The Session Initiation Protocol (SIP) is a signalling In the public switched telephone network, , in-band signalling is the exchange of signalling (call control) information within the same channel that the telephone call itself is using. An example is DTMF 'Dual-Tone multi-frequency' signalling, which is used on most telephone lines to exchanges protocol, widely used for setting up and tearing down multimedia Multimedia is media and content that uses a combination of different content forms. The term can be used as a noun or as an adjective describing a medium as having multiple content forms. The term is used in contrast to media which only use traditional forms of printed or hand-produced material. Multimedia includes a combination of text, audio, communication sessions In computer science, in particular networking, a session is a semi-permanent interactive information interchange, also known as a dialogue, a conversation or a meeting, between two or more communicating devices, or between a computer and user . A session is set up or established at a certain point in time, and torn down at a later point in time such as voice and video calls over Internet Protocol (IP). Other feasible application examples include video conferencing A videoconference is a set of interactive telecommunication technologies which allow two or more locations to interact via two-way video and audio transmissions simultaneously. It has also been called visual collaboration and is a type of groupware. It differs from videophone in that it is designed to serve a conference rather than individuals, streaming multimedia However, computer networks were still limited, and media was usually delivered over non-streaming channels, such as by downloading a digital file from a remote web server and then saving it to a local drive on the end user's computer or storing it as a digital file and playing it back from CD-ROMs distribution, instant messaging Instant messaging is a form of real-time communication between two or more people based on typed text. The text is conveyed via devices connected over a network such as the Internet, presence information In computer and telecommunications networks, presence information is a status indicator that conveys ability and willingness of a potential communication partner--for example a user--to communicate. A user's client provides presence information via a network connection to a presence service, which is stored in what constitutes his personal and online games An online game is a game played over some forms of computer network. At the present, this almost always means the Internet or equivalent technology; but games have always used whatever technology was current: modems before the internet, and hard wired terminals before modems. The expansion of online gaming has reflected the overall expansion of. The protocol can be used for creating, modifying and terminating two-party (unicast In computer networking, unicast transmission is the sending of information packets to a single network destination) or multiparty (multicast Multicast addressing is a network technology for the delivery of information to a group of destinations simultaneously using the most efficient strategy to deliver the messages over each link of the network only once, creating copies only when the links to the multiple destinations split) sessions consisting of one or several media streams. The modification can involve changing addresses or ports, inviting more participants, adding or deleting media streams A stream is a flowing body of water with a current, confined within a bed and stream banks. Depending on its locale or certain characteristics, a stream may be referred to as a branch, brook, beck, burn, creek, crick, kill, lick, rill, river syke, bayou, rivulet, or run. In some countries or communities a stream may be defined by its size. In the, etc.

SIP was originally designed by Henning Schulzrinne and Mark Handley starting in 1996. The latest version of the specification is RFC 3261[1] from the IETF The Internet Engineering Task Force develops and promotes Internet standards, cooperating closely with the W3C and ISO/IEC standard bodies and dealing in particular with standards of the TCP/IP and Internet protocol suite. It is an open standards organization, with no formal membership or membership requirements. All participants and leaders are Network Working Group.[2] In November 2000, SIP was accepted as a 3GPP The 3rd Generation Partnership Project is a collaboration between groups of telecommunications associations, to make a globally applicable third generation (3G) mobile phone system specification within the scope of the International Mobile Telecommunications-2000 project of the International Telecommunication Union (ITU). 3GPP specifications are signaling protocol and permanent element of the IP Multimedia Subsystem The IP Multimedia Subsystem is an architectural framework for delivering Internet Protocol (IP) multimedia services. It was originally designed by the wireless standards body 3rd Generation Partnership Project (3GPP), as a part of the vision for evolving mobile networks beyond GSM. Its original formulation (3GPP R5) represented an approach to (IMS) architecture for IP-based streaming multimedia services in cellular systems.

The SIP protocol is a TCP/IP The Internet Protocol Suite is the set of communications protocols used for the Internet and other similar networks. It is named from two of the most important protocols in it: the Transmission Control Protocol (TCP) and the Internet Protocol (IP), which were the first two networking protocols defined in this standard. Today's IP networking-based Application Layer Application Layer is a term used in categorizing protocols and methods in architectural models of computer networking. Both the OSI model and the Internet Protocol Suite contain an application layer protocol. Within the OSI model The Open systems Interconnection Reference Model is an abstract description for layered communications and computer network protocol design. It was developed as part of the Open Systems Interconnection (OSI) initiative. In its most basic form, it divides network architecture into seven layers which, from top to bottom, are the Application, it is sometimes placed in the session layer The Session Layer provides the mechanism for opening, closing and managing a session between end-user application processes, i.e. a semi-permanent dialogue. Communication sessions consist of requests and responses that occur between applications. Session Layer services are commonly used in application environments that make use of remote procedure. SIP is designed to be independent of the underlying transport layer; it can run on TCP The Transmission Control Protocol is one of the core protocols of the Internet Protocol Suite. TCP is one of the two original components of the suite (the other being Internet Protocol, or IP), so the entire suite is commonly referred to as TCP/IP. Whereas IP handles lower-level transmissions from computer to computer as a message makes its way, UDP The User Datagram Protocol is one of the core members of the Internet Protocol Suite, the set of network protocols used for the Internet. With UDP, computer applications can send messages, in this case referred to as datagrams, to other hosts on an Internet Protocol (IP) network without requiring prior communications to set up special transmission, or SCTP In computer networking, the Stream Control Transmission Protocol is a Transport Layer protocol, serving in a similar role as the popular protocols Transmission Control Protocol (TCP) and User Datagram Protocol (UDP). Indeed, it provides some of the same service features of both, ensuring reliable, in-sequence transport of messages with congestion. It is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol Hypertext Transfer Protocol is an application-level protocol for distributed, collaborative, hypermedia information systems. Its use for retrieving inter-linked resources led to the establishment of the World Wide Web (HTTP) and the Simple Mail Transfer Protocol Simple Mail Transfer Protocol is an Internet standard for electronic mail (e-mail) transmission across Internet Protocol (IP) networks. SMTP was first defined in RFC 821 (STD 10), and last updated by RFC 5321 (2008) which includes the extended SMTP (ESMTP) additions, and is the protocol in widespread use today (SMTP)[3], allowing for easy inspection by administrators.

The Internet Protocol Suite The Internet Protocol Suite is the set of communications protocols used for the Internet and other similar networks. It is named from two of the most important protocols in it: the Transmission Control Protocol (TCP) and the Internet Protocol (IP), which were the first two networking protocols defined in this standard. Today's IP networking
Application Layer Application Layer is a term used in categorizing protocols and methods in architectural models of computer networking. Both the OSI model and the Internet Protocol Suite contain an application layer
BGP The Border Gateway Protocol is the core routing protocol of the Internet. It maintains a table of IP networks or 'prefixes' which designate network reachability among autonomous systems (AS). It is described as a path vector protocol. BGP does not use traditional IGP metrics, but makes routing decisions based on path, network policies and/or · DHCP Dynamic Host Configuration Protocol is a network application protocol used by devices (DHCP clients) to obtain configuration information for operation in an Internet Protocol network. This protocol reduces system administration workload, allowing devices to be added to the network with little or no manual intervention · DNS The Domain Name System is a hierarchical naming system for computers, services, or any resource participating in the Internet. It associates various information with the domain names assigned to each of the participants. Most importantly, it translates domain names meaningful to humans into the numerical (binary) identifiers associated with · FTP File Transfer Protocol is a standard network protocol used to exchange and manipulate files over an Internet Protocol computer network, such as the Internet. FTP is built on a client-server architecture and utilizes separate control and data connections between the client and server applications. Client applications were originally interactive · GTP GPRS Tunnelling Protocol is a group of IP-based communications protocols used to carry General Packet Radio Service (GPRS) within GSM and UMTS networks · HTTP Hypertext Transfer Protocol is an application-level protocol for distributed, collaborative, hypermedia information systems. Its use for retrieving inter-linked resources led to the establishment of the World Wide Web · IMAP The Internet Message Access Protocol is one of the two most prevalent Internet standard protocols for e-mail retrieval, the other being the Post Office Protocol. Virtually all modern e-mail clients and mail servers support both protocols as a means of transferring e-mail messages from a server, such as those used by Gmail, to a client, such as · IRC Internet Relay Chat is a form of real-time Internet text messaging (chat) or synchronous conferencing. It is mainly designed for group communication in discussion forums, called channels, but also allows one-to-one communication via private message as well as chat and data transfers via Direct Client-to-Client · Megaco Megaco is an implementation of the Media Gateway Control Protocol architecture for controlling Media Gateways on Internet Protocol (IP) networks and the public switched telephone network (PSTN). The general base architecture and programming interface was originally described in RFC 2805 and the current specific Megaco definition is ITU-T · MGCP MGCP is an implementation of the Media Gateway Control Protocol architecture for controlling Media Gateways on Internet Protocol networks and the public switched telephone network (PSTN). The general base architecture and programming interface is described in RFC 2805 and the current specific MGCP definition is RFC 3435 (obsoleted RFC 2705). It is · NNTP The Network News Transfer Protocol or NNTP is an Internet application protocol used primarily for reading and posting Usenet articles , as well as transferring news among news servers. Brian Kantor of the University of California, San Diego and Phil Lapsley of the University of California, Berkeley completed RFC 977, the specification for the · NTP The Network Time Protocol is a protocol for synchronizing the clocks of computer systems over packet-switched, variable-latency data networks. NTP uses UDP on port 123 as its transport layer. It is designed particularly to resist the effects of variable latency by using a jitter buffer. NTP also refers to a reference software implementation that · POP In computing, the Post Office Protocol is an application-layer Internet standard protocol used by local e-mail clients to retrieve e-mail from a remote server over a TCP/IP connection. POP and IMAP (Internet Message Access Protocol) are the two most prevalent Internet standard protocols for e-mail retrieval. Virtually all modern e-mail clients and · RIP The Routing Information Protocol is a dynamic routing protocol used in local and wide area networks. As such it is classified as an interior gateway protocol (IGP). It uses the distance-vector routing algorithm. It was first defined in RFC 1058 (1988). The protocol has since been extended several times, resulting in RIP Version 2 (RFC 2453). Both · RPC Remote procedure call is an Inter-process communication technology that allows a computer program to cause a subroutine or procedure to execute in another address space (commonly on another computer on a shared network) without the programmer explicitly coding the details for this remote interaction. That is, the programmer would write essentially · RTP The Real-time Transport Protocol defines a standardized packet format for delivering audio and video over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and first published in 1996 as RFC 1889, and superseded by RFC 3550 in 2003 · RTSP The Real Time Streaming Protocol is a network control protocol for use in entertainment and communications systems to control streaming media servers. The protocol is used to establish and control media sessions between end points. Clients of media servers issue VCR-like commands, such as play and pause, to facilitate real-time control of playback · SDP The Session Description Protocol is a format for describing streaming media initialization parameters in an ASCII string. The IETF published the original specification as an IETF Proposed Standard in April 1998, and subsequently published a revised specification as an IETF Proposed Standard as RFC 4566 in July 2006 · SIP · SMTP Simple Mail Transfer Protocol is an Internet standard for electronic mail (e-mail) transmission across Internet Protocol (IP) networks. SMTP was first defined in RFC 821 (STD 10), and last updated by RFC 5321 (2008) which includes the extended SMTP (ESMTP) additions, and is the protocol in widespread use today · SNMP Simple Network Management Protocol is used in network management systems to monitor network-attached devices for conditions that warrant administrative attention. SNMP is a component of the Internet Protocol Suite as defined by the Internet Engineering Task Force (IETF). It consists of a set of standards for network management, including an · SOAP Soap is an anionic surfactant used in conjunction with water for washing and cleaning, which historically comes either in solid bars or in the form of a viscous liquid · SSH Secure Shell or SSH is a network protocol that allows data to be exchanged using a secure channel between two networked devices. Used primarily on Linux and Unix based systems to access shell accounts, SSH was designed as a replacement for Telnet and other insecure remote shells, which send information, notably passwords, in plaintext, leaving · Telnet Telnet is a network protocol used on the Internet or local area networks to provide a bidirectional interactive communications facility. Typically, telnet provides access to a command-line interface on a remote host via a virtual terminal connection which consists of an 8-bit byte oriented data connection over the Transmission Control Protocol ( · TLS/SSL Transport Layer Security and its predecessor, Secure Sockets Layer (SSL), are cryptographic protocols that provide security and data integrity for communications over networks such as the Internet. TLS and SSL encrypt the segments of network connections at the Transport Layer end-to-end · XMPP Extensible Messaging and Presence Protocol is an open, XML-based protocol originally aimed at near-real-time, extensible instant messaging (IM) and presence information (e.g., buddy lists), but now expanded into the broader realm of message oriented middleware. It remains the core protocol of the Jabber Instant Messaging and Presence technology · (more)
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Contents

Protocol design

SIP employs design elements similar to HTTP-like request/response transaction model.[4] 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 uses the Session Description Protocol (SDP), to exchange the session content.[4]

SIP clients typically use TCP or UDP (typically on port 5060 and/or 5061) 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 was designed to enable the construction of functionalities of network elements designated 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, 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, SIP is distinguished by its proponents for 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 is a carrier for the Session Description Protocol (SDP), which describes the media content of the session, e.g. what UDP 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 network 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.[1]

A SIP phone is a hardware-based or software-based SIP user agent, that provides call functions such as dial, answer, reject, hold/unhold, and call transfer.[5][6] Examples include softphones like Ekiga, KPhone, Twinkle, Windows Live Messenger, X-Lite, and hardware phones from vendors like Avaya, Cisco, Leadtek, Polycom, Snom, and Nokia.

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[7] 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 SIP messages must be transported over Transport Layer Security (TLS).[1]

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[8], and it can be useful in diagnosing SIP compatibility problems.

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 proxy server "is 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."

The RFC specifies: "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 detailed description.
Various types of gateways at the edge between a SIP network and other networks (as a phone network)

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.[9] The first line of a response has a response code.

For SIP requests, RFC 3261 defines the following methods:[10]

The SIP response types defined in RFC 3261 fall in one of the following categories:[11]

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. During an instant message session, files can be transferred using, for example, MSRP (Message Session Relay Protocol).

Some efforts have been made to integrate SIP-based VoIP with the XMPP specification. 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).[12]

Applications

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

Disadvantages

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.

Firewalls typically block media packet types such as UDP, though one way around this is to use TCP tunneling and relays for media in order to provide NAT and firewall traversal. One solution involves tunneling the media packets within TCP or HTTP/HTTPS 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.

SIP-ISUP interworking

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[13] 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[14].

See also

References

  1. ^ a b c RFC 3261, SIP: Session Initiation Protocol
  2. ^ SIP working group charter
  3. ^ Johnston, Alan B. (2004). SIP: Understanding the Session Initiation Protocol, Second Edition. Artech House. ISBN 1580531687.
  4. ^ a b William Stallings, p.209
  5. ^ Azzedine (2006). Handbook of algorithms for wireless networking and mobile computing. CRC Press. p. 774. http://books.google.com/books?id=b8oisvv6fDAC&pg=PT774.
  6. ^ Porter, Thomas; Andy Zmolek, Jan Kanclirz, Antonio Rosela (2006). Practical VoIP Security. Syngress. pp. 76-77. http://books.google.com/books?id=BYxdyekyRlwC&pg=PA76.
  7. ^ RFC 3986, Uniform Resource Identifiers (URI): Generic Syntax, IETF, The Internet Society (2005)
  8. ^ "User-Agents We Have Known " VoIP User.org
  9. ^ Stallings, p.214
  10. ^ Stallings, pp.214-215
  11. ^ Stallings, pp.216-217
  12. ^ Experiences of Using TTCN-3 for Testing SIP and also OSP
  13. ^ RFC3372: SIP-T Context and Architectures
  14. ^ White Paper: "Why SIP-I? A Switching Core Protocol Recommendation"

External links

Categories: Application layer protocols | Session layer protocols | VoIP protocols | VoIP terminology & concepts

 

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