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There are essentially two basic forms of data transport available with IP networks, UDP
and TCP. With UDP the service does not guarantee delivery of data. Since packets are
never retransmitted the protocol will not add to the transit delay. With TCP the service is
reliable but delays can be introduced when packets in error are retransmitted. For these
reasons UDP and not TCP is used for VoIP data transport.

2.3.7 Erlang and network capacity
Voice networks use the Erlang as a standard measure of capacity. The Erlang is a measure
of total voice traf¬c in one hour, usually classi¬ed as the busy hour (BH), which is the
60-minute interval during a 24-hour period in which the traf¬c load is at a peak. One
Erlang is equivalent to one user talking for one hour on one telephone.
Consider that there are 45 calls in a one-hour period, and each call lasts for 3 minutes.
This equates to 135 minutes of calls. In hours, this is 135/60 = 2.25 Erlangs.
There are some variations in the Erlang model. The most common one is the Erlang
B, which is used to calculate how many lines are required to meet a given minimum
call blocking, usually 2“3%, during this BH. For cellular systems, it is used to estimate
capacity per cell at base stations. The Erlang B formula assumes that all calls that are
blocked are cleared immediately. This means that if a user attempts to connect and cannot,
they will not try again. An extended form of Erlang B factors in that a certain percentage
of users who are blocked will immediately try again. This is more applicable to the
cellular environment, since if blocked, many users will immediately hit the redial button.
The Erlang C model is the most complex since it assumes that a blocked call is placed
in a queue until the system can handle it. This model is useful for call centres.

2.3.8 Voice over IP (VoIP)
The use of IP to transport voice traf¬c is one of the most remarkable developments in
telecommunications in recent times. The development of the Internet as a global network
means that through the use of VoIP, the Internet (and intranets) can be developed into
a global telecommunications network. VoIP is a key enabler for the development of 3G
networks as the infrastructure moves to use IP packet switching exclusively.

Since there is already a very prominent abundance of circuit switched telecommunica-
tions networks available, one might ask what bene¬ts there are to be gained through the
use of VoIP:

• Lower transmission costs: due to economies of scale, and open and widespread com-
petition in the packet-switching market, the costs of transmission bandwidth have been
pushed extremely low.
• Data/voice integration: many corporations already have an extensive data communica-
tions infrastructure. By using this to transmit voice, phone and fax, operating costs can
be reduced. In particular, an organization which has a data communications network
spanning international boundaries can avoid costly long-distance tariffs.
• Flexible enhanced service: data sent over IP can be encrypted for security, redirected
to email voice mail services and routed via the Internet or PSTN. VoIP local area
networks are ideal for building such solutions as customer call centre systems.
• Bandwidth consolidation: packet switching uses bandwidth considerably more ef¬-
ciently than circuit switching. When there is no data to be sent, no bandwidth is used.
This is distinct from circuit switched networks, where the circuit is allocated the full
rate for the duration of the call.

There are also a number of problems associated with VoIP. The Internet itself is not
well suited to the transport of real-time sensitive traf¬c since it offers poor performance
in terms of delay and jitter. This is being addressed via a number of solutions to provide
quality of service. Effective Internet telephony protocols have only recently been in place
and equipment support is somewhat limited. With the development and widespread vendor
support for session initiation protocol (SIP), this problem is largely solved. Finally there
still remains a question mark over whether VoIP will still hold its cost/bene¬t advantage
now that enhanced service provider (ESP) status has been removed from ISPs in the
USA. This scheme had meant that ISPs were not required to pay access fees for telco
local access facilities, giving ISPs advantages in competing for voice customers. The
technical details of SIP are outlined in Chapter 9.
For VoIP, the delay must be minimal (telco standard minimum delay <100 ms) with
no variation in delay. However, bandwidth requirements are modest, depending on the
CODEC used, and unlike most data applications, some loss is acceptable but must be
under a certain threshold for the call quality to be acceptable.

2.3.9 Quality of service
Quality of service (QoS) relates to providing performance guarantees to those applications
that require it. Older packet switched protocols such as IP were originally intended to
support transport of data traf¬c, for which the best-effort model is suitable, where of
paramount importance is that data is delivered accurately and reliably, with delay and
delay variation of little importance. However, when packet switched networks are required
to transport real-time voice and video applications, the situation is much different and now

these mechanisms are required to provide guarantees of performance. As an example,
ATM technology builds this QoS mechanism in as a central aspect of the protocol. With
IP, the QoS solutions must be incorporated into the protocol suite. There are two basic
approaches to provide traf¬c with QoS: guaranteed QoS and class of service (CoS).
With guaranteed QoS, the network is expected to provide a minimum service de¬ned
by a set of quality parameters, including such things as minimum and average data rates,
maximum delay and jitter as well as maximum packet loss rate. This type of service
requires that the network has had some resources dedicated for the duration of the data
transfer. This resource allocation can be done statically so the resources remain allocated
even if the channel is not being used (as in the case of ATM permanent virtual circuits;
PVCs) or dynamically before each call is made. Within IP the protocol de¬ned which
provides guaranteed QoS called the resource reservation protocol (RSVP).
CoS, on the other hand, splits the traf¬c into priority groups. The network simply
guarantees to send high-priority traf¬c ¬rst, which works well provided the network has
been scaled carefully to carry the total required volume of traf¬c. The protocol which
provides CoS on IP networks is called DiffServ.
RSVP and DiffServ are presented in Chapter 5 while QoS in the context of ATM is
explained in Chapter 7.

In any communications system with many users, whether it be a ¬xed line or a wire-
less scheme, those users share some resource. Some mechanism must be employed to
enable this resource sharing, and this is referred to as a multiple access scheme. In the
wireless domain, the resource that is shared is frequency. For cellular communications,
a change in generation has generally meant a change in the multiple access scheme that
is implemented. The ¬rst generation of cellular systems used frequency division multiple
access (FDMA); the majority of second generation systems use time division multiple
access (TDMA) and most of the third generation schemes use code division multiple
access (CDMA). In addition, a shift has been made from the original analogue system to
a digital communications system.

As previously stated, a wireless system has the resource of frequency to share among
many users. The ¬rst approach to solving this problem is to split the available frequency
into a number of channels, each with a narrow slice of the frequency. This concept is
shown in Figure 2.7(a). Each user in the system that wishes to communicate is allocated a
frequency channel, and each channel has a certain gap, known as a guard band, between
it and the next channel so that the two do not interfere with each other. Once all the
channels are in use, a new user to the system must wait for a channel to become free
before communication can commence. Therefore, the system is limited in capacity as it
can only support as many simultaneous users as there are channels. This is known as a


channel 6

channel 5

channel 4
channel 3

channel 2
ch 1 ch 2 ch 3 ch 4 ch 5 ch 6
channel 1
time time
(a) (b)

Figure 2.7 Frequency division multiple access scheme

hard capacity system. Another problem is that if there is any external interference at a
particular frequency, then a whole channel may be blocked.
The concept of FDMA can be considered in the context of radio broadcasting. There
is a certain allocation of frequency resources, for example 88 MHz to 108 MHz for FM,
and each radio station in a particular region is given one channel within this on which
it transmits.


As wireless communications systems are expected to support more and more simultaneous
users, there are clearly severe limitations with the FDMA scheme. A more ef¬cient channel
usage is required. With TDMA, a frequency channel is divided up into a number of slices
of time, as shown in Figure 2.7(b). Here, a user is allocated a particular time slot, which
repeats periodically. In the diagram, the frequency is split into six time slots; a user
is allocated one slot in every six. Providing that the time slices are small enough and
occur frequently enough, a user is oblivious to the fact that they are only being allocated
a discrete, periodic amount of time. In this manner, the capacity can be dramatically
increased and hence the ef¬ciency of our system. Again, this is referred to as a hard
capacity type network.
As an example, the global system for mobile communications (GSM) employs both a
TDMA and FDMA approach. As with other mobile phone systems, an area to be covered
is split up into a number of cells, each of which is operating at a particular frequency
(frequency channel). Within a cell, the frequency being used is further split into time
slots using the TDMA principle. If more capacity is required, either more cells, packed
closer together, can be introduced, or another frequency channel can be deployed in a
cell, increasing the number of available time slots, and hence, the number of simultaneous
users. This does add some complication to the system, since the frequencies being used
must be carefully planned so no two frequencies that are the same may border each other.
This is the idea of frequency reuse; that is, a frequency can be used more than once in

the system as long as there is a suf¬cient distance between the repeated usage locations.
This idea is shown in Figure 2.8, where seven different frequencies, A, B, C, D, E, F and
G, are being reused.
Typically in rural areas these cells are of the order of 10 km across but in areas of high
usage (such as city centres) this may be reduced considerably to a few tens of metres.
Another advantage of the smaller cells is that less transmission power is required. This
in turn means that the battery of the mobile devices can be smaller and lighter, thus
reducing the overall weight of the devices. A single base station can control a number of
cells, with each cell using a different frequency. More effective coverage of a highway,
for example, can be attained through the use of sectored base stations as illustrated in
Figure 2.9. A sectored site is typically used to cover a larger geographical area. Note






Figure 2.8 Cellular frequency reuse

F1,F2 F3

F1 F2



Figure 2.9 Sectoring a base station for ef¬cient coverage

that in GSM the terms cell and sector are synonymous. A cell may also have more than
a single frequency allocated to it, as illustrated in Figure 2.9. A transceiver unit (TRX)
is the physical device located at the base station which controls each of these separate
frequencies. A cell having a number of frequencies will therefore have a number of TRXs.
In GSM, a TRX can handle at maximum eight full-rate simultaneous users.

If the previous multiple access schemes are considered in terms of ef¬ciency, each of them
involves only one user transmitting on a particular channel at a particular time, which is
clearly inef¬cient. For example, with GSM, in a given cell, only one user is transmitting
at any time; all other active users are waiting for their time slot to come around. If a
mechanism could allow more than one user to transmit at a time; then the resource usage
could be dramatically improved. CDMA is such a scheme, where all users are transmitting
at the same frequency at the same time. The effect of interference that users cause to each
other is discussed under the heading of noise. Having a system that is limited by a noise
target rather than speci¬cally allocating resources for the sole use of a particular mobile
device is known as a soft capacity system. Evidently, allowing multiple users to transmit
simultaneously is not the central issue; providing some system to separate them out again
is where the dif¬culty lies. This is the role of the codes. Much of the development
work associated with CDMA was accomplished by the eminent mathematician Andrew J.
Viterbi, who is also a cofounder of Qualcomm Inc. Thus, many of the patents associated
with CDMA are held by Qualcomm, which has resulted in considerable debate with
regard to the ownership of CDMA technology and much litigation against manufacturers


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