Session Initiation Protocol
||This article may be too technical for most readers to understand. (December 2011)|
|Internet protocol suite|
The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating sessions consisting of one or several media streams. SIP can be use for two-party (unicast) or multiparty (multicast) sessions. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.
Originally designed by Henning Schulzrinne and Mark Handley in 1996, SIP has been developed and standardized in RFC 3261 under the auspices of the Internet Engineering Task Force (IETF). It 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). It is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP).
SIP works in conjunction with several other application layer protocols that identify and carry the session media. Media identification and negotiation is achieved with the Session Description Protocol (SDP). For the transmission of media streams (voice, video) SIP typically employs the Real-time Transport Protocol (RTP) or Secure Real-time Transport Protocol (SRTP). For secure transmissions of SIP messages, the protocol may be encrypted with Transport Layer Security (TLS).
- 1 History
- 2 Protocol design
- 3 Network elements
- 4 SIP messages
- 5 Transactions
- 6 Instant messaging and presence
- 7 Conformance testing
- 8 Applications
- 9 SIP-ISUP interworking
- 10 Deployment issues
- 11 Encryption
- 12 See also
- 13 References
- 14 External links
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 IETF Network Working Group published RFC 3261 - as of 2013[update] the latest version of the specification - in June 2002.
The U.S. National Institute of Standards and Technology (NIST), Advanced Networking Technologies Division provides a public-domain implementation of the Java standard for SIP 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 6665 (Subscribe / Notify) and RFC 3262 (Provisional Reliable Responses) etc.
SIP employs design elements similar to the HTTP request/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.
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 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
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 but it does not provide real end-to-end security to prevent espionage and law enforcement interception, as the 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) rules 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]
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.
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.
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. 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 BlackBerry.
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, and it can be useful in diagnosing SIP compatibility problems.
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.
A registrar is a SIP endpoint that accepts REGISTER requests and places the information it receives in those requests into a location service for the domain it handles. The location service links one or more IP addresses to the SIP URI of the registering agent. The URI uses 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 receive the calls to the 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.
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
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.
- 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.
- 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).
- 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.
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.
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. MSRP (Message Session Relay Protocol) allows instant message sessions and file transfer.
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). The SIP developer community meets regularly at the SIP Forum SIPit events to test interoperability and test implementations of new RFCs.
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.
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-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 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. A Session Initiation Protocol (SIP) connection is a Voice over Internet Protocol (VoIP) service offered by many Internet telephony service providers (ITSPs) that connects a company's private branch exchange (PBX) telephone system to the public switched telephone network (PSTN) via the Internet.
Using a SIP connection may simplify administration for the organization as the SIP connection typically uses the same Internet access that is used for data. This often removes the need to install Basic Rate Interface (BRI) or Primary Rate Interface (PRI) telephone circuits.
If the call traffic runs on the same connection with other traffic, such as email or Web browsing, voice and even signaling packets may be dropped and the voice stream may be interrupted.
To mitigate this, many companies split voice and data between two separate internet connections. Alternately, some networks use the TOS precedence or DiffServ fields in the header of IPV4 packets to mark the relative time-sensitivity of SIP and RTP as compared to web, email, video and other types of IP traffic. This precedence marking method requires that all routers in the SIP and RTP paths support separate queues for different traffic types. Other options to control delay and loss include incorporating multiple VLANs (virtual local area networks), traffic shaping to avoid this resource conflict, but the efficacy of this solution is dependent on the number of packets dropped between the Internet and the PBX. Registration is required if the end user has a dynamic IP address, if the provider does not support static hostnames, or if NAT is used. In order to share several DID numbers on the same registration, the IETF has defined additional headers (for example "P-Preferred-Identity", see RFC 3325). This avoids multiple registrations from one PBX to the same provider. Using this method the PBX can indicate what identity should be presented to the Called party and what identity should be used for authenticating the call. This feature is also useful when the PBX redirects an incoming call to a PSTN number, for example a cell phone, to preserve the original Caller ID.
Users should also be aware that a SIP connection can be used as a channel for attacking the company's internal networks, similar to Web and Email attacks. Users should consider installing appropriate security mechanisms to prevent malicious attacks.
The increasing concerns about security of calls that run over the public Internet has made SIP encryption more popular. Because VPN is not an option for most service providers, most service providers that offer secure SIP (SIPS) connections use TLS for securing signalling. The relationship between SIP (port 5060) and SIPS (port 5061), is similar to that as for HTTP and HTTPS, and uses URIs in the form "sips:email@example.com". The media streams, which occur on different connections to the signalling stream, can be encrypted with SRTP. The key exchange for SRTP is performed with SDES (RFC 4568), or the newer and often more user friendly ZRTP (RFC 6189), which can automatically upgrade RTP to SRTP using dynamic key exchange (and a verification phrase). One can also add a MIKEY (RFC 3830) exchange to SIP and in that way determine session keys for use with SRTP.
- Rendezvous protocol
- Peer-to-peer SIP
- 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
- SIGTRAN (Signaling Transport)
- SIP Trunking
- SIP provider
- Skinny Client Control Protocol (SCCP)
- XIMSS (XML Interface to Messaging, Scheduling, and Signaling)
- RFC 4168, The Stream Control Transmission Protocol (SCTP) as a Transport for the Session Initiation Protocol (SIP), IETF, The Internet Society (2005)
- Johnston, Alan B. (2004). SIP: Understanding the Session Initiation Protocol, Second Edition. Artech House. ISBN 1-58053-168-7.
- "SIP core working group charter". Ietf.org. 2010-12-07. Retrieved 2011-01-11.
- "JAIN SIP project". Retrieved 2011-07-26.
- William Stallings, p.209
- RFC 3986, Uniform Resource Identifiers (URI): Generic Syntax, IETF, The Internet Society (2005)
- RFC 3261, SIP: Session Initiation Protocol
- Azzedine (2006). Handbook of algorithms for wireless networking and mobile computing. CRC Press. p. 774. ISBN 978-1-58488-465-1.
- Porter, Thomas; Andy Zmolek, Jan Kanclirz, Antonio Rosela (2006). Practical VoIP Security. Syngress. pp. 76–77. ISBN 978-1-59749-060-3.
- "BlackBerry MVS Software". Na.blackberry.com. Retrieved 2011-01-11.
- "User-Agents We Have Known "VoIP User.org
- Stallings, p.214
- Stallings, pp.214-215
- Stallings, pp.216-217
- James Wright. "SIP - An Introduction" (PDF). Konnetic. Retrieved 2011-01-11.
- Experiences of Using TTCN-3 for Testing SIP and also OSP[dead link]
- Jonsson, Lars; Mathias Coinchon (2008). "Streaming audio contributions over IP" (PDF). EBU Technical Review. Retrieved 2010-12-27.
- "RFC3372: SIP-T Context and Architectures". 2002-09. Retrieved 2011-01-11.
- White Paper: "Why SIP-I? A Switching Core Protocol Recommendation"