Computer Communications and Networks

Transcription

1 Computer Communications and Networks For further volumes:

2 The Computer Communications and Networks series is a range of textbooks, monographs and handbooks. It sets out to provide students, researchers and nonspecialists alike with a sure grounding in current knowledge, together with comprehensible access to the latest developments in computer communications and networking. Emphasis is placed on clear and explanatory styles that support a tutorial approach, so that even the most complex of topics is presented in a lucid and intelligible manner.

3 Lingfen Sun Is-Haka Mkwawa Emmanuel Jammeh Emmanuel Ifeachor Guide to Voice and Video over IP For Fixed and Mobile Networks

4 Lingfen Sun School of Computing and Mathematics, University of Plymouth, Plymouth, UK Is-Haka Mkwawa School of Computing and Mathematics, University of Plymouth, Plymouth, UK Emmanuel Jammeh School of Computing and Mathematics, University of Plymouth, Plymouth, UK Emmanuel Ifeachor School of Computing and Mathematics, University of Plymouth, Plymouth, UK Series Editors A.J. Sammes Centre for Forensic Computing Cranfield University Shrivenham campus Swindon, UK ISSN Computer Communications and Networks ISBN ISBN (ebook) DOI / Springer London Heidelberg New York Dordrecht Library of Congress Control Number: Springer-Verlag London 2013 This work is subject to copyright. All rights are reserved by the Publisher, whether the whole or part of the material is concerned, specifically the rights of translation, reprinting, reuse of illustrations, recitation, broadcasting, reproduction on microfilms or in any other physical way, and transmission or information storage and retrieval, electronic adaptation, computer software, or by similar or dissimilar methodology now known or hereafter developed. Exempted from this legal reservation are brief excerpts in connection with reviews or scholarly analysis or material supplied specifically for the purpose of being entered and executed on a computer system, for exclusive use by the purchaser of the work. Duplication of this publication or parts thereof is permitted only under the provisions of the Copyright Law of the Publisher s location, in its current version, and permission for use must always be obtained from Springer. Permissions for use may be obtained through RightsLink at the Copyright Clearance Center. Violations are liable to prosecution under the respective Copyright Law. The use of general descriptive names, registered names, trademarks, service marks, etc. in this publication does not imply, even in the absence of a specific statement, that such names are exempt from the relevant protective laws and regulations and therefore free for general use. While the advice and information in this book are believed to be true and accurate at the date of publication, neither the authors nor the editors nor the publisher can accept any legal responsibility for any errors or omissions that may be made. The publisher makes no warranty, express or implied, with respect to the material contained herein. Printed on acid-free paper Springer is part of Springer Science+Business Media (

5 Preface Since the release of the first Internet Phone in 1995, Voice over Internet Protocol (VoIP) has grown exponentially, from a lab-based application to today s established technology, with global penetration, for real-time communications for business and daily life. Many organisations are moving from the traditional PSTN networks to modern VoIP solutions and are using VoIP products such as audio/video conferencing systems for their daily business operation. We depend on different VoIP tools such as Skype, Google Talk and Microsoft Lync to keep contact with our business partners, colleagues, friends and family members, virtually any time and from anywhere. We now enjoy free or low cost VoIP audio or even high quality video calls which have made the world like a small village for real-time audio/video communications. VoIP tools have been incorporated into our mobile devices, tablets, desktop PCs and even TV sets and the use of VoIP tools is just an easy one-click task. Behind the huge success and global penetration of VoIP, we have witnessed great advances in the technologies that underpin VoIP such as speech/video signal processing and compression (e.g., from narrowband, wideband to fullband speech/audio compression), computer networking techniques and protocols (for better and more efficient transmission of multimedia services), and mobile/wireless communications (e.g., from 2G, 3G to 4G broadband mobile communications). This book aims to provide an understanding and a practical guide to some of the fundamental techniques (including their latest developments) which are behind the success of VoIP. These include speech compression, video compression, media transport protocols (RTP/RTCP), VoIP signalling protocols (SIP/SDP), QoS and QoE for voice/video calls, Next Generation Networks based on IP Multimedia Subsystem (IMS) and mobile VoIP, together with case studies on how to build a VoIP system based on Asterisk, how to assess and analyse VoIP quality, and how to set up a mobile VoIP system based on Open IMS and Android mobile. We have provided many practical examples including real trace data to illustrate and explain the concepts of relevant transport and signalling protocols. Exercises, illustrative worked examples in the chapters and end-of-chapter problems will also help readers to check their understanding of the topics and to stretch their knowledge. Step-bystep instructions are provided in the case studies to enable readers to build their own open-source based VoIP system and to assess voice/video call quality accordingly, or to set up their own mobile VoIP system based on Open IMS Core and IMSDroid with an Android mobile. Challenging questions are set up in the case studies to help them to think deeper and to practice more. v

6 vi Preface This book has benefitted from the authors research activities in VoIP and related activities of over 10 years. In particular, it has benefitted from the recent international collaborative projects, including the EU FP7 ADAMANTIUM project (Grant agreement no ), the EU FP7 GERYON project (Grant agreement no ) and the EU COST Action IC1003 European Network on Quality of Experience in Multimedia Systems and Services (QUALINET). The book has also benefitted from the authors teaching experience in developing and delivering modules on Voice and Video over IP to undergraduate and postgraduate students at Plymouth University in the past four years. Some of the contents of the book were drawn from the lecture notes and some of the case studies materials from the lab activities. This book can be used as a textbook for final year undergraduate and first year postgraduate courses in computer science and/or electronic engineering. It can also serve as a reference book for engineers in industry and for those interested in VoIP, for example, those who wish to have a general understanding of VoIP as well as those who wish to have an in-depth and practical understanding of key VoIP technologies. In this book, Dr. Sun has contributed to Chaps. 1 (Introduction), 2 (Speech Compression), 3 (Video Compression), 4 (Media Transport) and 6 (VoIP QoE); Dr. Mkwawa has contributed to Chaps. 1 (Introduction), 5 (SIP Signalling), 7 (IMS and Mobile VoIP), 8 (Case Study 1), 9 (Case Study 2) and 10 (Case Study 3); Dr. Jammeh has contributed to Chap. 3 (Video Compression) and Professor Ifeachor has contributed to Chap. 1 (Introduction) and the book editing. Due to the time constraints and the limitation of our knowledge, some errors or omissions may be inevitable in the book, we welcome any feedbacks and comments about the book. Finally, we would like to thank Simon Rees, our editor at Springer-Verlag for his encouragement, patience, support and understanding in the past two years in helping us complete the book. We would also like to express our deepest gratitude to our family for their love, support and encouragement throughout the process of this book.

7 Contents 1 Introduction OverviewofVoIP How VoIP Works and Factors That Affect Quality VoIPTools Microsoft slync Skype Google Talk X-Lite VoIPTrend VoIP Protocol Stack and the Scope of the Book Summary Speech Compression Introduction Speech Compression Basics Speech Signal Digitisation Speech Waveform and Spectrum How Is Human Speech Produced? Speech Compression and Coding Techniques WaveformCompressionCoding ParametricCompressionCoding Hybrid Compression Coding Analysis-by-Synthesis Narrowband to Fullband Speech Audio Compression Standardised Narrowband to Fullband Speech/Audio Codecs ITU-T G.711 PCM and G PCM-WB ITU-TG.726ADPCM ITU-T G.728 LD-CELP ITU-T G.729 CS-ACELP ITU-T G MP-MLQ/ACELP ETSIGSM ETSIAMR IETF silbc Skype/IETF s SILK ITU-TG.722ADPCM-WB vii

8 viii Contents ITU-T G Transform Coding ETSI AMR-WB and ITU-T G ITU-T G.719 Fullband Audio Coding Summary of Narrowband to Fullband Speech Codecs IllustrativeWorkedExamples Question Question Question Summary Problems Video Compression Introduction to Video Compression VideoCompressionBasics Digital Image and Video Colour Components Colour Sub-sampling Video Resolution and Bandwidth Requirement Video Compression Techniques LosslessVideoCompression LossyVideoCompression PredictiveCoding Quantisation TransformCoding Interframe Coding Video Coding Standards H H MPEG1& H MPEG H HighlyEfficiencyVideoCoding(HEVC) IllustrativeWorkedExamples Question Question Question Summary Problems Media Transport for VoIP MediaTransportoverIPNetworks TCPorUDP? Real-Time Transport Protocol RTP RTP Header RTP Header for Voice Call Based on Wireshark RTP Payload and Bandwidth Calculation for VoIP... 80

9 Contents ix IllustrativeWorkedExample RTP Header for Video Call Based on Wireshark RTP Control Protocol RTCP RTCP Sender Report and Example RTCP Receiver Report and Example RTCP Source Description and Example RTCPBYEPacketandExample Extended RTCP Report RTCP XR for VoIP Metrics CompressedRTP crtp Basic Concept of Compressed RTP crtp IllustrativeWorkedExample Summary Problems VoIP Signalling SIP What is Session Initiation Protocol? SIPNetworkElements User Agent ProxyServer RedirectServer Registrar LocationServer SIP Protocol Structure SIPMessageFormat Session Descriptions Protocol SessionDescription TimeDescription MediaDescription Attributes ExampleofSDPMessagefromWireshark SIP Messages Flow SessionEstablishment Summary Problems VoIP Quality of Experience (QoE) Concept of Quality of Service (QoS) What is Quality of Service (QoS)? QoSMetricsandMeasurements NetworkPacketLossandItsCharacteristics Delay, Delay Variation (Jitter) and Its Characteristics Quality of Experience (QoE) for VoIP What is Quality of Experience (QoE)? Factors Affect Voice Quality in VoIP OverviewofQoEforVoiceandVideooverIP Subjective Speech Quality Assessment...138

10 x Contents 6.4 Objective Speech Quality Assessment Comparison-Based Intrusive Objective Test (Full- Reference Model) Parameter-Based Measurement: E-Model A Simplified and Applicable E-Model Subjective Video Quality Assessment Objective Video Quality Assessment Full-Reference (FR) Video Quality Assessment Reduced-Reference (RR) Video Quality Assessment No-Reference Video Quality Assessment IllustrativeWorkedExamples Question Question Question Summary Problems IMS and Mobile VoIP What Is IP Multimedia Subsystem? WhatDoWeNeedIMSfor? IMSArchitecture IMSElements IMSServices IMS Signalling and Bearer Traffic Interfaces Mobile Access Networks Cellular Standards The GSM Standard The UMTS Standard Long-Term Evolution Summary Problems Case Study 1 Building Up a VoIP System Based on Asterisk WhatisAsterisk? Channel Modules Codec Translator Modules Application Modules File Format Modules Installing Asterisk WhatIsX-Lite UsingX-Lite VoiceandVideoInjectionTools Manycam Video Injection Tool VirtualAudioCableInjectionTool Lab Scenario Adding SIP Phones...205

11 Contents xi 8.6 Configuring Dial Plans Configuring DAHDI Channels Starting and Stopping Asterisk Setup SIP Phone Making Voice Calls Between SIP Phones Making Video Calls Between SIP Phones Making Voice Calls Between SIP and Analogue Phones Problems Case Study 2 VoIP Quality Analysis and Assessment WhatIsWireshark LiveCaptureandOfflineAnalysis Three-PanePacketBrowser VoIPAnalysis Wireshark Familiarization Introduction to Netem and tc Commands Addingqdisc Changing and Deleting qdisc Lab Scenario Challenges SIPRegistration Challenges SIPInvite Challenges VoIP Messages Flow Challenges VoIP Quality Assessment: Packet Losses Challenges VoIP Quality Assessment: Delay Variation Challenges Problems Case Study 3 Mobile VoIP Applications and IMS What Is Open Source IMS Core The Main Features of OSIMS Core P-CSCF The Main Features of OSIMS Core I-CSCF The Main Features of OSIMS Core S-CSCF The Main Features of OSIMS Core FHoSS Installation and Configuration of OSIMS Core WhatIsAndroid Android Smart Phone Market Share AndroidArchitecture TheHistoryofAndroid IMSDroidIMSClient Lab Scenario Configuring IMSDroid

12 xii Contents Adding OSIMS Core Subscribers MakingVoiceandVideoCalls PlacingaCall InCallScreen Problems Index...265