While VoIP (Voice over Internet Protocol) continues to make great stride in the telecoms industry, there are still many people who are not aware what VoIP is and what it actually offers.
What is VoIP?
VoIP is a system which has been developed over the last few years which allows telephone communication over the internet and private data networks alike. It has only recently become viable with the ever increasing popularity of broadband and great jumps in internet data transfer speeds of late.
Prior to the recent increase in data transfer rates, the quality of VoIP systems was not commercially viable, because of the quality of the sound but this has all changed now. VoIP offers telephone communication as a fraction of the cost of traditional networks, as data is transferred using the highly efficient internet superhighway.
How Does It work?
In simple terms VoIP systems allow the transfer of audio information into digital data packets which can then be transferred over the internet at great speeds. The system is able to located the intended recipient of the call via their IP address, and deliver the data packets to their VoIP system.
At the receiving end the data is then converted back into audio data and presented through the VoIP handset or business PBX, Crystal clear sound delivered at break neck speeds along the internet.
Recent development have also seen the introduction of so called telecom exchange switches which allow VoIP uses to communicate with traditional phone networks. When moving from the internet to the traditional phone networks, the data packets are again switched to a format compatible with traditional networks, and delivered in the normal manner.
SIP
SIP (Session Initiation Protocol) is open architecture protocol that is widely used by ITSP (Internet Telephony Service Providers) to provide trunks (lines) over the internet for connection to both home and business. The benefits of this is simple. An ITSP can now port your existing UK number and provide it anywhere you like, even if out of area. This gives the flexibility of being able to move your numbers to anywhere you chose without having the geographical issue that you would have via the PSTN. Another widely use benefit of SIP and indeed of internet telephony as a whole is the ability to seamlessly link your remote offices via the internet or private lines. This gives us the ability to transfer calls and see full extension status access node including that of remote and mobility workers.
SIP has become a strong, catalytic force shaping today's telecom industry. This IETF driven protocol presents a key ingredient in the converging world of telecommunications based applications. But SIP does not do everything, and it does not solve every problem. SIP has limits, and SIP works with other protocols to get the job done.
Its actually good news that SIP does not try to solve everything single-handedly. We can examine this statement more closely with a quick look at the H.323 approach to telephony. H.323 is not a single protocol but rather an entire suite of protocols that cover everything from soup to nuts - codexs, call control, conferencing, and many other functions in one vertically integrated stack.
The advantage to this approach is that by strictly controlling so many aspects of the implementation it is easier to ensure that H.323 based systems function well together. On the down side, H.323 becomes very heavy and cumbersome. Flexibility is sacrificed as one is tied to a single family of technologies.
For a mature technology this may not be a problem, since the best solutions are likely to have been discovered and incorporated into standards. However for a field as young and fast changing as IP telephony, where many problems and solutions are still under debate, flexibility is more important. SIP is part of this flexible approach, as it uses a wide variety of protocols, each addressing a different aspect of the problem space. The advantage is the ability to chose from among many competing technologies and move to newer and better ones as the emerge. This has always been the philosophy behind SIP and this the approach the IETF to the IP telephony in general.
SIP is an important piece of this modular approach to IP telephony protocols. SIP addresses the need for a protocol to deal with generalised sessions. This involves finding potentially call participants and contacting them as the move from place to place, changing their location and even the equipment they are using. Calls may require the use of multiple streams of various media, and very large number of participants might be involved in a call - and even joining and leaving in a constant changing topology! This is what SIP does.
