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 Voice over Internet Protocol (VoIP), also known as IP telephony, is a set of technologies used primarily for voice communication sessions over Internet Protocol (IP) networks, such as the Internet. VoIP enables voice calls to be transmitted as ...

, in private IP telephone systems, as well as mobile phone calling over LTE ( VoLTE). 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 In computing, text-based user interfaces (TUI) (alternately terminal user interfaces, to reflect a dependence upon the properties of computer terminals and not just text), is a retronym describing a type of user interface (UI) common as an ear ...

, incorporating many elements of the

Hypertext Transfer Protocol HTTP (Hypertext Transfer Protocol) 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, wher ...

(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 typ ...

(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 phones, tablet computers, smartwatches, desktops/laptops, or ...

, 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 c ...

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 Network packet, packets) to other hosts on an Internet Protoco ...

(UDP), the

Transmission Control Protocol The Transmission Control Protocol (TCP) is one of the main communications protocol, protocols of the Internet protocol suite. It originated in the initial network implementation in which it complemented the Internet Protocol (IP). Therefore, th ...

(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 ...

(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, such as the Internet. The protocol is widely used in applications such as email, instant messaging, and voice over ...

(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 applic ...

(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 multica ...

(SRTP).

History

SIP was originally designed by

Mark Handley Mark Handley is a playwright and screenwriter. Personal life In 1977, he and his wife moved to the Pacific Northwest where they lived in isolation in a log cabin that they built themselves. Career He is best known for his play ''Idioglos ...

, Henning Schulzrinne, Eve Schooler and Jonathan Rosenberg in 1996 to facilitate establishing

multicast In computer networking, multicast is a type of 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 differs from ph ...

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 ...

. The protocol was standardized as in 1999. In November 2000, SIP was accepted as a

3GPP The 3rd Generation Partnership Project (3GPP) is an umbrella term for a number of standards organizations which develop protocols for mobile telecommunications. Its best known work is the development and maintenance of: * GSM and related 2G and ...

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-sty ...

(IMS) architecture for IP-based streaming multimedia services in

cellular network A cellular network or mobile network is a telecommunications network where the link to and from end nodes is wireless network, wireless and the network is distributed over land areas called ''cells'', each served by at least one fixed-locatio ...

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) is the aggregate of the world's telephone networks that are operated by national, regional, or local telephony operators. It provides infrastructure and services for public telephony. The PSTN consists o ...

(PSTN) with a vision of supporting new multimedia applications. It has been extended for

video conferencing Videotelephony (also known as videoconferencing or video calling) is the use of audio signal, audio and video for simultaneous two-way communication. Today, videotelephony is widespread. There are many terms to refer to videotelephony. ''Vide ...

streaming media Streaming media refers to multimedia delivered through a Computer network, network for playback using a Media player (disambiguation), media player. Media is transferred in a ''stream'' of Network packet, packets from a Server (computing), ...

distribution,

instant messaging Instant messaging (IM) technology is a type of synchronous computer-mediated communication involving the immediate ( real-time) transmission of messages between two or more parties over the Internet or another computer network. Originally involv ...

presence information In computer network, computer and telecommunications networks, presence information is a status indicator that conveys ability and willingness of a potential communication partner—for example a user (computing), user—to communication, communicat ...

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 protoc ...

Internet fax The Internet (or internet) is the Global network, global system of interconnected computer networks that uses the Internet protocol suite (TCP/IP) to communicate between networks and devices. It is a internetworking, network of networks ...

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 PCs, consoles and mobile devices, a ...

s. SIP is distinguished by its proponents for having roots in the Internet community rather than in the

telecommunications industry The telecommunications industry within the sector of information and communication technology comprises all telecommunication/ telephone companies and Internet service providers, and plays a crucial role in the evolution of mobile communications ...

. SIP has been standardized primarily by the

Internet Engineering Task Force The Internet Engineering Task Force (IETF) is a standards organization for the Internet standard, Internet and is responsible for the technical standards that make up the Internet protocol suite (TCP/IP). It has no formal membership roster ...

(IETF), while other protocols, such as H.323, have traditionally been associated with the

International Telecommunication Union The International Telecommunication Union (ITU)In the other common languages of the ITU: * * is a list of specialized agencies of the United Nations, specialized agency of the United Nations responsible for many matters related to information ...

(ITU).

Protocol operation

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 (

) sessions. It also allows modification of existing calls. The modification can involve changing addresses or

ports Ports collections (or ports trees, or just ports) are the sets of makefiles and Patch (Unix), patches provided by the BSD-based operating systems, FreeBSD, NetBSD, and OpenBSD, as a simple method of installing software or creating binary packages. T ...

, 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

(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

(RTP) or

(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), formerly Universal Resource Identifier, is a unique sequence of characters that identifies an abstract or physical resource, such as resources on a webpage, mail address, phone number, books, real-world obje ...

(URI). The syntax of the URI follows the general standard syntax also used in

Web service A web service (WS) is either: * a service offered by an electronic device to another electronic device, communicating with each other via the Internet, or * a server running on a computer device, listening for requests at a particular port over a n ...

s and e-mail. The URI scheme used for SIP is ''sip'' and a typical SIP URI has the form ''sip:username@domainname'' or ''sip:username@hostport'', where ''domainname'' requires DNS SRV records to locate the servers for SIP domain while ''hostport'' can be an

IP address An Internet Protocol address (IP address) is a numerical label such as that is assigned to a device connected to a computer network that uses the Internet Protocol for communication. IP addresses serve two main functions: network interface i ...

or a

fully qualified domain name A fully qualified domain name (FQDN), sometimes also called 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 the top-level ...

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 the header fields, encoding rules and status codes of HTTP, providing a readable text-based format. SIP can be carried by several

protocols including

(TCP),

(UDP), and

(SCTP). SIP clients typically use TCP or UDP on

port number In computer networking, a port is a communication endpoint. At the software level within an operating system, a port is a logical construct that identifies a specific process or a type of network service. A port is uniquely identified by a numbe ...

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

(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 the 1970s that is used to setup and teardown telephone calls on most parts of the global public switched telephone network (PSTN). The protocol also performs ...

(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 mobile phone with advanced computing capabilities. It typically has a touchscreen interface, allowing users to access a wide range of applications and services, such as web browsing, email, and social media, as well as multi ...

s. In SIP, as in HTTP, the

user agent On the Web, a user agent is a software agent responsible for retrieving and facilitating end-user interaction with Web content. This includes all web browsers, such as Google Chrome and Safari A safari (; originally ) is an overland jour ...

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

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 middleboxes 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 Gateway (telecommunications), gateways that implement network address translation (NAT). NAT traversal te ...

. 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, 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

and

. Message Session Relay Protocol (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. TT ...

test specification language, developed by a task force at

ETSI The European Telecommunications Standards Institute (ETSI) is an independent, not-for-profit, standardization organization operating in the field of Information and communications technology, information and communications. ETSI supports the de ...

(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, 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 signifi ...

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.Ku ...

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 known as IP telephony, is a set of technologies used primarily for voice communication sessions over Internet Protocol (IP) networks, such as the Internet. VoIP enables Voice call, voice calls to be tran ...

(VoIP) services offered by many Internet telephony service providers (ITSPs). The service provides routing of telephone calls from a client's

private branch exchange A business telephone system is a telephone system typically used in business environments, encompassing the range of technology from the key telephone system (KTS) to the private branch exchange (PBX). A business telephone system differs from ...

(PBX) telephone system to the PSTN. Such services may simplify corporate information system infrastructure by sharing

Internet access Internet access is a facility or service that provides connectivity for a computer, a computer network, or other network device to the Internet, and for individuals or organizations to access or use applications such as email and the World Wide ...

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 Plain old telephone service, ...

(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 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 data distribution, distribution of sound, audio audiovisual content to dispersed audiences via a electronic medium (communication), mass communications medium, typically one using the electromagnetic spectrum (radio waves), ...

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 Outline of p ...

(NIST), Advanced Networking Technologies Division provides a public-domain

Java Java is one of the Greater Sunda Islands in Indonesia. It is bordered by the Indian Ocean to the south and the Java Sea (a part of Pacific Ocean) to the north. With a population of 156.9 million people (including Madura) in mid 2024, proje ...

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 ...

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.

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 International Telecommunication Union Telecommunication Standardization Sector (ITU-T) is one of the three Sectors (branches) of the International Telecommunication Union (ITU). It is responsible for coordinating Standardization, standards fo ...

, whereas SIP-T was defined by the

IETF The Internet Engineering Task Force (IETF) is a standards organization for the Internet standard, Internet and is responsible for the technical standards that make up the Internet protocol suite (TCP/IP). It has no formal membership roster ...

Encryption

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

(TLS). SIPS URIs take the form sips:[email protected].

End-to-end encryption End-to-end encryption (E2EE) is a method of implementing a secure communication system where only communicating users can participate. No one else, including the system provider, telecom providers, Internet providers or malicious actors, can ...

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 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 Hypertext Transfer Protocol Secure (HTTPS) is an extension of the Hypertext Transfer Protocol (HTTP). It uses encryption for secure communication over a computer network, and is widely used on the Internet. In HTTPS, the communication protoc ...

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 (), 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 ...

(). When SDES is used, the keys will be transmitted via insecure SIP unless SIPS is used. One may also add a MIKEY () exchange to SIP to determine session keys for use with SRTP.

Notes

References

* *

External links

IANA: SIP Parameters

IANA: SIP Event Types Namespace
{{Authority control VoIP protocols Videotelephony Application layer protocols