Technical papers | What is voice over IP?
What is Voice over IP?
Voice over IP, or Voice over Internet Protocol to use the full title, is simply a means of making
telephone calls over a data network instead of over the traditional analogue public switched
telephone network (PSTN).
The term VoIP describes the use of the Internet Protocol (IP) to transfer speech between two or
more sites. Inherent in the term is the management of the protocol. In general, this means that the
voice information is encoded into discrete digital packets and then transferred across an IP-based
network.
There are many advantages to this method of telephony, primarily the cost savings that can be
made by avoiding the use of the traditional PSTN. In addition to cost savings, the digital nature of
VoIP allows easy administration, the implementation of additional services such as voicemail, and
a reduction in the physical cabling required for new installations. However, due to the distributed
nature of the internet the Quality of Service (QoS) can often suffer, and there are still a number of
technical issues affecting the widespread adoption of VoIP.
Brief history
The term VoIP was first used by the VoIP Forum (a group of major companies including Cisco,
Vocalec and 3Com) to promote and develop the use of the International Telecommunications
Union’s (ITU) H.323 protocol. The forum also worked on the standardisation of directory services
and the use of touch-tone standards for accessing voicemail.
The ability to transfer voice over the internet, rather than the PSTN was first made possible in
February 1995 when Vocaltec released its Internet Phone software. This software was designed
to run on a standard personal computer (PC) equipped with a sound card, speakers, microphone,
and modem. The software encoded and compressed the voice signal, converting it into IP
packets that were then transmitted over the internet. However this PC-to-PC internet telephony
only worked if both parties were using the Internet Phone software. Inherent in this technology
were a number of severe limitations. For example, no formalised ringing protocols meant that it
was necessary for both parties to pre-arrange the time of the call in order to be available to make
the final connections themselves. Users also had to contend with the lack of any kind of directory
service, poor quality, and frequent delays.
Internet telephony has made a number of important advances since 1995. Many software
developers now offer PC telephony software but, more importantly, gateway servers are emerging
to act as an interface between the internet and the PSTN. Equipped with voice-processing cards,
these gateway servers enable users to communicate via standard telephones over great
distances without using the ‘long distance’ telephone network.
How does VoIP work?
When a traditional call is made using the PSTN, the analogue lines are kept open between the
two callers for the entire duration of the call; this is called a circuit. This small segment of the
network is exclusively used for this call and will be unavailable to any other users until the call has
finished. VoIP does not require such dedicated circuits as it translates voice signals into packets
of digital data. These packets are then transmitted over Ethernet or wireless networks, with no
part of the network used exclusively by callers.
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Valid at September 2004
Review at December 2004
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Becta | Technical paper | What is voice over IP?
VoIP requires the use of Codecs (COder-DECoders). These can be software, or hardware such
as microphones, IP telephones or other similar devices. They are required to convert analogue
signals (what we say and hear) into digital (for transmission over the network) and back into
analogue.
Figure 1 (below) shows how schools can combine their existing voice and data networks to carry
both voice and data traffic. In the example shown, the LEA provides schools A and B with a
broadband connection that is used to carry both data and VoIP traffic. The schools incur no call
costs when calling the LEA or each other as all calls are passed along existing connections.
School C has not yet converted to VoIP and therefore runs two separate networks, one for voice
and one for data. The LEA also has a VoIP gateway server that allows it to send and receive calls
on the PSTN. Should school A or B wish to call outside the VoIP service area the calls pass
through the gateway and out over the PSTN. Similarly, if an external site wishes to call school A
or B the calls are routed into the VoIP gateway at the LEA and over the network to the school.
Figure 1 – a typical H.323 implementation of VoIP with other networks
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Becta | Technical paper | What is voice over IP?
If part of the network uses the PSTN, then after a call is initiated it goes over the local PSTN to
the nearest gateway server. This digitises the analogue voice signal, compresses it into IP
packets, and moves it onto the IP network for transport to the gateway server at the receiving end.
This server converts the digital IP signal back to analogue and completes the call via the local
PSTN.
Figure 2 – the relationship between IP and PSTN systems
Figure 2 shows the relationship between VoIP, PSTN, the internet and the gateways. It is possible
to route calls using VoIP over existing infrastructure as long as the necessary gateways are in
place.
Today, VoIP networks operate by translating telephone numbers to IP addresses and placing an
H.323 or Session Initiation Protocol (SIP) call to the device. Should the route between the callers
involve all IP networks with no parts PSTN, then no gateways are required to translate the digital
data into analogue.
The easiest way to start using VoIP is to install a software client onto several networked PCs.
These clients are common and there is an abundance of software
[http://www.iptelephony.org/GIP/vendors/client-phones/] that allows people to talk to each other
providing they have a network connection, microphone and speakers. Person A can call person B
simply by typing the IP address of person B’s PC into the software. The voice data between the
two PCs is carried over the network along with normal data traffic. This is a relatively simple and
cheap way for people to communicate using an existing network infrastructure.
The next step up from this basic software implementation is to use IP telephones. These devices
look similar to standard telephones and plug straight into the existing data network. They contain
all the software and equipment needed to establish and receive VoIP calls; they behave as a
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Review at December 2004
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Becta | Technical paper | What is voice over IP?
normal telephone does, except that they transmit and receive data over computer networks and
not the PSTN. Such devices are normally completely independent and are not centrally controlled
by a public branch exchange (PBX) or server that would control a normal analogue system.
Larger, more complex VoIP networks include many IP telephones linked to one central server that
controls functions such as voicemail, extension numbers and caller groups. These larger, wide
area networks (WANs) often cover many geographically dispersed sites. Such networks use
hardware ‘bridges’ to transfer data between sites in the WAN and these bridges can also convey
VoIP traffic.
Where is VoIP being used?
The main application of VoIP is currently within intranets. As voice communications are
dependent upon continuous time-based audio (speech), it is imperative that the bandwidth across
which the IP packets are transferred is stable and that packet loss is kept to an absolute
minimum. Any packet loss during a VoIP call will result in the speech being transferred sounding
‘clipped’ and/or unintelligible. In addition, delays to the routing of the packets will cause
momentary pauses and delays in the received speech, reducing the quality of the call.
Consequently local conditions can adversely affect VoIP calls, especially when they are routed
across the internet. Therefore VoIP is best suited to use across local area networks (LANs) and
wide area networks (WANs) where routing via the internet is not required.
VoIP in practice
Australia’s Academic and Research Network (AARNet)
[http://www.aarnet.edu.au/services/voip/index.html] has implemented VoIP telephony throughout
its 37 member organisations. As long distance telephony is common in Australia, VoIP presents
significant cost savings – up to 70 per cent over PSTN solutions.
The advantages and disadvantages of VoIP
VoIP offers a number of potential advantages and benefits to its users:

reduced operating cost, including lower telecom charges

flexibility

reduced infrastructure

integrated services and greater user features.
The major advantage of VoIP is cost saving (although do not expect immediate cost savings due
to the upgrading of existing infrastructure and systems implementation). In addition to the financial
savings, there are a number of additional benefits; it is innately scalable, allowing the quick and
easy addition of new terminals and connections. The voice data can be easily processed using a
standard PC, allowing call monitoring, voicemail, and a range of other features to be
implemented.
A single network topology can be put in place, requiring only network cabling to be installed in a
building rather than both data and traditional telephone cabling. VoIP systems can also be
configured using standard networking tools such as SNMP (simple network management
protocol). A side benefit of VoIP is that many of the steps to full implementation are the same as
the steps required to implement video conferencing and similar technologies. Whilst VoIP is not
strictly speaking a subset of video conferencing there are many similarities, and a large number of
products on the market implement both technologies in a single package.
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Valid at September 2004
Review at December 2004
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Becta | Technical paper | What is voice over IP?
However, there are drawbacks:

the network has to be fast, reliable and offer high quality of service

compatibility with existing firewalls and security devices may cause problems.
(Firewalls need to be H.323 and possibly SIP compliant depending on the VoIP
solution.)
Are there any VoIP standards?
IP packets are unreliable and use various re-transmission techniques, hence the need for
standards to achieve an acceptable level of service. When considering implementing VoIP there
are a number of standards that decision makers should examine.

H.323 is the protocol ratified in 1996 by the ITU. As previously discussed, this has
become the protocol of choice for the transfer of voice information across the
internet. The H.323 standard has also become somewhat of an ‘umbrella’ for a
number of other protocols that add extra facilities.

SIP is becoming a more attractive option even though H.323 is the most
established protocol. SIPs ‘lighter’ system use in connection and transfer standards
means that a number of VoIP system designers are opting to use SIP. The
compatibility of SIP devices makes the implementation of this protocol an easier
proposition, allowing IT staff to choose equipment from a number of separate
manufacturers and still expect that it will be compatible with existing equipment.
As these two standards are quite different, interoperability between H.323 and SIP is only possible
via a mediator or gateway.
What should be considered when purchasing a VoIP solution?

Here are some, but by no means all, of the considerations you should discuss with
your supplier when they recommend a VoIP solution. Choose a VoIP supplier who
will asses your current structure and decide on the best solution for your needs.
This may possibly be a part-PSTN/part-VoIP solution. Asses the extra features that
come with VoIP such as phone grouping, instant messaging, instant messaging
(IM) and answerphone services.

Ensure that your network can handle the voice traffic load. Look at your existing
call usage such as volume, peek usage times and duration of calls. You will need
this information when deciding if your current network is able to cope with the
expected rise in bandwidth requirements.

Establish a test procedure. Check for voice quality, gaps in conversations, latency,
echo and consistency. Make calls internally and externally at various times of the
day to ensure the network can cope even at its busiest times.

Obtain high quality, preferably hardware, Codecs. These will play the most crucial
part in how the VoIP ‘feels’ to its users. Generally, the higher quality the Codec the
better the voice quality.

Administration of VoIP calls should be included, and call logs should be produced
automatically in an easy-to-read format. More detailed logging should also be
available.

Training should be comprehensive and thorough for both administrators and users.
What type of equipment will be needed?
The H.323 standard specifies four types of device that together form the core of a VoIP network:
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
Terminals/IP telephones are the real-time bi-directional devices capable of
broadcasting and receiving H.323 traffic.

Gateways perform a translation role, converting H.323 traffic to other protocols and
vice-versa. Thus a gateway is often employed to provide communication between a
H.323 terminal and the PSTN.

Gatekeepers provide additional services to a VoIP system such as address
controlling, authentication, billing, accounting and bandwidth management.

Multipoint control units (MCUs) provide connectivity for three or more terminals
allowing conference calling.
Whilst gateways, gatekeepers and MCUs are logically separate components of the H.323
standard they are often implemented in a single physical unit.
Fully functional IP telephones are available, as are hardware ‘gateways’ which usually perform
gateway, gatekeeper and MCU functions. Software solutions are also available for the majority of
functions, from simple internet-capable ‘soft terminals’ to sophisticated software gateways,
gatekeepers and MCUs.
SIP implementations require a substantially different array of equipment to H.323:

A SIP-compliant end device, such as a SIP telephone, is required and this can be
hardware or software.

A SIP registrar/location server accepts registration requests from end points and
maintains a whereabouts list of all its registered users.

SIP proxy servers stand between the end points and the location server. These
allow calls to be redirected to end points and to temporary contact addresses.
In order for VoIP to perform at acceptable levels, the network bandwidth must be great enough to
cope with the number of calls anticipated. Whilst the load of voice traffic is not high enough to
cause concern for the majority of LANs, care must be applied to bandwidth calculations for
distributed network use of VoIP. As all voice communication is converted to IP traffic via the VoIP
protocols and hardware, there is no need for network architecture to be altered other than to
accommodate VoIP equipment.
The future
As the take up of broadband increases and IP networks become more reliable, the use of VoIP
will grow. The normal course of replacing existing analogue communications with digital solutions
will continue, and the integration of the two separate networks into one will become more
common. However, some major obstacles will have to be overcome if VoIP is to succeed as a
complete replacement to PSTN.

Quality of service – the expectations in the communications market are that voice
communications should have a 100 per cent uptime and that voice quality should
be almost perfect.

Security – it is possible to tap almost any form of communication be it analogue or
digital, and VoIP manufactures will have to ensure their solutions are secure from
all but the most sophisticated attacks.

Charging – at the moment providers using the PSTN charge by distance called, but
with VoIP the model may change to one where telecommunication providers can
start reserving bandwidth and charge users accordingly.
The existing telecommunications providers may start replacing the national analogue networks
with digital equipment and this may lead to a gradual replacement of equipment inside homes and
schools. So-called ‘soft switches’ are currently in development, these switch voice traffic using
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Review at December 2004
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Becta | Technical paper | What is voice over IP?
software allowing all the old functionality found in a large exchange to be achieved whilst reducing
overall costs and increasing control, billing and other services.
Other sources of information
The International Engineering Consortium (IEC) VoIP Tutorial (PDF)
[http://www.conferzone.com/resource/wp/VOIP_whitepaper.pdf]
VoIP Watch — news site
[http://www.voipwatch.com/index.php3]
Voice over IP and IP telephony: reference links
[http://www.cis.ohio-state.edu/~jain/refs/ref_voip.htm]
The International Telecommunications Union
[http://www.itu.int/home/index.html]
The SIP Centre
[http://www.sipcenter.com/]
Palomar College VoIP Project Journal
[http://www.palomar.edu/nibblesandbits/PC_VoIP_Project.htm]
© Becta 2004
Valid at September 2004
Review at December 2004
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