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
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