retrieved 2011-03-20. Retrieved 2011-03-20. .1956-2006" (http:/ / www. int/ ITU-T/ info/ tsb/ info. ITU-T e-FLASH . [3] ITU-T and TSB General Information (http:/ / www. [14] apdip.itu. ITU-T Focus Groups (http:/ / www. int/ dms_pub/ itu-t/ oth/ 0A/ 0F/ T0A0F0000090001PDFE. pdf).int (http:/ / www. itu.pdf) . itu. highbeam. (s23-24).itu. com/ doc/ 1G1-169930353. 17. int/ ITU-T/ gap/ ) [25] itu.50 YEARS OF EXCELLENCE . [2] "About ITU" (http:/ / www. itu. itu. pdf).int/ITU-T/studygroups/) ITU-T Focus Groups (http://www. int/ ITU-T/ 50/ docs/ ITU-T_50. (s28-29). International Telecommunication Union. int/ ITU-T/ e-flash/ 022-jan06. html) [11] Free access for all to ITU-T standards (http:/ / www. aspx) [12] ITU-T Recommendations overview (http:/ / www. 14.int/ITU-T/50/docs/ITU-T_50. pdf).html) ITU-T Recommendations (http://www. Retrieved 2011-03-20. 16.itu. pdf) • 50 years of CCITT/ITU-T (PDF) (http://www. .Focus Groups (http:/ / www. int/ ITU-T/ studygroups/ index. . html) External links • • • • • • • • • International Telecommunication Union (ITU) (official site) (http://www.50 YEARS OF EXCELLENCE . Retrieved 2011-03-20.net (http:/ / www. int/ dms_pub/ itu-t/ oth/ 0A/ 0F/ T0A0F0000090001PDFE.net (http:/ / www.1956-2006" (http:/ / www.itu.1956-2006" (http:/ / www.itu. itu.itu. 22 (http:/ / www. aspx).int (http:/ / www. International Telecommunication Union. [9] ITU-T Standards now freely available online (http:/ / www. pdf). pdf) (p13) [5] "CCITT . int/ ITU-T/ studygroups/ com17/ ict/ index. International Telecommunication Union. itu. itu. itu. itu. [7] "CCITT .net (http:/ / www.Issue No. .html) ITU-T Newslog (RSS available) (http://www. itu. International Telecommunication Union. p.int/ITU-T/publications/recs. International Telecommunication Union. pdf).int/ITU-T/focusgroups/index. ITU-T Study Groups (Study Period 2009-2012) (http:/ / www. International Telecommunication Union. int/ en/ ITU-T/ publications/ Pages/ recs.itu. int/ dms_pub/ itu-t/ oth/ 0A/ 0F/ T0A0F0000090001PDFE. int/ ITU-T/ e-flash/ 022-jan06.int (http:/ / www. p.int/ITU-R/) ITU Development Sector (ITU-D) (http://www. itu. itu.Review of the International Telecommunication Regulations (ITRs) (http:/ / www. itu. Retrieved 2011-03-20. itu. int/ ITU-T/ 50/ docs/ ITU-T_50. . int/ dms_pub/ itu-t/ oth/ 0A/ 0F/ T0A0F0000090001PDFE. aspx). doc) [22] ITU-T . (s27). ITU-T e-FLASH . pdf). apdip. net/ publications/ iespprimers/ eprimer-igov. (s16). Free access for all to ITU-T standards (http:/ / www. itu. International Telecommunication Union. pdf).50 YEARS OF EXCELLENCE . int/ ITU-T/ itr-eg/ ) [23] itu. int/ dms_pub/ itu-t/ oth/ 0A/ 0F/ T0A0F0000090001PDFE. 8. int/ ITU-T/ 50/ docs/ ITU-T_50. itu. itu.int (http:/ / www. p.itu. itu.1956-2006" (http:/ / www. html) [4] apdip. . [6] "CCITT . doc) [24] itu.int (http:/ / www.Issue No. int/ dms_pub/ itu-t/ oth/ 0A/ 0F/ T0A0F0000090001PDFE. html) [10] Telecom Standards Newsletter. itu. itu. edu/ group/ eng/ blog/ 2007/ 09/ itut_standards_now_freely_avai. itu.itu. int/ net/ about/ index.int/dms_pub/itu-t/oth/0A/0F/T0A0F0000090001PDFE. int/ ITU-T/ 50/ docs/ ITU-T_50.

research bodies and users — in fact. In 2010.Used when the document contains guidance on handling of technical standardization activities. ETSI is officially responsible for standardization of Information and Communication Technologies (ICT) within Europe.e. This deliverable type is approved and adopted by the Industry Specification Group that drafted it. GSM cell phone system and the TETRA professional mobile radio system. This list is gathered from the European Telecommunications Standards Institute website. including giving public availability to information not produced within a technical committee. plug-tests and fora hosting (i. • ETSI Technical Report (TR) . contract work and partner funding.Used for various purposes. ETSI has been successful in standardizing the Low Power Radio. A list of current members can be found here [1]. Based in Sophia Antipolis (France). documents that are dynamically generated by a query to a database via the web. it is approved by the technical committee that drafted it. An SR is published by the technical committee in which it was produced. validation and maintenance are essential. telecommunications series (EN) . encouraging governments to pass their wiretapping laws accordingly. Significant ETSI standardisation bodies include TISPAN (for fixed networks and Internet convergence) and M2M (for machine-to-machine communications). • ETSI Guide (EG) . ETSI has 740 members from 62 countries/provinces inside and outside Europe. • ETSI Standard (ES) . ETSI was criticised for the preemptive inclusion of wiretapping capabilities in their telecommunication standards. all the key players in the ICT arena. e. standardization organization in the telecommunications industry (equipment makers and network operators) in Europe. the hosting of forums[2] ). with contributions coming from members.[4] .Used when the document contains mainly informative elements. (see Lawful Interception) ETSI deliverable types • European Standard. with worldwide projection. service providers.Used when the document contains normative requirements and it is necessary to submit the document to the whole ETSI membership for approval. ETSI SRs are also used for 'virtual' documents.European Telecommunications Standards Institute 104 European Telecommunications Standards Institute The European Telecommunications Standards Institute (ETSI) is an independent. ETSI's budget exceeded €22 million. commercial activities like sale of documents. and is a partner in 3GPP. it is submitted to the whole ETSI membership for approval. These technologies include telecommunications. • ETSI Group Specification (GS) .g.Used by Industry Specification Groups according to the decision making procedures defined in the group's Terms of Reference. • ETSI Technical Specification (TS) . [3] About 40% goes towards operating expenses and the remaining 60% towards work programs including competency centres and special projects.Used when the document is intended to meet needs specific to Europe and requires transposition into national standards. network operators. ETSI was created by CEPT in 1988 and is officially recognized by the European Commission and the EFTA secretariat.Used when the document contains normative requirements and when short time-to-market. ETSI inspired the creation of. broadcasting and related areas such as intelligent transportation and medical electronics. • Special Report (SR) . it is approved by the technical committee that drafted it. non-profit. administrations. ETSI is a founding partner organization of the Global Standards Collaboration initiative. Short Range Device. or when the drafting of the document is required under an EC/EFTA mandate. including manufacturers.

etsi. pdf ETSI. etsi. org/ Portal_IntegrateAppli/ QueryResult. aspx http:/ / www.etsi. aspx) External links • Official Website (http://www.European Telecommunications Standards Institute 105 References [1] [2] [3] [4] http:/ / portal. asp?Alone=1& SortBy=& SortDirection=& Param= http:/ / www. org/ WebSite/ Standards/ ETSIDeliverables.org/) • NORMAPME. Trades and Small and Medium sized Enterprises for Standardisation .normapme.com). org/ WebSite/ OurServices/ Forapolislandingpage. etsi. org/ WebSite/ document/ AR2010. The European Office of Crafts.com (http://www.org (http:/ / www. etsi.

The ADs are responsible for appointing working group chairs. The working groups are organized into areas by subject matter. along with a charter that describes its focus. Each area is overseen by an area director (AD). government's National Security Agency. cooperating closely with the W3C and ISO/IEC standards bodies and dealing in particular with standards of the TCP/IP and Internet protocol suite. Operations and Management.S. The IAB also manages the Internet Research Task Force (IRTF). The mailing list consensus is the final arbiter of decision-making. the current chairperson is funded by VeriSign and the U. General. form the Internet Engineering Steering Group (IESG). . The IETF is overseen by the Internet Architecture Board (IAB). though their work is usually funded by their employers or sponsors. together with the IETF Chair. with most areas having two co-ADs. 1986 Standards Organization Creating standards applying to the internet to improve internet usability. Each group is intended to complete work on that topic and then disband. All participants and managers are volunteers. as it operates on rough consensus process. which oversees the IETF Administrative Support Activity (IASA). which oversees its external relationships. Current areas include: Applications. the WG will instead have its charter updated to take on new tasks as appropriate. with no formal membership or membership requirements. and what and when it is expected to produce. which is responsible for the overall operation of the IETF. support for the IETF. which provides logistical. The IETF is formally a part of the Internet Society. and relations with the RFC Editor. The IAB is also jointly responsible for the IETF Administrative Oversight Committee (IAOC). The groups will normally be closed once the work described in its charter is finished. Each working group has an appointed chairperson (or sometimes several co-chairs).[2] Organization The IETF is organized into a large number of working groups and informal discussion groups (BoF)s.Internet Engineering Task Force 106 Internet Engineering Task Force Internet Engineering Task Force Abbreviation Formation Type Purpose/focus Region served IETF Chair IETF January 16. Worldwide Russ Housley Parent organization Internet Society Website ietf. etc. Security. It is open to all who want to participate. and holds discussions on an open mailing list or at IETF meetings. In some cases.org [1] The Internet Engineering Task Force (IETF) develops and promotes Internet standards. Real-time Applications and Infrastructure. There is no voting procedure. The area directors. Routing. with which the IETF has a number of cross-group relations. Internet. for instance. and Transport. It is an open standards organization. each dealing with a specific topic.

This has allowed its protocols to be used in many different systems. Because it relies on volunteers and uses "rough consensus and running code" as its touchstone. review and independent testing by participants. meeting sponsors and by the Internet Society via its organizational membership and the proceeds of the Public Interest Registry. Work within the IETF on ways to improve the speed of the standards-making process is ongoing but. international activity associated with the Internet Society. Because the IETF does not have members (nor is it an organisation per se). to have 3 in North America. it has been meeting 3 times a year. The maximum attendance during the first 13 meetings was only 120 attendees. Attendance declined with industry restructuring during the early 2000s. and its standards are routinely re-used by bodies which create full-fledged architectures (e. and is currently around 1200. This occurred at the 12th meeting held during January 1989. . For protocols like SMTP. The initial meetings were very small. results can be slow whenever the number of volunteers is either too small to make progress. IRTF.S. government to an independent. but from 1991. The list of past and future meeting locations can be found on the IETF meetings [4] page.-government-funded researchers. Most of its specifications are focused on single protocols rather than tightly-interlocked systems. Since that time all IETF meetings have been open to the public. and meeting attendance is not required for contributors. The IETF has strived to hold the meetings near where most of the IETF volunteers are located. However. it had a maximum attendance of almost 3000 at the December 2000 IETF held in San Diego. IETF activities are funded by meeting fees. Initially. IETF meetings vary greatly in where they are held. with 2 in North America and 1 in either Europe or Asia (alternating between them every other year). or when volunteers lack the necessary expertise. 1986. 2 in Europe and 1 in Asia. 3GPP IMS). the goal was 3 meetings a year. Interoperability is the chief test for IETF specifications becoming standards. The goal ratio is currently. Operations The details of its operations have changed considerably as it has grown. or so large as to make consensus difficult.. Recently the IETF has set up an IETF Trust that manages the copyrighted materials produced by the IETF. but the basic mechanism remains publication of draft specifications. It was a continuation of the work of the earlier GADS Task Force. The majority of the IETF's work is done on mailing lists. For a long time. the Internet Society provides the financial and legal framework for the activities of the IETF and its sister bodies (IAB. which is used to transport e-mail for a user community in the many hundreds of millions.S. during October of that year.[3] During the early 1990s the IETF changed institutional form from an activity of the U..Internet Engineering Task Force 107 History The first IETF meeting was on January 16.g. corporate sponsorship of the meetings is typically a more important factor and this schedule has not been kept strictly in order to decrease operational costs. there is also considerable resistance to any change that is not fully backwards compatible. it met quarterly. These meetings have grown in both participation and scope a great deal since the early 1990s. CA. consisting of 21 U. because the number of volunteers with opinions on it is very great. with fewer than 35 people in attendance at each of the first five meetings.. consensus mechanisms on how to improve have been slow. during a two year period.). Representatives from non-governmental entities were invited starting with the fourth IETF meeting. and republication.

Retrieved 2008-08-05. ietf.Internet Engineering Task Force 108 Chairs The IETF Chairperson is selected by the NOMCOM process specified in RFC 3777 for a 2-year term.org/) • MyIETF (http://myietf.ietf.ietf.html) • The Tao of the IETF (http://tools. networkworld. . Carolyn (July 26.org/meeting/past.html) • Past IESG Members and IETF Chairs (http://www. . the IETF Chair was selected by the IAB.html) • Early IETF Proceedings (http://www. html). Retrieved 2008-04-20. one for each volume) • Past Meetings of the IETF (http://www. "Q&A: Security top concern for new IETF chair" (http:/ / www.org/old/2009/proceedings_directory. org/ meetings/ past.ietf. Before 1993.ietf. renewable. Network World (IDG).org) Personalized notification service on RFC's and drafts with full archive of old drafts etc. html). Retrieved 2007-03-23.html) (note: large pdf files. . • • • • • • • Mike Corrigan (1986) Phill Gross (1986–1994) Paul Mockapetris (1994–1996) Fred Baker (1996–2001) Harald Tveit Alvestrand (2001–2005) Brian Carpenter (2005–2007) Russ Housley (2007–)[5] References [1] http:/ / ietf.org/iesg/past-members.ietf.unfix.org/old/2009/proceedings/directory2. html). 2007). com/ news/ 2007/ 073007-ietf-qa. meetings.org/) • IETF Online Proceedings (http://www.ietf. [4] http:/ / www. org [2] Duffy Marsan. org/ ietf_chairs_year. ietf. org/ meeting/ [5] "IETF Chairs by year" (http:/ / www.org/rfcmarkup?doc=fyi17): details on how IETF is organized • IAOC information (http://iaoc. [3] "Past Meetings of the IETF" (http:/ / www. External links • The official IETF site (http://ietf. ietf. .

Jon Postel served as the RFC editor. the Internet Architecture Board (IAB). or innovations applicable to the working of the Internet and Internet-connected systems. Crocker started attributing the RFC series to the "Network Working Group". the precursor step before being approved as an RFC. was another of the four first ARPANET nodes and the source of early RFCs.[6] . it was anyone who wanted to join in on meetings and discussions about the project. History The inception of the RFC format occurred in 1969 as part of the seminal ARPANET project. Reynolds continued to be part of the team until October 13. Feinler to distribute them along with other network information. behaviors. directed by Douglas Engelbart. information. either for peer review or simply to convey new concepts. 2006. and—to some extent—the global community of computer network researchers in general. the RFC emerged from an early working group discussion between Steve Crocker. his obituary was published as RFC 2468. The authors of the first RFCs typewrote their work and circulated hard copies among the ARPA researchers. it is the official publication channel for the Internet Engineering Task Force (IETF). RFC 1. In July 2007. The ARC became the first Network Information Center.) Following the expiration of the original ARPANET contract with the U. Steve Carr and Jeff Rulifson. The Augmentation Research Center (ARC) at Stanford Research Institute. which was managed by Elizabeth J. This wasn't so much a formal committee as a loose association of researchers interested in the ARPANET project. splitting the task into several roles. Association Management Solutions with Glenn Kowack serving as interim series editor. The IAB can publish its own documents. and Alice Hagens in 2005. IETF documents came from IETF working groups or submissions sponsored by an IETF area director from the Internet Engineering Steering Group. The RFC leaves questions open and is written in a less formal style. entitled "Host Software". This less formal style is now typical of Internet Draft documents. was written by Steve Crocker of the University of California. In effect. and and independent stream includes documents from other outside sources. engineers and computer scientists may publish discourse in the form of an RFC. with standards defined for their style. federal government. "streams" of RFCs were defined. many of the early RFCs were requests for comments. (On his death in 1998.[2] Bob Braden took over the role of RFC project lead.[3] A new model was proposed in 2008. Sandy Ginoza joined USC/ISI in 1999 to work on RFC editing. Unlike the modern RFCs. A research stream of documents comes from the Internet Research Task Force (IRTF). research. In RFC 3. and published in August 2009.[5] In January 2010 the RFC editor function was moved to a contractor.S.[1] Today. the Internet Society (acting on behalf of the IETF) contracted with the Networking Division of the University of Southern California (USC) Information Sciences Institute (ISI) to assume the editorship and publishing responsibilities (under the direction of the IAB). In December 1969. 1969. The IETF adopts some of the proposals published as RFCs as Internet standards. while Joyce K. so the editing duties could be divided. Many of the subsequent RFCs of the 1970s also came from UCLA. a Request for Comments (RFC) is a memorandum published by the Internet Engineering Task Force (IETF) describing methods.[4] The streams were also refined in December 2009. because UCLA was one of the first Interface Message Processors (IMPs) on ARPANET. refined. which first defined the RFC series. researchers began distributing new RFCs via the newly operational ARPANET. or (occasionally) engineering humor. From 1969 until 1998. Through the Internet Society. and published on April 7. Although written by Steve Crocker. Los Angeles (UCLA).Request for Comments 109 Request for Comments In computer network engineering.

mandatory IETF RFCs not on standards track. In other words. an RFC is never rescinded or modified. informational RFCs promoted by the IETF as specified in RFC 1150 (FYI 1). obsolete. and are usually produced by experts participating in working groups. or even obsoleted (sic). RFC 5742. For more details about RFCs and the RFC process. FYI For Your Information. which first publish an Internet Draft. and (4) independent submission. Therefore. Once assigned a number and published. and RFC 5744.[7] The RFC production process differs from the standardization process of formal standards organizations such as ISO. some RFCs supersede others.Request for Comments 110 Production and evolution The RFC Editor assigns each RFC a unique serial number. the superseded RFCs are said to be deprecated. Revision 3". IRTF and independent  RFCs are supposed to contain relevant info or experiments for the Internet at large not in conflict with IETF work. (3) IAB. the quality is assessed by an independent submission editorial board. Only the IETF creates BCPs and RFCs on standards track.[7] Sub-series The RFC series contains three sub-series for IETF RFCs: BCP Best Current Practice. in order to keep the RFCs consistent and easy to understand. This approach facilitates initial rounds of peer review before documents mature into RFCs. In 2011 RFC 6360 obsoleted FYI 1 and concluded this sub-series. (2) IRTF. committee-driven process typical of ISO and national standards bodies. experience-driven. Internet technology experts may submit an Internet Draft without support from an external institution. STD Standard. Standards-track RFCs are published with approval from the IETF. The RFC tradition of pragmatic. Augmented Backus–Naur Form (ABNF) (as defined by RFC 5234) as a metalanguage. Together. after-the-fact standards authorship accomplished by individuals or small working groups can have important advantages over the more formal. Streams There are four streams of RFCs: (1) IETF. this used to be the third and highest maturity level of the IETF standards track specified in RFC 2026 (BCP 9). the serialized RFCs compose a continuous historical record of the evolution of Internet standards and practices. compare RFC 4846. the authors publish a revised document. see below. "The Internet Standards Process. Most RFCs use a common set of terms such as "MUST" and "NOT RECOMMENDED" (as defined by RFC 2119). see RFC 2026. An independent submission is checked by the IESG for conflicts with IETF work. if the document requires amendments. In 2011 RFC 6410 (a new part of BCP 9) reduced the standards track to two maturity levels. and simple text-based formatting. .

Draft Standard. For example. This status is one of the following: Informational. Status "experimental" An experimental RFC can be an IETF document or an individual submission to the 'RFC Editor'. e. example: http://purl.org/rfc/rfc5000. Only the IETF.[5] Status Not all RFCs are standards.[10] Each RFC is assigned a designation with regard to status within the Internet standardization process. or Historic (sic). when an Internet Standard is updated. However. The definitive list of Internet Standards is itself an Internet Standard. and The Tao of IETF  RFC 4677 (FYI 17). and especially later updates. can approve standards-track RFCs. however. in 2007 RFC 3700 was an Internet Standard—STD 1—and in May 2008 it was replaced with RFC 5000. it is submitted again and assigned a new RFC number. keywords. Standards Track. author(s). represented by the Internet Engineering Steering Group (IESG). RFC 5000 became an Internet Standard.g. Standards-track documents are further divided into Proposed Standard. publication date.net/net/rfc/5000 [9] The official International Standard Serial Number (ISSN) of the RFC series is 2070-1721. Some informational RFCs formed the FYI sub-series. Best Current Practice (BCP). BCP n refers to a certain RFC or set of RFCs. A redirection sets some efficient parameters. If an RFC becomes an Internet Standard (STD). status.Request for Comments 111 Obtaining RFCs The official source for RFCs on the World Wide Web is the RFC Editor [8]. and Internet Standard documents. When STD 1 is updated again. Experimental. the RFC Editor site offers a search form with many features. for example RFC 5000. STD n. the Internet User's Glossary  RFC 1983 (FYI 18). STD 1: Internet Official Protocol Standards. In theory it is indeed experimental. its number stays the same and it simply refers to a different RFC or set of RFCs. Some old FYIs are still interesting. but may also be available in other formats. but later the same standard may be updated to be RFC z instead. Best Current Practices work in a similar fashion. Each RFC is static. errata. Almost any individual published RFC. but it will still be STD 1. and as of May 2008 STD 1 is RFC 5000. can be retrieved via the URL: http://www.. but which RFC or RFCs may change over time. if the document is changed. including abstract. may be RFCs x and y at a given time. as of 2008 the definitive version of any standards-track specification is the ASCII version.rfc-editor. it is assigned an STD number but retains its RFC number. The term Historic is applied to deprecated standards-track documents or obsolete RFCs that were published before the standards track was established. so RFC 3700 changed to Historic. in practice some documents are not promoted on standards track because there are no volunteers for the procedural details.[11] Status "informational" An informational RFC can be nearly anything from April 1 jokes over proprietary protocols up to widely recognized essential RFCs like Domain Name System Structure and Delegation (RFC 1591). A given Internet Standard.txt Every RFC is submitted as plain ASCII text and is published in that form. . For easy access to the metadata of an RFC. it will simply refer to a newer RFC that will have completed the standards track.

org/ rfc. J. Postel. . org/ ibin/ c5i?mid=6& rid=49& gid=0& k1=934& k2=7324& tid=1263251951). where it is unclear which status the document would get if it were published today. org/ html/ bcp9). com/ web/ about/ ac123/ ac147/ archived_issues/ ipj_13-1/ 131_rfc. BCP 9 (http:/ / tools. [8] http:/ / www. ietf. ietf. "RFC Streams. Retrieved August 7. Status "unknown" Status unknown is used for some very old RFCs. not intended to specify a protocol. Internet Standard). [6] Glenn Kowack (January7. Crocker. . 2011. . 6 April 2009 (http:/ / www. . cisco. RFC Editor. 2008-05-25. Bradner (October 1996).[12] or IETF administration. ietf. 2010). May 2008. Experimental. Ed (August 2009). . txt) (plain text). org/ html/ bcp38 . or Historic. rfc-editor. org/ html/ rfc4844). org/ html/ rfc1796). Draft Standard. ietf. Olaf Kolkman (December 2009). The BCP series also covers technical recommendations for how to practice Internet standards. If a document only affects the Internet Standards Process. Retrieved August 17. Some obsolete RFCs are not classified as historic. Retrieved 2008-05-19. because the Internet standards process generally does not allow normative references from a standards track RFC to another RFC with lower status. documents a protocol that is not considered interesting in the current Internet. Kolkman. org/ rfc-index2. "Not All RFCs are Standards (RFC 1796)" (http:/ / tools. administrative procedure.Request for Comments 112 Status "best current practice" The best current practice (BCP) subseries collects administrative documents and other texts which are considered as official rules and not only informational. but some are on the standards track. The border between standards track and BCP is often unclear. S. C. The Internet Protocol Journal (Cisco Systems) 13 (1). . Retrieved 2008-05-26. "RFC Editor in Transition: Past. "The RFC Series and RFC Editor" (http:/ / tools. "RFC Editor Model (Version 1)". html?_r=1& em) [2] Leslie Daigle (March 2010). nytimes. and Boilerplates" (http:/ / tools. 2011. [3] Leslie Diagle (July 2007). few are interested in working through the required procedural details to get RFCs classified as historic and update all RFCs normatively depending on it. IETF. Status "historic" A historic RFC is one that has been made obsolete by a newer version. an early RFC was often just that: a simple request for comments. How the Internet Got Its Rules. "RFC Editor Transition Announcement" (http:/ / www.. [5] Leslie Diagle. html). or Standards Track (Proposed Standard. it is clearly a BCP. ietf. 2011. .. The New York Times. org/ html/ rfc5741). Headers. or anything else for which the RFC series is used today. RFC Editor. Retrieved August 17. [12] Scott O. RFC 5741." [11] "Internet Official Protocol Standards (STD 1)" (ftp:/ / ftp. ietf. html [9] http:/ / purl. but which do not affect over the wire data. References [1] Stephen D. html). Present. Some of these RFCs would not be published at all today. most of these documents are BCPs. Retrieved August 17. RFC 5620. The Internet Engineering Task Force. If it only defines rules and regulations for Internet Assigned Numbers Authority (IANA) registries it is less clear. RFC 4844. . [13] http:/ / tools. rfc-editor. like BCP 9. [7] "RFC Index" (http:/ / www. [4] O. 2011. and Future" (http:/ / www. ietf. or has been removed from the standards track for other reasons. Crocker. org/ in-notes/ std/ std1. net/ net/ rfc/ 5000 [10] Huitema. org/ html/ rfc2026). Also. for instance the recommendation to use source filtering to make DoS attacks more difficult (RFC 2827: "Network Ingress Filtering: Defeating Denial of Service Attacks which employ IP Source Address Spoofing") is BCP 38 [13]. Retrieved 2011-04-26. "RFC 2026: The Internet Standards Process — Revision 3" (http:/ / tools. Retrieved 2011-10-31. com/ 2009/ 04/ 07/ opinion/ 07crocker. "[E]ach RFC has a status…: Informational. rfc-editor. (April 1995).

workshops. Communications and Computer Engineering. The two organizations formally merged as the IEEE on January 1. 1959. and Ivan Getting (IEEE. .html) • RFC Errata (http://www. electronics engineers usually became members of the IRE.org/rfc. consumer electronics. After World War II. Hewlett (IRE. read I-Triple-E) is a non-profit professional association headquartered in New York City that is dedicated to advancing technological innovation and excellence. and in 1961. information technology. Notable Presidents of IEEE and its founding organizations include Elihu Thomson (AIEE.ietf. and was formed from two smaller organizations. and symposia (many of which have associated published proceedings). the IEEE serves as a major publisher of scientific journals and organizer of conferences.org/errata. the two organizations became increasingly competitive. The major interests of the AIEE were wire communications (telegraphy and telephony) and light and power systems. Electronics.rfc-editor.org/rfc.rfc-editor. 1930). the leadership of both the IRE and the AIEE resolved to consolidate the two organizations. the Society of Wireless and Telegraph Engineers and the Wireless Institute.org/rfcxx00.org/rfcfaq. Ernst Weber (IRE.html) • IETF's RFC page (http://www. biomedical technology and healthcare.[3] It was formed in 1963 by the merger of the Institute of Radio Engineers (IRE. also mentions what other RFCs this one "updates" or is "updated by". including electric power and energy. 1954). This article was originally based on material from the Free On-line Dictionary of Computing. Lee De Forest (IRE. 1889–1890).Request for Comments 113 External links • RFC Editor (http://www.org/rfc-index2. 1901–1902). Charles Proteus Steinmetz (AIEE. about 45% of whom reside outside the United States. IEEE. the allied branches of engineering and the related arts and sciences. IEEE's Constitution defines the purposes of the organization as "scientific and educational. It is also a leading standards development organization for the development of industrial standards (having developed over 900 active industry technical standards) in a broad range of disciplines. Terman (IRE. With the rise of electronics in the 1930s.org/) • RFC Database (http://www. William R. Frederick E. telecommunications.html) • RFC Index (http://tools. Alexander Graham Bell (AIEE. but the applications of electron tube technology became so extensive that the technical boundaries differentiating the IRE and the AIEE became difficult to distinguish. 1963. 1978). founded 1912) and the American Institute of Electrical Engineers (AIEE.rfc-editor. 1941).html) (text) • Official RFC standardization status (http://www."[4] In pursuing these goals. founded 1884). 1891–1892).000 members in more than 160 countries.rfc-editor.ietf. 1963).org/rfc/) (HTML) With the text of each RFC.rfc-editor. information assurance.html) • RFC Index (http://www.[1] [2] History The IEEE is incorporated under the Not-for-Profit Corporation Law of the state of New York in the United States. which is licensed under the GFDL. Institute of Electrical and Electronics Engineers The Institute of Electrical and Electronics Engineers (IEEE. It has more than 400.rfc-editor.php) • RFC Frequently Asked Questions (http://www. as well as Computer Science. directed toward the advancement of the theory and practice of Electrical. The IRE concerned mostly radio engineering.

certifications and courses at a 10% discount. the IEEE Philadelphia Section. the IEEE Computer Society).[10] Standards in Education[11] and Continuing Education Units (CEUs). offers on-line degree programs. encouraging engineers to seek Continuing Education Units (CEUs) for their participation in continuing education programs. with more than 300 local organizations that hold regular meetings. The Standards in Education website explains what standards are and the importance of developing and using them. the engineering and scientific communities. The IEEE logo is a diamond-shaped design which illustrates the right hand grip rule embedded in Benjamin Franklin's kite. and it is anticipated that IEEE Expert Now courses will feature in the CPD listing for South Africa. Currently.[12] Picture of the place where an office of IEEE works in the District University of Bogotá. the profession and the public in the United States.. with 1 CEU being equivalent to 10 PDHs. Educational activities The IEEE provides learning opportunities within the engineering sciences. twenty-nine states in the United States require Professional Development Hours (PDH) to maintain a Professional Engineering license. and the general public. The site includes tutorial modules and case illustrations to introduce the history of standards. research. organized around specialized technical fields. and nanotechnology. CEUs readily translate into Professional Development Hours (PDHs). such as South Africa. Education Partners. IEEE South Africa Section [6]) and technical focus (e. [5] IEEE has a dual complementary regional and technical structure . and it was created at the time of the 1963 merger. The IEEE includes 38 technical Societies.with organizational units based on geography (e. the basic terminology. aerospace. the IEEE also publishes tutorials and the standards that are produced by its standardization committees.[7] The published content in these journals as well as the content from several hundred annual conferences sponsored by the IEEE are available in the IEEE online digital library for subscription-based access and individual publication purchases.[8] In addition to journals and conference proceedings. designed for self-paced learning. book reviews and links to other sites that contain information on standards. similarly require continuing professional development (CPD) credits.g. their applications and impact on products. The IEEE Standards Association is in charge of the standardization activities of the IEEE.[9] the Education Partners Program. IEEE eLearning Library is a collection of online educational courses Colombia.. 114 Publications IEEE produces 30% of the world's literature in the electrical and electronics engineering and computer science fields. Countries outside the United States. IEEE offers educational opportunities such as IEEE eLearning Library. as well as news related to standards. exclusive for IEEE members.Institute of Electrical and Electronics Engineers transportation. IEEE develops and participates in educational activities such as accreditation of electrical engineering programs in institutes of higher learning. It manages a separate organizational unit (IEEE-USA) which recommends policies and implements programs specifically intended to benefit the members. The goal of the IEEE education programs is to ensure the growth of skill and knowledge in the electricity-related technical professions and to foster individual commitment to continuing education among IEEE members. . and technology. publishing well over 100 peer-reviewed journals.g.

An individual can join the IEEE as a student member. Students of age 8–18. a professional member can apply for Senior Membership. country. which is the highest level of recognition that a professional member can directly apply for.Institute of Electrical and Electronics Engineers IEEE also sponsors a website[13] designed to help young people understand better what engineering means. and certain IEEE officer positions are available only to Senior (and Fellow) Members. degree field. except the right to vote and hold certain offices. IEEE standards affect a wide range of industries including: power and energy. or urban). or Honorary members and fulfill other rigorous requirements of education. biomedical and healthcare. This grade of membership is conferred by the IEEE Board of Directors in recognition of a high level of demonstrated extraordinary accomplishment. One of the more notable IEEE standards is the IEEE 802 LAN/MAN group of standards which includes the IEEE 802. and teachers can explore the site to prepare for an engineering career. and cannot be applied for directly by the member – instead the candidate must be nominated by others. with 500 standards under development. • Senior Members: Upon meeting certain requirements. at the time of enrollment. telecommunications. In order to qualify for membership.. play interactive games. but the organization's wide scope of interests has attracted people in other disciplines as well (e. tuition ranges. An associate membership is available to individuals whose area of expertise falls outside the scope of the IEEE or who does not. and mathematicians. Senior Membership is also one of the requirements for those who are nominated and elevated to the grade IEEE Fellow. Applicants for Senior Member must have at least three letters of recommendation from Senior. and review lesson plans. suburban. room and board ranges. and location (rural. and how an engineering career can be made part of their future. mechanical and civil engineering) as well as biologists. explore curriculum links. professional member.11 Wireless Networking standard. nanotechnology.3 Ethernet standard and the IEEE 802. remarkable contribution. size of student body. the individual must fulfil certain academic or professional criteria and abide to the code of ethics and bylaws of the organization. Students and Associates have all the privileges of members. In 2005. • Members: Ordinary or professional Membership requires that the individual have graduated from a technology or engineering program of an appropriately-accredited institution of higher education or have demonstrated professional competence in technology or engineering through at least six years of professional work experience. IEEE had close to 900 active standards. • Fellow Members: The Fellow grade of membership is the highest level of membership. a distinctive honor. which allows a limited form of participation in the work of a particular IEEE Society. . This website also allows students to search for accredited engineering degree programs in Canada and the United States. 115 Standards and development process IEEE is one of the leading standards-making organizations in the world. and many more. • Society Affiliates: Some IEEE Societies also allow a person who is not an IEEE member to become a Society Affiliate of a particular Society within the IEEE. or associate member. information assurance. parents. The Senior Members are a selected group. achievement. transportation.g. Fellow. computer science. physicists. ask experts engineering-related questions. meet all the requirements for full membership. and experience in the field. Membership and member grades Most IEEE members are electrical and electronics engineers. There are several categories and levels of IEEE membership and affiliation: • Student Members: Student membership is available for a reduced fee to those who are enrolled in an accredited institution of higher education as undergraduate or graduate students in technology or engineering. IEEE performs its standards making and maintaining functions through the IEEE Standards Association (IEEE-SA). visitors are able to search by state/province/territory. Information Technology (IT).

Packaging. Jr. planning. • Life Members and Life Fellows: Members who have reached the age of 65 and whose number of years of membership plus their age in years adds up to at least 100 are recognized as Life Members – and. other than those provided by corporate sponsors for some awards. Education Medal IEEE Alexander Graham Bell Medal (for communications engineering) IEEE Simon Ramo Medal (for systems engineering) IEEE Medal for Engineering Excellence IEEE Medal for Environmental and Safety Technologies IEEE Medal in Power Engineering IEEE Richard W. are administered by the IEEE Foundation. may receive Honorary Membership from the IEEE Board of Directors. Flanagan Speech and Audio Processing Award IEEE Andrew S. and administration) IEEE James H. society and the engineering profession. Noyce Medal (for microelectronics) IEEE Medal for Innovations in Healthcare Technology IEEE/RSE Wolfson James Clerk Maxwell Award IEEE Centennial Medal Technical field awards • • • • • • • • IEEE Biomedical Engineering Award IEEE Cledo Brunetti Award (for nanotechnology and miniaturization) IEEE Components. 116 Awards Through its awards program. Hamming Medal (for information technology) IEEE Heinrich Hertz Medal (for electromagnetics) IEEE John von Neumann Medal (for computer-related technology) IEEE Jack S. as Life Fellows. the IEEE recognizes contributions that advance the fields of interest to the IEEE. Funds for the awards program. For nearly a century. and Manufacturing Technologies Award IEEE Control Systems Award IEEE Electromagnetics Award IEEE James L. in the case of Fellow members. Kilby Signal Processing Medal IEEE Dennis J.Institute of Electrical and Electronics Engineers • Honorary Members: Individuals who are not IEEE members but have demonstrated exceptional contributions. Grove Award (for solid-state devices) IEEE Herman Halperin Electric Transmission and Distribution Award • IEEE Masaru Ibuka Consumer Electronics Award • IEEE Internet Award . the IEEE Awards Program has paid tribute to technical professionals whose exceptional achievements and outstanding contributions have made a lasting impact on technology. Picard Medal for Radar Technologies and Applications IEEE Robert N. such as being a recipient of an IEEE Medal of Honor. Mulligan. Medals • • • • • • • • • • • • • • • • • • IEEE Medal of Honor IEEE Edison Medal IEEE Founders Medal (for leadership.

Institute of Electrical and Electronics Engineers • • • • • • • • • • • • • • • • • • • • • • • IEEE Reynold B. Johnson Data Storage Device Technology Award IEEE Reynold B. Johnson Information Storage Systems Award IEEE Richard Harold Kaufmann Award (for industrial systems engineering) IEEE Joseph F. Keithley Award in Instrumentation and Measurement IEEE Gustav Robert Kirchhoff Award (for electronic circuits and systems) IEEE Leon K. Kirchmayer Graduate Teaching Award IEEE Koji Kobayashi Computers and Communications Award IEEE William E. Newell Power Electronics Award IEEE Daniel E. Noble Award (for emerging technologies) IEEE Donald O. Pederson Award in Solid-State Circuits IEEE Frederik Philips Award (for management of research and development) IEEE Photonics Award IEEE Emanuel R. Piore Award (for information processing systems in computer science) IEEE Judith A. Resnik Award (for space engineering) IEEE Robotics and Automation Award IEEE Frank Rosenblatt Award (for biologically and linguistically motivated computational paradigms such as neural networks IEEE David Sarnoff Award (for electronics) IEEE Charles Proteus Steinmetz Award (for standardization) IEEE Marie Sklodowska-Curie Award (for nuclear and plasma engineering) IEEE Eric E. Sumner Award (for communications technology) IEEE Undergraduate Teaching Award IEEE Nikola Tesla Award (for power technology) IEEE Kiyo Tomiyasu Award (for technologies holding the promise of innovative applications)

117

Recognitions
• • • • • IEEE Haraden Pratt Award IEEE Richard M. Emberson Award IEEE Corporate Innovation Recognition IEEE Ernst Weber Engineering Leadership Recognition IEEE Honorary Membership

Prize paper awards
• IEEE Donald G. Fink Prize Paper Award • IEEE W.R.G. Baker Award

Scholarships
• IEEE Life Members Graduate Study Fellowship in Electrical Engineering was established by the IEEE in 2000. The fellowship is awarded annually to a first year, full time graduate student obtaining their masters for work in the area of electrical engineering, at an engineering school/program of recognized standing worldwide.[14] • IEEE Charles LeGeyt Fortescue Graduate Scholarship was established by the IRE in 1939 to commemorate Charles Legeyt Fortescue's contributions to electrical engineering. The scholarship is awarded for one year of full-time graduate work obtaining their masters in electrical engineering an ANE engineering school of recognized standing in the United States.[15]

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Societies
IEEE is supported by 38 societies, each one focused on a certain knowledge area. They provide specialized publications, conferences, business networking and sometimes other services.[16] [17]
• • • • • • • • • • • • • • • • • • • IEEE Aerospace and Electronic Systems Society IEEE Antennas & Propagation Society IEEE Broadcast Technology Society IEEE Circuits and Systems Society IEEE Communications Society IEEE Computational Intelligence Society IEEE Computer Society IEEE Consumer Electronics Society IEEE Control Systems Society IEEE Dielectrics & Electrical Insulation Society IEEE Education Society IEEE Electromagnetic Compatibility Society IEEE Electron Devices Society IEEE Engineering in Medicine and Biology Society IEEE Geoscience and Remote Sensing Society IEEE Industrial Electronics Society IEEE Industry Applications Society IEEE Information Theory Society • • • • • • • • • • • • • • • • • • IEEE Instrumentation & Measurement Society IEEE Intelligent Transportation Systems Society IEEE Magnetics Society IEEE Microwave Theory and Techniques Society IEEE Nuclear and Plasma Sciences Society IEEE Oceanic Engineering Society IEEE Photonics Society IEEE Power Electronics Society IEEE Power & Energy Society IEEE Product Safety Engineering Society IEEE Professional Communication Society IEEE Reliability Society IEEE Robotics and Automation Society IEEE Signal Processing Society IEEE Society on Social Implications of Technology IEEE Solid-State Circuits Society IEEE Systems, Man & Cybernetics Society IEEE Ultrasonics, Ferroelectrics & Frequency Control Society IEEE Vehicular Technology Society

IEEE Components, Packaging & Manufacturing Technology Society •

Technical councils
IEEE technical councils are collaborations of several IEEE societies on a broader knowledge area. There are currently seven technical councils:[16] [18] • • • • • • • IEEE Biometrics Council IEEE Council on Electronic Design Automation IEEE Nanotechnology Council IEEE Sensors Council IEEE Council on Superconductivity IEEE Systems Council IEEE Technology Management Council

Technical committees
To allow a quick response to new innovations, IEEE can also organize technical committees on top of their societies and technical councils. There are currently two such technical committees:[16] • IEEE Committee on Earth Observation (ICEO) • IEEE Technical Committee on RFID (CRFID)

Organizational units
• Technical Activities Board (TAB)

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119

IEEE Foundation
The IEEE Foundation is a charitable foundation established in 1973 to support and promote technology education, innovation and excellence.[19] It is incorporated separately from the IEEE, although it has a close relationship to it. Members of the Board of Directors of the foundation are required to be active members of IEEE, and one third of them must be current or former members of the IEEE Board of Directors. Initially, the IEEE Foundation's role was to accept and administer donations for the IEEE Awards program, but donations increased beyond what was necessary for this purpose, and the scope was broadened. In addition to soliciting and administering unrestricted funds, the foundation also administers donor-designated funds supporting particular educational, humanitarian, historical preservation, and peer recognition programs of the IEEE.[19] As of the end of 2009, the foundation's total assets were $27 million, split equally between unrestricted and donor-designated funds.[20]

Copyright policy
The IEEE requires authors to transfer their copyright for works they submit for publication.[21] [22] The IEEE generally does not create its own research. It is a professional organization that coordinates journal peer-review activities and holds subject-specific conferences in which authors present their research. The IEEE then publishes the authors' papers in journals and other proceedings, and authors are required to give up their exclusive rights to their works.[21] Section 6.3.1 IEEE Copyright Policies – subsections 7 and 8 – states that "all authors…shall transfer to the IEEE in writing any copyright they hold for their individual papers", but that the IEEE will grant the authors permission to make copies and use the papers they originally authored, so long as such use is permitted by the Board of Directors. The guidelines for what the Board considers a "permitted" use are not entirely clear, although posting a copy on a personally-controlled website is allowed. The author is also not allowed to change the work absent explicit approval from the organization. The IEEE justifies this practice in the first paragraph of that section, by stating that they will "serve and protect the interests of its authors and their employers".[21] [22] The IEEE places research papers and other publications such as IEEE standards behind a "pay wall"[21] , although the IEEE explicitly allows authors to make a copy of the papers that they authored freely available on their own website. As of September 2011, the IEEE also provides authors for most new journal papers with the option to pay to allow free download of their papers by the public from the IEEE publication website.[23] IEEE publications have received a Green[24] rating the from SHERPA/RoMEO guide[25] for affirming "authors and/or their companies shall have the right to post their IEEE-copyrighted material on their own servers without permission" (IEEE Publication Policy 8.1.9.D[26] ). This open access policy effectively allows authors, at their choice, to make their article openly available. Roughly 1/3 of the IEEE authors take this route . Some other professional associations do not impose the same requirements on authors. For example, the USENIX association[21] requires that the author only give up the right to publish the paper elsewhere for 12 months (in addition to allowing authors to post copies of the paper on their own website during that time). The organization operates successfully even though all of its publications are freely available online.[21]

2010.3 Technical activities objectives [5] "IEEE . their professions. . ieee. Retrieved November 7. html). ieee. html).tv/) — a video content website operated by the IEEE. 0/ us/ ).3. IEEE.IEEE Continuing Education Units (http:/ / www. IEEE. Most IEEE Journals are now Open Access (http:/ / theinstitute. October 2010. uk/ romeo. This section uses content available under the CC-BY-SA 3.ieee. Washington College of Law Intellectual Property Brief (American University). net/ 2011/ 03/ 12/ shaking-down-science-matt-blazes-criticism-of-the-acm-and-the-ieee/ ). org/ organizations/ foundation/ overview. ieee. org/ licenses/ by/ 3.ieee.org/) • IEEE Global History Network (http://www. org/ portal/ pages/ about/ awards/ sums/ lmfellow1. IEEE. Retrieved on 2010-01-23.1 IEEE Incorporation [4] "IEEE Technical Activities Board Operations Manual" (http:/ / www.0 License. 2010. org/ about/ volunteers/ tab_operations_manual. uk/ romeoinfo. [24] Sherpa Romeo color code (http:/ / www. za [7] About IEEE (http:/ / www. [18] "IEEE Technical Councils" (http:/ / www.tv (http://ieee. org/ web/ education/ ceus/ index. .0 United States License and hosted by Dan Rosenthal (http:/ / creativecommons. [15] Charles LeGeyt Fortescue Graduate Scholarship (http:/ / www.org/) — an online library of more than 200 self-study multimedia short courses and tutorials in technical fields of interest to the IEEE. its members. October 7. IEEE. ieee. [17] "IEEE Society Memberships" (http:/ / www. section 1. ac. 2010. html). . html) • External links • Official IEEE website (http://www. and their technologies. .1. html) [21] Johns. ieee. 2010. Retrieved November 7.org/) — a wiki-based website containing information about the history of IEEE. . ac. • IEEE eLearning Library (http://ieee-elearning. org/ about/ today/ at_a_glance. org/ portal/ pages/ about/ awards/ sums/ fortescue. html). [22] "6. . section 1. ipbrief. org/ publications_standards/ publications/ rights/ copyrightpolicy. pdf).D (http:/ / www. html#colours) [25] Sherpa Romeo site (http:/ / www. with over 2.org/) — the IEEE Xplore Digital Library. 2010. org/ web/ education/ partners/ eduPartners. ieee. • IEEE. Retrieved March 7. Retrieved 2011-04-17. html) [10] IEEE . [6] http:/ / www. [19] IEEE Foundation Home page (http:/ / www. Retrieved on 2010-01-23. [16] "IEEE Societies & Communities" (http:/ / www. Retrieved 2011-01-28.ieeeghn. . org/ organizations/ foundation/ index. org/ societies_communities/ societies/ about_technical_councils. Retrieved 2011-04-17. ieee. tryengineering. org/ portal/ cms_docs/ education/ setf/ index.The IEEE Standards Education pages have moved (http:/ / www. ieee. .9.. org/ ) [14] IEEE Life Member Graduate Study Fellowship (http:/ / www. 2011).1 IEEE Copyright Policies" (http:/ / www. org. [2] "IEEE 2009 Annual Report" (http:/ / www.6 million technical documents available online for purchase. 2011. sherpa. ieee. ieee. html) [20] IEEE Foundation Overview page (http:/ / www. IEEE. The Institute. html). Chris (March 12. org/ about/ ) [8] IEEE's online digital library (http:/ / ieeexplore. IEEE. org) [9] IEEE . . ieee. ieee. pdf). ieee. org/ publications_standards/ publications/ subscriptions/ prod/ elearning_overview. 2011. org/ societies_communities/ index. Retrieved November 8. org/ about/ volunteers/ tab_operations_manual. Amanda. IEEE. html) [12] IEEE . org/ briefings/ business/ most-ieee-journals-are-now-openaccess). Retrieved November 10. html). 2011. • IEEE Xplore (http://ieeexplore. [23] Davis. org/ documents/ ieee_annual_report_09_complete. ieee. Retrieved December 7.IEEE Education Partners Program (http:/ / www. ieee. www. 2010. ieee. org/ web/ publications/ rights/ policies. sherpa.. php) [26] IEEE Publication Policy 8. ieee.org.Institute of Electrical and Electronics Engineers 120 References [1] "IEEE at a Glance > IEEE Quick Facts" (http:/ / www. ieee.org (http:/ / www. ieee. The American University Washington College of Law Intellectual Property Brief is licensed by Dan Rosenthal under a Creative Commons Attribution 3. 2010. html#sect1). ieee.Master Brand and Logos" (http:/ / www. December 31. . Retrieved November 11. html) [13] Welcome to TryEngineering. . org/ about/ toolkit/ masterbrand/ index. html) (Available online).IEEE Expert Now (http:/ / www. pdf). ieee. html) [11] IEEE . IEEE. "Matt Blaze’s criticism of the ACM and the IEEE" (http:/ / www. ieee.ieee. org/ web/ membership/ societies/ index. [3] "IEEE Technical Activities Board Operations Manual" (http:/ / www.

[5] Within an SDP message there are three main sections. SDP started off as a component of the Session Announcement Protocol (SAP).[4] Whitespace is not allowed immediately to either side of the =. time. Values are typically a UTF-8 encoding. if present m= (media name and transport address) . SDP is designed to be extensible to support new media types and formats. and parameter negotiation.e. within the session. detailing the session. Each message may contain multiple timing and media descriptions. one per line. Session description A session is described by a series of fields. or media. and all associated properties. and media descriptions. i. Session description v= (protocol version) o= (originator and session identifier) s= (session name) i=* (session information) u=* (URI of description) e=* (email address) p=* (phone number) c=* (connection information—not required if included in all media) b=* (zero or more bandwidth information lines) One or more time descriptions ("t=" and "r=" lines.[2] SDP is intended for describing multimedia communication sessions for the purposes of session announcement. The set of properties and parameters are often called a session profile. Real-time Streaming Protocol (RTSP).[3] The form of each field is as follows. format. see below) z=* (time zone adjustments) k=* (encryption key) a=* (zero or more session attribute lines) Zero or more media descriptions Time description t= (time the session is active) r=* (zero or more repeat times) Media description. Session Initiation Protocol (SIP) and even as a standalone format for describing multicast sessions.[1] and subsequently published a revised specification as an IETF Proposed Standard as RFC 4566 in July 2006. SDP does not deliver media itself but is used for negotiation between end points of media type. but found other uses in conjunction with Real-time Transport Protocol (RTP). timing. <character>=<value> Where <character> is a single case-significant character and value is structured text whose format depends upon attribute type.[6] Optional values are specified with =* and each field must appear in the order shown below. Names are only unique within the associated syntactic construct.Session Description Protocol 121 Session Description Protocol The Session Description Protocol (SDP) is a format for describing streaming media initialization parameters. session invitation. The IETF published the original specification as an IETF Proposed Standard in April 1998.

org/ html/ rfc4566). with a single repeat time a week later at the same time can be represented as: t=3487140000 3487143600 r=604800 3600 0 Or using typed time: t=3487140000 3487143600 r=7d 3600 0 Notes [1] Handley. [4] session information and session name values are subject to the encoding specified in any charset attribute of the section. iana. New attributes are added to the standard occasionally through registration with IANA. Van Jacobson. hours ('h'). [3] Each line is separated from the next by a carriage return/line feed sequence. pdf) [7] Internet Assigned Numbers Authority SDP site (http:/ / www. "SDP: Session Description Protocol (RFC 4566)" (http:/ / tools. . Colin Perkins (2006-07). Attributes can appear within the Session or Media sections and are scoped accordingly as session-level or media-level. . Mark.Session Description Protocol i=* (media title) c=* (connection information—optional if included at session level) b=* (zero or more bandwidth information lines) k=* (encryption key) a=* (zero or more media attribute lines) 122 Attributes SDP uses attributes to extend the core protocol. Mark. Thus an hour meeting from 10am on 1 August 2010. Time Formats Absolute times are represented in Network Time Protocol format (the number of seconds since 1900). Van Jacobson (1998-04). org/ html/ rfc2327). Retrieved 2008-04-19. [5] RFC 4566 Section 5 [6] An In-Depth Overview of SDP (http:/ / www. "SDP: Session Description Protocol (RFC 2327)" (http:/ / tools. ietf. Retrieved 2008-04-19. konnetic. IETF. If the stop time is 0 then the session is "unbounded. IETF. minutes ('m') and seconds ('s')) sequence. org/ assignments/ sdp-parameters) . Implementations are allowed to relax this to omit the carriage return and supply only the line feed. • A value form: a=<attribute>:<value> provides a named parameter. [2] Handley. com/ Documents/ KonneticSDPTechnicalOverview. Intervals can be represented with Network Time Protocol times or in typed time: a value and time units (days ('d'). ietf." Unbounded and permanent sessions are discouraged but not prohibited." If the start time is also zero then the session is considered "permanent.[7] Attributes take two forms: • A property form: a=<flag> conveys a property of the session.

Session Description Protocol 123 References External links • Rosenberg. (2002-06). Retrieved 2010-07-25.org/html/rfc3264).ietf.. IETF. Schulzrinne. J. "An Offer/Answer Model with the Session Description Protocol(RFC 3264)" (http://tools. H. .

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Ananth t. Bobguy7. Regin LARSEN. Cmdawson. Chuckiesdad. GaelicWizard. Nikevich. Rick Sidwell. Tristantech. Kvng. Swpb. Roberta F. Ukexpat. HarisM. The Anome. ShaunMacPherson. Trevor MacInnis. Etu. Navedahmed123. Jncraton. BonsaiViking. Albanaco. Katalaveno. Bobo192.andrew. Acce245. Ellywa. Ryanmcdaniel.79. Moazami.php?oldid=461392792  Contributors: 16@r. Hm2k. Luna Santin. ^demon. Intelati. Vipinhari. Johnblade. Brian2wood. Alek Baka. Ale2006. JTN. Bansaribarot. Zahakiel. William Avery. SpaceRocket. Liempt. Corruptcopper. Matusz.org/w/index. Hollomis. Jmdavid1789. Rayward. Kyng. Wikiklrsc. Tezdog. MischaO. Wikibob. 237 anonymous edits Public switched telephone network  Source: http://en. TheOtherJesse. Longhair. Electron9. Amillar. Weylinp. Wayiran.. Ryamigo. Lerdsuwa. J. Rememberway. OttoTheFish. Krellis. Asimzaidi123. RaNo. Rich257. Jack Phoenix. Tiuks. Echuck215. Glane23. Helix84. SasiSasi. Isdnip. Manop. Jbonocore. Rich Farmbrough. Rkononenko. Epbr123. Boffin.xxx. TkGy. Thorpe. 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Alejo2083. 250 anonymous edits Voice over IP  Source: http://en. VoxLuna. Noono. Rahulmoksha. Denisutku. Barnacle157. J. Enric Naval. Brion VIBBER. Calle. Sir Arthur Williams. Kim Rubin. Mysid. XJamRastafire. JohnOwens. Avi. Sirgorpster. RyanCross. Pegasus1138. The Anome. Graham87. Zac439. Eeekster. IANCEE. Bonius. Crissov. Pfalstad. L Kensington. Barberio. Rohaq. Calltech. BobbyLee. NGNWiki.dorfman. Agnvoip. Glenn. R'n'B. Zvezda1111. Erik Sandberg. Rjwilmsi. Dgtsyb. Versus22. Looxix. 787 anonymous edits Signalling System No. SuperWiki. Accountwiki2. Fæ. Epolk. Gabi S. Jamelan. Inomyabcs. I am Me true. Reliablesources. William Avery. Adoresoftphone. Amungale. Moogwrench. Johnuniq. RedWolf. Charivari. Bernard François. Glenn. Nasz. Konman72. OwenX. DARTH SIDIOUS 2. Max Schwarz. Jnc. Casey Abell. Tarquin. Aggelos. Hemanshu. Globemasterthree. DavidDouthitt.. Yukoba. Template namespace initialisation script. X!. Drphilharmonic. Akram1997. Mike Rosoft. Conversion script. Goatbilly. Law. Atlant. Bender235. Here. Erodium. Nixdorf. Thingg. Rabbit67890. JustAGal. JTN. Mion. Kgfleischmann. Rkrikorian. Joaopaulo1511. Rgclegg. Biot.biboudis. Angelo. Mwanner. AJR. O. Nmacu. MikkoM. 203. Beland. Abdullais4u. Tide rolls. HenryLi. Dnas. DavidCary. ArkansasHistory. Avé. Eptin. Poweroid. Alphachimp. EoGuy. RexNL.delanoy. Guy Harris. Requestion. Jsoon eu. AlistairMcMillan. Ykhwong. Alansohn. Pillaidee. Antiuser. Agoode. Hannes Hirzel. Darkhalfactf. Coolcaesar. Dogcow. SEWilco. Paulkramer. Adrian Sampson. Falcon Kirtaran. Rsduhamel. Alohawolf. Fsiler. Tirupur123451. Beefman. Bert490. Kasperd. I-baLL. Alexander Straub. Steverapaport. Paul Koning. UncleBubba. Daniel Staal. Wik. Chetvorno. Bluezy. Suruena. Yekrats. Martyman. Carlo. Hcberkowitz. Edwardando. Manuelamsp. Txomin. ZNott. Varnav. Alvin-cs. Jl20777. Apapadop. Tobias Bergemann. Defyant. Addihockey10. Mandarax. THEN WHO WAS PHONE?. The Transhumanist (AWB). Shii. Xojo. Alex de carvalho. Shoeofdeath. Zondor. Lotje. JonHarder. Kyz.Article Sources and Contributors 124 Article Sources and Contributors Communications protocol  Source: http://en. Theresa knott. Graham87. Banes. Elmindreda. IlyaKorolev. TinaSDCE. Andre Engels. Fredrik. Cyclonenim. Inhumandecency. Yunghkim. Derek Ross. Harryzilber. Kaare. Bryan Derksen.roberts. DanielCD. Richardelainechambers. GordonMcKinney. Oxwil. Golbez. Swhitehead. Krauss. Bunnyhop11. Hairy Dude. Urvabara. Skarg. Scientizzle. Miym. Kubigula. Jim. Amalia07. Logictheo. Capricorn42. Denisarona. Bloodshedder. Mange01. Bridgecross. Ahoerstemeier. Doradus. Sietse Snel. 7  Source: http://en. Closedmouth. Pokeywiz. Petruss. John Vandenberg. Bradams. NickBush24. Timwi. Liangent. Krellis. Pmj. Ebernat. NMChico24. Stephan Leeds. Galmicmi. Beno1000. Kgfleischmann. Aelman. The Anome. Gloop. Expensivehat. Harryzilber. Dwheeler. Loupeter. Papadopa. Ranjandutta. Nixdorf. Philip Trueman. Joe102283. Patrick. Oli Filth. Jnc. Rich Farmbrough.. Bellhead. JanCeuleers. Kusma.php?oldid=463646455  Contributors: -ramz-.Ward. Seaphoto. Ruthherrin. EoGuy. Duffman. Oscarthecat. Johnuniq. Riick.xxx. Itai. B4hand. Thunderboltz. Zundark. Mange01. Crissov. Ferkelparade. Aldie. Noformation. Coralmizu. Rjd0060. Daniel. Master Conjurer. Intgr. Ashton1983. DeadEyeArrow. Jimgeorge.wikipedia. Biot. Radiojon. KnowledgeOfSelf. Xeesh. Chuunen Baka. 6birc. Dicklyon. NeoNorm. Badanedwa. AxelBoldt. Can't sleep. Acceptus. SimonInns. Jsoon eu. Mailtomeet. . Jatkins. Bennor. Matt Dunn. PPBlais. Una Smith. Tarquin. Bentogoa. Bluezy. Stephenb. Adclark88. CecilWard. Helix84. MER-C. Unyoyega. Austin512. Helland. Camilo Sanchez. JTN. Axcess. Aresnika. Mulad. Jorunn. DanielCD. Jimys salonika. Jim.Biboudis. Justsee. Joshaidan. Brim. Wknight94. Forton. Jeff Carr. Rfc1394. Public Menace. Sfisher. GaryW. Borislav. Coolcaesar. Beboy10. Rick Sidwell. SnoFox. CanadianLinuxUser. JeremyA. Aldie. Yyy. The Rambling Man. Coasting. DavidDW. Zipz0p. Velella. Equendil. Ian Geoffrey Kennedy. H. 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php?title=File:H.wikipedia.svg  Source: http://en.png  License: Public Domain  Contributors: Dewikipeder.php?title=File:Protocol_and_Software_layering.svg  License: GNU Free Documentation License  Contributors: en:User:Cburnett original work.gif  Source: http://en.svg  License: GNU Free Documentation License  Contributors: en:User:Kbrose File:UDP encapsulation.wikipedia.php?title=File:H.Image Sources.323_Stack.wikipedia.wikipedia.png  Source: http://en.org/w/index. Original uploader was ITU-T at en. Noir.323_High-level_call_flow.png  Source: http://en.org/w/index.svg  License: GNU Free Documentation License  Contributors: en:User:Cburnett original work.wikipedia.org/w/index.323 Architecture.png  License: Public Domain  Contributors: ITU-T (talk) Image:Ejemplo ENUM.wikipedia.245_exchange.php?title=File:A_typical_H.wikipedia.png  License: Public Domain  Contributors: ITU-T (talk).wikipedia.org/w/index.jpg  License: Public domain  Contributors: Conscious. Joey-das-WBF.svg  Source: http://en.wikipedia.svg  License: Creative Commons Attribution-Sharealike 3.jpg  License: Creative Commons Attribution-Sharealike 3.php?title=File:SIP_signaling.php?title=File:Ejemplo_ENUM.0  Contributors: Jsoon eu (talk) File:Protocol and Software layering.jpg  Source: http://en.php?title=File:SRI_Packet_Radio_Van.svg  Source: http://en.png  Source: http://en.php?title=File:IP_stack_connections.323_Architecture.org/w/index. Petr K Image:Call Forwarding with ENUM.wikipedia.php?title=File:Typical_H.org/w/index. Licenses and Contributors 128 Image Sources.jpg  License: Public domain  Contributors: Conscious. 1 anonymous edits Image:Http request telnet ubuntu.php?title=File:Internet_layering.svg  License: Creative Commons Attribution-Sharealike 3.jpg  Source: http://en.png  Source: http://en.org/w/index.org/w/index.svg  License: GNU Free Documentation License  Contributors: en:User:Kbrose Image:UDP encapsulation. Licenses and Contributors File:Internet layering.wikipedia File:H.0  Contributors: Jsoon eu (talk) File:Message flows.org/w/index.wikipedia. colorization by en:User:Kbrose File:SRI First Internetworked Connection diagram. Kozuch.jpg  License: Creative Commons Attribution-Sharealike 3.org/w/index.png  License: Public Domain  Contributors: ITU-T (talk) File:H.0  Contributors: User:Russavia File:SRI Packet Radio Van.wikipedia.org/w/index.org/w/index.323 High-level call flow.svg  License: Creative Commons Attribution-Sharealike 3.wikipedia.svg  License: Creative Commons Attribution-Sharealike 3.png  License: Public Domain  Contributors: ITU-T (talk) File:Establishment of an H. colorization by en:User:Kbrose File:Voip-typical.org/w/index.php?title=File:IEED_UD. Ysangkok.php?title=File:Establishment_of_an_H. Petr K File:IEED UD.wikipedia.245 exchange.jpg  License: Creative Commons Attribution-Sharealike 3.php?title=File:IP_stack_connections.wikipedia.php?title=File:Call_Forwarding_with_ENUM.png  License: Public Domain  Contributors: TheJosh File:Typical H.jpg  Source: http://en.svg  Source: http://en.0  Contributors: User:Countblack .org/w/index.png  Source: http://en.php?title=File:Voip-typical.php?title=File:SRI_First_Internetworked_Connection_diagram.wikipedia.org/w/index.gif  License: Creative Commons Attribution 1.323 Stack.png  Source: http://en. Joey-das-WBF.jpg  Source: http://en.0  Contributors: Jsoon eu (talk) File:Message flows and Routing.org/w/index.org/w/index.wikipedia.0  Contributors: User:Russavia Image:IP stack connections.org/w/index.php?title=File:Message_flows.wikipedia.svg  Source: http://en.org/w/index. Incnis Mrsi.wikipedia.wikipedia.php?title=File:UDP_encapsulation.svg  Source: http://en.jpg  Source: http://en. ‫זרם‬-‫טבעת‬ File:SIP signaling.php?title=File:UDP_encapsulation.org/w/index.php?title=File:Http_request_telnet_ubuntu.svg  Source: http://en.org/w/index.0  Contributors: Jsoon eu (talk) File:IP stack connections.0 Generic  Contributors: JMPerez.svg  Source: http://en.wikipedia.php?title=File:Message_flows_and_Routing.png  License: Public Domain  Contributors: ITU-T (talk) File:A typical H.png  Source: http://en.323 call.323_call.

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