The Session Initiation Protocol (SIP) is a
signaling protocol A signaling protocol is a type of communications protocol for encapsulating the signaling between communication endpoints and switching systems to establish or terminate a connection and to identify the state of connection.
The following is a list ...
used for initiating, maintaining, and terminating
communication session
In computer science and networking in particular, a session is a time-delimited two-way link, a practical (relatively high) layer in the tcp/ip protocol enabling interactive expression and information exchange between two or more communication de ...
s that include voice, video and messaging applications.
SIP is used in
Internet telephony
Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. The terms Internet t ...
, in private IP telephone systems, as well as mobile phone calling over
LTE (
VoLTE
Voice over LTE (VoLTE) is an LTE high-speed wireless communication standard for mobile phones and data terminals, including Internet of things (IoT) devices and wearables. VoLTE has up to three times more voice and data capacity than older 3G U ...
).
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
The Hypertext Transfer Protocol (HTTP) is an application layer protocol in the Internet protocol suite model for distributed, collaborative, hypermedia information systems. HTTP is the foundation of data communication for the World Wide Web, ...
(HTTP) and the
Simple Mail Transfer Protocol
The Simple Mail Transfer Protocol (SMTP) is an Internet standard communication protocol for electronic mail transmission. Mail servers and other message transfer agents use SMTP to send and receive mail messages. User-level email clients typica ...
(SMTP).
A call established with SIP may consist of multiple
media streams, but no separate streams are required for applications, such as
text messaging
Text messaging, or texting, is the act of composing and sending electronic messages, typically consisting of alphabetic and numeric characters, between two or more users of mobile devices, desktops/laptops, or another type of compatible compute ...
, that exchange data as payload in 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
The Session Description Protocol (SDP) is a format for describing multimedia communication sessions for the purposes of announcement and invitation. Its predominant use is in support of streaming media applications, such as voice over IP (VoIP) ...
(SDP), which is carried as payload in SIP messages. SIP is designed to be independent of the underlying
transport layer
In computer networking, the transport layer is a conceptual division of methods in the layered architecture of protocols in the network stack in the Internet protocol suite and the OSI model. The protocols of this layer provide end-to-end ...
protocol and can be used with the
User Datagram Protocol
In computer networking, the User Datagram Protocol (UDP) is one of the core communication protocols of the Internet protocol suite used to send messages (transported as datagrams in packets) to other hosts on an Internet Protocol (IP) network. ...
(UDP), the
Transmission Control Protocol
The Transmission Control Protocol (TCP) is one of the main protocols of the Internet protocol suite. It originated in the initial network implementation in which it complemented the Internet Protocol (IP). Therefore, the entire suite is commonly ...
(TCP), and the
Stream Control Transmission Protocol
The Stream Control Transmission Protocol (SCTP) is a computer networking communications protocol in the transport layer of the Internet protocol suite. Originally intended for Signaling System 7 (SS7) message transport in telecommunication, the p ...
(SCTP). For secure transmissions of SIP messages over insecure network links, the protocol may be encrypted with
Transport Layer Security
Transport Layer Security (TLS) is a cryptographic protocol designed to provide communications security over a computer network. The protocol is widely used in applications such as email, instant messaging, and voice over IP, but its use in securi ...
(TLS). For the transmission of media streams (voice, video) the SDP payload carried in SIP messages typically employs the
Real-time Transport Protocol
The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applicatio ...
(RTP) or the
Secure Real-time Transport Protocol
The Secure Real-time Transport Protocol (SRTP) is a profile for Real-time Transport Protocol (RTP) intended to provide encryption, message authentication and integrity, and replay attack protection to the RTP data in both unicast and multicast ...
(SRTP).
History
SIP was originally designed by
Mark Handley
Mark Handley is a playwright and screenwriter.
In 1977, he and his wife moved to the Pacific Northwest where they lived in isolation in a log cabin that they built themselves. He is best known for his play ''Idioglossia
An idioglossia (from ...
,
Henning Schulzrinne
Henning Schulzrinne is a German-American computer scientist, who led research and development of the voice over IP network protocols.
Life
Schulzrinne studied engineering management at the Department of Electrical Engineering and Information T ...
, Eve Schooler and
Jonathan Rosenberg in 1996 to facilitate establishing
multicast
In computer networking, multicast is group communication where data transmission is addressed to a group of destination computers simultaneously. Multicast can be one-to-many or many-to-many distribution. Multicast should not be confused with ...
multimedia sessions on the
Mbone
Mbone (short for "multicast backbone") was an experimental backbone and virtual network built on top of the Internet for carrying IP multicast traffic on the Internet. It was developed in the early 1990s and required specialized hardware and soft ...
. The protocol was standardized as in 1999. In November 2000, SIP was accepted as a
3GPP signaling protocol and permanent element of the
IP Multimedia Subsystem
The IP Multimedia Subsystem or IP Multimedia Core Network Subsystem (IMS) is a standardised architectural framework for delivering IP multimedia services. Historically, mobile phones have provided voice call services over a circuit-switched-styl ...
(IMS) architecture for IP-based streaming multimedia services in
cellular network
A cellular network or mobile network is a communication network where the link to and from end nodes is wireless. The network is distributed over land areas called "cells", each served by at least one fixed-location transceiver (typically thre ...
s. In June 2002 the specification was revised in 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
The public switched telephone network (PSTN) provides Communications infrastructure, infrastructure and services for public Telecommunications, telecommunication. The PSTN is the aggregate of the world's circuit-switched telephone networks that ...
(PSTN) with a vision of supporting new multimedia applications. It has been extended for
video conferencing
Videotelephony, also known as videoconferencing and video teleconferencing, is the two-way or multipoint reception and transmission of audio and video signals by people in different locations for real time communication.McGraw-Hill Concise Ency ...
,
streaming media
Streaming media is multimedia that is delivered and consumed in a continuous manner from a source, with little or no intermediate storage in network elements. ''Streaming'' refers to the delivery method of content, rather than the content it ...
distribution,
instant messaging
Instant messaging (IM) technology is a type of online chat allowing real-time text transmission over the Internet or another computer network. Messages are typically transmitted between two or more parties, when each user inputs text and trigge ...
,
presence information In computer and telecommunications networks, presence information is a status indicator that conveys ability and willingness of a potential communication partner—for example a user—to communicate. A user's client provides presence information (p ...
,
file transfer File transfer is the transmission of a computer file through a communication channel from one computer system to another. Typically, file transfer is mediated by a communications protocol. In the history of computing, numerous file transfer protocol ...
,
Internet fax
Internet fax, e-fax, or online fax is the use of the internet and internet protocols to send a fax (facsimile), rather than using a standard telephone connection and a fax machine. A distinguishing feature of Internet fax, compared to other Intern ...
and
online game
An online game is a video game that is either partially or primarily played through the Internet or any other computer network available. Online games are ubiquitous on modern gaming platforms, including PC game, PCs, Console game, consoles and ...
s.
SIP is distinguished by its proponents for having roots in the Internet community rather than in the
telecommunications industry The telecommunications industries within the sector of information and communication technology is made up of all telecommunications/telephone companies and internet service providers and plays a crucial role in the evolution of mobile communication ...
. SIP has been standardized primarily by the
Internet Engineering Task Force
The Internet Engineering Task Force (IETF) is a standards organization for the Internet and is responsible for the technical standards that make up the Internet protocol suite (TCP/IP). It has no formal membership roster or requirements and a ...
(IETF), while other protocols, such as
H.323
H.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. The H.323 standard addresses call signaling and control, m ...
, have traditionally been associated with the
International Telecommunication Union
The International Telecommunication Union is a specialized agency of the United Nations responsible for many matters related to information and communication technologies. It was established on 17 May 1865 as the International Telegraph Unio ...
(ITU).
Protocol operation
![SIP session setup example](https://upload.wikimedia.org/wikipedia/commons/9/9c/SIP_session_setup_example.svg)
SIP is only involved in the signaling operations of a media communication session and is primarily used to set up and terminate voice or video calls. SIP can be used to establish two-party (
unicast
Unicast is data transmission from a single sender (red) to a single receiver (green). Other devices on the network (yellow) do not participate in the communication.
In computer networking, unicast is a one-to-one transmission from one point in ...
) or multiparty (
multicast
In computer networking, multicast is group communication where data transmission is addressed to a group of destination computers simultaneously. Multicast can be one-to-many or many-to-many distribution. Multicast should not be confused with ...
) sessions. It also allows modification of existing calls. The modification can involve changing addresses or
ports
A port is a maritime facility comprising one or more wharves or loading areas, where ships load and discharge cargo and passengers. Although usually situated on a sea coast or estuary, ports can also be found far inland, such as Ham ...
, 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 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
The Session Description Protocol (SDP) is a format for describing multimedia communication sessions for the purposes of announcement and invitation. Its predominant use is in support of streaming media applications, such as voice over IP (VoIP) ...
(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
The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applicatio ...
(RTP) or
Secure Real-time Transport Protocol
The Secure Real-time Transport Protocol (SRTP) is a profile for Real-time Transport Protocol (RTP) intended to provide encryption, message authentication and integrity, and replay attack protection to the RTP data in both unicast and multicast ...
(SRTP).
Every resource of a SIP network, such as user agents, call routers, and voicemail boxes, are identified by a
Uniform Resource Identifier
A Uniform Resource Identifier (URI) is a unique sequence of characters that identifies a logical or physical resource used by web technologies. URIs may be used to identify anything, including real-world objects, such as people and places, conc ...
(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 record
A Service record (SRV record) is a specification of data in the Domain Name System defining the location, i.e., the hostname and port number, of servers for specified services. It is defined iRFC 2782 and its type code is 33. Some Internet protoco ...
s to locate the servers for SIP domain while ''hostport'' can be an
IP address
An Internet Protocol address (IP address) is a numerical label such as that is connected to a computer network that uses the Internet Protocol for communication.. Updated by . An IP address serves two main functions: network interface ident ...
or a
fully qualified domain name
A fully qualified domain name (FQDN), sometimes also referred to as an ''absolute domain name'', is a domain name that specifies its exact location in the tree hierarchy of the Domain Name System (DNS). It specifies all domain levels, including th ...
of the host and port. If
secure transmission
In computer science, secure transmission refers to the transfer of data such as confidential or proprietary information over a secure channel. Many secure transmission methods require a type of encryption. The most common email encryption is called ...
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 the header fields, encoding rules and status codes of HTTP, providing a readable text-based format.
SIP can be carried by several
transport layer
In computer networking, the transport layer is a conceptual division of methods in the layered architecture of protocols in the network stack in the Internet protocol suite and the OSI model. The protocols of this layer provide end-to-end ...
protocols including
Transmission Control Protocol
The Transmission Control Protocol (TCP) is one of the main protocols of the Internet protocol suite. It originated in the initial network implementation in which it complemented the Internet Protocol (IP). Therefore, the entire suite is commonly ...
(TCP),
User Datagram Protocol
In computer networking, the User Datagram Protocol (UDP) is one of the core communication protocols of the Internet protocol suite used to send messages (transported as datagrams in packets) to other hosts on an Internet Protocol (IP) network. ...
(UDP), and
Stream Control Transmission Protocol
The Stream Control Transmission Protocol (SCTP) is a computer networking communications protocol in the transport layer of the Internet protocol suite. Originally intended for Signaling System 7 (SS7) message transport in telecommunication, the p ...
(SCTP).
SIP clients typically use TCP or UDP on
port number
In computer networking, a port is a number assigned to uniquely identify a connection endpoint and to direct data to a specific service. At the software level, within an operating system, a port is a logical construct that identifies a specific ...
s 5060 or 5061 for SIP traffic to servers and other endpoints. Port 5060 is commonly used for non-encrypted signaling traffic whereas port 5061 is typically used for traffic encrypted with
Transport Layer Security
Transport Layer Security (TLS) is a cryptographic protocol designed to provide communications security over a computer network. The protocol is widely used in applications such as email, instant messaging, and voice over IP, but its use in securi ...
(TLS).
SIP-based telephony networks often implement call processing features of
Signaling System 7
Signalling System No. 7 (SS7) is a set of telephony signaling protocols developed in 1975, which is used to set up and tear down telephone calls in most parts of the world-wide public switched telephone network (PSTN). The protocol also perform ...
(SS7), for which special SIP protocol extensions exist, although the two protocols themselves are very different. SS7 is a centralized protocol, characterized by 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.
Network elements
The network elements that use the Session Initiation Protocol for communication are called ''SIP user agents''. Each ''user agent'' (UA) performs 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 the client-server model implemented in user agent clients and servers.
User agent
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 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
A VoIP phone or IP phone uses voice over IP technologies for placing and transmitting telephone calls over an IP network, such as the Internet. This is in contrast to a standard phone which uses the traditional public switched telephone network ...
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 a hardware device or as a
softphone
A softphone is a software program for making telephone calls over the Internet using a general purpose computer rather than dedicated hardware. The softphone can be installed on a piece of equipment such as a desktop, mobile device, or other compu ...
. 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
smartphone
A smartphone is a portable computer device that combines mobile telephone and computing functions into one unit. They are distinguished from feature phones by their stronger hardware capabilities and extensive mobile operating systems, whic ...
s.
In SIP, as in HTTP, the
user agent
In computing, a user agent is any software, acting on behalf of a user, which "retrieves, renders and facilitates end-user interaction with Web content". A user agent is therefore a special kind of software agent.
Some prominent examples of us ...
may identify itself using a message header field (''User-Agent''), containing 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 the display of service status.
Proxy server
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 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.
Redirect server
A redirect server 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.
Registrar
![SIP-registration-flow](https://upload.wikimedia.org/wikipedia/commons/e/ed/SIP-registration-flow.png)
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 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 controller
Session border controller
A session border controller (SBC) is a network element deployed to protect SIP based voice over Internet Protocol (VoIP) networks.
Early deployments of SBCs were focused on the borders between two service provider networks in a peering environme ...
s (SBCs) serve as
middlebox
A middlebox is a computer networking device that transforms, inspects, filters, and manipulates traffic for purposes other than packet forwarding. Examples of middleboxes include firewalls, network address translators (NATs), load balancers, and ...
es between user agents and SIP servers for various types of functions, including network topology hiding and assistance in
NAT traversal
Network address translation traversal is a computer networking technique of establishing and maintaining Internet protocol connections across gateways that implement network address translation (NAT).
NAT traversal techniques are required for m ...
. SBCs are an independently engineered solution and are not mentioned in the SIP RFC.
Gateway
Gateways can be used to interconnect a SIP network to other networks, such as the PSTN, which use different protocols or technologies.
SIP messages
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 the request should be sent. The first line of a response has a ''response code''.
Requests
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
The Session Initiation Protocol (SIP) is a signalling (telecommunications), signalling communications protocol, protocol used for controlling communication sessions such as Voice over IP telephone calls. SIP is based on request/response transaction ...
, which return a result code of the transaction, and generally indicate the success, failure, or other state of the transaction.
Responses
Responses are sent by the user agent server indicating the result of a received request. Several classes of responses are recognized, determined by the numerical range of result codes:
* 1xx: Provisional responses to requests indicate the request was valid and is being processed.
* 2xx: Successful completion of the request. As a response to an INVITE, it indicates a call is established. The most common code is 200, which is an unqualified success report.
* 3xx: Call redirection is needed for completion of the request. The request must be completed with a new destination.
* 4xx: The request cannot be completed at the server for a variety of reasons, including bad request syntax (code 400).
* 5xx: The server failed to fulfill an apparently valid request, including server internal errors (code 500).
* 6xx: The request cannot be fulfilled at any server. It indicates a global failure, including call rejection by the destination.
Transactions
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 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''.
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
Instant messaging (IM) technology is a type of online chat allowing real-time text transmission over the Internet or another computer network. Messages are typically transmitted between two or more parties, when each user inputs text and trigge ...
and
presence information In computer and telecommunications networks, presence information is a status indicator that conveys ability and willingness of a potential communication partner—for example a user—to communicate. A user's client provides presence information (p ...
.
Message Session Relay Protocol In computer networking, the Message Session Relay Protocol (MSRP) is a protocol for transmitting a series of related instant messages in the context of a communications session. An application instantiates the session with the Session Description Pr ...
(MSRP) allows instant message sessions and file transfer.
Conformance testing
The SIP developer community meets regularly at conferences organized by SIP Forum to test interoperability of SIP implementations. The
TTCN-3 TTCN-3 ''(Testing and Test Control Notation version 3)'' is a strongly typed testing language used in conformance testing of communicating systems. TTCN-3 is written by ETSI in the ES 201 873 series, and standardized by ITU-T in the Z.160 Series.
T ...
test specification language, developed by a task force at
ETSI
The European Telecommunications Standards Institute (ETSI) is an independent, not-for-profit, standardization organization in the field of information and communications. ETSI supports the development and testing of global technical standard ...
(STF 196), is used for specifying conformance tests for SIP implementations.
Performance testing
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 the call load.
The software measures performance indicators like answer delay,
answer/seizure ratio The answer-seizure ratio (ASR) is the percentage of telephone calls which are answered, a measure of network qualityITU SG2 Recommendation E.411: International network management - Operational guidance. and call success rates in telecommunications. ...
, RTP
jitter
In electronics and telecommunications, jitter is the deviation from true periodicity of a presumably periodic signal, often in relation to a reference clock signal. In clock recovery applications it is called timing jitter. Jitter is a significa ...
and
packet loss
Packet loss occurs when one or more packets of data travelling across a computer network fail to reach their destination. Packet loss is either caused by errors in data transmission, typically across wireless networks, or network congestion.Kur ...
,
round-trip delay time
In telecommunications, round-trip delay (RTD) or round-trip time (RTT) is the amount of time it takes for a signal to be sent ''plus'' the amount of time it takes for acknowledgement of that signal having been received. This time delay includes p ...
.
Applications
''SIP connection'' is a marketing term for
voice over Internet Protocol
Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. The terms Internet t ...
(VoIP) services offered by many
Internet telephony service provider An Internet telephony service provider (ITSP) offers digital telecommunications services based on Voice over Internet Protocol (VoIP) that are provisioned via the Internet.
ITSPs provide services to end-users directly or as whole-sale suppliers to ...
s (ITSPs). The service provides routing of telephone calls from a client's
private branch exchange
A business telephone system is a multiline telephone system typically used in business environments, encompassing systems ranging in technology from the key telephone system (KTS) to the private branch exchange (PBX).
A business telephone syst ...
(PBX) telephone system to the PSTN. Such services may simplify corporate information system infrastructure by sharing
Internet access
Internet access is the ability of individuals and organizations to connect to the Internet using computer terminals, computers, and other devices; and to access services such as email and the World Wide Web. Internet access is sold by Internet ...
for voice and data, and removing the cost for
Basic Rate Interface
Basic Rate Interface (BRI, 2B+D, 2B1D) or Basic Rate Access is an Integrated Services Digital Network (ISDN) configuration intended primarily for use in subscriber lines similar to those that have long been used for voice-grade telephone service ...
(BRI) or
Primary Rate Interface (PRI) telephone circuits.
SIP trunking SIP trunking is a voice over Internet Protocol (VoIP) technology and streaming media service based on the Session Initiation Protocol (SIP) by which Internet telephony service providers (ITSPs) deliver telephone services and unified communications t ...
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 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
Audio over IP (AoIP) is the distribution of digital audio across an IP network such as the Internet. It is used increasingly to provide high-quality audio feeds over long distances. The application is also known as audio contribution over IP (ACI ...
for
broadcasting
Broadcasting is the distribution (business), distribution of sound, audio or video content to a dispersed audience via any electronic medium (communication), mass communications medium, but typically one using the electromagnetic spectrum (radio ...
applications where it provides an interoperable means for audio interfaces from different manufacturers to make connections with one another.
Implementations
The U.S.
National Institute of Standards and Technology
The National Institute of Standards and Technology (NIST) is an agency of the United States Department of Commerce whose mission is to promote American innovation and industrial competitiveness. NIST's activities are organized into physical sci ...
(NIST), Advanced Networking Technologies Division provides a public-domain
Java
Java (; id, Jawa, ; jv, ꦗꦮ; su, ) is one of the Greater Sunda Islands in Indonesia. It is bordered by the Indian Ocean to the south and the Java Sea to the north. With a population of 151.6 million people, Java is the world's List ...
implementation that serves as a
reference implementation
In the software development process, a reference implementation (or, less frequently, sample implementation or model implementation) is a program that implements all requirements from a corresponding specification. The reference implementation o ...
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 in full and a number of extension RFCs including (event notification) and (reliable provisional responses).
Numerous other commercial and open-source SIP implementations exist. See
List of SIP software
This list of SIP software documents notable software applications which use Session Initiation Protocol (SIP) as a voice over IP (VoIP) protocol.
Servers
Free and open-source license
A SIP server, also known as a SIP proxy, manage ...
.
SIP-ISUP interworking
SIP-I, Session Initiation Protocol with encapsulated
ISUP, 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
The ITU Telecommunication Standardization Sector (ITU-T) is one of the three sectors (divisions or units) of the International Telecommunication Union (ITU). It is responsible for coordinating standards for telecommunications and Information Commu ...
, whereas SIP-T was defined by the
IETF
The Internet Engineering Task Force (IETF) is a standards organization for the Internet and is responsible for the technical standards that make up the Internet protocol suite (TCP/IP). It has no formal membership roster or requirements and a ...
.
Encryption
Concerns about the security of calls via the public Internet have been addressed by encryption of the SIP protocol for
secure transmission
In computer science, secure transmission refers to the transfer of data such as confidential or proprietary information over a secure channel. Many secure transmission methods require a type of encryption. The most common email encryption is called ...
. The URI scheme SIPS is used to mandate that SIP communication be secured with
Transport Layer Security
Transport Layer Security (TLS) is a cryptographic protocol designed to provide communications security over a computer network. The protocol is widely used in applications such as email, instant messaging, and voice over IP, but its use in securi ...
(TLS). SIPS URIs take the form sips:user@example.com.
End-to-end encryption
End-to-end encryption (E2EE) is a system of communication where only the communicating users can read the messages. In principle, it prevents potential eavesdroppers – including telecom providers, Internet providers, malicious actors, and even ...
of SIP is only possible if there is a direct connection between communication endpoints. While a direct connection can be made via
Peer-to-peer SIP or via a
VPN
A virtual private network (VPN) extends a private network across a public network and enables users to send and receive data across shared or public networks as if their computing devices were directly connected to the private network. The be ...
between the endpoints, most SIP communication involves multiple hops, with the first hop being from a user agent to the user agent's
ITSP An Internet telephony service provider (ITSP) offers digital telecommunications services based on Voice over Internet Protocol (VoIP) that are provisioned via the Internet.
ITSPs provide services to end-users directly or as whole-sale suppliers to ...
. 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
Hypertext Transfer Protocol Secure (HTTPS) is an extension of the Hypertext Transfer Protocol (HTTP). It is used for secure communication over a computer network, and is widely used on the Internet. In HTTPS, the communication protocol is enc ...
protocol provides end-to-end security as it 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
SDES (Session Description Protocol Security Descriptions) for Media Streams is a way to negotiate the key for Secure Real-time Transport Protocol. It has been proposed for standardization to the IETF in July 2006 (see RFC khh.)
How it works
The ...
(), or with
ZRTP
ZRTP (composed of Z and Real-time Transport Protocol) is a cryptographic key-agreement protocol to negotiate the keys for encryption between two end points in a Voice over IP (VoIP) phone telephony call based on the Real-time Transport Protocol. I ...
(). When SDES is used, the keys will be transmitted via insecure SIP unless SIPS is used. One may also add a
MIKEY
Mikey is a masculine given name, often a diminutive form (hypocorism) of Michael. It may also refer to:
People
* Mikey Ambrose (born 1993), American Major League Soccer player
* Mikey Arroyo (born 1969), Filipino actor and politician, son of Phili ...
() exchange to SIP to determine session keys for use with SRTP.
See also
*
Computer telephony integration
Computer telephony integration, also called computer–telephone integration or CTI, is a common name for any technology that allows interactions on a telephone and a computer to be coordinated. The term is predominantly used to describe desktop- ...
(CTI)
*
Computer-supported telecommunications applications Computer-supported telecommunications applications (CSTA) is an abstraction layer for telecommunications applications. It is independent of underlying protocols. It has a telephone device model that enables CTI applications to work with a wide ran ...
(CSTA)
*
H.323
H.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. The H.323 standard addresses call signaling and control, m ...
protocols
H.225.0
H.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. The H.323 standard addresses call signaling and control, ...
and
H.245
*
IP Multimedia Subsystem
The IP Multimedia Subsystem or IP Multimedia Core Network Subsystem (IMS) is a standardised architectural framework for delivering IP multimedia services. Historically, mobile phones have provided voice call services over a circuit-switched-styl ...
(IMS)
*
Media Gateway Control Protocol
The Media Gateway Control Protocol (MGCP) is a signaling and call control communication protocol used in voice over IP (VoIP) telecommunication systems. It implements the media gateway control protocol architecture for controlling media gatewa ...
(MGCP)
*
Mobile VoIP
Mobile VoIP or simply mVoIP is an extension of mobility to a voice over IP network. Two types of communication are generally supported: cordless telephones using DECT or PCS protocols for short range or campus communications where all base stati ...
*
MSCML The Media Server Control Markup Language (MSCML) is a protocol used in conjunction with the Session Initiation Protocol (SIP) to enable the delivery of advanced multimedia conferencing services over IP networks. The MSCML specification has been pub ...
(Media Server Control Markup Language)
*
Network convergence
Network convergence refers to the provision of telephone, video and data communication services within a single network. In other words, one company provides services for all forms of communication. Network convergence is primarily driven by devel ...
*
Rendezvous protocol
A rendezvous protocol is a computer network protocol that enables resources or P2P network peers to find each other. A rendezvous protocol uses a handshaking model, unlike an eager protocol which directly copies the data. In a rendezvous protocol ...
*
RTP payload formats The Real-time Transport Protocol (RTP) specifies a general-purpose data format and network protocol for transmitting digital media streams on Internet Protocol (IP) networks. The details of media encoding, such as signal sampling rate, frame size an ...
*
SIGTRAN SIGTRAN is the name, derived from ''signaling transport'', of the former Internet Task Force (I) working group that produced specifications for a family of protocols that provide reliable datagram service and user layer adaptations for Signaling ...
(Signaling Transport)
*
SIP extensions for the IP Multimedia Subsystem
*
SIP provider
*
Skinny Client Control Protocol
The Skinny Client Control Protocol (SCCP) is a proprietary network terminal control protocol originally developed by Selsius Systems, which was acquired by Cisco Systems in 1998.
SCCP is a lightweight IP-based protocol for session signaling with ...
(SCCP)
*
T.38
T.38 is an ITU recommendation for allowing transmission of fax over IP networks (FoIP) in real time.
History
The T.38 fax relay standard was devised in 1998 as a way to permit faxes to be transported across IP networks between existing Group 3 ...
*
XIMSS
CommuniGate Pro (CGatePro) is a highly scalable carrier grade unified communications server, as well as a development platform. The system is Internet Protocol based. The server is a single process, multi-threaded application written entirely in ...
(XML Interface to Messaging, Scheduling, and Signaling)
Notes
References
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External links
IANA: SIP ParametersIANA: SIP Event Types Namespace
{{Authority control
VoIP protocols
Videotelephony
Application layer protocols