VoIP is an abbreviation for Voice over Internet Protocol and it is a number of technologies that allow telephony to be conducted over an IP network instead of using the PSTN (Public Switched Telephone Network). VoIP has been around for something like 17 years or so,ever since the Israeli company Vocaltech first released their software-based Internet phone.

So how does VoIP operate?

In order to explain VoIP we will just look at a very brief explanation of traditional telephony.

In the traditional telecommunications world a fixed line telephone user is connected to a local telephone exchange by a pair of copper wires, which enable both analogue voice signals (speech) to be passed from the user, through a network of telecoms switches to the intended recipient in another area of the network. In order for this to happen the telecommunications network must have a number of components in place as well as a number of basic procedures. Firstly the user must be able to communicate with the Local Exchange (LEX) and signal its intent to connect to another user. A signalling protocol known as CAS (Channel Associated Signalling) is used for this purpose using telephone numbers that are broken down rather like postal codes. The telecommunications network must convey this signalling information in order to connect to a remote receiver. Telecommunications networks us something known as CCS (Common Channel Signalling) to do this. Finally, once the signalling CAS and CCS has resulted in the recipients phone ringing at the remote end, a circuit is set up through the network to pass the speech which is digitised prior to entering the network using a common Codec (Coder / Decoder). We refer to this method of communications as Circuit Switching, where the digitised speech is passed over a telecommunications circuit between telecoms switches where the circuit is created in advance of the flow of voice media.

True Voice over IP utilises a packet switched network using the Internet Protocol and associated protocols within the TCP/IP protocol suite. The analogue speech from the talker is first digitised using one of a number of industry recognised codecs and then broken up into small chunks which are then packetized ready for transmission using IP. The digitised speech is routed towards the recipient by means of IP Addressing and Routing Protocols in the same manner that other data such as HTTP and FTP are routed.

Just like traditional telecommunications, a signalling protocol is needed to allow the sender to indicate which remote ip phone should receive the call. In the late 1990s there were two signalling protocols competing for role. These two protocols were the ITU (International Telecommunications Union) H.323 protocol suite and the IETF (Internet Engineering Task Force) SIP (Session Initiation Protocol). H.323 lost the battle and SIP has become the dominant signalling protocol for VoIP. IP Addresses and Telephone numbers need to be linked together and SiP actually uses URLs in a similar manner to which the HTTP protocol uses URLs. This allows SIP to use the services of DNS (Domain Name Service), and therefore is routable across the Internet.

Just like traditional telephony can use PBX (Private Branch Exchanges) to provide local telephony services to an office or organisation, VoIP has spawned the introduction of IP PBXs to carry out the sale role. The IP PBX can have a trunk connection via a telecommunications network to allow VoIP calls to be routed across the existing telecoms networks. Gateways are used to interface the VoIP area of the network with the telecoms area of the network which can translate not only the signalling between different formats, but also media formats where the codec formats might differ.

VoIP has had an impact in the telecommunications world by being one factor in the reduction of telephone charges for consumers.

It is important for company and organisation telephony engineers, who are used to managing, troubleshooting and supporting traditional telephony in the workplace to receive training on VoIP and SIP when the company or organisation decides to switch to this maturing technology.

There are many training organisations offering VoIP training courses in the UK, and doing a search in Google or one of the many other search engines will produce many training provider websites offering Voice over IP and SIP training.

Perpetual Solutions training consultancy are based in London and have 9 different VoIP training courses for delivery as either Public Scheduled or available for onsite training worldwide. Some of their training courses are:

Understanding Voice over IP (2 days)

Hands-On Voice over IP (5 days)

Voice over IP using SIP (3 days)

Voice over IP Foundations (5 days

In the North of England, Network Systems Training (UK) Ltd have a number of VoIP training courses for delivery including:

Voice over IP Fundamentals (2 days)

VoIP with SIP (3 days)

Practical Voice over IP (5 days)

This article on VoIP and Voice over IP Training was written by David Christie, MD at NSTUK Ltd, Website http://www.nstuk.com . NSTUK Ltd offer a range of Data Networking Instructor-Led Training Courses including VoIP and SIP, and deliver those courses within the UK and throughout the World. Other training resources can be found at http://www.ipexpress.co.uk/info/Training