Background

The fast growth of the internet as a direct rival to the standard telephone network has led too strong economic and technological reasons for converged services and architectures to be developed. A signalling protocol was required to be able to allow a flow of information between standard telephony and IP based technology.

Many different organisations put forward ideas to solve the problem, each with their own varying interests and priorities. The internet based organisations suggested introducing innovative services (based on XML) and more open, peer-to-peer type protocols and call structures. The IETF themselves on the other hand offered SIP.

SIP had been originally developed to allow the invitation of people to large scale dial up type conferences on the internet Multicast Backbone (Mbone). IP telephony didn't really exist around this time, though it soon became apparent that SIP could be used to set up single point-to-point conferences - otherwise known as phone calls.

The SIP communication design approach excells at being a classic Internet type innovation: build only what you require and address only what is lacking in existing solutions. Because the SIP approach is totally modular and free from any underlying protocol or architectural constraints, and because the protocols themselves are incredibly simple, SIP has caught on fast as an alternative to H.323 (typically used for video conferencing) and to vendor-proprietary mechanisms for transporting SS7 protocols over the IP network.

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