SOFTWARE CORNER Session Initiation Protocol – Pushing

telephony toward reality By Curtis A. Schwaderer

Introduction Multipoint Controller Units manage all infancy, flexibility in the solution is The Internet continues to impact every aspects of multipoint conferences by man- needed. More specification work would facet of how individuals and enterprises aging streams of conferencing also still be needed to support implemen- throughout the world communicate with data and coordinating participants within tation of advanced features required now each other. Data, e-mail, and information a multipoint conference. Gateways pro- and into the future. access are mainstays of the Internet. Voice, vide the interface between the H.323 net- the last bastion of communications thought work and the broader telephony networks That’s not to say SIP threw everything to be too critical for Internet technology, including the PSTN or ISDN networks. away and started all over. Quite the con- is moving gradually toward becoming a Gatekeepers are optional elements in the trary Ð SIP is a very narrowly focused viable alternative to Public Switched Tele- H.323 network. They are used to control specification and leverages (but doesn’t phone Networks (PSTN) and Integrated access to conferences by H.323 terminals, depend on) a number of industry standard Services Digital Network (ISDN) tele- providing authorization and authentica- protocols and services. Many of which are phony. There are many attractive cost and tion services. also used by the H.323 standards. technology benefits supporting this evolu- tion. Internet communication is far less Figure 1 shows the H.323 protocol stack. SIP was originally developed for large- expensive and is also better equipped to You’ll notice that the entire stack is speci- scale multipoint conference signaling on support multimedia interactivity and ser- fied including CODECs, registration, call the Internet Multicast Backbone (Mbone). vices. The Internet Engineering Task Force control signaling, authentication, real-time However, it was soon realized that SIP is (IETF) continues to be active in creating data streaming, and endpoint message equally able to provide connection control protocols to bring reliability and advanced streams. For example, the H.323 call con- for point-to-point calls in addition to con- intelligent network services telephony to trol signaling is based on another ITU ferences. SIP is laser-beam focused on the Internet. Among the newer initiatives in specification called Q.931 which is used as providing a single service for VoIP sys- this area is the Session Initiation Protocol. call control signaling in ISDN networks. tems Ð call control.

The Session Initiation Protocol (SIP) is a The early strength of the H.323 specifi- SIP is a modular call control protocol and protocol governed by the IETF and cation was the fact that it is completely free from underlying protocol or architec- designed for providing advanced tele- specified. This complete specification for tural constraints. The focus of the SIP pro- phony and video services across the VoIP services helped with interoperability tocol along with its definition makes it Internet. SIP isn’t the first standardized between product implementations. How- simple and flexible. The IETF standards protocol for audio and video conferenc- ever, the entire stack is very complex. process lends itself better to specification ing. The H.323 protocol covers all aspects Things not supported by H.323 can’t be evolution for IP based services as well. of Internet-based telephony. However, deployed, so flexibility is a problem. For these reasons, SIP is gaining popular- there are a number of issues with the Since VoIP as a viable, reliable replace- ity over H.323. For example, SIP is the H.323 approach that SIP and the IETF ment to standard telephone service is in its session management protocol of choice specification process intend to solve. In this month’s column, we’ll look at SIP, how it works, where it’s being deployed, and who the vendors are.

Voice over IP (VoIP) history The first widely adopted set of VoIP pro- tocols was developed by the International Union (ITU). The protocol was governed by the H.323 spec- ification entitled Packet Based Multi- media Communications Systems.

The H.323 specifies H.323 terminals, Multipoint Controller Units (MCU), gateways, and gatekeepers. The H.323 terminals are the IP-phones, or endpoints in the IP telephony network. Figure 1

Reprinted from CompactPCI Systems / April 2003 ©Copyright 2003 SOFTWARE CORNER for the 3rd Generation Partnership Project There are four kinds of servers defined How SIP works (3GPP) for the development of an all-IP by SIP: A SIP session can be a simple two-way wireless network. telephone call or it could be a collabora- ■ Proxy server tive multimedia conference session. The What is SIP? ■ Redirect server large range of support that SIP offers SIP is an signaling pro- ■ User agent server enables a large number of new services tocol for establishing, manipulating, and ■ Registrar such as voice-enriched e-commerce, Web- tearing down sessions over an Internet page click-to-dial, instant messaging with Protocol (IP) based network. SIP’s main Clients typically send connection requests buddy lists, and IP services. purpose is to help session originators to proxy servers. The proxy server will deliver invitations to potential session par- either handle the request itself or forward Figure 3 shows some of the detailed mes- ticipants wherever they may be. The SIP the request to another server after some saging behind the SIP protocol. You’ll protocol is text based and similar to other translation. The proxy server enables user notice from the syntax that it is similar to text based protocols like HTTP and agent servers to be hidden from external HTTP or SMTP. Connection setup starts SMTP. SIP provides the ability to register user agent clients. SIP registrars message with a user agent client (UAC) sending an locations and establish and tear-down with user agents to generate and maintain a INVITE message. A unique call identifier multimedia sessions. location database that can be used by the is generated by the caller and contained in redirect servers to reach the correct end- the message. The call ID will identify the SIP services include: points when user clients attempt to call SIP control messaging for this unique other user agents in the SIP-based network. message transaction. There is also a con- ■ Name translation and user Redirect servers use the current location tent delimiter in SIP. This enables SIP to location. SIP is able to track users information generated by the registrars to send additional payload as part of the con- and establish sessions wherever the respond to user agents with a new translated nection control process. Remember, SIP is users are located. This is accom- address. This way, the redirect server func- just the connection control protocol Ð it plished through SIP registrar and tion implements call forwarding and “fol- handles nothing else in terms of the kinds redirect server services. SIP users can low me” services within the VoIP network. of connections being set up. In the case of register a variety of locations and have Figure 3, you’ll notice something called the ability to message where to have Figure 2 shows the basic SIP messaging SDP as the content type. This is the their calls forwarded to at any involved with a redirect server. Prior to the Session Description Protocol. SDP is used moment in time. messaging shown in Figure 2, SIP user to negotiate the type of connections being ■ Feature management. SIP provides the agents can register with the SIP registrar. set up and torn down. While we won’t ability to negotiate different levels of The registrar works with the redirect server cover the SDP protocol in any detail, service, so endpoints with varying capa- to know the location of endpoints at any you’ll notice SDP is further negotiating bilities can still be session participants. given time. Registrations involve a list of all the use of the Real Time Protocol (RTP) This provides a robust environment possible locations for the user and an indi- for this session. where legacy equipment may at least be cation on where to reach them at any given able to be baseline-interoperable with time. So, prior to initiating the call, a user The callee responds with a RINGING SIP user endpoints. Likewise, advanced can issue a connection request which may acknowledgement if the callee is found SIP user equipment can enjoy the full be sent to a redirect server. The redirect and user notification is underway. Once benefits of multimedia conferencing as server can then look in the user database to the callee answers the SIP call, an OK their capabilities permit. map the requested address to the current is sent back to the user agent client. ■ Call control features. SIP provides location of the user and send this informa- You’ll see in Figure 3 that another SDP the ability to add, remove, or transfer tion back to the requestor. This message is payload accompanies the OK packet. The users in addition to putting users on sent back using a Contact: message. Once caller sends an ACK packet indicating hold. This provides features standard the requestor has the new address informa- that the negotiated SDP session para- in telephony on the Internet. This kind tion, it can then contact the user at the cur- meters are OK and the SIP call is set up of standardized capability is important rent location to set up the VoIP call. end-to-end. to provide widely deployed SIP services using IP based networks. ■ Call feature changes. SIP provides the ability to change the characteristics of an ongoing session. If things like video or data transmission are needed during the course of a session, SIP can dynamically enable and disable these.

SIP is based on a classic client/server model. The user agent client (UAC) is an applica- tion program that originates SIP requests. The server is the entity that responds to these requests. Calls are originated by a user agent client and terminate at a server. Figure 2

Reprinted from CompactPCI Systems / April 2003 ©Copyright 2003 Status codes are sent in SIP messages in VoIP. In order to provide telephony ser- between traditional networks and Telia’s response to requests. These status codes vices there a number of other standards new IP network. Number portability was are encoded for easy classification of and protocols. The Reliable Transport chosen as a pilot service to keep track of responses. The first digit of the Status- Protocol (RTP) ensures that multimedia the current phone number of a user as they Code defines the class of response. The payload gets received and processed prop- change IP address. Services were created last two digits are assigned unique num- erly. RADIUS and DIAMETER are two that spanned across underlying wireless bers to identify specific responses within protocols that can be used to authenticate and wire line networks (IP, GSM, ISDN, the class. For this reason, any response users. User directory services are provided PSTN); non-SIP phones are viewed as SIP with a status code between 100 and 199 by the Lightweight Directory Access phones using the Telia Golden Gate pro- is referred to as a 1xx response, any Protocol (LDAP). Quality of service and ject technology. To demonstrate opera- response with a status code between 200 legacy telephony equipment interworking tion, a call screening service was created. and 299 as a 2xx response, and so on. is provided by protocols like RSVP and This service directs incoming phone calls SIP/2.0 specifies 6 classes listed below YESSIR. All these protocols can be used to different numbers, depending on the taken from RFC3261: in support of a SIP-based VoIP network. time and who is calling. The Golden Gate project output is projected to be the plat- ■ 1xx: Provisional Ð request received, Who is using SIP? form for future commercial SIP service continuing to process the request Besides SIP being the specified protocol offerings from Telia. ■ 2xx: Success Ð action was successfully for the 3GPP project, a number of other received, understood, and accepted companies are using SIP in their network Level 3 Communications offers advanced ■ 3xx: Redirection Ð further action deployments and have begun trials using wholesale voice services, terminating needs to be taken in order to complete the technology. Three of these activities voice traffic via its dedicated IP backbone the request are described below. This information and to gateway locations worldwide. Level 3 ■ 4xx: Client Error Ð the request more can be found by accessing the SIP uses SIP signaling as a part of this service. contains bad syntax or cannot be Center Web site at www.sipcenter.com. Level 3 calls this their (3) Voice service. fulfilled at this server Customers of the (3) Voice service can ■ 5xx: Server Error Ð the server failed Telia started a project called Golden Gate develop personalized Application Service to fulfill an apparently valid request in 1999. The goal of this project was to Provider and enhanced services over the ■ 6xx: Global Failure Ð the request develop an interoperable service platform open architecture of a SIP-based network. cannot be fulfilled at any server between multiple networks. A SIP net- Level 3 has also created a self-certification work architecture and SIP-enabled ser- program that allows VoIP equipment and SIP and SDP are just two of the modular vices was chosen as the base technology service providers to interoperate with the protocols used in the IETF architecture for for the project. A gateway was built Level 3 (3) Voice network. Participants

Figure 3

Reprinted from CompactPCI Systems / April 2003 ©Copyright 2003 SOFTWARE CORNER will execute test plans relating to SIP sig- SIP references less manufacturers creating products for naling and RTP transport. The SIP Center Web site referenced in this third-generation wireless networks are article provides an amazing amount of already adopting an all-IP network with Reuters Messaging is a third interesting technical, market, and vendor information SIP as the protocol for providing session application of SIP based services. Reuters on SIP. It’s extremely well organized and control between endpoints. Internet infra- messaging is a real-time communications contains lots of information on SIP tech- structure continues to advance and prolif- tool designed for financial industry pro- nology and events. The SIP center URL is erate, providing bandwidth and maturing fessionals. Instant messaging gives users www.sipcenter.com. the quality of service objectives required in the financial industry immediate access for telephony over the Internet. Major to key market contacts around the world. Another good source of information on telecommunications equipment manu- Security is also important in the financial the SIP specifications is the IETF RFC facturers continue to evolve telephone industry and SSL encryption and message Web site at www.ietf.org. Here you can networks with increased pressure to pro- logging ensures conversations remain find RFC3261 which governs the SIP pro- vide additional services and features at confidential and compliant with current tocol in addition to a number of related lower cost points. Next-generation tele- financial services regulations. RFCs for RTP, RSVP, SDP, and others phony services using IP promise wide that make up the modular building blocks spread multimedia features that can be There are literally hundreds of companies of the SIP-based VoIP network. widely deployed. The SIP protocol offering SIP related products and services. promises to be a foundation piece to this They broadly fall into the domains of SIP Summary evolution. stacks and software consulting services, The promise for multimedia enriched SIP board-level platforms, SIP OEM telephony offered through the Internet is products, and SIP based applications. The undeniable. Of course, high-availability SIP center Web site also has a complete and reliability issues relating to Internet listing of vendors of SIP related products. services must be addressed. However, can There are a large number of wireless and you imagine telephony services offered wire line network equipment manufactur- over the Internet in the future? The three ers prominently identified at the SIP major forces on communications seems Center Web site Ð Alcatel, Ericsson, Nokia, to be headed to a convergence point with and Siemens being just a few. the Internet Protocol at the nexus. Wire-

Reprinted from CompactPCI Systems / April 2003 ©Copyright 2003