Session Initiation Protocol

'Session Initiation Protocol' (SIP) is a protocol developed by the IETF MMUSIC Working Group and proposed standard for setting up sessions between one or more clients. It is currently (2005) the leading signaling protocol for Voice over IP, gradually replacing H.323 in this role.

Contents

1 Protocol Design 2 SIP Network Elements 3 Instant Messaging (IM) and Presence 4 Commercial Application 5 Software 6 See also 7 External links

Protocol Design

A goal for SIP was to provide a superset of the call processing functions and features present in the public switched telephone network (PSTN). As such, features that permit familiar telephone-like operations are present: dialing a number, causing a phone to ring, hearing ringback tones or a busy signal. Implementation and terminology are different.

SIP also implements many of the more advanced call processing features present in Signalling System 7 (SS7), though the two protocols themselves could hardly be more different. SS7 is a highly centralized protocol, characterized by highly complex central network architecture and dumb endpoints (traditional telephone handsets). SIP is a peer-to-peer protocol. As such it requires only a very simple (and thus highly scalable) core network with intelligence distributed to the network edge, embedded in endpoints (terminating devices built in either hardware or software). Many SIP features are implemented in the communicating endpoints as opposed to traditional SS7 features, which are implemented in the network.

Although many other VoIP signaling protocols exist, SIP is characterized by its roots in the IP community rather than the telecom industry. SIP is being standardized and governed by the IETF while older, more complex VoIP protocols were proposed by the ITU.

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, very simple and request-response. 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. It shares many HTTP status codes, such as the familiar '404 not found'. Much of the promise of SIP is that the rapid innovation and application development that has characterized the Web will now mark the telephony industry too. SIP is not limited to voice but can mediate any kind of communication session from voice to video to future, unrealized applications.

SIP Network Elements

Hardware endpoints, devices with the look, feel, and shape of a traditional telephone, but that use SIP and RTP for communication, are commercially available from several vendors. Some of these can use Electronic Numbering (ENUM) to translate existing phone numbers to SIP addresses using DNS, so calls to other SIP users can bypass the telephone network, even though your service provider might normally act as a gateway to the PSTN network for traditional phone numbers (and charge you for it).

Today, software SIP endpoints are more common. Microsoft Windows Messenger uses SIP and in June, 2003, Apple Computer announced, and released in public beta, iChat AV, a new version of their AOL Instant Messenger compatible client that supports audio and video chat through SIP.

SIP also requires proxy and registrar network elements to work as a practical service. 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 softswitch implementations (by Nortel, Sonus and many more) which can act as proxy and registrar. Other companies, led by Ubiquity Software and Dynamicsoft have implemented products based on the proposed standards, building on the Java JAIN specification.

Instant Messaging (IM) and Presence

A standard instant messaging protocol based on SIP, called SIMPLE, has been proposed and is under development. SIMPLE can also carry Presence Information, conveying a person's willingness and ability to engage in communications. Presence information is most recognizable today as buddy status in IM clients such as MSN Messenger and AIM. Some efforts have been made to integrate SIP-based VoIP with the XMPP presence specification used by Jabber.

Commercial Application

The Real-time Transport Protocol (RTP) used to carry the media stream does not traverse NAT routers. Most SIP clients can use STUN to traverse full cone, restricted cone, and port restricted cone NAT but not symmetrical NAT. Also some newer routers now recognize and pass SIP traffic. RTP Proxies, special purpose SIP line speed processors analogous to HTTP proxies commonly used in the early 1990s, enable CALEA and traversal of older, SIP-unaware NAT devices.

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. However, as commercial SIP services begin to take off practical solutions to these problems are being proven. Standards being developed by such organizations as 3GPP and 3GPP2 define applications of the basic SIP model which facilitate commercialization and enable support for network-centric capabilities such as CALEA.

Companies such as Vonage and SIPphone were consumer SIP pioneers and have a fast growing subscriber base. Major carriers like AT&T and Level(3) are now following suit. Software vendors like Rostrvm Solutions are also embracing SIP to provide a standards-based platform for call routing and contact management functions for call centre applications. The traditional telecommunications industry (including companies such as Lucent Technologies and Nortel) is now focused on developing systems based on the architecture model and SIP extensions as defined by 3GPP in their IP Multimedia Subsystem, IMS. [Packet8 (https://www.packet8.net/)] service from 8x8.com uses SIP to provide VOIP calling services and VIDEO PHONE services. It is the only company to provide IP based video phone.

Some VoIP phone companies, such as BroadVoice, allow customers to bring their own SIP devices, including SIP-capable telephone sets, the Asterisk PBX, or softphones. The new market for consumer SIP devices continues to expand.

Software

• SIP Servers:
  • BEA WebLogic SIP Server (http://www.bea.com/sip/)
  • Cisco SIP Proxy Server (http://www.cisco.com/univercd/cc/td/doc/product/voice/sipproxy/)
  • IPTel.org free SIP Express Router [1] (http://www.iptel.org/ser/)
  • SIP Proxy server, OnDO SIP Server[2] (http://www.brekeke.com), is available for Windows, Linux, and Mac OS X. It is free for personal and educational use.
• SIP Clients:
  • PhoneGaim is free software for SIP, based on Gaim.
  • X-Lite is a popular Windows softphone.
  • Kphone is a free Linux softphone using Qt libraries.
  • Konference is a video-conferencing application for KDE supporting SIP as signalling protocol
  • Linphone is a free Linux softphone using GTK libraries
  • SJPhone [3] (http://www.sjlabs.com/sjp.html) is a free SIP client that runs on Mac OS X, Windows, PocketPC and Linux
  • unofficial configuration [4] (http://chrisholland.blogspot.com/2005/05/configuring-sjphone-for-earthlink-sip.html) guidelines for EarthLink

See also

• Earthlink SIPshare
• Media Gateway Control Protocol (MGCP)

External links

• Page of Henning Schulzrinne's Co-author of the SIP Internet standards-track (http://www.cs.columbia.edu/~hgs/sip)
• Session Initiation Protocol (SIP) and Its Functions (http://www.windowsnetworking.com/articles_tutorials/Session-Initiation-Protocol-Functions.html)
• The SIP Forum page (http://www.sipforum.org)
• The SIP Specifications (RFCs):
  • RFC 3853: S/MIME Advanced Encryption Standard (AES) Requirement for the Session Initiation Protocol (SIP)
  • RFC 3702: Authentication, Authorization, and Accounting Requirements for the Session Initiation Protocol (SIP)
  • RFC 3261: SIP: Session Initiation Protocol html version (http://www.rfcsearch.org/rfcview/RFC/3261.html)
  • RFC 3515: The Session Initiation Protocol (SIP) Refer Method
• Packetizer's VoIP Protocol Information (http://www.packetizer.com/voip/sip/)
• VoIP User Community Resource (http://www.voipuser.org/)
• 3GPP - The Third Generation Partnership Project (http://www.3gpp.org/)de:Session Initiation Protocol

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