It has been suggested that SIP connection be merged into this article or section. (Discuss) Proposed since April 2010.

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The Session Initiation Protocol (SIP) is an IETF-defined signaling protocol widely used^{[citation needed]} for controlling communication sessions such as voice and video calls over Internet Protocol (IP). The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions. Sessions may consist of one or several media streams.

Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer and online games^{[citation needed]}.

The SIP protocol is an Application Layer protocol designed to be independent of the underlying Transport Layer; it can run on Transmission Control Protocol (TCP), User Datagram Protocol (UDP), or Stream Control Transmission Protocol (SCTP).^[1] It is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP).^[2]

History[link]

SIP was originally designed by Henning Schulzrinne and Mark Handley in 1996. 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 systems. The latest version of the specification is RFC 3261 from the IETF Network Working Group published in June 2002.^[3]

The National Institute of Standards and Technology (NIST), Advanced Networking Technologies Division provides a public domain implementation of the JAVA Standard for SIP^[4] which serves as a reference implementation for the standard. The stack can work in proxy server or user agent scenarios and has been used in numerous commercial and research projects. It supports RFC 3261 in full and a number of extension RFCs including RFC 3265 (Subscribe / Notify) and RFC 3262 (Provisional Reliable Responses) etc.

Protocol design[link]

SIP employs design elements similar to the HTTP request/response transaction model.^[5] 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.

Each resource of a SIP network, such as a user agent or a voicemail box, is identified by a uniform resource identifier (URI), based on the general standard syntax^[6] also used in Web services and e-mail. A typical SIP URI is of the form: sip:username:password@host:port. The URI scheme used for SIP is sip:.

If secure transmission is required, the scheme sips: is used and mandates that each hop over which the request is forwarded up to the target domain must be secured with Transport Layer Security (TLS). The last hop from the proxy of the target domain to the user agent has to be secured according to local policies. TLS protects against attackers which try to listen on the signaling link. It does not provide real end-to-end security, since encryption is only hop-by-hop and every single intermediate proxy has to be trusted.

SIP works in concert with several other protocols and is only involved in the signaling portion of a communication session. SIP clients typically use TCP or UDP on port numbers 5060 and/or 5061 to connect to SIP servers and other SIP endpoints. Port 5060 is commonly used for non-encrypted signaling traffic whereas port 5061 is typically used for traffic encrypted with Transport Layer Security (TLS). SIP is primarily used in setting up and tearing down voice or video calls. It also allows modification of existing calls. The modification can involve changing addresses or ports, 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. A suite of SIP-related Internet Engineering Task Force (IETF) define behavior for such applications. The voice and video stream communications in SIP applications are carried over another application protocol, the Real-time Transport Protocol (RTP). Parameters (port numbers, protocols, codecs) for these media streams are defined and negotiated using the Session Description Protocol (SDP) which is transported in the SIP packet body.

A motivating goal for SIP was to provide a signaling and call setup protocol for IP-based communications that can support a superset of the call processing functions and features present in the public switched telephone network (PSTN). SIP by itself does not define these features; rather, its focus is call-setup and signaling. The features that permit familiar telephone-like operations: dialing a number, causing a phone to ring, hearing ringback tones or a busy signal - are performed by proxy servers and user agents. Implementation and terminology are different in the SIP world but to the end-user, the behavior is similar.

SIP-enabled telephony networks can also implement many of the more advanced call processing features present in Signaling System 7 (SS7), though 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 peer-to-peer protocol, thus it requires only a simple (and thus scalable) core network with intelligence distributed to the network edge, embedded in endpoints (terminating devices built in either hardware or software). SIP features are implemented in the communicating endpoints (i.e. at the edge of the network) contrary to traditional SS7 features, which are implemented in the network.

Although several other VoIP signaling protocols exist (such as BICC, H.323, MGCP, MEGACO), SIP is distinguished by its proponents for having roots in the IP community rather than the telecommunications industry. SIP has been standardized and governed primarily by the IETF, while other protocols, such as H.323, have traditionally been associated with the International Telecommunication Union (ITU).

The first proposed standard version (SIP 1.0) was defined by RFC 2543. This version of the protocol was further refined to version 2.0 and clarified in RFC 3261, although some implementations are still relying on the older definitions.^[specify]

Network elements[link]

SIP also defines server network elements. Although two SIP endpoints can communicate without any intervening SIP infrastructure, which is why the protocol is described as peer-to-peer, this approach is often impractical for a public service. RFC 3261 defines these server elements.

User Agent[link]

A SIP user agent (UA) is a logical network end-point used to create or receive SIP messages and thereby manage a SIP session. A SIP UA can perform the role of a User Agent Client (UAC), which sends SIP requests, and the User Agent Server (UAS), which receives the requests and returns a SIP response. These roles of UAC and UAS only last for the duration of a SIP transaction.^[7]

A SIP phone is a SIP user agent that provides the traditional call functions of a telephone, such as dial, answer, reject, hold/unhold, and call transfer.^[8][9] SIP phones may be implemented as a hardware device or as a softphone. As vendors increasingly implement SIP as a standard telephony platform, often driven by 4G efforts, the distinction between hardware-based and software-based SIP phones is being blurred and SIP elements are implemented in the basic firmware functions of many IP-capable devices. Examples are devices from Nokia and Research in Motion.^[10]

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/product involved. The User-Agent field is sent in request messages, which means that the receiving SIP server can see this information. SIP network elements sometimes store this information,^[11] and it can be useful in diagnosing SIP compatibility problems.

Proxy server[link]

An intermediary entity that acts as both a server (UAS) and a client (UAC) for the purpose of making requests on behalf of other clients. A proxy server primarily plays the role of routing, which means its job is to ensure that a request is sent to another entity "closer" to the targeted user. Proxies are also useful for enforcing policy (for example, making sure a user is allowed to make a call). A proxy interprets, and, if necessary, rewrites specific parts of a request message before forwarding it.

Registrar[link]

A server that accepts REGISTER requests and places the information it receives in those requests into the location service for the domain it handles which registers one or more IP addresses to a certain SIP URI, indicated by the sip: scheme, although other protocol schemes are possible (such as tel:). More than one user agent can register at the same URI, with the result that all registered user agents will receive a call to the SIP URI.

SIP registrars are logical elements, and are commonly co-located with SIP proxies. But it is also possible and often good for network scalability to place this location service with a redirect server.

Redirect server[link]

A user agent server that generates 3xx (Redirection) responses to requests it receives, directing the client to contact an alternate set of URIs. The redirect server allows proxy servers to direct SIP session invitations to external domains.

Session border controller[link]

Session border controllers Serve as middle boxes between UA and SIP server for various types of functions, including network topology hiding, and assistance in NAT traversal.

Gateway[link]

Gateways can be used to interface a SIP network to other networks, such as the public switched telephone network, which use different protocols or technologies.

SIP messages[link]

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.^[12] The first line of a response has a response code.

For SIP requests, RFC 3261 defines the following methods:^[13]

• REGISTER: Used by a UA to indicate its current IP address and the URLs for which it would like to receive calls.
• INVITE: Used to establish a media session between user agents.
• ACK: Confirms reliable message exchanges.
• CANCEL: Terminates a pending request.
• BYE: Terminates a session between two users in a conference.
• OPTIONS: Requests information about the capabilities of a caller, without setting up a call.

A new method has been introduced in SIP in RFC 3262: ^[14]

• PRACK (Provisional Response Acknowledgement): PRACK improves network reliability by adding an acknowledgement system to the provisional Responses (1xx). PRACK is sent in response to provisional response (1xx).

The SIP response types defined in RFC 3261 fall in one of the following categories:^[15]

• Provisional (1xx): Request received and being processed.
• Success (2xx): The action was successfully received, understood, and accepted.
• Redirection (3xx): Further action needs to be taken (typically by sender) to complete the request.
• Client Error (4xx): The request contains bad syntax or cannot be fulfilled at the server.
• Server Error (5xx): The server failed to fulfill an apparently valid request.
• Global Failure (6xx): The request cannot be fulfilled at any server.

Transactions[link]

SIP makes use of transactions to control the exchanges between participants and deliver messages reliably. The transactions maintain an internal state and make use of timers. Client Transactions send requests and Server Transactions respond to those requests with one-or-more responses. The responses may include zero-or-more Provisional (1xx) responses and one-or-more final (2xx-6xx) responses.

Transactions are further categorized as either Invite or 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).

Because of these transactional mechanisms, SIP can make use of un-reliable transports such as User Datagram Protocol (UDP).

If we take the above example, User1’s UAC uses an Invite Client Transaction to send the initial INVITE (1) message. If no response is received after a timer controlled wait period the UAC may have chosen to terminate the transaction or retransmit the INVITE. However, once a response was received, User1 was confident the INVITE was delivered reliably. User1’s UAC then must acknowledge the response. On delivery of the ACK (2) both sides of the transaction are complete. And in this case, a Dialog may have been established.^[16]

Instant messaging and presence[link]

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. MSRP (Message Session Relay Protocol) allows instant message sessions and file transfer.

Conformance testing[link]

TTCN-3 test specification language is used for the purposes of specifying conformance tests for SIP implementations. SIP test suite is developed by a Specialist Task Force at ETSI (STF 196).^[17] The SIP developer community meets regularly at the SIP Forum SIPit events to test interoperability and test implementations of new RFCs.

Applications[link]

The market for consumer SIP devices continues to expand; there are many devices such as SIP Terminal Adapters, SIP Gateways, and SIP Trunking services providing replacements for ISDN telephone lines.

Many VoIP phone companies allow customers to use their own SIP devices, such as SIP-capable telephone sets, or softphones.

SIP-enabled video surveillance cameras can make calls to alert the owner or operator that an event has occurred; for example, to notify that motion has been detected out-of-hours 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.^[18]

SIP-ISUP interworking[link]

SIP-I, or the 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^[19] 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, which 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. SIP-I was defined by the ITU-T, where SIP-T was defined via the IETF RFC route.^[20]

See also[link]

• Computer telephony integration (CTI)
• Computer-supported telecommunications applications (CSTA)
• H.323 protocols H.225.0 and H.245
• IP Multimedia Subsystem (IMS)
• IP phone
• List of SIP software
• Media Gateway Control Protocol (MGCP)
• Message Session Relay Protocol (MSRP)
• Mobile VoIP
• MSCML (Media Server Control Markup Language)
• Network convergence
• RTP audio video profile
• Secure Real-time Transport Protocol (SRTP)
• SIGTRAN (Signaling Transport)
• SIP Trunking
• Skinny Client Control Protocol (SCCP)
• XIMSS (XML Interface to Messaging, Scheduling, and Signaling)
• ZRTP

References[link]

External links[link]

• Computers/Internet/Protocols/SIP/ at the Open Directory Project
• Henning Schulzrinne's SIP homepage hosted by Columbia University
• SIP Latest specifications hosted by SIPKnowledge
• IANA: SIP Parameters
• IANA: SIP Event Types Namespace

This article needs additional citations for verification. Please help improve this article by adding citations to reliable sources. Unsourced material may be challenged and removed. (July 2009)

In the field of computer networking, a URI scheme is the top level of the uniform resource identifier (URI) naming structure. All URIs and absolute URI references are formed with a scheme name, followed by a colon character (":"), and the remainder of the URI called (in the outdated RFCs 1738 and 2396, but not the current STD 66/RFC 3986) the scheme-specific part. The syntax and semantics of the scheme-specific part are left largely to the specifications governing individual schemes, subject to certain constraints such as reserved characters and how to "escape" them.

URI schemes are frequently erroneously referred to as "protocols", or specifically as URI protocols or URL protocols, since most were originally designed to be used with a particular protocol, and often have the same name. The http scheme, for instance, is generally used for interacting with Web resources using HyperText Transfer Protocol. Today, URIs with that scheme are also used for other purposes, such as RDF resource identifiers and XML namespaces, that are not related to the protocol. Furthermore, some URI schemes are not associated with any specific protocol (e.g. "file") and many others do not use the name of a protocol as their prefix (e.g. "news").

URI schemes should be registered with IANA, although non-registered schemes are used in practice. RFC 4395 describes the procedures for registering new URI schemes.

Generic syntax[link]

Internet standard STD 66 (also RFC 3986) defines the generic syntax to be used in all URI schemes. Every URI is defined as consisting of four parts, as follows:

<scheme name> : <hierarchical part> [ ? <query> ] [ # <fragment> ]

The scheme name consist of a sequence of characters beginning with a letter and followed by any combination of letters, digits, plus ("+"), period ("."), or hyphen ("-"). Although schemes are case-insensitive, the canonical form is lowercase and documents that specify schemes must do so with lowercase letters. It is followed by a colon (":").

The hierarchical part of the URI is intended to hold identification information hierarchical in nature. Usually this part begins with a double forward slash ("//"), followed by an authority part and an optional path.

• The authority part holds an optional user-information part, terminated with "@" (e.g. username:password@); a hostname (e.g., domain name or IP address); and an optional port number, preceded by a colon ":".

• The path part is a sequence of segments (conceptually similar to directories, though not necessarily representing them) separated by a forward slash ("/"). Each segment can contain parameters separated from it using a semicolon (";"), though this is rarely used in practice.

The query is an optional part, separated by a question mark ("?"), that contains additional identification information that is not hierarchical in nature. The query string syntax is not generically defined, but it is commonly organized as a sequence of <key>=<value> pairs, with the pairs separated by a semicolon^[1][2][3] or an ampersand. For example:

Semicolon: key1=value1;key2=value2;key3=value3
Ampersand: key1=value1&key2=value2&key3=value3

The fragment is an optional part separated from the front parts by a hash ("#"). It holds additional identifying information that provides direction to a secondary resource, e.g., a section heading (in an article) identified by the remainder of the URI. When the primary resource is an HTML document, the fragment is often an id attribute of a specific element and web browsers will make sure this element is visible.

Examples[link]

The following figure displays two example URIs (foo://username:password@example.com:8042/over/there/index.dtb?type=animal&name=narwhal#nose and urn:example:animal:ferret:nose) and their component parts. (The examples are derived from RFC 3986 — STD 66, chapter 3).

  foo://username:password@example.com:8042/over/there/index.dtb?type=animal&name=narwhal#nose
  \_/   \_______________/ \_________/ \__/            \___/ \_/ \______________________/ \__/
   |           |               |       |                |    |            |                |
   |       userinfo         hostname  port              |    |          query          fragment
   |    \________________________________/\_____________|____|/ \__/         \__/ 
   |                    |                          |    |    |    |           |         
scheme              authority                    path   |    |    interpretable as keys
 name   \_______________________________________________|____|/       \____/     \_____/ 
   |                         |                          |    |          |           |
   |                 hierarchical part                  |    |    interpretable as values
   |                                                    |    |
   |            path               interpretable as filename |
   |   ___________|____________                              |
  / \ /                        \                             |
  urn:example:animal:ferret:nose               interpretable as extension

 scheme
  name  userinfo  hostname       query
  _|__   ___|__   ____|____   _____|_____
 /    \ /      \ /         \ /           \
 mailto:username@example.com?subject=Topic

Official IANA-registered schemes[link]

The official URI schemes registered with the IANA are as follows (note that "IETF Draft" in no way is a finalized specification and must not be treated so; refer to Section 2.2 of RFC 2026 for more details):

Unofficial but common URI schemes[link]

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