368 115 10MB
English Pages 574 Year 2007
WCDMA FOR UMTS – HSPA EVOLUTION AND LTE Fourth Edition
Edited by Harri Holma and Antti Toskala Both of Nokia, Finland
WCDMA FOR UMTS
WCDMA FOR UMTS – HSPA EVOLUTION AND LTE Fourth Edition
Edited by Harri Holma and Antti Toskala Both of Nokia, Finland
Copyright # 2007
John Wiley & Sons Ltd, The Atrium, Southern Gate, Chichester, West Sussex PO19 8SQ, England Telephone (+44) 1243 779777
Email (for orders and customer service enquiries): [email protected] Visit our Home Page on www.wiley.com All Rights Reserved. No part of this publication may be reproduced, stored in a retrieval system or transmitted in any form or by any means, electronic, mechanical, photocopying, recording, scanning or otherwise, except under the terms of the Copyright, Designs and Patents Act 1988 or under the terms of a licence issued by the Copyright Licensing Agency Ltd, 90 Tottenham Court Road, London W1T 4LP, UK, without the permission in writing of the Publisher. Requests to the Publisher should be addressed to the Permissions Department, John Wiley & Sons Ltd, The Atrium, Southern Gate, Chichester, West Sussex PO19 8SQ, England, or emailed to [email protected], or faxed to (þ44) 1243 770620. Designations used by companies to distinguish their products are often claimed as trademarks. All brand names and product names used in this book are trade names, service marks, trademarks or registered trademarks of their respective owners. The Publisher is not associated with any product or vendor mentioned in this book. All trademarks referred to in the text of this publication are the property of their respective owners. This publication is designed to provide accurate and authoritative information in regard to the subject matter covered. It is sold on the understanding that the Publisher is not engaged in rendering professional services. If professional advice or other expert assistance is required, the services of a competent professional should be sought. Other Wiley Editorial Offices John Wiley & Sons Inc., 111 River Street, Hoboken, NJ 07030, USA Jossey-Bass, 989 Market Street, San Francisco, CA 94103-1741, USA Wiley-VCH Verlag GmbH, Boschstr. 12, D-69469 Weinheim, Germany John Wiley & Sons Australia Ltd, 42 McDougall Street, Milton, Queensland 4064, Australia John Wiley & Sons (Asia) Pte Ltd, 2 Clementi Loop #02-01, Jin Xing Distripark, Singapore 129809 John Wiley & Sons Canada Ltd, 22 Worcester Road, Etobicoke, Ontario, Canada M9W 1L1 Wiley also publishes its books in a variety of electronic formats. Some content that appears in print may not be available in electronic books. Wiley Bicentennial Logo: Richard J. Pacifico Library of Congress Cataloging-in-Publication Data WCDMA for UMTS : radio access for third generation mobile communications / Harri Holma and Antti Toskala. – 4th ed. p. cm. ISBN 978-0-470-31933-8 (cloth) 1. Code division multiple access. 2. Wireless communication systems–Standards. 3. Mobile communication systems–Standards. 4. Global system for mobile communications. I. Holma, Harri, 1970- II. Toskala, Antti. TK5103.452.W39 2007 621.3845–dc22 2007019042 British Library Cataloguing in Publication Data A catalogue record for this book is available from the British Library ISBN 978-0-470-31933-8 (HB) Typeset in 10/12pt Times by Thomson Digital, New Delhi. Printed and bound in Great Britain by Antony Rowe Ltd, Chippenham, England. This book is printed on acid-free paper responsibly manufactured from sustainable forestry in which at least two trees are planted for each one used for paper production.
Contents Preface
xvii
Acknowledgements
xxi
Abbreviations
xxiii
1 Introduction
1
Harri Holma and Antti Toskala 1.1 WCDMA in Third-Generation Systems 1.2 Spectrum Allocations for Third-Generation Systems 1.3 Requirements for Third-Generation Systems 1.4 WCDMA and its Evolution 1.5 System Evolution References
1 2 3 4 6 6
2 UMTS Services
9
Harri Holma, Martin Kristensson, Jouni Salonen and Antti Toskala 2.1 Introduction 2.2 Person-to-Person Circuit Switched Services 2.2.1 AMR-NB and AMR-WB Speech Services 2.2.2 Video Telephony 2.3 Person-to-Person Packet Switched Services 2.3.1 Messaging 2.3.2 Push-to-Talk over Cellular 2.3.3 Voice over IP 2.3.4 Multiplayer Games 2.4 Content-to-Person Services 2.4.1 Browsing 2.4.2 Audio and Video Streaming 2.4.3 Content Download 2.5 Business Connectivity 2.6 Location Services 2.6.1 Cell-Coverage-Based Location Calculation 2.6.2 Assisted GPS
9 10 10 13 15 15 19 21 21 22 22 23 24 24 26 27 28
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2.7 2.8
QoS Differentiation Capacity and Cost of Service Delivery 2.8.1 Capacity per Subscriber 2.8.2 Cost of Voice and Data Delivery 2.9 Summary References
29 34 34 35 37 38
3 Introduction to WCDMA
39
Peter Muszynski and Harri Holma 3.1 Introduction 3.2 Summary of the Main Parameters in WCDMA 3.3 Spreading and Despreading 3.4 Multipath Radio Channels and Rake Reception 3.5 Power Control 3.6 Softer and Soft Handovers References
39 41 44 47 50 52
4 Background and Standardisation of WCDMA
53
Antti Toskala 4.1 Introduction 4.2 Background in Europe 4.2.1 Wideband CDMA 4.2.2 Wideband TDMA 4.2.3 Wideband TDMA/CDMA 4.2.4 OFDMA 4.2.5 ODMA 4.2.6 ETSI Selection 4.3 Background in Japan 4.4 Background in Korea 4.5 Background in the United States 4.5.1 W-CDMA N/A 4.5.2 UWC-136 4.5.3 cdma2000 4.5.4 TR46.1 4.5.5 WP-CDMA 4.6 Creation of 3GPP 4.7 How does 3GPP Operate? 4.8 Creation of 3GPP2 4.9 Harmonisation Phase 4.10 IMT-2000 Process in ITU 4.11 Beyond 3GPP Release 99 References
53 53 54 55 55 56 56 57 57 58 58 58 58 58 59 59 59 61 62 62 62 63 65
39
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5 Radio Access Network Architecture
67
Fabio Longoni, Atte La¨nsisalmi and Antti Toskala 5.1 System Architecture 5.2 UTRAN Architecture 5.2.1 The Radio Network Controller 5.2.2 The Node B (Base Station) 5.3 General Protocol Model for UTRAN Terrestrial Interfaces 5.3.1 General 5.3.2 Horizontal Layers 5.3.3 Vertical Planes 5.4 Iu, the UTRAN–CN Interface 5.4.1 Protocol Structure for Iu CS 5.4.2 Protocol Structure for Iu PS 5.4.3 RANAP Protocol 5.4.4 Iu User Plane Protocol 5.4.5 Protocol Structure of Iu BC, and the Service Area Broadcast Protocol 5.5 UTRAN Internal Interfaces 5.5.1 RNC–RNC Interface (Iur Interface) and the RNSAP Signalling 5.5.2 RNC–Node B Interface and the NBAP Signalling 5.6 UTRAN Enhancements and Evolution 5.6.1 IP Transport in UTRAN 5.6.2 Iu Flex 5.6.3 Stand-Alone SMLC and Iupc Interface 5.6.4 Interworking Between GERAN and UTRAN, and the Iur-g Interface 5.6.5 IP-Based RAN Architecture 5.7 UMTS CN Architecture and Evolution 5.7.1 Release 99 CN Elements 5.7.2 Release 5 CN and IP Multimedia Subsystem References
67 70 71 72 72 72 72 73 74 74 76 77 78 79 80 80 83 85 85 85 86 86 86 87 87 88 89
6 Physical Layer
91
Antti Toskala 6.1 Introduction 6.2 Transport Channels and their Mapping to the Physical Channels 6.2.1 Dedicated Transport Channel 6.2.2 Common Transport Channels 6.2.3 Mapping of Transport Channels onto the Physical Channels 6.2.4 Frame Structure of Transport Channels 6.3 Spreading and Modulation 6.3.1 Scrambling 6.3.2 Channelisation Codes
91 92 93 93 95 95 96 96 96
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6.3.3 Uplink Spreading and Modulation 6.3.4 Downlink Spreading and Modulation 6.3.5 Transmitter Characteristics 6.4 User Data Transmission 6.4.1 Uplink Dedicated Channel 6.4.2 Uplink Multiplexing 6.4.3 User Data Transmission with the Random Access Channel 6.4.4 Uplink Common Packet Channel 6.4.5 Downlink Dedicated Channel 6.4.6 Downlink Multiplexing 6.4.7 Downlink Shared Channel 6.4.8 Forward Access Channel for User Data Transmission 6.4.9 Channel Coding for User Data 6.4.10 Coding for TFCI Information 6.5 Signalling 6.5.1 Common Pilot Channel (CPICH) 6.5.2 Synchronisation Channel (SCH) 6.5.3 Primary Common Control Physical Channel (Primary CCPCH) 6.5.4 Secondary Common Control Physical Channel (Secondary CCPCH) 6.5.5 Random Access Channel (RACH) for Signalling Transmission 6.5.6 Acquisition Indicator Channel (AICH) 6.5.7 Paging Indicator Channel (PICH) 6.6 Physical Layer Procedures 6.6.1 Fast Closed-Loop Power Control Procedure 6.6.2 Open-Loop Power Control 6.6.3 Paging Procedure 6.6.4 RACH Procedure 6.6.5 Cell Search Procedure 6.6.6 Transmit Diversity Procedure 6.6.7 Handover Measurements Procedure 6.6.8 Compressed Mode Measurement Procedure 6.6.9 Other Measurements 6.6.10 Operation with Adaptive Antennas 6.6.11 Site Selection Diversity Transmission 6.7 Terminal Radio Access Capabilities 6.8 Conclusions References
98 103 105 106 106 109 111 112 112 114 116 116 116 118 118 118 119 119 120 121 122 122 123 123 124 124 125 126 127 128 129 131 132 133 134 137 137
7 Radio Interface Protocols
139
Jukka 7.1 7.2 7.3
Viale´n and Antti Toskala Introduction Protocol Architecture The Medium Access Control Protocol 7.3.1 MAC Layer Architecture 7.3.2 MAC Functions 7.3.3 Logical Channels
139 140 141 141 142 143
Contents
ix
7.3.4
Mapping Between Logical Channels and Transport Channels 7.3.5 Example Data Flow Through the MAC Layer 7.4 The Radio Link Control Protocol 7.4.1 RLC Layer Architecture 7.4.2 RLC Functions 7.4.3 Example Data Flow Through the RLC Layer 7.5 The Packet Data Convergence Protocol 7.5.1 PDCP Layer Architecture 7.5.2 PDCP Functions 7.6 The Broadcast/Multicast Control Protocol 7.6.1 BMC Layer Architecture 7.6.2 BMC Functions 7.7 Multimedia Broadcast Multicast Service 7.8 The Radio Resource Control Protocol 7.8.1 RRC Layer Logical Architecture 7.8.2 RRC Service States 7.8.3 RRC Functions and Signalling Procedures 7.9 Early UE Handling Principles 7.10 Improvements for Call Setup Time Reduction References
144 144 145 146 147 148 150 150 150 151 151 151 152 152 152 153 157 171 172 173
8 Radio Network Planning
175
Harri Holma, Zhi-Chun Honkasalo, Seppo Ha¨ma¨la¨inen, Jaana Laiho, Kari Sipila¨ and Achim Wacker 8.1 Introduction 8.2 Dimensioning 8.2.1 Radio Link Budgets 8.2.2 Load Factors 8.2.3 Capacity Upgrade Paths 8.2.4 Capacity per km2 8.2.5 Soft Capacity 8.2.6 Network Sharing 8.3 Capacity and Coverage Planning and Optimisation 8.3.1 Iterative Capacity and Coverage Prediction 8.3.2 Planning Tool 8.3.3 Case Study 8.3.4 Network Optimisation 8.4 GSM Co-planning 8.5 Inter-operator Interference 8.5.1 Introduction 8.5.2 Uplink versus Downlink Effects 8.5.3 Local Downlink Interference 8.5.4 Average Downlink Interference 8.5.5 Path Loss Measurements 8.5.6 Solutions to Avoid Adjacent Channel Interference
175 176 177 180 192 193 194 197 198 198 199 200 204 207 209 209 210 211 213 213 215
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8.6 8.7
WCDMA Frequency Variants UMTS Refarming to GSM900 Band 8.7.1 3GPP Blocking Requirements 8.7.2 Uncoordinated GSM900 þ UMTS900 8.7.3 Coordinated GSM900 þ UMTS900 8.7.4 Remaining GSM900 Voice Capacity References
216 217 217 217 218 220 221
9 Radio Resource Management
223
Harri Holma, Klaus Pedersen, Jussi Reunanen, Janne Laakso and Oscar Salonaho 9.1 Interference-Based Radio Resource Management 9.2 Power Control 9.2.1 Fast Power Control 9.2.2 Outer Loop Power Control 9.3 Handovers 9.3.1 Intra-frequency Handovers 9.3.2 Inter-system Handovers between WCDMA and GSM 9.3.3 Inter-frequency Handovers within WCDMA 9.3.4 Summary of Handovers 9.4 Measurement of Air Interface Load 9.4.1 Uplink Load 9.4.2 Downlink Load 9.5 Admission Control 9.5.1 Admission Control Principle 9.5.2 Wideband Power-based Admission Control Strategy 9.5.3 Throughput-Based Admission Control Strategy 9.6 Load Control (Congestion Control) References
223 224 224 231 237 237 246 250 251 253 253 255 256 256 257 259 259 260
10 Packet Scheduling
261
Jeroen Wigard, Harri Holma, Renaud Cuny, Nina Madsen, Frank Frederiksen and Martin Kristensson 10.1 Transmission Control Protocol (TCP) 10.2 Round Trip Time 10.3 User-specific Packet Scheduling 10.3.1 Common Channels (RACH/FACH) 10.3.2 Dedicated Channel (DCH) 10.3.3 Downlink Shared Channel (DSCH) 10.3.4 Uplink Common Packet Channel (CPCH) 10.3.5 Selection of Transport Channel 10.3.6 Paging Channel States 10.4 Cell-specific Packet Scheduling 10.4.1 Priorities
261 268 270 271 272 274 274 274 278 278 280
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10.4.2 10.4.3 10.5 Packet 10.5.1 10.5.2 10.6 Packet 10.6.1 10.6.2 10.6.3 10.6.4 10.6.5 References
Scheduling Algorithms Packet Scheduler in Soft Handover Data System Performance Link Level Performance System Level Performance Data Application Performance Introduction to Application Performance Person-to-person Applications Content-to-person Applications Business Connectivity Conclusions on Application Performance
281 281 283 283 284 286 287 288 292 294 297 298
11 Physical Layer Performance
299
Harri Holma, Jussi Reunanen, Leo Chan, Preben Mogensen, Klaus Pedersen, Kari Horneman, Jaakko Vihria¨la¨ and Markku Juntti 11.1 Introduction 11.2 Cell Coverage 11.2.1 Uplink Coverage 11.2.2 Downlink Coverage 11.3 Downlink Cell Capacity 11.3.1 Downlink Orthogonal Codes 11.3.2 Downlink Transmit Diversity 11.3.3 Downlink Voice Capacity 11.4 Capacity Trials 11.4.1 Single Cell Capacity Trials 11.4.2 Multicell Capacity Trials 11.4.3 Summary 11.5 3GPP Performance Requirements 11.5.1 Eb =N 0 Performance 11.5.2 RF Noise Figure 11.6 Performance Enhancements 11.6.1 Smart Antenna Solutions 11.6.2 Multiuser Detection References
299 299 302 311 312 312 317 319 321 321 335 337 339 339 342 343 343 350 359
12 High-Speed Downlink Packet Access
363
Antti Toskala, Harri Holma, Troels Kolding, Preben Mogensen, Klaus Pedersen and Jussi Reunanen 12.1 Release 99 WCDMA Downlink Packet Data Capabilities 12.2 HSDPA Concept 12.3 HSDPA Impact on Radio Access Network Architecture 12.4 Release 4 HSDPA Feasibility Study Phase 12.5 HSDPA Physical Layer Structure
363 364 366 367 367
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12.5.1 High-Speed Downlink Shared Channel (HS-DSCH) 12.5.2 High-speed Shared Control Channel (HS-SCCH) 12.5.3 Uplink High-speed Dedicated Physical Control Channel (HS-DPCCH) 12.5.4 HSDPA Physical Layer Operation Procedure 12.6 HSDPA Terminal Capability and Achievable Data Rates 12.7 Mobility with HSDPA 12.7.1 Measurement Event for Best Serving HS-DSCH Cell 12.7.2 Intra-Node B HS-DSCH to HS-DSCH Handover 12.7.3 Inter-Node–Node B HS-DSCH to HS-DSCH Handover 12.7.4 HS-DSCH to DCH Handover 12.8 HSDPA Performance 12.8.1 Factors Governing Performance 12.8.2 Spectral Efficiency, Code Efficiency and Dynamic Range 12.8.3 User Scheduling, Cell Throughput and Coverage 12.8.4 HSDPA Network Performance with Mixed Non-HSDPA and HSDPA Terminals 12.9 HSPA Link Budget 12.10 HSDPA Iub Dimensioning 12.11 HSPA Round-Trip Time 12.12 Terminal Receiver Aspects 12.13 Evolution in Release 6 12.14 Conclusions References
368 371
13 High-Speed Uplink Packet Access
403
Antti 13.1 13.2 13.3 13.4 13.5 13.6
13.7 13.8 13.9
Toskala, Harri Holma and Karri Ranta-aho Release 99 WCDMA Downlink Packet Data Capabilities HSUPA Concept HSUPA Impact on Radio Access Network Architecture 13.3.1 HSUPA Iub operation HSUPA Feasibility Study Phase HSUPA Physical Layer Structure E-DCH and Related Control Channels 13.6.1 Enhanced Dedicated Physical Data Channel (E-DPDCH) 13.6.2 Enhanced Dedicated Physical Control Channel (E-DPCCH) 13.6.3 E-DCH Hybrid ARQ Indicator Channel (E-HICH) 13.6.4 E-DCH Relative Grant Channel (E-RGCH) 13.6.5 E-DCH Absolute Grant Channel (E-AGCH) HSUPA Physical Layer Operation Procedure 13.7.1 HSUPA and HSDPA Simultaneous Operation HSUPA Terminal Capability HSUPA Performance 13.9.1 Increased Data Rates 13.9.2 Physical Layer Retransmission Combining 13.9.3 Node B-Based Scheduling
373 374 376 377 378 378 380 381 382 382 383 386 391 393 396 397 397 399 401 401
403 404 405 407 407 408 408 408 410 411 412 412 413 414 416 416 417 417 417
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13.9.4 HSUPA Link Budget Impact 13.9.5 Delay and QoS 13.9.6 Overall Capacity 13.10 Conclusions References
419 419 419 421 421
14 Multimedia Broadcast Multicast Service
423
Harri 14.1 14.2 14.3 14.4
Holma, Martin Kristensson and Jorma Kaikkonen Multimedia Broadcast Multicast Service Concept MBMS Impact on Network Architecture High-Level MBMS Procedures MBMS Radio Interface Channel Structure 14.4.1 Logical Channels 14.4.2 Transport Channels 14.4.3 Physical Channels 14.4.4 Point-to-Point and Point-to-Multipoint Connections 14.4.5 Example Radio Interface Procedure During MBMS Session Start 14.5 MBMS Terminal Capability 14.5.1 Selective Combining and Soft Combining 14.6 MBMS Performance 14.6.1 The 3GPP Performance Requirements 14.6.2 Simulated MBMS Cell Capacity 14.6.3 Iub Transport Capacity 14.7 MBMS Deployment and Use Cases 14.8 Benchmarking of MBMS with DVB-H 14.9 3GPP MBMS Evolution in Release 7 14.10 Summary References
15 High-Speed Packet Access Evolution (HSPA+) in 3GPP Release 7 Harri Holma, Antti Toskala, Karri Ranta-aho, Juho Pirskanen and Jorma Kaikkonen 15.1 Introduction 15.2 Setup Time Reduction 15.3 Peak Data Rate Increase with MIMO and 16QAM/64QAM 15.4 Layer 2 Optimisation 15.5 Throughput Evolution with Enhanced Terminals 15.6 Mobile Power Consumption Reduction with Continuous Packet Connectivity 15.7 Voice-over-IP (VoIP) Capacity Enhancements 15.8 Flat Architecture 15.9 Summary References
423 426 428 430 430 430 430 431 432 433 433 435 435 436 438 439 440 440 442 442
445
445 445 448 450 453 456 458 459 461 461
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Contents
16 UTRAN Long-Term Evolution Antti 16.1 16.2 16.3 16.4
463
Toskala and Harri Holma Background Multiple Access and Architecture Decisions LTE Impact on Network Architecture LTE Multiple Access 16.4.1 OFDMA Principles 16.4.2 SC-FDMA Principles 16.5 LTE Physical Layer Design and Parameters 16.6 LTE Protocols 16.7 Performance 16.7.1 Peak Bit Rates 16.7.2 Spectral Efficiency 16.7.3 Link Budget and Coverage 16.8 Summary References
463 464 466 467 467 470 473 475 477 477 478 480 483 483
17 UTRA TDD Modes
485
Antti Toskala, Harri Holma, Otto Lehtinen and Heli Va¨a¨ta¨ja¨ 17.1 Introduction 17.1.1 Time Division Duplex (TDD) 17.1.2 Differences in the Network-Level Architecture. 17.2 UTRA TDD Physical Layer 17.2.1 Transport and Physical Channels 17.2.2 Modulation and Spreading 17.2.3 Physical Channel Structures, Slot and Frame Format 17.2.4 UTRA TDD Physical Layer Procedures 17.3 UTRA TDD Interference Evaluation 17.3.1 TDD–TDD Interference 17.3.2 TDD and FDD Coexistence 17.3.3 Unlicensed TDD Operation 17.3.4 Conclusions on UTRA TDD Interference 17.4 HSDPA Operation with TDD 17.5 Release 7 TDD Enhancements 17.6 Concluding Remarks and Future Outlook on UTRA TDD References
485 486 487 488 489 489 490 495 499 499 501 503 503 504 505 505 506
18 Terminal Radio-Frequency Design Challenges
507
Laurent Noe¨l, Antti Toskala and Dominique Brunel 18.1 Introduction 18.2 Transmitter Chain System Design Challenges 18.2.1 The Adjacent Channel Leakage Ratio/Power Consumption Trade-off 18.2.2 Phase Discontinuity
507 509 509 514
Contents
18.3
xv
Receiver Chain Design Challenges 18.3.1 UE Reference Sensitivity System Requirements 18.3.2 Inter-Operator Interference 18.3.3 Impact of RF Impairments on HSDPA System Performance 18.4 Multi-Mode/Band Challenges 18.4.1 From Mono-Mode/Mono-Band to Multi-Mode/Multi-Band and Diversity 18.4.2 New Requirements due to Coexistence 18.4.3 Front-End Integration Strategies and Design Trends 18.4.4 Impact on Today’s Architectures 18.5 Conclusions References
515 516 523 526 527 527 527 529 531 532 532
Index
535
Preface Second-generation telecommunication systems, such as the Global System for Mobile Communications (GSM), enabled voice traffic to go wireless: the number of mobile phones exceeds the number of landline phones and the mobile phone penetration is approaching 100% in several markets. The data-handling capabilities of second-generation systems are limited, however, and third-generation systems are needed to provide the high bit-rate services that enable high-quality images and video to be transmitted and received, and to provide access to the Web with higher data rates. These third-generation mobile communication systems are referred to in this book as the Universal Mobile Telecommunication System (UMTS). Wideband Code Division Multiple Access (WCDMA) is the main thirdgeneration air interface in the world, and deployment has been started in Europe and Asia, including Japan and Korea, in the same frequency band, around 2 GHz. WCDMA has also been deployed in the USA in the US frequency bands. During writing of this 4th edition, the number of WCDMA subscribers has exceeded 130 million globally in 150 commercial networks. And most of those networks have already launched the next phase of WCDMA, i.e. High-Speed Downlink Packet Access (HSDPA). The growth continues, as there are more commercial networks, more terminals across all categories and more data services being deployed. The large market for WCDMA and its flexible multimedia capabilities will create new business opportunities for manufacturers, operators, and the providers of content and applications. This book gives a detailed description of the WCDMA air interface and its utilisation. The contents are summarised in Figure 1. The book is structured as follows. Chapters 1–4 provide an introduction to the technology and its standardisation. Chapters 5–7 give a detailed presentation of the WCDMA standard, and Chapters 8–11 cover the utilisation of the standard and its performance. Chapters 12–17 present WCDMA evolution. Chapter 18 covers terminal radio-frequency (RF) design challenges. Chapter 1 introduces the third-generation air interfaces, the spectrum allocation, the time schedule, and the main differences from second-generation air interfaces. Chapter 2 presents example UMTS applications, concept phones and the quality-of-service classes. Chapter 3 introduces the principles of the WCDMA air interface, including spreading, Rake receiver, power control and handovers. Chapter 4 presents the background to WCDMA, the global harmonisation process and the standardisation. Chapter 5 describes the architecture of the radio access network, interfaces within the radio access network between base stations and radio network controllers, and the interface between the radio access network and the core network. Chapter 6 covers the physical layer (layer 1), including spreading, modulation, user data and signalling transmission, and the main physical layer procedures of power control,
xviii
Preface
Figure 1. Contents of this book
paging, transmission diversity and handover measurements. Chapter 7 introduces the radio interface protocols, consisting of the data link layer (layer 2) and the network layer (layer 3). Chapter 8 presents the guidelines for radio network dimensioning, gives an example of detailed capacity and coverage planning, and covers GSM co-planning. Chapter 9 covers the radio resource management algorithms that guarantee the efficient utilisation of the air interface resources and the quality of service. These algorithms are power control, handovers, admission and load control. Chapter 10 depicts packet access and presents the performance of packet protocols of WCDMA. Chapter 11 analyses the coverage and capacity of the WCDMA air interface. Chapter 12 presents the significant Release 5 feature HSDPA, and Chapter 13 the corresponding uplink counterpart High-Speed Uplink Packet Access (HSUPA) in Release 6. Chapter 14 presents Multimedia Broadcast Multicast System (MBMS). Chapter 15 introduces High-Speed Packet Access (HSPA) Evolution (HSPAþ) in Release 7 and Chapter 16 Long-Term Evolution (LTE) in Release 8. The time division duplex mode of the WCDMA air interface is discussed in Chapter 17. The terminal design aspects are presented in Chapter 18. In the earlier editions of this book there was also a chapter on cdma2000 multi-carrier operation, but that chapter has been omitted from this edition as this is no longer under practical consideration for deployment. The 2nd edition contained coverage of the recently introduced key features of 3rd Generation Partnership Project (3GPP) Release 5 specifications, such as HSDPA and Internet Protocol Multimedia Subsystem. The 3rd edition of the book continues to deepen the coverage of several existing topics both based on the field experiences and based on more detailed simulation studies. The 3rd edition covers the main updates in 3GPP standard Release 6. The 4th edition covers in detail 3GPP Release 6 features including HSUPA in Chapter 13 and MBMS in Chapter 14. 3GPP Release 7 has brought HSPAþ, covering an extensive set of
Preface
xix
improvements on top of earlier releases. HSPAþ is described in Chapter 15. 3GPP is currently working on Release 8 LTE, which is described in Chapter 16. During the writing of the book, Release 7 was in practise finalised, while Release 8 was under heavy standardisation work. Chapter 18 was added with 4th edition covering terminal RF design aspects. This book is aimed at operators, network and terminal manufacturers, service providers, university students and frequency regulators. A deep understanding of the WCDMA air interface, its capabilities and its optimal usage is the key to success in the UMTS business. This book represents the views and opinions of the authors, and does not necessarily represent the views of their employers.
Acknowledgements The editors would like to acknowledge the time and effort put in by their colleagues in contributing to this book. Besides the editors, the contributors were Dominique Brunel, Leo Chan, Renaud Cuny, Frank Frederiksen, Zhi-Chun Honkasalo, Seppo Ha¨ma¨la¨inen, Markku Juntti, Jorma Kaikkonen, Troels Kolding, Martin Kristensson, Janne Laakso, Jaana Laiho, Ukko Lappalainen, Otto Lehtinen, Fabio Longoni, Atte La¨nsisalmi, Nina Madsen, Preben Mogensen, Peter Muszynski, Laurent Noel, Klaus Pedersen, Juho Pirskanen, Karri Ranta-aho, Jussi Reunanen, Oscar Salonaho, Jouni Salonen, Kari Sipila¨, Jukka Vialen, Heli Va¨a¨ta¨ja¨, Jaakko Vihria¨la¨, Achim Wacker, Jeroen Wigard and Juha Ylitalo. While we were developing this book, many of our colleagues from Nokia and Nokia Siemens Networks offered their help in suggesting improvements and finding errors. Also, a number of colleagues from other companies have helped us in improving the quality of the book. The editors are grateful for the comments received from Heikki Ahava, Erkka Ala-Tauriala, David Astely, Erkki Autio, Mattew Baker, Luis Barreto, Kai Heikkinen, Kari Heiska, Kimmo Hiltunen, Klaus Hugl, Alberg Ho¨glund, Kaisu Iisakkila, Ann-Louise Johansson, Susanna Kallio, Istvan Kovacs, Ilkka Keskitalo, Pasi Kinnunen, Tero Kola, Petri Komulainen, Lauri Laitinen, Anne Leino, Arto Leppisaari, Pertti Lukander, Esko Luttinen, Jonathan Moss, Jari Ma¨kinen, Olli Nurminen, Tero Ojanpera¨, Lauri Oksanen, Kari Pajukoski, Kari Pehkonen, Mika Rinne, David Soldani, Rauno Ruisma¨ki, Pekka Talmola, Kimmo Tera¨va¨, Mitch Tseng, Antti To¨lli, Veli Voipio and Helen Wait. The team at John Wiley & Sons participating in the production of this book provided excellent support and worked hard to keep the demanding schedule. The editors especially would like to thank Sarah Hinton and Mark Hammond for assistance with practical issues in the production process, and especially the copy-editor, for his efforts in smoothing out the engineering approach to the English language expressions. We are extremely grateful to our families, as well as the families of all the authors, for their patience and support, especially during the late night and weekend editing sessions near different production milestones. Special thanks are due to our employer, Nokia Siemens Networks, for supporting and encouraging such an effort and for providing some of the illustrations in this book. Finally, we would like to acknowledge the efforts of our colleagues in the wireless industry for the great work done within the 3rd Generation Partnership Project (3GPP) to produce the global WCDMA standard in merely a year and thus to create the framework for this book. Without such an initiative this book would never have been possible. The editors and authors welcome any comments and suggestions for improvements or changes that could be implemented in forthcoming editions of this book. The feedback is welcome to the editors’ e-mail addresses [email protected] and [email protected].
Abbreviations 3GPP 3GPP2 AAL2 AAL5 ACELP ACIR ACK ACIR ACLR ACTS ADC AGC A-GW AICH ALCAP AM AM AMD AMR AMR-NB AMR-WB ARIB AOL APN ARP ARQ ASC ASN.1 ATM AWGN BB BB SS7 BCCH
3rd Generation Partnership Project (produces WCDMA standard) 3rd Generation Partnership Project 2 (produced cdma2000 standard) ATM Adaptation Layer type 2 ATM Adaptation Layer type 5 Algebraic code excitation linear prediction Adjacent channel interference ratio, caused by the transmitter non-idealities and imperfect receiver filtering Acknowledgement Adjacent channel interference ratio Adjacent channel leakage ratio, caused by the transmitter non-idealities, the effect of receiver filtering is not included Advanced communication technologies and systems, EU research projects framework Analogue-to-Digital Converter Adaptive Gain Control Access gateway Acquisition indication channel Access link control application part Acknowledged mode Amplitude modulation Acknowledged mode data Adaptive Multirate (speech codec) Narrowband AMR Wideband AMR Association of Radio Industries and Businesses (Japan) America On-line Access point name Allocation and retention priority Automatic repeat request Access service class Abstract syntax notation one Asynchronous transfer mode Additive white Gaussian noise Baseband Broad band signalling system #7 Broadcast channel (logical channel)
xxiv
BCH BCFE BCH BER BLER BMC BM-SC BoD BPF BPSK BS BSS BSC CA-ICH CB CBC CBS CCCH CCH CCH CCSA CD-ICH CDF CDMA CFN CIF CIR CM CM CMOS CN C-NBAP CODIT CP CPC CPCH CPICH CQI CRC CRNC C-RNTI CS CSCF CSICH CTCH CWTS DAB
Abbreviations
Broadcast channel (transport channel) Broadcast control functional entity Broadcast channel (transport channel) Bit error rate Block error rate Broadcast/multicast control protocol Broadcast multicast service centre Bandwidth on demand Band-pass Filter Binary phase shift keying Base station Base station subsystem Base station controller Channel assignment indication channel Cell broadcast Cell broadcast centre Cell broadcast service Common control channel (logical channel) Common transport channel Control channel China Communications Standards Association Collision detection indication channel Cumulative distribution function Code division multiple access Connection frame number Common intermediate format Carrier to interference ratio Connection management Cubic metric Complementary metal–oxide–semiconductor Core network Common NBAP Code division test bed, EU research project Compression point Continuous packet connectivity Common packet channel Common pilot channel Channel quality indicator Cyclic redundancy check Controlling RNC Cell-RNTI, radio network temporary identity Circuit Switched Call state control function CPCH status indication channel Common traffic channel China wireless telecommunications standard group Digital audio broadcasting
xxv
Abbreviations
DAC DC DCA DCA DCCH DCFE DCH DCR DCS DECT DF DL DMB D-NBAP DNS DPCCH DPDCH DRM DRNC DRX DS-CDMA DSCH DSL DSP DTCH DTX DVB-T/H E-AGCH E-DCH E-DPCCH E-DPDCH EDGE EFR EGSM E-HICH EIRP EP E-RGCH ETSI E-UTRAN EVM FACH FBI FCC FCS FDD FDMA
Digital-to-Analogue Converter Direct Current Dynamic channel allocation Direct Conversion Architecture Dedicated control channel (logical channel) Dedicated control functional entity Dedicated channel (transport channel) Direct conversion receiver Digital cellular system (GSM 1800) Digital enhanced cordless telephone Decision feedback Downlink Digital multimedia broadcasting Dedicated NBAP Domain name system Dedicated physical control channel Dedicated physical data channel Digital Radio Mondiale Drift RNC Discontinuous reception Direct spread code division multiple access Downlink shared channel Digital subscriber line Digital Signal Processing Dedicated traffic channel Discontinuous transmission Digital video broadcast terrestrial / handheld E-DCH absolute grant channel Enhanced uplink DCH Enhanced dedicated physical control channel Enhanced dedicated physical data channel Enhanced data rates for GSM evolution Enhance full rate Extended GSM E-DCH hybrid ARQ indicator channel Equivalent isotropic radiated power Elementary Procedure E-DCH relative grant channel European Telecommunications Standards Institute Evolved UTRAN Error Vector Magnitude Forward access channel Feedback information Federal communication commission Fast cell selection Frequency division duplex Frequency division multiple access
xxvi
FDPCH FER FFT FM FP FRAMES FTP GERAN GFSK GGSN GMSC GMSK GPRS GPS GSIC GSM GTP-U GW HARQ HLR HP HPF HPSK H-RNTI HSDPA HS-DPCCH HS-DSCH HS-SCCH HSUPA HSS HTTP HW IC IC ID IETF IFFT IL IMD IMEISV IMS IMSI IMT-2000 IN IP IPDL IPI
Abbreviations
Fractional DPCH Frame error ratio Fast Fourier transform Frequency modulation Frame protocol Future radio wideband multiple access system, EU research project File transfer protocol GSM/EDGE Radio Access Network Gaussian Frequency Shift Keying Gateway GPRS support node Gateway MSC Gaussian Minimum Shift Keying General packet radio system Global positioning system Groupwise serial interference cancellation Global system for mobile communications User-plane part of GPRS tunnelling protocol Gateway Hybrid automatic repeat request Home location register High Power High pass filter Hybrid PSK HS-DSCH Radio Network Temporary Identity High-speed downlink packet access Uplink High-Speed Dedicated Physical Control Channel High-Speed Downlink Shared Channel High-Speed Shared Control Channel High-Speed Uplink Packet access Home subscriber server Hypertext transfer protocol Hardware Interference cancellation Integrated Circuit Identity Internet engineering task force Inverse Fast Fourier transform Insertion loss Inter-modulation distortion International Mobile Station Equipment Identity and Software Version IP multimedia subsystem International mobile subscriber identity International mobile telephony, third-generation networks are referred to as IMT-2000 within ITU Intelligent network Internet protocol Idle periods in downlink Inter-path interference
Abbreviations
xxvii
IRC IS-2000 IS-136 IS-95 ISDN ISI ITU ITUN Iu BC L2 LAI LAN LCS LCD LO LP LPF LTE MAC MAI MAP MBMS MCCH MCS MCU ME MF MGCF MGW MHA MIMO MLSD MM MME MMS MMSE MOS MPEG MR-ACELP MRF MS MSC/VLR MSCH MSN MT MTCH MTP3b MUD
Interference rejection combining IS-95 evolution standard, (cdma2000) US-TDMA, one of the second-generation systems, mainly in Americas cdmaOne, one of the second-generation systems, mainly in Americas and in Korea Integrated services digital network Inter-symbol interference International telecommunications union SS7 ISUP Tunnelling Iu broadcast Layer 2 Location area identity Local area network Location services Liquid crystal display Local oscillator Low pass Low-pass filter Long-Term Evolution Medium access control Multiple access interference Maximum a posteriori Multimedia broadcast multicast service MBMS point-to-multipoint control channel Modulation and coding scheme Multipoint control unit Mobile equipment Matched filter Media gateway control function Media gateway Mast head amplifier Multiple input multiple output Maximum likelihood sequence detection Mobility management Mobility management entity Multimedia message Minimum mean square error Mean opinion score Motion picture experts group Multirate ACELP Media resource function Mobile station Mobile services switching centre/visitor location register MBMS scheduling channel Microsoft network Mobile termination MBMS point-to-multipoint control channel Message transfer part (broadband) Multiuser detection
xxviii
NADC NAS NBAP NMT NRT OCNS ODMA OFDMA O&M OSS OTDOA OVSF PA PAD PAE PAR PBCH PC PCB PCCC PCCCH PCCH PCCPCH PCH PCPCH PCRF PCS PDC PDCP PDN PDP PDSCH PDU PEP PER PF PHY PI PIC PICH PLL PLMN PM PNFE POC PRACH PS
Abbreviations
North American Digital Cellular Non access stratum Node B application part Nordic Mobile Telephone Non-real time Orthogonal channel noise simulator Opportunity driven multiple access Orthogonal frequency division multiple access Operation and maintenance Operations support system Observed time difference of arrival Orthogonal variable spreading factor Power Amplifier Padding Power added efficiency Peak-to-average ratio Physical Broadcast Channel Power control Printed Circuit Board Parallel concatenated convolutional coder Physical common control channel Paging channel (logical channel) Primary common control physical channel Paging channel (transport channel) Physical common packet channel Policy and Charging Rules Function Persona communication systems, second-generation cellular systems mainly in Americas, operating partly on IMT-2000 band Personal digital cellular, second-generation system in Japan Packet data converge protocol Public data network Packet data protocol Physical downlink shared channel Protocol data unit Performance enhancement proxy Packed encoding rules Proportional fair Physical layer Page indicator Parallel interference cancellation Paging indicator channel Phase Locked Loop Public land mobile network Phase Modulation Paging and notification control function entity Push-to-talk over cellular Physical random access channel Packet switched
xxix
Abbreviations
PSCH PSTN P-TMSI PU PUCCH PUSCH PDCCH PVC QAM QCIF QoS QPSK QVGA RAB RACH RAI RAN RANAP RB RF RLC RMC RNC RNS RNSAP RNTI ROHC RR RRC RRC RRM RSN RSSI RSVP RT RTCP RTP RTSP RU SAAL-NNI SAAL-UNI SABP SAE SAP SAP SAS SAW SCCP
Physical shared channel Public switched telephone network Packet-TMSI Payload unit Physical uplink control channel Physical uplink shared channel Physical downlink control channel Predefined Virtual Connection Quadrature amplitude modulation Quarter common intermediate format Quality of service Quadrature phase shift keying Quarter video graphics array Radio access bearer Random access channel Routing area identity Radio access network RAN application part Radio bearer Radio frequency Radio link control Reference measurement channel Radio network controller Radio network subsystem RNS application part Radio network temporary identity Robust header compression Round robin Radio resource control Root-raised cosine Radio resource management Retransmission sequence number Received signal strength indicator Resource reservation protocol Real time Real-time transport control protocol Real-time protocol Real-time streaming protocol Resource unit Signalling ATM adaptation layer for network to network interfaces Signalling ATM adaptation layer for user to network interfaces Service Area Broadcast Protocol System architecture evolution Service access point Session announcement protocol Stand-alone SMLC Surface Acoustic Wave Signalling connection control part
xxx
SCCPCH SC-FDMA SCH SCTP SDD SDP SDU SEQ SF SFN SGSN SIP SHO SIB SIC SID SINR SIP SIR SM SMS SMLC SN SNR SPDT SQ-PIC SRB SRNC SRNS SS7 SSCF SSCOP SSDT STD STTD TCH TCP TCTF TD/CDMA TDD TDMA TD-SCDMA TE TF TFCI TFCS TFI
Abbreviations
Secondary common control physical channel Single carrier frequency division multiple access Synchronisation channel Simple control transmission protocol Space division duplex Session description protocol Service data unit Sequence Spreading Factor System frame number Serving GPRS support node Session initiation protocol Soft handover System information block Successive interference cancellation Silence indicator Signal-to-noise ratio where noise includes both thermal noise and interference Session initiation protocol Signal-to-interference ratio Session management Short message service Serving mobile location centre Sequence number Signal-to-noise ratio Single pole double throw Soft quantised parallel interference cancellation Signalling radio bearer Serving RNC Serving RNS Signalling System #7 Service-specific coordination function Service-specific connection oriented protocol Site selection diversity transmission Switched transmit diversity Space time transmit diversity Traffic channel Transport control protocol Target channel type field Time division CDMA, combined TDMA and CDMA Time division duplex Time division multiple access Time division synchronous CDMA, 1.28 Mcps TDD Terminal equipment Transport format Transport format combination indicator Transport format combination set Transport format indicator
xxxi
Abbreviations
TFRC THP TMGI TMSI TPC TR TS TSTD TTA TTC TTI TxAA UDP UE UHF UL UM UMD UMTS URA URL U-RNTI USCH USIM US-TDMA UTRA UTRA UTRAN UWB VAD VCO VoIP VPN WAP WARC WCDMA WiFi WiMAX WLAN WLL WML WPAN WWW XHTML ZF
Transport format and resource combination Traffic handling priority Temporary mobile group identity Temporary mobile subscriber identity Transmission power control Transparent mode Technical specification Time switched transmit diversity Telecommunications Technology Association (Korea) Telecommunication Technology Commission (Japan) Transmission time interval Transmit adaptive antennas User datagram protocol User equipment Ultra High Frequency Uplink Unacknowledged mode Unacknowledged mode data Universal mobile telecommunication services UTRAN registration area Universal resource locator UTRAN RNTI Uplink shared channel UMTS subscriber identity module IS-136, one of the second-generation systems mainly in USA UMTS Terrestrial radio access (ETSI) Universal Terrestrial radio access (3GPP) UMTS Terrestrial radio access network Ultrawideband Voice activation detection Voltage Controlled Oscillator Voice over IP Virtual private network Wireless application protocol World administrative radio conference Wideband CDMA, Code division multiple access Wireless fidelity (WLAN based on IEEE 802.11) World Wide Interoperability for Microwave Access Wireless local area network Wireless local loop Wireless markup language Wireless personal area network World Wide Web Extensible hypertext markup language Zero forcing
1 Introduction Harri Holma and Antti Toskala
1.1
WCDMA in Third-Generation Systems
Analog cellular systems are commonly referred to as first-generation systems. The digital systems, such as Global System for Mobile Communications (GSM), PDC, cdmaOne (IS-95) and US-TDMA (IS-136), are second-generation systems. These systems have enabled voice communications to go wireless in many of the leading markets, and customers are also increasingly finding value in other services, such as text messaging and internet access, which are starting to grow rapidly. Third-generation systems are designed for multimedia communication: with these, person-to-person communication can be enhanced with high-quality images and video, and access to information and services on public and private networks will be enhanced by the higher data rates and new flexible communication capabilities of third-generation systems. This, together with the continuing evolution of the second-generation systems, will create new business opportunities not only for manufacturers and operators, but also for the providers of content and applications using these networks. In the standardisation forums, Wideband Code Division Multiple Access (WCDMA) technology has emerged as the most widely adopted third-generation air interface. Its specification has been created in the 3rd Generation Partnership Project (3GPP), which is the joint standardisation project of the standardisation bodies from Europe, Japan, Korea, the USA and China. Within 3GPP, WCDMA is called Universal Terrestrial Radio Access (UTRA) Frequency Division Duplex (FDD) and Time Division Duplex (TDD), the term WCDMA being used to cover both FDD and TDD operations. Throughout this book, the chapters related to specifications use the 3GPP terms UTRA FDD and TDD, the others use the term WCDMA. This book focuses on the WCDMA FDD technology. The WCDMA TDD mode and its differences from the WCDMA FDD mode are presented in Chapter 17, including a description of TD-SCDMA. UTRA is the radio access part of the Universal Mobile Telephone System (UMTS) network.
WCDMA for UMTS – HSPA Evolution and LTE, fourth edition. and Antti Toskala # 2007 John Wiley & Sons, Ltd
Edited by Harri Holma
2
1.2
WCDMA for UMTS – HSPA Evolution and LTE
Spectrum Allocations for Third-Generation Systems
Work to develop third-generation mobile systems started when the World Administrative Radio Conference (WARC) of the International Telecommunications Union (ITU), at its 1992 meeting, identified the frequencies around 2 GHz that were available for use by future International Mobile Telephony 2000 (IMT-2000) mobile systems, both terrestrial and satellite. Within the IMT-2000 framework, five air interfaces are defined for third-generation systems, based on either CDMA or TDMA technology, as described in Chapter 3. The original target of the third-generation process was a single global IMT-2000 air interface. In practice, the third-generation systems are closer to this target than were second-generation systems, since WCDMA has clearly turned out to be the most dominant IMT-2000 standard in commercial deployments. The same WCDMA air interface is deployed in Europe, Asia, Australia, in North and South America and in Africa. Most of the WCDMA deployments use the identified IMT-2000 spectrum around 2 GHz: 1920–1980 MHz for uplink and 2110–2170 MHz for downlink. This spectrum is in IMT2000 use in Europe, Asia (including Japan and Korea) and in Brazil. The first licences for that spectrum were granted in Finland in March 1999, followed by Spain in March 2000. No auction was conducted in Finland or in Spain. Also, Sweden granted the licences without auction in December 2000. However, in other countries, such as the UK, Germany and Italy, an auction similar to the US Personal Communication Services (PCS) spectrum auctions was conducted. WCDMA will also be deployed in the existing second-generation frequency bands that were also identified for IMT-2000 in WRC-2000 and are currently used by GSM or cdma. That approach is called refarming. The WCDMA deployment in the USA started by refarming WCDMA to the existing cellular bands at 850 MHz and to the PCS band at 1900 MHz, since there were no new frequencies available for WCDMA deployment. A new frequency band was auctioned in the USA in 2006. This band, the so-called Advanced Wireless Services (AWS) band, is located at 1700 MHz for uplink and at 2100 MHz for downlink. WCDMA deployment at that band has already started. The AWS band uplink happens to be within the GSM1800 uplink band and the downlink is within the UMTS downlink band. WCDMA refarming to GSM bands has also started in Europe and in Asia. The GSM900 band is attractive, since a lower frequency can provide better coverage than the IMT-2000
Figure 1.1. Frequency allocation around 2 GHz
3
Introduction
Figure 1.2. Frequency allocation around 800–900 MHz
band at 2 GHz. The coexistence of GSM and WCDMA in the same frequency band needs to be taken into account in the network planning. This is considered in Section 8.6. The new IMT-2000 band around 2.6 GHz with a total 190 MHz spectrum will soon be available for the deployment of IMT-2000 and other mobile systems. In Europe, the spectrum includes 2 70 MHz for FDD systems and 50 MHz in the middle gap that could be used, for example, for TDD. The same 2.6 GHz spectrum is also available for mobile usage, including IMT-2000, in the USA. The band is already licensed and is likely to be used for TDD deployments due to the operator spectrum being more in line with the TDD arrangement. The IMT-2000 mobile spectrum around 2 GHz is presented in Figure 1.1, around 800–900 MHz in Figure 1.2 and at 2.6 GHz in Figure 1.3. The WCDMA system is specified in 3GPP for all the frequency bands shown in these figures. There are further frequencies where WCDMA specification and deployment are also considered, including the 700 MHz band in USA, the 2.3 GHz (Wireless Communication Services (WCS) band in the USA and part of the existing broadcast frequencies between 400 and 700 MHz.
1.3
Requirements for Third-Generation Systems
The second-generation systems were built mainly to provide speech services in macro cells. To understand the background to the differences between second- and third-generation systems, we need to look at the new requirements of the third-generation systems, which are listed below: Bit rates up to 2 Mbps; Variable bit rate to offer bandwidth on demand;
Figure 1.3. Frequency allocation around 2.6 GHz
4
WCDMA for UMTS – HSPA Evolution and LTE Table 1.1. Main differences between WCDMA and GSM networks WCDMA/HSPA
Carrier spacing Frequency reuse factor Frequency diversity Power control frequency Circuit and packet switched protocols Packet scheduling and retransmission control Network architecture
MIMO Downlink modulation
5 MHz 1 5 MHz bandwidth gives multipath diversity with Rake receiver Up to 1500 Hz Same protocols in radio network In base station (HSPA)
GSM/EDGE 200 kHz 1–18 Frequency hopping with frequency diversity Up to 2 Hz Different protocols In base station controller
Flat architecture with two network elements in user plane (HSPA Release 7)
Four network elements in user plane
2 2 MIMO in downlink QPSK, 16QAM, 64QAM (HSPA Release 7)
– GMSK, 8PSK
Multiplexing of services with different quality requirements on a single connection, e.g. speech, video and packet data; Delay requirements from delay-sensitive real-time traffic to flexible best-effort packet data; Quality requirements from 10% frame error rate to 10–6 bit error rate; Coexistence of second- and third-generation systems and inter-system handovers for coverage enhancements and load balancing; Support of asymmetric uplink and downlink traffic, e.g. web browsing causes more loading to downlink than to uplink; High spectrum efficiency. Table 1.1 lists the main differences between WCDMA/High Speed Packet Access (HSPA) and GSM/Enhanced Data Rates for GSM Evolution (EDGE) networks. The differences reflect the new requirements of the third-generation systems. For example, the larger bandwidth of 5 MHz is needed to support higher bit rates. HSPA Release 7 has also added a Multiple Input Multiple Output (MIMO) multi-antenna solution and higher order modulation 64QAM to support even higher data rates. HSPA pushes more functionalities to the base station and allows flat architecture, which improves the efficiency and the Quality of Service (QoS) capabilities for packet services.
1.4
WCDMA and its Evolution
European research work on WCDMA was initiated in the European Union research projects CODIT [1] and FRAMES [2] and within large European wireless communications
5
Introduction
Figure 1.4. Standardisation and commercial operation schedule for WCDMA and its evolution
companies at the start of the 1990s [3]. Those projects also produced WCDMA trial systems to evaluate link performance [4] and generated the basic understanding of WCDMA necessary for standardisation. In January 1998 the European standardisation body ETSI decided upon WCDMA as the third-generation air interface [5]. Detailed standardisation work has been carried out as part of the 3GPP standardisation process. The first full set of specifications was completed at the end of 1999, called Release 99. The first commercial network was opened in Japan during 2001 for commercial use in key areas, and in Europe at the beginning of 2002 for the pre-commercial testing phase and for commercial use during 2003. 3GPP specified important evolution steps on top of WCDMA: HSPA for downlink in Release 5 and for uplink Release 6. The downlink solution, High Speed Downlink Packet Access (HSDPA) was commercially deployed in 2005 and the uplink counterpart, High Speed Uplink Packet Access (HSUPA), during 2007. Further HSPA evolution is specified in 3GPP Release 7, and its commercial deployment is expected by 2009. HSPA evolution is also known as HSPAþ. 3GPP is also working to specify a new radio system called Long-Term Evolution (LTE), where the target for finalizing 3GPP standardization is during 2007. Release-7 and -8 solutions for HSPA evolution will be worked in parallel with LTE development, and some aspects of LTE work are also expected to reflect on HSPA evolution. HSPA, its evolution and LTE are covered in this book. The schedule for 3GPP standardization and for commercial deployment is illustrated in Figure 1.4. The peak data rate evolution for WCDMA is illustrated in Figure 1.5. WCDMA Release 99 in theory enabled 2 Mbps, but in practice gave 384 kbps. HSPA in Release 5 and Release 6 pushes the peak rates to 14 Mbps in downlink and 5.7 Mbps in uplink. HSPA evolution in Release 7 brings a maximum 28 Mbps in downlink and 11 Mbps in uplink. LTE will then further push the peak rates beyond 100 Mbps in downlink and 50 Mbps in uplink by using a 20 MHz bandwidth.
3GPP R99
3GPP R5
3GPP R6
3GPP R7
te eak ra
link p
Down
0.4 Mbps 0.4 Mbps
14 Mbps 0.4 Mbps
28 Mbps
14 Mbps
3GPP R8
LTE: 160 Mbps HSPA: 42 Mbps1 LTE: 50 Mbps
11 Mbps
5.7 Mbps k rate
k pea
Uplin
1
Assuming 2 × 2 MIMO together with 64QAM
Figure 1.5. Peak data rate evolution for WCDMA
6
WCDMA for UMTS – HSPA Evolution and LTE
GSM
EDGE WCDMA
EDGE evol uti on HSPA
HSPA evolution (HSPA+) LTE
cdma
EV-DO
Figure 1.6. System evolution
1.5
System Evolution
GSM and WCDMA together account for 85% of the global mobile subscriptions, and their share keeps increasing. WCDMA is designed for coexistence with GSM, including seamless handovers and dual-mode handsets. Most of WCDMA networks are deployed on top of the existing GSM network. GSM, EDGE and EDGE evolution can be efficiently deployed together with WCDMA and its evolution. In the same way, LTE is designed for coexistence with GSM and WCDMA. The cdma2000 market share globally in terms of mobile subscribers has been decreasing since 2004 and is currently slightly above 10%. A number of major cdma operators are turning to GSM/WCDMA for voice evolution to get access to the benefits of the large and open GSM/WCDMA ecosystem and economics of scale for low-cost mobile devices. The cdma data solution EV-DO is currently deployed commercially. For further data evolution a number of cdma operators are looking for HSPA or Worldwide Interoperability for Microwave Access (WiMAX) in the short term and 3GPP LTE in the long run. The high-level system evolution is illustrated in Figure 1.6. Looking back at the history of GSM, we note that, since the opening of the first GSM network in July 1991 (Radiolinja, Finland), several countries have reached more than 80% cellular phone penetration and the global GSM subscriber count exceeded 2 billion in 2006 – 15 years after the opening of the first network. The global cellular phone penetration currently stands at over 40%, with more than 1 million new GSM subscribers signing every day. Early GSM experiences showed that growth rates were very high once there were small-sized attractive terminals available with low power consumption. WCDMA is foreseen to follow the same trend. It took 7 years for GSM to reach 100 million subscribers and less than 6 years for WCDMA. Currently, there are more than 150 commercial WCDMA networks with more than 130 million subscribers. Second-generation systems could already enable voice traffic to go wireless; now thirdgeneration systems face the challenge of making a new set of data services go wireless as well.
References [1] Andermo, P.-G. (ed.), ‘UMTS code division testbed (CODIT)’, CODIT final review report, September 1995. [2] Nikula, E., Toskala, A., Dahlman, E., Girard, L. and Klein, A., ‘FRAMES multiple access for UMTS and IMT-2000’, IEEE Personal Communications Magazine, April 1998, pp. 16–24.
Introduction
7
[3] Ojanpera¨, T., Rikkinen, K., Ha¨kkinen, H., Pehkonen, K., Hottinen, A. and Lilleberg, J., ‘Design of a 3rd generation multirate CDMA system with multiuser detection, MUD-CDMA’, Proc. ISSSTA’ 96, Mainz, Germany, September 1996, pp. 334–338. [4] Pajukoski, K. and Savusalo, J., ‘Wideband CDMA test system’, Proc. IEEE Int. Conf. on Personal Indoor and Mobile Radio Communications, PIMRC’97, Helsinki, Finland, 1–4 September 1997, pp. 669–672. [5] Holma, H., Toskala, A. and Latva-aho, M., ‘Asynchronous wideband CDMA for IMT-2000’, SK Telecom Journal, South Korea, Vol. 8, No. 6, 1998, pp. 1007–1021.
2 UMTS Services Harri Holma, Martin Kristensson, Jouni Salonen and Antti Toskala
2.1
Introduction
Second-generation systems like, for example, Global System for Mobile Communications (GSM) were originally designed for efficient delivery of voice services. Universal Mobile Telecommunication Services (UMTS) networks, on the contrary, are designed from the beginning for flexible delivery of any type of service, where each new service does not require particular network optimisation. In addition to the flexibility, the Wideband Code Division Multiple Access (WCDMA)/High-Speed Packet Access (HSPA) radio solution brings advanced capabilities that enable new services. Such capabilities are High bit rates, theoretically up to 2 Mbps in 3rd Generation Partnership Project (3GPP) Release 99, up to 14.4 Mbps in 3GPP Release 5 and up to 28.8 Mbps in Release 7. The practical bit rates are 1–2 Mbps with the first Release 5 deployments. Low delays with packet round trip times below 100 ms with Release 5 and even below 50 ms with Release 6. Seamless mobility also for packet data applications. Quality of Service (QoS) differentiation for high efficiency of service delivery. Simultaneous voice and data capability. Interworking with existing Global System for Mobile Communications (GSM)/General Packet Radio System (GPRS) networks. This chapter presents a few example UMTS services in Sections 2.2–2.5. The services are divided into person-to-person services, content-to-person services and business connectivity. Person-to-person refers to a peer-to-peer or intermediate server-based connection between two persons or a group of persons. Content-to-person services are characterised by the access to information or download of content. Business connectivity refers to laptop access WCDMA for UMTS – HSPA Evolution and LTE, fourth edition. and Antti Toskala # 2007 John Wiley & Sons, Ltd
Edited by Harri Holma
10
WCDMA for UMTS – HSPA Evolution and LTE
to internet or intranet using WCDMA as the radio modem. The location services are covered in Section 2.6. The QoS differentiation is presented in Section 2.7 and it is closely related to the efficiency of the service delivery. The cost of service delivery and the maximum system capacity are analysed in Section 2.8.
2.2
Person-to-Person Circuit Switched Services
This section considers Adaptive Multi-Rate (AMR) voice, wideband AMR voice and video. These services are initially provided through the circuit-switched (CS) core network in WCDMA, but they can be later provided also through the packet switched core network.
2.2.1
AMR-NB and AMR-WB Speech Services
The speech codec in UMTS will employ the AMR technique. The multi-rate speech coder is a single integrated speech codec with eight source rates: 12.2 (GSM-EFR), 10.2, 7.95, 7.40 (IS-641), 6.70 (PDC-EFR), 5.90, 5.15 and 4.75 kbps. The AMR bit rates can be controlled by the radio access network (RAN). To facilitate interoperability with existing cellular networks, some of the modes are the same as in existing cellular networks. The 12.2 kbps AMR speech codec is equal to the GSM EFR codec, 7.4 kbps is equal to the US-TDMA speech codec, and 6.7 kbps is equal to the Japanese PDC codec. The AMR speech coder is capable of switching its bit rate every 20 ms speech frame upon command. For AMR mode switching, in-band signalling is used. The AMR-narrow band (NB) coder operates on speech frames of 20 ms corresponding to 160 samples at the sampling frequency of 8000 samples per second, whereas AMR-wide band (WB) is based on the 16 000 Hz sampling frequency, thus extending the audio bandwidth to 50–7000 Hz. The coding scheme for the multi-rate coding modes is the socalled Algebraic Code Excited Linear Prediction Coder (ACELP). The multi-rate ACELP coder is referred to as MR-ACELP. Every 160 speech samples the speech signal is analysed to extract the parameters of the CELP model (LP filter coefficients, adaptive and fixed codebooks’ indices and gains). The speech parameter bits delivered by the speech encoder are rearranged according to their subjective importance before they are sent to the network. The rearranged bits are further sorted based on their sensitivity to errors and are divided into three classes of importance: A, B and C. Class A is the most sensitive, and the strongest channel coding is used for class A bits in the air interface. During a normal telephone conversation, the participants alternate so that, on average, each direction of transmission is occupied about 50% of the time. The AMR has three basic functions to utilise effectively discontinuous activity: Voice Activity Detector (VAD) on the transmission (TX) side. Evaluation of the background acoustic noise on the TX side, in order to transmit characteristic parameters to the receiver (RX) side. The transmission of comfort noise information to the RX side is achieved by means of a Silence Descriptor (SID) frame, which is sent at regular intervals. Generation of comfort noise on the RX side during periods when no normal speech frames are received.
11
UMTS Services
Discontinuous TX (DTX) has some obvious positive implications: in the user terminal, battery life will be prolonged or a smaller battery could be used for a given operational duration. From the network point of view, the average required bit rate is reduced, leading to a lower interference level and, hence, increased capacity. The AMR specification also contains error concealment. The purpose of frame substitution is to conceal the effect of lost AMR speech frames. The purpose of muting the output in the case of several lost frames is to indicate the breakdown of the channel to the user and to avoid generating possibly annoying sounds as a result of the frame substitution procedure [1,2]. The AMR speech codec can tolerate about a 1% frame error rate (FER) of class A bits without any deterioration of speech quality. For class B and class C bits a higher FER is allowed. The corresponding bit error rate (BER) of class A bits will be about 104. The bit rate of the AMR speech connection can be controlled by the RAN depending on the air interface loading and the quality of the speech connections. During high loading, such as during busy hours, it is possible to use lower AMR bit rates to offer higher capacity while providing slightly lower speech quality. Also, if the mobile is running out of the cell coverage area and using its maximum transmission power, then a lower AMR bit rate can be used to extend the cell coverage area. The capacity and coverage of the AMR speech codec is discussed in Chapter 11. With the AMR speech codec it is possible to achieve a trade-off between the network’s capacity, coverage and speech quality according to the operator’s requirements. 2.2.1.1 AMR Source-Based Rate Adaptation: Higher Voice Capacity [3] Currently, the AMR codec uses source-based rate adaptation with VAD-driven DTX to optimise the network capacity and the power consumption of the mobile terminal. In an AMR speech codec, VAD is used to lower the bit rate only during silence periods. However, active speech is coded by fixed bit rate that is selected by the radio network according to network capacity and radio channel conditions. Although the network capacity is optimised during silence periods using VAD/DTX, it can be further optimised during active speech with source-controlled rate adaptation. Thus, AMR codec mode is selected for each speech frame depending on the source signal characteristics; see Figure 2.1. The speech codec mode can be updated in every 20 ms frame in WCDMA. 0.8
12.2 kbps
2
Signal amplitude
0.6 0.4
1
4.75 kbps
7.40 kbps
0.2 0 -0.2
DTX
-0.4 -0.6 35.6
35.8
36
36.2 Time (s)
36.4
36.6
1 = AMR with source adaptation changes its bit rate according to the input signal 2 = AMR today uses fixed bit rate (+DTX) Figure 2.1. AMR source-based mode selection as a function of time and speech content
12
WCDMA for UMTS – HSPA Evolution and LTE
Bit rate reduction with source adaptation To tal bandwidth
100% 80% 60% Source adaptation Fixed AMR
40% 20% 0% 12.2 kbps with DTX
7.4 kbps with DTX
Figure 2.2. Reduction of required bit rate with equal voice quality
AMR source adaptation allows one to provide the same voice quality with a lower average bit rate. The bit rate reduction is typically 20–25% and is illustrated in Figure 2.2. The reduced AMR bit rate can be utilised to lower the required transmission power of the radio link, and thus further enhance AMR voice capacity. The WCDMA flexible layer 1 allows adaptation of the bit rate and the transmission power for each 20 ms frame. The estimated capacity gain is 15–20%. The bit stream format of source-adapted AMR is fully compatible with the existing fixed-rate AMR speech codec format; therefore, the decoding part is independent of source-based adaptation. The AMR source-based adaptation can be added as a software upgrade to the networks to enhance WCDMA downlink capacity without any changes to the mobiles. The AMR source-controlled adaptation can also be implemented with an AMR-WB speech codec. 2.2.1.2 Wideband AMR: Better Voice Quality [4] 3GPP Release 5 introduces the AMR-WB speech codec, which brings substantial voice quality enhancements compared with the AMR-NB codec or compared with a standard fixed telephone line. In the case of packet switched streaming, AMR-WB is already part of Release 4. The AMR-WB codec has also been selected by the ITU-T in the standardisation activity for a wideband codec around 16 kbps. This is of significant importance, since this is the first time that the same codec has been adopted for wireless and wireline services. This will eliminate the need of transcoding, and ease the implementation of wideband voice applications and services across a wide range of communications systems. The AMR-WB codec operates on nine speech-coding bit-rates between 6.6 and 23.85 kbps. The term wideband comes from the sampling rate, which has been increased from 8 kHz to 16 kHz. This allows one to cover twice the audio bandwidth compared with the classical telephone voice bandwidth of 4 kHz. While all the previous codecs in mobile communication operate on a narrow audio bandwidth limited to 200–3400 Hz, AMR-WB extends the audio bandwidth to 50–7000 Hz. Figure 2.3 shows the listening test result, where AMR-WB is compared with AMR-NB. The results are presented as the Mean Opinion Score (MOS), where a higher number indicates that a better voice quality is experienced. The MOS results show that AMR-WB is able to improve the voice quality without increasing the required radio bandwidth. For example, AMR-WB 12.65 kbps provides a clearly higher MOS than AMR-NB 12.2 kbps. The improved voice quality can be obtained because of higher sampling frequency.
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3.0 2.5 2.0 1.5
.2 R k N B A 12 M .2 R k W B A 6. M 60 R k W A B M 8. R 85 W k B A 12 M .6 R 5k W B A 1 M 4. R 25 W k B A 15 M .8 R 5k W B 18 .2 5k
k
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k 40
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6.
90 5. B
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75
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Figure 2.3. Mean opinion score (MOS) example with wideband and narrowband AMR
2.2.2
Video Telephony
3GPP has specified that ITU-T Recommendation H.324M shall be used for video telephony in circuit-switched connections and Session Initiation Protocol (SIP) for supporting IP multimedia applications, including video telephony in a 3GPP Release 5 core network environment. 2.2.2.1 Multimedia Architecture for Circuit-Switched Connections Originally, ITU-T Recommendation H.324 was intended for multimedia communication over a fixed telephone network (PSTN). 3GPP modified the H.324 Recommendation to make the system more suitable for the digital domain and more robust against transmission errors. The overall picture of the H.324M system is shown in Figure 2.4 [5,6]. H.324M consists of the following mandatory elements: H.223 for multiplexing and H.245 for control. Elements that are optional, but which are typically employed, are the H.263/ MPEG-4 video codec and the G.723.1/AMR speech codec. The recommendation defines the seven phases of a call: set-up, speech only, modem training, initialisation, message, end, and clearing. Level 0 of H.223 multiplexing is exactly the same as that of H.324, thus providing backward compatibility with older H.324 terminals. With a standardised in-band negotiation procedure the terminal can adapt to the prevailing radio link conditions by selecting the appropriate error resiliency level. One of the recent developments of H.324 is an operating mode that makes it possible to use an H.324 terminal over ISDN links. This mode of operation is defined in Annex D of the H.324 recommendation and is also referred to as H.324/I. The H.324/I terminals use the I.400-series ISDN user-network interface in place of the V.34 modem. The output of the H.223 multiplex is applied directly to each bit of the digital channel, in the order defined by H.223. Operating modes are defined bit rates ranging from 56 kbps to 1920 kbps, so that H.324/I allows the use of several 56 or 64 kbps links at the same time, thus providing direct interoperability with H.320 ISDN terminals.
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The scope of TS 26.111
Video I/O equipment
Video codec H.263, MPEG-4, H.264, …
Audio I/O equipment
Speech codec AMR, G.723.1, …
User data applications T.120, …
Data protocols V.14, LAPM, …
System control
H.245
Multiplex/ Demultiplex H.223 annex A, B, (C, D)
Optional multilink H.324M annex H
3GPP network
Optional receive path delay
CCSRL
NSRP LAPM/V.42
Call set-up
Figure 2.4. Scope of ITU Rec. H.324M
2.2.2.2 Video Codec It is recommended that all H.324M terminals should support both H.263 and MPEG-4 video codecs. Error resiliency and high efficiency make the MPEG-4 video codec particularly well suited for mobile video telephony. MPEG-4 Visual is organised into Profiles. Within a Profile, various Levels are defined. Profiles define subsets of tool sets. Levels are related to computational complexity. Among these Profiles, Simple Visual Profile provides error resilience (through data partitioning, RVLC, resynchronisation marker and header extension code) and low complexity. It is recommended that the Simple Visual Profile @ Level 0 is supported to achieve adequate error resilience for transmission error and low complexity simultaneously. No other Profiles are recommended to be supported. Higher Levels for the Simple Visual Profile may be supported, depending on the terminal capabilities [7]. MPEG-4 allows various input formats, including general formats such as Common Intermediate Format (CIF) and Quarter CIF (QCIF). H.324M encoders and decoders are recommended to support the 1:1 pixel format (square format). Encoders should signal this capability using H.245 capability exchange and the appropriate header fields in video codecs so that unnecessary pixel shape conversions can be avoided. It is also baseline compatible with H.263. Regardless of which specific video codec standard is used, all video decoder implementations should, in practice, include basic error concealment techniques. These techniques may include replacing erroneous parts of the decoded video frame with interpolated picture material from previous decoded frames or from spatially different locations of the erroneous frame. The decoder should aim to prevent the display of substantially corrupted parts of the
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picture. In any case, it is recommended that the terminal could tolerate every possible bit stream without catastrophic behaviour (such as the need for a user-initiated reset of the terminal). Video telephony has roughly similar delay requirements to the speech service. However, owing to the nature of video compression, higher compression factor, the BER requirement is more demanding than that of speech. Other possible circuit-switched data services are, for example: Video conferencing. Point-to-point or multipoint session between mobiles or with office video conferencing systems. Video streaming. Mobile TV with multiple live TV channels, video-on-demand for content such as news and movies. Multimedia multiparty gaming. Playing in real time with other remote users. Figure 2.5 shows an example 3G video telephony application.
Figure 2.5. Nokia N73 video telephony application
2.3 2.3.1
Person-to-Person Packet Switched Services Messaging
Excluding the mobile speech service, the short messaging service (SMS) is in all likelihood the most successful mobile service ever. Picture messaging was developed on top of the SMS to convey simple grey-scale bitmap pictures along with text. Multimedia messaging service (MMS) was then a natural evolution step toward a rich person-to-person messaging. MMS is an example of a store-and-forward type of service, where a message is composed on a mobile device, consisting typically of a still image taken with an in-built digital camera
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and a short descriptive text. An MMS is sent to a server where it is stored until fetched by the recipient’s device. The fetching of the message is triggered by a WAP Push message, which is fundamentally an SMS, including such details as the sender’s MSISDN, subject field and the location of the message on the server. Most of the handsets nowadays also support so-called Synchronised Multimedia Integration Language (SMIL), allowing users to create rich timed multimedia presentations with multiple images or videos as well as text and speech. As the MMS service is of a store-and-forward type, it does not inherently impose rules on delivery time, thereby suggesting timewise a loose 3GPP QoS class. What is more important for the users is that the content is delivered with a high probability and that the delivered message is as close to the original one as possible. There are hundreds, if not thousands, of MMS-capable device models, as well as a few WAP gateways and Multimedia Messaging Service Centres (MMSCs) available, all of them should work seamlessly together, without modifying the original message too much. In order to facilitate interoperability, Open Mobile Alliance (OMA), which works closely with 3GPP, has created MMS specifications. The OMA MMS version 1.2 specification defines a minimum set of requirements and conformance to enable end-to end interoperability of MMS applications, MMS-capable handsets and servers, and content provisioning. For instance, it limits the maximum MMS size to 300 kB. The forthcoming specification OMA MMS 1.3 has lifted the maximum size to 600 kB [8,9]. Another important requirement from an end user’s perspective is that it shall be possible to send MMS messages simultaneously while having a telephone call. This, in turn, calls for support for multiple radio access bearers. 2.3.1.1 Audio Messaging Audio messaging is a special type of MMS that consists only of the audio component in an SMIL presentation. Unlike voice mail, audio messages can be stored on the handset to listen to later. They can also be forwarded to other users with MMS-capable handsets. Audio messaging is easy to implement in any MMS network, as it needs no additional hardware or software. Subscribers can also start using it on any MMS-enabled handset as soon as they have an MMS service. For operators, this is a new way of generating revenue from their existing MMSC investments. The service is very cost effective and can be priced to differentiate it from other MMS-based services. It is a service that holds great potential to boost messaging in new growth regions where written skills may be less common than in mature markets. A one-minute audio message takes up only about 35 kB. 2.3.1.2 Instant Messaging Instant messaging is a very popular internet service that is now also available for mobile users. All major internet service providers, such as Yahoo, MSN, AOL, ICQ, Jabber and Google, offer mobile clients to access the service with the smart phone. There are also mobile messaging solutions that combine access to all aforementioned services in a single application. The main features of the instant messaging service are real-time chatting, sending/receiving images or other media elements and sending/receiving documents. 2.3.1.3 Mobile Email All of us use email on every business day, but how many of us are able to use mobile email to connect either to a business email or to a private email? It has been estimated that there are
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more than 650 million corporate email boxes in the world, but only about 10 million of them have mobile connectivity, and only half of those deploy push email. According to a survey commissioned by the GSM association, multimedia messaging has become a popular service among mobile users worldwide and mobile email has even greater potential. More than 40% of the 3061 consumers surveyed by Circle Research in Europe, North America and Asia regard MMS as an indispensable service, while 38% see the mobile email in the same light. When asked which mobile date services they prefer, the respondents ranked SMS first, email second and MMS third [10]. There are different ways to connect to an email server and deliver email messages. Now, as mobile browsers have become full HTML browsers, it is possible to connect directly with a browser to an email service on the internet, for instance MSN/hotmail, Gmail, Yahoo. Correspondingly, it is possible to connect email services possessing an XHTML interface. Often mobile devices have an in-built email client, which makes it straightforward to connect to the service, provided that the user can configure the device or request the configuration settings over the air. The most recent evolution is push email, where either the entire email message or its header is pushed silently (no user action) to the device. This may be based on the proprietary protocols or the standardised OMA Email notification 1.0 standard. Push email is in fact very similar to MMS. A message is sent via the packet data network to the server, which will send a push notification to a recipient device that will in turn fetch the message. The main differences between MMS and push email are that MMS supports SMIL presentation, but is limited in size, and email does not support SMIL, but can cope with the content of several megabytes. From the network point of view, email is a typical service that clearly belongs to a background QoS class – delay is not so important; instead, error-free delivery is. 2.3.1.4 Video Sharing Having live video or video clips in real time during a normal voice call allows users to enrich their communication even more. Users can add and remove the video element as they want, sharing live camera views or video clips from the device. Figure 2.6 shows the flow of a video sharing session between Malcolm and Keith. Keith and Malcolm are in a standard CS call. During the ongoing CS voice call, Keith chooses to share the live video. They both have devices capable of video sharing and are registered for the service. The registration may be in always-on mode, or
Figure 2.6. The flow of video sharing
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alternatively in when needed mode. The latter is triggered on after the CS call is connected. Keith confirms Malcolm as a recipient, then Malcolm receives video request from Keith and accepts it. The system shows the acceptance to Keith, who activates the sending of the video stream. Malcolm’s device starts showing the same video as Keith’s device. They can discuss it via the voice call. Keith ends the video sharing when he has shown what he wanted. The voice call between Keith and Malcolm remains active. The video sharing has both professional and private use cases: sharing vacation experiences, showing real-estate property for real-estate brokers, and explaining what the situation is when there is a need to repair equipment. Typical end-user QoS requirements for the video sharing service are the following: Both spatial and temporal resolution of the video stream shall be high enough to enable the scanning of the scene with the camera. One-way delay between the sender and the recipient shall be low enough to enable true interactivity. The resolution of the video as perceived by Malcolm is affected by several factors: the quality of Keith’s camera, his video shooting skills, video source coding and the bit rate of the video. The amount of movement also indirectly influences the perceived quality: the less movement, the better the quality. With H.263 and MPEG-4 video codecs, reasonably good quality on 2.400 QVGA displays is typically obtained with 64–100 kbps, whereas H.264 video coding provides sufficient quality already with less than 64 kbps. According to tests presented in [11], the H.264 standard achieves 50% average coding gain over MPEG-2, 47% average coding gain over H.263 baseline, and 24% average coding gain over H.263 high-profile encoders. By comparing video sharing with video telephony and content-to-person streaming, we can conclude the following: video telephony is the most interactive by nature, thus requiring the shortest delay; content-to-person streaming demands the least delay; and video sharing delay falls in between these two. The voice communication and its logical linkage to the video content sets the practical limit for the delay. A bearable one-way delay between the moment when the video is taken and the moment when it is rendered by the other device is in the order of some seconds ( 20 dB the UE shall be able to report measurements within 200 ms from an already identified cell and within 800 ms from a new cell belonging to the monitored set. We will consider these performance requirements, together with typical common channel power allocations from Chapter 8 and typical handover parameter values. The scenario is illustrated in Figure 9.18 where the UE is
Figure 9.18. Intra-frequency handover measurement scenario
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connected to cell1 and it needs to identify cell2 that is approaching Window_add value. The resulting Ec =I0 is obtained as follows: 1. We allocate 10 % for CPICH and for SCH giving Ec =I0r ¼ 10 dB. 2. We assume Window_add ¼ 3 dB where the UE needs to identify the cell when it is 3 dB below the strongest cell. This gives I0r =I0c1 ¼ 3 dB. 3. We further assume that the interference from other cells is 3 dB higher than the signal power from the best server. This gives I0c2 =I0c1 ¼ 3 dB. Ec Ec Ec Ec ¼ ¼ ¼ ¼ 8:5 dB ¼ 18:5 dB I0 I0r þ I0c1 þ I0c2 I0r ð1 þ 100:3 þ 100:6 Þ I0r
ð9:1Þ
The Ec =I0 in this scenario is 18.5 dB which is better than –20 dB given in the performance requirements. Phase (2): System Frame Number (SFN) Decoding In phase (2) of Figure 9.17 the UE decodes the system frame number from BCH that is transmitted on P-CCPCH. If we allocate 5 dB power for P-CCPCH compared to CPICH, the resulting Ec =I0 is 23.5 dB. The performance requirement for BCH decoding with BLER ¼ 1 % is 22.0 dB [2]. As higher BLER levels can be allowed, the planned P-CCPCH power allocation should be adequate. These calculations of the cell identification and the system frame number decoding show that the assumed common channel powers in Chapter 8 are high enough to guarantee accurate handover measurements. It may even be possible to optimise the common channel powers to be lower than shown in Chapter 8. Before the pilot Ec =I0 is used by the active set update algorithm in the UE, some filtering is applied to make the results more reliable. The measurement is filtered both on Layer 1 and on Layer 3. The Layer 3 filtering can be controlled by the network. The WCDMA handover measurement filtering is described in Figure 9.19.
Figure 9.19. Handover measurement filtering and reporting
The handover measurement reporting from the UE to RNC can be configured to be periodic, as in GSM, or event-triggered. According to [7] the event-triggered reporting provides the same performance as periodic reporting but with less signalling load. Before we leave the area of handover measurements we note the size of the neighbourlist. The maximum number of intra-frequency cells in the neighbourlist is 32. Additionally, there
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can be 32 inter-frequency and 32 GSM cells on the neighbourlist. The inter-frequency and inter-system measurements are covered later in this chapter. The maximum number of cells in the neighbourlist is shown in Table 9.5. Table 9.5. Maximum number of monitored cells Intra-frequency 32 1
Inter-frequency 1
32
Inter-system GSM 32
The 32 inter-frequency neighbours can be on two frequencies.
The neighbourlist can be defined by the network planning tool or it can also be tuned automatically by network optimisation algorithms based on the UE measurements. If there is a neighbour cell missing from the list, it can be noticed based on the UE measurements, since UE is required to identify a new detectable cell that does not belong to the monitored set within 30 s [6]. 9.3.1.3 Soft Handover Link Gains The primary purpose of soft handover is to provide seamless handover and added robustness to the system. This is mainly achieved via three types of gain provided by the soft handover mechanism, i.e., Macro diversity gain: A diversity gain over slow fading and sudden drops in signal strength due to, e.g., UE movement around a corner. Micro diversity gain: A diversity gain over fast fading. Downlink load sharing: A UE in soft handover receives power from multiple Node Bs, which implies that the maximum transmit power to a UE in X-way soft handover is multiplied by factor X (i.e. improved coverage). These three soft handover gains can be mapped to improved coverage and capacity of the WCDMA network. This section presents results of micro diversity soft handover gains that have been obtained by means of link level simulations. The micro diversity gains are presented relative to the ideal hard handover case, where the UE would be connected to the Node B with the highest pilot Ec =I0 . The results were presented and discussed in more detail in [3]. Figures 9.20 and 9.21 show the simulation results of 8 kbps speech in an ITU Pedestrian A channel, at 3 km/h, assuming that the UE is in soft handover with two Node Bs. The relative path loss from the UE to Node B #1 compared to Node B #2 is 0, 3, –6 or –10 dB. The highest gains are obtained when the path loss is the same to both Node Bs, i.e. the relative path loss difference is 0 dB. Figure 9.20 shows the soft handover gain in uplink transmission power with two branch Node B receive antenna diversity. Figure 9.21 shows the corresponding gains in downlink transmission power without transmit or receive antenna diversity. The gains are relative to the single link case where the UE only is connected to the best Node B. It should be noted that ITU Pedestrian A channel has only little multipath diversity, and thus the micro diversity soft handover gains are relatively high. With more multipath diversity, the gains tend to decrease.
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[dB]
1 0 −1 −2 −3 −10
−6
−3
0
Relative path loss to BS1 compared to BS2 [dB]
Figure 9.20. Soft handover gain in uplink transmission power (positive value ¼ gain, negative value ¼ loss) 3 2
[dB]
1 0 −1 −2 −3 −10
−6
−3
0
Relative path loss to BS1 compared to BS2 [dB]
Figure 9.21. Soft handover gain in downlink transmission power (positive ¼ gain, negative ¼ loss)
In Figure 9.20 the maximum reduction of the UE transmission power due to soft handover is observed to be 1.8 dB if the path loss is the same to both Node Bs. If the path loss difference is very large, the uplink soft handover should – in theory – never increase the UE transmission power since there is no extra transmission but only more Node Bs trying to detect the signal. In practice, if the path loss difference is very large, the soft handover can cause an increase in the UE transmission power. This increase is caused by the signalling errors of the uplink power control commands, which are transmitted in the downlink. But, typically, the Node B would not be in the active set of the UE if the path loss were 3–6 dB larger than the path loss to the strongest Node B in the UE’s active set. In the downlink the maximum soft handover gain is 2.3 dB (Figure 9.21), which is more than in the uplink (Figure 9.20). The reason is that no antenna diversity is assumed in the downlink, and thus in the downlink there is more need for micro diversity soft handover gain. In the downlink, soft handover causes an increase in the required downlink transmission power if the path loss difference is more than 4–5 dB for the current example. In that case
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the UE does not experience a gain from the signal transmitted from the Node B with the largest path loss. Hence, the power transmitted from that Node B to the UE will only contribute to the total interference in the network These simulation results also suggest values for Window_add and for Window_drop. Typical values for those parameters are shown in Table 9.6. Table 9.6. Typical handover parameters Window_add
Window_drop
1–3 dB
2–5 dB
9.3.1.4 Network Capacity Gains The potential capacity gain of soft handover mainly depends on the soft handover overhead (i.e. on the relative proportion of UEs in soft handover), the soft handover link gain, and the applied power control algorithm. Note that there are two different downlink power control algorithms for UEs in soft handover; 1. Conventional power control as described in Section 9.2.1.3; and 2. Site selection diversity transmission (SSDT) scheme presented in Chapter 6. Recall that SSDT power control relies on feedback information from the UE, so only one of the Node Bs in the active set transmits data, while the other Node Bs only transmit physical layer control information. Thus, SSDT is equivalent to selection transmit diversity, while conventional power control of UEs in soft handover may be characterised as equal gain transmit diversity. The potential gain of SSDT comes from the reduced interference in the downlink, which should compensate for the loss of diversity gain in the downlink for the user data. From the conceptual point of view, it is obvious that the gain of SSDT is larger at high data rates where the overhead from the control information is marginal. The capacity gain of soft handover in combination with SSDT, is on the same order of magnitude as the gain of soft handover and conventional power control. No significant gain of SSDT is observed, and in some cases the gain even turns into a loss. The reason for these observations can be explained as follows. A UE in soft handover periodically sends feedback commands to the Node Bs in the active set, which dictates which of the Node Bs should be transmitting the data. This results in larger power fluctuations at the different Node Bs, because the transmission to UEs is switched on/off on a relatively fast basis, as dictated by the UEs in soft handover. The alternation of Node B transmission towards UEs in soft handover is not within the control of the network, but purely UE controlled. Hence, although the SSDT scheme results in a reduction of the average total transmit power from Node Bs, the variance of the total transmit power also increases. The increased variance of the total transmit power maps to larger required power control headroom, which tends to reduce the potential gain of SSDT. Other aspects to note from a performance point of view include the impact of UE velocity, as with higher velocities the UE feedback is not well synchronised with the actual channel state. At some velocities, resonance problems do occur so the UE constantly asks for the ‘wrong’ Node B to transmit, via the feedback signalling to the
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network. This effect does, e.g., become dominant when the fading rate is roughly equal to the feedback rate. 9.3.1.5 Soft Handover Overhead The soft handover overhead is a common metric, which often is used to quantify the soft handover activity in a network. The soft handover overhead ( ) is defined as ¼
N X
nPn 1
ð9:2Þ
n¼1
where N is the active set size and Pn is the average probability of a UE being in n-way soft handover. In this context one-way soft handover refers to a situation where the UE is connected to one Node B, while two-way soft handover means that the UE is connected to two Node Bs, etc as shown in Figure 9.22. As each connection between a UE and Node B requires logical baseband resources, reservation of transmission capacity over the Iub, and RNC resources, the soft handover overhead may also be regarded as a measure of the additional hardware/transmission resources required for implementation of soft handover. Radio network planning is responsible for proper handover parameter setting and site planning so that the soft handover overhead is planned to be on the order of 20–40 % for a standard hexagonal cell grid with three sector sites. Note that an excessive soft handover overhead could decrease the downlink capacity as shown in Figure 9.21. In the downlink each soft handover connection increases the transmitted interference to the network. When the increased interference exceeds the diversity gain, the soft handover does not provide any gain for system performance.
Figure 9.22. Soft handover overhead
The soft handover overhead can, to a large extent, be controlled by proper selection of the parameters Window_add, Window_drop, and the active set size. However, there are also factors which influence the soft handover overhead, which are not controllable via the soft handover parameter settings. Some of these are: The network topology: How are the sites located relative to each other, how many sectors per site, etc.?
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The Node B antenna radiation patterns. The path loss and shadow fading characteristics. The average number of Node Bs that a UE can synchronise to. As an example, the soft handover overhead is plotted in Figure 9.23 for a standard hexagonal cell grid with three sector sites. These results are obtained from a dynamic network level simulator. Results are presented for a cell radii of 666 metres and 2000 metres. A standard 65 degree antenna is assumed for each sector. The deterministic path loss is modelled according to the Okumura–Hata model, while the shadow fading component is assumed to be log-normal distributed with 8 dB standard deviation. The transmit power of the CPICH is fixed at 10 % and 20 % of the maximum Node B transmit power for the small and large cell radii, respectively. The power of the SCH is 3.0 dB relative to the P-CPICH. The active set size is limited to three.
1.9 Cell radii : 2000 m Cell radii : 666 m
Soft handover overhead [-]
1.8 1.7 1.6 1.5 1.4 1.3 1.2 1.1
1
1.5
2
2.5 3 3.5 Window−add [dB]
4
4.5
5
Figure 9.23. The soft handover overhead versus the soft handover parameter Window_add for a hexagonal cell grid with three sector sites, and two different cell radii. Window_drop ¼ Window_add þ 2.0 dB
It is observed that the soft handover overhead increases approximately linearly when Window_add and Window_drop are increased. For the same soft handover parameter settings, the soft handover overhead is typically larger for the scenario with small cells, compared to large cells. This behaviour is observed because UEs in the large cell grid can only synchronise to a few Node Bs, while UEs in the small cell grid typically can synchronise to many Node Bs. Assuming that the design goal is to have a soft handover overhead of 20–40 %, then the results in Figure 9.23 indicate that appropriate parameter settings are Window_add ¼ 1–3 dB in small cells and slightly larger values in large cells. This conclusion is, however, only valid for a network topology with three sector sites. For the same soft handover parameter settings, the soft handover overhead increases when migrating from three sector sites to six sector sites. As a rule of thumb, the soft handover
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overhead increases by approximately 30 % when comparing results for three and six sector site configurations. This calls for selection of lower Window_add/Window_drop when increasing the number of sectors. An example of soft handover overhead in a live WCDMA network in a dense urban area is shown in Figure 9.24. The average overhead, including softer handovers, is 38 %. From the transmission point of view the overhead is less than 38 % because softer handover combining takes place in Node B and does not require extra transmission resources. The Window_add has been 2–4 dB, the Window_drop 4–6 dB, addition timer 320 ms and drop timer 640 ms.
Figure 9.24. Example soft handover overhead in a live network. Softer handovers are included in this figure
9.3.1.6 Active Set Update Rate Active set update rate is counted as the time between consecutive active set update commands and includes all the events: addition, removal and replacement. This measure is relevant for RNC dimensioning since active set update signalling causes load to the RNC. Active set update rates are shown in Figure 9.25. The average value in this example is 12 seconds, i.e. 5 active set updates per minute. The value depends on the average mobility of the users, on the cell size, on the network planning and on the handover parameters. Typically, soft handover overhead and active set update rates are related: smaller overhead can be obtained at the expense of a higher active set update rate.
9.3.2
Inter-system Handovers between WCDMA and GSM
WCDMA and GSM standards support handovers both ways between WCDMA and GSM. These handovers can be used for coverage or load balancing reasons. At the start of WCDMA deployment, handovers to GSM are needed to provide continuous coverage, and handovers from GSM to WCDMA can be used to lower the loading in GSM cells. This scenario is shown in Figure 9.26. When the traffic in WCDMA networks increases, it is important to have load reason handovers in both directions. The inter-system handovers are
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Figure 9.25. Example active set update rates in a live network
GSM
GSM
WCDMA
GSM
WCDMA
GSM
GSM
WCDMA
GSM
Handover WCDMA → GSM for coverage extension
Handover GSM → WCDMA for capacity extension
Figure 9.26. Inter-system handovers between GSM and WCDMA
triggered in the source RNC/BSC, and from the receiving system point of view, the intersystem handover is similar to inter-RNC or inter-BSC handover. The handover algorithms and triggers are not standardised. A typical inter-system handover procedure is shown in Figure 9.27. The inter-system measurements are not active all the time but are triggered when there is a need to make intersystem handover. The measurement trigger is an RNC vendor-specific algorithm and could be based, for example, on the quality (block error rate) or on the required transmission power. When the measurements are triggered, the UE measures first the signal powers of the GSM frequencies on the neighbourlist. Once those measurements are received by RNC, it commands the UE to decode the BSIC (base station identity code) of the best GSM candidate. When the BSIC is received by RNC, a handover command can be sent to the UE. The measurements can be completed in approximately 2 seconds. 9.3.2.1 Compressed Mode WCDMA uses continuous transmission and reception and cannot make inter-system measurements with a single receiver if there are no gaps generated in the WCDMA signals. Therefore, compressed mode is needed both for inter-frequency and for inter-system
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WCDMA for UMTS – HSPA Evolution and LTE
(1) RNC commands UE to start inter-system measurements with compressed mode.
Measurement trigger is RAN vendor specific algorithm
(2) UE measures signal power of GSM frequencies in the neighbourlist.
Typically, 1 second of measurements needed to average fading out
(3) RNC commands the UE to decode BSIC of the best GSM candidate
On average takes 1 second
(4) RNC sends handover command to the UE
Figure 9.27. Inter-system handover procedure
Figure 9.28. Measured UE transmission power during compressed mode
measurements. The compressed mode procedure is described in Chapter 6. UE transmission power during compressed mode is illustrated in Figure 9.28. During the gaps of the compressed mode, the fast power control cannot be applied and part of the interleaving gain is lost. Therefore, a higher Eb =N0 is needed during compressed frames, leading to a capacity degradation. An example calculation for the capacity effect is
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Radio Resource Management Table 9.7. Effect of compressed mode to the capacity 100 % UEs in compressed mode
10 % UEs in compressed mode
58 % 19 %
Every frame compressed Every 3rd frame compressed
5.8 % 1.9 %
shown in Table 9.7. In here it is assumed that the Eb =N0 needs to be 2.0 dB higher during the compressed frames. If all UEs used compressed mode in every frame, the interference levels would increase 58 % (¼10^0.2), and the capacity would reduce correspondingly. The measurements capability is typically fast enough if every 3rd frame is compressed. With every 3rd frame compressed, the UE can measure six samples in every gap, i.e. 480 ms/ 30 ms 6 ¼ 96 samples per 480 ms [6]. That measurement capability is similar to that in GSM only mobile for intra-GSM measurements. If we have every 3rd frame compressed, the capacity degradation is still 19 %. The results clearly show that compressed mode should only be activated when there is a need to make inter-system and inter-frequency handover. If we assume that 10 % of the UEs are simultaneously using the compressed mode, the capacity effect is reduced down to 1.9 %. These results show that the effect of the compressed mode to the capacity is negligible if the compressed mode is used only when needed. The RNC algorithms for activating the compressed mode are important to guarantee reliable handovers while maintaining low compressed mode usage. Compressed mode also affects the uplink coverage area of the real time services where the bit rate cannot be lowered during the compressed mode. Therefore, the coverage reason inter-system handover procedure has to be initiated early enough at the cell edge to avoid any quality degradation during the compressed mode. This situation is shown in Figure 9.29. The compressed mode affects coverage in two ways: 10 ms
10 ms
10 ms
Normal frame
Gap
Normal frame
Extra power needed because of compressed mode
Compressed frame
Figure 9.29. Effect of compressed mode on the coverage
1. The same amount of data is transmitted in a shorter time. This effect is 10 log10(15/ (15 7)) ¼ 2.7 dB, with a 7-slot gap in a 15-slot frame. 2. The Eb =N0 performance degrades during the compressed mode. The degradation is assumed to be 2.0 dB here. An example effect to the coverage area of speech is shown in Table 9.8, where the coverage is reduced by 2.4 dB with 20 ms interleaving. AMR voice connection uses 20 ms
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WCDMA for UMTS – HSPA Evolution and LTE Table 9.8. Effect of compressed mode on the coverage Interleaving
Reduction in coverage
10 ms 20 ms
2.7 dB þ 2.0 dB ¼ 4.7 dB (2.7 dB þ 2.0 dB)/2 ¼ 2.4 dB
interleaving. This 2.4 dB coverage degradation can be compensated by lowering instantaneously the AMR bit rate during the compressed mode if the UE hits its maximum power, see Section 12.2.1.2. Inter-system handovers from GSM to WCDMA are initiated in GSM BSC. No compressed mode is needed for making WCDMA measurements from GSM because GSM uses discontinuous transmission and reception. The service interruption time in the inter-system handovers is 40 ms maximum. The interruption time is the time between the last received transport block on the old frequency and the time the UE starts transmission of the new uplink channel. The total service gap is slightly more than the interruption time because the UE needs to get the dedicated channel running in GSM. The service gap is typically below 80 ms which is similar to that in intraGSM handovers. Such a service gap does not degrade voice quality. The service gap is illustrated in Figure 9.30.
WCDMA
Interruption time
Start GSM transmission