Session Initiation Protocol

SIPSession Initiation Protocol (SIP)SIP TrunkingSIP (Session Initiation Protocol)5060proxySIP addressSIP protocolSIP TrunksSIP-based
The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video and messaging applications.wikipedia
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Session (computer science)

sessionsessionssession management
SIP is used for signaling and controlling multimedia communication sessions in applications of Internet telephony for voice and video calls, in private IP telephone systems, in instant messaging over Internet Protocol (IP) networks as well as mobile phone calling over LTE (VoLTE).
A Session Initiation Protocol (SIP) based Internet phone call

Jonathan Rosenberg (SIP author)

Jonathan Rosenberg
SIP was originally designed by Mark Handley, Henning Schulzrinne, Eve Schooler and Jonathan Rosenberg in 1996.
Network World has referred to him as "a pioneer [in] the development of the SIP protocol", and he was included in the 2002 TR35 list of the world's top under-35 innovators, as published by MIT Technology Review.

IP Multimedia Subsystem

IMSHSSIP Multimedia Subsystem (IMS)
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.
IMS uses IETF protocols wherever possible, e.g., the Session Initiation Protocol (SIP).

Real-time Transport Protocol

RTPReal-Time ProtocolRTP / RTCP
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).
RTP is one of the technical foundations of Voice over IP and in this context is often used in conjunction with a signaling protocol such as the Session Initiation Protocol (SIP) which establishes connections across the network.

Instant messaging

instant messengerIMinstant message
SIP is used for signaling and controlling multimedia communication sessions in applications of Internet telephony for voice and video calls, in private IP telephone systems, in instant messaging over Internet Protocol (IP) networks as well as mobile phone calling over LTE (VoLTE). It has been extended for video conferencing, streaming media distribution, instant messaging, presence information, file transfer, Internet fax and online games. The Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions (SIMPLE) is the SIP-based suite of standards for instant messaging and presence information.
There have been several attempts to create a unified standard for instant messaging: IETF's Session Initiation Protocol (SIP) and SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE), Application Exchange (APEX), Instant Messaging and Presence Protocol (IMPP), the open XML-based Extensible Messaging and Presence Protocol (XMPP), and Open Mobile Alliance's Instant Messaging and Presence Service developed specifically for mobile devices.

SRV record

SRVnetwork service typesserver record
The URI scheme used for SIP is sip and a typical SIP URI has the form or, where domainname requires DNS SRV records to locate the servers for SIP domain while hostport can be an IP address or a fully qualified domain name of the host and port.
It is defined in RFC 2782, and its type code is 33. Some Internet protocols such as the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP) often require SRV support by network elements.

Signaling protocol

signalling protocolsignaling
The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video and messaging applications.
Session Initiation Protocol

VoIP phone

IP PhoneIP PhonesEthernet-based phone
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.
Digital IP-based telephone service uses control protocols such as the Session Initiation Protocol (SIP), Skinny Client Control Protocol (SCCP) or various other proprietary protocols.

Session border controller

SBCSession Border Controllers
Session border controllers serve as middle boxes between user-agent (UA) and SIP servers for various types of functions, including network topology hiding and assistance in NAT traversal.
A Session Border Controller (SBC) is a network element deployed to protect SIP based Voice over Internet Protocol (VoIP) networks.

Session Description Protocol

SDP
Most commonly, media type and parameter negotiation and media setup is performed with the Session Description Protocol (SDP), which is carried as payload in SIP messages.
SDP started off as a component of the Session Announcement Protocol (SAP), but found other uses in conjunction with Real-time Transport Protocol (RTP), Real-time Streaming Protocol (RTSP), Session Initiation Protocol (SIP) and even as a standalone format for describing multicast sessions.

Henning Schulzrinne

SIP was originally designed by Mark Handley, Henning Schulzrinne, Eve Schooler and Jonathan Rosenberg in 1996.
He co-designed the Session Initiation Protocol along with Mark Handley, the Real Time Streaming Protocol, the Real-time Transport Protocol, the General Internet Signaling Transport Protocol,

Videotelephony

videoconferencingvideo conferencingvideo chat
It has been extended for video conferencing, streaming media distribution, instant messaging, presence information, file transfer, Internet fax and online games.
Videoconferencing in the late 20th century was limited to the H.323 protocol (notably Cisco's SCCP implementation was an exception), but newer videophones often use SIP, which is often easier to set up in home networking environments.

User agent

user-agentuser agent stringuser agents
In SIP, as in HTTP, the user agent may identify itself using a message header field (User-Agent), containing a text description of the software, hardware, or the product name.
The Session Initiation Protocol (SIP) protocol (based on HTTP) followed this usage.

Message Session Relay Protocol

MSRPMSRP protocol
MSRP (Message Session Relay Protocol) allows instant message sessions and file transfer.
An application instantiates the session with the Session Description Protocol (SDP) over Session Initiation Protocol (SIP) or other rendezvous methods.

Transport Layer Security

SSLTLSSSL/TLS
For secure transmissions of SIP messages over insecure network links, the protocol may be encrypted with Transport Layer Security (TLS).
TLS is also a standard method to protect Session Initiation Protocol (SIP) application signaling.

SIMPLE (instant messaging protocol)

SIMPLESIP for Instant Messaging and Presence Leveraging ExtensionsPresence
The Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions (SIMPLE) is the SIP-based suite of standards for instant messaging and presence information.
SIMPLE, the Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions, is an instant messaging (IM) and presence protocol suite based on Session Initiation Protocol (SIP) managed by the Internet Engineering Task Force.

List of SIP software

client softwareList of Session Initiation Protocol (SIP) software
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.

Presence information

presenceonline presencepresence events
It has been extended for video conferencing, streaming media distribution, instant messaging, presence information, file transfer, Internet fax and online games. The Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions (SIMPLE) is the SIP-based suite of standards for instant messaging and presence information.
In 2001, the SIMPLE working group was formed within IETF to develop a suite of CPP-compliant standards for presence and instant messaging applications over the Session Initiation Protocol (SIP).

Softphone

soft phonePC-to-PC calling
SIP phones may be implemented as a hardware device or as a softphone.
Many service providers use the Session Initiation Protocol (SIP) standardized by the Internet Engineering Task Force (IETF).

Business telephone system

PBXprivate branch exchangekey telephone system
The service provides routing of telephone calls from a client's private branch exchange (PBX) telephone system to the public switched telephone network (PSTN).
The modern key system now supports SIP, ISDN, analog handsets (in addition to its own proprietary handsets - usually digital) as well as a raft of features more traditionally found on larger PBX systems.

Stream Control Transmission Protocol

SCTPStream Control Transmission Protocol (SCTP)
SIP is designed to be independent of the underlying transport layer protocol, and can be used with the User Datagram Protocol (UDP), the Transmission Control Protocol (TCP), and the Stream Control Transmission Protocol (SCTP).
Session Initiation Protocol (SIP) – which may initiate multiple streams over SCTP, TCP or UDP

Internet telephony service provider

ITSPsNet telephonyVoIP services
SIP connection is a marketing term for voice over Internet Protocol (VoIP) services offered by many Internet telephony service providers (ITSPs).
ITSPs use a variety of signaling and multimedia protocols, including the Session Initiation Protocol (SIP), the Media Gateway Control Protocol (MGCP), Megaco, and the H.323 protocol.

Mark Handley (computer scientist)

Mark HandleyHandley
SIP was originally designed by Mark Handley, Henning Schulzrinne, Eve Schooler and Jonathan Rosenberg in 1996.
He is the (co-)author of 34 RFCs, including the Session Initiation Protocol, Multipath TCP and a series of other network protocols.

SIP extensions for the IP Multimedia Subsystem

SIP
Extensions to the Session Initiation Protocol for the IP Multimedia Subsystem
The Session Initiation Protocol (SIP) is the signaling protocol selected by the 3rd Generation Partnership Project (3GPP) to create and control multimedia sessions with two or more participants in the IP Multimedia Subsystem (IMS), and therefore is a key element in the IMS framework.

TTCN-3

The TTCN-3 test specification language, developed by a task force at ETSI (STF 196), is used for specifying conformance tests for SIP implementations.
TTCN-3 has been used to define conformance test suites to SIP, WiMAX, and DSRC standard protocols.