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LTE Technology for Engineers

Training Guide
K025

The information in this document is subject to change without notice and describes
only the product defined in the introduction of this documentation. This document is
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The information or statements given in this document concerning the suitability,
capacity, or performance of the mentioned hardware or software products cannot be
considered binding but shall be defined in the agreement made between AIRCOM
International and the customer. However, AIRCOM International has made all
reasonable efforts to ensure that the instructions contained in the document are
adequate and free of material errors and omissions. AIRCOM International will, if
necessary, explain issues, which may not be covered by the document.
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Other product names mentioned in this document may be trademarks of their
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Copyright AIRCOM International 2010. All rights reserved.

Contents
1

Introduction to LTE

1.1

Where are we?

1.2

Release 99

11

1.3

UTRAN

15

1.4

3G Services and QoS Classes

21

1.5

HS-PDSCH

22

1.6

HSDPA

23

1.7

HSUPA

24

1.8

HSPA+

Error! Bookmark not defined.

1.9

Upgrade Paths to LTE

28

1.10 R8 LTE

29

1.11 Scalability of Bandwidth

33

1.12 LTE Key Features

34

IP Core Network Overview

41

2.1

The TCP/IP Layers

41

2.2

Transport Layer Protocols

43

2.3

Application Layer Services

47

2.4

TCP/IP Inter-networks

52

2.5

IP Datagram Format

53

2.6

Maximum Transfer Unit (MTU)

55

2.7

Fragmentation

56

2.8

Time To Live (TTL)

57

2.9

IP Addresses

59

2.10 Dotted Decimal Notation

60

2.11 Address Classes

61

2.12 Special IP Addresses

62

2.13 Realtime Transport Protocol (RTP)

63

2.14 Base Header Format

66

2.15 The Need for QoS

72

2.16 IP Precedence

73

2.17 Type of Service

75

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2.18 Differentiated Services


2.18.1
2.18.2
2.18.3

81

Assured Forwarding
Weighted Fair Queing
RNC Scheduling

86
88
89

2.19 Questions

Layer 2 Switching91
3.1

Introduction

91

3.2

Spectral Efficiency

93

3.3

The TCP/IP Layers

95

3.4

Message Switching

97

3.5

Frames

98

3.6

Hardware Addresses

101

3.7

Store & Forward

102

3.7.1
3.7.2

Cut Through Switching


Modified Cut-Through

103
104

3.8

Switches and VLANs

105

3.9

Link Aggregation

109

3.10 Carrier Ethernet Overview

114

3.11 The Three Essential Functions of Connection-Oriented Ethernet

118

3.12 Colour Marking

124

3.13 Traffic Policing and Shaping

125

3.14 MEF Bandwidth Profiles

126

3.15 Mapping at the UNIs

128

3.16 Class of Service (CoS)

130

3.17 IEEE 802.1Q Virtual LANS (VLANS)

131

3.18 Carrier Ethernet Services

132

3.19 Questions

136

LTE Basic Air Interface 137


4.1

New Air Interface

137

4.2

OFDMA

138

4.3

Advanced Antenna Techniques

139

4.4

Cyclic Delay Diversity

143

4.5

FDD/TDD

144

4.5.1
4.5.2
4.5.3
4.5.4

4.6

FDD
TDD
LTE TDD / TD-LTE Subframe Allocations
Flexible Carrier Bandwidths

145
146
149
150

Slot Structure and Physical Resources

4.6.1
Page 6

90

152

Transmission Bandwidths

155
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4.7

Orthagonality

4.7.1
4.7.2

156

Single-Frequency Network Multicast Services


Single Carrier Frequency Division Multiple Access (SC-FDMA)

4.8

Cyclic Prefix

162

4.9

Delay Spread

163

4.10 Slot Structure and Physical Resources

166

4.11 Scheduler

167

4.12 ASSET - LTE

172

4.13 Downlink Data Transmission

173

4.14 Modulation and Subcarriers

175

4.15 Downlink Reference Signal Structure

177

4.15.1

Configuration of Carrier

178

4.16 Reference Signal Received Power (RSRP)

181

4.17 Channel Quality Indicator Reporting

183

4.18 Downlink Shared Channel (DL-SCH)

187

4.18.1
4.18.2

Physical Cell Identity (PCI)


Physical Downlink Control Channel

188
189

4.19 Questions

LTE Network Architecture and Protocols

190

191

5.1

LTE Architecture

192

5.2

Roaming Architecture

195

5.3

Bearer Establishment Procedure

199

5.4

KPI-RAB Success

204

5.5

KPI Dropped Call Ratio ?

205

5.6

The Home Subscriber Service

210

5.7

PDN Gateway

212

5.8

Network Sharing

217

5.9

Session Initiation Protocol Architecture

219

5.10 LTE Functional Nodes

224

5.11 Physical Channels, Transport Channels & Logical Channels

227

5.11.1
5.11.2
5.11.3

Logical Channels
Transport Channels
Physical Channels

5.12 Modulation and Coding


5.12.1
5.12.2

PDSCH
PUSCH

5.13 Functional Nodes


5.13.1
5.13.2

Functional Nodes - UE
Functional Nodes - eNodeB

5.14 RLC Modes


5.14.1

RLC Modes - Qos

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160

228
231
235

238
239
241

242
243
244

245
246
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5.15 Medium Access Control (MAC)

247

5.16 Physical Control Channel

251

5.16.1

Physical Downlink Control Channel

252

5.17 Multimedia Broadcast/Multicast Service (MBMS)

257

5.18 Reference Signal Received Power (RSRP)

264

5.19 Quality Channel Indicator Reporting

266

Mobility Management
6.1

UE States

271

6.2

UE Power-up

273

6.2.1
6.2.2

EPS Mobility Management


Tracking Area Update

275
277

6.3

LTE Functional Modes - MME

279

6.4

RRC States

282

6.5

Reference Signal Received Power (RSRP)

295

6.6

Cell Re-selection

299

6.7

Handover RRC Connected

302

6.7.1
6.7.2
6.7.3
6.7.4
6.7.5
6.7.6

6.8

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271

LTE Reference Signal Received Quality (RSRQ)


Received Signal Strength Indicator (RSSI)
Reference Signal Received Power (RSRP)
Handover
Automatic Neighbour Relations
Measurement Configuration

306
307
308
309
315
322

Questions

324

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CHAPTER 1

Introduction to LTE
1.1 Where are we?

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1.2 Release 99

UMTS/W-CDMA was initially conceived as a circuit switched based system and was
not well-suited to IP packet based data traffic. Once the basic UMTS system was
released and deployed, the need for better packet data capability became clear,
especially with the rapidly increasing trend towards Internet style packet data
services which are particularly bursty in nature. It supports Cell-DCH and typical
speeds 384kb/s.
Release 5: This release included the core of HSDPA itself. It provided for downlink
packet support, reduced delays, a raw data rate (i.e. including payload, protocols,
error correction, etc.) of 14 Mbps and gave an overall increase of around three times
over the 3GPP UMTS Release 99 standard.
Release 6: This included the core of HSUPA with an enhanced uplink with improved
packet data support. This provided reduced delays, an uplink raw data rate of 5.74
Mbps and it gave an increased capacity of around twice that offered by the original
Release 99 UMTS standard. Also included within this release was the Multimedia
Broadcast Multicast Services (MBMS), providing improved broadcast services such as
Mobile TV.
Release 7: This release of the 3GPP standard included downlink MIMO operation as
well as support for higher order modulation up to 64 QAM in the uplink and 16 QAM
in the downlink. However, it only allows for either MIMO or higher order
modulation. It also introduced protocol enhancements to allow support for
Continuous Packet Connectivity (CPC).
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Release 8: This release of the standard defines dual carrier operation as well as
allowing simultaneous operation of the high order modulation schemes and MIMO.
Further to this, latency is improved to keep it in line with the requirements for many
new applications being used.

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Rb_phy includes DPDCH (User data + L3 control) + Error protection + DPCCH (L1
control)
DPCCH = Dedicated Physical Control Channel
In UL, symbol rate=ch bit rate.
The DPDCH channel bit rate is less than channel bit rate because the latter contains
both DPDCH and DPCCH ch bit rates. The exact DPDCH bit rate depends on the slot
format. DPDCH is shared by logical/transport channels (DCCH/DCH +
DTCH/DCH).
The exact DTCH bit rate depends on the selected channel configuration or transport
format, for example with AMR 12.2, the DTCH is 12.2 kbit/s and DCCH is 3.7 kbit/s
by default (SF = 128).
For the channel coding, three options are supported: convolutional coding, turbo
coding, or no channel coding. Channel coding selection is indicated by upper layers.
For example, with single DPDCH in UL:
960kbps can be obtained with SF=4, no coding
400-500 kbps with coding
With 3 codes, up to 5740 kbps uncoded or 2Mbps (or even more) with coding.
Error Correction Coding Parameters
Transport channel type

Coding scheme

Coding rate

BCH, PCH, RACH

Convolutional code

1/2

CPCH, DCH, DSCH, FACH

Convolutional code

1/3, 1/2

CPCH, DCH, DSCH, FACH

Turbo code

1/3

CPCH, DCH, DSCH, FACH

No coding

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1.3 UTRAN

In UMTS, the UTRAN is formed from RNCs, Node Bs and their defined interfaces. An
RNC is the UMTS equivalent of a GSM BSC, but has greater functionality. One of the
main differences between UMTS (Release 99) and GSM (Release 99) is that there is an
Iur interface interconnecting multiple RNCs. This additional interface permits the
RNCs to communicate with each other, allowing soft handover (not present in GSM).
Another difference between GSM and UMTS is that some of the Mobility
Management (MM) has been moved to the RNC from the Core Network, allowing
UTRAN initiated paging. The RNC has greater control over a group of cells and the
functionality allows soft handover.

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Core Network
The Core Network is divided in circuit switched and packet switched domains. Some
of the circuit switched elements are Mobile services Switching Centre (MSC), Visitor
location register (VLR) and Gateway MSC. Packet switched elements are Serving
GPRS Support Node (SGSN) and Gateway GPRS Support Node (GGSN). Some
network elements, like EIR, HLR, VLR and AUC are shared by both domains.
The functions of RNC are:

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Radio Resource Control

Admission Control

Channel Allocation

Load Control

Handover Control

Macro Diversity

Ciphering

Segmentation / Reassembly

Broadcast Signalling

Open Loop Power Control

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Co-existence with legacy standards and systems: LTE users should be able to make
voice calls from their terminal and have access to basic data services even when they
are in areas without LTE coverage. LTE therefore allows smooth, seamless service
handover in areas of HSPA, WCDMA or GSM/GPRS/EDGE coverage. Furthermore,
LTE/SAE supports not only intra-system and intersystem handovers, but interdomain handovers between packet switched and circuit switched sessions.

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1.4 3G Services and QoS Classes

In UMTS, four Quality of Service classes have been defined:


Conversational class is the QoS class for delay-sensitive real time services such as
speech telephony.
Streaming class is also regarded as a real-time QoS class. It is also sensitive to delays; it
carries traffic, which looks real time to a human user. An application for streaming
class QoS is audio streaming, where music files are downloaded to the receiver. There
may be an interruption in the transmission, which is not relevant for the user of the
application, as long as there is still enough data left in the buffer of the receiving
equipment for seamless application provision.
Interactive class is a non-real time QoS class. It is used for applications with limited
delay-sensitivity (so-called interactive applications). But many applications on the
internet still have timing constraints, such as http, ftp, telnet, and smtp. A response to
a request is expected within a specific period of time. This is the QoS offered by the
interactive class.
Background class is a non-real time QoS class for background applications, which are
not delay sensitive. Example applications are email and file downloading.

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1.5 HS-PDSCH

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1.6 HSDPA

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1.7 HSUPA

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[CL] From one phase to another, the main limitation is the product development which resources you can offer to the user.
TTI - 2 ms (HS-DSCH), 10 ms, 20 ms, 40 ms, and 80 ms
TTI is length of transmission on the radio link

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1.8 Upgrade Paths to LTE

Who Needs LTE?


Less than a decade on from the launch of the first 3G/UMTS networks, why is the
cellular industry considering additional investments in its radio access and core
network infrastructures?
The answer lies in a changing market landscape, where user expectations are
constantly increasing. In the fixed world, broadband connectivity is now ubiquitous
with multi-megabit speeds available at reasonable cost to customers and business
users via DSL and cable connections.

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1.9 R8 LTE

The result of these radio interface features is significantly improved radio


performance, yielding up to five times the average throughput of HSPA. Downlink
peak data rates are extended up to a theoretical maximum of 300 Mbit/s per 20 MHz
of spectrum. Similarly, LTE theoretical uplink rates can reach 75 Mbit/s per 20 MHz
of spectrum, with theoretical support for at least 200 active users per cell in 5 MHz.
Reduced latency: By reducing round-trip times to 10ms or even less (compared with
4050ms for HSPA), LTE delivers a more responsive user experience. This permits
interactive, real-time services such as high-quality audio/videoconferencing and
multi-player gaming.

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1.10 Scalability of Bandwidth

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1.11 LTE Key Features

Urban areas:

Most likely LTE will be deployed.

Stepwise deployment in UMTS 2.1 bands will be possible at a later stage.

Rural areas:
Option 1: deploy UMTS in 900 MHz band.
Advantage: rollout can start now
Disadvantage: a block of 5 MHz need to be taken out of the GSM band. Not a lot of
operators can affort to take out this much spectrum due to heavy usage in this band
Option 2: Introduce LTE in 900 MHz band
Advantages: reuse of GSM 900 Sites; step by step introduction of LTE with smaller
granularity (1.4 / 3 / 5 /MHz).

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CHAPTER 2

IP Core Network
Overview
2.1 The TCP/IP Layers

TCP/IP can be represented by the US DoD Model. This model describes the
relationship between the main protocols used by TCP/IP.

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Prior to the development of this model most network protocols were vendor
dependent. The architecture behind TCP/IP is different in the sense that the same
protocol model can be run on a multitude of different computer systems without
modification of the operating system or hardware architecture. TCP/IP is designed to
run as an application.
The protocol was primarily used to support application-orientated functions and
process-to-process communications between hosts. Specific applications to provide
basic network services for users were written to run with TCP/IP. The objective of the
lower protocols was to provide support for the network layer application services.

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2.2 Transport Layer Protocols

Transport layer protocols provide two basic functions to the application layer services
- quality of service and application multiplexing through port numbers. TCP/IP has
two main transport layer protocols TCP and UDP.
UDP provides a simple datagram delivery service adding application multiplexing
and a checksum to the underlying IP layer. It therefore provides the same unreliable,
connectionless delivery service as IP. It does not use acknowledgements to confirm
that messages have arrived, it does not provide any flow control mechanisms, and it
does no sequencing - UDP messages can be duplicated, arrive out of order or not at
all. UDP works well on LANs where error rates are low and delays small, but on
WANs it behaves poorly, especially for large data transfers.
TCP provides a reliable, connection-oriented, stream based delivery system by adding
acknowledgements, sequencing and flow control to IP. This makes TCP much more
efficient on WANs and for large data transfers, but it has a large protocol overhead
which makes it slower and less efficient than UDP in certain applications.
Most applications tend to use TCP because it provides reliable delivery, but timesensitive, transactional and broadcast based applications need to use UDP.

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2.3 Application Layer Services

The power behind TCP/IP is not the sophisticated and powerful nature of the
protocol architecture, but rather it is the absolute simplicity of the protocol. This is
equally true of the application-level services which are designed to provide network
services for users.
TCP/IP provides a consistent application front end to users regardless of the
operating system, platform, or network architecture which is used. Many of the
application level services retain the look and feel of simple character-oriented
applications. Even today, with TCP/IP providing GUI, once the superficial GUI is
removed, the same basic element of code is used to provide the network service.
The Transport layer protocols (TCP/UDP) use Port Numbers to uniquely identify
each application level service. The client usually generates a port number above 1,023
to identify the process and the server always uses a well known port number.

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2.4 TCP/IP Inter-networks

The term internetworking is used to describe a number of discrete physical networks


that are connected together to form an internet. A characteristic of such an internet is
that the underlying physical network structure should be invisible to network users.
Internetworking is defined as a combination of interconnection and interoperation
(the ability to physically exchange data and make some sense from it).

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2.5 IP Datagram Format

VERS

Protocol version (currently 4)

HLEN

Length of header in 32 bit words (normally 5)

Service Type

Sets a precedence and Type of Service for the packet (normally 0)

Total Length
Length of IP datagram in octets including header & data Maximum of 65535
Identification

Unique ID for each datagram, used for fragmentation

Flags
fragments)

Controls fragmentation (DF - don't fragment and MF - more

Fragment Offset Position of data in this fragment compared to original datagram units of 8 octets
Time To Live
internet.

Specifies how long (in router hops) the datagram is to remain in the

Protocol ID of transport protocol - UDP, TCP, (ICMP) etc.


Checksum

Checksum of the header only

Source IP Address

32 bit IP address of source

Destination IP Address 32 bit IP address of destination


IP Options
Option type and data for additional facilities - network
management and debugging
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Padding

Padding to extend options data to multiple of 4 octets

DATA

The higher level Protocol Data Unit (PDU)

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2.6 Maximum Transfer Unit (MTU)

Each physical network has a defined limit on the size of protocol which it can
support. This is known as the Maximum Transfer Unit (MTU) and is generally
considered a hard limit of the network which cannot be increased. The MTU is the
maximum size of software protocol which can be sent, and not the maximum size of
frame which can be supported. If hardware control information (such as physical
addresses) is added to the MTU, the maximum frame size can be derived.
Ethernet limits transfers to 1500 bytes of data, which FDDI permits approximately
4470 bytes of data per frame. MTUs vary considerably in size. Local area networks,
which generally use high bandwidth, low bit error rate media have relatively large
MTUs, while wide area networks have much smaller MTUs. Limiting datagram size
to fit the smallest possible MTU in the internet makes transfers inefficient when those
datagrams pass across a network which can carry larger size frames. However,
allowing datagrams to be larger than the minimum network MTU in an internet
means that a datagram may not always fit into a single network frame.
Instead of making IP datagrams adhere to the constraints of physical networks,
TCP/IP software chooses a convenient initial datagram size and arranges a way to
divide large datagrams into smaller pieces when the datagram needs to traverse a
network that has a small MTU. The small pieces into which a datagram is divided are
called fragments, and the process of dividing a datagram is known as fragmentation.

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2.7 Fragmentation

Hosts can choose to send datagrams up to the supported MTU of their own network.
Routers interconnect different physical networks with varying MTUs. Routers can
fragment datagrams if desired to permit transport across networks which have
smaller MTUs.
The IP protocol does not limit datagram size nor does it guarantee that the datagram
will be delivered without fragmentation. The source can choose any datagram size it
thinks appropriate; fragmentation and reassembly occur automatically, without
taking action. The IP specification states that routers must accept datagrams up to the
maximum of the MTUs of the networks to which they are attached.
Once a datagram has been fragmented, the fragments travel all the way to the final
destination, where they reassembled. This has a number of disadvantages - small
fragments must be carried by networks which could support larger MTUs, and
reassembling the datagrams at the destination can lead to inefficiency, particularly if
fragments are lost. If fragments are lost, the original datagram cannot be reassembled.
The receiving machine starts a reassembly timer when it receives an initial fragment.
If the timer expires before all fragments arrive, the receiving machine discards the
surviving pieces without processing the datagram. Performing reassembly at the
ultimate destination works well, and permits each fragment to be routed
independently as well as sparing resources on routers.

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2.8 Time To Live (TTL)

The TIME To LIVE specifies how long, in seconds, the datagram is permitted to
remain in the internet. Whenever a host injects a datagram into the internet, it sets a
maximum time that the datagram should survive. Router and hosts that process
datagrams must decrement the TIME To LIVE field as time passes and remove the
datagram from the internet when the value in this field reaches zero.
Estimating exact time is difficult because routers do not usually know the transit time
of physical networks. A few rules simplify processing and make it easy to handle
datagrams without synchronise clocks. First, each router along the path from source
to destination is required to decrement the TIME To LIVE field when the datagram
header is processed. Furthermore, to handle cases of overloaded routers that
introduce long delays, each router records the local time when the datagram arrives
and decrements the TIME To LIVE by the number of seconds that the datagram
remained inside the router waiting for service.

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2.9 IP Addresses

TCP/IP uses a 32-bit binary address to uniquely identify a device on a TCP/IP


internetwork. The binary address string is a network layer logical address which must
be configured by the network manager. The address is used to identify the device in a
virtual network.
The 32-bit address structure is divided into a single-level hierarchy where the leading
bits in the address are used to describe a network in logical terms and the remaining
bits are used to describe the host on the logical network. The number of bits which are
used in each case varies, and will be covered later. The leading bits which make up
the logical network address are used to provide a routing (packet forwarding)
mechanism between logical networks. This allows for far more efficient routing than a
flat address space.

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2.10 Dotted Decimal Notation

Binary address strings are very difficult to work with. To overcome this problem and
make logical addressing easier to comprehend, the 32-bit address string is divided
into 8-bit bytes and then converted into the corresponding decimal notation. It is this
dotted decimal notation which is used to configure hosts on a TCP/IP network.
However, it should be noted that decimal addresses are a human and humane
interface to TCP/IP. As far as the host is concerned, the address appears and is used
as a binary string. This is the cause of much confusion.

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2.11 Address Classes

There are five main classes of IP addresses but only three are directly usable: A, B and
C.
For a Class A address, 8-bits are used to logically identify the network. For Class B,
16-bits are used, and for Class C, 24-bits are used. In each case, once the network bits
have been allocated, the remaining bits are used to logically identify the node.
Class E addresses are reserved for testing and development by the IETF and cannot be
assigned to any device. Class D addresses are software multicast addresses and
reserved for the use of routing protocols such as OSPF, RIPv2 and so on.
The address categorisation is derived from the high bit order rule of the first byte. The
high bit order rule is interrupted by every TCP/IP stack as soon as an address is
entered. This rule is also used to define the decimal ranges in the first byte of each
address.

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2.12 Special IP Addresses

Some binary bit patterns are reserved for management reasons and cannot be
allocated to devices on a TCP/IP network.
Although the Internet Protocol has been stable for a considerable number of years, the
way IP addresses are used and interpreted has changed over the years.
In general, 1s indicate "All" and 0s indicate "Any" - the local broadcast address is an
obvious exception; a broadcast to all hosts on all networks (on the Internet) would
cause chaos!

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2.13 Realtime Transport Protocol (RTP)

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2.14 Base Header Format

Although the IPv6 header must accommodate larger addresses, an IPv6 base header
contains less information than an IPv4 header. Options and some of the fixed fields
that appear in an IPv4 header have been moved to extension headers. Changes in the
datagram header reflect changes in the protocol:

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Alignment has been changed from 32-bit multiple to 64-bit multiples.

The header length field has been eliminated, and the datagram length field has
been replaced by a Payload

Length field.

The size of the source and destination address fields has been increased to 16 bytes
each.

Fragmentation information has been moved out of fixed fields in the base header
into an extension header.

The Time-to-Live field has been replaced by a Hop Limit field.

The Service Type field has been replaced by a Flow Label field.

The Protocol field has been replaced by a field that specifies the type of the next
header.

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IPv6 handles packet length specification in a new way. Firstly, because the size of the
base header is fixed at 40 bytes, the header does not include a field for the header
length. Secondly, IPv6 replaces IPv4 packet length field with a 16-bit Payload Length
field that specifies the number of octets carried in the packet excluding the header. An
IPv6 packet can contain 64k bytes of data.

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The fixed header occupies the first 40 octets (320 bits) of the IPv6 packet. It contains
the source and destination addresses, traffic classification options, a hop counter, and
a pointer for extension headers if any. The Next Header field, present in each extension
as well, points to the next element in the chain of extensions

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2.15 The Need for QoS

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2.16 IP Precedence

IP Precedence was designed in IPv4 by the IETF.


It uses 3 bits of the 8-bit Type of Service (TOS) field of an IP header. There are 8
classes of services in IP Precedence. The classification range is 0-7 where 0 (zero) is the
lowest and 7 is the highest priority.
The original intention of the TOS field was for a sending host to specify a preference
for how the datagram would be handled as it made its way through an internet. For
instance, one host could set its IPv4 datagrams' TOS field value to prefer low delay,
while another might prefer high reliability.

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IP Precedence provides the ability to classify network packets at Layer 3. With IP


Precedence configured, network packets traverse IP Precedence devices according to
the priority you set. Priority traffic is always serviced before traditional traffic.

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2.17 Type of Service

The Type of Service field in the IP header was originally defined in RFC 791.
It defined a mechanism for assigning a priority to each IP packet as well as a
mechanism to request specific treatment such as high throughput, high reliability or
low latency.
Differentiated Services Code Point
In RFC 2474 the definition of this entire field was changed. It is now called the "DS"
(Differentiated Services) field and the upper 6 bits contain a value called the "DSCP"
(Differentiated Services Code Point). Since RFC 3168, the remaining two bits (the two
least significant bits) are used for Explicit Congestion Notification.

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In theory, a network could have up to 64 (i.e. 26) different traffic classes using
different markings in the DSCP. The DiffServ RFCs recommend, but do not require,
certain encodings. This gives a network operator great flexibility in defining traffic
classes. In practice, however, most networks use the following commonly-defined
Per-Hop Behaviors:

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Default PHBwhich is typically best-effort traffic

Expedited Forwarding (EF) PHBdedicated to low-loss, low-latency traffic

Assured Forwarding (AF) PHB which gives assurance of delivery under


conditions

Class Selector PHBswhich are defined to maintain backward compatibility with


the IP Precedence field

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The DS field consists of a 6-bit differentiated services code point (DSCP) RFC 2474.
Explicit Congestion Notification occupies the least-significant 2 bits. ECN allows endto-end notification of network congestion without dropping packets

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Because the CE indication can only be handled effectively by an upper layer protocol
that supports it, ECN is only used in conjunction with upper layer protocols (for
example, TCP).

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2.18 Differentiated Services

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If you need to mark packets in your network and all of the devices support IP DSCP
marking and matching, use the IP DSCP marking to mark your packets. This is
because the IP DSCP markings provide more packet marking options. 64 individual
values can be marked using IP DSCP marking, while only 8 individual values can be
marked using IP precedence marking.

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Expedited Forwarding (EF) PHBdedicated to low-loss, low-latency traffic. These


characteristics are suitable for voice, video and other realtime services.
EF traffic is often given strict priority queuing above all other traffic classes. Because
an overload of EF traffic will cause queuing delays and affect the jitter and delay
tolerances within the class, EF traffic is often strictly controlled through admission
control, policing and other mechanisms.

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2.19 Assured Forwarding

There are four assured forwarding (AF) classes, AF1x through AF4x. The first number
corresponds to the AF class and the second number (x) refers to the level of drop
preference within each AF class. There are three drop probabilities, ranging from 1
(low drop) through 3 (high drop). Depending on a network policy, packets can be
selected for a PHB based on required throughput, delay, jitter, loss, or according to
the priority of access to network services

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2.20 Weighted Fair Queing

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2.20.1

RNC Scheduling

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2.21 Questions

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CHAPTER 3

Layer 2 Switching
3.1 Introduction

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With the introduction of 4G systems, wireless networks are evolving to nextgeneration packet architectures capable of supporting enhanced broadband
connections. Simple text messaging and slow email downloads are being replaced by
high-speed connections that support true mobile office applications, real time video,
streaming music, and other rich multimedia applications. 4G wireless networks will
approach the broadband speeds and user experience now provided by traditional
DSL and cable modem wireline service.
From the wireless operators perspective, 4G systems are vastly more efficient at
using valuable wireless spectrum. These spectral efficiency improvements support
new high-speed services, as well as larger numbers of users. The additional speeds
and capacity provided by 4G wireless networks put additional strains on mobile
backhaul networks and the carriers providing these backhaul services. Not only are
the transport requirements much higher, but there is also a fundamental shift from
TDM transport in 2G and 3G networks to packet transport in 4G networks.
Understanding the impact of 4G on mobile backhaul transport is critical to deploying
efficient, cost-effective transport solutions that meet wireless carrier expectations for
performance, reliability and cost.

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3.2 Spectral Efficiency

The amount of bandwidth on a wireless network is ultimately constrained by two


factors: the spectral efficiency of the wireless interface and the amount of licensed
spectrum a carrier owns. Spectral efficiency is a fancy way of saying how much
information can be transmitted over a given radio channel (i.e., Hz). Spectral
efficiency is measured as the amount of data (bps) that can be transmitted for every
Hz of spectrum; the higher the number (bps/Hz), the better. Newer technologies,
such as LTE, use advanced modulation schemes (OFDM) that support higher spectral
efficiencies and higher data rates than 2G and 3G wireless networks.

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3.3 TCP/IP Layers

TCP/IP can be represented by the US DoD Model. This model describes the
relationship between the main protocols used by TCP/IP.
Prior to the development of this model most network protocols were vendordependent. The architecture behind TCP/IP is different in the sense that the same
protocol model can be run on a multitude of different computer systems without
modification of the operating system or hardware architecture. TCP/IP is designed to
run as an application.
The protocol was primarily used to support application-orientated functions and
process-to-process communications between hosts. Specific applications to provide
basic network services for users were written to run with TCP/IP. The objective of the
lower protocols was to provide support for the network layer application services.

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3.4 Message Switching

Message switching moves the entire message from connecting point to connecting
point, one step at a time. This method is sometimes referred to as store & forward.
Message switching creates a virtual or dedicated connection to the next switching
station. The entire message is transmitted and then the connection is terminated. The
receiving station must buffer the entire message and then create a connection to the
next switching station and forward the entire message. The message is forwarded one
step at time until it is received at the final destination.
The best example of message switching is E-mail servers.

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3.5 Frames

A frame is the fundamental unit of data transfer used on LANs. It has specific
network characteristics which relate to the type of network it originated on. All IEEE
compliant frames have a similar structure and start with a preamble which is
followed by hardware addresses for source and destination stations. Some network
frames (Token Ring) have some special fields for specific MAC control. After the
source MAC address, the LLC protocol data unit follows. Generally, LLC is three
bytes in size followed by the network layer protocol. The maximum and minimum
size of the protocol which follows LLC is dependent on the type of network. The
frame is finished by a four-byte trailer used for error checking.

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3.6 Hardware Addresses

A universal address is assigned to a Network Interface Card (NIC) on manufacture.


The address is stored in an EPROM and, as the device is initialised, the address is
copied into RAM where it can be used by software. The address is 48 bits in size,
divided into six eight-bit fields. The first byte of the address (left hand side) is byte 0
and the last byte (right hand side) is byte 5. The first three bytes of the address (0, 1
and 2) indicate the vendor, and are unique to that vendor. The last three bytes of the
address (3, 4 and 5) can be assigned by the vendor (approx. 16M addresses per
Vendor Code).
For IEEE addressing, two other bits are important in the address. The least significant
bit of the first byte (byte 0) indicates an Individual/Group address. A value of 0
indicates a unicast address and 1 indicates a multicast. The second bit of this byte
indicates a universally assigned address or locally managed (where the address has
been changed by software). Locally managed addresses are supported by different
LAN technologies, but caution must be used when setting locally managed addresses.
It is the Network Managers responsibility to ensure that the assigned addresses are
unique.
Hardware addresses are Data Link characteristics of the OSI protocol stack. They are
separate from software protocol addresses, which are Network layer addresses.

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3.7 Store & Forward

A store and forward switching hub stores the full incoming frame in a buffer. This
enables the switch to perform a CRC check to see if the frame contains any errors. If
the frame is error-free, the switch uses an address lookup table to obtain a destination
port. Once the address is obtained, the switch performs a cross-connect operation and
forwards the frame to the destination.
Since the frame must be buffered in shared RAM, this results in greater latency than
that provided by cut through switching. A key advantage of store and forward
switches, results from the buffering of the frames in the switch. Since they are placed
in memory, this enables frame processing functions to be added to the switch,
permitting vendors to support a variety of filtering operations and the gathering of
statistics for management reports.

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3.7.1 Cut Through Switching


A cut through switch examines the destination address of each frame entering the
port. It then searches a table of addresses associated with ports to obtain a port
destination. Once the port destination is determined, the switch initiates the crossconnection between incoming and outgoing port.
Cut through switching minimises the delay or latency associated with placing a frame
received on one port onto another port. Since the switching decision is made once the
destination address is read, this means that the full frame is not examined. Thus the
switch cannot perform error checking on a frame. This limitation does not present a
problem on most LANs, due to the extremely low error rates. However, when
erroneous frames are encountered, they are passed from one network segment to
another. This results in an unnecessary increase in network utilisation on the
destination segment, as a store and forward switch would discard the frames
containing one or more bit errors.

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3.7.2 Modified Cut-Through


Modified cut-through switches attempt to offer the best of both worlds by holding an
incoming Ethernet frame until the first 64 bytes have been received. If the frame is
bad, it is nearly always detected within the first 64 bytes, so a trade-off between
switch latency and error-checking is achieved. In effect, modified cut-though switches
act as store and forward switches for short frames. For large frames they act like cutthrough switches.

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3.8 Switches and VLANs

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Within corporate networks system administrators like to segregate users into separate
networks. The old method would have required that all users of one network be
connected to the same physical devices.
With modern switches, it is possible to isolate users into separate networks, whilst
connected to the same layer 2 devices.
Users belonging to the same network are simply attached to ports on a layer 2 switch.
These ports are then programmed to be part of the same virtual network. A virtual
network is known as a Virtual LAN (VLAN) and user ports on the same virtual
network would share the same VLAN ID.
You may have users who wish to be part of the same VLAN whilst connected to
different layer 2 switches. A physical connection between the switches needs to be
established, and the ports at either end of the link would have to be part of the same
VLAN.
VLAN Tagging (IEEE 802.1q)
The diagram above also shows the additional fields within the layer 2 ethernet frame
that can be used to identify traffic from separate VLANs.
The fields are:

TPI

Priority Bits.

Canonical, format of MAC addresses.

VI

Tag Protocol Identifier.

VLAN ID.

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3.9 Link Aggregation

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3.10 Carrier Ethernet Overview

The Metro Ethernet Forum (MEF) has led the industry in propagating Carrier
Ethernet and has identified five key attributes that distinguish Carrier Ethernet from
traditional LAN based Ethernet. These are:

Standardised services

Scalability

Service manageability

Quality of service

Reliability

Making Ethernet Connection-Oriented


Historically, Ethernet has been a connectionless technology by design. In classic LAN
environments, the connectionless capabilities of Ethernet MAC bridging and
CSMA/CD provided considerable flexibility, simplicity, and economy in networking
latency-insensitive traffic within a single, well-bounded administrative domain.

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3.11 The Three Essential Functions of ConnectionOriented Ethernet

The three essential functions that make Ethernet connection-oriented are:

Predetermined EVC paths

Resource reservation and admission control

Per-connection traffic engineering and traffic management

The ability to predetermine the EVC path through the Ethernet network is
fundamental to making Ethernet connection-oriented. In classic connectionless
Ethernet bridging, Ethernet frames are forwarded in the network according to the
MAC bridging tables in the learning bridge. If a destination MAC address is
unknown, the bridge floods the frame to all ports in the broadcast domain. Spanning
tree protocols like IEEE 802.1s are run to ensure that there are no loops in the
topology and to provide network restoration in the event of failure. Depending upon
the location and sequence of network failures, the path EVCs take through the
network may be difficult to predetermine.
Predetermining the EVC patheither through a management plane application or via
an embedded control planeensures that all frames in the EVC pass over the same
sets of nodes. Therefore, intelligence regarding the connection as a whole can now be
imparted to all nodes along the path.

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Resource reservation and CAC is the next critical function. Now that the EVC path
through the network has been explicitly identified, the actual bandwidth and queuing
resources required for each EVC are reserved in all nodes along the path. This is vital
to ensure the highest possible levels of performance in terms of packet loss, latency,
and jitter. CAC ensures that the requested resource is actually available in each node
along the path prior to establishing the EVC.

Once the path has been determined and the resources allocated, the traffic
engineering and traffic management functions ensure that the requested connection
performance is actually delivered. After packets have been classified on network
ingress, a variety of traffic management functions must be provided in any packetbased network. These include:

Policing

Shaping

Queuing

Scheduling

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3.12 Colour Marking

Packet classification is the processes of identifying to which EVCs the incoming


frames belong. The Ingress equipment can examine a variety of Ethernet and IP layer
information to make this decision. Once the incoming frame is classified, policing is
then applied to ensure that all frames coming into the network conform to the traffic
contract, known as the bandwidth profile, agreed to upon connection setup. Twolevel, three-colour marking allows incoming frames that conform to the CIR to be
admitted to the network. Frames that exceed even the EIR are discarded immediately,
and frames that exceed the CIR, but not the EIR, are marked for possible discard later,
should the network become congested. An EVC can be subject to a single such policer
if the bandwidth profile is applied to the entire EVC. EVCs can also include
bandwidth profiles for each of many CoS classes within the EVC. In this case, a single
EVC can be subject to multiple policers.

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3.13 Traffic Policing and Shaping

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3.14 MEF Bandwidth Profiles

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3.15 Mapping at the UNIs

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3.16 Class of Service (CoS)

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3.17 IEEE 802.1Q Virtual LANS (VLANS)

802.1Q provides for tagging Ethernet frames with VLAN IDs. It provides the
mechanism that enables multiple-bridged networks to transparently share the same
physical network while maintaining the isolation between networks. Ethernet
switches deliver packets within the same VLAN and send the traffic between different
VLANs to internal or external routers to perform the routing function. 802.1Q only
supports up to 4094 VLANs, which is a scaling constraint for service providers.

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3.18 Carrier Ethernet Services

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Based on the Metro Ethernet Forums (MEF) definitions, there are two broad
categories of Carrier Ethernet services: point-to-point, referred to as E-Line services;
and multipoint, referred to as E-LAN services. Both E-line and E-LAN services are
often provided with multiple classes of service (CoS); where a single Ethernet virtual
connection (EVC) can carry traffic with one or more CoS. Service providers desire to
build networks that offer all services simultaneously on a single converged
infrastructure.

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3.19 Questions

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CHAPTER 4

LTE Basic Air Interface


4.1 New Air Interface

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4.2 OFDMA

The downlink transmission scheme for E-UTRA FDD and TDD modes is based on
conventional OFDM. In an OFDM system, the available spectrum is divided into
multiple carriers, called subcarriers. Each of these subcarriers is independently
modulated by a low rate data stream. OFDM is used as well in WLAN, WiMAX and
broadcast technologies like DVB. OFDM has several benefits including its robustness
against multipath fading and its efficient receiver architecture.

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4.3 Advanced Antenna Techniques

LTE uses advanced antenna techniques and wider spectrum allocations to provide
higher data rates throughout the cell area. LTE supports MIMO, SDMA and
beamforming . These techniques are complementary and can be used to trade off
between higher sector capacity, higher user data rates, or higher cell-edge rates, and
thus enable operators to have finer control over the end-user experience.
DL MIMOLTE supports up to 4x4 MIMO in the DL, which uses four transmit
antennas at the Node B to transmit orthogonal (parallel) data streams to the four
receive antennas at the user equipment (UE). Using additional antennas and signal
processing at the receiver and transmitter, MIMO increases the system capacity and
user data rates without using additional transmit power or bandwidth. To be most
effective, MIMO needs a high signal-to-noise ratio (SNR) at the UE and a rich
scattering environment. High SNR ensures that the UE is able to decode the incoming
signal, and a rich scattering environment ensures the orthogonality of the multiple
data streams. The MIMO benefit is therefore maximised in a dense urban
environment, where there is enough scattering and the small cell sizes provide an
environment of high SNRs at the UE.

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SU-MIMO
Similarly, on the UL, SDMA enables two users in the cell to simultaneously send data
to the eNode B, using the same time-frequency resource. Even though the
transmissions are simultaneous, the spatial separation ensures that the two data
streams do not interfere with each other. Allowing these concurrent transmissions
increases the cell capacity in both the DL and the UL. LTE does not support
simultaneous MIMO and SDMA operation to a user; hence, there is a tradeoff
between higher user data rates and higher system capacity in the DL.

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Beamforming
Beamforming increases the user data rates by focusing the transmit power in the
direction of the user, effectively increasing the received signal strength at the UE.
Beamforming provides the most benefits to users in weaker-signal-strength areas, like
the edge of the cell coverage. Beamforming ensures that cell-edge rates are high, and
enables the operator to deploy high-bandwidth services without concern for service
degradation at the cell edge.

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4.4 Cyclic Delay Diversity

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4.5 FDD/TDD

LTE can be used in both paired (FDD) and unpaired (TDD) spectrum. Leading
suppliers first product releases will support both duplex schemes. In general, FDD is
more efficient and represents higher device and infrastructure volumes, while TDD is
a good complement, for example in spectrum centre gaps.
All cellular systems today use FDD, and more than 90 per cent of the worlds mobile
frequencies available are in paired bands. With FDD, downlink and uplink traffic is
transmitted simultaneously in separate frequency bands.
With TDD the transmission in uplink and downlink is discontinuous within the same
frequency band. As an example, if the time split between down- and uplink is 1/1, the
uplink is used half of the time. The average power for each link is then also half of the
peak power. As peak power is limited by regulatory requirements, the result is that
for the same peak power, TDD will offer less coverage than FDD.

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4.5.1 FDD

Two frame structure types are defined for E-UTRA: frame structure type 1 for FDD
mode, and frame structure type 2 for TDD mode.
For the frame structure type 1, the 10 ms radio frame is divided into 20 equally sized
slots of 0.5ms. A sub-frame consists of two consecutive slots, so one radio frame
contains ten sub-frames.

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4.5.2 TDD

The frame structure for the type 2 frames used on LTE TDD is somewhat different.
The 10 ms frame comprises two half frames, each 5 ms long. The LTE half-frames are
further split into five sub-frames, each 1ms long.
With TDD the transmission in uplink and downlink is discontinuous within the same
frequency band. As an example, if the time split between down- and uplink is 1/1, the
uplink is used half of the time. The average power for each link is then also half of the
peak power. As peak power is limited by regulatory requirements, the result is that
for the same peak power, TDD will offer less coverage than FDD.

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The special subframes consist of the three fields:

DwPTS (Downlink Pilot Timeslot)

GP (Guard Period)

UpPTS (Uplink Pilot Timeslot).

The special frames replace what would be a normal sub-frame.

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The DL to UL switching method ensures that the high power downlink transmissions
from the eNodeB from other neighbour cells do not interfere when the eNodeB UL
reception is going in the current cell.

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4.5.3 LTE TDD / TD-LTE Subframe Allocations

One of the advantages of using LTE TDD is that it is possible to dynamically change
the up and downlink balance and characteristics to meet the load conditions. In order
that this can be achieved in an ordered fashion, a number of standard configurations
have been set within the LTE standards.
A total of seven up/downlink configurations have been set, and these use either 5 ms
or 10 ms switch periodicities. In the case of the 5ms switch point periodicity, a special
sub-frame exists in both half frames. In the case of the 10 ms periodicity, the special
subframe exists in the first half frame only. It can be seen from the table above that the
sub-frames 0 and 5 as well as DwPTS are always reserved for the downlink. It can
also be seen that UpPTS and the sub-frame immediately following the special
subframe are always reserved for the uplink transmission.

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4.5.4 Flexible Carrier Bandwidths

LTE is defined to support flexible carrier bandwidths from below 1.4MHz up to


20MHz, in many spectrum bands and for both FDD and TDD deployments. This
means that an operator can introduce LTE in both new and existing bands.

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LTE supports a range of bandwidths up to 20 MHz, as depicted above. LTE also


supports devices that can work on various system-bandwidth combinations, therefore
reducing the need to tailor specific device profiles to each combination. This allows an
operator to deploy LTE in 10 or 20 MHz combinations, without worrying about
device-compatibility issues. LTE devices are mandated to support 20 MHz bandwidth
in the DL and the UL. The available peak rates and average user rates for an
individual user, however, scale with the deployment bandwidth.
LTE supports both FDD and TDD modes, allowing operators to address all available
spectrum resources.

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4.6 Slot Structure and Physical Resources

The subcarriers in LTE have a constant spacing of f = 15 kHz. In the frequency


domain, 12 subcarriers form one resource block. The resource block size is the same for
all bandwidths.
To each OFDM symbol, a cyclic prefix (CP) is appended as guard time. One downlink
slot consists of 6 or 7 OFDM symbols, depending on whether extended or normal cyclic
prefix is configured respectively. The extended cyclic prefix is able to cover larger cell
sizes with higher delay spread of the radio channel.

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Data symbols are independently modulated and transmitted over a high number of
closely spaced orthogonal subcarriers. In E-UTRA, downlink modulation schemes
QPSK, 16QAM, and 64QAM are available.

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4.6.1 Transmission Bandwidths

LTE must support the international wireless market and regional spectrum
regulations and spectrum availability. To this end the specifications include variable
channel bandwidths selectable from 1.4 to 20 MHz, with subcarrier spacing of 15 kHz.
If the new LTE eMBMS is used, a subcarrier spacing of 7.5 kHz is also possible.
Subcarrier spacing is constant regardless of the channel bandwidth.
3GPP has defined the LTE air interface to be "bandwidth agnostic," which allows the
air interface to adapt to different channel bandwidths with minimal impact on system
operation. The smallest amount of resource that can be allocated in the uplink or
downlink is called a resource block (RB). An RB is 180 kHz wide and lasts for one 0.5
ms timeslot. For standard LTE, an RB comprises 12 subcarriers at a 15 kHz spacing,
and for eMBMS with the optional 7.5 kHz subcarrier spacing an RB comprises 24
subcarriers for 0.5 ms. The maximum number of RBs supported by each transmission
bandwidth is given above.

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4.7 Orthagonality

Depending on the required data rate, each UE can be assigned one or more resource
blocks in each transmission time interval of 1 ms. The scheduling decision is made in
the base station (eNodeB).

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For standard LTE, an RB comprises 12 subcarriers at a 15 kHz spacing, and for


eMBMS with the optional 7.5 kHz subcarrier spacing an RB comprises 24 subcarriers
for 0.5 ms.

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4.7.1 Single-Frequency Network Multicast Services

LTE specifies a high-capacity multicast and broadcast service, using a singlefrequency network (also called multicast-broadcast single-frequency network or
MBSFN). As depicted above, all cells in the network (or a geographical area) transmit
time-synchronized, identical DL signals. At the user terminal, these multiple timesynchronized transmissions appear as a single transmission with high signal strength,
and thus can be easily decoded. In addition to the benefits of time-synchronised
transmissions, the robustness of OFDM to multipath propagation ensures that the
inter-cell interference is reduced.
The capacity benefits of the single-frequency network are highest when the same
content is transmitted in all cells of the macro network.

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4.7.2 Single Carrier Frequency Division Multiple Access (SCFDMA)

LTE has ambitious requirements for data rate, capacity, spectrum efficiency, and
latency. In order to fulfill these requirements, LTE is based on new technical
principles. LTE uses new multiple access schemes on the air interface: OFDMA
(Orthogonal Frequency Division Multiple Access) in downlink and SC-FDMA (Single
Carrier Frequency Division Multiple Access) in uplink.
While OFDMA is seen optimum to fulfil the LTE requirements in downlink, OFDMA
properties are less favourable for the uplink. This is mainly due to weaker peak-toaverage power ratio (PAPR) properties of an OFDMA signal, resulting in worse
uplink coverage.
Thus, the LTE uplink transmission scheme for FDD and TDD mode is based on SCFDMA (Single Carrier Frequency Division Multiple Access) with cyclic prefix. SC-FDMA
signals have better PAPR properties compared to an OFDMA signal. This was one of
the main reasons for selecting SCFDMA as LTE uplink access scheme. The PAPR
characteristics are important for cost-effective design of UE power amplifiers.

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4.8 Cyclic Prefix

In the time domain, a guard interval may be added to each symbol to combat interOFDM-symbol-interference due to channel delay spread. In EUTRA, the guard
interval is a cyclic prefix which is inserted prior to each OFDM symbol.
Delay spread is a type of distortion that is caused when an identical signal arrives at
different times at its destination. The signal usually arrives via multiple paths and
with different angles of arrival. The time difference between the arrival moment of the
first multipath component (typically the Line of sight component) and the last one, is
called delay spread.

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4.9 Delay Spread

The data to be transmitted on an OFDM signal is spread across the carriers of the
signal, each carrier taking part of the payload. This reduces the data rate taken by
each carrier. The lower data rate has the advantage that interference from reflections
is much less critical. This is achieved by adding a guard band time or guard interval
into the system. This ensures that the data is only sampled when the signal is stable
and no new delayed signals arrive that would alter the timing and phase of the signal.
The distribution of the data across a large number of carriers in the OFDM signal has
some further advantages. Nulls caused by multi-path effects or interference on a
given frequency only affect a small number of the carriers, the remaining ones being
received correctly. By using error-coding techniques, which does mean adding further
data to the transmitted signal, it enables many or all of the corrupted data to be
reconstructed within the receiver. This can be done because the error correction code
is transmitted in a different part of the signal. It is this error coding which is referred
to in the "Coded" word in the title of COFDM which is often seen.

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To each OFDM symbol, a cyclic prefix (CP) is appended as guard time. One downlink
slot consists of 6 or 7 OFDM symbols, depending on whether extended or normal
cyclic prefix is configured, respectively. The extended cyclic prefix is able to cover
larger cell sizes with higher delay spread of the radio channel.

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4.10 Slot Structure and Physical Resources

Data is allocated to the UEs in terms of resource blocks, i.e. one UE can be allocated
integer multiples of one resource block in the frequency domain. These resource
blocks do not have to be adjacent to each other. In the time domain, the scheduling
decision can be modified every transmission time interval of 1 ms. The scheduling
decision is done in the base station (eNodeB). The scheduling algorithm has to take
into account the radio link quality situation of different users, the overall interference
situation, Quality of Service requirements, service priorities, and so on.

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4.11 Scheduler

Scheduler in eNB (base station) allocates resource blocks (which are the smallest
elements of resource allocation) to users for predetermined amount of time. Slots
consist of either 6 (for long cyclic prefix) or 7 (for short cyclic prefix) OFDM symbols
Longer cyclic prefixes are desired to address longer fading. The number of available
subcarriers changes depending on transmission bandwidth (but subcarrier spacing is
fixed).

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Round Robin
The aim of this scheduler is to share the available/unused resources equally among
the RT terminals (i.e. the terminals requesting RT services) in order to satisfy their RTMBR demand.
This is a recursive algorithm and continues to share resources equally among RT
terminals, until all RT-MBR demands have been met or there are no more resources
left to allocate
Proportional Fair
The aim of this Scheduler is to allocate the available/unused resources as fairly as
possible in such a way that, on average, each terminal gets the highest possible
throughput achievable under the channel conditions.
This is a recursive algorithm. The remaining resources are shared between the RT
terminals in proportion to their bearer data rates. Terminals with higher data rates get
a larger share of the available resources. Each terminal gets either the resources it
needs to satisfy its RT-MBR demand, or its weighted portion of the available/unused
resources, whichever is smaller. This recursive allocation process continues until all
RT-MBR demands have been met or there are no more resources left to allocate.

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Proportional Demand
The aim of this scheduler is to allocate the remaining unused resources to RT
terminals in proportion to their additional resource demands. This is a non-recursive
allocation process and results in either satisfying the RT-MBR demands of all
terminals or the consumption of all of the resources.

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Max SINR
The aim of this Scheduler is to maximise the terminal throughput and, in turn, the
average cell throughput. This is a non-recursive resource allocation process, where
terminals with higher bearer rates (and consequently higher SINR) are preferred over
terminals with lower bearer rates (and consequently lower SINR). This means that
resources are allocated first to those terminals with better SINR/channel conditions,
thereby maximising the throughput.

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4.12 ASSET - LTE

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4.13 Downlink Data Transmission

Data is allocated to the UEs in terms of resource blocks. A physical resource block
consists of 12 (24) consecutive sub-carriers in the frequency domain for the Nf=15 kHz
(Nf=7.5 kHz) case. In the time domain, a physical resource block consists of DL
Nsymb consecutive OFDM symbols, DL Nsymb is equal to the number of OFDM
symbols in a slot.
Depending on the required data rate, each UE can be assigned one or more resource
blocks in each transmission time interval of 1 ms. The scheduling decision is done in
the base station (eNodeB).

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The user data is carried on the Physical Downlink Shared Channel (PDSCH).

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4.14 Modulation and Subcarriers

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4.15 Downlink Reference Signal Structure

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4.15.1

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Configuration of Carrier

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4.16 Reference Signal Received Power (RSRP)

Reference Signal Received Power (RSRP), is determined for a considered cell as the
linear average over the power contributions (in [W]) of the resource elements that
carry cell-specific reference signals within the considered measurement frequency
bandwidth. For RSRP determination, the cell-specific reference signals R0 and, if
available, R1 can be used. If receiver diversity is in use by the UE, the reported value
shall not be lower than the corresponding RSRP of any of the individual diversity
branches.
E-UTRA Carrier RSSI
E-UTRA Carrier Received Signal Strength Indicator, comprises the total received
wideband power observed by the UE from all sources, including co-channel serving
and non-serving cells, adjacent channel interference, thermal noise and so on.
Reference Signal Received Quality (RSRQ)
RSRQ is defined as the ratio NRSRP / (E-UTRA carrier RSSI), where N is the number
of RBs of the E-UTRA carrier RSSI measurement bandwidth. The measurements in the
numerator and denominator shall be made over the same set of resource blocks.

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4.17 Channel Quality Indicator Reporting

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4.18.1

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Physical Cell Identity (PCI)

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Physical Downlink Control Channel

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4.19 Questions

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CHAPTER 5

LTE Network
Architecture and
Protocols

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5.1 LTE Architecture

LTE capabilities include:

Downlink peak data rates up to 326 Mbps with 20 MHz bandwidth.

Uplink peak data rates up to 86.4 Mbps with 20 MHz bandwidth.

Operation in both TDD and FDD modes.

Scalable bandwidth up to 20 MHz, covering 1.25, 2.5, 5, 10, 15, and 20 MHz in the
study phase.

Reduced latency, to 10 msec round-trip time between user equipment and the base
station, and to less than 100 msec transition time from inactive to active.

The overall intent is to provide an extremely high-performance radio-access


technology that offers full vehicular speed mobility and that can readily coexist with
HSPA and earlier networks. Because of scalable bandwidth, operators will be able to
easily migrate their networks and users from HSPA to LTE over time.

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5.2 Roaming Architecture

A network run by one operator in one country is known as a Public Land Mobile
Network (PLMN). Roaming is where users are allowed to connect to PLMNs other
than those to which they are directly subscribed.

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5.3 Bearer Establishment Procedure

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5.4 KPI-RAB Success

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5.5 KPI Dropped Call Ratio

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Radio Bearer Reconfiguration procedure allows the modification of the following


parameters:
DRX/DTX Re-configuration for UE in RRC_CONNECTED state (Radio Bearer
Reconfiguration procedure is used to reconfigure the RRC Connection.)
Modification of QoS parameters (QoS parameters are listed below).

Modification of long lived PRB allocation

Modification of fixed MCS allocation

QoS definitions for Radio Bearers which can be modified are listed below:

QoS-Label/ QoS Profile ID

UL Guaranteed Bit rate[1]GBR

UL Maximum Bit rate[2]

DL Guaranteed Bit rate GBR

DL Maximum Bit rate

Allocation / Retention Priority

[1] Guaranteed bit rate (GBR) specifies the guaranteed number of bits delivered by EUTRA within a period of time (provided there is data to deliver).
[2] Maximum bit rate (MBR) specifies a maximum number of bits delivered by UMTS
within a period of time

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5.6 The Home Subscriber Service

The HSS contains users SAE subscription data such as the EPS-subscribed QoS
profile and any access restrictions for roaming. It also holds information about the
PDNs to which the user can connect.
This could be in the form of an Access Point Name (APN) (which is a label according
to DNS1 naming conventions describing the access point to the PDN), or a PDN
Address (indicating subscribed IP address(es)). In addition, the HLR holds dynamic
information such as the identity of the MME to which the user is currently attached or
registered. The HLR may also integrate the Authentication Centre (AuC) which
generates the vectors for authentication and security keys.
Security functions are the responsibility of the MME for both signalling and user data.
When a UE attaches with the network, a mutual authentication of the UE and the
network is performed between the UE and the MME/HSS. This authentication
function also establishes the security keys which are used for encryption of the
bearers.
All data sent over the radio interface is encrypted

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5.7 PDN Gateway

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5.8 Network Sharing

The LTE architecture enables service providers to reduce the cost of owning and
operating the network by allowing the service providers to have separate CN (MME,
SGW, PDN GW) while the E-UTRAN (eNBs) is jointly shared by them. This is enabled
by the S1-flex mechanism by enabling each eNB to be connected to multiple CN
entities. When a UE attaches to the network, it is connected to the appropriate CN
entities based on the identity of the service provider sent by the UE.

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5.9 Session Initiation Protocol Architecture

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5.10 LTE Functional Nodes

Radio Resource Management:


Radio Bearer Control, Radio Admission Control, Connection
Bearer control is the allocation of resources to the UE, so the number of sub carriers,
coding scheme, power levels and so on.
Mobility Control, Dynamic allocation of resources to UEs in both UL and DL
(scheduling).
Mobility control includes handovers, which can now be achieved in some LTE cases
without the need for any decision making (and minimal signalling) by the Core
Network.
The allocation of resources to the UE includes the number of sub carriers, coding
scheme, power levels and so on. Obviously, this includes load sharing, enabling
everyone in the cell to get appropriate resources depending on the number of users in
the cell and cell capacity.
IP header compression and encryption of user data stream.
Header compression allows more user data information to be sent rather than using
resources. The purpose of IP header compression algorithm is to improve the ratio of
the overhead versus the payload for an IP packet. It is of tremendous importance
since the increase of the address space when shifting to IPv6 translates into an
increase of the header size.
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All data will be encrypted for over the air transmission, ensuring user confidentiality.
Scheduling and transmission of paging messages (from the MME);
A UE will have to be paged when there is data to send to it or when an incoming
phone call is being made.
mobility and scheduling.
UEs inform the network which cells it is receiving and the power level and quality of
those signals. The eNB can provide the UE assistance, by providing a list of
frequencies, scrambling codes (UTRAN) etc, and perhaps even a list of preferred
networks and specific frequencies to measure. We will talk more about this later.

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5.11 Physical Channels, Transport Channels &


Logical Channels

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5.11.1

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Logical Channels

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5.11.2

Transport Channels

The LTE transport channels vary between the uplink and the downlink, as each has
different requirements and operates in a different manner. Physical layer transport
channels offer information transfer to medium access control (MAC) and higher
layers.
Broadcast Channel (BCH): The LTE transport channel maps to Broadcast Control
Channel (BCCH)
Downlink Shared Channel (DL-SCH):This transport channel is the main channel for
downlink data transfer. It is used by many logical channels.
Paging Channel (PCH): To convey the PCCH
Multicast Channel (MCH): This transport channel is used to transmit MCCH information
to set up multicast transmissions

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Downlink Transport Channels


BCH (Broadcast Channel): It has a fixed transport format, provided by the specifications.
It is used for transmission of the information on the BCCH logical channel. It can be
characterised by fixed, predefined transport format and the requirement to be
broadcast in the entire coverage area of the cell.
DLSCH (Downlink Shared Channel): DL-SCH is the transport channel used for
transmission of downlink data in LTE. It supports LTE features such as dynamic rate
adaptation and channel dependent scheduling in the time and frequency domain,
hybrid ARQ, and spatial multiplexing.
PCH (Paging Channel): It is used for transmission of paging information on the PCCH
logical channel. The PCH supports DRX to allow the mobile terminal to save battery
power by sleeping and waking up to receive the PCH only at predefined time
instances.
MCH (Multicast Channel): It is used to support MBMS. It is characterised by a semi-static
transport format and semi-static scheduling. In case of multi-cell transmission using
MBSFN, the scheduling and transport format configuration is coordinated among the
cells involved in the MBSFN transmission
Uplink Transport Channels UL-SCH (Uplink Shared Channel): It is characterised by the
possibility to use beamforming; support for HARQ, dynamic link adaptation by
varying the transmit power and potentially modulation and coding and also for both
dynamic and semi-static resource allocation.
RACH (Random Access Channel(s)): It is characterised by limited control information and
collision risk. The possibility of using open loop power control depends on the
physical layer solution.
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5.11.3

Physical Channels

PBCH (Physical Broadcast Channel):


The coded BCH transport block is mapped to four subframes within a 40 ms interval.
This 40 ms timing is blindly detected, i.e. there is no explicit signalling indicating 40
ms timing. Each subframe is assumed to be self-decodable, i.e. the BCH can be
decoded from a single reception, assuming sufficiently good channel conditions.

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Physical Broadcast Channel (PBCH): This physical channel carries system information for
UEs requiring to access the network.
Physical Downlink Control Channel (PDCCH): The main purpose of this physical channel is
to carry mainly scheduling information.
Physical Hybrid ARQ Indicator Channel (PHICH): As the name implies, this channel is used
to report the Hybrid ARQ status.
Physical Downlink Shared Channel (PDSCH): This channel is used for unicast and paging
functions.
Physical Multicast Channel (PMCH): This physical channel carries system information for
multicast purposes.
Physical Control Format Indicator Channel (PCFICH): This provides information to enable
the UEs to decode the PDSCH

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5.12 Modulation and Coding

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5.12.1

PDSCH

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5.12.2

PUSCH

PUSCH (Physical Uplink Shared Channel): Carries the UL-SCH data, CQI, PMI and RI.
RI (Rank Indicator): RI indicates the number of spatial layers that can be supported by
the UE, based on the channel conditions. The transmission rank selected to be used is
dependent on RI as well as other factors (depending on the vendor) such as traffic
pattern, available transmission bandwidth and so on. RI is compulsory for both open
and closed loop spatial multiplexing.
PMI (Precoding Matrix Indicator): PMI ensures that the correct spatial domain precoding
matrix is applied by the eNodeB so that the transmitted signal matches with the
spatial channel experienced by the UE. It is denoted by the Transmit Precoding Matrix
Indicator (TPMI) that consists of 3 bit or 6 bit information field for 2 or 4 transmit
antennas, respectively. It is compulsory for closed loop spatial multiplexing.
CQI (Channel Quality Indicator): It is a 4 bit index pointing into a table of 16 different
modulation and coding schemes. It indicates or suggests a combination of modulation
and coding scheme that the eNodeB should use to ensure that the BLER (Block Error
Ratio) experienced by the UE remains less than 10%.

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5.13 Functional Nodes

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5.13.1

Functional Nodes - UE

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5.13.2

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Functional Nodes - eNodeB

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5.14.1

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RLC Modes - Qos

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5.15 Medium Access Control (MAC)

PDCCH (Physical Downlink Control Channel): Informs the UE about the resource allocation
of PCH and DL-SCH, and Hybrid ARQ information related to DL-SCH. It also carries
the uplink scheduling grant. The downlink control signalling (PDCCH) is located in
the first n OFDM symbols where n 3 and consists of:

Transport format, resource allocation, and hybrid-ARQ information related to DLSCH, and PCH

Transport format, resource allocation, and hybrid-ARQ information related to ULSCH

QPSK modulation is used for all control channels

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QoS differentiation, i.e. prioritisation of different services according to their


requirements becomes extremely important when the system load gets higher. The
most relevant parameters of QoS classes are:
Transfer Delay: This represents how delay-sensitive the traffic is. For example, the VoIP
class is meant for very delay sensitive traffic, while the P2P File Sharing class is delayinsensitive.
Guaranteed Bit rate: Delay sensitive QoS Classes have guaranteed bit rate requirements.
This defines the minimum bearer bit rate that the E-UTRAN must provide and it can
be used in admission control and in resource allocation. Each guaranteed bit rate
service also has a maximum bit rate demand, i.e. it can't exceed this limit.
Allocation and Retention Priority (ARP): Within each QoS class there are different
allocation and retention priorities. The primary purpose of ARP is to decide whether a
bearer establishment/modification request can be accepted or needs to be rejected in
case of resource limitations (typically available radio capacity in case of GBR bearers).
In addition, the ARP can be used (for example, by the eNodeB) to decide which
bearer(s) to drop during exceptional resource limitations (for example, at handover).
It is important to remember that pure prioritisation in packet scheduling alone is not
enough to provide full QoS differentiation gains. Users within the same QoS class and
ARP class will share the available capacity. If the number of users is simply too high,
then they will suffer from bad quality. In that case it is better to block a few users to
guarantee the quality of existing connections, like streaming videos.
The radio network can estimate the available radio capacity and block an incoming
user.

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5.16 Physical Control Channel

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5.16.1

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Physical Downlink Control Channel

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PCFICH (Physical Control Format Indicator Channel): Informs the UE about the number of
OFDM symbols used for the PDCCHs. It is transmitted in every subframe.

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5.17 Multimedia Broadcast/Multicast Service


(MBMS)

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5.18 Reference Signal Received Power (RSRP)

Physical Layer Measurements and Indicators


Reference Signal Received Power (RSRP), is determined for a considered cell as the
linear average over the power contributions (in [W]) of the resource elements that
carry cell-specific reference signals within the considered measurement frequency
bandwidth. For RSRP determination, the cell-specific reference signals R0 and, if
available, R1 can be used. If receiver diversity is in use by the UE, the reported value
shall not be lower than the corresponding RSRP of any of the individual diversity
branches.
E-UTRA Carrier RSSI
E-UTRA Carrier Received Signal Strength Indicator, comprises the total received
wideband power observed by the UE from all sources, including co-channel serving
and non-serving cells, adjacent channel interference, thermal noise and so on.
Reference Signal Received Quality (RSRQ)
RSRQ is defined as the ratio NRSRP / (E-UTRA carrier RSSI), where N is the number
of RBs of the E-UTRA carrier RSSI measurement bandwidth. The measurements in
the numerator and denominator shall be made over the same set of resource blocks.

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5.19 Quality Channel Indicator Reporting

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CHAPTER 6

Mobility Management
6.1 UE States

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Co-existence with legacy standards and systems: LTE users should be able to make voice
calls from their terminal and have access to basic data services even when they are in
areas without LTE coverage. LTE therefore allows smooth, seamless service handover
in areas of HSPA, WCDMA or GSM/GPRS/EDGE coverage. Furthermore, LTE/SAE
supports not only intra-system and intersystem handovers, but inter-domain
handovers between packet switched and circuit switched sessions.

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6.2 UE Power-up

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6.2.1 EPS Mobility Management

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6.2.2 Tracking Area Update

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6.3 LTE Functional Modes - MME

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6.4 RRC States

In order to provide seamless service continuity, ensuring mobility between LTE and
legacy technologies is therefore very important. These technologies include
GSM/GPRS and WCDMA/HSPA.

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6.5 Reference Signal Received Power (RSRP)

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6.6 Cell Re-selection

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If a neighboring cell was ranked with the highest value R, will the UE start the cell reselection?
If it is a GSM or TDD cell, then indeed the UE performs the cell re-selection process to
this cell. If it is an FDD cell, it depends on the used quality measure.
There are two options: CPICH RSCP or CPICH Ec/No.
The UE learns from the system information, which quality measure to use.
If the quality measure CPICH RSCP is used, the UE perform the cell re-selection. If the
quality measure Ec/No is used, the UE has to make a second ranking based on the
same measurement quantity. The UE performs cell re-selection to the FDD cell, which
was ranked best in the second ranking process.
Is the cell re-selection initiated immediately after the UE ranks a neighbouring cell to
be the best?
If so, we could face a ping-pong effect a UE often performing cell reselection
between two neighbouring cells.
To avoid this, the operator uses the time interval value Treselection, whose value
ranges between 0 and 31 seconds. Only when a cell was ranked Treselection seconds
better then the serving cell, a cell reselection to this cell takes place. In addition to this,
a UE must camp at least 1 second on a serving cell, before the next cell re-selection
may take place.
How often are the cell re-selection criteria evaluated?
This is done at least once every DRX cycle for cells, for which new measurement
results are available.
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6.7 Handover RRC Connected

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6.7.1 LTE Reference Signal Received Quality (RSRQ)

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6.7.2 Received Signal Strength Indicator (RSSI)

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6.7.3 Reference Signal Received Power (RSRP)

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6.7.4 Handover

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6.7.5 Automatic Neighbour Relations

The mobiles continuously measure the RSRP from the serving cell and candidate cells
(cells in the vicinity of the mobile that might be considered as handover candidates).
A measurement report is typically triggered when the RSRP from a candidate cell is
within a threshold D dB from the serving cell RSRP.
The measurement report contains information about the PCI and the corresponding
RSRP of the candidate cell. The serving cell may order the mobile to read the GID
(transmitted on the broadcast channel from each cell) of a cell with a certain PCI and
report that back to the serving cell.
This could be done for example if the PCI is associated with a cell with handover
failures in the past or if a central node such as the OSS has requested it. In any case,
the GID of a neighbouring cell can be obtained with help from a mobile station upon
request from the serving cell. In case the serving cell decides to set up a relation to the
neighbouring cell it contacts the central configuration server in the network and
obtains the IP address.

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Is the PCI of the candidate cell already known in the serving cell (i.e. is the neighbour
relation already established)?
Yes: Initiate handover decision procedure.
No: Consider the candidate cell as a NCR list candidate.
Order the UE to report GID. Obtain connectivity information for the candidate cell
and signal to the candidate cell, directly or through the core network, about a mutual
addition to the NCR lists of the two cells.

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6.7.6 Measurement Configuration

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6.8 Questions

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