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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 intended for the use of AIRCOM International's customers only for the purposes of the agreement under which the document is submitted, and no part of it may be reproduced or transmitted in any form or means without the prior written permission of AIRCOM International. The document has been prepared to be used by professional and properly trained personnel, and the customer assumes full responsibility when using it. AIRCOM International welcomes customer comments as part of the process of continuous development and improvement of the documentation. 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. AIRCOM International's liability for any errors in the document is limited to the documentary correction of errors. AIRCOM International WILL NOT BE RESPONSIBLE IN ANY EVENT FOR ERRORS IN THIS DOCUMENT OR FOR ANY DAMAGES, INCIDENTAL OR CONSEQUENTIAL (INCLUDING MONETARY LOSSES), that might arise from the use of this document or the information in it. This document and the product it describes are considered protected by copyright according to the applicable laws. ASSET is a registered trademark of AIRCOM International. Other product names mentioned in this document may be trademarks of their respective companies, and they are mentioned for identification purposes only. Copyright AIRCOM International 2010. All rights reserved.

Contents
1 Introduction to LTE
1.1 1.2 1.3 1.4 1.5 1.6 1.7 1.8 1.9 1.11 Where are we? Release 99 UTRAN 3G Services and QoS Classes HS-PDSCH HSDPA HSUPA HSPA+ Upgrade Paths to LTE Scalability of Bandwidth

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9 11 15 21 22 23 24 Error! Bookmark not defined. 28 29 33 34

1.10 R8 LTE 1.12 LTE Key Features

IP Core Network Overview


2.1 2.2 2.3 2.4 2.5 2.6 2.7 2.8 2.9 2.11 The TCP/IP Layers Transport Layer Protocols Application Layer Services TCP/IP Inter-networks IP Datagram Format

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41 43 47 52 53 55 56 57 59 60 61 62 63 66 72 73 75

Maximum Transfer Unit (MTU) Fragmentation Time To Live (TTL) IP Addresses Address Classes

2.10 Dotted Decimal Notation 2.12 Special IP Addresses 2.13 Realtime Transport Protocol (RTP) 2.14 Base Header Format 2.15 The Need for QoS 2.16 IP Precedence 2.17 Type of Service

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2.18 Differentiated Services


2.18.1 2.18.2 2.18.3 Assured Forwarding Weighted Fair Queing RNC Scheduling

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86 88 89

2.19 Questions

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Layer 2 Switching91
3.1 3.2 3.3 3.4 3.5 3.6 3.7 Introduction Spectral Efficiency The TCP/IP Layers Message Switching Frames Hardware Addresses Store & Forward
Cut Through Switching Modified Cut-Through

91 93 95 97 98 101 102
103 104

3.7.1 3.7.2

3.8 3.9 3.11

Switches and VLANs Link Aggregation The Three Essential Functions of Connection-Oriented Ethernet

105 109 114 118 124 125 126 128 130 131 132 136

3.10 Carrier Ethernet Overview 3.12 Colour Marking 3.13 Traffic Policing and Shaping 3.14 MEF Bandwidth Profiles 3.15 Mapping at the UNIs 3.16 Class of Service (CoS) 3.17 IEEE 802.1Q Virtual LANS (VLANS) 3.18 Carrier Ethernet Services 3.19 Questions

LTE Basic Air Interface 137


4.1 4.2 4.3 4.4 4.5 New Air Interface OFDMA Advanced Antenna Techniques Cyclic Delay Diversity FDD/TDD
FDD TDD LTE TDD / TD-LTE Subframe Allocations Flexible Carrier Bandwidths Transmission Bandwidths K025 LTE Technology for Engineers
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145 146 149 150

4.5.1 4.5.2 4.5.3 4.5.4

4.6

Slot Structure and Physical Resources

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155

4.6.1 Page 6

4.7

Orthagonality
Single-Frequency Network Multicast Services Single Carrier Frequency Division Multiple Access (SC-FDMA)

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158 160

4.7.1 4.7.2

4.8 4.9 4.11

Cyclic Prefix Delay Spread Scheduler

162 163 166 167 172 173 175 177


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4.10 Slot Structure and Physical Resources 4.12 ASSET - LTE 4.13 Downlink Data Transmission 4.14 Modulation and Subcarriers 4.15 Downlink Reference Signal Structure
4.15.1 Configuration of Carrier

4.16 Reference Signal Received Power (RSRP) 4.17 Channel Quality Indicator Reporting 4.18 Downlink Shared Channel (DL-SCH)
4.18.1 4.18.2 Physical Cell Identity (PCI) Physical Downlink Control Channel

181 183 187


188 189

4.19 Questions

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LTE Network Architecture and Protocols


5.1 5.2 5.3 5.4 5.5 5.6 5.7 5.8 5.9 5.11 LTE Architecture Roaming Architecture Bearer Establishment Procedure KPI-RAB Success KPI Dropped Call Ratio ? The Home Subscriber Service PDN Gateway Network Sharing Session Initiation Protocol Architecture

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192 195 199 204 205 210 212 217 219 224 227
228 231 235

5.10 LTE Functional Nodes Physical Channels, Transport Channels & Logical Channels
Logical Channels Transport Channels Physical Channels PDSCH PUSCH Functional Nodes - UE Functional Nodes - eNodeB RLC Modes - Qos 5.11.1 5.11.2 5.11.3 5.12.1 5.12.2 5.13.1 5.13.2 5.14.1

5.12 Modulation and Coding

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239 241

5.13 Functional Nodes

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243 244

5.14 RLC Modes

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5.15 Medium Access Control (MAC) 5.16 Physical Control Channel


5.16.1 Physical Downlink Control Channel

247 251
252

5.17 Multimedia Broadcast/Multicast Service (MBMS) 5.18 Reference Signal Received Power (RSRP) 5.19 Quality Channel Indicator Reporting

257 264 266

Mobility Management
6.1 6.2 UE States UE Power-up

271
271 273
275 277

6.2.1 6.2.2

EPS Mobility Management Tracking Area Update

6.3 6.4 6.5 6.6 6.7

LTE Functional Modes - MME RRC States Reference Signal Received Power (RSRP) Cell Re-selection Handover RRC Connected
LTE Reference Signal Received Quality (RSRQ) Received Signal Strength Indicator (RSSI) Reference Signal Received Power (RSRP) Handover Automatic Neighbour Relations Measurement Configuration

279 282 295 299 302


306 307 308 309 315 322

6.7.1 6.7.2 6.7.3 6.7.4 6.7.5 6.7.6

6.8

Questions

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C HAPTER 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 BCH, PCH, RACH CPCH, DCH, DSCH, FACH CPCH, DCH, DSCH, FACH CPCH, DCH, DSCH, FACH Coding scheme Convolutional code Convolutional code Turbo code No coding Coding rate 1/2 1/3, 1/2 1/3 -

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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: 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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C HAPTER 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 HLEN Service Type

Protocol version (currently 4) Length of header in 32 bit words (normally 5) 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 Flags fragments) Unique ID for each datagram, used for fragmentation 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. Checksum Specifies how long (in router hops) the datagram is to remain in the

Protocol ID of transport protocol - UDP, TCP, (ICMP) etc. Checksum of the header only 32 bit IP address of source Source IP Address

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 DATA

Padding to extend options data to multiple of 4 octets 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: 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: 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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C HAPTER 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 P C VI = Tag Protocol Identifier. Priority Bits. Canonical, format of MAC addresses. 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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C HAPTER 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

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 Downlink Shared Channel (DL-SCH)

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4.18.1

Physical Cell Identity (PCI)

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Physical Downlink Control Channel

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4.19 Questions

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C HAPTER 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

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

Functional Nodes - eNodeB

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5.14 RLC Modes

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5.14.1

RLC Modes - Qos

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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

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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5.18 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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C HAPTER 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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