The early days of Voice over IP were dominated by use over the Internet to get low cost international calls. With the "best efforts" nature of the Internet, though, quality was always going to be low and the threat to traditional telephony services was slight compared with other factors in the European market. Since then, most of the development of VoIP technology has been to create seriously high quality, high reliability services that business and residential end users alike cannot distinguish from the traditional PSTN services they know and like. Reaching this goal lays a path for convergence of voice and data networks. This is something that the telecoms industry has tried and largely failed to do over the last two decades but is now more important than ever to achieve with the growing need for mobile data services. SIP, the Session Initiation Protocol, is one of a number of protocols developed to achieve this and is therefore significant in that regard alone.

However, network convergence itself is no longer regarded as the sufficient end-point it was in the days when incumbent telcos started introducing ISDN. Now it represents a means of introducing new (as in "not seen before") services that can be rapidly introduced and that users will find useful and easy to use, will hopefully become reliant on and thereby generate new service revenues. In this area of rapid service creation, SIP is now also becoming increasingly significant.

The heart of the matter

Originating in 1996 as part of the development of multicasting, SIP came to prominence only once the direction of VoIP technology development moved from "low cost" to "value add". It is a lightweight, text-based protocol that is easily programmed, highly flexible and readily scalable – all attributes where it differs from its closest rival in the IP telephony services arena: the ITU’s H.323 umbrella recommendation.

As the name implies, SIP is about initiating interactive communications sessions between users. It also handles termination and, most interestingly, modifications of sessions in progress as well. Such a modification might mean converting a one-to-one telephony call into a multi-party videoconference as part of one single, continuous session, for example. It might also mean finding a particular user wherever he (or she) is and whatever type of terminal he’s using at any particular time. "Initiating a session" requires determining where the user to be contacted is actually located at that particular time. He might have a PC at work, another PC at home, a fixed line phone at work, at home or elsewhere. A call for that user might need to ring all phones and send instant messages to all PCs at once. He might be mobile – one day at work, the next visiting somewhere else. This dynamic location information needs to be taken into account in order to find him.

Once the user has been located, the correct session for the type of terminal he’s using at the time needs to be established. SIP achieves all of this.

The SIP advantage

SIP is a lightweight client easily embedded in end-user devices. Wherever there is a requirement for real-time sessions to be established, SIP can reside in the communications device and handle these sessions.

On-line gaming, an application with substantial revenue potential, can take advantage of this. Gaming requires real-time data sessions to be set up. Currently, many games also use instant messaging for communication between players during a game, to co-ordinate the play. SIP not only supports this functionality, but also opens the possibility of players being able to talk to one another. Games could become more voice-driven, more natural and more interactive as a result.

Mobile handsets are also destined to take advantage of this. 3GPP (Third Generation Partnership Project), which is setting the standards for UMTS, has now standardised on SIP for call control and signalling on third generation mobile networks. All IP voice and multimedia call signalling will be performed by SIP, end to end, providing a basis for rapid new service introductions and integration with fixed network IP services (such as streamed content) once the basic platform is in place.

Then there’s web commerce. SIP allows browsers to gain multimedia capability with powerful, but simple, services like click-to-dial becoming possible. Web call centres, where a particular web page can be popped onto an agent’s screen as a direct result of a user’s click-to-dial action on the web site, are now destined to evolve to multimedia contact centres, offering huge new service revenue opportunities.

There are many more potential areas of use. IP Centrex, instant messaging, presence management, desktop call management and unified messaging, to name a few.

Looking ahead

It is often forgotten that whereas data is now driving traffic growth in backbone networks, telephony is still where most of the revenue is. SIP now looks set to play a key role in extending the revenue potential of voice into many other application areas. The news that Microsoft’s Windows XP, due for release in October, will be SIP-enabled clearly adds to this potential.

There are still key issues to resolve, though. For example, there is an increasing shortage of IP v4 address numbers to accommodate all the new devices heading towards IP networks. This leads to rationing and re-allocation of addresses. A similar problem arises with the growing use of network address translators (NATs). The need for this IP address translation typically arises when a corporate network's internal IP addresses cannot be used outside the network either because they are invalid for use outside, or because the internal addressing must be kept private from the external network. Both of these could impact on the promise of SIP.

Nevertheless, the prospect of SIP-inspired services generating significant new revenues not only looks feasible. Given the current state of the industry it could prove particularly timely.

© e-principles 2001

Robin Duke-Woolley

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