Session Initiation Protocol
Internet telephony | |
Introduction | March 1999 |
---|---|
OSI layer | Application layer (Layer 7) |
Port(s) | 5060, 5061 |
RFC(s) | 2543, 3261 |
Internet protocol suite |
---|
Application layer |
Transport layer |
Internet layer |
Link layer |
The Session Initiation Protocol (SIP) is a
The protocol defines the specific format of messages exchanged and the sequence of communications for cooperation of the participants. SIP is a
SIP works in conjunction with several other protocols that specify and carry the session media. Most commonly, media type and parameter negotiation and media setup are performed with the Session Description Protocol (SDP), which is carried as payload in SIP messages. SIP is designed to be independent of the underlying transport layer protocol and can be used with the User Datagram Protocol (UDP), the Transmission Control Protocol (TCP), and the Stream Control Transmission Protocol (SCTP). For secure transmissions of SIP messages over insecure network links, the protocol may be encrypted with Transport Layer Security (TLS). For the transmission of media streams (voice, video) the SDP payload carried in SIP messages typically employs the Real-time Transport Protocol (RTP) or the Secure Real-time Transport Protocol (SRTP).
History
SIP was originally designed by
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
SIP is distinguished by its proponents for having roots in the Internet community rather than in the telecommunications industry. SIP has been standardized primarily by the Internet Engineering Task Force (IETF), while other protocols, such as H.323, have traditionally been associated with the International Telecommunication Union (ITU).
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 (
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 (SDP) data unit, which specifies the media format, codec and media communication protocol. Voice and video media streams are typically carried between the terminals using the Real-time Transport Protocol (RTP) or Secure Real-time Transport Protocol (SRTP).[3][8]
Every resource of a SIP network, such as user agents, call routers, and voicemail boxes, are identified by a Uniform Resource Identifier (URI). The syntax of the URI follows the general standard syntax also used in Web services and e-mail.[9] 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 or a fully qualified domain name of the host and port. If secure transmission is required, the scheme sips is used.[10][11]
SIP employs design elements similar to the HTTP request and response transaction model.[12] 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
SIP-based telephony networks often implement call processing features of
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.[15]
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.[6]
A SIP phone is an
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 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,[18] 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 controllers (SBCs) serve as middleboxes between user agents and SIP servers for various types of functions, including network topology hiding and assistance in NAT traversal. SBCs are an independently engineered solution and are not mentioned in 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.[19] 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
Request name | Description | Notes | RFC references |
---|---|---|---|
REGISTER | Register the URI listed in the To-header field with a location server and associates it with the network address given in a Contact header field. | The command implements a location service. | RFC 3261
|
INVITE | Initiate a dialog for establishing a call. The request is sent by a user agent client to a user agent server. | When sent during an established dialog (reinvite) it modifies the sessions, for example placing a call on hold. | RFC 3261
|
ACK | Confirm that an entity has received a final response to an INVITE request. | RFC 3261
| |
BYE | Signal termination of a dialog and end a call. | This message may be sent by either endpoint of a dialog. | RFC 3261
|
CANCEL | Cancel any pending request. | Usually means terminating a call while it is still ringing, before answer. | RFC 3261
|
UPDATE | Modify the state of a session without changing the state of the dialog. | RFC 3311
| |
REFER | Ask recipient to issue a request for the purpose of call transfer. | RFC 3515
| |
PRACK | Provisional acknowledgement. | PRACK is sent in response to provisional response (1xx). | RFC 3262
|
SUBSCRIBE | Initiates a subscription for notification of events from a notifier. | RFC 6665
| |
NOTIFY | Inform a subscriber of notifications of a new event. | RFC 6665
| |
PUBLISH | Publish an event to a notification server. | RFC 3903
| |
MESSAGE | Deliver a text message. | Used in instant messaging applications. | RFC 3428
|
INFO | Send mid-session information that does not modify the session state. | This method is often used for DTMF relay. | RFC 6086
|
OPTIONS | Query the capabilities of an endpoint. | It is often used for NAT keepalive purposes. | RFC 3261
|
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:[20]
- 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 and presence information. 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.[22] The TTCN-3 test specification language, developed by a task force at ETSI (STF 196), is used for specifying conformance tests for SIP implementations.[23]
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.
Applications
SIP connection is a marketing term for
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.[25][26]
SIP-enabled video surveillance cameras can initiate calls to alert the operator of events, such as the motion of objects in a protected area.
SIP is used in audio over IP for broadcasting applications where it provides an interoperable means for audio interfaces from different manufacturers to make connections with one another.[27]
Implementations
The U.S.
Numerous other commercial and open-source SIP implementations exist. See List of SIP software.
SIP-ISUP interworking
SIP-I, Session Initiation Protocol with encapsulated
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 Transport Layer Security (TLS). SIPS URIs take the form sips:[email protected].
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
See also
- Computer telephony integration (CTI)
- Computer-supported telecommunications applications (CSTA)
- H.225.0 and H.245
- IP Multimedia Subsystem (IMS)
- Media Gateway Control Protocol (MGCP)
- Mobile VoIP
- MSCML (Media Server Control Markup Language)
- Network convergence
- Rendezvous protocol
- RTP payload formats
- SIGTRAN (Signaling Transport)
- SIP extensions for the IP Multimedia Subsystem
- SIP provider
- Skinny Client Control Protocol (SCCP)
- T.38
- XIMSS(XML Interface to Messaging, Scheduling, and Signaling)
Notes
- ^ ISUP detail is important as there are many country-specific variants of ISUP that have been implemented over the last 30 years, and it is not always possible to express all of the same detail using a native SIP message.
References
- ^ Network World. May 11, 2004.
- ^ "4G | ShareTechnote". www.sharetechnote.com. Retrieved 2023-03-09.
- ^ ISBN 9781580531689.
- ^ "SIP core working group charter". Internet Engineering Task Force. 2010-12-07. Retrieved 2011-01-11.
- ^ "Search Internet-Drafts and RFCs". Internet Engineering Task Force.
- ^ .
- ^ Rouse, Margaret. "Session Initiation Protocol (SIP)". TechTarget.
- ISBN 9781894887038.
- .
- ^ Miikka Poikselkä et al. 2004.
- ^ Brian Reid & Steve Goodman 2015.
- ^ "SIP: Session Initiation Protocol". IETF.
- .
- S2CID 27215205.
- S2CID 3873601.
- ISBN 978-1-58488-465-1.
- ISBN 978-1-59749-060-3.
- ^ "User-Agents We Have Known". VoIP User. Archived from the original on 2011-07-16.
- ^ Stallings, p.214
- ^ Stallings, pp.216-217
- ^ Wright, James. "SIP - An Introduction" (PDF). Konnetic. Retrieved 2011-01-11.
- ^ "SIPit Wiki". Retrieved 2017-10-07.
- ^ Experiences of Using TTCN-3 for Testing SIP and also OSP (PDF), archived from the original (PDF) on March 30, 2014
- ^ "Performance and Stress Testing of SIP Servers, Clients and IP Networks". StarTrinity. 2016-08-13.
- ^ "AT&T Discusses Its SIP Peering Architecture". sip-trunking.tmcnet.com. Retrieved 2017-03-20.
- ^ "From IIT VoIP Conference & Expo: AT&T SIP transport PowerPoint slides". HD Voice News. 2010-10-19. Retrieved 2017-03-20.
- ^ Jonsson, Lars; Mathias Coinchon (2008). "Streaming audio contributions over IP" (PDF). EBU Technical Review. Retrieved 2010-12-27.
- ^ "JAIN SIP project". Retrieved 2011-07-26.
- .
- ^ "Why SIP-I? A Switching Core Protocol Recommendation" (PDF). Archived from the original (PDF) on 2012-03-17.
- Brian Reid; Steve Goodman (22 January 2015), Exam Ref 70-342 Advanced Solutions of Microsoft Exchange Server 2013 (MCSE), Microsoft Press, p. 24, ISBN 9780735697904)
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