The Session Initiation Protocol ( SIP ) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. SIP is used in Internet telephony , in private IP telephone systems, as well as mobile phone calling over LTE ( VoLTE ).
95-632: The protocol defines the specific format of messages exchanged and the sequence of communications for cooperation of the participants. SIP is a text-based protocol , incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP). A call established with SIP may consist of multiple media streams , but no separate streams are required for applications, such as text messaging , that exchange data as payload in
190-502: A MIKEY ( RFC 3830 ) exchange to SIP to determine session keys for use with SRTP. Text-based protocol A communication protocol is a system of rules that allows two or more entities of a communications system to transmit information via any variation of a physical quantity . The protocol defines the rules, syntax , semantics , and synchronization of communication and possible error recovery methods . Protocols may be implemented by hardware , software , or
285-402: A tunneling arrangement to accommodate the connection of dissimilar networks. For example, IP may be tunneled across an Asynchronous Transfer Mode (ATM) network. Protocol layering forms the basis of protocol design. It allows the decomposition of single, complex protocols into simpler, cooperating protocols. The protocol layers each solve a distinct class of communication problems. Together,
380-504: A SIP network, such as user agents, call routers, and voicemail boxes, are identified by a Uniform Resource Identifier (URI). The syntax of the URI follows the general standard syntax also used in Web services and e-mail. The URI scheme used for SIP is sip and a typical SIP URI has the form sip:username@domainname or sip:username@hostport , where domainname requires DNS SRV records to locate
475-669: A coarse hierarchy of functional layers defined in the Internet Protocol Suite . The first two cooperating protocols, the Transmission Control Protocol (TCP) and the Internet Protocol (IP) resulted from the decomposition of the original Transmission Control Program, a monolithic communication protocol, into this layered communication suite. The OSI model was developed internationally based on experience with networks that predated
570-427: A combination of both. Communicating systems use well-defined formats for exchanging various messages. Each message has an exact meaning intended to elicit a response from a range of possible responses predetermined for that particular situation. The specified behavior is typically independent of how it is to be implemented . Communication protocols have to be agreed upon by the parties involved. To reach an agreement,
665-513: A complex central network architecture and dumb endpoints (traditional telephone handsets). SIP is a client-server protocol of equipotent peers. SIP features are implemented in the communicating endpoints, while the traditional SS7 architecture is in use only between switching centers. The network elements that use the Session Initiation Protocol for communication are called SIP user agents . Each user agent (UA) performs
760-599: A computer environment (such as ease of mechanical parsing and improved bandwidth utilization ). Network applications have various methods of encapsulating data. One method very common with Internet protocols is a text oriented representation that transmits requests and responses as lines of ASCII text, terminated by a newline character (and usually a carriage return character). Examples of protocols that use plain, human-readable text for its commands are FTP ( File Transfer Protocol ), SMTP ( Simple Mail Transfer Protocol ), early versions of HTTP ( Hypertext Transfer Protocol ), and
855-426: A database of time zones for knowing when and where daylight adjustments will be needed, the repeat times are assumed to be all defined within the same time zone, and SDP supports the indication of NTP absolute times when a daylight offset (expressed in seconds or using a type time) will need to be applied to the repeated start time or end time falling at or after each daylight adjustment. All these offsets are relative to
950-405: A de facto standard operating system like Linux does not have this negative grip on its market, because the sources are published and maintained in an open way, thus inviting competition. Session Description Protocol The Session Description Protocol ( SDP ) is a format for describing multimedia communication sessions for the purposes of announcement and invitation. Its predominant use
1045-544: A direct connection can be made via Peer-to-peer SIP or via a VPN between the endpoints, most SIP communication involves multiple hops, with the first hop being from a user agent to the user agent's ITSP . For the multiple-hop case, SIPS will only secure the first hop; the remaining hops will normally not be secured with TLS and the SIP communication will be insecure. In contrast, the HTTPS protocol provides end-to-end security as it
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#17327721190501140-486: A hardware device or as a softphone . As vendors increasingly implement SIP as a standard telephony platform, the distinction between hardware-based and software-based SIP phones is blurred and SIP elements are implemented in the basic firmware functions of many IP-capable communications devices such as smartphones . In SIP, as in HTTP, the user agent may identify itself using a message header field ( User-Agent ), containing
1235-456: A machine rather than a human being. Binary protocols have the advantage of terseness, which translates into speed of transmission and interpretation. Binary have been used in the normative documents describing modern standards like EbXML , HTTP/2 , HTTP/3 and EDOC . An interface in UML may also be considered a binary protocol. Getting the data across a network is only part of the problem for
1330-457: A networking protocol, the protocol software modules are interfaced with a framework implemented on the machine's operating system. This framework implements the networking functionality of the operating system. When protocol algorithms are expressed in a portable programming language the protocol software may be made operating system independent. The best-known frameworks are the TCP/IP model and
1425-417: A packet-switched network, rather than this being a service of the network itself. His team was the first to tackle the highly complex problem of providing user applications with a reliable virtual circuit service while using a best-effort service , an early contribution to what will be the Transmission Control Protocol (TCP). Bob Metcalfe and others at Xerox PARC outlined the idea of Ethernet and
1520-439: A protocol may be developed into a technical standard . A programming language describes the same for computations, so there is a close analogy between protocols and programming languages: protocols are to communication what programming languages are to computations . An alternate formulation states that protocols are to communication what algorithms are to computation . Multiple protocols often describe different aspects of
1615-554: A protocol. The data received has to be evaluated in the context of the progress of the conversation, so a protocol must include rules describing the context. These kinds of rules are said to express the syntax of the communication. Other rules determine whether the data is meaningful for the context in which the exchange takes place. These kinds of rules are said to express the semantics of the communication. Messages are sent and received on communicating systems to establish communication. Protocols should therefore specify rules governing
1710-626: A public-domain Java implementation that serves as a reference implementation for the standard. The implementation can work in proxy server or user agent scenarios and has been used in numerous commercial and research projects. It supports RFC 3261 in full and a number of extension RFCs including RFC 6665 (event notification) and RFC 3262 (reliable provisional responses). Numerous other commercial and open-source SIP implementations exist. See List of SIP software . SIP-I, Session Initiation Protocol with encapsulated ISUP ,
1805-565: A reference model for communication standards led to the OSI model , published in 1984. For a period in the late 1980s and early 1990s, engineers, organizations and nations became polarized over the issue of which standard , the OSI model or the Internet protocol suite, would result in the best and most robust computer networks. The information exchanged between devices through a network or other media
1900-477: A response code in the form 1xx , and one or multiple final responses (2xx – 6xx). Transactions are further categorized as either type invite or type non-invite . Invite transactions differ in that they can establish a long-running conversation, referred to as a dialog in SIP, and so include an acknowledgment (ACK) of any non-failing final response, e.g., 200 OK . The Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions (SIMPLE)
1995-570: A session as a group of fields in a text-based format, one field per line. The form of each field is as follows. Where <character> is a single case-sensitive character and <value> is structured text in a format that depends on the character. Values are typically UTF-8 encoded. Whitespace is not allowed immediately to either side of the equal sign. Session descriptions consist of three sections: session, timing, and media descriptions. Each description may contain multiple timing and media descriptions. Names are only unique within
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#17327721190502090-478: A set of cooperating processes that manipulate shared data to communicate with each other. This communication is governed by well-understood protocols, which can be embedded in the process code itself. In contrast, because there is no shared memory , communicating systems have to communicate with each other using a shared transmission medium . Transmission is not necessarily reliable, and individual systems may use different hardware or operating systems. To implement
2185-673: A single communication. A group of protocols designed to work together is known as a protocol suite; when implemented in software they are a protocol stack . Internet communication protocols are published by the Internet Engineering Task Force (IETF). The IEEE (Institute of Electrical and Electronics Engineers) handles wired and wireless networking and the International Organization for Standardization (ISO) handles other types. The ITU-T handles telecommunications protocols and formats for
2280-432: A single repeat time a week later at the same time can be represented as: Or using typed time: When repeat times are specified, the start time of each repetition may need to be adjusted to compensate for daylight saving time changes so that it will occur at the same local time in a specific time zone throughout the period between the start time and the stop time. Instead of specifying this time zone and having to support
2375-456: A standardization process. Such protocols are referred to as de facto standards . De facto standards are common in emerging markets, niche markets, or markets that are monopolized (or oligopolized ). They can hold a market in a very negative grip, especially when used to scare away competition. From a historical perspective, standardization should be seen as a measure to counteract the ill-effects of de facto standards. Positive exceptions exist;
2470-418: A text description of the software, hardware, or the product name. The user agent field is sent in request messages, which means that the receiving SIP server can evaluate this information to perform device-specific configuration or feature activation. Operators of SIP network elements sometimes store this information in customer account portals, where it can be useful in diagnosing SIP compatibility problems or in
2565-430: A transfer mechanism of a protocol is comparable to a central processing unit (CPU). The framework introduces rules that allow the programmer to design cooperating protocols independently of one another. In modern protocol design, protocols are layered to form a protocol stack. Layering is a design principle that divides the protocol design task into smaller steps, each of which accomplishes a specific part, interacting with
2660-514: A user is allowed to make a call. A proxy interprets, and, if necessary, rewrites specific parts of a request message before forwarding it. SIP proxy servers that route messages to more than one destination are called forking proxies. The forking of a SIP request establishes multiple dialogs from the single request. Thus, a call may be answered from one of multiple SIP endpoints. For identification of multiple dialogs, each dialog has an identifier with contributions from both endpoints. A redirect server
2755-430: A web browser only acts as a client, and never as a server, SIP requires both peers to implement both roles. The roles of UAC and UAS only last for the duration of a SIP transaction. A SIP phone is an IP phone that implements client and server functions of a SIP user agent and provides the traditional call functions of a telephone, such as dial, answer, reject, call hold, and call transfer. SIP phones may be implemented as
2850-520: Is a protocol used to create, modify, and terminate communication sessions based on ISUP using SIP and IP networks. Services using SIP-I include voice, video telephony, fax and data. SIP-I and SIP-T are two protocols with similar features, notably to allow ISUP messages to be transported over SIP networks. This preserves all of the detail available in the ISUP header. SIP-I was defined by the ITU-T , whereas SIP-T
2945-502: Is a user agent server that generates 3xx (redirection) responses to requests it receives, directing the client to contact an alternate set of URIs. A redirect server allows proxy servers to direct SIP session invitations to external domains. A registrar is a SIP endpoint that provides a location service. It accepts REGISTER requests, recording the address and other parameters from the user agent. For subsequent requests, it provides an essential means to locate possible communication peers on
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3040-431: Is an audio stream on port 49170 using RTP/AVP payload type 0 (defined by RFC 3551 as PCMU ), and the second is a video stream on port 51372 using RTP/AVP payload type 99 (defined as "dynamic"). Finally, an attribute is included which maps RTP/AVP payload type 99 to format h263-1998 with a 90 kHz clock rate. RTCP ports for the audio and video streams of 49171 and 51373, respectively, are implied. The SDP specification
3135-407: Is commonly used for non-encrypted signaling traffic whereas port 5061 is typically used for traffic encrypted with Transport Layer Security (TLS). SIP-based telephony networks often implement call processing features of Signaling System 7 (SS7), for which special SIP protocol extensions exist, although the two protocols themselves are very different. SS7 is a centralized protocol, characterized by
3230-525: Is distinguished by its proponents for having roots in the Internet community rather than in the telecommunications industry . SIP has been standardized primarily by the Internet Engineering Task Force (IETF), while other protocols, such as H.323 , have traditionally been associated with the International Telecommunication Union (ITU). SIP is only involved in the signaling operations of a media communication session and
3325-474: Is done with a direct connection and does not involve the notion of hops. The media streams (audio and video), which are separate connections from the SIPS signaling stream, may be encrypted using SRTP. The key exchange for SRTP is performed with SDES ( RFC 4568 ), or with ZRTP ( RFC 6189 ). When SDES is used, the keys will be transmitted via insecure SIP unless SIPS is used. One may also add
3420-453: Is governed by rules and conventions that can be set out in communication protocol specifications. The nature of communication, the actual data exchanged and any state -dependent behaviors, is defined by these specifications. In digital computing systems, the rules can be expressed by algorithms and data structures . Protocols are to communication what algorithms or programming languages are to computations. Operating systems usually contain
3515-403: Is in support of streaming media applications, such as voice over IP (VoIP) and video conferencing . SDP does not deliver any media streams itself but is used between endpoints for negotiation of network metrics, media types, and other associated properties. The set of properties and parameters is called a session profile . SDP is extensible for the support of new media types and formats. SDP
3610-521: Is primarily used to set up and terminate voice or video calls. SIP can be used to establish two-party ( unicast ) or multiparty ( multicast ) sessions. It also allows modification of existing calls. The modification can involve changing addresses or ports , inviting more participants, and adding or deleting media streams. SIP has also found applications in messaging applications, such as instant messaging, and event subscription and notification. SIP works in conjunction with several other protocols that specify
3705-480: Is purely a format for session description. It is intended to be distributed over different transport protocols as necessary, including SAP , SIP , and RTSP . SDP could even be transmitted by email or as an HTTP payload. SDP uses attributes to extend the core protocol. Attributes can appear within the Session or Media sections and are scoped accordingly as session-level or media-level . New attributes are added to
3800-449: Is referred to as communicating sequential processes (CSP). Concurrency can also be modeled using finite state machines , such as Mealy and Moore machines . Mealy and Moore machines are in use as design tools in digital electronics systems encountered in the form of hardware used in telecommunication or electronic devices in general. The literature presents numerous analogies between computer communication and programming. In analogy,
3895-457: Is the SIP-based suite of standards for instant messaging and presence information . Message Session Relay Protocol (MSRP) allows instant message sessions and file transfer. The SIP developer community meets regularly at conferences organized by SIP Forum to test interoperability of SIP implementations. The TTCN-3 test specification language, developed by a task force at ETSI (STF 196),
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3990-408: Is the synchronization of software for receiving and transmitting messages of communication in proper sequencing. Concurrent programming has traditionally been a topic in operating systems theory texts. Formal verification seems indispensable because concurrent programs are notorious for the hidden and sophisticated bugs they contain. A mathematical approach to the study of concurrency and communication
4085-414: Is used for specifying conformance tests for SIP implementations. When developing SIP software or deploying a new SIP infrastructure, it is important to test the capability of servers and IP networks to handle certain call load: number of concurrent calls and number of calls per second. SIP performance tester software is used to simulate SIP and RTP traffic to see if the server and IP network are stable under
4180-776: The National Physical Laboratory in the United Kingdom, it was written by Roger Scantlebury and Keith Bartlett for the NPL network . On the ARPANET , the starting point for host-to-host communication in 1969 was the 1822 protocol , written by Bob Kahn , which defined the transmission of messages to an IMP. The Network Control Program (NCP) for the ARPANET, developed by Steve Crocker and other graduate students including Jon Postel and Vint Cerf ,
4275-423: The OSI model . At the time the Internet was developed, abstraction layering had proven to be a successful design approach for both compiler and operating system design and, given the similarities between programming languages and communication protocols, the originally monolithic networking programs were decomposed into cooperating protocols. This gave rise to the concept of layered protocols which nowadays forms
4370-638: The PARC Universal Packet (PUP) for internetworking. Research in the early 1970s by Bob Kahn and Vint Cerf led to the formulation of the Transmission Control Program (TCP). Its RFC 675 specification was written by Cerf with Yogen Dalal and Carl Sunshine in December 1974, still a monolithic design at this time. The International Network Working Group agreed on a connectionless datagram standard which
4465-761: The Transmission Control Protocol (TCP), and the Stream Control Transmission Protocol (SCTP). For secure transmissions of SIP messages over insecure network links, the protocol may be encrypted with Transport Layer Security (TLS). For the transmission of media streams (voice, video) the SDP payload carried in SIP messages typically employs the Real-time Transport Protocol (RTP) or the Secure Real-time Transport Protocol (SRTP). SIP
4560-547: The finger protocol . Text-based protocols are typically optimized for human parsing and interpretation and are therefore suitable whenever human inspection of protocol contents is required, such as during debugging and during early protocol development design phases. A binary protocol utilizes all values of a byte , as opposed to a text-based protocol which only uses values corresponding to human-readable characters in ASCII encoding. Binary protocols are intended to be read by
4655-590: The public switched telephone network (PSTN). As the PSTN and Internet converge , the standards are also being driven towards convergence. The first use of the term protocol in a modern data-commutation context occurs in April 1967 in a memorandum entitled A Protocol for Use in the NPL Data Communications Network. Under the direction of Donald Davies , who pioneered packet switching at
4750-450: The PSTN. Such services may simplify corporate information system infrastructure by sharing Internet access for voice and data, and removing the cost for Basic Rate Interface (BRI) or Primary Rate Interface (PRI) telephone circuits. SIP trunking is a similar marketing term preferred for when the service is used to simplify a telecom infrastructure by sharing the carrier access circuit for voice, data, and Internet traffic while removing
4845-461: The SIP RFC. Gateways can be used to interconnect a SIP network to other networks, such as the PSTN, which use different protocols or technologies. SIP is a text-based protocol with syntax similar to that of HTTP. There are two different types of SIP messages: requests and responses. The first line of a request has a method , defining the nature of the request, and a Request-URI, indicating where
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#17327721190504940-581: The SIP message. SIP works in conjunction with several other protocols that specify and carry the session media. Most commonly, media type and parameter negotiation and media setup are performed with the Session Description Protocol (SDP), which is carried as payload in SIP messages. SIP is designed to be independent of the underlying transport layer protocol and can be used with the User Datagram Protocol (UDP),
5035-456: The approval or support of a standards organization , which initiates the standardization process. The members of the standards organization agree to adhere to the work result on a voluntary basis. Often the members are in control of large market shares relevant to the protocol and in many cases, standards are enforced by law or the government because they are thought to serve an important public interest, so getting approval can be very important for
5130-450: The associated syntactic construct. Fields must appear in the order shown; optional fields are marked with an asterisk: Below is a sample session description from RFC 4566. This session is originated by the user "jdoe", at IPv4 address 10.47.16.5. Its name is "SDP Seminar" and extended session information ("A Seminar on the session description protocol") is included along with a link for additional information and an email address to contact
5225-448: The basis of protocol design. Systems typically do not use a single protocol to handle a transmission. Instead they use a set of cooperating protocols, sometimes called a protocol suite . Some of the best-known protocol suites are TCP/IP , IPX/SPX , X.25 , AX.25 and AppleTalk . The protocols can be arranged based on functionality in groups, for instance, there is a group of transport protocols . The functionalities are mapped onto
5320-429: The call load. The software measures performance indicators like answer delay, answer/seizure ratio , RTP jitter and packet loss , round-trip delay time . SIP connection is a marketing term for voice over Internet Protocol (VoIP) services offered by many Internet telephony service providers (ITSPs). The service provides routing of telephone calls from a client's private branch exchange (PBX) telephone system to
5415-442: The client-server model implemented in user agent clients and servers. A user agent is a logical network endpoint that sends or receives SIP messages and manages SIP sessions. User agents have client and server components. The user agent client (UAC) sends SIP requests. The user agent server (UAS) receives requests and returns a SIP response. Unlike other network protocols that fix the roles of client and server, e.g., in HTTP, in which
5510-442: The content being carried: text-based and binary. A text-based protocol or plain text protocol represents its content in human-readable format , often in plain text encoded in a machine-readable encoding such as ASCII or UTF-8 , or in structured text-based formats such as Intel hex format , XML or JSON . The immediate human readability stands in contrast to native binary protocols which have inherent benefits for use in
5605-414: The display of service status. A proxy server is a network server with UAC and UAS components that functions as an intermediary entity for the purpose of performing requests on behalf of other network elements. A proxy server primarily plays the role of call routing; it sends SIP requests to another entity closer to the destination. Proxies are also useful for enforcing policy, such as for determining whether
5700-673: The field of computer networking, it has been historically criticized by many researchers as abstracting the protocol stack in this way may cause a higher layer to duplicate the functionality of a lower layer, a prime example being error recovery on both a per-link basis and an end-to-end basis. Commonly recurring problems in the design and implementation of communication protocols can be addressed by software design patterns . Popular formal methods of describing communication syntax are Abstract Syntax Notation One (an ISO standard) and augmented Backus–Naur form (an IETF standard). Finite-state machine models are used to formally describe
5795-399: The first start time): As SDP announcements for repeated sessions should not be made to cover very long periods exceeding a few years, the number of daylight adjustments to include in the z= parameter should remain small. Sessions may be repeated irregularly over a week but scheduled the same way for all weeks in the period, by adding more tuples in the r parameter. For example, to schedule
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#17327721190505890-484: The function of a user agent client (UAC) when it is requesting a service function, and that of a user agent server (UAS) when responding to a request. Thus, any two SIP endpoints may in principle operate without any intervening SIP infrastructure. However, for network operational reasons, for provisioning public services to users, and for directory services, SIP defines several specific types of network server elements. Each of these service elements also communicates within
5985-418: The header fields, encoding rules and status codes of HTTP, providing a readable text-based format. SIP can be carried by several transport layer protocols including Transmission Control Protocol (TCP), User Datagram Protocol (UDP), and Stream Control Transmission Protocol (SCTP). SIP clients typically use TCP or UDP on port numbers 5060 or 5061 for SIP traffic to servers and other endpoints. Port 5060
6080-426: The horizontal message flows (and protocols) are between systems. The message flows are governed by rules, and data formats specified by protocols. The blue lines mark the boundaries of the (horizontal) protocol layers. The software supporting protocols has a layered organization and its relationship with protocol layering is shown in figure 5. To send a message on system A, the top-layer software module interacts with
6175-643: The internet as a reference model for general communication with much stricter rules of protocol interaction and rigorous layering. Typically, application software is built upon a robust data transport layer. Underlying this transport layer is a datagram delivery and routing mechanism that is typically connectionless in the Internet. Packet relaying across networks happens over another layer that involves only network link technologies, which are often specific to certain physical layer technologies, such as Ethernet . Layering provides opportunities to exchange technologies when needed, for example, protocols are often stacked in
6270-476: The layers make up a layering scheme or model. Computations deal with algorithms and data; Communication involves protocols and messages; So the analog of a data flow diagram is some kind of message flow diagram. To visualize protocol layering and protocol suites, a diagram of the message flows in and between two systems, A and B, is shown in figure 3. The systems, A and B, both make use of the same protocol suite. The vertical flows (and protocols) are in-system and
6365-427: The layers, each layer solving a distinct class of problems relating to, for instance: application-, transport-, internet- and network interface-functions. To transmit a message, a protocol has to be selected from each layer. The selection of the next protocol is accomplished by extending the message with a protocol selector for each layer. There are two types of communication protocols, based on their representation of
6460-442: The media format and coding and that carry the media once the call is set up. For call setup, the body of a SIP message contains a Session Description Protocol (SDP) data unit, which specifies the media format, codec and media communication protocol. Voice and video media streams are typically carried between the terminals using the Real-time Transport Protocol (RTP) or Secure Real-time Transport Protocol (SRTP). Every resource of
6555-402: The module directly below it and hands over the message to be encapsulated. The lower module fills in the header data in accordance with the protocol it implements and interacts with the bottom module which sends the message over the communications channel to the bottom module of system B. On the receiving system B the reverse happens, so ultimately the message gets delivered in its original form to
6650-483: The need for PRI circuits. SIP-enabled video surveillance cameras can initiate calls to alert the operator of events, such as the motion of objects in a protected area. 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. The U.S. National Institute of Standards and Technology (NIST), Advanced Networking Technologies Division provides
6745-785: The network. The location service links one or more IP addresses to the SIP URI of the registering agent. Multiple user agents may register for the same URI, with the result that all registered user agents receive the calls to the URI. SIP registrars are logical elements and are often co-located with SIP proxies. To improve network scalability, location services may instead be located with a redirect server. Session border controllers (SBCs) serve as middleboxes between user agents and SIP servers for various types of functions, including network topology hiding and assistance in NAT traversal . SBCs are an independently engineered solution and are not mentioned in
6840-470: The other parts of the protocol only in a small number of well-defined ways. Layering allows the parts of a protocol to be designed and tested without a combinatorial explosion of cases, keeping each design relatively simple. The communication protocols in use on the Internet are designed to function in diverse and complex settings. Internet protocols are designed for simplicity and modularity and fit into
6935-457: The possible interactions of the protocol. and communicating finite-state machines For communication to occur, protocols have to be selected. The rules can be expressed by algorithms and data structures. Hardware and operating system independence is enhanced by expressing the algorithms in a portable programming language. Source independence of the specification provides wider interoperability. Protocol standards are commonly created by obtaining
7030-401: The protocol, creating incompatible versions on their networks. In some cases, this was deliberately done to discourage users from using equipment from other manufacturers. There are more than 50 variants of the original bi-sync protocol. One can assume, that a standard would have prevented at least some of this from happening. In some cases, protocols gain market dominance without going through
7125-539: The protocol. The need for protocol standards can be shown by looking at what happened to the Binary Synchronous Communications (BSC) protocol invented by IBM . BSC is an early link-level protocol used to connect two separate nodes. It was originally not intended to be used in a multinode network, but doing so revealed several deficiencies of the protocol. In the absence of standardization, manufacturers and organizations felt free to enhance
7220-418: The request should be sent. The first line of a response has a response code . Requests initiate a functionality of the protocol. They are sent by a user agent client to the server and are answered with one or more SIP responses , which return a result code of the transaction, and generally indicate the success, failure, or other state of the transaction. Responses are sent by the user agent server indicating
7315-465: The responsible party, Jane Doe. This session is specified to last for two hours using NTP timestamps, with a connection address (which indicates the address clients must connect to or — when a multicast address is provided, as it is here — subscribe to) specified as IPv4 224.2.17.12 with a TTL of 127. Recipients of this session description are instructed to only receive media. Two media descriptions are provided, both using RTP Audio Video Profile. The first
7410-487: The result of a received request. Several classes of responses are recognized, determined by the numerical range of result codes: SIP defines a transaction mechanism to control the exchanges between participants and deliver messages reliably. A transaction is a state of a session, which is controlled by various timers. Client transactions send requests and server transactions respond to those requests with one or more responses. The responses may include provisional responses with
7505-429: The servers for SIP domain while hostport can be an IP address or a fully qualified domain name of the host and port. If secure transmission is required, the scheme sips is used. SIP employs design elements similar to the HTTP request and response transaction model. Each transaction consists of a client request that invokes a particular method or function on the server and at least one response. SIP reuses most of
7600-445: The session section. The parameters v , s , and o are mandatory, must not be empty, and should be UTF-8-encoded. They are used as identifiers and are not intended to be displayed to users. A few other attributes are also present in the example, either as a session-level attribute (such as the attribute in property form a=recvonly ), or as a media-level attribute (such as the attribute in value form a=rtpmap:99 h263-1998/90000 for
7695-612: The specification was revised in RFC 3261 and various extensions and clarifications have been published since. SIP was designed to provide a signaling and call setup protocol for IP-based communications supporting the call processing functions and features present in the public switched telephone network (PSTN) with a vision of supporting new multimedia applications. It has been extended for video conferencing , streaming media distribution, instant messaging , presence information , file transfer , Internet fax and online games . SIP
7790-442: The standard occasionally through registration with IANA. Attributes are either properties or values: Two of these attributes are specially defined: In both cases, text fields intended to be displayed to a user are interpreted as opaque strings, but rendered to the user or application with the values indicated in the last occurrence of the fields charset and sdplang in the current media section, or otherwise their last value in
7885-400: The start time on Sunday 1 August 2010 at 10am UTC), and this will be the only daylight adjustment to apply in the scheduled period which would occur between 1 August 2010 up to the 28 November 2010 at 10 am UTC (the stop time of the repeated 1-hour session which is repeated each week at the same local time, which occurs 88 days later), this can be specified as: If the weekly 1-hour session
7980-479: The start time, they are not cumulative. NTP supports this with field z , which indicates a series of pairs whose first item is the NTP absolute time when a daylight adjustment will occur, and the second item indicates the offset to apply relative to the absolute times computed with the field r . For example, if a daylight adjustment will subtract 1 hour on 31 October 2010 at 3 am UTC (i.e. 60 days minus 7 hours after
8075-514: The top module of system B. Program translation is divided into subproblems. As a result, the translation software is layered as well, allowing the software layers to be designed independently. The same approach can be seen in the TCP/IP layering. The modules below the application layer are generally considered part of the operating system. Passing data between these modules is much less expensive than passing data between an application program and
8170-506: The transmission. In general, much of the following should be addressed: Systems engineering principles have been applied to create a set of common network protocol design principles. The design of complex protocols often involves decomposition into simpler, cooperating protocols. Such a set of cooperating protocols is sometimes called a protocol family or a protocol suite, within a conceptual framework. Communicating systems operate concurrently. An important aspect of concurrent programming
8265-406: The transport layer. The boundary between the application layer and the transport layer is called the operating system boundary. Strictly adhering to a layered model, a practice known as strict layering, is not always the best approach to networking. Strict layering can have a negative impact on the performance of an implementation. Although the use of protocol layering is today ubiquitous across
8360-617: The video in the example). Absolute times are represented in Network Time Protocol (NTP) format (the number of seconds since 1900). If the stop time is 0 then the session is unbounded . If the start time is also zero then the session is considered permanent . Unbounded and permanent sessions are discouraged but not prohibited. Intervals can be represented with NTP times or in typed time: a value and time units (days: d , hours: h , minutes: m and seconds: s ) sequence. Thus an hour meeting from 10 am UTC on 1 August 2010, with
8455-502: Was defined by the IETF . Concerns about the security of calls via the public Internet have been addressed by encryption of the SIP protocol for secure transmission . The URI scheme SIPS is used to mandate that SIP communication be secured with Transport Layer Security (TLS). SIPS URIs take the form sips:user@example.com . End-to-end encryption of SIP is only possible if there is a direct connection between communication endpoints. While
8550-415: Was first implemented in 1970. The NCP interface allowed application software to connect across the ARPANET by implementing higher-level communication protocols, an early example of the protocol layering concept. The CYCLADES network, designed by Louis Pouzin in the early 1970s was the first to implement the end-to-end principle , and make the hosts responsible for the reliable delivery of data on
8645-698: Was originally a component of the Session Announcement Protocol (SAP), but found other uses in conjunction with the Real-time Transport Protocol (RTP), the Real-time Streaming Protocol (RTSP), Session Initiation Protocol (SIP), and as a standalone protocol for describing multicast sessions. The IETF published the original specification as a Proposed Standard in April 1998 (RFC 2327). Revised specifications were released in 2006 (RFC 4566), and in 2021 (RFC 8866). The Session Description Protocol describes
8740-596: Was originally designed by Mark Handley , Henning Schulzrinne , Eve Schooler and Jonathan Rosenberg in 1996 to facilitate establishing multicast multimedia sessions on the Mbone . The protocol was standardized as RFC 2543 in 1999. In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IP Multimedia Subsystem (IMS) architecture for IP-based streaming multimedia services in cellular networks . In June 2002
8835-588: Was presented to the CCITT in 1975 but was not adopted by the CCITT nor by the ARPANET. Separate international research, particularly the work of Rémi Després , contributed to the development of the X.25 standard, based on virtual circuits , which was adopted by the CCITT in 1976. Computer manufacturers developed proprietary protocols such as IBM's Systems Network Architecture (SNA), Digital Equipment Corporation's DECnet and Xerox Network Systems . TCP software
8930-479: Was redesigned as a modular protocol stack, referred to as TCP/IP. This was installed on SATNET in 1982 and on the ARPANET in January 1983. The development of a complete Internet protocol suite by 1989, as outlined in RFC 1122 and RFC 1123 , laid the foundation for the growth of TCP/IP as a comprehensive protocol suite as the core component of the emerging Internet . International work on
9025-408: Was repeated every Sunday for one full year, i.e. from Sunday 1 August 2010 3 am UTC to Sunday 26 June 2011 4 am UTC (stop time of the last repeat, i.e. 360 days plus 1 hour later, or 31107600 seconds later), so that it would include the transition back to Summer time on Sunday 27 March 2011 at 2 am (1 hour is added again to local time so that the second daylight transition would occur 209 days after
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