Session Initiation Protocol

The Session Initiation Protocol (SIP) is a signalling protocol, widely used for setting up and tearing down multimedia communication sessions such as voice and video calls over the Internet. Other feasible application examples include video conferencing, streaming multimedia distribution, instant messaging, presence information and online games. The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams. The modification can involve changing addresses or ports, inviting more participants, adding or deleting media streams, etc.

SIP was originally designed by Henning Schulzrinne (Columbia University) and Mark Handley (UCL) starting in 1996. The latest version of the specification is RFC 3261 [ [http://www.ietf.org/rfc/rfc3261.txt RFC 3261] ] from the IETF SIP Working Group. [ [http://www.ietf.org/html.charters/sip-charter.html SIP working group charter] ] In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IMS architecture for IP-based streaming multimedia services in cellular systems.

The SIP protocol is situated at the session layer in the OSI model, and at the application layer in the TCP/IP model. SIP is designed to be independent of the underlying transport layer; it can run on TCP, UDP, or SCTP. SIP has the following characteristics:
* Transport-independent, because SIP can be used with UDP, TCP, SCTP, etc.
* Text-based, allowing for humans to read and analyze SIP messages.

Protocol design

SIP clients typically use TCP or UDP (typically on port 5060) to connect to SIP servers and other SIP endpoints. SIP is primarily used in setting up and tearing down voice or video calls. However, it can be used in any application where session initiation is a requirement. These include Event Subscription and Notification, Terminal mobility and so on. There are a large number of SIP-related RFCs that define behavior for such applications. All voice/video communications are done over separate session protocols, typically RTP. 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. However, it has been designed to enable the building of such features in network elements known as Proxy Servers and User Agents. These are features that permit familiar telephone-like operations: dialing a number, causing a phone to ring, hearing ringback tones or a busy signal. 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. As such 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) as opposed to traditional SS7 features, which are implemented in the network.

Although many other VoIP signaling protocols exist, SIP is characterized by its proponents as having roots in the IP community rather than the telecom industry. SIP has been standardized and governed primarily by the IETF while the H.323 VoIP protocol has been traditionally more associated with the ITU. However, the two organizations have endorsed both protocols in some fashion.

SIP works in concert with several other protocols and is only involved in the signaling portion of a communication session. SIP acts as a carrier for the Session Description Protocol (SDP), which describes the media content of the session, e.g. what IP ports to use, the codec being used etc. In typical use, SIP "sessions" are simply packet streams of the Real-time Transport Protocol (RTP). RTP is the carrier for the actual voice or video content itself.

The first proposed standard version (SIP 2.0) was defined in RFC 2543. The protocol was further clarified in RFC 3261, although many implementations are still using interim draft versions. Note that the version number remains 2.0.

SIP is similar to HTTP and shares some of its design principles: It is human readable and request-response structured. SIP shares many HTTP status codes, such as the familiar '404 not found'. SIP proponents also claim it to be simpler than H.323. However, some would counter that while SIP originally had a goal of simplicity, in its current state it has become as complex as H.323. Others would argue that SIP is a stateless protocol, hence making it possible to easily implement failover and other features that are difficult in stateful protocols such as H.323. SIP and H.323 are not limited to voice communication but can mediate any kind of communication session from voice to video or future, unrealized applications.

IP network elements

SIP User Agents (UAs) are the end-user devices, used to create and manage a SIP session. A SIP UA has two main components, the User Agent Client (UAC) send messages and answers with SIP responses, the User Agent Server (UAS) responds to SIP requests sent by the peer. SIP UAs may work in point to point mode. Typical implementations of a UA are SIP softphones, SIP hardphones and SIP-enabled ATAs.

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 impractical for a public service. There are various implementations that can act as SIP servers:

RFC 3261 defines these server elements:

:" Proxy, Proxy Server: An intermediary entity that acts as both a server and a client 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 server that accepts REGISTER requests and places the information it receives in those requests into the location service for the domain it handles."

:"A redirect server is a user agent server that generates 3xx responses to requests it receives, directing the client to contact an alternate set of URIs.The redirect server allows SIP Proxy Servers to direct SIP session invitations to external domains."

"It is an important concept that the distinction between types of SIP servers is logical, not physical."

Other SIP related network elements are :Session border controllers (SBC), they serve as "man in the middle" between UA and SIP server, see the article SBC for a detailled description.

:Various types of gateways at the edge between a SIP network and other networks (as a phone network)

Instant messaging (IM) 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.Some efforts have been made to integrate SIP-based VoIP with the XMPP specification used by Jabber. Most notably Google Talk, which extends XMPP to support voice, plans to integrate SIP. Google's XMPP extension is called Jingle and, like SIP, it acts as a Session Description Protocol carrier.

Conformance testing

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). [ [http://portal.etsi.org/ptcc/downloads/TTCN3SIPOSP.pdf Experiences of Using TTCN-3 for Testing SIP and also OSP] ]

Commercial applications

Firewalls typically block media packet types such as UDP, though one way around this is to use TCP tunnelling and relays for media in order to provide NAT and firewall traversal. One solution involves tunnelling the media packets within TCP or HTTP packets to a relay. This solution uses additional functionality in conjunction with SIP, and packages the media packets into a TCP stream which is then sent to the relay. The relay then extracts the packets and sends them on to the other endpoint. If the other endpoint is behind a symmetrical NAT, or corporate firewall that does not allow VoIP traffic, the relay would transfer the packets to another tunnel. One disadvantage of this approach is that TCP was not designed for real time traffic such as voice, so an optimized form of the protocol is sometimes used.

As envisioned by its originators, SIP's peer-to-peer nature does not enable network-provided services. For example, the network can not easily support legal interception of calls (referred to in the United States by the law governing wiretaps, CALEA). Emergency calls (calls to E911 in the USA) are difficult to route. It is difficult to identify the proper Public Service Answering Point, PSAP because of the inherent mobility of IP end points and the lack of any network location capability.

Many VoIP phone companies allow customers to bring their own SIP devices, as SIP-capable telephone sets, or softphones. The new market for consumer SIP devices continues to expand.

The free software community started to provide more and more of the SIP technology required to build both end points as well as proxy and registrar servers leading to a commodification of the technology, which accelerates global adoption. SIPfoundry has made available and actively develops a variety of SIP stacks, client applications and SDKs, in addition to entire IP PBX solutions that compete in the market against mostly proprietary IP PBX implementations from established vendors.

The National Institute of Standards and Technology (NIST), Advanced Networking Technologies Division provides a public domain implementation of the JAVA Standard for SIP [http://jain-sip.dev.java.net JAIN-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 3265 (Subscribe / Notify) and RFC 3262 (Provisional Reliable Responses) etc.

IP-ISUP interworking

IP-I

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 [ [http://www.3gamericas.org/pdfs/3G_Americas_SIP-I_White_Paper_August_2007-FINAL.pdf White Paper: "Why SIP-I? A Switching Core Protocol Recommendation"] ] .

ee also

*List of SIP request methods
*List of SIP response codes
*List of SIP software
*IP phone
*Media Gateway Control Protocol (MGCP)
*MSCML (Media Server Control Markup Language)
*IP Multimedia Subsystem
*Inter-Asterisk eXchange
*Voice over Internet Protocol
*Mobile VoIP
*Private branch exchange (PBX)
*Session Initiation Protocol (Java)
*SIGTRAN
*Skinny Client Control Protocol (SCCP)
* SRTP
*ZRTP

References

External links

*dmoz|Computers/Internet/Protocols/SIP/|Computers/Internet/Protocols/SIP/
* [http://www.cs.columbia.edu/sip/ Henning Schulzrinne's SIP homepage] hosted by Columbia University
* [http://www.sipknowledge.com/SIP_RFC.htm All SIP related RFCs categorized by IETF WGs]