Babet Final Report

Broadband applications on limited bandwidth networks
A survey on the suitability of current and emerging access technologies
Colophon
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01.11.01 1 Babet TI/RS/2001/100 https://doc.telin.nl/dscgi/ds.py/Get/File-18723/Babet_Final_Report.pdf Anyone Final S. M. Iacob Telematica Instituut (http://www.telin.nl ) Martin Alberink, Mortaza Bargh, Frank Biemans, Henk Eertink, Sorin Iacob, Daan Velthausz
Copyright 2001 Ministerie van Verkeer en Waterstaat Persoonlijk gebruik is toegestaan. U heeft toestemming nodig van of via het Ministerie van Verkeer en Waterstaat (http://www.minvenw.nl/dgtp) voor het kopieren en/of publiceren van dit materiaal voor reclame of promotionele doeleinden of voor het maken van verzamelde werken met als doel verkoop of distributie via servers of lijsten of voor het hergebruik van enig auteursrechtelijk beschermd deel van dit werk in andere werken. Personal use of this material is permitted. However, permission to reprint/republish this material for advertising or promotional purposes or for creating new collective works for resale or redistribution to servers or lists, or to reuse any copyrighted component of this work in other works must be obtained from or via Ministerie van Verkeer en Waterstaat.
Table of Contents
S am envat ting Con clu s ie s Su mm ar y Con clu s ion s 1 Int rodu ct ion 1.1 Goal and scope of the project 1.2 Structure of the report 2 Com mun ic at ion s e rvic e s an d app li c atio ns 2.1 Networked multimedia applications 2.1.1 Relevant categories of networked multimedia applications 2.1.2 Bandwidth requirements 2.2 Quality of service parameters 2.3 Graceful degradation and adaptability 3 Comp r e ss ion te ch niqu e s 3.1 Source coding 3.1.1 Lossless compression 3.1.2 Lossy compression 3.1.3 Pushing the limits 3.1.4 Theoretical limits of compression 3.1.5 Beyond the limits? 3.2 Compression standards 3.2.1 Image compression 3.2.2 The MPEG standard 3.3 Compression technologies 3.3.1 RealVideo from RealNetworks 3.3.2 Quicktime 3.3.3 Windows media player 3.3.4 DivX from DivXNetworks 3.3.5 On2 Technologies 3.3.6 Wavelet-based technologies 3.3.7 Fractals 4 Multimed i a d is tr ibut ion t ec hni que s 4.1 Network-level multicasting or broadcasting 4.2 Distribution technology 4.3 Storage versus transport 5 Dev elo pm ent s in a cce s s t ec hno log ie s 5.1 DSL networks 5.1.1 Description 5.1.2 Deployment 5.1.3 Innovation 5.2 CATV-based networks 5.2.1 Description 5.2.2 Deployment 5.2.3 Innovation 5.2.4 Comparison between Cable and DSL technologies 1 13 15 25 27 27 29 31 31 31 32 32 34 37 37 37 39 41 43 45 45 46 47 51 51 51 52 52 52 52 53 55 55 56 57 59 60 60 61 61 62 62 63 63 63
T A B L E
O F
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5.3 5.4 5.4.1 5.4.2 5.5 5.6 5.7 5.7.1 5.7.2 5.8 5.8.1 5.8.2 5.8.3
Power Line Communication (PLC) Wireless local loop Data-access: fixed wireless connections Free space optics (FSO) Cellular technologies Hot-spot technologies and home networks Hybrid networks Multiple communication channels Complementary hybrid networks New developments in access technologies Future cellular technologies Bi-directional satellite links T-spectrum technology
64 66 67 69 71 71 74 74 75 77 77 77 78 79 79 80 80 81 81 82 87 91 91 91 91 91 93 94 95 96 96 96 97 97 98 99 100 10 3 103 103 103 107 108 11 3 113 116 120
6 Wra p u p 6.1 Bandwidth requirements for different categories of applications 6.2 Compression 6.3 Channel coding techniques 6.4 Multimedia distribution mechanisms 6.5 Access technologies 6.6 Suitability of access technologies for the given applications Ref e re nc e s A p p end ix A : So me the o r y on comp r e ss ion A.1 Lossless compression techniques A.1.1 Shannon's theory of coding for noiseless channels A.1.2 The counting argument A.1.3 Pixel based compression techniques A.1.4 Redundancy and auto-correlation A.1.5 Predictive techniques A.1.6 The Karhunen-Loeve transform A.1.7 The Discrete Cosine Transform (DCT) A.2 Lossy compression techniques A.2.1 Block truncation coding A.2.2 Quantisation of transform coefficients A.2.3 Vector quantisation A.2.4 Perceptual coding A.2.5 Hybrid techniques - the JPEG algorithm A.2.6 MPEG-1 video compression A p p end ix B: Modul atio n and c ha nn el co ding B.1 Channel coding techniques B.1.1 Communication channel and message transmission B.1.2 Channel coding B.2 Modulation techniques B.3 Multiple access protocols A p p end ix C: A c c e s s te c hn olog i es - Te ch nic a l det a il s C.1 Cable TV access networks C.2 Copper twisted pairs (subscriber loops) C.3 WLAN standards
VI
T E L E M A T I C A
I N S T I T U U T
Samenvatting
De beschikbaarheid van telecommunicatiediensten wordt gezien als een publiek belang en is n van de overheidstaken waarvoor het Directoraat1 Generaal Telecommunicatie en Post (DGTP) zorg heeft. Dit beleid houdt in dat er op wordt toegezien dat elke Nederlander toegang heeft tot (spraak)telefonie tegen een betaalbaar tarief. Dit wordt ook wel universele dienst (UD) genoemd. Sinds een aantal jaren kennen we nieuwe telecomdiensten gebaseerd op Internet-technologie zoals e-mail, web en chat-diensten. Het is mogelijk dat over een aantal jaren bepaalde Internetdiensten ook als universele dienst worden benoemd. Daarvoor zou kunnen worden gekozen wanneer toegang tot die diensten voor alle burgers van dermate belang is (geworden) dat deelname aan activiteiten van sociale en/of economische activiteiten er van af hangt, en wanneer de toegang daartoe (soms) problematisch is. Een bepaalde categorie Internetdiensten is die van breedband. Hierbij wordt dan gedacht aan zaken zoals teleconferencing, telewerken, onderwijs op afstand, de toepassing van webcameras in de gezondheidszorg, het raadplegen van overheidsinformatie, enzovoort. Voor breedbanddiensten is ook een breedbandig toegangsnetwerk nodig. Dit wordt vaak geassocieerd met het leggen van glasfibers naar iedere woning (fibre to the home). Uitrol van zon op glasvezel gebaseerd toegangsnet is al gaande. Maar het moment waarop een dergelijke infrastructuur nationale dekking heeft bereikt zal waarschijnlijk nog een flink aantal jaren duren. Sterker nog, in de komende jaren zal moeten blijken of dit in wat afgelegen gebieden berhaupt haalbaar is. Deze studie is een verkenning van de perspectieven voor particulieren en kleine bedrijven in gebieden zonder glasvezeltoegang, vanuit een technologisch perspectief. De centrale vraag in dit rapport is:
"In hoeverre maken nieuwe aanstaande technologische ontwikkelingen, waaronder compressie, het mogelijk om de bestaande capaciteit in het huidige en aanstaande vaste en mobiele aansluitnet beter te benutten zodat over een aantal jaren daarlangs ook breedband-informatie- en communicatiediensten naar particulier en kleinbedrijf te transporteren zijn. Hoever zal de praktijk over 5 jaar zijn gevorderd om via deze netwerken diensten aan te bieden die nu als 'breedband' te boek staan (zoals bijvoorbeeld video en audiostreaming), door gebruik te maken van compressie en/of slimme algoritmes en protocollen. Waar zal naar verwachting ongeveer de grens van het mogelijke liggen gerekend in termen van de transmissie-bandbreedte die betreffende diensten vereisen."
In zijn algemeenheid is ons dus gevraagd de ontwikkelingen te inventariseren die in de komende jaren bandbreedtebesparend kunnen gaan werken, d.w.z. technologien en ontwikkelingen die het mogelijk maken dat ondanks een geringe(re) bandbreedte in het toegangsnetwerk toch diensten kunnen worden ontvangen die als (relatief) breedbandig kunnen worden getypeerd.
Dit directoraat is onderdeel van het Ministerie van Verkeer en Waterstaat.
S A M E N V A T T I N G
Bovenstaande vragen kunnen en zullen niet met een absoluut antwoord worden afgedaan. Het antwoord is resultante van interacterende maar ook onzekere effecten van technologie, markt en ook van beleid. In verband hiermee moet dit rapport eerder als een visie worden beschouwd dan als een voorspelling. Met het oog op de onzekerheid voor wat betreft de behoefte aan elektronische diensten en de snelheid van de technologische ontwikkelingen kijken we tot vijf jaar in de toekomst. In Nederland is op dit moment sprake van een behoorlijk hoge dichtheid van de communicatieinfrastructuur, zowel voor vaste als draadloze toegangsnetwerken. Het is daarom zinvol om vast te stellen in welke mate huidige en aankomende toegangstechnologien waaronder ADSL, kabeltelevisie, GPRS, satellietcommunicatie en vast-draadloze verbindingen (WLL) kunnen voorzien in de eisen van essentile breedbanddiensten in de nabije toekomst. Hierbij is het ook belangrijk om vast te stellen of op dit gebied nog belangrijke technologische ontwikkelingen kunnen worden verwacht of dat de huidige prestaties het maximaal haalbare al dicht benaderen. Deze studie beschrijft de voor DGTP van belang zijnde technologische ontwikkelingen op het genoemde gebied. De aanpak Om vast te stellen in welke mate breedbanddiensten kunnen worden aangeboden via bestaande toegangsnetwerken moet eerst worden bekeken welke eisen door de belangrijkste toepassingen en diensten daarvan worden gesteld. Het probleem hierbij is natuurlijk dat het niet mogelijk is om exact te voorspellen welke breedbanddiensten de komende jaren als essentieel zullen worden beschouwd. In overleg met het Directoraat-Generaal Telecommunicatie en Post hebben we vier categorien basisdiensten gekozen die als karakteristiek en dekkend worden gezien voor de essentile breedbandtoepassingen en waartoe in zeer brede kring eventueel toegang moet worden geboden. Daarnaast hebben we verschillende mogelijk essentile toepassingen genoemd op het gebied van onderwijs, de gezondheidszorg, telewerken, enzovoort. Dit staat weergegeven in onderstaande tabel. Voorbeelden van essentile toepassingen Video (nieuws) op aanvraag, asynchrone virtuele radio- en TV-zenders Internetradio en -TV, leren op afstand voor besloten groepen, controle of toezicht op afstand Adviseren op afstand, Internettelefonie, videoconferencing, telewerken Toegang tot digitale bibliotheken, nationale archieven, multimedia e-mail, individueel afstandsonderwijs, telewerken Bijbehorende Basisdiensten Audio en video streaming Audio en video multicast
Conferencing Interactieve toegang tot documenten en berichten
Deze basisdiensten stellen zodanig verschillende eisen aan de infrastructuur voor wat betreft bandbreedte en quality of service (QoS) dat hiermee waarschijnlijk aan de eisen van toekomstige essentile toepassingen kan worden voldaan. De basisdiensten stellen verschillende eisen aan netwerkinfrastructuren. Het betreft eisen op het gebied van 1. de door de netwerken te leveren bandbreedte voor de volledige realisatie van de diensten
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2. de door de netwerken te realiseren quality of service (vertraging, verlies van data en dergelijke) 3. de door de netwerken te leveren bandbreedte voor een beperkte realisatie van de diensten Onderstaande tabel geeft de eisen aan de bandbreedte weer voor de basisdiensten. Niet alleen de hoeveelheid data maar ook eisen aan de tijd tussen het verzenden en het ontvangen van data bepalen de vereiste bandbreedte. De kolom kwaliteit geeft aan in welke gevallen strikte eisen aan de vertraging en variaties in de vertraging worden gesteld. Laag betekent dat er in een bepaald geval geen sprake is van strikte eisen omdat er hierbij geen sprake is van audio- of videosignalen. Medium kwaliteit geeft aan dat vertraging niet zo erg is zolang deze maar constant is. Met een constante vertraging kan men namelijk nog wel een constante kwaliteit van beeld en geluid garanderen. Hoge kwaliteit betekent dat slechts een beperkte vertraging is toegestaan omdat het interactieve communicatie betreft. Basisdienst Toepassing Minimum bandbreedte [Mbit/s] 0.05 0.08 20 0.4 0.128 0.064 Maximum bandbreedte [Mbit/s] 0.2 0.2 340 8 1.5 0.25 Vereiste kwaliteit Medium Medium Medium Medium Hoog Hoog
A/V streaming en A/V multicast
Asynchrone radiozenders MPEG audio streaming HDTV MPEG video TV Videoconferencing Internettelefonie
Conferencing (bandbreedte per deelnemende partij) Interactieve toegang tot documenten en berichten
e-learning, e-government Gezondheidszorg Telewerken
0.5 0.5 0.014
2 20 6
Laag Laag Laag
Bij deze tabel moet overigens wel opgemerkt worden dat applicaties als gezondheidszorg, telewerken ook componenten als videoconferencing of internettelefonie kunnen bevatten. In dat geval zijn de kwaliteitseisen voor die functies natuurlijk niet Laag, maar juist Hoog. Om concrete uitspraken over technologische innovaties te kunnen doen, hebben we eerst gekeken naar de typische structuur van de toepassingen. Elk van deze toepassingen is opgebouwd uit verschillende componenten. In de eerste plaats de encoding-decoding component waarbij natuurlijke of fysische signalen worden omgezet in digitale datastromen. Deze worden via netwerken verstuurd naar andere plaatsen (bijvoorbeeld van een televisiestudio naar een eindgebruiker of tussen eindgebruikers bij een conferencingdienst), gebruikmakend van distributietechnieken. Deze studie kijkt naar technologische innovaties in elk van deze drie gebieden (de gekleurde vlakken in onderstaande figuur).
applicatie encoder transcoder Multimedia Distributie Technieken (Toegangs) netwerken
applicatie decoder
(Toegangs) netwerken
We bespreken nu eerst kort deze gebieden, en daarna wordt een en ander gentegreerd en gellustreerd middels een scenario.
Ontwikkelingen in data compressietechnieken
Compressie is een van de meest doeltreffende manieren om de efficintie van dataoverdracht te vergroten. Vooral data afkomstig van natuurlijke audio- of videosignalen kan flink worden gecomprimeerd met bepaalde compressietechnieken waarbij verlies van informatie optreedt maar niet van de menselijke waarneming van die informatie. Het oorspronkelijke signaal kan hierbij niet worden gereconstrueerd. Compressietechnieken worden gemplementeerd in encoders, transcoders en decoders. Data compressie technieken kunnen onderverdeeld worden in twee categorien: 1) kwaliteitsbehoudende compressie en 2) kwaliteitsverlagende compressie. Bij kwaliteitsbehoudende compressie gaat het altijd om een verlaging van de redundantie. De mate van compressie is dan ook niet zo heel hoog (een typische reductie is 2:1 bij standaard afbeeldingen). Door gebruik te maken van verschillende intrinsieke eigenschappen van de te comprimeren objecten kunnen kwaliteitsbehoudende compressietechnieken echter wel verbeterd worden. Bijvoorbeeld, wanneer een afbeelding bestaat uit verschillende objecten, kan elk van die objecten onafhankelijk gecomprimeerd worden. Dit levert een optimale representatie van de afbeelding, omdat de deelobjecten context-onafhankelijk kunnen worden behandeld. Dit soort methoden brengt echter de maximale haalbare compressiefactor niet verder dan 5:1. Voor een videofilm is, door bovendien extra redundantie tussen de verschillende frames te benutten, zo een maximale kwaliteitsbehoudende compressie in de ordegrootte van 10:1 haalbaar. Maar voor TV kwaliteit video signalen (640x480 resolutie, 24 bits kleur, 25 frames/seconde = 18 Mbits/s) resulteert dit nog steeds in een gecomprimeerde bandbreedte van 2 Mbit/s. Let wel, zonder geluid! Bij kwaliteitsverlagende compressietechnieken vinden bepaalde onomkeerbare transformaties plaats op de originele data met als gevolg dat het origineel niet voor 100% gereconstrueerd kan worden. Dit is acceptabel wanneer nog steeds dezelfde persoonlijke communicatie tot stand gebracht wordt, of wanneer de verschillen onmerkbaar zijn. Oftewel: kwaliteitsverlagende compressie verbergt niet alleen de redundantie binnen de originele data maar ook de irrelevante details. Er bestaan een groot aantal verschillende kwaliteitsverlagende compressietechnieken voor spraak, geluid, afbeeldingen, fotos en/of video. Hierbij worden tactieken gebruikt die zijn gebaseerd op benadering van het origineel (bijvoorbeeld door middel van kwantitatieve transformatie codering) of door een model te definiren waarmee de data gereconstrueerd kan worden met een kleinere set van parameters en een groot aantal van tevoren opgeslagen prototypes (bijvoorbeeld met vector kwantisatie of componenten analyse). Slechts een klein aantal van deze technieken is gestandaardiseerd (en alleen gestandaardiseerde
T E L E M A T I C A
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formaten worden breed gebruikt). Momenteel zijn de JPEG ISO serie and de MPEG open standaard specificaties koploper bij beelden videocompressie. Moderne producten die gebaseerd zijn op JPEG2000 and MPEG4 bieden niet alleen een zeer hoge compressieverhouding (100:1 voor MPEG4 video met DVD kwaliteit) maar bieden ook aanpassingsmechanismen om tegemoet te komen aan de verschillende bandbreedtes en mogelijkheden van terminals. Echter, de complexiteit van het coderen is hoog. Dit leidt tot een aanzienlijke vertraging waardoor hun gebruik in peer-to-peer applicaties zoals videoconferencing nog steeds geremd wordt. Ondanks een scala aan methodieken en algoritmen voor kwaliteitsverlagende compressie is het onmogelijk om een theoretisch maximum voor deze categorie compressietechnieken te bepalen. Hiervoor zijn 2 redenen: Ten eerste is er geen eenduidig perceptiemodel dat de minimale informatie uitrekent die nodig is voor het begrijpen van een visuele scne. Ten tweede, ook al zouden er zulke modellen bestaan, dan nog kunnen context afhankelijkheden, ervaringen, verwachtingen een invloed hebben op de algehele perceptie van het beeld. Het gevolg hiervan is dat er verschillende limieten gedefinieerd worden voor verschillende toepassingen. Bij beeldtelefonie, bijvoorbeeld, communiceren de betrokken partijen niet alleen verbaal, maar mogelijk ook emotioneel. Dit wordt als vanzelfsprekend bereikt door zowel een audio als een video communicatiekanaal te gebruiken. Desalniettemin, de enige informatie die nodig is, is datgene wat de andere partij zegt en hoe de partij zich gedraagt tijdens het gesprek. Met andere woorden, het versturen van de woorden en een interpretatie van de non-verbale communicatie zou in feite voldoende zijn om hetzelfde resultaat te behalen. Samen met een textto-speech applicatie die een audiosignaal genereert uit deze woorden via een van tevoren opgenomen audiomodel van de andere partij, en mogelijk zelfs een animatie van het hoofd gebaseerd op emotionele gezichtsuitdrukkingen, zou een bitrate van 300 bit/s voldoende zijn om deze communicatievorm te ondersteunen. Echter, deze methoden zijn erg academisch van aard en vereisen heel wat aanpassingen aan de terminals van de eindgebruikers en aan de distributietechnieken om de verschillende modellen (mogelijk off-line) te synchroniseren. Het is daarom zeer onwaarschijnlijk dat we zulke scenario's binnen de eerst komende vijf jaar buiten de laboratoria zullen zien. Kortom, wanneer we de bestaande (vrijwel) gestandaardiseerde compressietechnieken in ogenschouw nemen, kunnen we concluderen dat het onwaarschijnlijk is dat deze compressietechnieken binnen de komende vijf jaar sterk zullen verbeteren. Wel zullen recent gestandaardiseerde adaptieve coderingsschema's zoals MPEG4 de komende jaren meer en meer gebruikt worden. Ondanks de hoge encoding-kosten bieden deze coderingen veel toegevoegde waarde ten opzichte van de huidige standaarden, met name voor mensen met een lage toegangscapaciteit tot het Internet.
Ontwikkelingen in distributietechnologie
Distributiemechanismen worden ingezet voor een efficinter transport van multimedia objecten. Dit soort technieken wordt met name gebruikt door operators voor het optimaliseren en ontlasten van hun netwerk. Voorbeelden hiervan zijn onder andere het cachen van informatie en het repliceren van informatie naar die locaties binnen het netwerk die dicht bij de klant staan. Dit soort technieken valt hierdoor buiten de scope van dit onderzoek, omdat het met name betrekking heeft op backbone netwerken van operators. Echter, een tweetal aspecten van distributietechnologien zijn wel interessant voor eindgebruikers: 1. Intelligente (adaptieve) broadcasting van informatie vanaf n verzender naar meerdere ontvangers. Binnen academische en bedrijfslaboratoria wordt op dit moment gewerkt aan oplossingen waarbij de intermediaire systemen niet alleen gewoon de informatie verspreiden naar de klanten, maar indien nodig ook de informatie in een specifiek formaat 5
doorstuurt. Deze technieken kunnen dus gebruikt worden om op een schaalbare manier klanten een optimale kwaliteit aan te bieden, afhankelijk van bijvoorbeeld de toegangsnelheid. Gelijksoortige technieken kunnen gebruikt worden binnen conferenties waar meerdere partijen met verschillende terminals met elkaar communiceren. 2. Pro-actief ophalen van data. Met deze techniek wordt de overtollige bandbreedte gebruikt voor het van tevoren binnenhalen van informatie. De mogelijkheden van deze methode zijn groot, hoewel enige terughoudendheid op zijn plaats is: de methode is alleen bruikbaar als de benodigde objecten van tevoren bepaald kunnen worden (zoals bijvoorbeeld het geval is bij asynchrone tele-educatie scenario's). Die distributie kan eventueel zelfs via CD-ROMs of DVDs plaatsvinden. On-line kun je dan, bijvoorbeeld, de toegang regelen of alleen de parameters versturen waarmee de data gepresenteerd kan worden aan eindgebruikers.
Ontwikkelingen in Toegangstechnologie
Toegangsnetwerken bestaan uit een fysieke laag (koper, coax, glas, ether, ) waar bovenop coderings- en modulatietechnieken gebruikt worden voor de versturing van data. De service providers bieden aan eindgebruikers altijd een combinatie van fysieke laag en codering/modulatie aan (uitgezonderd de dark fibre oplossingen). In deze studie wordt alleen gekeken naar draadloze media, kabeltelevisienetwerken en telefoonbekabeling. Nieuwe modulatietechnieken kunnen zorgen voor grote winst in efficintie (vergelijk bijvoorbeeld ADSL met analoge telefonie). Voor bestaande vaste toegangsnetwerken zullen modulatietechnieken alleen niet bijdragen in een efficintieverhoging: de theoretische limieten van de verbindingscapaciteiten zijn al bereikt. Als gevolg hiervan blijven er nog maar enkele manieren over om de capaciteit in deze netwerken te verhogen: Beperk de lengte van de (telefoon)kabel, waardoor hogere frequenties gebruikt worden2 met als resultaat meer bandbreedte. Scenario's met glasvezelnetwerken naar de stoep (fibre to the curb, FttC) maken gebruik van dit principe en leveren zo toegangssnelheden van 50 Mbit/s over een afstand van maximaal 300 meter. Verhoog het frequentiespectrum dat gebruikt wordt voor kabel TV gebaseerde netwerken (in CATV netwerken wordt momenteel maar een klein frequentiebereik gebruikt voor datatoegang).
Voor draadloze netwerken wordt momenteel veel onderzoek gedaan naar oplossingen voor nieuwe modulatietechnieken die optimaal zijn voor bepaalde nieuwe frequenties. Voorbeelden van dit soort digitale modulatietechnieken zijn frequency shift keying (FSK), phase shift keying (PSK), quadrature PSK (QPSK). De selectie van de modulatiemethode gebeurt op grond van de fysieke karakteristieken van het medium, en dan met name de frequentie- en de ruiskarakteristieken. Omdat vele gebruikers vaak hetzelfde communicatiekanaal delen, wordt de bandbreedte vaak onderverdeeld in meerdere kanalen. Voor de toewijzing van die kanalen zijn Multiple Access (MAC) technieken nodig; hiermee wordt geregeld hoe die opsplitsing werkt. Belangrijke MAC technieken zijn frequency division multiple access (FDMA), time division
Waarbij vaak ook andere modulatiemethoden gekozen worden die beter werken bij de nieuwe fysische structuur van het netwerk. De innovatie komt echter uit de veranderde eigenschappen van de fysieke netwerkverbinding.
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multiple access (TDMA), code division multiple access (CDMA), wide-band CDMA (WCDMA), enz. Nieuwe ontwikkelingen in deze toegangsmethodes staan zelfs hogere bandbreedtes toe per eindgebruiker of ondersteunen standaard QoS reserveringsmethoden. Echter, veel vooruitgang door innovaties op het gebied van modulatietechnieken of MAC technologie wordt niet verwacht. Daadwerkelijke bandbreedtewinst kan geput worden uit ontwikkelingen binnen nieuwe frequentiegebieden. Zo zijn er WLL oplossingen (punt-naar-punt; niet mobiel, wel draadloos) met een maximale capaciteit van 155 Mbit/s over een afstand van meerdere kilometers, en ook de wireless LANs (beperkt mobiel; nieuwe ontwikkelingen als 802.11a levert een 55 Mbit/s capaciteit over een korte afstand van max. 30 meter). Met de uitrol van landelijk dekkende mobiele netwerken zullen ook de toegangssnelheden in vergelijking met het huidige GSM sterk verbeteren. GPRS wordt momenteel uitgerold (30 Kbit/s), UMTS zal binnen 5 jaar uitgerold worden, en zal, afhankelijk van de architectuur, zorgen voor een toegangssnelheid van enkele honderden Kbit/s met een theoretisch maximum van 2 Mbit/s. Daarnaast is er een aantal nieuwe, opkomende, technieken die via nieuwe fysieke media een efficinte oplossing kunnen bieden voor het ondersteunen van breedband diensten. Communicatie via het elektriciteitsnet (power-line networks), bi-directionele satellietverbindingen, T-spectrum draadloze verbindingen en free-space optics zijn enkele van deze opkomende technologien die de potentie hebben om eindgebruikers een breedbandige verbinding te bieden. De keuzemogelijkheden nemen dus toe, de komende jaren.
Het grote geheel
Voor concrete applicaties kunnen keuzes gemaakt worden uit het voorafgaande portfolio aan mogelijkheden. Dit resulteert dan in specifieke oplossingen waarmee voor een bepaalde applicatie optimaal omgegaan kan worden met de beperkingen in toegangssnelheid. Om een beetje een gevoel te creren voor de opties, beschrijven we in deze sectie een hypothetische medische teleconsultatiedienst.
Een thuisgebonden patint wil een arts consulteren. Zowel de arts als de patint hebben een computer die uitgerust is met een webcam, microfoon en luidsprekers. De arts heeft een ADSL netwerkverbinding (1 Mbit/s downstream en 256 kbit/s upstream), terwijl de patint gebruikt maakt van een ISDN verbinding (128 kbit/s verzamelde bandbreedte). Laten we nu gaan kijken in hoeverre een applicatie de werkelijke, levensechte situatie kan benaderen door gebruik te maken van bestaande technologien. De patint maakt een connectie met zijn Internet service provider via zijn inbel account. Daarna opent hij de webpagina van de tele-consultatiedienst van de arts. De Internetverbinding van de arts is "always on" zodat hij direct opmerkt dat er een verzoek voor een tele-consultatiesessie is. Hij start onmiddellijk zijn videoconferentie applicatie. Ondanks de lage kwaliteit van de beelden (180x140 pixels, 8 bit per pixel en 5 beelden per seconde), komt de arts toch na een aantal vragen tot de voorlopige diagnose van griep. Hij vraagt vervolgens de patint om goed in het licht te gaan zitten en om de applicatie op 'hoge kwaliteit opname' te zetten. Deze instelling verandert de standaard modelgebaseerde MPEG4 compressie naar MPEG2. Deze omzetting vereist natuurlijk veel meer bandbreedte voor de audio/video gegevens maar zal de arts duidelijker beeldmateriaal van de patint geven, maar met hoog verlies aan frames (320x240 pixels, 16 bit per pixel, en een beeld per seconde). De applicatie van de arts zelf zal nog steeds de modelgebaseerde compressie gebruiken om het door hem gegenereerde verkeer naar de patint toe tot een minimum te beperken. Hierdoor houdt de patint een goede verbinding met de arts.
De arts zal in de ogen van de patint moeten kijken. Hiervoor heeft hij een afbeelding van hoge kwaliteit nodig. Om dit te doen vraagt hij de patint om in plaats van de huidige video transmissie een snapshot te sturen met de hoogste kwaliteit van zijn camera(800x600 pixels, 24 bit per pixel) en om deze foto dan ook zonder kwaliteitsverlies ("lossless" gecomprimeerd) door te sturen. Door de grote omvang van de foto duurt het een halve minuut voordat de arts de foto ontvangt. Vervolgens wil de arts de reflexen van de patint testen. Hiervoor vraagt hij de patint om de 'hoge-snelheid' instelling van de video te selecteren. De beeldsnelheid wordt nu 25 beeldjes per seconde met een kleurdiepte van 8 bits per pixel. Dit houdt wel in dat de maximaal haalbare resolutie nog maar 120 bij 80 pixels zal zijn. Deze instelling is onbruikbaar. De arts besluit dat het beter is om wat meer compressienauwkeurigheid te verliezen; details zijn niet voor deze test niet zo heel belangrijk. De applicatie schakelt daarom intern om naar MPEG4 compressie met een kwaliteitsfactor van 70. Het gevolg van deze acties is dat de arts de gewenste beelden binnenkrijgt: een video met 25 beeldjes per seconde en een beeldresolutie van 320 bij 240 pixels. Dit is genoeg om een goede indruk te krijgen van de reflexen van de patint. Dan komt er een collega van de arts binnen, iemand die in dezelfde groepspraktijk werkt. De collega heeft enkele hoge resolutie rntgen scans nodig uit de database van het ziekenhuis; hij heeft daarvoor meer bandbreedte nodig. De arts schakelt terug naar de standaard MPEG4 compressie en zet het consult als een gewone tele-consultatie voort.
De arts heeft intussen voldoende zekerheid over de diagnose van griep. Hij stelt de patint hiervan op de hoogte en adviseert een paar dagen het bed te houden. Hij ziet dat deze mededeling de patint gerust stelt. De twee besluiten dat het tele-consult kan worden beindigd. Dit scenario illustreert dat voor elke gebruikte applicatie verschillende kwaliteitsinstellingen mogelijk, en wellicht ook nodig zijn. Hierdoor moeten situatie-specifieke en applicatie-specifieke afwegingen en instellingsmogelijkheden zijn nodig om een optimaal communicatiekanaal tot stand te brengen. Een eenduidig advies is niet eenvoudig te geven. In de onderstaande tabellen wordt een overzicht gegeven van de bandbreedte eisen van de verschillende basisdiensten, de effecten van compressietechnieken, en de bruikbaarheid van de verschillende inzetbare toegangstechnieken. De tabellen illustreren per basisdienst de bandbreedte eisen in situaties met en zonder het gebruik van bandbreedte reducerende technologien. Uit de tabellen kan geconcludeerd worden dat de huidige bandbreedte reducerende technologien al behoorlijk effectief zijn in de zin dat ze de bruikbaarheid van de toegangstechnologien voor het toepassen van de diensten sterk verbeteren. Merk op dat deze gegevens gelden voor n gebruiker tegelijkertijd; wanneer er meerdere sessies lopen over hetzelfde netwerk - iets wat vrij normaal is in veel huishoudens - heeft dit een sterk negatief effect op de bruikbaarheid van een bepaald toegangsnetwerk. De onderstaande tabel is gebaseerd op het gebruik van n enkele applicatie op een bepaald tijdstip.
T E L E M A T I C A
I N S T I T U U T
Toegangstechniek
Beschikbare Bandbreedte (down/upstream) [Mbit/s] 5/0.64 51/6.5
Basisdiensten (geselecteerd door DGTP)

A/V stream. A/V mcast. Conf. Interactive rich doc. access + +
Opmerkingen3
ADSL VDSL
+ (MPEG) +
+ (MPEG) +
+/+
Tot 3.66 km - Tot 300 m - Testversie beschikbaar Voor 200 abonnees per netwerkdeel Voor 50 gebruikers per netwerkdeel Verwachte bandbreedte voor 2006 Voor 16-QAM, TDMA en 200 gebruikers Een verbeterde versie (54 Mbit/s) is naar verwachting binnen enkele jaren beschikbaar
CATV
0.2/0.015
+ (MPEG)
+/(Audio) - (Video)
+/-
PLC UMTS
0.15symmetrisch 0.2symmetrisch 2.8symmetrisch 11symmetrisch
+ (MPEG) + (MPEG) + (MPEG) + (MPEG)
+/+/-
+/+/-
LMDS
+ (MPEG) + (MPEG)
IEEE802.11 b
FSO satelliet Near Tspectrum T-spectrum
155symmetrisch 0.5- 0.05 622symmetrisch 100000symmetrisch
+ + (MPEG) + +
+ + (MPEG) + +
+ + +
+ +/+ +
Nog niet overal beschikbaar Tot 400 m Binnen vier jaar beschikbaar
De getallen in deze kolom zijn de werkelijke getallen of realistische schattingen.
Table 1. Geschiktheid van toegangstechnologien voor ruwe audio-video data streaming
Source video format

Spatial resolution [pixels] 720x575 320x240 180x120 180x120 Temporal resolution [fps] 25 25 25 12 Pixel accuracy [bit/pixel] 16 16 16 8
Source audio format

Sampling frequency [kHz] 44 44 44 44 Sample accuracy [bits/sample] 16 16 16 16 Number of channels 2 2 2 1
Relative quality class
Required bandwidth [Mbit/s]
Geschiktheid van verschillende toegangstechnologien

ISDN CATV UMTS ADSL IEEE 802.11b + +
Broadcast TV/DVD Near VHS QuarterVHS -
160 120 9.5 2.5
Table 2. Geschiktheid van toegangstechnologien voor met MPEG-4 gecomprimeerde data voor audio/video streaming
Source video format

Source audio format


ISDN CATV UMTS ADSL IEEE 802.11b + + + +
1.5 0.3 0.1 0.05
+/+ +
+ +
+ + + +
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I N S T I T U U T
Table 3. Geschiktheid van toegangstechnologien voor ruwe data voor videoconferencing
Source video format

Spatial resolution [pixels] 720x560 360x280 360x280 360x280 360x280 180x140 180x140 Temporal resolution [fps] 25 25 12.5 12.5 12.5 12,5 4 Pixel accuracy [bit/pixel] 16 16 8 8 8 8 8
Source audio format

Sampling frequency [kHz] 44 44 44 22 22 44 11 Sample accuracy [bits/sample] 16 16 16 16 8 16 8 Number of channels 2 2 2 2 1 2 1
Number of parties
Downstream/ upstream bandwidth [Mbit/s]

ISDN CATV UMTS ADSL IEEE 802.11b
4 4 4 2 4 4 4
TV/DVD Near VHS Near VHS Near VHS Near VHS 1/4 VHS -
465/155 117/39 32/10.9 10.9/10.9 31/10.3 11/3.7 2.79/0.9
+/-
+/+/+
Table 4. Geschiktheid van toegangstechnologien voor gecomprimeerde data voor videoconferencing, met compressieniveau zoals bij MPEG-4
Source video format

Source audio format

Number of parties
Downstream/ upstream bandwidth [Mbit/s]

4 4 4 2 4 4 4
3/1 1.1/0.36 0.56/0.2 0.2/0.2 0.54/0.18 0.16/0.06 0.05/0.03
+/+ +
+/+
+/+ + + + + +
+ + + + + + +
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Conclusies
Als we de gekozen categorien van breedbanddiensten als uitgangspunt nemen en vervolgens kijken naar de eisen die deze diensten aan het toegangsnetwerk stellen, naar de vooruitgang in de compressietechnieken en naar de ontwikkelingen binnen de toegangsnetwerken voor de komende jaren dan komen we tot de volgende conclusies: 1. Bestaande toegangstechnologien, met name DSL, kunnen de meest essentile diensten zoals beschreven in dit document faciliteren voor de komende vijf jaar. Hierbij nemen we aan dat per huishouden niet meer dan n persoon tegelijkertijd gebruik maakt van breedbanddiensten. Merk echter op dat technologische randvoorwaarden de uitrol van DSLtechnologie beperken, zodat in een bepaald gebied niet alle huishoudens ervan gebruik zullen kunnen maken. Het is zeer onwaarschijnlijk dat compressietechnieken een substantile bijdrage zullen leveren aan het toepasbaar maken van de verschillende diensten over de verschillende toegangsnetwerken binnen een periode van vijf jaar. Spectaculaire verbeteringen zijn op deze termijn niet te verwachten. Wel kunnen intelligente, applicatie- en domeinspecifieke compressietechnieken ingezet worden om zeer hoge compressie verhoudingen te behalen. Zonder veel afbreuk te doen aan semantische of emotionele aspecten van de over te brengen informatie, kan die informatie toch in verschillende kwaliteitsniveaus aangeboden worden. Dit kan door verschillende formaten aan te leveren, die verschillen in kleurdiepte of resolutie. Hierdoor kan men, bij een gegeven bandbreedte, de semantische waarde van communicatie toe laten nemen door het optimale kwaliteitsniveau te kiezen. Het op deze manier degraderen van informatie is een goede mogelijkheid om via de huidige netwerkinfrastructuur toch een bepaalde mate van toegang tot pseudo-breedband informatie te bieden. Een groot aantal nieuwe en interessante toegangstechnologien staan op het punt om door te breken voor praktisch gebruik. Voorbeelden van deze technologien zijn communicatie via hoogspanningskabels, hybride netwerken, free-space optics, bi-directionele satelliet verbindingen, draadloze verbindingen, et cetera. Deze technologien vergroten de mogelijkheden om breedband diensten naar de eindgebruikers te brengen, met name in dun bevolkte gebieden waar de glasvezel naar het huis nog lange tijd op zich zal laten wachten. Voor bedrijven en huishoudens in perifere gebieden is het met wireless local loop technologie ('fixed-wireless') eenvoudig mogelijk om aansluitingen van enkele Mbit/s te realiseren over afstanden van enkele tientallen kilometers. Via wireless LAN technieken kan vervolgens deze bandbreedte gedeeld worden over meerdere locaties in een straal van enkele tientallen meters.
2.
3.
4.
5.
C O N C L U S I E S
13
Summary
The Directorate4-General for Telecommunications and Post's mission includes the development of policies to safeguard the public interest with respect to telecommunication services access. This policy of safeguarding the public interest has ensured that everyone has access to telephone services for reasonable prices. In the future it is possible that public interest will also include particular Internet services, even in the broadband domain, that are considered very desirable or even essential for citizens to participate in basic social or economic activities. Consider teleconferencing, telecommuting, services to follow education, webcam application in healthcare, consulting governmental information, et cetera. Although it is widely recognised that deployment of a fibre access infrastructure will enable real service integration, it is also an economic reality that the national rollout of such an infrastructure may take a considerable number of years, assuming that an all-fibre access connection is a realistic perspective for households located in rural areas. This study explores the perspectives for households and companies in regions without a fibre-infrastructure from a technology point of view. The central question that is addressed in this report is:
To what extent do new and upcoming technological developments, such as data compression, contribute to more efficient use of the current capacity of actual and future fixed and mobile communication networks, so that these networks can support high-bandwidth information and communication services between small-scale business companies and private users in a couple of years? In five years time, to what extent will it be possible to deploy services via these networks which are called high-bandwidth services now, such as services based on video and audio streaming, by data compression or smart algorithms or protocols? What is by approximation the expected highest possible transmission bandwidth required for these services?
We were asked to answer these questions and to assess the impact of new, so-called bandwidth reducing, technologies. These are technologies that could potentially be used to reduce the need for data transfer while delivering broadband services. In other words, technologies that allow the delivery of broadband services over current, common access networks that are typically referred to as narrow or medium band. It must be pointed out that the above questions cannot be answered in absolute terms. The answers will depend both on policy decisions that qualify particular services as universal as well as on technological developments. Both cannot be predicted with absolute certainty. This report therefore reflects our vision rather than absolute wisdom. Moreover, given the uncertainties regarding the need for electronic services and the speed of technological developments we will consider a time frame of five years. The Netherlands already has a quite dense communication infrastructure for both fixed and mobile services, covering virtually the whole country. So it is worth investigating to what extent,
4
The Directorate is part of the Dutch Ministry of Transport and Communication (Ministerie van Verkeer en Waterstaat).
S U M M A R Y
15
and under what circumstances, current and emerging access technologies (ADSL, CATV, GPRS, satellite, WLL, etc.) can satisfy the bandwidth requirements of essential broadband services in the near future. Moreover, it is important to evaluate whether significant technological improvements can be expected here, or whether their current performances are close to their intrinsic limits. This study gives insight in the technological developments in this domain that are important for DGTP.
The approach
A key issue is to evaluate to what extent broadband services can be deployed over existing access networks. Therefore, the first thing to estimate is the bandwidth requirements for most important applications and services, and their relevance with respect to the overall network traffic. However, it is impossible to predict exactly which broadband services will be considered essential in the coming years. Together with the Directorate-General for Telecommunications and Post we have identified four categories of "basic" services that are considered to be characteristic for the essential broadband applications to which universal access should be guaranteed. In other words, we have listed numerous potential essential applications for education, healthcare, telecommuting, etc. Subsequently, we have identified four basic categories of services that include most of these essential applications. The following table summarises the result of this analysis. Examples of essential applications Video (news) on demand, time-shifted virtual radio-TV stations Internet radio and TV, distance learning for closed groups, tele monitoring Tele consulting, internet telephony, videoconference, telecommuting Access to digital libraries, national archives, multimedia e-mail, individual distance learning, telecommuting Basic categories of services Audio and video streaming Audio and video multicast Conferencing Interactive rich documents access
We conclude that, whereas the exact nature of the future essential applications is unknown, we may reasonably assume that they can be deployed as long as the basic services listed in the table can be deployed. Moreover, these basic services put such diverse requirements on the infrastructure in terms of bandwidth and quality of service that it is likely that they will cover the requirements of future essential applications. It should be noted that the above categories do not cover the entire range of applications that could require a broadband connection. For instance, applications where data is not directly destined to a human (inter-system applications) cannot be included in any of these categories. Although some inter-system applications (e.g. grid computing) have been given much attention lately, we estimate that there will be no need for such applications in the residential or small business environment within five years. Also, information provisioning by end-users is not part of the list. Although this may become quite important, providers can easily supply hosting services for this. Hence, it is not relevant for this discussion.
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T E L E M A T I C A
I N S T I T U U T
We started with an analysis of the requirements put onto network infrastructures by the four categories of basic services. These are requirements in terms of 1. 2. 3. Bandwidth to be offered by the networks to deploy the services to their full extent, Quality of service (delay, packet loss, etc.) to be offered by the networks, Bandwidth to be offered by the networks to deploy the services in a degraded mode.
The table below summarises the bandwidth requirements for the basic categories of services. Note that not only the volume of data, but also temporal relations between sent and received data have an important impact on the required bandwidth. The quality column indicates whether strong requirements exist with respect to delay variance and delay. "Low" means that no strong requirements exist (there is no audio/video involved), medium quality means that delay is not very relevant (as long as the delay is constant), and high quality means that low delay is necessary (communication is interactive).
Basic service
Application
Minimum bandwidth [Mbit/s] 0.05 0.08 20 0.4 0.128 0.064 0.5 0.5 0.014
Maximum bandwidth [Mbit/s] 0.2 0.2 340 8 1.5 0.25 2 20 6
Required quality Medium Medium Medium Medium High High Low Low Low
A/V streaming A/V Multicast Conferencing (bandwidth per party) Interactive rich documents access
Time-shifted radio stations MPEG Audio streaming HDTV MPEG video TV Video conferencing Internet telephony e-learning, e-government Tele-medicine Telecommuting
It should be noted that applications like tele-medicine, or telecommuting may include videoconferencing or internet telephony as well. In such cases, the quality requirements will, of course, be "High" rather than "Low". Each of the broadband applications in the above table makes use of several components. First, there is the encoding-decoding component that is used to map natural or physical signals to digital streams. These streams are distributed to other entities (e.g. from a television studio to an end-user, or between end-users in conferencing services) using networks. This is illustrated below.
S U M M A R Y
17
application encoder transcoder Multimedia Distribution Techniques (Access) network
application decoder
(Access) network
This study addresses efficiency gains in the coloured parts of this figure. In more detail, we focus on advances in the following areas:
Compression is one of the most effective ways of increasing transport efficiency. Particularly
data resulting from natural aural or visual signals is highly compressible using lossy compression techniques (i.e. the original signal cannot be reconstructed completely). Compression techniques are implemented in encoders, transcoders and decoders. Distribution techniques are used to optimise the delivery of content. Examples are networklevel multicasting, on-the-fly transcoding, or pre-fetching of content. Coding and modulation techniques are intrinsic parts of (access) network technologies, and map data onto physical communication media (this study limits the physical media to wireless techniques, cable TV networks and telephony copper wires). New modulation techniques may result in large efficiency gains (compare ADSL with analogue telephony). These techniques are implemented in interface boards.
Advances in data compression
Data compression methods can be subdivided in two categories: lossless compression and lossy compression. Lossless compression basically performs redundancy reduction. Their compression ratio is rather poor (e.g. 2:1 for natural images). Advances of lossless compression techniques take advantage of additional properties of original data to achieve a more convenient statistical representation of data samples. In the case of images, for instance, lossless compression can be improved when applied separately to the different objects that make up a given scene. This way, an optimal representation is achieved for the image as a whole, and not only for every pixel separately. Either way, the usual compression ratio achievable for natural images does not get higher than 5:1. In the case of video sequences, lossless compression can reach a ratio of about 10:1. For TVquality video signals (640x480 resolution, 24 bits colour, 25 frames/second = 18 Mbit/s) this still amounts to a bandwidth of 2 Mbit/s. Without audio! Lossy compression techniques imply certain irreversible transformations on the original data, causing thus some unrecoverable loss of information. This is though acceptable when compressed data mediates an interpersonal communication, and if a human observer cannot perceive data alteration. This way, lossy compression reduces not only the redundancy of original data, but also its irrelevancies. A large number of lossy compression techniques have been proposed for speech, audio, graphics, image, and video. The strategies followed in all these are either to find an approximation of the original data samples (e.g. quantised transform coding), or to define a model (recipe) by which data can be reconstructed from a reduced set of parameters and a large number of previously stored prototypes (e.g. vector quantisation, principal components analysis).
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T E L E M A T I C A
I N S T I T U U T
Only a few of these techniques have been widely accepted, and eventually incorporated in major compression standards and technologies. For now, the ISO series of JPEG and MPEG open standard specifications led to the development of most efficient technologies for image and video compression. The state of the art products based on JPEG2000 and MPEG4 offer not only high compression ratios (e.g. 100:1 for MPEG4 video at near DVD quality), but also provide adaptation mechanisms to accommodate different bandwidths and terminal capabilities. However, the complexity of the encoder is high, and products may suffer from a significant delay that still prohibits their usage in peer-to-peer applications like videoconferencing. Despite the plethora of methods and algorithms for lossy compression, it is still hard to point out a theoretical limit for lossy compression. The reason is twofold. First, there is no comprehensive model of perception to allow evaluating the minimum amount of information needed for understanding a certain aural or visual scene. Second, even if such a model was available, subjects experience, immediate goals, and context would always influence the perception. Consequently, different limits for lossy compression could eventually be defined for different applications. For instance, in video telephony the parties involved only need to communicate verbally and possibly emotionally. This is achieved, obviously, through an audio-video communication. Nevertheless, the only information they really need is what the other party is saying, and how does he or she act during the conversation. So one could only send the words and the facial expressions to achieve the same results. Together with an application that actually generates audio from the words using a pre-fetched audio-model of the other party, and possibly even an animation of a pre-fetched model of the head based on the transmitted facial expressions, a bitrate of 300 bit/s would be sufficient to have a realistic discussion. However, these methods are quite academic, and require a lot of configuration on end-users terminals. Therefore, it is not likely that these will be fully deployed within the next five years. If we take into account the compression techniques currently known and subject to standardisation we conclude that it is unlikely that compression techniques will be improved very much for the next five years. However, adaptive coding schemes like MPEG4 will be used more and more, in particular aimed at people with lower access-speed.
Advances in distribution technology
Distribution mechanisms can be used to gain efficiency in the transport of multimedia. They are typically used inside networks, to optimise operators resource usage. For instance, by caching content, replication of content to locations close to particular user-communities, and so on. Therefore, they fall out of scope for this study. However, two aspects of distribution technology are interesting:
Adaptive multicasting. This is used in point-to-multipoint content distribution settings, where

each intermediate system is not only splitting content to its clients, but also may transcode the content from one format into another. These techniques can be used to give low-speed access to high quality content in a scalable way. Similar techniques can be used in conferencing bridges to support multi-party conferences with heterogeneous terminals. Pre-fetching of content. With this technique unused bandwidth is used to pre-fetch large amounts of data. This is a promising method, although reservations apply: it is only useful when the types of data that need to be accessed can be determined ahead of time (which may be the case for asynchronous telecommuting or tele-education scenarios).
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Advances in Access technology
For existing wireline access networks, modulation techniques will not result in efficiency gains: the theoretical limits of line capacities have already been reached. Hence, the only possible ways to enhance the capacity of these networks are to:
limit the length of each link in DSL networks (which allows one to use higher frequencies,
and, hence, obtain more bandwidth). Fibre-to-the-curb scenarios use this method, and deliver access speeds of 50 Mbit/s over distances of 300 meters max. Increase the frequency range used for cable-based networks (in CATV networks currently only a small frequency range is used for data-access). For wireless networks there is a lot of effort going on in defining new modulation techniques that are optimal on particular new frequencies. Examples of digital modulation techniques are frequency shift keying (FSK), phase shift keying (PSK), quadrature PSK (QPSK), etc. Modulation methods should be carefully selected based on the physical characteristics of the medium, notably the frequency and the noise characteristics. For wireless networks also several logical channels are multiplexed in the same frequency band (using e.g. frequency division multiplexing (FDM), time division multiplexing (TDM)). As many users often share a single communication channel, Multiple Access (MAC) techniques are necessary that define how a medium is subdivided effectively into multiple channels. Important MAC techniques are frequency division multiple access (FDMA), time division multiple access (TDMA), code division multiple access (CDMA), wide band CDMA (W-CDMA), etc. There are advances in these access methods that allow for higher bandwidths or that natively support QoS reservation methods. However, many advances cannot be expected from innovative modulation techniques or MAC-developments. Real bandwidth gains can be obtained from the deployment of new frequency ranges in WLLsolutions (point-to-point with up to 155 Mbit/s over distances of approximately 40 km) and in wireless LANs (802.11a has 55 Mbit/s capacity; short range). The roll-out of mobile networks will also improve access speeds compared to the current GSM figures: GPRS is currently rolled out (30Kbit/s), UMTS will be rolled out within 5 years, and, depending on the deployment architecture, will provide at most a few hundreds of kbit/s (theoretical maximum is 2 Mbit/s, when close to a base station). There are also emerging technologies that might be deployed within the next year. These technologies could provide an efficient solution for supporting residential broadband services. Power line communication, free-space optics, bi-directional satellite links, "T-spectrum" wireless links are just a few of the emerging technologies having a high potential for providing broadband connections to residential users. This wealth of opportunities asks for a proper gateway-concept between the in-house infrastructure and the different types of access networks, both wireline and wireless. This residential gateway concept still requires additional research.
Putting it all together
To give the reader a clearer perspective on the complex relations between bandwidth requirements, quality of service, and application functionality, we describe a hypothetical service for medical tele consulting. This example illustrates the possibilities and limitations imposed by
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I N S T I T U U T
the performances of current access networks, and gives insight into some of the possible solutions that exist when application-specific measurements are taken.
Imagine that a practitioner needs to consult a patient who cannot leave his house. Both the practitioner and the patient have a computer equipped with microphones, webcams, and speakers. The practitioner has a network connection via ADSL (1Mbit/s downstream and 256 kbit/s upstream), while the patient uses an ISDN connection (128 kbit/s aggregated bandwidth). Let us analyse to what extent such an application can meet real life requirements using the currently available technologies. The patient connects to his Internet service provider using his dial-in account. Then, he opens the web page of the practitioner's tele-consulting service. The practitioner's Internet connection is "always on", so he immediately notices a request for a tele-consulting session, and starts his videoconference application. He knows that the images on his screen (180x140 pixels, 8 bit per pixel, and 5 frames per second) are not accurate enough to rely on them, but after a few questions he decides his patient shows flu symptoms. He then instructs the patient to sit properly in the light, and to select from his application menu the settings that are for close examination. This setting changes the default compression from model-based MPEG4 to the plain old MPEG2. Of course, this will require a lot more bandwidth for the audio-video data, but at least will show the practitioner the "actual" images of the patient at a spatial resolution of 320x240 pixels, a colour depth of 16 bit per pixel, and a temporal resolution of one frame per second. The practitioner instead, will continue using his model-based compressor in order keep the traffic generated by him to a minimum, while still giving his patient good feedback. Initially, the practitioner needs to look at the eyes of his patient, and for this he needs a quite good picture quality. Therefore, he asks the patient to stop the current video transmission, to take a snapshot at the maximum quality of his camera (800x600 pixels, 24 bit per pixel), and then send it lossless compressed. Due to the size of this image, it takes about half a minute before the practitioner finally receives it. Next, he wants to assess the alertness of the patient. Therefore, he asks the patient to select from his video settings menu the high-speed setting. This sets the frame rate to 25 frames per second, and a colour depth of 8 bits per pixel. Because this means that the maximum achievable resolution is degraded to only 120 by 80 pixels, which is really useless, the practitioner decides it's better to lose some compression accuracy since the details are not that important for this phase of diagnosis, and asks the patient to select a region-based MPEG4 compressor with a quality factor of 70. At the same time, they agree to stop the audio communication and use a chat application for a while. This way, the practitioner is able to receive the needed information: a video stream of 25 frames per second, and image resolution of 320 by 240 pixels, which is quite enough for assessing the alertness of his patient. Then a colleague pops up; he needs to download a few highresolution X-ray images from the hospital's database and needs more bandwidth. Hence, the practitioner writes the patient to switch back on the model-based MPEG4, so they can continue their dialogue in audio/video mode. The practitioner has now enough evidence to set the flu diagnostic. He informs his patient and recommends him to stay in bed for a few days. Then both agree to end this tele consulting session.
This scenario illustrates that during each concrete application the quality requirements may differ. Hence, situation-specific and application-specific decisions and controls can be used to optimise application-behaviour for particular situations.
S U M M A R Y
21
The following tables summarise the bandwidth requirements of some of the basic services, the effects of bandwidth reducing techniques, and the suitability of various access techniques for their deployment. The tables illustrate per basic category of service the bandwidth requirements for two cases, i.e. without and with the use of bandwidth reducing technologies. It may be concluded that current bandwidth reducing technologies are already quite effective in the sense that they improve the suitability of access technologies for deploying the basic services. Note that running multiple user sessions over the same access link (as may be very common in many households; particularly those with children!) may have a strong negative impact on the suitability of particular access networks. The table below is based on the use of a single application at a given moment in time.
Access Technique
Achievable Bandwidth (down/upstream) [Mbit/s] 5/0.64 51/6.5
Basic categories of services (selected by DGTP)

A/V stream. A/V mcast. Conf. Interactive rich doc. access + +
Remarks 5
ADSL VDSL
+ (MPEG) +
+ (MPEG) +
+/+
Assuming a range of 3.66 km - Assuming a range of 300 m - almost available Assuming 200 subscribers per network segment. Assuming 50 users per section The bandwidth expected for 2006 Assuming 16-QAM, TDMA, and 200 users - another version providing 54 Mbit/s will be available within a few years
CATV
0.2/0.015
+ (MPEG)
+/(Audio) - (Video)
+/-
PLC UMTS
0.15symmetric 0.2symmetric 2.8symmetric 11symmetric
+/+/-
+/+/-
LMDS
+ (MPEG) + (MPEG)
IEEE802.11b
FSO satellite
155symmetric 0.5- 0.05
+ + (MPEG) + +
+ + (MPEG) + +
+ -
+ +/-
- Not yet widely available - over a range of 400 m To be deployed within four years
Near Tspectrum T-spectrum
622symmetric 100000symmetric
+ +
+ +
The figures in this column are either actual, or realistic estimations of what is possible to achieve
22
Table 5. Suitability of access technologies for raw data audio/video streaming
Source video format

Source audio format

Suitability of several access technologies

160 120 9.5 2.5
Table 6. Suitability of access technologies for MPEG-4 compressed data audio/video streaming
Source video format

Source audio format


1.5 0.3 0.1 0.05
+/+ +
+ +
+ + + +
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23
Table 7. Suitability of access technologies for raw data video conferencing
Source video format

Source audio format

Number of parties
Downstrea m/ upstream bandwidth [Mbit/s]

4 4 4 2 4 4 4
465/155 117/39 32/10.9 10.9/10.9 31/10.3 11/3.7 2.79/0.9
+/-
+/+/+
Table 8. Suitability of access technologies for compressed data video conferencing, assuming a compression approaching MPEG-4 performance
Source video format

Source audio format

Number of parties

4 4 4 2 4 4 4
3/1 1.1/0.36 0.56/0.2 0.2/0.2 0.54/0.18 0.16/0.06 0.05/0.03
+/+ +
+/+
+/+ + + + + +
+ + + + + + +
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T E L E M A T I C A
I N S T I T U U T
Conclusions
Taking into account the basic categories of services, the requirements they put onto access networks, the advances in compression techniques and the advances in access networks for the coming years, we reached the following conclusions. 1. Existing infrastructure technologies, in particular DSL, can accommodate most of the essential services considered in this study for the coming five years, assuming that at a given point in time, one person per household would make use of one such service. Note, however, that rollout of DSL-technology is hampered by physical limitations that may restrict deployment to only a part of the households. Compression techniques are unlikely to substantially alter our assessment of the suitability of the various existing infrastructure technologies for the deployment of essential services, since no spectacular improvements are likely to occur within five years. However, intelligent, application and domain-specific compression techniques can be used to obtain much higher compression rates. A service can be delivered at different levels of quality, since information can in many cases be properly degraded without losing the semantic or emotional communication value. Many of the future services offered in high quality (for example colour video) could be gracefully degraded to a lower quality (for example black and white video) without losing their core functionality. Hence, given a fixed bandwidth, one can increase the semantic value of communication by degrading the perceptual quality of the signals. Graceful degradation therefore is a fallback option to deploy essential services via current network infrastructures. An extensive number of new access technologies that could be interesting for the studied applications are on the verge of deployment. Examples of these technologies are power line communication, hybrid networks, free-space optics, bi-directional satellite links, wireless links, et cetera. These have a high potential for providing broadband connections to residential users, particularly in rural areas where fibre to the home or fibre to the curb is not available.
2.
3.
4.
5. Wireless local loop (fixed wireless) technologies are possible solutions for offering
connections of a few Mbit/s over distances of tens of kilometres to residential or business users situated in peripheral areas. Using wireless LAN techniques, the bandwidth can then be used from several locations within a range of a few tens of meters.
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25
1 Introduction
1.1
Goal and scope of the project
The Directorate-General for Telecommunications and Post's mission includes the development of policies to safeguard the public interest with respect to access to telecommunication services. This policy of safeguarding the public interest has ensured that everyone has access to telephone services for reasonable prices. In the future it is possible that public interest will also include particular Internet services, even in the broadband domain, that are considered very desirable or even essential for citizens to participate in basic social or economic activities. Consider teleconferencing, telecommuting, services to follow education, webcam application in healthcare, consulting governmental information, et cetera. Although it is widely recognised that deployment of a fibre access infrastructure will enable real service integration, it is also an economic reality that the national rollout of such an infrastructure may take a considerable number of years, assuming that an all-fibre access connection is a realistic perspective for households located in rural areas. This study explores the perspectives for households and companies in regions without a fibre-infrastructure from a technology point of view. This is not only based on existing access technologies and expected advances in access technologies, but also on possible breakthroughs in compression techniques. There has been a lot of attention lately on such breakthroughs that might result in extremely low-bandwidth video transmissions. In this report we try to evaluate to what extent current and emerging access services would give everyone the possibility of using broadband applications. A more accurate description of this goal is given by the following research question formulated by DGTP: "In hoeverre maken nieuwe aanstaande technologische ontwikkelingen, waaronder compressie, het mogelijk om de bestaande capaciteit in het huidige en aanstaande vaste en mobiele aansluitnet beter te benutten zodat over een aantal jaren daarlangs ook breedband-informatie- en communicatiediensten naar particulier en kleinbedrijf te transporteren zijn. Hoever zal de praktijk over 5 jaar zijn gevorderd om via deze netwerken diensten aan te bieden die nu als 'breedband' te boek staan (zoals bijvoorbeeld video en audiostreaming), door gebruik te maken van compressie en/of slimme algoritmes en protocollen. Waar zal naar verwachting ongeveer de grens van het mogelijke liggen gerekend in termen van de transmissie-bandbreedte die betreffende diensten vereisen." ("To what extent do new and upcoming technological developments, such as data compression, contribute to more efficient use of the current capacity of actual and future fixed and mobile communication networks, so that these networks can support high-bandwidth information and communication services between small-scale business companies and private users in a couple of years? In five years time, to what extent will it be possible to deploy services via these networks which are called high-bandwidth services now, such as services based on video and audio streaming, by data compression or smart algorithms or protocols? What is by approximation the expected highest possible transmission bandwidth required for these services?")
I N T R O D U C T I O N
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Therefore, the context of our research will be limited to the existing access networks infrastructure, and to the one likely to emerge within five years, and will only address the problem of optimising data traffic by: 1. Minimising the bandwidth required for transmitting a certain quantity of information. This is achieved by keeping from the original data only those elements that are needed for a (more or less) accurate reconstruction of the original data. This process will be referred to as compression. Maximising the data flow through a certain physical communication channel, while maintaining the maximum transmission error of messages below a given limit. This is obtained by efficient coding methods, modulation and multiple access techniques inside access networks. Using optimal distribution techniques in order to minimise duplicate encoded streams over the network. This helps in preventing overloaded backbone connections and servers by efficient content-distribution mechanisms and replicated data-storage.
2.
3.
The relation between these aspects is depicted below. Networked applications use compression techniques implemented in encoding-decoding components, which map natural or physical signals to digital streams. These streams are distributed to other entities (e.g. from a television studio to an end-user, or between end-users in conferencing services) using networks. This is illustrated in the system description below.
application encoder transcoder Multimedia Distribution Techniques (Access) network
application decoder
(Access) network
However, solving these three problems separately, may not lead to a global optimum with respect to bandwidth usage for a given network infrastructure. On the other hand, a theoretically optimal solution may often be impossible to achieve in practice due to a number of side issues, like implementation costs, legacy components, inadequate business models, or user resilience, et cetera. Therefore, our conclusions will be formulated in terms of i) Estimated bandwidth requirements for residential and (small) business users; ii) Assessment of improvements that can be achieved for compression, coding and distribution techniques; iii) Overview of state-of-the-art technologies for compression, coding and distribution; iv) Current and near-future capabilities of existing access networks; v) Possibly discovery of (very) relevant adjacent developments as aspects, vi) Mapping user requirements (i), weighted by a factor resulted from (ii) and (iii) onto bandwidth availability (iv). Obviously, these conclusions do not have an absolute character, being the result of a limited perspective of the whole matter. The elements taken here into account are mainly technical. Some side-steps are made to directly adjacent matters, like perception and quality of service. 28
Economical and social aspects were ignored, although they may have a great influence on the evolution of both requirements and availability of network resources.
1.2 Structure of the report
A key issue in the research question stated above is to evaluate to what extent broadband (wideband) services can be deployed over wideband (narrowband) networks. Consequently, on a higher level, the logical structure of this report is as follows:
Identification of bandwidth requirements for most important applications and services.

Which of these are broadband (wideband), what is their relevance with respect to the overall network traffic?
Assessment of the possibilities for reducing the bandwidth requirements while preserving
the quality of service parameters by means of compression and distribution techniques. What are the theoretical limits, if any, what are the practical constraints, what are the current achievements?
Assessment of the technical characteristics of current and emerging access-network

technologies.
Matching the original bandwidth requirements, downscaled with a factor resulted from
applying bandwidth saving techniques, with network parameters. Section 2 is dedicated to the analysis of relevant applications and services. After selecting several categories, we present their most important requirements in terms of bandwidth and quality of service. Their relevance for the present study is assessed according to their bandwidth requirements. Section 3 gives a survey of the most important principles for reducing bandwidth requirements in multimedia communication services using compression. Section 4 addresses distribution techniques. Although they are not part of the main focus, some advances here might influence the efficiency of access networks. Section 5 addresses the key issue in developing new communications services, the so-called last mile. A description of currently used and emerging access technologies is given there, along with a description of the technology used inside these networks. In section 6 we combine the results of sections 3, 4, and 5 with the requirements put forward in section 2. For the reader interested in deeper technical details, background information and examples, these can be found in the three annexes. These discuss compression algorithms, datalink and physical layer technologies, and detailed descriptions of access technologies, respectively.
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2 Communication services and applications
In this section we intend to further define the scope of our research, by identifying those broadband applications or services that may become more general in the next five years' time and to which the government may adhere. We will estimate their bandwidth and quality of service requirements. These applications will be further used throughout the remaining of the report as main guidelines.
2.1 Networked multimedia applications
Currently, thousands of different multimedia applications and services are offered and used. Virtually, every field of economic and social activities makes use of some kind of (multi-) media communication. So we will identify categories of applications or services that include essential functionality elements that are shared by most multimedia applications. In order to that, we propose a number of criteria that will help identifying such relevant categories: media types used for information exchange: data, graphic, voice, audio, image, video temporal dependencies: Interactive applications, like telephony, require a very low delay and low delay variance (synchronous). Non-interactive applications have lower delay requirements. These can be subdivided into isochronous applications like video streaming or voice-mail access that have requirements with respect to delay variance (jitter), and asynchronous applications like web-browsing that have no strong quality-requirements. number of participants/topology: point to point, point to multipoint, multipoint to multipoint
2.1.1 Relevant categories of networked multimedia applications
For the present study we selected a number of categories of networked multimedia applications that are most relevant and representative with respect to their bandwidth requirements and their usage. These are presented in Table 9, together with some of their most relevant properties.
Table 9. Categories of applications
Category of applications Audio and video streaming Video multicast
Typical topology/configuration Point-to-point, asymmetric
Temporal dependencies Isochronous, constant delay Synchronous
Examples video on demand, timeshifted virtual radio and TV channels Real-time virtual radioTV channels, distance learning Internet telephony, video conference e-government applications, ecommerce, transactions services, multimedia email, digital libraries, etc.
Point-to-multipoint, asymmetric (Multi-) point-to-(multi-) point, symmetric Point-to-point, asymmetric
Conferencing applications Interactive richdocument access
Synchronous Asynchronous, variable delay
Note that the difference between streaming and multicast applications is not directly related to bandwidth, but rather to quality of service, which may influence the way these applications can be used in combination with others, or in multiple instances.
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2.1.2
Bandwidth requirements
The table below gives an estimate of the bandwidth requirements for several applications from each of the categories considered earlier. Note that some of the applications fall into several categories. For instance, tele-medicine may consist of only access to existing patient files (which is part of interactive rich documents access), but might as well consist of remote video observation of patients (conferencing). Some applications may even combine these features into a single application.
Table 10. Bandwidth requirements for several examples of applications
Category
Application
Minimum bandwidth (Mbit/s) 0.05 0.08 20 0.4 0.128 0.064 0.5 0.5 0.014
Maximum bandwidth (Mbit/s) 0.2 0.2 340 8 1.5 0.25 2 20 6
A/V streaming A/V Multicast Conferencing (bandwidth per party) Interactive rich documents access and file exchange
Time-shifted radio stations MPEG Audio streaming HDTV MPEG video TV Video conferencing: e.g. in telemedicine, telecommuting Internet telephony e-learning, e-government Tele-medicine (multimedia document access only) Telecommuting (file access)
It should be noted that the above categories do not cover the entire range of applications that could require a broadband connection. For instance, applications where data is not directly destined to a human (inter-system applications) cannot be included in any of these categories. Although some inter-system applications (e.g. grid computing) were given much attention lately, we estimate that there will be no need for such applications in the residential or small business environment within five years. Also, information provisioning by end-users is not part of the list. Although this may become quite important, providers can easily supply hosting services for this. Hence, it is not relevant for this discussion.
2.2 Quality of service parameters
Quality of service (QoS) is the ability of a network element to have some level of assurance that its traffic and service requirements can be satisfied [50]. Quality of service does not have a direct relation with bandwidth, but rather with the way this is allocated to different concurrent applications. Still, it is important to point out several QoS parameters and to show how their values or the lack of bandwidth can influence different applications. In the technical literature there are referred a large number of quality of service parameters. Still, for the purpose of this study it is appropriate to discuss only those that directly influence data traffic efficiency: delay, jitter, skew, packet loss, and average error rate. Although these terms are generally used in many different contexts, for our discussion we implicitly assume a packet switched network, so the parameters defined below characterise the behaviour of data packets transported over such networks.
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T E L E M A T I C A
I N S T I T U U T
Delay is the elapsed time for a packet to be passed from the sender, through the network, to
the receiver. The higher the delay, the greater the stress that is placed on the transport protocol to operate efficiently. Higher levels of delay imply greater amounts of data held "in transit" in the network. Besides the effects on the transport protocols, there is a more obvious effect at the application level. For example, in real time or interactive applications large values of delay would render these non-responsive. Jitter is the variation in end-to-end transit delay. In mathematical terms it is measurable as the absolute value of the first differential of the sequence of individual delay measurements. High levels of jitter in applications based on continuous media, such as audio or video, causes the signal to be distorted, since there is a strict temporal relation between consecutive data packets. However, the effect of jitter can be compensated by increasing the receiver's reassembly playback queue (or buffer). This is mainly appropriate for streaming applications, but makes interactive sessions very cumbersome to maintain. Skew is an effect that may occur in multiple flow transmissions, where a strict temporal relation between flows has to be kept (synchronism). For example, in an audio-video streaming application, a difference between the delay values for audio and video data packets can lead to skew, i.e. a relative shift between the audio and video data flows of a particular stream. Packet loss and average error rates are two different QoS parameters that define the reliability of a transmission system, or a transmission medium. Reliability can also be a characteristic of the switching system, in that a poorly performing switching system can alter the order of packets in transit, or even drop packets through transient routing loops. Erroneous or missing data packets may have different effects on the transmission, depending on the kind of application and on the underlying transport protocols. For instance, a missing or erroneous packet in a transmission of MPEG video can lead to the impossibility of correctly decoding subsequent data, up to a certain temporal extent (error propagation). On the other hand, some transport protocols will request a retransmission every time a packet loss is detected, inducing thus an artificial increase in traffic. ITU-T Y.1541 Draft Recommendation [25] defines four classes of QoS for IP (Internet Protocol) networks, and for each of them suggests some values for the above parameters, as shown in Table 11. These values are still under study, and some applications already suggest higher requirements. For example, a service class 1 MPEG-2 video streaming requires IP packet loss rate of about 10-5.
Table 11. Recommended values for several QoS parameters in IP networks
QoS parameter
Class 0 (synchronous)
Class 1 (interactive) 400 ms 50 ms 10-3 10

-4
Class 2 (noninteractive) 1s 1s 10-3 10

-4
Class 3 (unspecified) Unbounded Unbounded Unbounded Unbounded
Delay (IP packet transfer delay) Jitter (IP packet delay variation) Packet loss rate Packet error rate
150 ms 50 ms 10-3 10
-4
Turning back to the categories of applications under considerations, Table 12 gives an estimation of the required QoS class for some of them. Although there is no explicit dependency between bandwidth requirements and QoS parameters, by combining Table 10 and Table 12, one can see a correlation between bandwidth and QoS requirements.
Table 12. QoS classes required by different applications
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A N D
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33
Category of applications A/V streaming & A/V multicast Conferencing Interactive rich document access
Example application Time-shifted radio stations MPEG Audio streaming HDTV MPEG video TV Video conferencing Internet telephony e-learning, e-government Tele-medicine Telecommuting
Recommended QoS class 1 1 1 1 0 0 1 1 1
Minimal QoS class 2 2 1 1 1 1 2 2 2
2.3
Graceful degradation and adaptabilit y
Bandwidth and quality of service are either impossible to guarantee for some networks or protocols (e.g. the transfer control protocol - TCP), or otherwise it may be the service provider's policy to allow a certain variation of these parameters. In such cases, certain applications will cease functioning, while others will provide less functionality, but still perform essential operations. Generally, graceful degradation refers to the property of a certain system to decrease its performances in a continuous and smooth manner, when parts of it are altered or removed. On the contrary, if a system ceases functioning when altering a single parameter, or removing a single part, it has the property of catastrophic degradation. In a networked multimedia communication service, graceful degradation is an important requirement. Besides the mechanisms implemented at the network or data transport level, graceful degradation can also be implemented within the applications that require this property. For instance, a web page could display, besides the core information, high resolution images, animations, play sounds associated with some events, if there is enough bandwidth and QoS for the whole path between the web server and the end user. When network performance decreases, the user's web browser could only display the textual information, or lower resolution images. In other applications, graceful degradation is an intrinsic property of transmitted data. This is the case for the so-called progressive formats for images, where visual information is ordered in a hierarchy of importance. The most important fraction (about 2%) is transmitted first to provide a low-resolution version, then quality improves as more information arrives. From the point of view of functionality, graceful degradation is related to adaptability, in the sense that some quality parameters can be modified to match a certain performance level. The difference, however, resides in the trigger of the adaptation process. Here, user's terminal capabilities may require the alteration of quality parameters. For instance, a streaming service could provide a full resolution (720x575 pixels), true colour (24 bit per pixel), and full motion (25 frames per second) MPEG2 video. A user instead could only display on his (hypothetical) mobile video terminal only 180x144 pixel images, with 16 bits per pixel, at 10 frames per second. Consequently, an adaptable application would first assess the capability of user's terminal and then send the only data that can be effectively displayed by that particular terminal, which in this case would be about 60 times less than the original. Actually, such an adaptation process is more complex since it also has to take into account the bandwidth availability and QoS parameters. This is also covered in the chapter on distribution technology (section 4).
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I N S T I T U U T
Both graceful degradation and adaptability cannot easily sustain arbitrarily large alterations within a session. So every application is only gracefully degradable or adaptable within certain limits. Table 13 gives a few examples in this respect, for the same set of sample applications considered earlier.
Table 13. Limits of graceful degradation
Category of applications A/V streaming
Example application Time-shifted radio stations, MPEG audio streaming HDTV8 MPEG video TV
Gracefully degradable, or adaptable parameter(s) Sample resolution and accuracy7
Absolute or relative limits6 8 k samples/sec, 8bit/sample (mono)
A/V multicast
Spatial and temporal resolution, pixel accuracy Spatial and temporal resolution, pixel accuracy Sample resolution and accuracy Image and graphics resolution, animations, sound samples resolution and accuracy, video resolution (spatial and temporal) and accuracy
300x200 pixels, 25 frames/second, 16 bit/pixel 180x144 pixels, 2 frames/sec, 8 bit/pixel 8 kilo samples per second, 8 bit/sample Instance dependent
Conferencing
Video conferencing
Internet telephony Interactive rich document access e-learning, egovernment, telecommuting, telemedicine
These limits are not specified by a certain standard. They only reflect our opinion of a lower bound for maintaining a minimal functionality. 7 Assuming a decoded/de-compressed sequence of samples. 8 Since this application differs from MPEG video TV only by signal's quality, graceful degradation and adaptation do not apply here.
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3 Compression techniques
In this chapter we investigate different possibilities for reducing the amount of data prior to sending it over a communication channel. These techniques are referred to as compression, or more generally source coding. Because of the high complexity of this topic, we will restrict the scope to what we believe to be the most challenging part, which is audio and video compression, although some of the techniques are suited for generic data as well. Compression techniques are implemented in encoders and decoders, either as software components (for low-complexity tasks) or as hardware components when either real-time performance is needed application application for encoding, or for decoding in lowtranscoder decoder encoder performance environments like cellMultimedia Distribution Techniques phones. These are the greyed areas in this figure. (Access) network (Access) network We start here by describing several principles for data compression, and illustrating them with a few particular examples. Next, we analyse their performances in terms of (possible) theoretical limits and practical constraints. The remainder of this chapter is dedicated to a short, but comprehensive summary of the most relevant compression standards and technologies.
3.1 Source coding
By source coding we mean any data compression technique, without any restriction on linearity or reversibility, and assuming an ideal communication channel between encoder (compressor) and decoder (de-compressor). Consequently, data output by the de-compressor is either identical with the original, or the only differences are a result of the encoding process. In the case of so-called loss-less compression, there is no difference between the original data and the de-compressed data. In the case of "lossy" compression, data produced by a de-compressor may be different, to some extent, from the original ones. In this section we present some of the most effective principles for both loss-less and lossy compression, and try to give an estimate of the maximum compression that can be achieved for different types of data, in different contexts.
3.1.1 Lossless compression
The underlying assumption in loss-less compression is that parts of the original data are redundant. This means that either their representation or their values can be somehow reduced, without altering the quantity and quality of the information they carry. Entropy coding refer to a number of techniques that address the minimisation of data representation by assuming that each data value occurs as an independent random event, according to a certain probability function. Then the minimum average number of bits needed to represent those values equals the entropy of the random sequence (see A.1.1). Assume for instance that one has to encode the numbers resulted by casting a dice. Then one needs log2 6 = 2,58 bits to encode these numbers. Actually, in computing this value we implicitly assumed that
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each number appears with the same probability 1/6. In this case, the occurrence of any of the six numbers will eliminate the incertitude of one out of six possible events. In other words, each event (number) carries a quantity of information that depends on its probability. In our example, this is -log2 (1/6) = 2,58, as computed before. The entropy of the information source represented by the dice results as the average of the quantities of information carried by each source symbol (number). However, if one of dice's facets, say 3, faces up five times more often than the others, then this event is more predictable, and consequently number three will bring less information than any of the other five numbers. Indeed, the probability of a three is now 5/10, or 0,5, while the probability of any other number is 1/10. Then the information carried by the value three is log2 (0,5) = 1 bit. The information carried by any of the other five possible numbers will result higher in this case, namely -log2 (0,1) = 3,32 bits. However, the entropy of the said information source results as a weighted sum of the information carried by each symbol, that is (5*3,32 + 5*1)/10 = 2,16. Consequently, fewer bits are needed, on average, to encode the data generated by such a biased information source. Different algorithms have been developed for entropy coding. Although very simple, these algorithms are usually difficult to implement efficiently in practice, due to the requirement for an accurate probability function of data values. In the case of image and video compression, pixel values distribution may vary significantly from one image to another, making a certain entropy code rather inefficient. However, different combinations of other compression techniques with entropy coding led to good practical solutions, as shown in A.2.5. The above considerations indicate these techniques to be most suitable for compressing data having a quite constant statistical distribution, and requiring for perfect recovery of the original values. Usually, text files meet these requirements. Arithmetic coding achieves compression by allocating a certain real number between 0 and 1 to a whole sequence of original symbols. As in the case of entropy coding, the code words are a function of the symbols' probabilities (see A.1.3 for more details). The compression ratio for this technique depend directly on the length of the sequence of symbols encoded by one real number, and approaches to the limit the value given by Shannon's theorem (A.1.1). Other compression techniques stem from the assumption that there is a certain relation between data values, and therefore some of them may be reconstructed or predicted from the others. A trivial example is the image of this page. Considering it as a raster-scan image at a resolution of 600 dpi, we realise that many consecutive pixels have the same "white" value. It would be more efficient then to store or transmit such a sequence as one value representing "white" and another one representing the number of consecutive white pixels. So a few thousand values required for a blank line could be represented by only two values. This technique is known as run-length encoding. In general, complex data, like sound, natural images or video, can not be treated in this straightforward manner. However, similar principles can be derived from the auto-correlation function of their sample values (see A.1.4 for more details). Several compression techniques based on this property are known as predictive compression. These are explained in more detail in A.1.5. As explained in A.1.4, the maximum lossless compression of a set of samples is achieved when the samples get fully de-correlated. The techniques described above cannot, generally, reach full de-correlation for a finite set of sample values. Full de-correlation can be instead achieved by a linear transformation called the Karhunen-Love transform. By this transformation, the whole
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energy of the original samples is concentrated into a minimum number of transform coefficients. Of course, in order to achieve an effective compression, the values of these coefficients have to be truncated (or quantised), leading thus to a lossy compression (see A.1.6). Moreover, the optimality of this transformation can only be ensured if the statistics of the original data set is known in advance. This makes it unpractical for most real-life applications. However, other linear transformations (Fourier, cosine, sine, wavelet, Haar, Hadamard, Slant, etc.) approximate very well its behaviour, while having much simpler implementations.
3.1.2 Lossy compression
The underlying assumption in lossy compression is that parts of the original data are either perceptually, or statistically irrelevant. This means that one can drop these parts, without altering the perceived information in the de-compressed data, or its statistical properties. Therefore, a decompressed signal in this case will be an approximation of the original. The most trivial lossy compression is the digitisation, by which a continuous, indefinitely differentiable function representing a certain signal is converted to a discrete set of samples taking a finite number of values. These techniques rely on a number of psychophysical limitations of the human visual and aural system. Therefore, they are only suited for interpersonal audio/visual communications. Block truncation coding (BTC) is a statistical compression technique by which the values of the pixels within a rectangular window of arbitrary size are replaced by some coefficients derived from the average and variance of the original pixels (see A.2.1 for details). Transform coding starts with a certain linear transform (see A.1.6 and A.1.7), by which the total energy of the original data set (e.g. image) is concentrated into a reduced number of values, such that every other data point in the transformed space can be discarded. In reality, however, most transform coefficients actually have very small values, but still not zero, and a few of them will result in very large real values. Therefore, in order to improve compression, one has to quantise these values in order to obtain a finite representation of coefficient values, and (consequently) map to zero sufficiently small coefficients (see A.2.2). This way, some of the original information if irreversibly lost. To maximise compression and minimise losses, special quantisation schemes were proposed (and some of them standardised) for different input data types and formats (e.g. RGB or YUV images, mono or stereo sound, etc.). Vector quantisation (VQ) is a lossy data compression method based on the principle of block coding, i.e. it assigns one code-word for a group of data samples. See A.2.3 for further details. Fractal compression Fractal methods have been developed for both noisy and noise free coding methods. Images of natural scenes are likely candidates because of the fractal structure of the scene content [61]. In the mid-eighties Dr Michael Barnsley reported the use of "Iterated Function System" for image compression and synthesis. Using sets of affine transformations developed for a given image, and a principal result known as the "collage theorem", intra-frame (singular image) compressions in excess of 10,000:1 and inter-frame (sequence of images) compression in excess of 1,000,000:1 were reported. The collage theorem states that if an image can be covered (approximately) with compressed affine transformations of itself, then the image can be (approximately) reconstructed
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by computing the attractor (in the sense of non linear dynamic systems) of this set of affine transformations. By the end of 1988 Iterated Systems had developed the patented technique called the 'Fractal Transform' which has become the basis of their current product range. The development allowed a real world image to be reduced to a set of fractal equations based on the image being processed, rather than a huge library of pre-calculated, reference, fractal patterns. Image compression algorithms that are noise-free have been reported to be developed from this transform for real time automatic image compression at ratios between 10:1 and 100:1. Researchers at BellCore have developed a compression method that incorporates a Peano Scan with a fractal based coding schema for intra-frame compression, so making it possible to archive high picture quality with bit rates of less than one bit per pixel. A fractal-based method was reported earlier by Walach while the concepts of using a Peano Scan had been used for storage, compression and display. The BellCore researchers have combined these two fractal concepts into an efficient implementation, which is now being incorporated into a VLSI chip. Compression results show good quality imagery at a rate of 0.8-1 bit per pixel. The technique may be implemented in an adaptive fashion since a local estimate of fractal dimension provides an objective measure of image complexity. An alternative approach to image compression, also incorporating some features of fractal geometry, is based on orthogonal pyramid transforms. These transforms decompose the image as a weighted sum of basis functions. The basis functions have several attractive features including localisation in space and spatial frequency as well as self similarity. This latter feature reflects the notion that basis functions are scaled version of one another, and thus are able to capture image information at all scales in a uniform way. The pyramid transform techniques are based, in part, on an earlier concept of quadrature mirror filters. The pyramid structure results from dealing with the image at multiple resolutions. Results reported show high quality imagery at rates of 0.65 bits per pixel. The mean square error is comparable to 16X16 block DCT transform coding algorithms but superior in visual image quality. Perceptual compression is a generic technique based on the property of noise masking by the signal itself. It can be applied in audio, image, and video compression alike. Generally, compression artefacts (distortions) are not perceptible if their magnitude is lower than a certain relative value. This value, however, is not constant, but depends on its spatial, temporal, or spectral localisation [27]. For instance, if a low-level noise occurs simultaneously with a high amplitude audio tone, the human ear will not perceive it. Similar phenomena take place for images and video. The quality of a perceptually compressed signal is given by the just noticeable distortion (JND). Ideally, the noise level for every point in the signals space has to be equal to the threshold of JND (see A.2.4 for an example). This way, the compressed signal will have a perceptually perfect quality and the minimal size (or bit rate) for a particular lossy compression technique. Perceptual compression can be thus achieved by adapting the magnitude of compression artefacts to the threshold of JND. This process is referred to as noise shaping. Hybrid techniques. Each of the techniques described earlier performs very well with respect to some of the signals properties. None of them however is capable of totally removing signals redundancy and irrelevancies. Hybrid techniques combine some of these basic compression methods to achieve better performances. Some examples of hybrid compression (i.e. JPEG, MPEG) are described in A.2.5 and A.2.6.
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Adaptive compression is a further improvement of the hybrid techniques, where better compression is achieved by choosing the most suited compression algorithms for different components (parts) of the original signal. For instance, in the case of image compression one could first separate a given input image into different components (contours, textures, gradients, etc.) and then compress each of these separately. These techniques usually have a high computational complexity due to the required analysis stage. An extreme case of adaptive compression is based on principal component analysis. Actually, this is, to some extent, a generalisation of the Karhunen-Love transform, where the original data points are not the signal's samples, but whole sequences of samples. Therefore, to obtain good compression ratios, the data to be encoded must be highly specific. A common application of this technique is the compression of images of human faces. Here, a certain number of images, or regions thereof, are used as a "training set" to obtain a reduced set of parameters (the principal components). Linear combinations of these parameters can produce then optimal representations of any new image. Obviously, the optimality of the representations depends on how much the new data to be encoded differ from the training set. Of course, many other compression techniques are described in the scientific and technical literature. However, they do not differ fundamentally from the ones described above.
3.1.3 Pushing the limits
As shown in the previous two paragraphs, maximal compression of a data sequence is achieved when redundancy is completely removed. If that data sequence represents aural, visual or other kind of perceptual information, then eliminating its perceptually irrelevant parts can further reduce the amount of data. Apparently, this is the farthest one can go with compression. There are, however, at least two strategies for further minimising the amount of data needed in an interpersonal communication: 1. Minimise or even eliminate the information that is semantically irrelevant for the targeted user. 2. Transmit data models together with a few parameters, instead of data values. We will further refer to these strategies as to semantic redundancy reduction, and data synthesis (or regeneration). Although most of the techniques based on these strategies are still in the experimental phase, we shall treat them as possible near-future developments, or at least as possible ultimate limits in data compression9.
3.1.3.1 Semantic redundancy reduction
There are certain multimedia applications, particularly of the operative kind, where only a small part of the transferred information is significant (i.e. has a relatively high semantic value) for the end user. For instance, a news reporter who has to write an article about a football match may not need to watch again the whole video of the game, but could only use the "highlights". As another example, in a multiparty videoconference only the active participants at a certain instant require a full-motion video transmission, while for the others a slower refresh rate might suffice. More generally, when compression can not further reduce the bitrate, then reducing the amount of original data is the only option. This reduction can be easily achieved by e.g. sub-sampling. For a video, this means reducing the spatial or temporal resolution, pixel accuracy. However, such a
9
A more accurate term in this case would be data reduction.

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uniform sub-sampling can lead to an unacceptable decrease in quality. On the other hand, some images from the video may be more relevant than others (e.g. the highlights in a sports event). The techniques for reducing semantic redundancy attempt to identify the most salient parts of a certain data set to form a relevant summary that maximises the amount of useful information contained in a given amount of data.
3.1.3.2 Data re-synthesis and modality conversion
A way to save considerable amounts of bandwidth is by distinguishing objects in video and audio and manipulate these by modifying a parameter set in time, where each parameter describes some feature of an object. This is similar to the case in graphics, where each object is mathematically described and movement or changing characteristics of objects are modelled by a changing parameter. Scenes built up by real sources such as moving video, or by synthetic sources such as computer-generated graphics can be modified by deleting, adding, or repositioning objects, or to set behaviour of the objects for which the processing is done on the client side. Thus, visual objects in a scene can be described mathematically and can be placed in a two- or three-dimensional space. Similarly, audio objects are placed in a sound space. When placed in 3D space, the video or audio object need only be defined once and based on the aspect angle of the viewer, the objects are rendered. The processing for this is performed locally, at the user's terminal, which is advantageous for fast response and when the available bit-rate is limited, or when there is no return channel. It is possible to create a predefined face and let it express emotions by changing parameters that determine the shape of the face. Written text can be transformed in movement of lips of a face while speaking the text. An audio object could be a soundtrack. In an object-based approach, objects are modelled by means of parameters changing in time, which describe characteristics of the audio object. For playing such an audio object, only the values of parameters need be transmitted. Thus, a human voice could be generated by sending the parameters that define the voice characteristics of a certain sex, age, accent et cetera. Three-dimensional effects can be generated as well. Objects can be made scalable, which means that they can be rendered with a quality which is just sufficient for an application or with the maximum quality which can be delivered by the communication channel bandwidth. Scalable objects could have a basic-quality representation and representations for improved quality, such as black and white video versus colour video. It could be the case that parts of a scene are delivered only when enough bandwidth is available. Scaling allows for graceful degradation if little bandwidth is available, and for adapted video content. In some cases backgrounds can be left unchanged so that it needs virtually no bandwidth. A new view from a different angle can be calculated if the angle is known. The processing is done on the client side. Modality conversion attempts to reduce the amount of data by converting a media type into another. The typical example is speech to text conversion. Obviously textual information takes a lot less than speech to express the same idea. A telephony application making use of speech-totext and text-to-speech would only have to transmit over a network the textual information. Other techniques are being developed for converting facial expressions into face parameters, so video communications could only make use of just a reduced set of parameters.
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3.1.4
Theoretical limits of compression
The research in multimedia compression is still very active and new compression techniques are continuously developed. So a legitimate question would be: is there a limit? Unfortunately, such a question cannot be answered generally with yes or no, but particular answers can be given for particular contexts. However, the limits of lossless compression are somewhat clearer than the ones of lossy techniques. Therefore, for this latter category we will indicate different possible limits, according to the perceptual or semantic quality level required by different applications.
3.1.4.1 Limits of loss-less compression
Without any doubt, the limits of loss-less data compression have been set by Shannon in his theory on compression for noiseless channels: the minimum amount of binary information units (bits) needed to represent the symbols of a certain information source, equals the entropy of that source. For example, the entropy of the image below is 5.489. Since it is represented as an 8bpp bitmap, the achievable loss-less compression ratio in this case is 8/5.49 = 1.46. There are, however, some important remarks here. First, in the case of electronic images, either digital or analogue, an important part of the entropy is due to noise, since the values of additive noise decrease the auto-correlation between adjacent pixels. Consequently, a further increase of loss-less compression is achievable in practice by suppressing the noise (see the example in Figure 1).
Figure 1. Left: a grey-scale image with 8 bits per pixel that can be compressed, without any loss of information, to a minimum of about 5.5 bits per pixel (see the text). Right: the same image, after applying noise reduction, can be compressed to less than 5.2 bits per pixel.
With the second remark we emphasise that the entropy of the visual information in an image also depends on the way image samples are chosen. The conventional way is to consider every picture element (pixel) as one symbol generated by some information source. However, it could be more efficient to consider groups of "similar" pixels as being produced by different sources. For instance, the picture in Figure 1 could be seen as a combination of samples produced by three different sources, one for each of the three main "objects" in the scene: the snow, the skier, and the dark background. Since the pixels composing each object have similar values, one may expect lower entropy for each of the three objects. Generally, this is not true. Nevertheless, the average of the entropy values of the three objects is always less or equal to the entropy of the whole image (see Figure 2).
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Figure 2. Left: "snow" segment (144920 pixels, entropy 4.49 bits). Centre: "skier" segment (12139 pixels, entropy 6.91 bits). Right: "fence" segment (6151 pixels, entropy 5.35 bits). These yield an average entropy of 4.70 bits for the whole image.
The limits of lossless compression are given by Shannon's theory. Current technologies are able of reaching performances close to this limit. Therefore, no important improvements can be expected in this area
3.1.4.2 Limits of lossy compression
Currently, best performance (good quality at minimal bit rates) compression can be achieved with 1 bit per sample for speech, and 2 bits per sample for audio. There is no expectation for important improvements in the near future. A compression ratio increase with a factor of two for audio is not likely to occur sooner than five years from now. For video compression, the state-of-the-art technologies can achieve a compression of about 100:1, reducing thus the bitrate of near DVD quality videos (640x480 pixels, full motion) down to about 400 kbit/s. For a lower quality, compression factors of 175:1 were also reported. For near VHS quality (320x240 pixels, full motion) the currently achievable bitrate is 150 kbit/s. Reduced quality videos (192x144 pixels, 20 fps) can be currently transmitted over a 56 kbit/s connection. As in the case of audio, there are no spectacular improvements expected in the next few years. At least, there is no recent scientific development to promise important improvements in compression. Still there is room for improvements in the adaptation of the encoding process, the pre- and post- processing steps, so compression ratios could be increased by a factor of maximum two within five years. Still, for particular applications extreme compression ratios are achievable. Relevant in this respect is the case of videoconference based on face animation and text-to-speech conversion, where the whole information needed for video and audio would require a bandwidth of 300 bit/s/channel. In terms of compression ratios, this is about 1400 times more than state-of-the-art video compressors. Fractal compression, or, compression techniques based on prototype libraries, can also provide extreme compression ratio for domain-specific applications.
Pixel or tile based lossy compression technologies (e.g. MPEG) already achieve a high data compaction for a given distortion level. Sample based perceptual audio compression techniques also approach the maximal values of admissible compression artefacts. In conclusion, improvements in this field are only possible within narrow limits. Higher improvements in lossy compression can be achieved for particular applications and in particular contexts by using e.g. fractal compression, or large prototype libraries.
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Semantics preserving techniques, like summarisation or modality conversion could also be very effective, but again with limited applicability.
Beyond the limits?
3.1.5
At least two different sensational claims for extreme compression technologies were extensively presented during last few months. First of them, the so-called "Sloot Digital Coding System" was specially designed for compressing digital movies, and is supposed to have been capable of recording one complete, full resolution movie, on an eight kilo byte memory. For a 100-minute movie, at TV quality, this means a compression ratio of approximately 14.25 million to one. Note that current DVDs use the MPEG2 specifications to achieve a compression ratio of about 15 to 1, and the latest MPEG4 encoders can compress a movie about 100 to 175 times, but then with a considerable loss in quality. So, there is a legitimate interest in finding out whether these claims are credible, possible, and reproducible. Unfortunately, there is no scientific or technical information to support these claims, so we can only conclude for now that if a movie has been "compressed" and stored on an 8kB memory chip, then a large amount of the original information should have been available within the "de-compressor" (player). Indeed, further investigations confirmed that the 8kB memory was not used for storing the movie data, but rather a "recipe" to reconstruct an approximate version of the video frames. A huge library for image prototypes had to be used in combination with the 8 kilobytes of externally stored information. Even so, the quality of the movies played by this system was rather low. Another remark refers to the encoding process, which apparently had to be adapted to each particular movie. This suggests that the supposed prototype libraries were not generic (i.e. suited for encoding any movie). The other alleged breakthrough compression technology is known as I2BP. According to the original announcements, this technology allows streaming a full resolution DVD movie over a 2kbit/s connection. Taking into account the bit-rate of 156 Mbit/s for an uncompressed digital video stream (4:2:2, CCIR 601-2 compliant), the resulting compression ratio needed is approximately 80000. Again, the authors of this technology give no scientific or technical information. Moreover, the online demonstration of this technology, originally planned for April 2001, has been postponed to an unspecified date. Several demonstrations have been performed over a local network, in the presence of a few journalists and specialists, but with no possibility of controlling the parameters of transmission. Besides these, there were many other such claims over the years, not only for lossy compression, but also for lossless. A very popular "invention" in this respect is the so-called lossless recursive compression. Generally, such techniques claim to be able to compress any file, thus also files already compressed using those particular techniques. However, there is a straightforward demonstration of the impossibility of such a process (see A.1.2). The most recent claims for spectacular compression ratios, the "Sloot Digital Coding System" and I2BP cannot be confirmed. All available information about these claims indicates that these are most likely misinterpretations. There is no recursive compression technique
Compression standards
3.2
In our opinion, current and emerging standards, either open or proprietary, will steer further developments in compression technologies. Generally, new compression methods or technologies need at least five years to become a standard and to become deployed in the mainstream of
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applications and services. Therefore, we start this section with a brief description of the most prominent standards for audio, graphics, image, and video compression.
3.2.1 Image compression
The GIF standard (Graphics Interchange Format) is, together with the JPEG format, the de facto standard file format for graphic images on the World Wide Web and elsewhere on the Internet. Companies that make products that exploit the GIF format need to license its use from Unisys. There are two versions of the format, 87a and GIF89a. Version 89a (July 1989) allows for the possibility of an animated GIF, which is a short sequence of images within a single GIF file. A GIF89a can also be specified for interlaced GIF presentation. A patent-free replacement for the GIF, the Portable Network Graphics (PNG) format, has been developed by an Internet committee, and all recent releases of the major browsers support it. The PNG standard10 (Portable Network Graphics) is a file format for image compression that, in time, is expected to replace the patent protected GIF format that is widely used on today's Internet. The PNG format is undergoing standardisation by the ISO/IEC, is developed to be patent-free and provides a number of improvements over the GIF format. Like a GIF, a PNG file is compressed in lossless fashion. A PNG file is not intended to replace the JPEG format, which is based on lossy compression, but lets the creator make a trade-off between file size and image quality when the image is compressed. Typically, an image in a PNG file can be 10 to 30% more compressed than in a GIF format. Extra features of the PNG format includes controlling the degree of transparency, interlacing and tuning the image in terms of colour brightness. JPEG-LS is the latest ISO/ITU-T standard for lossless coding of colour still images. It has relatively low complexity and a compression performance that is near to the best achievable. The JPEG standard11 (Joint Photographic Experts Group) is one of the most popular coding standards for lossy compression of still images. It was developed for compressing full colour or grey scale images of natural scenes and thus, it works well on photographs and naturalistic work. It does not work well on line drawings, cartoons or text in black and white. The JPEG compression mechanism is based on exploiting known limitations of the human eye. JPEG allows for trading off file size against output image quality. Also decoding speed can be exchanged to image quality. JPEG can typically achieve compression by a factor of 10 to 20 of full colour data. A factor 30 to 50 is possible with moderate degradation. Motion JPEG (MJPEG) is a compression technique for videos, in which frames are compressed individually using JPEG. This leads to lesser encoding delays (compared with MPEG), because no differences between frames need to be computed. The disadvantage is that the format needs more bandwidth. However, the delay advantages are quite high, and therefore motion jpeg encoders are often used in high-quality video conferencing applications. GIF does better than JPEG on images with only a few distinct colours, such as line drawings and simple cartoons. For such images, GIF is lossless. The MPEG-4 VTC standard. VTC (Visual Texture Coding) is a texture and still image compression tool that has been standardised within the MPEG-4. VTC is aimed at compressing textures that are used in photo realistic three-dimensional rendering applications, in which the same texture can be visualised at a variety of different resolutions. A unique feature of MPEG-4
10 11
http://www.libpng.org/pub/png/ See also http://www.faqs.org/faqs/jpeg-faq/

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VTC is the capability to code arbitrarily shaped objects. Several objects can be encoded separately, possibly at different qualities, and then composed at the decoder to obtain the final decoded image. MPEG-4 VTC does not support lossless coding12. JPEG 2000 is the newest ISO/ITU-T standard for still image coding along with the standards JPEG-LS and MPEG-4 VTC which have been introduced recently and that start appearing in various applications. JPEG2000 is a similar texture compression method to MPEG-4 VTC, but it was standardised for another application domain, viz. for very high quality two-dimensional image compression applications while MPEG-4 was standardised with three-dimensional image features in mind. JPEG 2000 offers scalability and the possibility to encode images with so-called arbitrarily shaped Regions of Interest. The image compression standards compared In [46], the performances of JPEG-2000, JPEG-LS and PNG were assessed for lossless image compression. It turned out that in almost all cases of the test session, the best performance was obtained by JPEG-LS. JPEG 2000 provided, in most cases, competitive compression ratios with the added benefit of scalability. PNG performance was similar to the one of JPEG 2000. Typical compression ratios for lossless compression were in the range from 1.5 to 10. Also in [46], the performances of JPEG-2000, JPEG-LS and MPEG-4 VTC were assessed for lossy image compression. Here JPEG 2000 outperforms all other algorithms. JPEG provides, as expected for older technology, inferior results, showing a considerable quality difference at any given bitrate. MPEG-4 VTC provides results in between JPEG and JPEG 2000. In any case, the choice of a standard for a particular application or product will depend on its requirements. In the cases where lossy compression is of interest and low complexity is of high priority, JPEG still provides a good solution. JPEG-LS stands out as the best option when only lossless compression is of interest, providing the best compression efficiency at a low complexity. PNG is also of interest in such cases, although the complexity of the encoder is much higher than that of JPEG-LS. As for MPEG-4 VTC, it appears to be of limited interest, except when the ability to code arbitrarily shaped objects is required. JPEG 2000 provides the most flexible solution, combining good compression performance with a rich set of features.
3.2.2 The MPEG standard
MPEG is a working group in a subcommittee of ISO/IEC in charge of developing international standards for compression, decompression, processing, and coded representation of moving pictures, audio, and their combination. In particular, MPEG defines the syntax of low bit rate video and audio bit streams of synthetic and natural sources, descriptions of their structure and content, and standardises the operation of conformant decoders of these bit streams. The encoder algorithms are not defined by MPEG, which allows for continuous improvement of encoders and their adaptation to specific applications. Along with the video and audio coding, MPEG also defines means to multiplex several video, audio and information streams synchronously in one single bit stream, describes methods to test conformance of bit streams and decoders to the standard, and publishes technical reports containing software describing the decoder operation and software describing examples of encoder operation.
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See also the MPEG-4 SNHC FAQ: http://www.cselt.it/mpeg/faq/mp4-snh/mp4-snh.htm
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The MPEG working group works in phases denoted by numbers such as in MPEG-1 and MPEG2. These phases do not describe different versions of one standardisation effort, but are completely different coexisting standards that all handle different aspects of multimedia communication. Therefore, the later phases do not replace the earlier phases but complement them. Concerning MPEG-1 and MPEG-2, there is the notion of layers. Both in MPEG-1 and in MPEG-2 BC, three different layers are defined. These layers represent a family of coding algorithms. The layers are denoted by roman figures as Layer I, Layer II and Layer III. The notion of versions is used in the context of MPEG-4. Version 1 of MPEG-4 provides a set of tools for audio coding. With Version 2, new tools are added to provide additional functionality, while being fully backward compatible to Version 1. MPEG video compression is used in many current and emerging products. It is at the heart of digital television set-top boxes, DSS, HDTV decoders, DVD players, video conferencing, Internet video, and other applications. The MPEG-1 and MPEG-2 standards are similar in basic concepts: they both are based on motion compensated block-based transform coding techniques, while MPEG-4 deviates from these more traditional approaches in its usage of image construct descriptors for target bit rates in the very low range: below 64 kbit/s. MPEG-1 is an ISO/IEC standard for medium quality and medium bit rate video and audio compression. It allows video to be compressed by the ratios in the range of 50:1 to 100:1, depending on image sequence type and desired quality. The encoded data rate is targeted at 1.5Mbit/s for this is a reasonable transfer rate of a double-speed CD-ROM player (rate includes audio and video). VHS-quality playback is expected from this level of compression. MPEG-1 is designed to produce bit rates of 1.5Mbit/s or less, and is intended to be used with images of size 352x288 at 24-30 frames per second. This results in data rates of 55.7-69.6 Mbit/s [5]. MPEG-1 audio coding provides single-channel ('mono') and two-channel ('stereo' or 'dual mono') coding of digitised sound waves at 32, 44.1, and 48 kHz sampling rate. MPEG-1 audio standardises three different coding schemes for digitised sound waves called Layers I, II and III. The predefined bit rates range from 32 to 448 kbit/s for Layer I, from 32 to 384 kbit/s for Layer II, and from 32 to 320 kbit/s for Layer III13. MPEG-1 does not standardise the encoder, but rather the type of information that an encoder has to produce and write to an MPEG-1 conformant bit stream, as well as the way in which the decoder has to parse, decompress, and re-synthesise this information in order to regain the encoded sound. The encoded sound bit stream can be stored together with an encoded video bit stream and other data streams in a so-called MPEG-1 system stream. Within the professional and consumer market, four fields of typical applications of MPEG-1 audio can be identified: broadcasting, storage, multimedia, and telecommunication. This variety of applications is possible because of the wide range of bit rates and the numerous configurations allowed within the MPEG-1 Audio standard. Some of the most important applications are: Consumer Recording (DCC) Disc based storage (CD-i, CD-Video) DVD Disc based Editing, audio broadcasting station automation Solid State Storage Audio
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See also the MPEG Audio FAQ site: http://www.tnt.uni-hannover.de/project/mpeg/audio/faq/mpeg2.html

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Digital Audio Broadcasting (e.g. ADR, DAB, US-Digital Radio, Worldspace Radio) Internet Radio Computer based Multimedia Stand-alone electronic information systems
MPEG-1 audio addresses generic sound waves, i.e. it is not restricted to e.g. speech signals, but codes all types of sound signals. It performs perceptual audio coding rather than lossless coding. A perceptual audio codec aims at eliminating those parts of the sound signal that are not audible by the human ear (see also A.2.4). The different layers in MPEG-1 have been defined because they all have their merits. Basically, the complexity of the encoder and decoder, the encoder/decoder delay, and the coding efficiency increase when going from Layer I via Layer II to Layer III. Layer I has the lowest complexity and is specifically suitable for applications where the encoder complexity plays an important role. Layer II requires a more complex encoder and a slightly more complex decoder, and is directed towards 'one to many' applications, i.e. one encoder serves many decoders. Compared to Layer I, Layer II is able to remove more of the signal redundancy and to apply the psycho-acoustic threshold more efficiently. Layer III (also referred to as MP3) is again more complex and is directed towards lower bit rate applications. The three layers have been defined to be compatible in a hierarchical way, i.e. a Layer N decoder is able to decode bit streams encoded in Layer N and all layers below N. MPEG-1 video coding. The objective of the development of MPEG-1 was to encode VHS quality video together with associated audio of CD quality at a total bit rate of approximately 1.5 Mbit/s, to allow storage of movies on CD-ROMs. For this MPEG-1 was optimised to work at video resolutions of 352x240 pixels at 30 frames/sec (based on the NTSC standard used in America) or 352x288 pixels at 25 frames/sec (based on the PAL standard used in Europe). This is referred to as Source Input Format (SIF) video. MPEG-1 may in fact go as high as 4095x4095 at 60 frames/sec. MPEG-1 is defined for progressive frames only, and has no direct provision for interlaced video applications, such as in broadcast television applications. MPEG-2 is mainly used in digital television. It produces the video quality needed in HDTV. MPEG-2 audio supports the same applications as MPEG-1, extending the MPEG-1 audio capabilities to applications that require very low bitrates and to applications that require more than two channels (e.g. for professional sound or for multi-lingual channels). The MPEG-2 audio standard provides a backward compatible multi-channel extension to MPEG-1: up to 5 main channels plus a low frequency enhancement channel can be coded. The bitrate range is extended up to about 1 Mbit/s. It also extends MPEG-1 towards lower sampling rates: 16, 22.05, and 24 kHz, for bitrates ranging from 32 to 256 kbit/s (Layer I) and from 8 to 160 kbit/s (Layer II & Layer III). MP3-Pro [31] is an extension of the MPEG-1 Layer III standard aiming for improving compression ratio while maintaining the near CD quality of audio and the backward compatibility with MP3 format. To increase compression, MP3-Pro encodes only a low-bandwidth version of the original signal, the high frequencies being reproduced, through a predictive method, in the receiver. This method is referred to as Spectral Band Replication. To ensure an accurate reproduction of the high-frequency (10-20 kHz) components of the original audio signal, a lowbitrate set of parameters is transmitted together with the MP3 bit stream. This method can be very efficient for harmonic sounds, but it will not work for noise-like ones. The developers of MP3-
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Pro estimate an improvement of two times compared with MP3, while other experiments [35] suggest 15 to 30% improvements. MPEG-2 AAC (Advanced Audio Coding) is a new coding scheme, incompatible with MPEG-1. It provides a very high-quality audio coding standard for 1 to 48 channels at sampling rates of 8 to 96 kHz, with multi-channel, multilingual, and multi-programme capabilities. AAC works at bit rates from 8 kbit/s for a monophonic speech signal, up to over 160 kbit/s per channel for very high quality audio. AAC adheres to the same basic coding paradigm as MPEG-1/2 Layer-3, but adds new coding tools and improves on details. AAC is approximately 30% more bit rate efficient than MPEG-1 Layer 3. MP3 is the current choice for near-CD quality digital audio. However, AAC is its designated successor as it is able to provide the same sound quality with a larger compression rate. In addition it enables higher quality encoding and playback for high definition audio (at 96 kHz sampling rate). So AAC is the most promising candidate e.g. for new portable playback devices using solid state memory. The main areas of application of AAC are Internet Audio Audio for digital television and radio (both AM and FM radio successors) Portable playback devices MPEG-2 video coding addresses issues directly related to digital television broadcasting, such as the efficient coding of field-interlaced video and scalability. Also, the target bit-rate was raised to between 4 and 9 Mbit/s, resulting in potentially very high quality video, such as DVD quality [57]. Probably the most common MPEG-2 profile is 720 x 480 pixel resolution video at 30 frames/sec, at bit-rates up to 15 Mb/sec for NTSC video. Another MPEG-2 profile is the HDTV resolution of 1920x1080 pixels at 30 frame/sec, at a bit-rate of up to 80 Mbit/s. MPEG-414 was finalised in October 1998 and became an international standard in the first months of 1999. Application areas of MPEG-4 are digital television, interactive graphics applications, and interactive multimedia e.g. via the World Wide Web. MPEG-4 allows for interaction with content. It also brings multimedia to new networks, including those employing relatively low bit rate, and mobile ones. The flexibility of MPEG-4 video coding encourages many applications. MPEG seeks to avoid a multitude of proprietary, non-interoperable formats and players. Instead MPEG-4 provides standardised ways to represent units of aural, visual or audio-visual content of natural or synthetic origin, which can be either 2- or 3-dimensional, and to interact with the audio-visual scene generated at the receivers end. The bit rates supported by MPEG-4 Version 1 are typically between 5 kbit/s and 10 Mbit/s. MPEG-4s low-bit rate and error resilient coding allows for robust communication over limited rate wireless channels, useful for e.g. mobile videophones and space communication. There may also be roles in surveillance data compression since it is possible to have a very low or variable frame rate. It is likely that the standard will eventually support data-rates well beyond those of MPEG-2. A major application area is interactive web-based video. Software that provides live MPEG-4 video on a web page has already been demonstrated. There is much room for applications to make use of MPEG-4's object based characteristics. The binary and greyscale shape coding tools allow arbitrarily shaped video objects to be composed together with text and graphics. Doing so, a rich interactive experience for web-based presentations and advertising can be provided; this same
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scenario also applies to set-top box applications. Additionally, it is possible to make use of scalability tools to allow for a smooth control of user experience with terminal and data link capabilities. The games market is another area where the application of interactive MPEG-4 video shows much promise, with 3-D texture mapping of still images, live video, or extended prerecorded video sequences enhancing the player experience. Adding live video of users adds to the user experience multi-player 3-D games, as does use of arbitrarily shaped video, where transparency could be combined artistically with 3-D video texture mapping. MPEG-4 Audio coding15 provides: coding and composition of natural and synthetic audio objects, scalability of the bitrate of an audio bit stream, scalability of encoder or decoder complexity, Structured Audio: A universal language for score-driven sound synthesis, An interface for text-to-speech conversion systems (TTSI). MPEG-7 - Although not specifically designed for compression, the MPEG-7 standard allows one to implement advanced schemes for reducing semantic irrelevancies (see appendix A.2). For instance, under bandwidth-constrained circumstances, an MPEG-7 based application could retrieve, manage, and display only most important and relevant information, leading thus to a considerable reduction in data traffic.
3.3 Compression technologies
A number of proprietary video formats have been developed, such as QuickTime from Apple, AVI from Microsoft and RealVideo from RealNetworks. In this section, these will be explained together with their performances.
3.3.1 RealVideo from RealNetworks
RealNetworks16 companys business is in media delivery on the Internet. RealNetworks develops and markets software products and services designed to enable users of personal computers and other consumer electronic devices to send and receive audio, video and other multimedia services using the Web. The first RealPlayer was released in 1995, and to date over 200 million users have been registered. In the RealPlayer format, good quality video of PC screen size and average action is encoded at a bitrate between 500 and 1000 kbit/s. Lower quality video can be encoded at a bitrate as low as 20 kbit/s. CD quality audio is encoded at 64 kbit/s, while lower quality audio can reach 8 kbit/s17.
3.3.2 Quicktime
QuickTime 18 is software that allows PC users to play back audio and video on their computers. QuickTime is a suite of applications for playing back audio and video files, working with still images, flexible multimedia authoring, viewing media within a web page and delivering streaming media files on the Internet in real time.
15
See also http://www.tnt.uni-hannover.de/project/mpeg/audio/faq/mpeg4.html
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http://www.realnetworks.com
http://www.realnetworks.com/devzone/downlds/RV8_RA8_encode_settings.pdf 18 http://www.apple.com/quicktime
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3.3.3
Windows media player
Windows Media Player 7.119 is an integrated media player which supports a range of digital media activities, including playback of CD audio, streaming and downloading audio and video, media management, Internet radio. In download-and-play scenarios, Windows Media Video 8 can provide near-VHS quality video at bit rates as low as 250 Kbit/s and near-DVD quality video at rates as low as 500 Kbit/s.
3.3.4 DivX from DivXNetworks
DivX is a compression technology based on the MPEG-4 standard that can reduce an MPEG-2 video to ten percent of its original size according to its developer, DivXNetworks20. This means that it allows for compressing digital video such that full-screen DVD quality video can be effectively downloaded or streamed from the Internet over cable modems and DSL.
3.3.5 On2 Technologies
On2 Technologies21 have recently launched a series of video encoders based on MPEG4 specifications. Their performances are similar to the ones of DivX. According to product specifications, the achievable bitrate is about 400 kbit/s for near DVD quality video and less than 300 kbit/s for VHS video format.
3.3.6 Wavelet-based technologies
Wavelets refer to a way to represent pixel values of pictures based on a certain type of mathematical transformation. Some good moving image compression results were obtained by replacing the DCT operation in MPEG-1 by a wavelet transformation. Wavelets are likely to be the basis of the next generation of lossy image compression standards. DjVu is a new image compression technology developed since 1996 at AT&T Labs, and it was specially designed for the efficient storage and delivery of high resolution scanned documents and photographs. The DjVu format is progressive, so first an initial low-resolution version of the document is sent, and then the visual quality progressively improves as more bits arrive. Another distinctive feature is that background is separated from the foreground, allowing thus to apply different compression algorithms for the two kinds of images, to obtain a higher compression ratio. This way, the text can be kept at high resolution, while at the same time the backgrounds and pictures are compressed at lower resolution with a wavelet-based compression technique LeadTools JPEG2000 is a wavelet-based compression that stores the encoded images a hierarchical format. This means that there are several versions of the image, at different resolutions in the same file, without duplication. For that reason, a JPEG2000 compressed image can then be sent to a device in the resolution that best fits without additional storage overhead (adaptability). Moreover, there is another adaptability feature that allows the user selecting a region of interest to be displayed at full resolution, while keeping the rest of the image at a lower resolution, so the needed bandwidth, memory requirements, and processing power is minimised.
19 20
http://www.microsoft.com/windows/windowsmedia/EN/default.asp http://www.divx.com 21 http://www.on2.com
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3.3.7
Fractals
Traditional geometry with its straight lines and smooth surfaces does not resemble the geometry of objects in pictures reflecting reality, such as trees and clouds. Fractal geometry does. Fractals allow for the generation of artificial yet realistic looking scenes. However, the inverse problem of finding a fractal function which represents a given image remains unsolved. Fractal images do not become course grained when magnified. However, details that become visible then are not retained from reality, but generated by the fractal function. In fact, fractal image compression is an advanced form of interpolation useful for nicely looking graphics. It remains to be seen whether fractal compression will achieve breakthroughs in compression performance. Fractal compression ratios are typically between 20 and 60. The quality is not so good compared to wavelets or JPEG.
Conclusion
Although there is a lot of work on novel techniques like fractal-based compression or wavelet encoding, we firmly believe that Compression techniques are not likely to improve significantly in the next five years. An improvement of two to four times of current compression ratios for audio, video or text is an optimistic estimate. In certain application scenarios, however, with proper encoding of the environment model, much higher compression rates can be achieved. For instance, it is possible to create a predefined face and let it express emotions by changing parameters that determine the shape of the face. Also, written text can be transformed in movement of lips of a face while speaking the text. However, such applications need considerable amounts of context information at the destination, and processing power at both sender and destination in order to make this happen. There will not be any infrastructural means to assist in this realm to make this work for ad-hoc interactive sessions or web browsing.
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4 Multimedia distribution techniques
Novel solutions on several layers of the OSI reference model can improve the effectiveness of content distribution. These solutions are deployed inside the networks, typically at the edges, just before the last mile. They aim at optimising the resource usage of both the access network and the core networks. As we assume in this study an application application end-to-end IP network from client to server, this means decoder transcoder encoder that solutions must either be Multimedia Distribution Techniques provided inside the IP-layer, or using intermediate systems (Access) network (Access) network on top of the IP-layer (typically proxy-based). The IP-layer solutions are presented first in this chapter.
4.1 Network-level multicasting or broadcasting
Some applications are not point-to-point, but involve multiple senders and/or receivers. Primary examples of these are TV-broadcasts (one sender, multiple receivers), tele-conferences (multiple senders, multiple receivers) and tele voting (multiple senders, one receiver). We refer to broadcasting when everybody connected to the network is able to receive the data transmitted on the network. Multicasting is the term used when explicit subscription is necessary to receive (or send) data to others. In the traditional telephony environment, multicasting or broadcasting is not natively provided by the network22. Hence, only application-layer technology (see section 4.2) can provide multicast capabilities (typically using so-called conference bridges). Networks based on broadcast media (such as wireline or wireless Ethernet, DAB or DVB digital broadcasting, satellite networks) do provide broadcast services, but these are then confined to senders/receivers that are directly connected to that particular network. It is clear that native broadcasting greatly improves access-network performance when multiple users want to receive the same data at the same time. The Internet, a packet-based layer-3 network, also supports multicasting. For that purpose, sophisticated multicast-routing protocols have been developed that provide a many-to-many IPservice. Actual deployment of IP-multicast is not very high, although an increasing number of ISPs are providing multicast services to end-users23. This is notably due to lack of securitymechanisms, no standardised reliability protocol like TCP, and a limited multicast address space. It is likely that in the coming years standards will appear that provide secure and reliable communication for the single-sender, multiple receiver category of multicast applications. The security-service will enable access-controlled sessions for, e.g. tele-education and TV-broadcast sessions over IP-multicast, with encrypted content. This work is currently being done in the IETF, in the MSEC working group24. The reliability service would support transmission of applicationdata to receivers, e.g. for efficient replication of databases or web sites. This work is being
With the notable exception of SDH and ATM bearer connections, that support point-to-multipoint configurations 23 http://www.ietf.org/html.charters/mboned-charter.html 24 http://www.ietf.org/html.charters/msec-charter.html
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performed in the IETF-RMT working group25. Other categories of multicast sessions (e.g. manyto-one or many-to-many) will need proprietary application-layer solutions.
4.2 Distribution technology
When native broadcasting or multicasting is not available, or not sufficient, it is possible to use application-level techniques to actually combine or transcode traffic. Examples of such techniques are: Conference bridges. These devices accept multiple incoming voice or video connections, and have a single outgoing voice or video conference bridge connection that is multiplexed to all simultaneous members of the session. It is used com bine m ultiplex for multiparty conferences when only point-topoint connections are available. The bridge has functions like audio mixing, echo cancelling, video selection or video editing (e.g. only transmit the video of the person currently user 1 user 2 user n talking, or place each incoming video stream at a particular position in the outgoing video stream). These bridges are widely used for PSTN audio conferences, and H.320/H.323 audio/video conferences. Allocation of bridges is typically done by the end-user, by explicitly calling a conference bridge while presenting a previously established session-identification. Transcoding gateways. These are placed inside a stream; typically at the edges of networks, and translate audio or video traffic from a particular transcoder encoding into another encoding. This H.261 mpeg2 always happens at Voice-over-IP gateways, and at GSM/PSTN interfaces in which peer users often use different voice encoding. Increasingly, these are also deployed inside the Internet, to perform adaptation of content for, notably, low-bandwidth access networks. Typically, they are placed inside RTP streams. Allocation of such transcoding functions is done automatically, typically as a result of application-layer signalling (such as telephony call-setup). Transcoders are owned and controlled via standardised mechanisms (H.248 media control gateways) by parties that operate the signalling entities. It can be expected that in the future (2 to 10 years) such transcoders can be allocated by third parties, for instance when Grid-like secure resource allocation algorithms become available. This allows for more powerful multimedia peer-to-peer applications that can exploit logic "inside" the network. The use of transcoding gateways is quite interesting, in particular when combined with conference bridges, to allow for heterogeneous sessions between people with different bandwidth or codec availability. Multiplexers. Live or scheduled broadcasting over the Internet is difficult when IP-multicast is not available. To achieve scalability, so-called multiplexers can be exploited such that hierarchical distribution of content can be achieved, and server overload and backbone bandwidth explosion is prevented.
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http://www.ietf.org/html.charters/rmt-charter.html
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user
m u lt ip le x e r
user
user s t r e a m in g s e rve r m u lt ip le x e r user
user
Multiplexers are often deployed inside an enterprise, or inside the core networks of Internet service providers inside application-level proxies in combination with content-replication or caching (see section 4.3). In that case, they need to be explicitly configured in the clientapplication (using the proxy-settings of a browser or multimedia player), or. They can also be deployed by the content-owner. In that case, often a web page is displayed that lists several servers that can be used to access the content. The user then has to make a choice for the logically nearest server (which is not always easy to do). Of course, multiplexers are also used inside IPmulticast networks and SDH splitters, but that is another network layer. Above the IP-level the proxies can use available knowledge about the actual transported media-types, and content-level accounting mechanisms can be applied easily. Example product in this area is the Real Proxy server26.
4.3 Storage versus transport
The most optimal bandwidth saving technique is of course to not send the content at all. In other words, there is a trade-off between storage and transport for document-access applications or streaming applications; for conferencing applications the availability of local storage or prefetching of content is irrelevant. At one side of the spectrum are the set-top boxes that implement digital storage of TV-channels, such that delay TV is handled completely locally inside the TVset. Another example of this is the local web-cache that is managed by current web-browsers. Cache-hits will result in (almost) zero bandwidth usage. The other side of the spectrum is the thin-client approach, where no local storage exists and everything is downloaded from the server. Examples of these are streaming clients, TV-sets, distributed file systems and client-server databases. Of course, content can also be stored at intermediate locations between clients and servers, as shown in the figure below. This is called caching (when data is kept due to a particular request) or replication (when data is pro-actively put on servers at particular locations in the network). Caches are exploited by almost all ISPs, and most enterprises. Active replication is performed by content-providers themselves, or by emerging Content Distribution Network Service Providers like Akamai 27 and Digital Island28. Possible functionality of a CDN includes:
Redirection and delivery services to direct a request to the cache server that is the closest
and most available, e.g., using mechanisms to bypass congested areas of the Internet or technologies like IP-multicast. Distribution services to ensure that intermediaries (replica servers and/or caches) contain up-to-date content Content negotiation services that automatically take care of network and terminal capabilities as well as handle user preferences (this is still a research issue!)
26 27
http://www.real.com http://www.akamai.com 28 http://www.digitalisland.com

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Content adaptation services to perform all kinds of content-level modifications, such as

format conversion, virus checking, or to include (localised) advertisements. Management services to handle, e.g., accounting or digital rights management or services to monitor and report on content usage.
accounting connection 2 nd request data connection 1 st request data connection
cache
Internet
Access network
Client Origin Server Streaming Intermediary
It is interesting to note that some CDN-services actually use a completely different infrastructure to perform the distribution among intermediaries. There are a number of distribution networks that use satellite networks to actually replicate the content, and traditional networks (IP-based or CATV networks) to provide the distribution to end-users. An overview of the technology, current state of the art and ongoing research-subjects on IP-based CDNs can be found in [22]. Finally, there is the possibility of pre-storing content at the customers premises, e.g. on a secured DVD, for tele-education like purposes. Then the on-line connection is only necessary for access control and accounting (e.g. to obtain session-keys that allow one to decrypt DVD-content during a limited period of time), and for interactive communication with other students or teachers, but not for document download.
Conclusions
In the current topologies of access networks advanced distribution mechanisms can not change much the traffic patterns and loads in the local loop; primary optimisations are obtained inside the core networks. Caching and replication techniques become effective only in more complex topologies. Content negotiation and adaptation could make use of local caching and user profiles for minimising the amount of data in the access part of the network.
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5 Developments in access technologies
One of the last great obstacles to a wide scale deployment of the information society is the last mile issue: how are end-user premises connected to the global infrastructure. We use the term access networks for the network that delivers this type of connectivity. Access networks are interconnected to Core, or backbone, networks, that can much easily be scaled to higher bandwidths (because the cost of upgrades can be shared among all customers of all access networks that interconnect through that core network). Hence, the issues in the backbone are mainly economical, not technical. Furthermore, the core networks are currently based on fibre technology that can scale quite easily to gigabits of bandwidth using modulation techniques like DWDM. Hence, for application-deployment, a p p lic a tio n a p p lic a tio n the bottlenecks lie within d e co d e r tra n sc o d e r e n co d e r the access networks. M u ltim e d ia D istrib u tio n T e ch n iq u e s This is the main focus of this chapter.
(A c ce s s) n e tw o rk (A cc e ss ) n e tw o rk
A categorisation of access networks can be obtained from the way information is carried, via electromagnetic waves, from a transmitter to a receiver: guided (wired), or unguided (wireless). We will structure this section according to the physical media, as follows:
wired technologies DSL-technology (using the existing telephony infrastructure) Cable modems (using the existing CATV infrastructure) Power-line communication (using the electricity infrastructure) wireless technologies Wireless local loop solutions (radio and free space optics) Mobile networks (cellular radio) Hot-spot technology (radio-based: wireless LAN, bluetooth) Hybrid networks (e.g. CATV or satellite technology combined with DSL)
Fibre-to-the-home solutions like Gbit Ethernet are out of scope for this report. We will discuss these networks separately. For each network, we will indicate what type of physical infrastructure they need, what realistic expectations with respect to gross bandwidth and bandwidth per user are. It is important to distinguish between gross bandwidth and per-user bandwidth because in a number of technologies the available medium-capacity must be shared between end-users (e.g. in CATV networks, or in cellular radio networks). For such sharing of resources, an access network deploys a particular medium-access-control (MAC) solution. This MAC-functionality is part of the technology standard, and implemented in each of the interface boards. Different approaches of MAC-layers exist, and they are described in Appendix B.3. Also, in Appendix B: we discuss various channel coding methods and modulation techniques. Newly developed modulation techniques are described in the sections on wireless local loop and mobile networks. For existing wired infrastructures improvements on modulation and coding cannot be expected in the coming years. Possible improvements in capacity are mainly due to the usage of
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higher frequency rates (which may lead to different modulation techniques, but that is a secondary issue).
5.1 5.1.1 DSL networks Description
DSL stands for Digital Subscriber Loop. The subscriber loop is the physical connection between the end-user premises and the switching equipment of the service provider. The subscriber loop may carry analogue POTS traffic, digital ISDN switched traffic, or some kind of DSL (digital subscriber line) configuration. We focus here on the DSL technologies. A DSL connection consists of a copper-wire physical link together with a pair of modems at the two ends of the subscriber loop. A DSL with the mentioned modems creates a (virtual) digital line over the existing subscriber lines. The term xDSL refers to different variations of DSL, such as IDSL, HDSL, HDSL2, ADSL, SDSL, MSDSL, VDSL (see appendix C.2 for more details). The subscriber loop consists of a twisted pair of copper wires. Technically speaking, the subscriber loop connects customer premises equipment (CPE) to Central Offices (CO), as shown in Figure 3. The traffic on a wire pair is bi-directional and two so-called hybrids (that include, among others, the modems) are used in CPE and CO to accommodate two-way traffic on such a single pair of wires.
Hybrid Hybrid CPE

Figure 3: Analogue subscriber loop with twisted pair
CO
A subscriber loop is capable of carrying about 1.5 Mbit/s (for most of the current subscriber loops; distances less than 4 km). More specifically, the bit rate carrying capability of these wires depends on the length of the loop, the type of the wire used, and the presence of loading coils and existence of taps29on the wire. Figure 4 illustrates the capacity of subscriber loops in terms of loop distance (from [34]). Near End Crosstalk (NEXT; i.e. interference between different pairs of wires) is the main limiting factor for the capacity of subscriber loops.
29 Tap is basically a branch that has been left from an earlier use of the wire pair, which interferes with high frequency traffic
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9 8 7 6 5 4 3 2 1
.
Figure 4: Bit-rate capacity of subscriber loops [34].
This data rate versus reach is the major bone of contention when it comes to DSL in general. This trade-off is critical to DSL providers because it defines the coverage area for their DSL offering. As for symmetry, some DSLs provide highly asymmetric connections in order to optimise for specific applications. Table 14 compares DSL technologies based on bandwidth, range (length) and the symmetry of the traffic.
Table 14: Comparison of xDSL techniques, cf. [54].
DSL TYPE IDSL SDSL HDSL (two pairs) HDSL2 (one pair) ADSL G.lite ADSL VDSL
DOWNSTREAM (MBIT/S) 0.128 1.544 1.544 1.544 1.5 6 51.84 25.92 25.92 12.96 12.96
UPSTREAM (MBIT/S) Symmetric Symmetric Symmetric Symmetric 0.256 0.64 6.48 Symmetric 3.24 Symmetric 3.24
LOOP RANGE (KM) 5.5 3.05 3.66 3.66 5.5 3.66 0.305 (Short range) 0.914 (Medium range) 1.37 (Long range)
5.1.2
Deployment
ADSL and HDSL are currently rolled-out in The Netherlands, so there is no wide-scale deployment yet. VDSL experiments have taken place in the US.
5.1.3 Innovation
It is likely that higher-speed DSL technology will be rolled out. The VDSL equipment is not very stable yet, and only very few experiments have been done with broadband VDSL modems, but it is promising. A large advantage of DSL compared to cable modem technology is that individual 61
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subscribers may use different DSL technologies. Also, a multiple service-provider setting is much easier to obtain with DSL technology than with cable modem technology. DSL is often rolled out as last-mile option in fibre-to-the-curb networks. Costs of fibre-to-the-home are decreased by a significant factor when the last few hundred meters can be covered by existing infrastructure. A comparison between DSL and cable access networks can be found on page
5.2 5.2.1 CATV-based networks Description
CATV networks are hierarchically structured. Their deployment is based on efficient broadcasting of signals from a head-end to individual customers. The infrastructure is a hybrid fibre-coax network (HFC); that consists of fibre connections from the head-end to individual neighbourhoods, whereas the last mile consists of tapped COAX shared medium cables. Each shared cable serves about 100 to 2000 users. CATV networks have a gross capacity of 300-1000 MHz, which is subdivided into 6MHz channels. Each channel can be used for analogue TV, or for 16 digital TV channels (MPEG2 encoded channels; useful for interactive-TV like applications). Two-way HFC networks dedicate two channels for data-transport: one upstream and one downstream channel. The architecture is depicted in the diagram below
Figure 5. Cable Infrastructure
Deploying this infrastructure means that the cable companies must interface to some backbone IP network (or also to the PSTN if telephony services are offered on the CATV network), and that active components must be installed in the fibre nodes and the individual homes. The ITU standard for cable modems is DOCSIS. This standard specifies downstream traffic transfer rates between 27 and 36 Mbit/s over a radio frequency (RF) path in the 50 MHz to 750+ MHz range, and upstream traffic transfer rates between 320 Kbit/s and 10 Mbit/s over a RF path between 5 and 42 MHz. ETSI has standardised the DVB/DAVIC-based EuroModem specification, that has slightly different characteristics (8MHz downstream bandwidth, aggregate data rates of 42 Mbit/s downstream and 3 Mbit/s upstream). But because data over cable travels
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on a shared loop individual end-users will see transfer rates drop as more users gain access. DOCSIS and EuroModem are implemented in CATV-modems and HDTV-sets for web TVapplications.
5.2.2 Deployment
CATV networks are deployed on a large scale in The Netherlands. Migration towards bidirectional HFC networks is ongoing. Both Casema and Essent-kabel (major service providers in The Netherlands) are part of the EuroModem camp.
5.2.3 Innovation
There is not much innovation to be expected in this area. The main business of the cable companies is TV-broadcasting, with value-added services surrounding that. Large-scale investments in the physical infrastructure would be needed to upgrade the network. These investments are similar to those needed for optical networks, and because the capacity of optical networks is much higher, it is not likely that CATV-networks will provide higher quality services than the estimated shared 42 Mbit/s downstream. Other possible technical advancements include the reservation of more frequencies for cable modems. That would need a standardisation effort that is currently not on its way. For more details on CATV networks see Appendix C.1
5.2.4 Comparison between Cable and DSL technologies
While DSL-technologies may serve as a transition strategy, they suffer from a number of limitations in essentially they are trying to modify and adapt infrastructure that was designed for another purpose. Cable modems use an underlying infrastructure that was designed to deliver one way video to the home and DSL modems use an underlying infrastructure that was originally designed to deliver two-way voice to the home. Because of this, there are a number of technical challenges that remain unresolved at this point in terms of scalability and serviceability. For the cable modem the challenge is increasing the bandwidth and minimising the return channel noise on most cable systems. For xDSL deployments the challenge is extending the service reach beyond 5 kilometres and minimising the cross talk interference between adjacent copper pairs. The solutions for these problems involve an increasing amount of technical complexity within the modems themselves and at the servers in the central office or head end. As a result the costs are likely to remain high for the customer premises modem particularly compared to a relatively simple Ethernet interface that is now readily available for most home computers. The impact of crosstalk (near end crosstalk: NEXT, and far end crosstalk: FEXT) on the widespread deployment of xDSL on existing subscriber lines varies depending on the quality of the cable, the number of pairs and the bandwidth of the signal. Since few systems have been deployed, the extent of the problem is unknown. Before carriers can proceed with the wide scale deployment of ADSL, a large number of field measurements will be needed to assess the quality of the system. However, a recent study [4] indicates that only about 7% of the total number of households in The Netherlands can get access to an ADSL service. This conclusion is based on the announcement of KPN that by the end of the year 2001 only 70% of the households will be within the range of an ADSL-connection30, and on the conclusions of an earlier report of TNO
30
see http://www.mxstream.nl/support/nieuws_100000.html
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[55], according to which less than 10% of the cables within a bundle can be simultaneously used for a broadband (ADSL) connection. We do not fully support this conclusion, since several aspects were not taken into account (for instance the statistical distribution of sessions). Nevertheless, this 7% can be interpreted as a lower bound. Another fundamental limitation to cable and xDSL modem services is that customer channels are mixed together and the data rates are highly asymmetric. So while the wire speed of cable or xDSL modem service may be in the order of 1-10 Mbit/s the actual effective throughput may be considerably less and will vary depending on how many other simultaneous users are accessing the service. More importantly because of the highly asymmetrical nature of cable modem or xDSL service it will be difficult for a home, school or small business to be a data exporter rather than a data consumer. Cable and xDSL modems, in the short term may provide relative high-speed access to the home. In the long run, however, both technologies have fundamental limitations and will not be suitable for higher bandwidth applications and services, particularly those that require high data volumes originating from the home or school or involve real time interactive video. Advantages of cable are: cable is an established technology, cable companies are akin to utilities (they are in business for the long-run), installation is generally easy and quick, less expensive, good for home users. Disadvantages of cable are: cable companies are not at the forefront of innovation, utility-like service, cable is often not available in commercial buildings, cable become choked as more users sign on, need to build firewall. Advantages of DSL are: more reliability, some providers co-ordinate installation and turn-up, more technological innovation, versatile, good for business users, multiple services and multiple users. Disadvantages of DSL are: more expensive, installation can be tricky (contingent upon multiple service providers), innovations can be difficult to implement, and users face uncertainties due to competition among providers.
5.3 Power Line Communication (PLC)
31
PLC technology provides connection and data transfer over existing electricity networks, or socalled, "power lines". The electricity network is available in every room and in almost every house in the developed world. The ubiquity of the PLC infrastructure, therefore, is denser than that of todays existing telephony infrastructure. In terms of coverage area, data transmission solutions via power lines include in-house PLC and last-mile PLC. In-house PLC system exclusively operates within a building and uses in-house electrical cabling to carry the signals between the different PLC devices. In-house PLC is out of the scope of this document. Last mile PLC system, on the other hand, connects a backbone connection point, where the telephony and data traffic is fed in, with a connection point at the customers electrical feeding point. Generally, the low voltage transformer station is used as the backbone connection point and the low voltage distribution network is used as the connection to the customers building and premises.
Figure 6 shows a possible structure of a PLC network or the corresponding low-voltage supply
network. As we see, there are generally several network sections from the transformer to the users. Sections can have different structures. These structures (and, therefore, the structures of the
31
See also: http://www.plcforum.org/
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PLC networks) depend on several factors such as location of the PLC network (locating in urban/rural and residential/business areas), user density (number of the users, user concentration), network length and network design (number of sections) [8]. Generally speaking, there are about 5 sections in a PLC network, where around 50 to 80 users might be connected to each section. A PLC network, consisting of 250 to 400 users, may span over a length of 500 meters.
Base station
Figure 6: Above: PLC network structure, below: the shared access model of PLC.
Each section of a PLC network holds a bus structure, i.e., shared access medium configuration, in the upstream and downstream directions [8]. From this point of view, PLC network is similar to a CATV network. Unlike in CATV network, however, the physical medium of power lines is not isolated from outside world. Therefore, the frequency range allocated for PLC as a whole is limited from 1 to 30 MHz and there are constraints on the maximum transmission power. These measures minimise and limit the interference of PLC signals with legacy systems. On the other hand, PLC in a power line suffers from noise signals radiated from on line electrical appliances, distortions caused by natural and artificial sources, signal attenuation due to cable material and network elements, reflection by network elements. Table 15 compares the throughputs and distances that can be overcome in PLC systems with those of twisted-pair copper, optic fibre and wireless. This table presents figures that can be attained in the futures using these technologies. Note that the figures given for the throughput of power line network is the total throughput for a section of the network, which typically may have 50 to 80 users.
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Table 15: A rough comparison of channel throughput for four mediums, cf. [13].
Physical medium Typical technology Throughput today Throughput tomorrow Typical distance
Copper xDSL, modem, cable modem 9.6 Kbit/s to 8.0 Mbit/s 60 Mbit/s 300 m to 5km
Optic fibre FO-modem, MM/SMconverter 2.0 Mbit/s to 2.5 Gbit/s 6.4 Tbit/s 70 km
Wireless WLL-PMP satellite VLAN 9.6 Kbit/s to 2.0 Mbit/s 155 Mbit/s 3 km
PLC OFDM, QPSK, FSK, FH/DSCDMA 1.0 Mbit/s to 8.0 Mbit/s 100 Mbit/s 300 m
5.4
Wireless local loop
Wireless local loop (WLL) is a name for a number of technologies that use radio technology, typically in unlicensed bands, to provide wireless telephony-style connectivity. The advantages of WLL are [2]: Low installation and maintenance Costs. There is no need for local loop cabling to be laid and no physical connections to be made between the local loop switch and the subscribers premises. However, for high-speed technologies line-of-sight is a requirement. For lowerspeed connections near line of sight is necessary. Rapid deployment. WLL systems can be added quickly and easily with small efforts. Depending on the used technology, it is scalable from ISDN-speed (standard WLL delivers ISDN-like connections at 64 kbit/s) to 155 Mbit/s (fixed wireless point-to-point) Outside plant access and maintenance. Unlike a wired local loop, WLL lacks problems regarding access or right of way are overcome if a radio local loop is used. Unlike in wired local loops, the distance between the subscriber premises and the backbone access point does not affect the expense of WLL deployment (see Figure 7).
2000 1500
Cost [$]
1000 500 0
Distance
wireless wireline
Figure 7: Comparison between the operating costs of WLL and Wired local loop.
The market for WLL is growing. WLL services can be developed not only in developed areas (for competitive bypass) but also in undeveloped regions (as the primary POTS technology). Areas that are refused service by wired local loop providers because of deployment expense are also candidates for WLL technologies. By the year 2002, 339 million WLL lines are expected worldwide.
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5.4.1
Data-access: fixed wireless connections
In The Netherlands a number of companies already provide wireless local loops, e.g. 3TEL and Skybernet. They typically deploy equipment that conforms to standards set by the CISCOinitiated Broadband Wireless Internet Forum32. These wireless local loop solutions operate in the 38GHz range, in point-to-multipoint mode (bandwidth up to 44 Mbit/s) and point-to-point mode (bandwidth up to 155 Mbit/s) covering distances of up to 40 km. Technically speaking, this technology combines the functionality of a vector orthogonal frequency division multiplexing (VOFDM) physical layer (see appendix B.2) with the media access control (MAC) layer of the DOCSIS cable standard. The point-to-point mode is a cheaper alternative for, e.g., Free Space Optics (FSO) solutions (discussed below). For the point-to-multipoint configuration also a number of competitive approaches exist, that each use different physical layers or MAC-layers. Below we mention two of these, viz. LMDS and MMDS. Other technologies that are often used reside in the unlicensed 2.4GHz band, e.g. based on 802.11 technology (see also section 5.6; when using appropriate antennas, and placed optimally, they can cover distances of over 30 km). Fixed wireless connections are cheaper than digital leased lines, certainly for higher connection speeds. Local Multipoint Distribution System (LMDS) LMDS is a broadband wireless technology to deliver services to residential and commercial customers. It is expected that LMDS services will be a combination of voice, video and data [52]. LMDS operates in the range of 20-40GHz (depending on the country licensing). In this frequency range the coverage area is limited to a single site with a transmitter range of 8km (in metropolitan centres). Downstream signals, from the base station to the subscribers, are transmitted in a pointto-point or point-to-multipoint mode (broadcast). Upstream signals, from the subscribers to the base station, are point-to-point. Generally, in the downstream direction (from the base station to CPE) Time Division Multiplexed (TDM) is used in either a point-to-point mode or a point-tomultipoint mode. The upstream uses access technologies of TDMA33, FDMA and CDMA. The choice between these access technologies depends on the system operator business case, service strategy and target market (see [52] for a comparison). As a deployment example, in the US a total bandwidth of 1.3 MHz is allocated to LMDS in the 25 GHz and higher spectrum. In a 1GHz spectrum, the LMDS system provides 500MHz of useable spectrum per sector (with frequency reuse of 2). Assuming symmetrical upstream and downstream links, Table 16 summarises the total34 throughput for upstream (or downstream) direction.
32 33
http://www.bwif.org/ It seems advanced or generalised TDMA as described in [23]. 34 Considering the total traffic of all users.
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Table 16: Typical total (for all users) throughputs for upstream/downstream in a LMDS system (from [23]).
Modulation Spectral efficiency Upstream/downstream throughput for FDMA Upstream/downstream throughput for TDMA
4 QAM 1.5 bit/sec/Hz 375 Mbit/s 300 Mbit/s
16-QAM 3.5 bit/sec/Hz 875 Mbit/s 700 Mbit/s
64-QAM 5.5 bit/sec/Hz 1250 Mbit/s (only for short distances) Not used
Note that, compared to the FDMA method, the TDMA method yields a lower total throughput. However, TDMA makes efficient use of the capacity of the shared access link when users transmit data in burst. Typical access rates for individual customers lie below 34 Mbit/s. Products are currently offered by a number of companies (e.g. Ericsson) with ATM or IP connectivity, targeted at the SME market. Multichannel Multipoint Distribution System (MMDS) The MMDS frequencies have traditionally been used to provide a one-way analogue wireless cable TV broadcast service. Therefore, MMDS is also referred to as "wireless cable". MMDS can carry signals on a line of sight as far as 50 km from the transmitter. Traditionally data is returned upstream from customers through dial-up telephone modem connections. Nowadays MMDS providers are permitted to offer interactive two-way services for the Internet, making it a more legitimate local access platform for the delivery of high-quality data, video and voice services. Using cable modems and 64-QAM (64 point quadrature amplitude modulation), a single 6 MHz MMDS channel can deliver up to 27 Mbit/s of data throughput, or peak downstream rates of up to 1.5 Mbit/s to more than 1,000 simultaneous users35. A two-way service, however, reduces the effective range of MMDS to about 10 km. Other bands adjacent to the MMDS frequency of the spectrum such as MDS (Multipoint Distribution Service) and ITFS (Instructional Television Fixed Service) have frequently been aggregated together with MMDS. The frequency range allocated to MMDS is the 2.1 GHz to 2.7 GHz band36. Higher frequency bands are not chosen for MMDS due to higher free space or path attenuation. Note that bandwidth allocated to LMDS (the 27.5 to 29.5 GHz band in the US) has limited range of transmission (3 to 5 miles radius), therefore, LMDS is not a good choice to provide wide area coverage of digital television service.
Innovations in WLL technologies
The last decade has seen the emergence of many theoretical and practical techniques in the field of wireless telecommunications. These techniques exploit the spatial dimension in a more effective way than more diversity combining [43]. These new techniques can broadly be classified under four categories, namely: smart antenna technology, transmit diversity schemes, spatial multiplexing, and space-time coding.
35
See http://florin.stanford.edu/~t361/Fall2000/zyuan/main_page.html According to http://www.wcai.com/mmds.htm, MMDS networks are allocated only 200 MHz of spectrum (between 2.5 GHz to 2.7 GHz, for more information on the frequency ranges refer to http://www.cabledatacomnews.com/wireless/cmic10.html). For TV signals with 6 MHz bandwidth, merely 33 channels can be fit into the spectrum .
36
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Smart Antennas By using smart antennas, beams from the transmitter antenna can be focused to the receiver antenna. Such electronic focussing that takes place by using adaptive and phased-array antennas enables improvement in link margins, thus enabling longer reach with same power. Moreover, this method has the capability of significantly improving spectrum reuse because two focussed pencil beams may use the same channel and still could produce very little interference to each other. Smart antennas are key elements for providing such pencil beams for each user. Although the concept is extremely interesting and has the potential of significantly improving spectral efficiency (especially in fixed wireless applications), the work in this area is still in early stages. There are some commercial products that incorporate very early versions of smart antennas. Nevertheless, it will take a few years for products providing significant space diversity to emerge [28]. Transmit Diversity From the base station a signal is transmitted simultaneously from two antennas on the same carrier. By appropriate coding of the data streams put on the two antennas, the receiver can reduce the fade margin significantly. This method needs feedback from the mobile/portable to the base station to obtain the channel state information and optimise performance. Spatial Multiplexing Different data streams can spatially be multiplexed on the same carrier by using multiple antennas at a (fixed) subscriber terminal and at the base station. Here one transmits the streams through different antennas. Use of L antennas can give (nearly) an L-fold increase in data rate. This technique is also called MIMO (multi-input multi-output) processing. V-Blast [14] is one popular version of spatial multiplexing and it has been shown to yield spectral efficiencies as high as 30 bits/sec/Hz in short-range, fixed wireless application. Space time Coding Error-control coding, in conjunction with MIMO processing, can further increase range and/or bit-rate. This coding across spatial channels use ideas similar to Trellis Coded Modulation (TCM) to define mappings of symbols to antenna ports [51]. Some of these techniques are also being currently used to provide broadband wireless access to homes at rates up to 2Mbit/s.
5.4.2 Free space optics (FSO)
FSO is a technology for delivering (extremely) high data rates using optical signals propagated through free space. Hereby, the output of a semiconductor laser is focused on a receiver unit in a thin beam. Although FSO bandwidth capacity is small compared to optical fibre, it is large compared to other broadband and it allows vendors to avoid expensive fibre trenching costs altogether. Moreover, if the customer demands fibre, FSO can be used temporarily while fibre is being trenched. This allows a vendor to acquire a customer before they make an infrastructure investment. FSO can provide high bandwidth for both the business market and the residential market. It can be used in backbone (e.g., to connect different units of a large business) and in access points. FSO is a line-of-sight technology; i.e., the sender and receiver should be directly in the sight of each other. FSO links are point-to-point by nature. An FSO link refers to a full duplex communications
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link and, unlike MMDS and LMDS, it allows the same bandwidth in both directions. The FSO point-to-point links can be used to construct different network topologies including rings, meshes, and stars. FSO equipment operates in two frequency-bands of 780nm-900nm and 1500-1600nm. The frequency spectrum of FSO is unregulated by the FCC, being greater than 300 GHz. Therefore, there is no need for any spectral licensing. Furthermore because the footprint of the beam is so small, the interference that can arise in unlicensed radio pose no problem for FSO. Since it is a line-of-sight technology and does not generate the side lobes that one could expect from a directional microwave antenna, it significantly minimises the chance of interference and simplifies installations. Other advantages of FSO include ease of installation, low cost. The installation of an FSO point-to-point link takes about 3-4 hours. Though the channel characteristics are different in FSO as compared to fixed fibre, most of the electronic and electrooptical components are shared. FSO bandwidth is allocated per link, not per area like in LMDS, so bandwidth sharing is not required here. This gives service providers a high flexibility in organising the required links. Most likely providers would have some very high-speed links (622Mbit/s and above) from which they would feed off lower speed links to customers (20Mbit/s to 155Mbit/s). Another possibility is for providers to have the entire network running at high speed, and use different techniques to pull off bandwidth for each customer. The bandwidth allocations could be packet based, circuit based, or as FSO moves towards the WDM, wavelength based. Commercial FSO systems such as the LightPointe37 and GigaLink systems are available today, which allow speeds in excess of full duplex 1Gbit/s per link. Multi-wavelength laboratory tests have been performed showing systems with link speeds in excess of 160Gbit/s (full duplex). It can be expected that FSO capacity will always stay close behind that of fibre [33]. A typical FSO system will have a divergence angle in the range of 3-6 mRad, which is used to combat building sway and scintillation. This results in a cone diameter of 3.6 m at 600 m. The typical coverage range of FSO is less than 1 km. FSO links can be affected by a variety of atmospheric phenomena, such as fog and scintillation. This may affect the availability of the FSO system and can be combated with multi-beam systems, careful link placement and link budget analysis. In rare situations, other techniques may be used to improve availability such as re-routing the traffic to either a backup radio link or through other paths on the network.
Reflections on wireless local loop
Wireless local loop solutions (fixed wireless, broadband wireless, FSO) are areas where innovations happen very quickly. Rapid performance increases are possible to much higher bandwidths than can be obtained with DSL or cable infrastructures. However, deployment is quite complicated, due to interference of signals, landscape architecture, and frequency usage. Furthermore, the price of wireless local loop solutions is still a few factors higher than DSL or cable. For companies or individuals requiring higher bandwidths (and willing to pay for it), it is a good alternative for fibre-connections, as the installation cost and monthly cost is much lower compared to digital leased line connections (less than Kfl 700/month for 2Mbit/s bi-directional). Deployment of point-to-multipoint solutions in densely populated areas, however, is quite easy to realise.
37
http://www.lightpointe.com
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5.5
Cellular technologies
Although mobile networks are slightly out of scope for this study, we include them because 2.5G and 3G solutions will become available during the next 5 years, and will deliver access speeds of at least 100 kbit/s. Currently a migration is on its way from GSM (9.6 kbit/s) to 2.5G standards GPRS (theoretical maximum 112 kbit/s, in reality about 30-60Kbit/s), HSCSD and, much later, Edge. The real revolution here is GPRS, which is a packet-switched network. HSCSCD basically is a software solution that allows one to bundle 4 GSM channels as one single connection. UMTS is the third generation mobile networking technology. There are three main carriertechnologies that are capable of offering UMTS requirements, namely: wideband CDMA, CDMA2000 and SD-WCDMA (in China). All UMTS technologies offer significant increases to traffic capacity per carrier when compared to GPRS. Cell ranges in UMTS are also different due to technology and frequency levels. The number of base stations needed is about six times larger that for a GSM900 network.
220 200 180 160 140 120 100 80 60 40 20 0 2001 2002 2003 2004 2005
GSM HSCSD GPRS EDGE UMTS
UMTS, which has promised to deliver up to 2 Mbit/s, will initially provide an average bandwidth range of 40 kbit/s, which might increase to 200 kbit/s by 2006. The expectation is that the average data rate to grow further as the technology develops. The figure above depicts real world data speeds, which predicts that opportunities for realistically sized streaming audio and video media will not become realistic before 2006 and 2008 respectively. However, video and audio targeted at small screens like those of cell-phones, wristwatches or PDAs will be delivered already in 2002, based on GPRS. This might be sufficient for some kind of conferencing scenario, but certainly not for entertainment or effective working at home.
5.6 Hot-spot technologies and home networks
Home area technologies provide wireless technologies to indoor spaces (within a building or apartment office) and public hotspots. As a major class of these technologies, a wireless LAN (WLAN) is a flexible data communication system implemented as an extension to, or as an alternative for, a wired LAN within a building or campus. WLANs transmit and receive data over the air and enhance user mobility. WLANs have gained strong popularity in a number of vertical
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markets (e.g., the health-care, retail, manufacturing, warehousing, and academic arenas) due to productivity gains by using hand-held terminals and notebook computers to transmit real-time information to centralized hosts for processing [54]. Another emerging technology in this field is BlueTooth. We will discuss these two solutions in this section. Alternatives for public access and hotspots might be the point-to-multipoint WLL solutions discussed in section 5.4, although the necessary modem equipment is not available yet as plug-in PC-cards.
Wireless LAN (802.11)
Two important configurations can be used in the implementation of WLANs: an ad-hoc networking (independent WLANs) approach and an infrastructure based approach [6]. The simplest WLAN configuration is an ad-hoc or independent (or peer-to-peer) WLAN that connects a set of PCs with wireless adapters. Any time two or more wireless adapters are within range of each other, they can set up an independent network. The corresponding stations dynamically configure themselves to set up such a temporary network. An infrastructure-based WLAN has a centralized controller for each cell, often referred to as access point. Access points link WLANs to the wired network and allow users to efficiently share network resources. The access points not only provide communication with the wired network but also mediate wireless network traffic in the immediate neighborhood. Multiple access points can provide wireless coverage for an entire building or campus.
Server
Access point
Figure 8. Configuration of a WLAN providing multiple access points.
Technology Manufacturers of wireless LANs have a range of technologies to choose from, each technology comes with its own set of advantages and limitations. Most wireless LAN systems use spreadspectrum technology, which is designed to trade off bandwidth efficiency for reliability, integrity, and security. In other words, more bandwidth is consumed than in the case of narrow-band transmission, but the tradeoff produces a signal that is, in effect, louder and thus easier to detect, provided that the receiver knows the parameters of the spread-spectrum signal being broadcast. If
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a receiver is not tuned to the right frequency, a spread-spectrum signal looks like background noise. There are two types of spread spectrum radio: frequency hopping and direct sequence. Frequency-hopping spread-spectrum (FHSS) uses a narrow-band carrier that changes frequency in a pattern known to both transmitter and receiver. Properly synchronized, the net effect is to maintain a single logical channel. To an unintended receiver, FHSS appears to be short-duration impulse noise. Direct-sequence spread-spectrum (DSSS) generates a redundant bit pattern for each bit to be transmitted. This bit pattern is called a chip (or chipping code). For longer chip-sizes the probability that the original data can be recovered is higher (at the cost of using more bandwidth). Even if one or more bits in the chip are damaged during transmission, statistical techniques embedded in the radio can recover the original data without the need for retransmission. To an unintended receiver, DSSS appears as low-power wideband noise and is rejected (ignored) by most narrowband receivers. Current WLAN standards
Table 17 gives a synthetic overview of some WLAN standards specifications. More details
regarding these standards are presented in Appendix C.3. According to [39], two of these WLAN technologies are expected to be dominant over the next five years: Wi-Fi (and its 802.11g extensions), and 802.11a. The latter includes most concepts originally defined in HiperLAN/2.
Table 17: Overview of WLAN standards
Standard IEEE802.11b Fi) (WI-
Available spectrum 83.5 MHz at 2.5 GHz band 83.5 MHz at 2.5 GHz band 83.5 MHz at 2.5 GHz band 300 MHz at 5 GHz band
Data rate 11 Mbit/s 24 Mbit/s 22 Mbit/s 54 Mbit/s
Throughput 5-7 Mbit/s 10-11 Mbit/s 10-11 Mbit/s 31 Mbit/s
Range & data rate (a fair comparison) 100m at 11 Mbit/s 100m at 12 Mbit/s 100m at 11 Mbit/s 50m at 9 Mbit/s (15m at 36 Mbit/s) (10m at 54 Mbit/s)
IEEE802.11gOFDM IEEE802.11gPBCC IEEE802.11a
Bluetooth
Bluetooth38 wireless technology is a de facto standard, as well as a specification for small-form factor, low-cost, short-range radio links between mobile PCs, mobile phones and other portable devices. It is one of the main technologies for so-called personal area networks. It uses the unlicensed 2.5GHz band, offering aggregate bandwidth of up to 800 Kbit/s per micro-cell. It is currently rolled out as replacement for serial or infrared connections between devices. However, a next generation Bluetooth devices is planned with increased bandwidth, which is also positioned for technology that serve hotspots inside airports or railway stations. However, this technology is out of scope for residential access to individual households or small enterprises.
38
http://www.bluetooth.com
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Concluding remarks
Home area networks tend to grow to public areas, possibly as high-speed access inside a UMTS network. They have an advantage over WLL solutions, because the CPE-interface cards are well standardised, small, low power and inexpensive. Hence, these networks can be deployed in public areas (railway stations, libraries, and public governmental offices) to give people quick access to multimedia content using devices that they carry with them. However, for in-house access in rural areas, these technologies are not useful due to the inability to cover long distances. For these application areas WLL solutions (including certain 802.11 solutions) are more beneficial.
5.7 Hybrid networks
Hybrid networks are the result of complementary, interoperable integration of terrestrial, satellite and wireless networks [3]. Considering the diversity of telecommunication technologies, many different configurations can be considered for hybrid networks. In the following subsections some of the most important ones are described.
5.7.1 Multiple communication channels
Sometimes there are two or more physical mediums between two points for communication (parallel communication channels). It is also possible for an end-point to be connected to one or more core networks (e.g., Internet, telephone networks) via different access networks (a star connectivity). Note that in the latter case, the egress points of the access networks to the corresponding core networks are not necessarily the same. Communicating parties in this setting can benefit from the presence of these extra channels. For example, in the case of parallel communication channels, extra channel(s) can improve the availability of the connection between two end-points. An example technology of such kind is briefly described in what follows. The configuration of multiple connections to an end-point provides a higher aggregate bandwidth to and from the end-point. This is basically a case of combining some access networks, which are individually explained in previous sections and, therefore, we are not going to further elaborate on this. Note that dividing data into parallel streams and feeding them via different ingress points (to the corresponding access networks) is a matter of data management.
Hybrid FSO/microwave communication networks
As mentioned in section 5.4, FSO communication systems are capable of providing high-speed local loop connectivity in metropolitan areas. These laser-beam-based connections, however, are subject to attenuation due to molecular absorption, molecular scattering, aerosol absorption and aerosol scattering, which are mainly caused by water drops in fog, rain, snow etc. Typical availability figures of the FSO system are given in [9] for different cities in the USA. These figures vary in the range of 97.4 and 99.9 percent. Such availability figures are not acceptable for many applications where having a near always connectivity is very crucial. For example, voice/data applications require a 99.999 percent statistical availability [9]. To achieve carrier class availability figures using FSO over longer distances and at higher speed, different methods can be applied such as scaling down the distance between the communication locations, increasing the launch power, increasing the receiver optics diameter, or using a more preferable wavelength range. In many cases these methods are neither feasible nor economical.
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An FSO system in conjunction with a slower speed microwave backup link, designed by LightPointe Communications Inc., can improve the statistical availability of free space laser systems. With respect to the weather impact, optical and microwave technologies are complementary. While the optical communication path suffers signal degradation and attenuation from small size particles such as fog, RF/microwave systems in frequencies below 10GHz are far less impacted by these kinds of weather conditions. The basic assumption in this approach is that losing a connection completely is certainly much worse than communicating at slower speed for a short period of time. The current hybrid network architecture provides a high-speed full duplex 100Mbit/s Ethernet primary connection. The backup system is an 11Mbit/s wireless Ethernet bridge operating in the unlicensed 2.4GHz band. The hybrid network architecture provides an automatic switchover facility in the event of a malfunction. Switching is performed automatically without user intervention and within a timeframe on the order of 100 milliseconds. The system automatically resumes its regular high-speed operation when the failure disappears. Due to the lower bandwidth of microwave systems, this approach is certainly not applicable to any kind of transmission scenario. However, for commercial applications involving networking speeds on the order of 100-155Mbit/s, this approach might very well be a viable solution to boost the availability figure to 99.999%.
5.7.2 Complementary h ybrid networks
Nowadays communication networks are based on heterogeneous technologies. This means that a message from its source to its destination passes through networks with different infrastructures, e.g., cellular wireless, optical network, copper wires etc. In a general sense, these networks complement each other to pave the way for a message. In this subsection we narrow our scope and explain the heterogeneous access networks where the incoming and outgoing traffic to an end-point for a given application passes through different types of access networks. Note that this differs from the star configuration mentioned in the previous subsection, where the access networks differ if we consider different applications. In other words, in a star configuration the incoming and outgoing access networks to an end-point are the same for a given application.
Hybrid terrestrial/satellite communication networks
This category refers to the system that uses terrestrial networks to carry out some functions and a satellite network to carry out others. In this way, the resulting hybrid system takes advantage of the properties of each network to provide the whole service [16]. This approach is motivated by the explosive expansion of the Internet, where access to the Internet is either too slow or too expensive (e.g. in remote areas). It is however possible to exploit the following three remarks [17] to improve this situation: Satellites are able to offer high-bandwidth connections to a large geographic area; a receive-only VSAT is cheaper to manufacture and easier to install than one which can also transmit; and computer users, especially those in a home environment, typically wish to consume more data from an external network than they generate (asymmetric traffic). These facts indicate the solution of breaking a users TCP/IP connection into two physical channels: a conventional terrestrial dial-up link for carrying data from the user to the Internet, and a higher-speed one-way satellite link for delivering data from the Internet to the user. These return terrestrial links are used to forward users requests, usage data, and other information. In this case, terrestrial telephone lines are a lower cost solution than providing a satellite return link from each subscriber location. Although access to the Internet in future broadband satellite
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systems will not be exclusively asymmetric, such access is likely going to be asymmetric for most individual users (for reasons more economic than technical). As an example, DirectPC, designed by Hughes Network Systems, provides asymmetric access to the Internet. Using IP-tunnelling techniques, this system supports asymmetric bandwidthintensive Internet applications such as browsing the World Wide Web. With this system, a downlink (from satellite to the user) bitrate of 400 kbit/s can be provided. According to [16], more elaborate use of hybrid systems will occur in mobile satellite systems, where mobile terminals with dual-mode capability will be provided. These terminals can be used with terrestrial cellular systems (within the range of such systems) or operate through satellite. Registration, mobility management, and advanced intelligent network features all may involve terrestrial network mobile switches working in conjunction with satellite network switches.
Hybrid terrestrial/MMDS communication networks
A key issue facing fixed wireless technologies, such as MMDS, has been the lack of two-way capabilities. Similar to delivering wired cable data services, digital data such as Internet content are modulated onto radio frequency channels for broadcast transmission to roof-top antennas at subscriber locations. Coaxial cable is run from the antenna to a down-converter (which shifts the microwave signal frequency into the cable television band) and then into the cable modem inside the customer premises. The cable modem demodulates the incoming high-speed data signal and passes it on to an individual PC or local area network (LAN). Wireless operators offering Internet access have typically used a telephone-return path for upstream communication. Reliance on a telephone return path limits upstream transmission speeds for end-users. Furthermore, it adds costs to the wireless service provider for incoming telephone lines and dial-up modem pools. Therefore, in the U.S. the FCC is approving the use of MDS, MMDS and ITFS spectrum for upstream and downstream broadcasts. Upgrading wireless cable systems to support two-way transmission is technically challenging, as it requires operators to convert broadcast television systems into networks that more closely resemble a cellular telecommunications platform.
Concluding remarks
Hybrid networks are special in its kind. They are used either to increase capacity or reliability. In this section we described some configurations that are offered by service providers, notably to small and medium-size enterprises. However, it is questionable whether these integrated services are really useful for individual households. For individual households, an alternative hybrid network can be created by means of a residential gateway concept. This gateway, a zero-management device deployed by end-users, could interface to any available digital network (based on cable, power network or local loop infrastructure), and on the user premises side to an in-home infrastructure. Such a concept does not yet exist, but it may be the way to go, as it can implement a lot of distribution functionality (cf. section 4) and make end-users a lot more independent of the individual service providers.
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5.8 5.8.1
New developments in access technologies Future cellular technologies
Enhanced 3G Upgrades for W-CDMA systems that might support a theoretical speed of up to 8 Mbit/s downstream, called enhanced 3G, are the first step. Vendors who plan such upgrades have not disclosed technical details of how they plan to boost the data rate, but it is expected that these solutions will focus on more efficient packet coding. We expect a number of interesting investment opportunities to arise in the field of sophisticated technology solutions for enhanced 3G, as each of these will need to utilise the existing bandwidth more efficiently. Towards 4G The next step in the development towards 4G technologies will be separation of the up and downlink technologies. AT&T has already conducted feasibility tests for a system with an EDGE uplink and wideband OFDM (Orthogonal Frequency Division Multiplexing) downlink. Other possible configurations combine GPRS or UMTS for the uplink and WLAN technology for the downlink. Digital broadcasting systems such as DAB & DBV are also becoming more relevant as down link alternatives. Among the modulation technologies that hold the most potential for 4G applications is OFDM, initially developed by Flarion technologies. Its potential bandwidth was estimated to be as high as 100 Mbit/s for the aggregated capacity of one cell. Nortel Networks is aiming to provide 20 Mbit/s per user peak rates on multi-carrier OFDM. Some companies go as far as to quote theoretical peak rates of 54 Mbit/s (Wi-LAN Ltd. and HyperLAN Standards). Other modulation technologies such as UWB are also sometimes referred to as 4G. Only recently have 4G technology developments commenced. Many of these set out to achieve the performance that 3G originally intended to provide. Several large telecom companies including Ericsson, NTT DoCoMo and Lucent have committed extensive resources to develop 4G systems. They are focussing on creating a convergent network that offers seamless IP connectivity over several radio interfaces (WLAN, OFDM, W-CDMA) and wireline networks with very high data rates.
5.8.2 Bi-directional satellite links
Until recently, satellite-based access could only provide a downlink broadband connection. For the return channel, a separate ISP connection was needed to send data to the Internet, typically over an analogue modem. Recently has been announced the launch of the first consumer two-way satellite Internet service, called DirectPC Satellite Return39. This provides a bi-directional satellite connection, so Internet users no longer need to maintain a separate analogue modem connection. For now, services based on this device are only operational in limited areas. At this time, the service is really just being deployed. Another high-speed satellite Internet service is called StarBand40, and works with the DISH satellite network. StarBand claims download speeds of up to 500 kbit/s and minimum speeds of 150 kbit/s, and upload speeds up to 50 kbit/s in a two-way satellite connection.
39 40
http://www.direcpc.com/index2.html www.starband.com
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5.8.3
T-spectrum technology
Claimed to be the decade's most important trend [11], T-spectrum 41 technology, also referred to as wireless optics, represents the convergence of radio and fibre. These networks could provide a throughput of up to 100Gbit/sec by operating in a largely unused frequency spectrum between radio and light waves. Several companies are developing systems based on T-rays. The closest so far are Harmonix42, which uses 60GHz microwaves, and Endwave43, which started to develop transceivers at 94GHz, 140GHz, and 220GHz. The latter technologies will enable 100Gbit/sec services within four years. However, regulatory issues might delay the large deployment of these technologies. Governments have not yet decided how T-rays should be used, or whether they should be licensed at all. Moreover, the deployment of such access technology requires designing and developing special wireless techniques in order to accommodate such high rate data streams into an optical backbone.
41 42
Named after the frequencies used, ranging up to 1THz. http://www.hxi.com 43 http://www.endwave.com
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6 Wrap up
6.1
Bandwidth requirements for different categories of applications
The success of some communication services was mainly influenced by the degree to which they satisfied a business need. Examples: telephony, voice/e-mail, videoconference, file transfer, hypertext web. Consumer-oriented applications/services usually enter the mainstream when prices are below a certain value. Examples: TV, video. As availability and affordability of Internet increase, new usage patterns arise. Nowadays, uploading and peer distribution of multimedia seems to become more and more popular ("People are more willing to pay for sending than for receiving video"). This, however, could just be hype, so we do not have for now any reason to believe that local traffic will change its asymmetric character. Quality of multimedia may differ very much from one application to another. Business applications tend to require less perceptual quality, but higher reliability. Here, the "message" matters most. Entertainment applications will not by widely embraced, unless they offer a high perceptual quality and a certain (minimal) reliability. Examples:
a videoconference is acceptable for a CIF image format (360x240 pixels), or QCIF (180x120
pixels), at 10 (5) frames per second, 16 bit stereo sound, and latency of no more than 100ms. This results in a bandwidth requirement of less than 128 kbit/s (using H.263 standard). a video on demand application is barely acceptable at a VHS video resolution and CD quality for sound. Using state-of-the-art compression tools, this still requires a bandwidth of about 200 kbit/s. These remarks were taken into account for the evaluations in the next table, using the bandwidth figures from Table 22. Category Application Minimum bandwidth (Mbit/s) 0.05 0.08 0.2 20 0.4
44
Maximum bandwidth (Mbit/s) 0.2 0.2 8 340 8 1.5 0.25 2 20 6
A/V streaming
Time-shifted radio stations MPEG Audio streaming MPEG Video streaming
A/V Multicast Conferencing Interactive rich documents access and file exchange
HDTV MPEG video TV Video conferencing Internet telephony Tele-medicine Telecommuting

44
0.128 0.064 0.5 0.5 0.014
e-learning, e-government
The categories in this table are on-demand streaming applications, broadcasting applications, real-time conferencing applications, and asynchronous document access. Note that some applications may include features from more than one category (e.g. tele-medicine may include a
44 The bandwidth is given for each individual channel. The total bandwidth results, roughly, by multiplying this value with the number of participants.
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synchronous video-conferencing component). In that case, the bandwidth figures need to be added.
6.2 Compression
Variants in bandwidth requirements are due to smaller images, or better compression techniques. This report investigates progress in compression, as innovations in that area can lead to deployment of broadband applications over middle band or low bandwidth connections. Compression can be achieved either by:
optimising data representation (or minimising its redundancy), eliminating irrelevancies, or finding an approximation (or a parametric description) of original data.
The first is referred to as lossless compression, since it does not alter the original information. The last two methods lead to lossy compressions. The limits of lossless compression are given by Shannon's theorem of coding for noiseless channels. General, a compression factor of 2:1 is typical for natural images, when each pixel is seen as an independent data sample. An optimal representation of a whole image can lead to a compression ratio of 5:1. The limits of lossy compression may vary within a very wide range, depending on the purpose of communications. As long as the communication has to keep the perceptual quality of the original information, the maximum admissible loss of information is given by the threshold of "just noticeable distortion". Much higher reduction factors can be achieved when only the semantic quality of the message has to be kept. Example for videoconference: Using vectorisation and modality conversion (speech-driven face animation, speech-to-text and text-to-speech conversion), the whole information to be transmitted is one text (data) stream at a bitrate as low as 300 bit/s. However, this requires all parties to have stored face and voice parameters of all others, and their terminals to have enough memory and state-of-the-art computing power. Many spectacular compression techniques were reported over the years, both for lossy and lossless methods. All of these were proven sooner or later as frauds. For one of the most recent one, claimed by I2BP, we can only say that there is no evidence that compression factors of 80 000 to 1 have been achieved for video. In our opinion this is not possible now and will not be possible in the future using a pixel-based approach, as they suggested to have done. Theoretically, however, by using e.g. fractal methods it is possible to attain such compression ratios. Still, such a method wouldn't be practical, since it would require the adaptation of the encoding process to the specific visual content of each image in a video sequence.
6.3 Channel coding techniques
Improvements in channel coding are another means of improving bandwidth on existing infrastructures. However, it appears that this is not really possible, because channel capacity is generally a function of signal's power, its spectral distribution, and the parameters of additive or multiplicative noise affecting the channel. Hence, notably frequency-ranges and power consumption influence the bandwidth, and coding techniques are selected in the most appropriate way.
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Current coding, modulation, and multiplexing techniques have reached performances approaching the optimum for a certain channel type, signal power, bandwidth, etc. therefore only marginal improvements are expected in the field of channel coding for copper-wire and cellular wireless access networks. More significant technical improvements are expected in modulation and multiple access layers (MAC) for wireless channels, but these are still in the research phase. A good example in this respect is related to space-time processing (see Appendix B: ).
6.4 Multimedia distribution mechanisms
In the current topologies of access networks advanced distribution mechanisms will not change much the traffic patterns and loads in the local loop. In more complex local loop topologies (involving residential servers or local peer to peer links) they could be used for optimising the capacity in the access networks. Along the same idea, caching and replication techniques become effective only in more complex topologies. Theoretically, existent copper wire infrastructures (ADSL, CATV) can be adapted to allow for local peer-to-peer traffic. Then, residential servers could make use of both advanced distribution and caching mechanisms to optimise the local peer-to-peer traffic. Even then, however, only a small part of the applications could take advantage of these optimisation techniques. Applications based on ephemeral data, like conferences, do not benefit from these techniques. Content negotiation and adaptation could make use of local caching and user profiles for minimising the amount of data in the access part of the network. This, however, has a limited applicability in e.g. filtering multicast data. This way, undesired, inappropriate, or irrelevant data is detected prior to transmission by matching user profile parameters against the set of descriptors attached to each multicast programme.
6.5 Access technologies
Cable networks still have, at least theoretically, room for bandwidth improvements. In Netherlands, most cable operators allocate a number of 35 to 40 channels for analogue TV. Using digital set-top boxes, the same number of channels would fit into the bandwidth required for 8 to 10 channels (64 to 80 MHz), thus increasing the bandwidth available for data traffic. At the same time, the number of users per network branch can be decreased, so the statistical availability of bandwidth per user could increase. However, the costs involved by these improvements may be a barrier for such a development, legacy TV sets cannot cope with digital channels, and new modems would need to be developed. ADSL and its higher performance successors are also well suited for almost every application from the categories considered here. However, in Netherlands the availability of ADSL is for now very limited, and near future developments are mainly directed to densely populated areas, such that it will not be widely accessible (more than 90% of the population) for years. On the other hand, the offered level of performance is way below the technical limits. For instance, MXStream ADSL services offer a downstream/upstream bandwidth of 512/64 kbit/s for the basic version, and 1024/256 kbit/s for the extended version. If we also consider the protocols overhead, the achievable throughput is about 384 kbit/s for the downstream link of the basic version. Fibre-tothe-curb solutions will enable the use of VDSL, which will offer real broadband access to individual houses. That is a realistic deployment scenario for a lot of environments, as the rollout of fibre is manageable.
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Wireless access technologies (e.g. LMDS, MMDS) are as well suited for residential broadband services. Since most of them are relatively cheap and easy to deploy, they could compete with, or supplement ADSL and CATV. Nevertheless, the short range may be an obstacle for some of them to be used as residential access networks.
6.6 Suitability of access technologies for the given applications
Since the expected improvements in bandwidth saving technologies are in the limits of 2-4 for compression, and only in the range of fractions for channel coding and distribution techniques, the landscape of access networks suitability will remain about the same within five years. The most important short-term developments are expected in improving the performances of access technologies.
Table 18 gives an estimate of the extent to which different access technologies can accommodate
the bandwidth requirements of different applications. We took into account only the requirements for single users and single applications. In fact, more relevant would be the total traffic expected for a residential connection. Since there is no way to estimate it, one could elaborate realistic usage scenarios. For residential users, the most representative scenarios could be for a home entertainment environment, and for a small business. The bandwidth requirements can be obtained from Table 19 to Table 22, which list bandwidth figures for certain media types using preferred compression techniques.
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Table 18. Broadband applications or services that can be supported by some of the currently available or emerging access technologies, for a single user
Access Technique
Achievable Bandwidth (down/upstream) [Mbit/s] 5/0.64 51/6.5
Applications (selected by DGTP)

A/V stream. + (MPEG) + A/V mcast. + (MPEG) + Conf. i.r.d.a.
Remarks 45
ADSL VDSL
+/+
+ +
Assuming a range of 3.66 km - Assuming a range of 300 m - not yet available Assuming 200 subscribers per network segment. Assuming 50 users per section The bandwidth expected for 2006 Assuming 16-QAM, TDMA, and 200 users - another version providing 54 Mbit/s will be available within a few years
CATV
0.2/0.015
+ (MPEG)
+/(Audio) - (Video)
+/-
PLC UMTS
0.15symmetric 0.2symmetric 2.8symmetric 11symmetric
+/+/-
+/+/-
LMDS
+ (MPEG) + (MPEG)
IEEE802.11b
FSO satellite Near Tspectrum T-spectrum
155symmetric 0.5- 0.05 622symmetric 100000symmetric
+ + (MPEG) + +
+ + (MPEG) + +
+ + +
+ +/+ +
- Not yet widely available - over a range of 400 m To be deployed within four years
45
The figures in this column are either actual, or realistic estimations of what is possible to achieve
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Table 19. Suitability of access technologies for raw data audio/video streaming
Source video format

Source audio format


160 120 9.5 2.5
Table 20. Suitability of access technologies for MPEG-4 compressed data audio/video streaming
Source video format

Source audio format


1.5 0.3 0.1 0.05
+/+ +
+ +
+ + + +
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Table 21. Suitability of access technologies for raw data video conferencing
Source video format

Source audio format

Number of parties

4 4 4 2 4 4 4
465/155 117/39 32/10.9 10.9/10.9 31/10.3 11/3.7 2.79/0.9
+/-
+/+/+
Table 22. Suitability of access technologies for compressed data video conferencing, assuming a compression approaching MPEG-4 performance
Source video format

Source audio format

Number of parties

4 4 4 2 4 4 4
3/1 1.1/0.36 0.56/0.2 0.2/0.2 0.54/0.18 0.16/0.06 0.05/0.03
+/+ +
+/+
+/+ + + + + +
+ + + + + + +
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References
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Appendix A: Some theory on compression
A.1
A.1.1
Lossless compression techniques

Shannon's theory of coding for noiseless channels
Assume the samples of a certain signal as values, ranging from 0 to L-1, of an independent random variable. If each value occurs with a probability pi, i=0,1,,L-1, then the average information carried by each sample is given by the value of the entropy, defined by
H = - pi log 2 pi
i =0
L -1
According to Shannon [48], these values can be encoded using H + A bits/sample, where A is an arbitrary positive quantity. Therefore, the maximum compression ratio that can be achieved in this case is given by
C=
B , H+ A
where B is the original number of bits used for representing samples' values.
A.1.2 The counting argument
It is important to emphasise here that the only achievable compression is due to non-uniform statistical distribution of data values. Therefore, no compression algorithm will be able to compress any data set, without losing information. A very simple proof of this statement is the socalled counting argument, which starts by assuming that a certain compression technique will be able to compress an arbitrary data set of N bits by at least one bit, resulting thus a data set of length N-1. With N bits it is possible to form 2N different combinations, while only 2N-1 different values can be obtained for the compressed data set. Therefore, some of the original data sets can not be represented by any of the compressed versions, so some loss of information must have occurred in the compression process.
A.1.3 Pixel based compression techniques
Huffman and arithmetic compressors are two of the most representative techniques from this category. These both use the same approach for minimising data representation by allocating different number of bits for symbols occurring with different frequencies. So, the larger the probability of a certain symbol, the more compact will be its representation. Assuming an certain information source that produces the symbols s1, s2, s3, s4, with the probabilities P(s1)=P1, P(s2)=P2, P(s3)=P3, P(s4)=P4, the first step is to arrange these probability in a decreasing order. Without loss of generality, we may assume P1 > P2 > P3 > P4. To compute the Huffman code words for these four symbols, one has to construct a binary tree in the following way:
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1. 2.
Assign the first two nodes of the tree to the lowest two probabilities. Build a new node as the parent of the previous two, and label it with the sum of their probabilities. Re-arrange the resulting probabilities in decreasing order. Repeat the steps 1 to 3 until all probabilities are assigned to a node.
3.
For our example, the tree could look like the one in Figure 9. Then, the left and right branches can be arbitrarily assigned the binary values 0 and 1. The code words for each symbol are then obtained simply by parsing the tree, starting from the root. In our example, the code word for the most probable symbol is "10", while for the least probable it is "1111"
1 1 1 P6 1 P5 0 P1 0 P2 0 P3 P4 1
P5=P3+P4 P6=P2+P5
Figure 9. The construction of Huffman code words
The arithmetic compressor attempts to encode messages, rather than symbols, so a single code word, which in this case is a positive real number less than one, will be assigned to a whole group of symbols. Assume the messages we want to encode consist of three symbols. Then for each source symbol we allocate a range of real numbers equal to the probability of each symbol, as depicted in Figure 10 for a particular case, where P1=0.4, P2=0.3, P3=0.2, P4=0.1. Encoding one symbol means restricting the whole range from 0 to 1, to the sub-range allocated to the current symbol. A subsequent symbol would restrict further, and so on. To illustrate this process, consider a particular message, say "s2 s3 s1". When symbol s2 arrives, the whole range from 0 to 1 is restricted to (0.4, 0.7). Symbol s3 will further restrict the previously available interval to (0.61, 0.67). The third symbol from the message, s1, will take from this interval only the first 40%, that is the range (0.61, 0.634). Any number from this interval (e.g. 0.62) can now uniquely identify the input sequence.
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s2 0.7 0.67 0.61 0.664 0.652
s3 0.67
s1 0.634
s4 s3
1 0.9 0.7
s2 0.4 s1 0.52 0.634
0.4
0.61
0.61
Figure 10. Arithmetic coding
A.1.4
Redundancy and auto-correlation
The auto-correlation function of a one-dimensional data sequence representing the consecutive samples of a physical signal has a graph similar to the one in Figure 11.
r (t) xx
Figure 11. Auto-correlation function of a physical signal
Two extreme cases are presented in Figure 12. The first graph (a) shows the auto-correlation of a constant function, while (b) shows the auto-correlation of white noise (see annex 1 for details). It is obvious that the first sequence can be completely represented by only one of its samples, while the other is "incompressible". Physical (real) signals can never reach either of these extremes. So, the objective of compression can be formulated now as reducing the region of support for the auto-correlation function. This process is referred to as de-correlation.
rcc(t)
a)
0 rzz(t) t
b)
0 t
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Figure 12. Auto-correlation function of: a) the constant function, and b) the white noise
Although not explicit, there is an intrinsic relation between the redundancy of a signal, and its auto-correlation function defined as below (similar properties occur in the discrete domain):
1 r xx (t ) = lim x( t) x(t - t) d t , T 2 T -T
where x(t) is the original signal. A fully redundant signal, that is a constant value, will have a constant auto-correlation function:
rcc (t ) = lim
1 T 2T
-T
c(t)c(t - t)dt = lim 2T c

T
2T = c 2 .
At the opposite end, the white noise with a uniform spectral distribution, will be fully decorrelated, and thus have an auto-correlation function of type Dirac delta. This results, using the Wiener-Hincin theorems, from the inverse Fourier transform of the spectral density function:
-1
rnn ( t ) =
{ S nn ( n )} =
nn
( n )e j 2pnt dn .
Since the spectral density function is constant for the white noise,
, t = 0 rnn ( t ) = @ ( t ) = . 0, t 0
In Figure 13, the auto-correlation functions for a TV signal, along different axes, are presented. Note that m, n, and p represent the distance, in number of pixels, measured from the current pixel. For a PAL standard signal, the temporal distance between two consecutive points will be 74 ns for the first graph, 64 ms for the second, and 40 ms for the third.
R yy (m) 1 R y y (n) 1 R y y (p) 1
0,5
0,5
0,5
4 8
12
20
40 n 0
10
15
a b c Figure 13. Auto-correlation functions for a TV (luminance) signal: a) between consecutive pixels; b) between lines; c) between frames.
A.1.5
Predictive techniques
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The simplest predictive technique is the differential pulse code modulation (DPCM), where the value of a certain sample at location n can be expressed as a linear combination of the previous k samples, as shown in the following equation:
~ u (n) = j[u(n - k ), u(n - k + 1),..., u(n - 1)],

where
~ u (n) is the estimated value, and u(n-k) are the previously actual values. Therefore, the ~ A ( n ) = u( n ) - u ( n ).
only information that has to be encoded and transmitted is the prediction error
Since this has usually a smaller range, the average length of the code words will be shorter. An example of a commonly used linear predictor for certain image compression schemes is the linear two-dimensional predictor:
~ u ( m , n ) = a( k ,l )u( m - k , n - l ) )
( k ,l )W
To ensure the best46 prediction, coefficients a(k,l) are computed as solutions of a certain optimisation function. More complex and efficient predictive techniques can be derived from the above principles following different strategies: Applying some image transformations to achieve a higher correlation between adjacent transform coefficients Modifying the sampling scheme (or the ordering of the original samples) to increase their auto-correlation (e.g. Peano-Hilbert scans) Computing different prediction coefficients for different regions of the image to be compressed. This assumes that the image should be first segmented into a number of "homogeneous" (i.e. highly correlated) areas
A.1.6
The Karhunen-Loeve transform
The main idea in transform coding is to concentrate a large part of the signal's energy into a low number of transformed elements (coefficients). Although theoretically doesn't assume loss of information, in practice is mostly used in lossy compression techniques. For a one-dimensional data set represented as a vector u, having an auto-covariance matrix R, the KL transform is defined as by the matrix F, such that a de-correlated vector v can be obtained by using the relation v = F*Tu. To achieve a complete de-correlation of the elements of v, it is necessary and sufficient for matrix F to satisfy the relation
46 With respect to some property, e.g. the magnitude, average, variance, or other momentum of the prediction error.
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F*T R F = L, where L is the diagonal matrix of the eigenvalues of the auto-covariance matrix R.
A.1.7 The Discrete Cosine Transform (DCT)
Since KL transform is very difficult to implement, it is considered preferable to use this transform to achieve approximately similar performances. The formula below defines the forward DCT for a certain two-dimensional data set u(i, j):
N -1 N -1 F 2m + 1i F 2n + 1 j vm , n = am an u i , j cos cos 2 N 2N i =0 j = 0
Statistically, when u(i, j) are the pixel values of a natural image, v(m, n) will result in large values for small m and n, and decrease rapidly as m and n approach the upper limit. This property will allow one to use these transform values for achieving a higher compression.
A.2
A.2.1
Lossy compression techniques

Block truncation coding
This is a very simple example of lossy compression to illustrate the effects of approximation in the representation of images. The algorithm is performed in the following steps: The MN input image is segmented into rectangular blocks of 44 pixels. The pixels belonging to block k are denoted uk(i,j), i,j=0,...,3, k=0,...,[M/4][N/4]-1. For each block the pixels' average is computed:
mk =
1 3 3 u (i, j) . 16 i= 0 j= 0 k
Each pixel is assigned a value 1 if its magnitude is higher than the block's average, or 0 otherwise. This way, a binary mask is generated. For each of the two groups obtained before a new average is computed, say mk1, mk2. Denote q the number of pixels greater than the block's average. Then it is easy to show that
m k 2 = m k - sk m k1 = m k + sk
q 16 - q , 16 - q q
where sk is the standard deviation of pixels from block k,
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1 3 3 sk = | u k ( i, j) - m k |2 . 16 i = 0 j= 0
This way, for each 4X4 block will be enough to store the 16 element binary mask, and the quantised values for block's average, and its standard deviation, resulting thus a compression ratio of 4 to 1.
Figure 14. Example of BTC: original 8 bit per pixel image (left), image compressed at 2 bit per pixel (centre), and the difference image (right).
A.2.2
Quantisation of transform coefficients
After performing a linear transform, a straightforward way to achieve compression is to quantise the transform coefficients, such that a finite representation is achieved for large valued coefficients, and zero for the rest. As an example, Figure 15 shows the losses introduced by quantising and discarding DCT coefficients. In this case, a compression ratio of about 3:1 can be easily achieved without heavy alterations of the original image.
Figure 15. Images reconstructed from a reduced (27% left, 35% right) number of quantised DCT coefficients.
A.2.3
Vector quantisation
A vector quantiser can be defined mathematically as a transform operator T from a K-dimensional Euclidean space RK to a finite subset X in RK made up of N vectors. This subset X becomes the vector code-book, or, more generally, the code-book. The main problem in designing a vector quantiser is the choice of the set of vectors. The distortion produced by the transformation T is generally computed as the mean square error (MSE) between the "real" vector x in RK and the corresponding vector x' = T(x) in X. Therefore, the vectors in the code-book should be such as to minimise the Euclidean distance d between any point in the original data space and the closest vector in the code-book (code-word). Consequently, the set of code words will induce a partition on the original data set, such that each code word will be a centroid of one of the resulting subsets.
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Figure 16. Example of VQ of some two-dimensional data47. Asterisks show the positions of code-words, and lines delimit the Voronoi regions.
An optimum scalar quantiser was proposed by Lloyd and Max. Later on, Linde, Buzo and Gray resumed and generalised this method, extending it to the case of a vector quantiser. The algorithm (called LBG) is performed by iterating the following steps: subdivide the training set into N groups (called partitions or Voronoi regions), which are associated with the N code-words, according to a minimum distance criterion; the centroids of the Voronoi regions become the updated code-book vectors; compute the average distortion: if the percent reduction in the distortion (as compared with the previous step) is below a certain threshold, then STOP.
Once the code-book has been designed, the coding process simply consists in the application of the T operator to the vectors of the original image. In practice, each group of n pixels will be coded as an address in the vector code-book, that is, as a number from 1 to N. The LBG algorithm for the design of a vector code-book always reaches a local minimum for the distortion function, but often this solution is not the optimal one. A careful analysis of the LBG algorithm's behaviour allows one to detect two critical points: the choice of the starting codebook and the uniformity of the Voronoi regions' dimensions. For this reason some algorithms have been designed that give better performances. With respect to the initialisation of the LBG algorithm, for instance, one can observe that a random choice of the starting code-book requires a large number of iterations before reaching an acceptable amount of distortion. Moreover, if the starting point leads to a local minimum solution, the relative stopping criterion prevents further optimisation steps.
A.2.4 Perceptual coding
Example of noise masking for audio signals (cf. [41]):
47
http://www.data-compression.com/vq.html
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A.2.5
Hybrid techniques - the JPEG algorithm
Without giving all the details regarding the generation of a JPEG image file, we will describe the main stages of JPEG compression. In principle, this is nothing more than a packaging of some of the techniques already presented. A simplified block diagram of the JPEG encoder is given in Figure 17.
Quantisation tables Y U V 8x8 pixel block splitting
8x8 block DCT
Quantiser
Coefficient packaging
Code words tables
Buffer (FIFO)
Huffman encoder
Encoded bit stream

Figure 17. Simplified block diagram of a JPEG encoder
The JPEG encoding process operates independently on rectangular image blocks of 8x8 pixels. First, a DCT is applied to each block to so only a few of he resulting 64 coefficients will have relatively large values. The quantiser truncates the real values of the coefficients, according to
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their respective ranks. For this, a number of predefined quantisation tables are used to specify the accuracy
Figure 18. Compression artefacts in JPEG compressed images for 20:1 (left) and 7:1 (right) compression ratios.
A.2.6
MPEG-1 video compression
Figure 19 depicts a simplified view of the main processing stages in an MPEG-1 encoder. The A/D
converter produces the raw video data for the three input video signals: one luminance (Y) and two colour difference signals (U and V). Temporal redundancy reduction performs the so-called inter-frame coding by eliminating data values that remain unchanged over two or more consecutive images. This is a quite complex process, (not illustrated here), based on interpolation and linear prediction (see Figure 21). Spatial redundancy reduction, or intra-frame coding, optimises the representation of pixel values within the current image, by performing a twodimensional DCT transformation on blocks of eight by eight pixels. To achieve a specified compression ratio, the DCT coefficients are quantised according to a quantisation table and a quality parameter given by the bitrate control block. The Huffman encoder further reduces the redundancy of each sequence of DCT coefficients. Since the resulted code words have variable length, a buffer is needed to achieve a uniform bitrate for the encoded sequence. Bit-rate control Encoded sequence DCT coefficient quantising Huffman encoding Buffer
Temporal redundancy reduction
Spatial redundancy reduction
Figure 19. Block diagram of an MPEG video encoder.
MPEG video data hierarchy:
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Figure 20. MPEG video data hierarchy: a) original image; b) a block of luminance and colour differences samples; c) macro-block; d) slice; e) full picture; f) group of pictures; g) sequence.
Interpolation
Prediction
Figure 21. MPEG sequence
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Appendix B: Modulation and channel coding
Networks consists of several layers: the physical layer (can be wired or wireless) that uses a particular modulation technique to map data to physical elements. The data is often organized in channels, that use a particular channel encoding. This encoding provides for some redundancy, in order to correct transmission errors in the physical layer. Finally, networks may have mediumaccess control functions (MAC-layer) that grant access to (a well-defined part of ) the mediumcapacity to a particular user. In this appendix, we will describe the mechanisms used channel encoding (section B.1), modulation techniques (section B.2), and MAC-layer technologies (section B.3). These fields are quite well-known, and advances do not really exist for wireline cabling (except for fibre-based DWDM).
B.1
B.1.1
Channel coding techniques

Communication channel and message transmission
The goal of a telecommunication system is to reliably transfer data from one point or points to another point or points. The medium through which the data is transmitted is called the communication channel or channel for short. A communication channel can be specified by its input symbol, its output symbol, and different parameters (e.g. attenuation, noise) that try to alter the input symbol and deliver a different symbol at the output. There are many channel models for representing different types of physical communication channels. The examples
are:
Binary Symmetric Channel: a model for deep space channels with hard decision, Additive White Gaussian Noise (AWGN) channel: a model for binary deep space channels with soft decision, and telephone channels, Additive Coloured Gaussian Noise channel: a model for channels with memory Interference channels, fading channels: models for virtual channels, useful for quantitative description of the effects of multiplicative noise.
The data, e.g., video, audio, images, sound, files, etc., is transmitted in a sequence of messages. In this stage, one does not have to take into account data redundancy, so we just assume here that messages have been previously applied a "perfect" compression. Message transmission is mainly characterised by its reliability, defined here as a function of the probability of the received (decoded) message to be identical to the original one. An alternative measure for the quality of transmission is the error probability. This is defined as the probability of the received message to be erroneous.
B.1.2 Channel coding
Channel coding enables the transmission of data reliably and efficiently through communication channels. From a channel coding point of view, a communication system consists of three
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components, namely: channel encoder, communication channel and channel decoder, as indicated in Figure 22.
m=(x1, x2, , xn) Channel encoder
C=(y1, y2, , yN) Communication channel Cm=(xc1, xc2, , xcN)

Figure 22. A communication system
Channel decoder m=(x1, x2, , xn)
Transmission channel. Assume a channel is a device with one input, say x, and one output, y (see Figure 23). The symbols x and y are from the input and output alphabets X and Y, respectively. For example, in a binary input and output channel x and y take binary values of 0 or 1, i.e. the alphabets are X=Y={0,1}.
Communication Channel
Figure 23: A communication channel
Mathematically, a channel is defined by a probability transition matrix P(Y|X). More specifically, Px, y= P(Y=y |X=x), i.e., the (x, y) entry of the matrix, means that given the input of the channel to be symbol x, Px, y is the probability of the output of the channel to be y, where x and y are two symbols from X and Y, respectively. For example, Figure 24 shows a Binary Symmetric Channel (BSC), which is a binary input and binary output channel. As illustrated in the figure, a 0 input results in 0 or 1 output with probability 1-p and p, respectively. Similarly, a 1 input results in 1 or 0 output with probability 1p and p, respectively. The term symmetric reflects the symmetry of the crossover probabilities. In a BSC, the domain of p is [0, 0.5), which decreases with increasing the channel signal-to-noise ratio, i.e., with increasing the transmission power. As a specific case, a BSC with p=0 is called noiseless binary channel (a perfect channel), where the received symbol is always the same as the transmitted one. In real life, however, channel errors inevitably occur due to channel distortion, ambient noise and imperfections of physical signalling process. In other words, channel introduces noise into the transmission.
1-p 0 p p 1 1-p
Figure 24: a BSC.
The input and output alphabets of the channel can be discrete (e.g., 0 or 1 in our example) or continuous (a real value). Moreover, a channel can be with or without memory. In a channel without memory every output is dependent only of the corresponding input, however, in channels with memory the output depends also on the previous channel inputs or outputs. Different channels encountered in practice can be modelled in terms of the stochastic model mentioned above. Hereunder, a few important channel models are mentioned. 104
Message transmission Every message m, can be represented as a sequence of symbols from an alphabet X. Without loss of generality, we assume that the redundancy of all messages is extracted using perfect data compression techniques. Moreover, we assume that the set of messages, denoted by M, consists of equal length sequences with shortest length possible. The set of messages represents all possible sequences to be sent over the channel. Every time, the transmitter presents a message m from M to the channel in order to be sent to the receiver. Consequently, the symbols of the message are transmitted one by one through the channel. After the transmission, message m is recovered at the receiver site. Ideally speaking, one would like that his message is correctly recovered at the receiver, i.e., m=m.
m=(x1, x2, , xn)

Figure 25: message transmission.
Communication Channel
m=(y1, y2, , yn)
The reliability of the communication is measured in terms of the probability that mm, denoted Pm= Pr{mm}. This value is also called decoding error probability of message m or block error probability in short. The lower this probability, the more reliable is the communication channel. The maximum (average) value of this error probability can be set according to the requirements of the data in transmission (related to, e.g., those of the corresponding application, the setting and etc). As an example, lets assume that our application requires every message to be transmitted with block error probability Pm=10-2. Moreover, we assume that the channel is a BSC with crossover probability of p. Figure 26 shows the block error probability with respect to the message length n for values of p=10-2,10-3,10-4 and 10-5. To depict these graphs the first order approximation of Pm n p is applied. Pm 1 10-1 10-2 10-3 10-4 n 1 10 102 103 104 p=10-2 p=10-3 p=10-4 p=10-5
Figure 26: The message error probability
Now if the length of our messages is 1000 bits (or more), then only a BSC with crossover probability of 10-5 (or less)48 is capable of conveying messages with such a reliability requirement, i.e., Pm being at most 10-2. Now, how would it be possible to transmit large
48
Note that all these are very high quality channels.

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messages (e.g., of a length of 1000 bits) through a BSC with p=10-2 or 10-3 and maintain such a level of reliability? The following ways are possible to achieve this goal: Directly improve channel quality (in our example this means reduce the value of the crossover probability p). To do so, one can increase the transmission power. This seems out of question due to practical restrictions on the transmission power, or deploy efficient and effective modulation/multiplexing techniques.
Indirectly improve channel quality by using channel coding techniques (in our example this means virtually decrease the value of the crossover probability p). By channel coding one efficiently sends some extra information (increasing the length of the sequences representing the messages) to improve channel performance.
Channel coding An illustration of the channel coding process is given in Figure 27. The length of code word is N, where N n. During the transmission, the channel disturbs/alters a few symbols (or bits in our example) of the code word. At the receiver site, the channel decoder exploits the redundancy introduced by the encoding so that errors occurred during the transmission can be detected and corrected (up to some degree). Generally speaking, the more extra symbols the better (and easier) is the error detection and correction process.
Redundancy of channel coding 111001010 n N
1100010100011010100111101
Cm
1100010100011010100111101
1100010101010010100110101
111011010
1100010100011010100111101
Figure 27: an example to illustrate the concept of channel coding
As an example, consider a BSC where the encoder repeats three times every bit of the message to construct the corresponding code word. This is called a repetition code. After transmitting the result, the decoder can look at every group of three consecutive bits and decide based on a majority vote on the corresponding bit of the message. If there is at most one error in every group of three bits, this coding method can detect and correct all errors. Note that still the decoding might result in an erroneous message when two or three errors occur in a group of three consecutive bits. In other words, for a message of n bits now we have Pm n (3 p2) = 3np2. Figure 28 shows the block error probability with respect to message length for two values of p=10-2 and 10-3.
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Pm 1 10-1 10-2 10-3 10-4 n 1 N: The no. of transmitted bits 3 10 30 102 300 103 3103 104 3104 p=10-2 p=10-3
Figure 28: The message error probability after repeating every bit three times.
As we see in Figure 28, a message error probability of 10-2 is achieved for message length of 3300 bits, while the crossover probability p=10-3. Without such simple coding, the length of message should have been 10 bits to have the same error performance. The reader may notice by now that this improvement is obtained in the cost of extra transmissions, i.e. increasing the channel bandwidth or, equivalently, reducing channel throughput (three times per second, see the last row of Figure 28). In the following section we explain that there are smart coding methods which yield arbitrarily small error probability for messages with arbitrarily large lengths. The amount of redundancy, needed to deliver such a performance, is very marginal, e.g., for a BSC with p=10-3 the required redundancy is about 0.1 percent!
B.2
Modulation techniques
The message symbols at the output of the channel encoder are mapped into electrical signal in order to be transmitted through the physical channel (the waveform channel, see Figure 29). Modulation is used in communication system to match transmitted signals characteristics to channel characteristics, for reducing noise and interference, for simultaneously transmitting several signals over a single waveform channel, and for overcoming some equipment limitations. The stochastic model of a communication channel presented above is a mathematical tool that describes a physical system. From the viewpoint of this report, such a physical system comprises the following components.
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Communication channel
Channel encoder
Modulator Waveform channel
Channel decoder
Demodulator
Figure 29. Modulation within a communication channel
Modulation maps analogue information (e.g., a speech signal) and digital information (e.g., a digitised speech signal) into waveform signals. Analogue modulation techniques include, for example, the famous AM (amplitude modulation), FM (Frequency Modulation) and PM (Phase Modulation) methods. Examples of digital modulation techniques are ASK (Amplitude Shift Keying), PSK (Phase Shift Keying), FSK (Frequency Shift Keying), QPSK (Quadrature Phase Shift Keying), etc. Depending on the channel characteristics, system complexity, and the required transmission speed, one may choose a modulation method, see a comparison in Table 8.3 in [47].
B.3
Multiple access protocols
Communication networks make use of two important medium types as communication (or waveform) channels, namely point-to-point mediums49 and shared mediums.
core network
core network
separate physical medium
shared physical medium
Figure 30: Two important access network configurations.
There are pros and cons of shared medium solutions: bursty traffic: shared medium solutions can more easily support bursty traffic (bursty = high volumes of data during a very short periods of time).
49 In Error! Reference source not found. point-to-point mediums are referred to as separate physical mediums.
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concurrent users: shared medium solutions can support more concurrent users, but the amount of available bandwidth for each individual user will dramatically decrease due to collisions (or the cost of collision-avoidance). security. Shared medium solutions need mechanisms in (preferably) the medium access layer or otherwise in the network-layer to avoid that users receive information that is not meant for them.
Since multiple users share a common pipe, there is a need for a set of rules for sharing that resource in a fair and efficient manner. This set of rules is the Medium Access Control (MAC) protocol. There are different MACs, for instance: Time/frequency sharing: dividing the capacity equally among all users (or proportional to their needs). It is good for continuous (non-bursty) usage. More or less similar to dedicated circuit, only sharing a single physical channel. Random access (or statistical multiplexing of network traffic): It is good for bursty usage. Unlike with a dedicated connection, where the user has a bandwidth pipe tied up whether or not it is being used effectively, in a shared access network the user is only using the resource when it is needed. Advantages of this method are (according to [7]): data over a shared access network provides superior information burst capacity compared to a dedicated circuit a shared access network gives the data over cable subscriber the ability to be online all the time. Third, all users on a shared access network are connected to the same information pipe. This gives the content provider the unique ability to broadcast data streams. This can be an extremely efficient and effective way of delivering services such as streaming stock tickers, news feeds, multi-player games, software downloads, etc. In the dedicated circuit connection (used in, e.g., voice telephony, telephony data modems, ISDN and xDSL) a user establishes a dedicated (i.e., non-shared) connection with a host computer on the other end. This connection is set aside for the users benefit until it is terminated at the users decision (e.g., user hangs-up). Point-to-point mediums are dedicated to an (ordered) pair of users. The corresponding point-topoint channels are advantageous due to their non-interference feature whereby transmission between a pair of nodes has no effect on the transmission between another pair of nodes. This, however, requires that the topology to be fixed at the network design stage. The shared mediums can be used when the point-to-point ones are not economical, not available, or when dynamic topologies are required. In the corresponding shared channels, generally speaking, more than one receiver can potentially receive every transmitted message. Important advantages of a shared medium system are easy deployment, ability to support mobile users, and ability to dynamic channel allocation. Broadcast channels such as radio, satellite and some LANs are a type of shared medium channels, which are clearly superior when a message is destined to a large number of destinations. When a message is destined to a single or a very small number of destinations in a shared channel, however, there should be some processing to direct the message to the right destination(s). Moreover, the transmissions from different sources over such channels interfere. To avoid interference in a shared channel multiple access protocols are used. In this way, the shared resource of the channel is properly allocated to individual point-to-point communications. From the viewpoint of the OSI reference model, multiple access protocols reside within a special layer called Medium Access Control (MAC) layer (a sub-layer between Data Link and Physical layers. Here we use the classifications of [45] for multiple access protocols with some extensions
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and modifications. At the highest level of classification, there are two categories of conflict-free and contention protocols, each described in the following paragraphs. Conflict free protocols ensure a transmission succeeds, whenever made. In other words, the transmission will never be interfered with. Conflict-free transmission can be achieved by allocating the channel to the users either statically or dynamically. Here the channel resources can be viewed from a time (Time Division Multiplexing), a frequency (Frequency Division Multiplexing), a mixed time-and frequency, wavelength (Wavelength Division Multiplexing, WDM) or etc standpoint. Some of the resulting multiple access protocols are: Frequency Division Multiple Access (FDMA) divides the available frequency spectrum in a number of orthogonal frequency channels, which are assigned, on demand, to different users. FDMA can be used both for analogue as well as a digital communications. This simple technique, used extensively in first generation analogue mobile system, has though poor reuse characteristics. One way to increase reuse efficiency is by employing sectored or directional antennas at the cell site. Even with, for example, 3 sectors per cell the best planning gives a typical reuse of once in 7 cells [32], implying reuse factor of 1/7 = 0.143 per cell. Time Division Multiple Access (TDMA) is the most widely used multiple access technique today, both for mobile and fixed wireless local loop. The frequency spectrum in TDMA is also divided, but into a few (wide) bandwidth channels or carriers. Each carrier is used for transmission of multiple time-multiplexed channels. Each such orthogonal channel, or so-called time-slot, could be assigned to a user on demand. The technique can be used only for digital communication. The ability to work with smaller signal to interference ratio in digital domain gives TDMA better reuse factor in mobile communication as compared to the analogue FDMA. For example, with three sectors, a cell reuse factor of 1/4 or even 1/3 is achievable [32]. Code Division Multiple Access (CDMA) enables the use of the same frequency and time slots by different channels. Direct Sequence CDMA (DS-CDMA) emerged in the eighties. Based on spread spectrum techniques used extensively in defence applications for over twenty years, this technique enables definition of near-orthogonal channels in code-space. Every sender uniquely encodes each bit to be transmitted by spreading the bit into 64 or 256 or even 1024 chips. The receiver separates the data of the user by a decoder that correlates the received signal with the code vector associated with that user. On correlation, the interference from other users would become nearly zero and add only a small amount of noise, where as the desired signal will be enhanced considerably. The technique is useful in exploiting the inherent time-diversity from multi-path delay-spread (especially if the spreading is significant). The only problem with CDMA is that as completely orthogonal codes are not possible (especially on the up-links), the total bit-rate supportable from all users using this technique is significantly less than the total bit-rate supportable with TDMA and FDMA technique using the same frequency spectrum. This disadvantage in the CDMA system is made up by better reuse efficiency, as the same spectrum with different set of codes can almost totally be reused in every cell. The theoretical reuse efficiency could be as high as 1.0 (a little less in practice less). With sectored antennas, it is possible to reuse the spectrum in each sector, with a 3-sector cell site resulting in a reuse efficiency of nearly 0.5 per sector. An issue that is as important as reuse of frequency spectrum is fine power control so that more or less equal power from each subscriber set reaches a base station. Such a control mechanism was difficult to implement and delayed widespread use or CDMA for sometime. Fortunately, the problem has been largely overcome today.
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There are different classes of CDMA, some of which are: DS-CDMA : The information is directly multiplied by the code at the high chip rates. FH-CDMA : The carrier frequency is changed according to the code. TH-CDMA : The information is not transmitted continuously. Instead of transmitting continuously, the signal is sent in short bursts where the times of the bursts are decided by the code. Hybrid Modulation : Two or more are used together to combine the advantages of all kinds and to combat the disadvantages of them. MC-CDMA : Multicarrier CDMA, spreading is performed over frequency axis. MT-CDMA: Multitone CDMA, spreading is performed over time axis.(orthogonal frequency division multiplexing)
Multi-Carrier TDMA (MC-TDMA) is a modification of TDMA, where a time frame is divided into time slots, as in TDMA. However, in each time slot, a sender can use any of the several available frequencies. Therefore as in TDMA, the available spectrum is divided into a set of frequencies. The key is that no frequency or time-slot is assigned to any sender and almost all channels are available as a pool for every one to choose from. A technique known as Dynamic Channel Selection (DCS) governs the choice of the channel and is the key to high reuse efficiency. A reuse efficiency of 0.7 to 0.8 may be possible from cell to cell and with a 3 sector cell, one may get a reuse efficiency of almost 0.7 per sector. Fixed Channel Allocation (FCA) versus DCS FCA requires a prior allocation or assignment of certain number of channels (carrier frequencies) to the senders (in wireless communications within a given sector of a cell). This assignment is done by a procedure know as frequency planning. The planning had to be carried out using a worst case scenario assuming the nearest possible distance between the interfering signals. After assigning the channel pool to the base station of a sector, the base station can assign channels to subscribers in the sector on demand. DCS is a totally different approach that does no assignment of channels to any base station or its subscribers. All channels are available to every one. The radio equipment is designed to measure the signal strength that it receives on all channels (using something similar to a spectrum analyser) and thus determine the actual radio environment in its vicinity. It carries out this measurement on a continuous basis, whether it is using a channel or not. This complete knowledge of the radio environment enables a sender to select a channel in which it can communicate best. The key is that even while it is communicating on one channel, it is measuring the radio-environment in all other channels. If it finds that another channel can provide it better communication, it switches to this channel seamlessly. If it does succeed to communicate in the best channel, it tries the next best channel. The DCS is thus based not on worst case scenario, but on actual radio environment. Therefore, it is possible sometimes to reuse a channel even 25m from the other.
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No FCA with worst case planning can come close to this in reuse. DCS gives a factor of 2 to 4 in reuse advantage compared to FCA [28]. In users radio equipment, DCS requires fairly sophisticated radio environment measurement techniques, which nowadays can easily be included in integrated circuits and digital signal processors.
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Appendix C: Access technologies - Technical details
We give some background information on CATV networks (section C.1), subscriber loop access (section C.2) and wireless LAN standards (section C.3).
C.1
Cable TV access networks
Television signals can be transmitted in three ways: broadcast via radio waves using ground antennas, by coaxial tree network of CATV network, or via a satellite using the so-called direct broadcast system. The abbreviation CATV originally meant community antenna television. However, CATV is now usually understood to mean cable TV. It is a television distribution method in which signals from distant stations are received, amplified, and then transmitted by (coaxial or fibre) cable or microwave links to users. In most CATV systems, off-air signals were not available or were very weak because of the large distance of the receiver from television transmitters and multiple signal reflections and shadows cast by buildings in big cities. Cable television is made possible by the technology of coaxial cable. Rigid coaxial cable has a solid aluminium outer tube and a centre conductor of copper-clad aluminium. Flexible coaxial cables outer conductor is a combination of metal foil and braided wire, with a copper-clad, steel centre conductor. The characteristic impedance of the coaxial cable used in cable television is 75 ohms. The well-known principles of transmission line theory apply fully to cable television technology. The most important characteristic of coaxial cable is its ability to contain a separate frequency spectrum and to maintain the properties of that separate spectrum so that it behaves like over-theair spectrum. This means that a television receiver connected to a cable signal will behave as it does when connected to an antenna. Moreover, since the cable spectrum is tightly sealed inside an aluminium environment (the coax cable), a properly installed and maintained cable system can use frequencies assigned for other purposes in the over-the-air environment. This usage takes place without causing interference to these other applications or without having them cause interference to the cable service. New spectrum is created inside the cable by the reuse of spectrum. The principal negative of coaxial cable, however, is its relatively high loss. Coaxial cable signal loss is a function of its diameter, dielectric construction, temperature, and operating frequency (the higher the frequency, the more the loss). An approximate figure is 1 dB of loss per 30.5 meters. Therefore, for a long distance signals must repeatedly be amplified on the way. This introduces non-linearity (interference) to the signal at the end of the cable. The topology of CATV network that has evolved over the years is called tree-and-branch architecture. Many small-and intermediate-sized systems fit this model. When analysed, most large systems can be seen as having evolved from this prototype. There are five major parts to a traditional coaxial cable system described below. 1) Headend is the signal reception, origination and modulation point. 2) Trunk cable transports the signals to the neighbourhood (and vice versa). In doing so, its primary goal is to preserve the quality of signals in a cost-effective way. With coaxial cable, therefore, it is necessary to use broadband amplifiers about every 610 meters [8]. In new systems, so-called Hybrid Fibre Coax (HFC) architecture, fibre is used in the trunk part of the cable network. In this way, it is possible to transfer signals over substantial distances with minimal degradation.
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3) Distribution (or feeder) cable runs in the neighbourhood. The feeder is tapped in order for the flexible drop cables to be connected. The distribution cable interfaces to the trunk cable through an amplifier called a bridge amplifier. 4) Drop cable to connect the home and in-house wiring to the cable system. 5) Terminal equipment refers to consumer electronics.
BA BA
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Systems cable footage in the trunk, distribution and drop parts is 15, 40 and 45 percents respectively. Therefore, using fibre optics in the trunk part of HFC architecture does not require a substantial amount of changes in the infrastructure. Note that CATV originated in areas where good reception of direct broadcast TV was not possible. Now CATV also consists of a cable distribution system to large metropolitan areas in competition with direct broadcasting. Channel capacity. In a CATV network two flows of signals can be recognised: downstream signals, transmitted to the customers homes from the headend, and upstream signals transmitted from the customers homes to the headend. Channel capacity depends on Radio Frequency (RF) bandwidth and it is a useful characteristic for classifying cable systems. As shown in Table 23, there are three types of systems in terms of their highest operating frequency of the downstream signals.
Table 23: Operating frequencies of the downstream signals in CATV [8].
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Operating Frequencies (RF range)
Number of Channels each)
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170 MHz 220 MHz
50 MHz-220 MHz 50 MHz-270 MHz 50 MHz-330 MHz 50 MHz-400 MHz 50 MHz-450 MHz 50 MHz-550 MHz 50 MHz-750 MHz 50 MHz-1,000 MHz
12-22 (single coax) 30(single coax) 40 (single coax) 52 (single coax)/104 (dual coax) 60(single coax)/120 (dual coax) 80 (single coax) 110 (single coax) 150 (single coax)
Medium
280 MHz 350 MHz
Large
400 MHz 500 MHz 700 MHz 950 MHz
According to [8], small-capacity cable systems were constructed from the mid-50s to the late-70s. They account for less than 10% of total plant mileage. Medium-capacity systems account for about 75% of total plant mileage. They serve a wide range of communities, from rural areas (populations of 5,000 to 50,000) to some of the largest systems built in the late-70s. Largecapacity cable systems achieve high channel capacities through extended operating frequencies and through the installation of dual, co-located, coaxial cable. The Time Warner system in Queens, New York, has a 1-GHz (1,000 MHz) capacity with 150 channels. Large-capacity cable systems account for about 15% of total cable plant miles. They are primarily high-tech systems designed for large urban areas previously not cabled. They serve 50,000 to 150,000 customers and consist of 400 to 2,000 miles of plant. They began construction in 1981. Large-capacity systems are designed, and some operate, as two-way cable plant. In addition to the downstream signals to the customers (50 MHz to upper band edge), upstream signals are carried from customers to the cable central headend, or hub node. They are transmitted using frequencies between 5 and 30 MHz. Some recent systems operate at 5 to over 40 MHz. The Full Service Network in Orlando, Florida, also has an upstream path above 900 MHz. A bandwidth of 1 GHz (for both upstream and downstream) contains 160 slots of 6 MHz. These can be allocated to NTSC, HDTV, simulcast, and to new services (such as Internet). Now let determine the packing density of digital information (bits) in the 6-MHz cable channel. This is determined by the modulation method used. The two main competitors are Quadrature Amplitude Modulation (QAM) and Vestigial Side-Band (VSB) modulation. These come in various data-speed capacities. The two most interesting ones for cable applications are 256-QAM or 16-VSB. Both are operate at about the same data rate. The Advanced Television Test Centre (ATTC) completed tests of 16-VSB on cable in Charlotte, North Carolina. The 16-VSB system tested delivered 38.5 Mbit/s in all locations. 256-QAM is expected to be able to deliver similar results [8]. The total throughput in 1 GHz will be 160 38.5 = 6.16 Gbit/s (in total in both directions per CATV network). Thus, it is possible to transmit 12 Digital Video Compression (DVC) movies in 6 MHz at 3 Mbit/s or 1920 DVC movies. The 38.5 Mbit/s per 6 MHz seems that is obtained without applying any channel coding. Assuming that a CATV channel is an AWGN channel, a coding gain of 9dB can be attained. This means that a two dimensional constellation of 256 signal points can be replaced with a 2048 signal point constellation. The gain in bits is log22048/log2256 = 11/8=1.375. Therefore, the final 115
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throughput would be 38.5 Mbit/s 1.375 = 52.94 Mbit/s per 6 MHz or 160 52.94 = 8.47 Gbit/s (in total in both directions per CATV network).
C.2
Copper twisted pairs (subscriber loops)
Normally a twisted pair of copper wires, referred to as Subscriber Loop, connects a customer (subscriber) to the telephone switching offices. Technically speaking, the subscriber loop connects customer premises equipment (CPE) to Central Offices (CO), as shown in Figure 31. The traffic on a wire pair is bi-directional and two so-called hybrids are used in CPE and CO to accommodate two-way traffic on such a single pair of wires. Over 94 percent of U.S. homes and all business are connected to the telephone networks by means of twisted-pair copper wires, of which 40 percent are shorter than 3.2 km [34].
Hybrid Hybrid CPE Tap

Figure 31: Analogue subscriber loop with twisted pair
CO
9 8 7 6 5 4 3 2 1
.
Figure 32: Bit-rate capacity of subscriber loops [34].
A subscriber loop is capable of carrying about 1.5 Mbit/s. More specifically, the bit rate carrying capability of these wires depends on the length of the loop, the type of the wire used, and the presence of loading coils and the bridge taps50. Figure 32 illustrates the capacity of subscriber loops in terms of loop distance (from [34]). Near End Crosstalk (NEXT) is the main limiting factor for the capacity of subscriber loops. To save lines between a cluster of users and a CO, T1 and D1 systems, also known as (digital) Subscriber Loop Carriers (DLC), are used. The figure below shows the general arrangement. T1 is a data stream of 1.544 Mbit/s that consists of 24 digitised voice streams. In other words, each analogue voice signal is sampled and quantified into (8 kilo samples/sec) (8 bits/sample) = 64 kbit/s, then 24 digitised voice streams are multiplexed (together with some other information
50 Tap is basically a branch that has been left from an earlier use of the wire pair, which interferes with high frequency traffic
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bits) to yield a 1.544 Mbit/s data stream. T1 is AT&Ts approach. Within Europe E1 system, which multiplexes 30 voice channels, is adopted.
Users
Remote unit
Subscriber Loop Carrier
Exchange unit
CO
In the past few years some new developments in signal processing capabilities have enabled the traditional copper wires to support data communications at rates up to 5 Mbit/s. Future versions of this technology promise to deliver data rates approaching 50 Mbit/s. These technologies, i.e., analogue modems and xDSL, are described in the following subsections.
Analogue modems
Basically, public switched telephone network is/was designed to connect subscribers via SLs for voice communication. Analogue modems enable two computers to communicate via the public switched telephone network, by translating the computer digital data into a series of high-pitched sounds. These signals are sent over the very same analogue telephone network, i.e., through the subscriber loops, hybrids, and switches. Therefore, they should be restricted to certain power (energy) levels and in bandwidth so as not to interfere with or harm the telephone network in anyway. Moreover, they should use network control signals such as dialling, busy signals and so on in full complying with the requirements of the network.
Public switched telephone network
Although a subscriber loop has a large bandwidth (in the order of Mbit/s), the bandwidth of voice channels in a public switched telephone network is limited to approximately 3.3 kHz by the filters at the edge of the core network. Shannon introduced the concept of channel capacity for power limited Gaussian noise channels (our analogue telephone channel) in terms of channel bandwidth and channel signal-to-noise ratio. For a bandwidth of 2400 Hz to 2800 Hz and signal-to-noise ratio from 24 dB to 30 dB (the figures of a typical voice-telephony channel), a capacity of 24,000 bit/s can be obtained. Shannons theory does not mention how to achieve such a capacity, it simply states that the channel capacity can be approached with suitable techniques. With the advent of trellis coded modulation, the addition of echo cancellation methods and so on, 14,400 and 33,600 bit/s data transmission rates are achieved within V.32 and V.34 technologies. Such modems yield 10 bits per Hertz of bandwidth, a figure that approaches theoretical limits.
Digital Subscriber Line (DSL) variants
The term xDSL refers to different variations51 of DSL, such as IDSL, HDSL, HDSL2, ADSL, SDSL, MSDSL, and VDSL.
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See also http://www.xdsl.com/policies.asp

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IDSL, ISDN Digital Subscriber Line, provides DSL technology over existing ISDN lines. Although the transfer rates for IDSL are about the same as ISDN (144kbit/s v. 128kbit/s), IDSL circuits typically only carry data (not voice). The major benefits of switching to IDSL from ISDN are always-on connections, thus eliminating call set up delays; flat rate billing, instead of per minute fees; and transmission of data over the data network, rather than the PSTN. IDSL uses proven loop technology (based on existing ISDN circuits). IDSL can operate up to 5500m (or longer with repeaters), through a Digital Loop Carrier (DLC) based circuit (unlike other DSL technologies), and at higher speeds than analogue modems (and provides an always-on network connection). IDSL offers a fully symmetric service with low deployment cost of the service. Although based on the ISDN transceiver technology, IDSL does not operate over the ISDN voice switch; it is terminated at the DSLAM (digital subscriber line access multiplexer). This offloads the voice network of the high-utilisation data calls and ensures the service provider can make more efficient use of the existing networks. HDSL, High-rate Digital Subscriber Line, is method to transmit T1/E1 traffic52 over copper wires. Compared to previous methods, HDSL uses less bandwidth, has better noise immunity and does not require so many repeaters. More specifically, HDSL long reach requires repeaters only every 3660m, compared to every 1370m for other T1 provisioning techniques. HDSL technology works at low latency, which makes t suitable for voice communication. In HSDL technology T1 information is transmitted on two copper pairs. Each pair carries a full duplex connection of twelve 64 Kbit/s voice channels with some overhead which comes to 784 Kbit/s line transmission rate. 2-Pair HDSL T1 lines are deployed in masses in North America instead of the legacy T1 connections. A similar progression followed in Europe for E1 transmission. HDSL2, High Bit Rate Digital Subscriber Line 2, is the next generation of HDSL technology. It delivers a full-duplex T1 payload over one copper pair loop. Note that HDSL provides full T1 service (a bi-directional rate of 1.5M bit/sec) over two copper pairs. Moreover, HDSL2 resists better against noise resulting from coexistence with other DSL services. Additional benefits are found in HDSL2's fast deployment and customer turn up, which follows the example set by HDSL. Although HDSL is the most widely deployed DSL, it is not governed by a standard. The pending HDSL2 standard, as defined by the ANSI T1E1.4 committee, contains sufficient technical detail to let vendors deploy interoperable products. HDSL2 uses Pulse Amplitude Modulation (PAM) that offers low transmission latency, making HDSL2 suitable for voice communication. ADSL, Asymmetric Digital Subscriber Line, transmits two asymmetric data streams, where a much higher bandwidth is devoted to the downstream flow than to the upstream one. ADSL takes the advantages of the fact that most applications (video-on-demand, Internet access, remote LAN access, multimedia, etc.) require a low data rate in the upstream direction. In most cases, a 10-to1 ratio of downstream-to-upstream is required [1]. The asymmetry leads to a better performance within cables carrying many pairs of twisted wires (as the cables exit the COs). ADSL cannot support transmission of multiple voice calls and megabit data transfer over the same line. This is due to the significant transmission delay in ADSL modems, which prevents
T1/E1 is used for lines connecting intermediate points and it is not suitable for connections to individual subscribers. The main problem in Legacy T1 was the low reach it could achieve (about 1400 m), requiring repeaters at every such distance.
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them from carrying voice transmission. The only way ADSL can support both voice and data over the same line is by using additional POTS splitting technology. However, even with POTS splitting, ADSL can still only support transmission of one voice channel. The asymmetric nature of ADSL cannot support high-speed upstream transmission. Moreover, the ADSL solution has an inherent high power consumption [38]. SDSL, Single-line (Single-pair) Digital Subscriber Line, similarly to HDSL2 transmits T1/E1 signals over one twisted pair copper lines. Moreover, SDSL is also able to support POTS at the same time. SDSL technology is based on mature HDSL technology and uses single pair lines to transmit at rates from 128 kbit/s up to 2.3 Mbit/s. While HDSL technologies were used for replacing E1 and T1 connections mainly for voice applications, SDSL technology provides a symmetric link for data communication as well. Therefore, SDSL can be used to connect businesses, campuses and residential units to the data networks. Since it is based on HDSL technology with low latency, it can support well (unlike ADSL) both digital voice transmission and data transmission simultaneously on the same copper pair A natural evolution of SDSL is Multi-rate SDSL (MSDSL), a system that enables transmission of a few rates using the same platform. The MSDSL solution can automatically adapt its transmission rate to the physical properties of the copper wire, including wire diameter, line length and noise, enabling a trade-off between rate and distance. VDSL, Very-high-data-rate Digital Subscriber Line, delivers both narrow-band and wide-band services from fibre optic backbone to the subscribers, using the existing copper twisted pairs as the last mile connection. VDSL is the highest DSL technology available and it supports both symmetric and asymmetric applications. Compared to ADSL, VDSL has simpler implementation requirements and larger throughput (the trade off for increased speed is loop length. The VDSL spectrum is specified within the range of 200 kHz to 30 MHz. As a result, VDSL supports also POTSs and ISDN services. ATM over VDSL is the preferred implementation method [54]. VDSL used to be called VADSL, BDSL and ADSL. VDSL is designed to deliver a host of asymmetric broadband services, including digital broadcast TV, video-on-demand, high speed Internet access, distance learning and tele-medicine. VDSL is also designed to deliver symmetrical services for small and medium business customers, to corporate enterprise, high-speed data applications, video tele-conferencing and tele-consulting applications and etc. It can also provide short-haul T1 connections at nT1 data rates. VDSL offers the highest speeds, but is only in limited commercial use. For short distances, 304m to 1370m, it promises data rates of up to 50Mbit/s. As of early 2000, VDSL has appeared in some US-West areas, for example, Gilbert and central Phoenix. Those subscribers report great service, highly integrated -- TV, phone, Internet all down one short line from a local fibre connection53.
Theoretical capacity of subscriber loops
AT&T claims that CAP technology can handle bitrates of 20 to 125 Mbit/s over short distances up to 100m [34]. Generally, the highest bitrate that can be achieved is mainly limited by the particular configuration of the physical lines, so only experimental values can be relevant in this case.
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CATV access versus xDSL
To offer data services over a cable television network, operators must first upgrade the network in an entire region to two-way HFC. This can be a costly process of changing one way amplifiers throughout the network into two-way amplifiers. Next, the cable operator must make a large investment in equipment in the head end station, and allocate one television channel for downstream communication. A few cable operators have avoided some infrastructure investment to two-way HFC by having users send all their upstream data via regular telephone modems in a system known as Telco-Return. Once these infrastructure investments are made, the only additional cost for adding an additional subscriber is the cable modem at the subscribers end. This is in contrast to ADSL, where with each new subscriber the data CLEC must purchase a new line card at the central office in addition to the DSL modem that must be purchased on the subscribers end. Cable modem networks have a smaller marginal cost for an additional subscriber, but a greater up-front fixed cost from the region-wide network improvements that are necessary before anyone can be offered cable access service. The bandwidth of a cable modem network is asymmetric. The downstream bandwidth varies between 27 Mbit/s and 40 Mbit/s, while the upstream bandwidth varies between 500kbit/s and 10 Mbit/s (depending on the amount of spectrum allocated for the service, and the noisiness of the network.) Furthermore, the users at the end of each fibre trunk line, which can number between 500 - 5000 subscribers, share both the upstream and downstream bandwidth. In contrast, ADSL bandwidth is not shared among multiple subscribers.
C.3
WLAN standards
IEEE-802.11 standard defines a MAC layer and a physical layer for WLAN. The MAC layer provides to its users both connection-based and contention-free access control. The access methods in IEEE802.11 MAC are Distributed Coordination Function (DCS) -a Carrier Sense Multiple Access with Collision Avoidance (CSMA/CA) MAC protocol- and Point Coordination Function (similar to a polling system that uses a point coordinator to determine which station has the right to transmit). Three different technologies can be used as physical layer: Infrared, FHSS and DSSS. IEEE802.11 is standardized in 1997 [6]. Following IEEE802.11 standard, both 802.11b and 802.11a standards were proposed and ratified out of the US-based Institute of Electrical and Electronic Engineers (IEEE). These two, that use different unlicensed frequency bands, are explained below. Wi-Fi (IEEE-802.11b) standard can achieve raw data rates of 11Mbit/s, has a range of about 100 meters, and uses the Industrial, Scientific, and Medical (ISM) band between 2.4000 2.4835 GHz. Wi-Fi offers an 11 Mbit/s data rate, which translates into approximately 5-7 Mbit/s of actual message throughput. This amount is shared among all network users who are using it simultaneously, and is managed through a CSMA/CA technique modeled on its Ethernet wired equivalent. There are several potential speed increases in store for Wi-Fi, which will come from the IEEE 802.11 Task Group g. The IEEE is currently considering two proposals for adoption: 802.11 Packet Binary Convolutional Code (802.11gPBCC), and 802.11 Orthogonal Frequency Division Multiplexing (802.11gOFDM). IEEE-802.11a standard uses the Unlicensed National Information Infrastructure (U-NII) and ISM bands in the United States (in the 5GHz band, which is actually a conglomerate of several 120
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bands: 5.150 5.250, 5.250 5.350, 5.470 5.725, and 5.725 5.875). This technology can achieve raw data rates of up to 54 Mbit/s for short distances (for about 10-15 meters), and a data rate comparable to Wi-Fi (about 9-12 Mbit/s) for a range of about 50 meters. The IEEE-802.11a standard is based on OFDM modulation and will theoretically achieve a 54 Mbit/s data rate, or approximately 31 Mbit/s of throughput for a single network (shared among the users). This easily supports several simultaneous occurrences of streaming video. Additionally, 802.11a nodes share bandwidth efficiently using the same CSMA/CA techniques used in Wi-Fi systems, thus allowing a large number of users access to high wireless data rates. Providing high data rate (36-54 Mbit/s) levels close to the AP (10- to 15 meters), IEEE-802.11a becomes attractive for dense user environments that also require high throughput. An important purchasing consideration for any networking technology is the amount of data rate (bandwidth), or throughput, it provides to each network user or to all WLAN network users, and how well that throughput can support the applications running on the network.
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Babet Final Report

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Babet Final Report

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Broadband applications on limited bandwidth networks

A survey on the suitability of current and emerging access technologies

64 66 67 69 71 71 74 74 75 77 77 77 78 79 79 80 80 81 81 82 87 91 91 91 91 91 93 94 95 96 96 96 97 97 98 99 100 10 3 103 103 103 107 108 11 3 113 116 120

Dit directoraat is onderdeel van het Ministerie van Verkeer en Waterstaat.

Conferencing Interactieve toegang tot documenten en berichten

A/V streaming en A/V multicast

e-learning, e-government Gezondheidszorg Telewerken

0.5 0.5 0.014

Laag Laag Laag

applicatie encoder transcoder Multimedia Distributie Technieken (Toegangs) netwerken

Ontwikkelingen in data compressietechnieken

Beschikbare Bandbreedte (down/upstream) [Mbit/s] 5/0.64 51/6.5

Basisdiensten (geselecteerd door DGTP)

0.15symmetrisch 0.2symmetrisch 2.8symmetrisch 11symmetrisch

+ (MPEG) + (MPEG) + (MPEG) + (MPEG)

FSO satelliet Near Tspectrum T-spectrum

155symmetrisch 0.5- 0.05 622symmetrisch 100000symmetrisch

De getallen in deze kolom zijn de werkelijke getallen of realistische schattingen.

Table 1. Geschiktheid van toegangstechnologien voor ruwe audio-video data streaming

Source video format

Source audio format

Relative quality class

Required bandwidth [Mbit/s]

Geschiktheid van verschillende toegangstechnologien

Broadcast TV/DVD Near VHS QuarterVHS -

160 120 9.5 2.5

Source video format

Source audio format

Relative quality class

Required bandwidth [Mbit/s]

Geschiktheid van verschillende toegangstechnologien

Broadcast TV/DVD Near VHS QuarterVHS -

1.5 0.3 0.1 0.05

Table 3. Geschiktheid van toegangstechnologien voor ruwe data voor videoconferencing

Source video format

Source audio format

Relative quality class

Downstream/ upstream bandwidth [Mbit/s]

Geschiktheid van verschillende toegangstechnologien

465/155 117/39 32/10.9 10.9/10.9 31/10.3 11/3.7 2.79/0.9

Source video format

Source audio format

Relative quality class

Downstream/ upstream bandwidth [Mbit/s]

Geschiktheid van verschillende toegangstechnologien

3/1 1.1/0.36 0.56/0.2 0.2/0.2 0.54/0.18 0.16/0.06 0.05/0.03

Maximum bandwidth [Mbit/s] 0.2 0.2 340 8 1.5 0.25 2 20 6

application encoder transcoder Multimedia Distribution Techniques (Access) network

Advances in data compression

Adaptive multicasting. This is used in point-to-multipoint content distribution settings, where

Advances in Access technology

Achievable Bandwidth (down/upstream) [Mbit/s] 5/0.64 51/6.5

Basic categories of services (selected by DGTP)

0.15symmetric 0.2symmetric 2.8symmetric 11symmetric

+ (MPEG) + (MPEG) + (MPEG) + (MPEG)

155symmetric 0.5- 0.05

Near Tspectrum T-spectrum

Table 5. Suitability of access technologies for raw data audio/video streaming

Source video format

Source audio format

Relative quality class

Required bandwidth [Mbit/s]

Suitability of several access technologies