Session Initiation Protocolwikipedia
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.
SIPsession initiation protocolSession Initiation Protocol (SIP)SIP TrunkingSIP (Session Initiation Protocol)5060SIP addressSIP-basedproxySIPS

Voice over IP

VoIPvoice over IPvoice over Internet Protocol
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 SIP connection is a marketing term for voice over Internet Protocol (VoIP) services offered by many Internet telephony service providers (ITSPs).
These include RTCP Extended Report (RFC 3611), SIP RTCP Summary Reports, H.460.9 Annex B (for H.323), H.248.30 and MGCP extensions.

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

IMSIP Multimedia Subsystem (IMS)HSS
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).

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).

Real-time Transport Protocol

RTPreal-time protocolReal-Time Transport Protocol
For the transmission of media streams (voice, video) SIP 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.

SRV record

SRVSRV resource recordsserver 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.

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

session border controllerSBCSession 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.
While no one signalling protocol is mandated by the WebRTC specifications, SIP over Websockets (RFC 7118) is often used partially due to the applicability of SIP to most of the envisaged communication scenarios as well as the availability of open source software such as JsSIP.

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.

Instant messaging

instant messagingIMinstant messenger
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). 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. It has been extended for video conferencing, streaming media distribution, instant messaging, presence information, file transfer, Internet fax and online games.
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.

User agent

user agentuser-agentuser agent string
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.

SIMPLE (instant messaging protocol)

SIMPLESIP for Instant Messaging and Presence Leveraging ExtensionsSession Initiation Protocol for Instant Messaging and Presence Leveraging Extensions
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.

SIP extensions for the IP Multimedia Subsystem

SIP
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.

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.

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.

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,

Session Description Protocol

SDP
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.

Internet telephony service provider

VoIP servicesVoIP telephone companiesITSPs
A 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.

Mobile VoIP

mobile VoIPVoIPHotspot@Home
One implementation turns the mobile device into a standard SIP client, which then uses a data network to send and receive SIP messaging, and to send and receive RTP for the voice path.

MIKEY

Mikey
The key exchange for SRTP is performed with SDES, or with ZRTP . One may also add a MIKEY exchange to SIP to determine session keys for use with SRTP.
Key management is performed by including MIKEY messages within the SDP content of SIP signalling messages.

MSCML

Media Server Control Markup Language
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.

Signaling protocol

signaling protocolsignalling 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.

Softphone

softphonesoft 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).

ZRTP

ZRTP - an alternative to MIKEY as cryptographic key-agreement protocol for SRTPencrypted
The key exchange for SRTP is performed with SDES, or with ZRTP . One may also add a MIKEY exchange to SIP to determine session keys for use with SRTP.
ZRTP ("Z" is a reference to its inventor, Zimmermann; "RTP" stands for Real-time Transport Protocol) is described in the Internet Draft as a ''"key agreement protocol which performs Diffie–Hellman key exchange during call setup in-band in the Real-time Transport Protocol (RTP) media stream which has been established using some other signaling protocol such as Session Initiation Protocol (SIP). This generates a shared secret which is then used to generate keys and salt for a Secure RTP (SRTP) session."'' One of ZRTP's features is that it does not rely on SIP signaling for the key management, or on any servers at all.