Università degli Studi di Roma “LA SAPIENZA”infocom.uniroma1.it/~robby/Tesi/Cioccari...
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Anno Accademico 2005/2006 Università degli Studi di Roma “LA SAPIENZA” Facoltà di Ingegneria Dipartimento di Scienza e Tecnica dell’Informazione Corso di Laurea Specialistica in Ingegneria delle Telecomunicazioni Tesi di Laurea Modelli architetturali per soluzioni di accesso innovative basate sulla convergenza fra reti 3G/4G, DVB-H e WiMAX, in rispetto del Framework IMS Candidato Cioccari Francesca Relatore Chiar.mo Prof. Roberto Cusani Correlatore Ing. Gennaro Galdo CONSEL - Junior Consulting
Transcript of Università degli Studi di Roma “LA SAPIENZA”infocom.uniroma1.it/~robby/Tesi/Cioccari...
Anno Accademico 2005/2006
Università degli Studi di Roma “LA SAPIENZA”
Facoltà di Ingegneria Dipartimento di Scienza e Tecnica dell’Informazione
Corso di Laurea Specialistica in Ingegneria delle Telecomunicazioni
Tesi di Laurea Modelli architetturali per soluzioni di
accesso innovative basate sulla convergenza
fra reti 3G/4G, DVB-H e WiMAX, in rispetto
del Framework IMS
Candidato
Cioccari Francesca
Relatore Chiar.mo Prof.
Roberto Cusani
Correlatore Ing. Gennaro Galdo
CONSEL - Junior Consulting
Alla mia famiglia
Table of Contents
I
TTAABBLLEE OOFF CCOONNTTEENNTTSS
INTRODUZIONE...................................................................................................... 1
Il Progetto Wind – “Convergent Access Network”.................................................... 5
Struttura della Tesi ..................................................................................................... 6
CHAPTER ONE
1. TECHNICAL OVERVIEW.............................................................................. 9
1.1 UMTS Release 6 (Universal Mobile Telecommunication System) .......... 11
1.1.1 Introduction........................................................................................ 11
1.1.2 UMTS Services and Features ............................................................ 11
1.1.3 UMTS Architecture ........................................................................... 13
1.1.3.1 Core Network..................................................................................... 14
1.1.3.1.1 CS Domain.................................................................................... 14
1.1.3.1.2 PS Domain .................................................................................... 15
1.1.3.1.3 IP Multimedia subsystem (IMS)................................................... 15
1.1.3.2 UTRAN.............................................................................................. 15
1.1.3.2.1 Radio Access................................................................................. 15
1.1.3.2.2 WCDMA channels........................................................................ 16
1.1.3.3 User Equipment ................................................................................. 17
1.1.4 General Protocol Architecture ........................................................... 18
1.2 Mobile WiMAX (Wireless Interoperability for Microwave Access)
[802.16e] ................................................................................................................ 19
1.2.1 Introduction........................................................................................ 19
1.2.2 Features .............................................................................................. 21
1.2.2.1 Physical Layer.................................................................................... 21
1.2.2.1.1 OFDMA ........................................................................................ 21
1.2.2.1.2 Scalable OFDMA ......................................................................... 22
1.2.2.1.3 TDD Frame Structure ................................................................... 23
1.2.2.1.4 Other Advanced PHY Layer Features .......................................... 25
1.2.2.2 MAC Layer ........................................................................................ 26
1.2.2.2.1 Quality of Service ......................................................................... 26
Table of Contents
II
1.2.2.2.2 MAC Scheduling Service ............................................................. 27
1.2.2.2.3 Mobility Management................................................................... 28
1.2.2.2.4 Security ......................................................................................... 29
1.2.2.3 Advanced Features of Mobile WiMAX............................................. 30
1.2.2.3.1 Smart Antenna Technologies........................................................ 30
1.2.2.3.2 Fractional Frequency Reuse.......................................................... 31
1.2.2.3.3 Multicast and Broadcast Service (MBS) ...................................... 33
1.2.3 End-to-End WiMAX Architecture..................................................... 34
1.3 DVB-H (Digital Video Broadcasting-Handheld) ...................................... 37
1.3.1 Introduction........................................................................................ 37
1.3.2 Overview of the system ..................................................................... 37
1.3.3 Structure of a DVB-H receiver .......................................................... 39
1.3.4 Standards and Protocol Stack ............................................................ 42
1.3.5 Data Link Layer ................................................................................. 46
1.3.5.1 Time-Slicing ...................................................................................... 47
1.3.5.1.1 Handover Considerations.............................................................. 48
1.3.5.2 MPE-FEC........................................................................................... 49
1.3.6 Physical Layer: .................................................................................. 50
1.3.6.1 4K Mode and In-depth Interleavers ................................................... 50
1.3.6.2 DVB-H Signalling: TPS-bit Signalling ............................................. 51
1.3.7 DVB-H Networks .............................................................................. 51
1.3.7.1 The IP-DataCasting System............................................................... 51
1.3.7.2 Broadcasting Spectrum ...................................................................... 52
1.3.7.3 DVB-H Network Architectures ......................................................... 53
1.3.7.3.1 DVB-H Standalone (Dedicated DVB-H Networks) ..................... 53
1.3.7.3.2 Sharing with DVB-T..................................................................... 54
CHAPTER TWO
2. ACCESS NETWORK ANALYSIS AND COMPARISON FOR UMTS,
WiMAX AND DVB-H ............................................................................................. 58
2.1 Introduction................................................................................................ 58
2.2 Access Networks Analysis: UMTS UTRAN............................................. 59
Table of Contents
III
2.2.1 Node B functionality.......................................................................... 61
2.2.2 RNC functionality.............................................................................. 61
2.2.3 UTRAN Functions ............................................................................. 63
2.3 Access Networks Analysis: WiMAX ........................................................ 71
2.3.1 Base Station ....................................................................................... 72
2.3.2 ASN GW (Access Service Network Gateway).................................. 73
2.3.3 Functions............................................................................................ 74
2.4 Access Networks Analysis: DVB-H.......................................................... 79
2.4.1 IP Encapsulator .................................................................................. 83
2.4.2 Multiplexer......................................................................................... 83
2.4.3 Modulator........................................................................................... 84
2.4.4 Features .............................................................................................. 84
2.5 Access Networks Comparison ................................................................... 86
2.6 A UNIQUE ARCHITECTURAL BLOCK MODEL FOR ACCESS
NETWORK ANALYSIS AND COMPARISON.................................................. 90
2.6.1 Air Interface Control Module ............................................................ 91
2.6.2 Enforcement Module ......................................................................... 91
2.6.3 Decision & Control Module .............................................................. 92
CHAPTER THREE
3. CORE NETWORK: IMS TECHNICAL DESCRIPTION.......................... 93
3.1 Introduction................................................................................................ 93
3.1.1 Why IMS?.......................................................................................... 94
3.1.2 The Introduction of IMS .................................................................... 95
3.2 The Architecture of IMS............................................................................ 96
3.2.1 The HSS and SLF Databases ............................................................. 98
3.2.2 The CSCF .......................................................................................... 98
3.2.2.1 P-CSCF .............................................................................................. 98
3.2.2.2 The I-CSCF........................................................................................ 99
3.2.2.3 The S-CSCF..................................................................................... 100
3.2.3 The AS ............................................................................................. 100
3.2.4 The MRF.......................................................................................... 101
Table of Contents
IV
3.2.5 The BGCF........................................................................................ 101
3.2.6 The PSTN/CS Gateway ................................................................... 102
3.2.7 Home and Visited Networks............................................................ 102
3.2.8 Identification in the IMS.................................................................. 102
3.2.8.1 Public User Identification ................................................................ 103
3.2.8.2 Private User Identities...................................................................... 103
3.2.8.3 The relation between Public and Private User Identities................. 103
3.2.8.4 Public Service Identities .................................................................. 103
3.2.9 SIM, USIM and ISIM in 3GPP........................................................ 103
3.3 Protocols .................................................................................................. 105
3.3.1 Session Control Protocol ................................................................. 105
3.3.2 Media Plane Protocol....................................................................... 105
3.3.3 Security and Authentication Protocol .............................................. 105
3.4 SIP Protocol ............................................................................................. 106
3.4.1 SIP functionality .............................................................................. 107
3.4.2 SIP Entities ...................................................................................... 108
3.4.3 Messages .......................................................................................... 109
3.4.3.1 Start Line.......................................................................................... 109
3.4.3.2 Headers ............................................................................................ 110
3.4.3.3 Message Body.................................................................................. 110
3.4.4 Management of a SIP session .......................................................... 111
3.4.4.1 Session Establishment and Termination .......................................... 111
3.4.4.2 Call Redirection ............................................................................... 112
3.4.4.3 Call Proxying ................................................................................... 113
3.5 IPv6.......................................................................................................... 115
3.5.1 A new IP standard............................................................................ 115
3.5.1.1 A New Header ................................................................................. 115
3.5.1.2 Addressing ....................................................................................... 117
3.5.1.2.1 Unicast Addresses....................................................................... 118
3.5.1.2.2 Anycast Addresses ...................................................................... 119
3.5.1.2.3 Multicast Address ....................................................................... 119
3.5.1.3 Mobile in IPv6 ................................................................................. 119
Table of Contents
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3.5.2 Differences Between IPv4 and IPv6................................................ 121
CHAPTER FOUR
4. EDGE NETWORK PROPOSAL FOR UMTS, WiMAX AND DVB-H,
AND CONVERGENT ARCHITECTURE ......................................................... 122
4.1 Introduction.............................................................................................. 122
4.2 What is the Edge Network? ..................................................................... 123
4.3 The Edge Network for UMTS Release 6 (PS Domain)........................... 126
4.3.1 Serving GPRS Support Node (SGSN)............................................. 127
4.3.2 Gateway GPRS Support Node (GGSN) .......................................... 128
4.3.3 Data Bases....................................................................................... 129
4.3.3.1 Home Subscriber Server (HSS) ....................................................... 128
4.3.3.2 The Home Location Register (HLR) ............................................... 130
4.3.3.3 The Authentication Centre (AuC).................................................... 130
4.3.3.4 HSS logical functions ...................................................................... 131
4.3.4 The Equipment Identity Register (EIR)........................................... 132
4.3.5 Interfaces.......................................................................................... 133
4.3.5.1 Interface between SGSN and RNS (Iu_PS-interface) ..................... 133
4.3.5.2 Interface between SGSN and HLR (Gr-interface)........................... 133
4.3.5.3 Interface between SGSN and GGSN (Gn- and Gp-interface) ......... 133
4.3.5.4 Signalling Path between GGSN and HLR (Gc-interface) ............... 133
4.3.5.5 Interface between SGSN and EIR (Gf-interface) ............................ 134
4.3.6 Protocol Analysis and Definition..................................................... 134
4.4 The Edge Network for WiMAX .............................................................. 135
4.4.1 CSN-WAG....................................................................................... 136
4.4.2 Interfaces.......................................................................................... 137
4.4.3 Protocol Analysis and Definition..................................................... 138
4.5 The Edge Network for DVB-H................................................................ 139
4.5.1 Protocol Analysis and Definition..................................................... 141
4.6 CONVERGENT ARCHITECTURE PROPOSAL ................................. 143
4.6.1 Edge Network Evolutions................................................................ 146
4.6.2 Conclusions...................................................................................... 150
Table of Contents
VI
APPENDIX A: Implementation ........................................................................... 153
A.1 Introduction.............................................................................................. 153
A.2 Case Pilot of a Convergent Service: m-Advertising Game ..................... 153
A.3 “Mapping” the convergent service on the convergent architecture ......... 155
A.4 Proxy Simulation ..................................................................................... 157
APPENDIX B: Telecommunications Market Overview .................................... 165
B.1 The International Scenario....................................................................... 167
B.2 The Italian Scenario ................................................................................. 168
B.2.1 Players in the Italian Market ............................................................ 168
BIBLIOGRAPHY.................................................................................................. 170
Introduzione
1
IINNTTRROODDUUZZIIOONNEE
Il nuovo modo di comunicare sta evolvendo verso l’integrazione, in un'unica
struttura, di tutti i servizi che generalmente sono presenti in un’organizzazione (cioè
servizi dati, telefonici e video). Il protocollo IP sembra essere ormai la struttura di
riferimento per questo nuovo sviluppo tecnologico.
I modelli architetturali delle reti attuali, però, prevedono una netta separazione
dei servizi voce, video e dati. I servizi sono amministrati generalmente in modo
autonomo a livello di cablaggio, di apparati e a livello applicativo. Il nuovo modello
di rete, quindi, dovrebbe prevedere una struttura indipendente dalla applicazione, con
apparati dotati di caratteristiche multiservizio e flessibilità, adatti a concentrare le
diverse tipologie di traffico e convogliarle su canali unici sia in ambito LAN sia
WAN. Questa nuova occorrenza sta muovendo i grandi operatori delle
telecomunicazioni, e i protagonisti come Nortel, Samsung, Alcatel insieme con quelli
del settore del networking come Cisco e 3Com, stanno già avanzando le loro
proposte parlando di convergenza.1
Finora, lo sviluppo dei servizi voce, dati e video era avvenuto prevalentemente
in modo autonomo e la loro implementazione era gestita, molto spesso, da operatori
diversi. Dal punto di vista dell’utente, l’offerta dei servizi telematici era troppo
rigida. Dal punto di vista dell’efficienza e dell’ottimizzazione del traffico, una rete
dati richiede larghezze di banda decisamente superiore rispetto a quella voce,
soprattutto a causa dello sviluppo dei servizi come videoconferenza, video streaming,
e-commerce, ecc... La nascita di offerte e soluzioni innovative scaturiscono quasi
sempre da nuove esigenze o precise richieste. Nel settore delle telecomunicazioni e
del networking si sta assistendo ad una crescita vertiginosa del traffico dati contro un
andamento della domanda praticamente costante del traffico voce, tanto che si stanno
diffondendo nel mercato soluzioni che trattano la trasmissione voce come un
particolare pacchetto dati. Da queste realtà sono nate tecnologie mirate
all’ottimizzazione delle architetture di reti con l’obiettivo di trasportare su un unico
1 “Le reti convergenti”, Matteo Santucci
Introduzione
2
supporto video, voce e dati implementando, cioè, una rete convergente. D’altro canto
è ormai largamente riconosciuto che l’IP (Internet Protocol) sarà il protocollo di
riferimento per qualunque tipo di rete. L’adozione di questo protocollo da parte di
numerosi operatori e utilizzatori per il trasporto video, voce, dati implica quasi
obbligatoriamente che una rete convergente si basi anch’essa sull’IP. Anche il
protocollo SIP ha un ruolo cardine in questa evoluzione e ampliamento dell’offerta
dei servizi. La sua flessibilità permette, infatti, rimanendo in uno stesso contesto
tecnologico, di fornire servizi con caratteristiche molto differenziate, e di integrare,
in una esperienza cliente omogenea, diverse forme di comunicazione come la
conversazione, la messaggistica, lo streaming di video.
Questa dinamicità dell'industria mondiale delle telecomunicazioni, garantita
dai numerosi exploit tecnologici e dalle nuove soluzioni proposte di continuo dai
player, inducono gli analisti a mantenere alta l'attenzione sui driver che stanno
trainando il mercato, e cioè la domanda costante di connettività a banda larga con
elevata capacità e di servizi mobili sistematicamente più sofisticati e convergenti,
con un rapporto qualità-prezzo sempre più competitivo.2 L’intero settore delle
telecomunicazioni è attualmente caratterizzato da grandi innovazioni che ne stanno
cambiando radicalmente i connotati. E’ normale che, in queste situazioni di forte
transizione tecnologica, praticamente tutto venga sconvolto e messo in discussione. Il
vecchio declina mentre il nuovo per consolidarsi richiede scelte difficili,
investimenti, nuove dimensioni imprenditoriali e manageriali, e soprattutto tempo.
Come già detto, la grande dinamicità e il ristagnare delle rendite legate ai
tradizionali servizi di telefonia fanno da preludio ad un profondo mutamento: la
rivoluzione della convergenza. La strategia che stanno perseguendo tutti gli operatori
di mercato è quella che punta a realizzare un’infrastruttura unica di trasporto, basata
su protocollo IP, lo stesso della rete internet, in grado di supportare servizi innovativi
in banda larga sia per il fisso che per il mobile. “In un futuro ormai prossimo,
diciamo cinque anni, ci sarà la convergenza su un'unica piattaforma delle tecnologie
di comunicazione. L’utilizzo dei telefoni portatili cambierà: il cellulare, grazie alla
2 Insight Research, "The Future of Telecommunications 2006 – 2011”. Maggio 2006
Introduzione
3
convergenza e alla possibilità di trasmettere e ricevere una grande quantità di dati,
diventerà un apparecchio molto più simile ad un tradizionale PC. Ciò significa che
sulla stessa rete potremo contemporaneamente comunicare a voce, navigare su
Internet, vedere televisione, fare videogiochi da remoto, discutere in
videoconferenza.”. E’ quanto sottolinea Roberto Guadagni, responsabile
infrastrutture di rete e di calcolo dell'Enea (Ente per le nuove tecnologie, l’energia e
l’ambiente).3 Tale infrastruttura rappresenta la base per lo sviluppo di una
piattaforma comune attraverso la quale offrire gli stessi prodotti e servizi su
qualunque terminale (PC, TV, Telefonino, ecc), mediante diverse tecnologie (ADSL,
HSDPA, DVB-H, UMA, ecc) e consentendo al cliente di decidere dove, come, e
quando accedere a tali servizi.
In questo contesto, l’offerta di Mobile TV, la fruizione di contenuti televisivi su
apparecchi mobili tramite l’utilizzo della tecnologia DVB-H (Digital Video
Broadcasting - Handheld), appare ricca e ben distribuita. Grazie allo standard DVB-
H sarà possibile guardare la televisione sul cellulare in maniera interattiva e dunque
rendere concreto il concetto di portabilità. Con lo sviluppo di offerte multimediali
VoD (Video on Demand) e PVR (Personal Video Recorder), combinate con il lancio
della Mobile TV, il mercato entrerà inoltre nell’era della Personal TV.
La convergenza non rende più possibile la distinzione tra telefonia fissa e
mobile. Proprio quest’ultima sta acquisendo ampiezze di banda che per molti servizi
non farà rimpiangere la potenza delle reti fisse. E manterrà quel grande vantaggio
che è la mobilità totale, requisito ormai indispensabile per gli utenti finali. Ma le reti
fisse potranno riappropriarsi almeno di una parte del traffico che hanno perso con i
nuovi sviluppi degli accessi WiFi e WiMax, che consentiranno di fare con il
telefonino, da casa, dall’ufficio e forse dalla strada comunicazioni che di fatto
transiteranno sulla rete fissa.
3 Conferenza "e-Infrastrutture per lo sviluppo", organizzata dal Consortium GARR, ideatore e gestore della rete telematica nazionale per l'Istruzione, l'Università e la Ricerca Scientifica.
Introduzione
4
Ormai la marcia verso grandezze di banda sempre più importanti è
inarrestabile. Si calcola che entro il 2015 il 20 per cento degli europei potrà contare
su una banda di 100 mega, il che consentirà di fare assolutamente tutto».4
La banda larga, verso la quale si stanno rivolgendo con ingenti investimenti le
principali compagnie che offrono servizi di telefonia fissa, non è un semplice
passaggio dalla banda stretta della telefonia vocale a quella larga che ci consente già
di vedere la televisione attraverso le reti di telecomunicazione. E’ una svolta
progressiva che apre nuovi mondi di servizi, non solo di intrattenimento, ma
soprattutto di utilità per la produttività delle imprese e per lo sviluppo di
indispensabili servizi sociali. Molti servizi innovativi di comunicazioni mobili, in
primis la navigazione su Internet, sono limitati o quasi impossibili per la carenza di
banda, e lo stesso accade nelle comunicazioni fisse. Servono servizi innovativi per la
sanità, per i trasporti, per la sicurezza, per l’insegnamento a distanza, per la
formazione professionale, per la gestione aziendale per giustificare nuovi
investimenti in banda larga, e la convinzione generale è che sarà la crescita della
banda a stimolare la nascita dei servizi nuovi.
Infine, la forte avanzata delle tecnologie ti telecomunicazioni introducono
importanti fattori: la capacità dei sistemi basati su protocollo IP di combinare e
trasportare ovunque voce, video e dati; la forte varietà di soluzioni wired e wireless
proposte a livello locale; la continua richiesta di connettività e la conseguente guerra
al ribasso delle tariffe; la crescente domanda di soluzioni per garantire la sicurezza
informatica e, di pari passo, il compito sempre più arduo di proteggere con efficacia
gli internauti dai mille pericoli della Rete.
4 Carlo Mario Guerci, Professore Ordinario di Economia Politica presso la Università degli Studi di Milano
Introduzione
5
Il Progetto Wind – “Convergent Access Network”
Questo lavoro è il risultato di un progetto commissionato dalla divisione reti di
Wind Spa. Il progetto, della durata di sei mesi, prevedeva l’analisi del mercato delle
telecomunicazioni, ed in particolare lo studio dei soggetti coinvolti e delle loro
strategie di mercato. Sono state studiate le tecnologie e le innovazioni nel campo
della comunicazione e della telefonia, sia fissa che mobile, con una focalizzazione
sulle nuove potenzialità offerte. Sono stati proposti nuovi servizi convergenti che
incontrassero queste nuove esigenze e le possibilità offerte dalle tecnologie
innovative. Infine, è stato analizzato il mercato ed i suoi trend, e sono stati
evidenziati i bisogni espressi dai consumatori e le loro nuove esigenze.
Il lavoro è scritto in Inglese, su specifica richiesta del committente, che con la
recente acquisizione da parte di Weather Investments ha assunto un carattere
internazionale. L’attività è stata sottoposta a periodiche verifiche da parte del
committente e del referente aziendale, Maria Rita Spada, mediante diversi incontri,
con lo scopo di verificare lo stato di avanzamento dei lavori e di definire gli step
successivi. Infatti, per poter raggiungere l’obiettivo del lavoro si è proceduto tramite
diverse fasi, sintetizzate nella Figura 1.
Figura 1: Fasi del Progetto "Convergent Access Network"
Introduzione
6
Come si evince, il progetto ha una doppia valenza: una di carattere prettamente
tecnologico e l’altra di carattere economico. Proprio questa interdisciplinità ha
contribuito all’accrescimento delle mie competenze trasversali e all’approfondimento
delle tematiche d’interesse di questa tesi.
Il lavoro da me svolto nell’ambito di questo progetto è rivolto alla componente
tecnologica che ha reso possibile la proposta di un servizio integrato basato su una
nuova architettura convergente.
Sviluppato nell’ambito del programma formativo Junior Consulting, presso il
Consorzio ELIS (CONSEL) di Roma, il progetto ha visto lavorare, sotto il
coordinamento del Team Leader Gennaro Galdo, quattro laureandi in diverse
discipline: Grace Augustine, laureata in B.A. Organizational Studies presso
l’università del Michigan (U.S.A.), Andrea Chiarello, laureando in Ingegneria
Elettronica (indirizzo Microelettronica), presso l’università di Bari, e Tommaso
Chiocchi, laureando in Ingegneria Gestionale, presso l’università Tor Vergata di
Roma.
Struttura della Tesi
L’obiettivo di questa tesi è quello di individuare un possibile modello
architetturale che favorisca soluzioni di convergenza fra varie tecnologie di accesso
nomadiche, mobili e di broadcasting. Le tecnologie studiate per questo scopo sono:
DVB-H, UMTS Release 6, Mobile WiMAX (802.16e), IMS, SIP e IPv6.
Come prima fase è stato necessario uno studio approfondito sulle reti di
accesso delle principali tecnologie (UMTS, WiMAX e DVB-H) che sono state scelte
per l’integrazione [Capitolo 1]. In particolare, sono state analizzate le varie
caratteristiche che contraddistinguono queste tre tecnologie standardizzate e le
rendono le più efficienti nel loro specifico ambito d’interesse. Questo studio è stato
fatto per poter capire fino a che punto potersi ‘spingere’ al fine di ottenere la
convergenza verso un’unica rete di accesso.
Introduzione
7
La seconda fase, infatti, ha come scopo l’individuazione proprio di un modello
architetturale unico per una generica rete di accesso [Capitolo 2]. Per questo è stata
necessaria dapprima un’analisi delle reti di accesso rivolta principalmente ad
individuare dove e come queste tecnologie si agganciano al “mondo IP” e quali
specifiche funzionalità ogni singolo dispositivo fisico svolge [§ 2.2 - 2.4]. A valle di
questa analisi preliminare è stato possibile eseguire un paragone a livello funzionale
tra i vari elementi che compongono le diverse reti di accesso per sottolineare cosa
hanno in comune (funzioni parallele o simili) ed individuare le diversità e peculiarità
di ogni tecnologia [§ 2.5]. Ne deriva un modello architetturale unico tra le varie
tecnologie di accesso in cui vengono individuati dei blocchi funzionali. Ognuno di
questi blocchi è composto da funzionalità, necessarie per la rete di accesso, che sono
dislocate o accorpate nei dispositivi delle reti di accesso delle tre tecnologie prese in
considerazione [§ 2.6].
Come passo successivo si è ritenuto opportuno approfondire l’analisi della
piattaforma IMS (IP Multimedia Subsystem) [Capitolo 3], la piattaforma
multimediale che permette proprio l’integrazione tra diversi servizi, basandosi sul
protocollo SIP. La rete IMS è stata individuata, quindi, come Core Network all-IP
based.
Un ulteriore passo verso una totale convergenza ha reso necessario un
approfondimento su come poter legare le reti di accesso alla Core Network unica
IMS. Questo ha portato a dover tener conto di un insieme di elementi necessari per
ottenere tale connessione. Il risultato di questo lavoro è la proposta di una Edge
Network [Capitolo 4]. L’approccio utilizzato nella definizione di questa sezione di
rete è stato quello di suddividere il lavoro in varie fasi. Come primo passo sono stati
individuati i requisiti e le funzionalità che la Edge Network deve soddisfare [§ 4.2].
Sulla base di questo sono stati definiti gli elementi di cui la Edge Network è
composta per le tre specifiche tecnologie di accesso, UMTS, WiMAX e DVB-H. Per
ognuno di questi dispositivi è stata svolta un’analisi approfondita e soprattutto uno
studio ed una definizione del loro stack protocollare [§ 4.3 - 4.5]. Successivamente si
è presentato un modello di architettura convergente, composto da Access Network,
Edge Network e Core Network, ed è chiaramente spiegato in che modo la Edge
Network permette di ottenere tale architettura [§ 4.6]. Ovviamente, la proposta finale
Introduzione
8
di questa tesi è il risultato di ampie valutazioni e considerazioni, per cui è necessaria
anche una descrizione dell’evoluzione che la Edge Network, e di conseguenza
l’architettura convergente, subisce a partire dall’esistenza di tre reti di accesso del
tutto separate al raggiungimento di una rete completamente integrata, tenendo in
considerazione anche possibili sviluppi futuri [§ 4.6.1]. Terminando, sono indicati i
vantaggi che questa soluzione apporta e le sue caratteristiche [§ 4.6.2].
Infine, viene presentato come l’architettura convergente risultante può essere
implementata [Appendice A]. Per poter fare ciò viene preso come esempio un case
pilot di possibile servizio convergente, chiamato m-Advertising game [§ A.1]. È stato
scelto questo tipo di servizio perché mette in evidenza il ruolo svolto da uno degli
elementi della Edge Network, il Content Provider server, che è appunto un
dispositivo cruciale nell’implementazione della Edge Network [§ A.2]. Questo
componente della Edge Network ha il compito di connettere il sistema di
trasmissione broadcast DVB-H alla Core Network IMS, quindi oltre le funzionalità
che tale server già possiede deve essere in grado di svolgere funzioni aggiuntive. Una
di queste è l’instaurazione di sessione, la quale è messa in evidenza nel caso preso in
considerazione. Perciò, per simulare un ambiente IMS, in cui il Content Provider
server si trova, ho usato un SIP proxy che gestisce la funzione di presence. Si mostra
in questo modo come il Content Provider server può instaurare una sessione HTTP
con gli utenti [§ A.3].
Uno studio preliminare del mercato delle Telecomunicazioni è stato necessario
per poter capire verso cosa sono rivolte le continue evoluzioni nel mondo delle
telecomunicazioni e quali sono le possibilità che le innovazioni tecnologiche offrono,
quindi è presentato un overwiev sullo scenario internazionale ed italiano [Appendice
B].
Technical Overview
9
CCHHAAPPTTEERR OONNEE
1. TECHNICAL OVERVIEW
The objective of this thesis is a convergent architecture between different access
technologies. In order to achieve this convergence the access networks considered
are UMTS (Release 6), Mobile WiMAX and DVB-H.
These three technologies were chosen because of different reasons. First, they are
standard solutions. Standardization is very important when interworking is necessary,
especially in this case. ETSI (European Telecommunication Standards Institute) has
chosen DVB-H as European standard for the broadcasting mobile TV. UMTS is
standardized by 3GPP (3rd Generation Partnership Project) which is a co-operation
between ETSI in Europe and other Associations worldwide. WiMAX standard is
IEEE 802.16. This is the 16th working group of IEEE (Institute of Electrical and
Electronic Engineers) 802, which is specialized in wireless broadband access. The
IEEE 802.16 standard specifics just the air interface (physical and MAC layer), the
last 802.16e one allows this technology to be used by mobile and nomadic clients.
Furthermore, they are the most common, especially in Europe where UMTS is
already spread, DVB-H is now coming and WiMAX will soon be realized. Moreover
there is a wide industry support for them.
In addition, they cover the whole service spectrum: DVB-H is the one that provides
the broadcasting transmission to mobile terminals, UMTS is the mobile
telecommunication system and it also provides a wide range of services, at last
WiMAX gives the broadband wireless connection, mobile, portable and fixed.
A Convergent Access Network can be done with any kind of access technology but
in this thesis DVB-H, UMTS and WiMAX are taken into account because they are
the most efficient technologies in comparison with their competitors. For example,
DVB-H provides a spectral efficiency higher then MediaFLO5.
In Figure 1 the main property of each technology is summarized. DVB-H is based on
the DVB-T transmission but it provides mobility with high data rate in addition,
5 MediaFLO is one of the main Broadcasting systems
Technical Overview
10
thanks to its new features that allow battery saving, increased general robustness and
support for seamless handover. It is also important to underline that DVB-H is IP-
Based.
UMTS is very flexible, thanks to the CDMA6 it supplies a negotiable bit rate, delay
and BER and it offers different QoS parameters. UMTS’s maximum achievable bit
rate is 2 Mbps, very improved compared to other mobile systems like GSM/GPRS.
WiMAX is the broadband wireless access technology that can accomplish very high
data rates, up to 10 Mbps. It is able to allocate different QoS flow classes and it is
very scalable.
In this chapter these standardized technologies are introduced just to present them in
general in order to understand what their features are. In the successive chapter, they
are analyzed to see how convergence can be achieved.
Figure 1: Technology Features
6 CDMA (Code Division Multiple Access) is a method of multiple access that does not divide up the channel by time (as in TDMA), or frequency (as in FDMA), but instead encodes data with a special code associated with each channel and uses the constructive interference properties of the special codes to perform the multiplexing.
Technical Overview
11
1.1 UMTS Release 6 (Universal Mobile Telecommunication System)
1.1.1 Introduction
3G Systems are intended to provide a global mobility with wide range of services
including telephony, paging, messaging, Internet and broadband data. International
Telecommunication Union (ITU) started the process of defining the standard for
third generation systems, referred to as International Mobile Telecommunications
2000 (IMT-2000). In Europe European Telecommunications Standards Institute
(ETSI) was responsible of UMTS standardization process. In 1998 Third Generation
Partnership Project (3GPP) was formed to continue the technical specification work.
The first UMTS Release is “R99” that was finalized in 2000, the subsequently
Releases were numbered Rel4, Rel5 and Rel6. Release 99 defines the Core Network
evolution starting from the GSM/GPRS system one, in the Release 4 control and
transport planes are separated in the Core Network. Both Release 5 and 6 change the
architecture by introducing the IP Multimedia Subsystem which provides all IP-
Based services, including voice.
UMTS Release 6 takes a radical approach to the introduction of conversational and
real time interactive multimedia services over an end-to-end IP transport provided by
an enhanced general packet radio service in the packet switched domain.
1.1.2 UMTS Services and Features
UMTS offers teleservices (like speech or SMS) and bearer services, which provide
the capability for information transfer between access points. It is possible to
negotiate and renegotiate the characteristics of a bearer service at session or
connection establishment and during ongoing session or connection. Both
connection-oriented and connectionless services are offered for Point-to-Point and
Point-to-Multipoint communication.
Bearer services have different QoS parameters for maximum transfer delay, delay
variation and bit error rate.
Technical Overview
12
The main UMTS feature that distinguishes UMTS from GSM/GPRS is the maximum
achievable bit rate. Offered data rate targets depend on the coverage area and they
are:
• 144 kbits/s satellite and rural outdoor
• 384 kbits/s urban outdoor
• 2048 kbits/s indoor and low range outdoor
UMTS network services have different QoS classes for four types of traffic:
• Conversational class (voice, video telephony, video gaming)
• Streaming class (multimedia, video on demand, webcast)
• Interactive class (web browsing, network gaming, database access)
• Background class (email, SMS, downloading)
UMTS will also have a Virtual Home Environment (VHE). It is a concept for
personal service environment portability across network boundaries and between
terminals. Personal service environment means that users are consistently presented
with the same personalized features, User Interface customization and services in
whatever network or terminal, wherever the user may be located. UMTS also has
improved network security and location based services.
In order to provide such services UMTS has to be very flexible.
UMTS Release 6 introduces new features and services thanks to the IMS Core
Network. It is allowed network sharing (i.e. multiple radio access networks sharing
common core network) and WLAN interworking (use WLAN as access network for
IMS instead of PS domain). New services are:
• MBMS (Multimedia Broadcast and Multicast Service): downstream
broadcasting and multicast support enable resource and cost efficient data
transfer to many users in parallel
• Push Service: pushing of information from network to UE
• IMS Group Management: supporting service for other services
• IMS Presence Services: user can find out presence of others and is visibility
is defined to others
• IMS Messaging: Instant Messaging interworks with Presence Service
Technical Overview
13
1.1.3 UMTS Architecture
A UMTS network consist of three interacting domain:
• Core Network (CN)
• UMTS Terrestrial Radio Access Network (UTRAN)
• User Equipment (UE)
The main function of the Core Network is to provide switching, routing and transit
for user traffic. Core Network also contains the databases and network management
functions.
The basic Core Network architecture for UMTS is based on GSM network with
GPRS. All equipment has to be modified for UMTS operation and services.
The UTRAN provides the air interface access method for User Equipment. Base
Station is referred as Node-B and control equipment for Node-B's is called Radio
Network Controller (RNC).
It is necessary for a network to know the approximate location in order to be able to
page user equipment. Here is the list of system areas from largest to smallest:
• UMTS systems (including satellite)
• Public Land Mobile Network (PLMN)
• MSC/VLR or SGSN
• Location Area
• Routing Area (PS domain)
• UTRAN Registration Area (PS domain)
• Cell
• Sub cell
In particular, the architecture of UMTS Release 6 includes the UMTS Terrestrial
Access Network (UTRAN), the GPRS domain and the Internet protocol Multimedia
Subsystem Core Network (IMS CN). This architecture is shown in Figure 2.
Technical Overview
14
Figure 2: UMTS Release 6 Architecture
1.1.3.1 Core Network
The Core Network in the earlier releases (Rel99 and Rel4) is divided in circuit
switched and packet switched domains. Some of the circuit switched elements are
Mobile services Switching Centre (MSC), Visitor location register (VLR) and
Gateway MSC. Packet switched elements are Serving GPRS Support Node (SGSN)
and Gateway GPRS Support Node (GGSN). Some network elements, like EIR, HLR,
VLR and AUC are shared by both domains. The Asynchronous Transfer Mode
(ATM) is defined for UMTS core transmission. ATM Adaptation Layer type 2
(AAL2) handles circuit switched connection and packet connection protocol AAL5
is designed for data delivery. The architecture of the Core Network may change
when new services and features are introduced. Number Portability DataBase
(NPDB) will be used to enable user to change the network while keeping their old
phone number. Gateway Location Register (GLR) may be used to optimize the
subscriber handling between network boundaries. MSC, VLR and SGSN can merge
to become a UMTS MSC.
1.1.3.1.1 CS Domain
The CS domain refers to the set of all the CN entities offering "CS type of
connection" for user traffic as well as all the entities supporting the related signaling.
Technical Overview
15
A "CS type of connection" is a connection for which dedicated network resources are
allocated at the connection establishment and released at the connection release.
The entities specific to the CS domain are: MSC, GMSC, VLR. All the other CN
entities not defined as PS domain specific entities are common to the CS and to the
PS domains.
1.1.3.1.2 PS Domain
The PS domain refers to the set of all the CN entities offering "PS type of
connection" for user traffic as well as all the entities supporting the related signaling.
A "PS type of connection" transports the user information using autonomous
concatenation of bits called packets: each packet can be routed independently from
the previous one.
The entities specific to the PS domain are the GPRS specific entities, i.e. SGSN and
GGSN. [§ 4.3]
1.1.3.1.3 IP Multimedia subsystem (IMS)
The IM subsystem comprises all CN elements for provision of IP multimedia
services comprising audio, video, text, chat, etc. and a combination of them delivered
over the PS domain. The entities related to IMS are CSCF, MGCF, MRF, etc.
[Chapter 3].
1.1.3.2 UTRAN
The Universal Terrestrial Radio Access Network does not change in the different
releases. It is a set of RNS which are composed of RNC and Node B. Every element
is linked to another by using an ATM transport protocol.
1.1.3.2.1 Radio Access
Wideband CDMA7 technology was selected to for UTRAN air interface.
UMTS WCDMA is a Direct Sequence CDMA system where user data is multiplied
with pseudo-random bits derived from WCDMA Spreading codes. In UMTS, in
addition to channelization, Codes are used for synchronization and scrambling.
7 W-CDMA is a wideband spread-spectrum 3G mobile telecommunication air interface that utilizes code division multiple access (or CDMA the general multiplexing scheme, not to be confused with CDMA the standard).
Technical Overview
16
WCDMA has two basic modes of operation: Frequency Division Duplex (FDD) and
Time Division Duplex (TDD).
1.1.3.2.2 WCDMA channels
Due to the wide range of information that the UMTS is able to transport, the protocol
architecture of the UTRA FDD is layered in many levels. Because of this, there are
different channels corresponding the different layers of the protocol stack:
• Physical channels
• Transport channels
• Logical channels
Physical channels connect the UE and the Node B, logical and transport channels
transfer information from the RNC to the UE and vice versa.
UE Node B RNC
Figure 3: Physical, Logical and Transport Channels
In Figure 4 are shown the channels involved in the air interface.
Technical Overview
17
Figure 4: Radio Interface Protocol Architecture
1.1.3.3 User Equipment
The UMTS standard does not restrict the functionality of the User Equipment in any
way. Terminals work as an air interface counter part for Node-B and have many
different types of identities.
Most of these UMTS identity types are taken directly from GSM specifications:
• International Mobile Subscriber Identity (IMSI)
• Temporary Mobile Subscriber Identity (TMSI)
• Packet Temporary Mobile Subscriber Identity (P-TMSI)
• Temporary Logical Link Identity (TLLI)
• Mobile station ISDN (MSISDN)
• International Mobile Station Equipment Identity (IMEI)
• International Mobile Station Equipment Identity and Software Number
(IMEISV)
UMTS mobile station can operate in one of three modes of operation:
• PS/CS mode of operation: The MS is attached to both the PS domain and CS
domain, and the MS is capable of simultaneously operating PS services and
CS services.
Technical Overview
18
• PS mode of operation: The MS is attached to the PS domain only and may
only operate services of the PS domain. However, this does not prevent CS-
like services to be offered over the PS domain (like VoIP).
• CS mode of operation: The MS is attached to the CS domain only and may
only operate services of the CS domain.
UMTS IC card has same physical characteristics as GSM SIM card. It has several
functions:
• Support of one User Service Identity Module (USIM) application (optionally
more that one)
• Support of one or more user profile on the USIM
• Update USIM specific information over the air
• Security functions
• User authentication
• Optional inclusion of payment methods
• Optional secure downloading of new applications
1.1.4 General Protocol Architecture
In Figure 5 a simplified UMTS Architecture with the external reference points and
interfaces to the UTRAN is shown.
Figure 5: UMTS Architecture
Two interfaces are illustrated: Uu and Iu interfaces.
Technical Overview
19
The protocols over these interfaces are divided into two structures:
• User plane protocols: These are the protocols implementing the actual radio
access bearer service, i.e. carrying user data through the access stratum. The
radio access bearer service is offered from SAP (Service Access Point) to
SAP by the Access Stratum.
• Control plane protocols: These are the protocols for controlling the radio
access bearers and the connection between the UE and the network from
different aspects (including requesting the service, controlling different
transmission resources, handover & streamlining etc.). Also a mechanism for
transparent transfer of NAS messages is included.
1.2 Mobile WiMAX (Wireless Interoperability for Microwave
Access) [802.16e]
1.2.1 Introduction
In December 2005 the IEEE ratified the 802.16e amendment to the 802.16 standard
with the target of providing mobility to WiMAX (Wireless Interoperability for
Microwave Access) technology. WiMAX is a broadband wireless solution that will
enable the convergence of mobile and fixed broadband network. In order to
distinguish the new standard version from the old one, the IEEE 802.16-2004 is
called as “Fixed WiMAX” and the IEEE 802.16e as “Mobile WiMAX”. The Mobile
WiMAX Air Interface uses OFDMA (Orthogonal Frequency Division Multiple
Access) to improve the efficiency in a NLOS (No Line Of Sight) multipath context.
SOFDMA (Scalable OFDMA) is introduced to support scalable channel bandwidths
from 1.25 to 20 MHz. The IEEE 802.16 standard addresses the air interface
specification, to define end-to-end system solutions for a Mobile WiMAX network
the WiMAX forum Network Working Group (NWG) is developing a higher-level
networking specifications for Mobile WiMAX system solution.
Technical Overview
20
Some of the characteristics provides by the Mobile WiMAX are:
• High Data Rates: Mobile WiMAX in a 10 MHz channel will support peak
downlink (DL) data rates up to 63 Mbps per sector and peak uplink (UL) data
rates up to 28 Mbps per sector. This performance can be achieved thanks to:
- MIMO antenna technology
- Flexible sub-channelization schemes
- Advanced Coding and Modulation
• Quality of Service: IEEE 802.16 MAC architecture defining Service Flow,
which can map to DiffServ code points or MPLS (Multiprotocol Label
Switching) flow labels, enables end-to-end IP based QoS (Quality of
service).
• Scalability: Mobile WiMAX technology is thought to be scalable in order to
work in different channelizations from 1.25 to 20 MHz so it is conformable to
the different worldwide spectrum resource allocations and allows countries
with different economies to realize the configuration shaped to their specific
geographic needs such as providing affordable internet access in rural setting
versus high capacity in metro and suburban areas.
• Security: To provide security Mobile WiMAX to exploit:
- EAP-based authentication
- AES-CCM-based authenticated encryption
- CMAC and HMAC based control message protection schemes
Moreover it supports a diverse set of user credentials such as the following:
- SIM/USIM cards
- Smart Cards
- Digital Certificates
- Username/Password schemes
• Mobility: To provide real-time communications, such as VoIP, Mobile
WiMAX adopts optimized handover schemes (latencies less than 50
milliseconds). Security is maintained during handover thanks to flexible key
management.
Technical Overview
21
1.2.2 Features
1.2.2.1 Physical Layer
1.2.2.1.1 OFDMA
Orthogonal Frequency Division Multiple Access (OFDMA) is a multiple-access
multiplexing scheme for OFDM systems. It works by assigning a subset of
subcarriers to individual users. It provides multiplexing operation of data streams
from multiple users onto the downlink sub-channels and uplink multiple accesses by
means of uplink sub-channel.
OFDMA features are summarized in the following:
• OFDMA is the “multi-user” version of OFDM
• Functions essentially as OFDM-FDMA
• Each OFDMA user transmits symbols using subcarriers that remain
orthogonal to those of other users
• More than one subcarrier can be assigned to one user to support high rate
applications
• Allows simultaneous transmission from several users and so better spectral
efficiency
As the Figure 6 shows, the OFDMA symbol structure consists of three types of sub-
curriers:
• Data sub-carriers for data transmission
• Pilot sub-carriers for estimation and synchronization purposes
• Null sub-carriers for no transmission, used for guard bands and DC carriers
Technical Overview
22
Figure 6: OFDMA Sub-Carriers Structure
A sub-channel is a subset of Active (data end pilot) sub-carriers. Sub-channelization
is made both in DL and UL. The minimum frequency-time resource unit of sub-
channelization is one slot, which is equal to 48 data tones (sub-carriers).
1.2.2.1.2 Scalable OFDMA
The key difference between the Fixed and Mobile WiMAX standards is the more
efficient S-OFDMA modulation scheme. S-OFDMA (Scalable OFDM Access) can
assign a subset of sub-carriers to individual users. By using different sub-carriers
multiple people can connect at the same time on the same frequency without
interference. The number of sub-carriers can adjust dynamically for different
conditions. Unlike many other OFDM-based systems such as WLAN, the 802.16
standard supports variable bandwidth sizes, this wide range of bandwidths is thought
to flexibly address the need for various spectrum allocation and usage model
requirements. A scalable physical layer enables standard-based solutions to deliver
optimum performance in channel bandwidths ranging from 1.25 MHz to 20 MHz
with fixed sub-carrier spacing. The S-OFDMA’s architecture is based on a scalable
sub-channelization structure with variable Fast Fourier Transform (FFT) sizes
according to the channel bandwidth. The scalability is supported by adjusting the
FFT size while fixing the sub-carrier frequency spacing at 10.94 kHz, this is made to
meet the optimal operation point balancing protection against multipath and Doppler
shift. Since the resource unit sub-carrier bandwidth and symbol duration is fixed, the
impact to higher layers is minimal when scaling the bandwidth. The SOFDMA
parameters are listed in Table 1.
Technical Overview
23
Table 1: OFDMA Scalability Parameters
1.2.2.1.3 TDD Frame Structure
The 802.16e PYH supports TDD, FDD, and Half-Duplex FDD operation but the
initial release of Mobile certification profile will only include TDD. Some of the
reasons for preferring TDD are the following:
• It has a strong advantage in the case where the asymmetry of the uplink and
downlink data is variable. As the amount of uplink data increases, more
bandwidth can be allocated to that and as it shrinks it can be taken away so it
enables adjustments of the downlink/uplink ratio while with FDD both
downlink and uplink always have a fixed and generically equal DL and UL.
• TDD assures channel reciprocity for better support of link adaptation, MIMO
and other closed loop advanced antenna technologies.
• Unlike FDD, which requires a pair of channels, TDD only requires a single
channel for both downlink and uplink providing greater flexibility for
adaptation to varied global spectrum allocations.
• Transceiver designs for TDD implementations are less complex and therefore
less expensive.
The frames are divided into DL and UL sub-frames. To prevent DL and UL
transmission collision the sub-frames are separated by Transmit/Receive and
Receive/Transmit Gaps. The Figure 7 shows the OFDM frame structure for TDD
implementation.
Technical Overview
24
The fields are:
• Preamble: The preamble, used for synchronization, is the first OFDM symbol
of the frame.
• Frame Control Head (FCH): The FCH follows the preamble. It provides the
frame configuration information such as MAP message length and coding
scheme and usable sub-channels.
• DL-MAP and UL-MAP: The DL-MAP and UL-MAP provide sub-channel
allocation and other control information for the DL and UL sub-frames
respectively.
• UL Ranging: The UL ranging sub-channel is allocated for mobile stations
(MS) to perform closed-loop time, frequency, and power adjustment as well
as bandwidth requests.
• UL CQICH: The UL CQICH channel is allocated for the MS to feedback
channel state information.
• UL ACK: The UL ACK is allocated for the MS to feedback DL HARQ
acknowledgement.
Figure 7: OFDMA Frame Structure
Technical Overview
25
1.2.2.1.4 Other Advanced PHY Layer Features
To best meet the coverage and capacity in mobile applications the 802.16e introduces
the following features:
• Adaptive modulation and coding (AMC): The core idea of AMC is to
dynamically change the Modulation and Coding Scheme (MCS) in
subsequent frames with the objective of adapting the overall spectral
efficiency of the channel condition. The decision concerning selecting the
appropriate MCS is performed at the receiver side according to the observed
channel condition. Then the information is fed back to the transmitter in each
frame.
• Hybrid Automatic Request (HARQ): it is a variation of the ARQ error control
method. When the coded data block is received, the receiver first decodes the
error-correction code. If the channel quality is good enough, all transmission
errors should be ok, and the receiver can obtain the correct data block. If the
channel quality is bad and some transmission errors can not be corrected, the
receiver will detect this situation using the error-correction code and then the
received coded data block is discarded and a retransmission is requested by
the receiver, similar to ARQ.
• Fast Channel Feedback (CQICH).
Table 2: Code and Modulation options
As can be seen in Table 2, WiMAX supports different digital modulation schemes, in
the DL QPSK, 16QAM and 64QAM are mandatory while in the UL 64QAM is
optional. It can support Convolutional Code (CC) and Convolutional Turbo Code
(CTC) with variable code rate and repetition coding. Optionally, it can support Block
Turbo Code and Low Density Parity Check Code (LDPC).
Combining the various modulation and code rates WiMAX can provide a variety of
data rates that are adaptable to different needs.
Technical Overview
26
1.2.2.2 MAC Layer
The WiMAX standard wants to reach the goal of providing a set of heterogeneous
broadband services like voice, data and video. To achieve this target it may be able to
supporting simultaneously on the same channel bursty data traffic with high peak
rate demand, streaming video and latency sensitive voice traffic. A MAC scheduler
has the capability of changing dynamically the throughput by varying the resources
allocated to one terminal from a single time slot to an entire frame. The allocation
information is transmitted by the MAP messages at the beginning of each frame. In
this way, the scheduler can adapt the resource allocation frame-by-frame.
1.2.2.2.1 Quality of Service
To provide the different types of services, Mobile WiMAX needs to support different
QoS requirements. Some of the characteristics that enable it to achieve this
peculiarity are:
• Fast air link
• Symmetric UL/DL capacity
• Fine resource granularity
• Flexible resource allocation mechanism
To manage the QoS the Mobile WiMAX MAC layer uses a service flow that is a
unidirectional flow of packets that contain the particular set of QoS parameters.
First of all the base station and the user terminal establish a unidirectional logic link
called a connection between the peer MACs. The outbound MAC then associates
packets crossing the MAC interface into a service flow to be delivered over the
connection. The QoS parameters associated with the service flow define the
transmission ordering scheduling on the air interface.
The service flow parameters can be dynamically managed through MAC messages to
accommodate the dynamic service demand. The service flow-based QoS mechanism
applies to both DL and UL and provides improved QoS in both directions. Mobile
WiMAX supports a wide range of data services and applications with varied QoS
requirements. These are summarized in Table 3.
Technical Overview
27
Table 3: Quality of service options
1.2.2.2.2 MAC Scheduling Service
The goal of scheduling and link adaptation is to provide the desired QoS treatment to
the traffic traversing the airlink, while optimally utilizing the resources of the airlink.
Some of the most important scheduler’s characteristics are the following:
• It is located at each base station to enable rapid response to traffic
requirements and channel condition.
• It can correctly determine the packet’s transmission ordering, thanks to the
data packets association of the service flow, with well defined QoS
parameters in the MAC layer.
• It can choose the appropriate coding and modulation for each allocation
taking advantage of the CQICH channel fast feedback.
The scheduling is provided for both DL and UL traffic. In order for the MAC
scheduler to make an efficient resource allocation and provide the desired QoS in the
UL, the UL must feedback accurate and timely information reaching the traffic
condition and QoS requirements. The UL service flow defines the feedback
mechanism for each uplink connection. The MAC scheduler manages the data
Technical Overview
28
transport on a connection-by-connection basis. Each connection is associated with a
single data service with a set of QoS parameters that quantify the aspects of its
behavior. With the ability to dynamically allocate resources in both DL and UL, the
scheduler can provide superior QoS for both DL and UL traffic. Particularly with
uplink scheduling, the uplink resources are more efficiently allocated, performance is
more predictable, and QoS is better enforced.
1.2.2.2.3 Mobility Management
To enable power-efficient MS operation and save battery Mobile WiMAX works in
two way of working:
• Sleep Mode: When working in Sleep Mode state the MS conducts pre-
negotiated periods of absence from the Serving Base Station air interface.
• Idle Mode: When working in Idle Mode state, the MS is periodically
available for DL broadcast traffic messaging without registering at a specific
base station.
To manage the handoff Mobile WiMAX supports three different methods:
• Hard Handoff (HHO): It is the only mandatory. With hard handoff, the link
to the prior base station is terminated before or as the user is transferred to the
new cell’s base station. That is to say that the mobile is linked to no more
than one base station at a given time. Initiation of the handoff may begin
when the signal strength received on the mobile from next base station is
greater than that of the prior base station.
• Fast Base Station Switching (FBSS): With FBSS the MS and the BS have a
list called Active set in which they maintain the BSs involved in FBSS with
the MS. The MS continuously monitors the BS in the Active Set among
which defines the Anchor BS. MS only communicates with the Anchor BS
for uplink and downlink messages including management and traffic
connections. The Anchor BS switching is performed without of explicit HO
signaling messages, only communicating the BS signal strength via the
Channel Quality Indicator (CQI). A FBSS handover begins with a decision
by the BS to receive or transmit data from the Anchor BS that may change
Technical Overview
29
within the Active set. The MS scans the neighbor BSs and selects those that
are suitable to be included in the active set. The MS reports the selected BSs
and the active set update procedure is performed by the BS and MS. The MS
continuously monitors the signal strength of the BSs that are in the active set
and selects one BS from the set to be the Anchor BS. The MS reports the
selected Anchor BS on CQICH or MS initiated HO request message. An
important requirement of FBSS is that the data is simultaneously transmitted
to all members of an active set of BSs that are able to serve the MS.
• Macro Diversity Handover (MDHO): Also in this case the BS has a list called
Active set in which they maintain the BSs involved in MDHO with the MS.
The communications either in uplink or in downlink. A MDHO begins when
a MS decides to transmit or receive unicast messages and traffic from
multiple BSs at the same time. For downlink MDHO, two or more BSs
provide synchronized transmission of MS downlink data such that diversity
combining is performed at the MS. For uplink MDHO, the transmission from
a MS is received by multiple BSs where selection diversity of the information
received is performed. Selection diversity is the simplest diversity approach.
Using multiple antennas with overlapping coverage, this approach selects the
antenna with the highest received signal power, mitigating fading.
The sleep Mode provides battery economizing and seamless handoff enables
switching from one base station to another without interrupting the connection. In
this way it tries to solve two of the most important barriers to the mobility.
1.2.2.2.4 Security
Mobile WiMAX supports best in class security features by adopting the best
technologies available today. Support exists for mutual device/user authentication,
flexible key management protocol, strong traffic encryption, control and
management plane message protection and security protocol optimization for fast
handovers. The usage aspects of the security features are:
• Key Management Protocol: Privacy and Key Management Protocol Version 2
(PKMv2) is the basis of Mobile WiMAX security as defined in 802.16e. This
protocol manages the MAC security using PKM-REQ/RSP messages. PKM
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EAP authentication, Traffic Encryption Control, Handover Key Exchange
and Multicast/Broadcast security messages all are based on this protocol.
• Device/User Authentication: Mobile WiMAX supports Device and User
Authentication using IETF EAP protocol by providing support for credentials
that are SIM-based, USIM-based or Digital Certificate or
UserName/Password-based. Corresponding EAP-SIM, EAP-AKA, EAP-TLS
or EAP-MSCHAPv2 authentication methods are supported through the EAP
protocol. Key deriving methods are the only EAP methods supported.
• Traffic Encryption: AES-CCM is the cipher used for protecting all the user
data over the Mobile WiMAX MAC interface. The keys used for driving the
cipher are generated from the EAP authentication. A Traffic Encryption State
machine that has a periodic key (TEK) refresh mechanism enables sustained
transition of keys to further improve protection.
• Control Message Protection: Control data is protected using AES based
CMAC, or MD5-based HMAC schemes.
• Fast Handover Support: A 3-way Handshake scheme is supported by Mobile
WiMAX to optimize the re-authentication mechanisms for supporting fast
handovers. This mechanism is also useful to prevent any man-in-the-middle-
attacks.
1.2.2.3 Advanced Features of Mobile WiMAX
1.2.2.3.1 Smart Antenna Technologies
Smart antenna technologies typically involve complex vector or matrix operations on
signals due to multiple antennas. OFDMA allows smart antenna operations to be
performed on vector-flat sub-carriers. Complex equalizers are not required to
compensate for frequency selective fading. OFDMA therefore, is very well-suited to
support smart antenna technologies.
Mobile WiMAX supports a full range of smart antenna technologies to enhance
system performance. The smart antenna technologies supported include:
• Beamforming: With beamforming (a signal processing technique used with
arrays of transmitters or receivers that controls the directionality of, or
sensitivity to, a radiation pattern) the system uses multiple-antennas to
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transmit weighted signals to improve coverage and capacity of the system and
reduce outage probability.
• Space-Time Code (STC): a method employed to improve the reliability of
data transmission in wireless communication systems using multiple transmit
antennas. STCs rely on transmitting multiple, redundant copies of a data
stream to the receiver in the hope that at least some of them may survive the
physical path between transmission and reception in a good enough state to
allow reliable decoding. Transmit diversity such as Alamouti code is
supported to provide spatial diversity and reduce fade margin.
• Spatial Multiplexing (SM): Spatial multiplexing is supported to take
advantage of higher peak rates and increased throughput. With spatial
multiplexing, multiple streams are transmitted over multiple antennas. If the
receiver also has multiple antennas, it can separate the different streams to
achieve higher throughput compared to single antenna systems. In UL, each
user has only one transmit antenna, two users can transmit collaboratively in
the same slot as if two streams are spatially multiplexed from two antennas of
the same user. This is called UL collaborative SM.
The supported features in the Mobile WiMAX performance profile are listed in
Table 4.
Table 4: Smart Antenna Options
Mobile WiMAX supports adaptive switching between these options to maximize the
benefit of smart antenna technologies under different channel conditions.
1.2.2.3.2 Fractional Frequency Reuse
Mobile WiMAX supports frequency reuse of one, i.e. all cells/sectors operate on the
same frequency channel to maximize spectral efficiency. However, due to heavy
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Cochannel Interference (CCI) in frequency reuse one deployment, users at the cell
edge may suffer degradation in connection quality. With Mobile WiMAX, users
operate on subchannels, which only occupy a small fraction of the whole channel
bandwidth; the cell edge interference problem can be easily addressed by
appropriately configuring subchannel usage without applying the traditional
frequency planning. In Mobile WiMAX, the flexible sub-channel reuse is facilitated
by sub-channel segmentation and permutation zone. A segment is a subdivision of
the available OFDMA sub-channels (one segment may include all sub-channels).
One segment is used for deploying a single instance of MAC. Permutation Zone is a
number of contiguous OFDMA symbols in DL or UL that use the same permutation.
The sub-channel reuse pattern can be configured so that users close to the base
station operate on the zone with all sub-channels available. While for the edge users,
each cell or sector operates on the zone with a fraction of all sub-channels available.
In Figure 8, F1, F2, and F3 represent different sets of sub-channels in the same
frequency channel. With this configuration, the full load frequency reuse one is
maintained for center users to maximize spectral efficiency and fractional frequency
reuse is implemented for edge users to assure edge-user connection quality and
throughput. The sub-channel reuse planning can be dynamically optimized across
sectors or cells based on network load and interference conditions on a frame by
frame basis. All the cells and sectors therefore, can operate on the same frequency
channel without the need for frequency planning.
Figure 8: Fractional frequency Reuse
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1.2.2.3.3 Multicast and Broadcast Service (MBS)
Multicast and Broadcast Service (MBS) supported by Mobile WiMAX combines the
best features of DVB-H, MediaFLO and 3GPP E-UTRA and satisfies the following
requirements:
• High data rate and coverage using a Single Frequency Network (SFN)
• Flexible allocation of radio resources
• Low MS power consumption
• Support of data-casting in addition to audio and video streams
• Low channel switching time
The Mobile WiMAX Release 1 profile defines a toolbox for initial MBS service
delivery. The MBS service can be supported by either constructing a separate MBS
zone in the DL frame along with unicast service (embedded MBS) or the whole
frame can be dedicated to MBS (DL only) for standalone broadcast service. Figure 9
shows the DL/UL zone construction when a mix of unicast and broadcast service are
supported. The MBS zone supports multi-BS MBS mode using Single Frequency
Network (SFN) operation and flexible duration of MBS zones permits scalable
assignment of radio resources to MBS traffic. It may be noted that multiple MBS
zones are also feasible. There is one MBS zone MAP IE descriptor per MBS zone.
The MS accesses the DL MAP to initially identify MBS zones and locations of the
associated MBS MAPs in each zone. The MS can then subsequently read the MBS
MAPs without reference to DL MAP unless synchronization to MBS MAP is lost.
The MBS MAP IE specifies MBS zone PHY configuration and defines the location
of each MBS zone via the OFDMA Symbol Offset parameter. The MBS MAP is
located at the 1st sub-channel of the 1st OFDM symbol of the associated MBS zone.
The multi-BS MBS does not require the MS be registered to any base station. MBS
can be accessed when MS in Idle mode to allow low MS power consumption. The
flexibility of Mobile WiMAX to support integrated MBS and unicast services
enables a broader range of applications.
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1.2.3 End-to-End WiMAX Architecture
The IEEE only defined the Physical (PHY) and Media Access Control (MAC) layers
in 802.16. This approach has worked well for technologies such as Ethernet and
WiFi, which rely on other bodies such as the IETF (Internet Engineering Task Force)
to set the standards for higher layer protocols such as TCP/IP, SIP, VoIP and IPSec.
In the mobile wireless world, standards bodies such as 3GPP and 3GPP2 set
standards over a wide range of interfaces and protocols because they require not only
airlink interoperability, but also inter-vendor inter-network interoperability for
roaming, multi-vendor access networks, and inter-company billing. Vendors and
operators have recognized this issue, and have formed additional working groups to
develop standard network reference models for open inter-network interfaces. Two
of these are the WiMAX Forum’s Network Working Group, which is focused on
creating higher-level networking specifications for fixed, nomadic, portable and
mobile WiMAX systems beyond what is defined in the IEEE 802.16 standard, and
Service Provider Working Group which helps write requirements and prioritizes
them to help drive the work of Network WG.
Before delving into some of the details of the architecture, a few basic principles that
have guided the WiMAX architecture development are first noted:
1. The architecture is based on a packet-switched framework, including native
procedures based on the IEEE 802.16 standard and its amendments, appropriate
IETF RFCs and Ethernet standards.
2. The architecture permits decoupling of access architecture (and supported
topologies) from connectivity IP service. Network elements of the connectivity
system are agnostic to the IEEE 802.16 radio specifics.
3. The architecture allows modularity and flexibility to accommodate a broad range
of deployment options such as:
• Small-scale to large-scale (sparse to dense radio coverage and capacity)
WiMAX networks
• Urban, suburban, and rural radio propagation environments
• Licensed and/or licensed-exempt frequency bands
• Hierarchical, flat, or mesh topologies, and their variants
• Co-existence of fixed, nomadic, portable and mobile usage models.
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WiMAX Forum industry participants have identified a WiMAX Network Reference
Model (NRM) that is a logical representation of the network architecture. The NRM
identifies functional entities and reference points over which interoperability is
achieved between functional entities. The architecture has been developed with the
objective of providing unified support of functionality needed in a range of network
deployment models and usage scenarios (ranging from fixed – nomadic – portable –
simple mobility – to fully mobile subscribers).
Figure 10 illustrates the NRM, consisting of the following logical entities: MS, ASN,
and CSN and clearly identified reference points for interconnection of the logical
entities. The figure shows the key normative reference points R1-R5.
The intent of the NRM is to allow multiple implementation options for a given
functional entity, and yet achieve interoperability among different realizations of
functional entities. Interoperability is based on the definition of communication
protocols and data plane treatment between functional entities to achieve an overall
end-to-end function, for example, security or mobility management. Thus, the
functional entities on either side of a reference point represent a collection of control
and bearer plane end-points.
Figure 9: NRM Network Reference Model
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The Network Access Provider (NAP) is a business entity that provides WiMAX
radio access infrastructure to one or more WiMAX Network Service Providers
(NSPs). A NAP implements this infrastructure using one or more Access Service
Networks (ASN). The ASN defines a logical boundary and represents a convenient
way to describe aggregation of functional entities and corresponding message flows
associated with the access services. The ASN represents a boundary for functional
interoperability with WiMAX clients, WiMAX connectivity service functions and
aggregation of functions embodied by different vendors. Mapping of functional
entities to logical entities within ASNs as described in the NRM may be performed
in different ways.
The Network Service Provider (NSP) is business entity that provides IP
connectivity and WiMAX services to WiMAX subscribers consistent with the
Service Level Agreement it establishes with WiMAX subscribers. To provide these
services, an NSP establishes contractual agreements with one or more NAPs. An
NSP may also establish roaming agreements with other NSPs and contractual
agreements with third-part application providers (e.g. ASP or ISPs) for providing
WiMAX services to subscribers.
The ASP (Application Service Provider) provides value added services, Layer 3+
(e.g. IMS, corporate access, etc...) furthermore it provides and manages applications
on top of IP.
Connectivity Service Network (CSN) is defined as a set of network functions that
provide IP connectivity services to the WiMAX subscriber(s). A CSN may comprise
network elements such as routers, AAA proxy/servers, user databases and
Interworking gateway devices. A CSN may be deployed as part of a Greenfield
WiMAX Network Service Provider (NSP) or as part of an incumbent WiMAX NSP.
The WiMAX Forum is in the process of network specifications in a manner that
would allow a variety of vendor implementations that are interoperable and suited for
a wide diversity of deployment requirements.
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1.3 DVB-H (Digital Video Broadcasting-Handheld)
1.3.1 Introduction
DVB-H (Digital Video Broadcasting-Handheld) is a standard that allows
broadcasting transmission in handheld devices. It is based on the DVB-T
transmission system which has proven its ability to serve fixed, portable and mobile
terminals, but handheld terminals (defined as light-weight, battery-powered
apparatus) require specific features from the transmission system serving them. In
effect, the transmission system shall offer the possibility to repeatedly turn the power
off to some parts of the reception chain. This will reduce the average power
consumption of the receiver. The transmission system shall also ensure it is easy for
receivers to move from one transmission cell to another while maintaining the DVB-
H service. For a number of reception scenarios (indoor, outdoor, pedestrian and
inside a moving vehicle), the transmission system shall offer sufficient flexibility and
scalability to allow the reception of DVB-H services at various speeds, whilst
optimizing transmitter coverage. Furthermore, as services are expected to be
delivered in environments that suffer high levels of man-made noise, the
transmission system shall offer the means to mitigate their effects on the
performance of the receiving terminal. As DVB-H aims to provide a generic way to
serve handheld terminals in various part of the world, the transmission system shall
offer the flexibility to be used in various transmission bands and channel bandwidths.
1.3.2 Overview of the system
A full DVB-H system is a combination of elements of the physical and link layer, as
well as service information. DVB-H makes use of the following technological
elements for the link and physical layers:
• Link layer:
- Time Slicing in order to reduce the average power consumption of the
receiving terminal and enable smooth and seamless frequency
handover. Time-slicing is mandatory for DVB-H.
- Forward Error Correction for Multiprotocol Encapsulated Data (MPE-
FEC) for an improvement in C/N-performance (C/N: Carrier to Noise
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ratio) and Doppler performance in mobile channels, also to improve
the tolerance to impulse interference. MPE-FEC is not mandatory for
DVB-H.
• Physical layer: DVB-T with the following technical elements specifically
targeting DVB-H use:
- DVB-H signaling in the TPS-bits to enhance and speed up service
discovery. Cell identifier is also carried on TPS-bits to support quicker
signal scan and frequency handover on mobile receivers;
- 4K-mode for trading off mobility and SFN cell size, allowing single
antenna reception in medium SFNs at very high speed, adding thus
flexibility in the network design;
- In-depth symbol interleaver for the 2K and 4K-modes for further
improving their robustness in mobile environment and impulse noise
conditions.
It should be emphasized that neither time slicing nor MPE-FEC technology elements,
as they are implement in the link layer, touch the DVB-T physical layer in any way.
This means that the existing receivers for the DVB-T are not disturbed by DVB-H
signals (DVB-H is totally backward compatible to DVB-T). It is also important to
notice that the payload of DVB-H is IP-datagrams or other network layer datagrams
encapsulated into MPE-sections. In view of the restricted data rates suggested for
individual DVB-H services and the small displays of typical handheld terminals, the
classical audio and video coding schemes used in digital broadcasting do not suit
DVB-H well. It is therefore suggested to exchange MPEG-2 video by H.264/AVC or
other high-efficiency video coding standards.
As said before, the physical layer has four extensions to the existing DVB-T physical
layer. First, the bits in transmitter parameter signaling (TPS) have been upgraded to
include two additional bits to indicate presence of DVB-H services and possible use
of MPE-FEC to enhance and speed up the service discovery. Second, a new 4K
mode orthogonal frequency division multiplexing (OFDM) mode is adopted for
trading off mobility and single-frequency network (SFN) cell size, allowing single-
antenna reception in medium SFNs at very high speeds. This gives additional
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flexibility for the network design. 4K mode is an option for DVB-H complementing
the 2K and 8K modes that are as well available. Also all the modulation formats,
QPSK, 16QAM and 64QAM with nonhierarchical or hierarchical modes, are
possible to use for DVB-H. Third, a new way of using the symbol interleaver of
DVB-T has been defined. For 2K and 4K modes, the operator may select (instead of
native interleaver that interleaves the bits over one OFDM symbol) the option of an
in-depth interleaver that interleaves the bits over four or two OFDM symbols,
respectively. This approach brings the basic tolerance to impulse noise of these
modes up to the level attainable with the 8K mode and also improves the robustness
in mobile environment. Finally, the fourth addition to DVB-T physical layer is the 5-
MHz channel bandwidth to be used in no-broadcast bands. This is of interest, e.g., in
the United States, where a network at about 1.7 GHz is running using DVB-H with a
5-MHz channel.
1.3.3 Structure of a DVB-H receiver
The conceptual structure of DVB-H user equipment is depicted in Figure 10.
Figure 10: Conceptual Structure of DVB-H Receiver
It includes a DVB-H receiver (which includes a DVB-T demodulator, a time-slicing
module, and an optional MPE-FEC module) and a DVB-H terminal.
The DVB-T demodulator recovers the MPEG-2 transport stream (TS) packets from
the received DVB-T RF signal. It offers three transmission modes: 8K, 4K, and 2K
with the corresponding signaling (Transmitter Parameter Signaling, TPS).
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The time-slicing module controls the receiver to decode the wanted service and shut
off during the other service bits. It aims to reduce receiver power consumption while
also enabling a smooth and seamless frequency handover.
The MPE-FEC module, provided by DVB-H, offers in addition to the error
correction in the physical layer transmission, a complementary FEC function that
allows the receiver to cope with particularly difficult reception situations.
Let’s consider an example of using DVB-H for transmission of IP-services. In this
example, given in Figure 11, both traditional MPEG-2 services and time-sliced
“DVB-H services” are carried over the same multiplex. The handheld terminal
decodes/uses IP-services only. It is important to note that 4K mode and the in-depth
interleavers are not available, for compatibility reasons, in cases where the multiplex
is shared between services intended for fixed DVB-T receivers and services for
DVB-H devices.
Figure 11: A conceptual description of using DVB-H system (sharing a MUX with MPEG-2
services)
Some of the basic parameters of DVB-H physical layer are given in Tables 5 –7.
Table 5 gives the frequency domain parameters for the 8-MHz channel. For other
bandwidths, simple scaling offers the parameters where narrowing channel
bandwidth means increased symbol length. Note that the number of active carriers is
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smaller than directly proposed by the FFT size. As in DVB-T, this is due to having
some guard band with zero amplitude carriers.
Table 5: Frequency Domain Parameters for DVB-H OFDM Signal (8 MHz Channel)
Table 6 gives the OFDM symbol lengths in time domain with and without guard
intervals. It is worth noting that with the longest guard interval and using 4K mode
one can build SFN networks using up to about 33–35-km transmitter distances. The
maximum distance is dictated by the transmission delay between the transmitter
sites. This should be smaller than the guard interval length.
Table 6: Time Domain Parameters for DVB-H OFDM Signal (8 MHz Channel)
Table 7 gives some examples of the achievable multiplex capacities with various
modulation schemes and convolutional coding rates. The given numbers assume that
MPE-FEC has been used with code rate 3/4. It should be noted that the DVB-H
standard allows use of various code rates for MPE-FEC or even having no MPE-FEC
at all. Again the figures can be scaled directly to other code rates and/or channel
bandwidths where needed. For practical purposes, in networks aiming to serve
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mobile handheld terminals, mainly the strongest code rates (i.e., 1/2 or 2/3) for
convolutional coding lead to networks with good coverage and total performance.
1.3.4 Standards and Protocol Stack
In the DVB-H transmission are involved some standards: DVB, MPEG-2 and DSM-
CC. The DVB-H system specification represents the central document, referencing
all other necessary standards.
The physical layer specification has been incorporated in the DVB-T standard. Time
slicing and MPE-FEC have been described in a new chapter of the DVB Data
Broadcast specification. This document also defines the Multi-Protocol
Encapsulation (MPE). DVB-H-specific signaling has been integrated into the DVB
Service Information (SI) specification.
The system specification determines mandatory and optional elements. As said
before, time slicing is mandatory for all DVB-H services and therefore has become a
characteristic feature of them. The system specification is complemented by DVB-H
Implementation Guidelines which contain hints for the use and practical
implementation of the standard.
Table 7: Useful Net Bitrates (Mb/s) for Nonhierarchical Systems in 8-MHz Channel With MPE-
FEC Code Rate 3/4; Full Multiplex Assumed to be DVB-H
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Figure 12: The DVB-H standards family
In general, the data broadcasting is considered an extension of a DVB transmission
standards based on MPEG-2. These standards define techniques for delivering
MPEG-2 Transport Stream (TS) which typically contains audio and video streams,
through a variety of transmission media.
As shown in Figure 13, data are transported within the MPEG-2 TS by application
areas. Each application area has different features about filtering, overhead, size, and
so on.
The application areas are:
• Data Piping: end-to-end asynchronous data transmission mechanism through
a DVB compliant network. This specification shows how to put data into
MPEG-2 TS packets.
• Data Streaming: it supports the transmission of services which need an end-
to-end streaming-oriented delivery, using an asynchronous, synchronous or
synchronized mode through a DVB compliant network. Data transmitted in
this way are transported within PES (Program Elementary Stream) packets
defined in MPEG-2.
• MPE (Multi-Protocol Encapsulation): it supports sending services which
need communication protocol datagrams transmission through a DVB
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network. Datagram transmission based on the MPE specifications is realized
by encapsulating datagrams into DSM-CC sections. DSM-CC sections are
structures of data suitable for private sections defined by MPEG-2. This
specification both contains also signaling mechanism for IP/MAC services
inside the DVB networks and allows to implement DVB receiver which are
automatically tuned when they access to IP/MAC flows on one or more TS.
• Data Carousel: it allows to transmit services which need a periodic
transmission of modules through DVB networks. These modules have a fixed
size and can be updated, added or deleted inside the data carousel. They can
be also grouped if this is required by the service. Data sent according to this
specification are transmitted within DSM-CC data carousel.
• Object Carousel: this specification was added in order to support the
transmission of services which need a periodic sending of User-to-User
DSM-CC through DVB networks. Data sent according to this specification
are transmitted within both DSM-CC object carousel and DSM-CC data
carousel defined by the MPEG-2 DSM-CC standard.
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Figure 13: Standards in the Protocol Stack of the Broadcast Channel
Each application area is defined by the following two parts:
• Transport: it indicates how data are encapsulated in the suitable structures
• Control: it specifies the information that allows receiver to automatically
synthesize the services selected by user. It also indicates how data can be
extracted from TS and how they can be built again.
Both information will be a part of one bit flow (TS).
The transmission of datagrams IP through DVB network is based on the protocol
stack ISO/OSI compliant shown in Figure 14.
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Figure 14: Protocol Stack, OSI Layer 1 to 3
1.3.5 Data Link Layer
The standard DVB way of carrying IP datagrams in an MPEG-2 TS is to use Multi-
Protocol Encapsulation (MPE). With MPE each IP datagram is encapsulated into one
MPE section. A stream of MPE sections are then put into an elementary stream (ES),
i.e., a stream of MPEG-2 TS packets with a particular program identifier (PID). Each
MPE section has a 12-B header, a 4-B cyclic redundancy check (CRC-32) tail and a
payload length, which is identical to the length of the IP datagram, which is carried
by the MPE section. A typical situation for future handheld DVB-H devices may be
to receive audio/video services transmitted over IP on ESs having a fairly low bitrate,
probably in the order of 250 kb/s. The MPEG-2 TS may, however, have a bitrate of
e.g., 10 Mb/s. The particular ES of interest thus occupies only a fraction (in this
example, 2.5%) of the total MPEG-2 TS bitrate. In order to drastically reduce power
consumption, one would ideally like the receiver to demodulate and decode only the
2.5% portion of interest, and not the full MPEG-2 TS. With time slicing this is
possible, since the MPE sections of a particular ES are sent in high bitrate bursts
instead of with a constant low bitrate. During the time between the bursts (the off-
time) no sections of the particular ES are transmitted and this allows the receiver to
power off completely during off-times. This feature is necessary in handheld mobile
user terminals.
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1.3.5.1 Time-Slicing
The objective of time-slicing is to reduce the average power consumption of the
terminal and enable smooth and seamless service handover. Time-slicing consists of
sending data in bursts using significantly higher instantaneous bit rate compared to
the bit rate required if the data were transmitted using traditional streaming
mechanisms.
To indicate to the receiver when to expect the next burst, the time (delta-t) to the
beginning of the next burst is indicated within the burst. Between the bursts, data of
the elementary stream is not transmitted, allowing other elementary streams to use
the bandwidth otherwise allocated. Time-slicing enables a receiver to stay active
only a fraction of the time, while receiving bursts of a requested service. It is
important to note that the transmitter is constantly on (i.e. the transmission of the
transport stream is not interrupted).
In Figure 15 is illustrated how the Time-Slicing works.
The peak bit rate of the bursts may potentially be the full MPEG-2 TS bit rate, but
could also be any lower peak value allocated for the ES. If the value is lower than the
peak bitrate, the MPEG-2 TS packets of a particular burst may be interleaved with
MPEG-2 TS packets belonging to other ESs (DVB-H or other, e.g., SI or MPEG-2
audio/video).
Thanks to the flexible delta_t signaling there are no requirements to have fixed burst
sizes or fixed time between bursts. A variable-bit-rate coded video stream could
Figure 15: Principle of Time-Slicing
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therefore use a variable burst size and/or a variable time between bursts. It should be
noted that one burst could contain several services, which would then share PID but
could e.g., be discriminated by different IP addresses. If the average bitrate of the ES
is 500 kb/s, the peak bitrate is 10 Mb/s and the burst size is 2 Mb (maximum allowed
value), the burst time becomes 200 ms, and the burst cycle time 4 s. The receiver,
however, has to wake up a little bit before the burst to synchronize and be prepared
to receive the sections. Assuming a figure of 200 ms for the total preparation time,
including some margin for delta_t jitter, the power saving in the example becomes
90%. It is probable that the actual parameters used for Time Slicing will be a
compromise between power consumption and other factors, such as service access
time and RF performance.
Time-slicing also supports the possibility to use the receiver to monitor neighbouring
cells during the off-times (between bursts). By accomplishing the switching of the
reception from one transport stream to another during an off period it is possible to
accomplish a quasi-optimum handover decision as well as seamless service
handover.
Time-slicing is always used in DVB-H.
1.3.5.1.1 Handover Considerations
DVB-H supports very efficient handover behavior including seamless handover.
This is due to the existence of the off periods in time slicing, where the receiver may
scan other frequencies in order to find the best potential alternative frequency, or
actually execute the handover.
It should be emphasized that the possibility of “silently” evaluating alternative
frequencies, without disturbing the ongoing reception of the service, is a very
important feature of the DVB-H system.
If the same TS is available in a number of adjacent cells, the transmission of the TS
should preferably be time synchronized. This is in principle straightforward to
achieve, since the same methods could be used as in SFNs and the required time
accuracy is much less strict than in the SFN case. If the transmissions of the TS on
different frequencies are time synchronized, a receiver will receive the next burst at
the time indicated by delta_t also on any new frequency carrying this TS. Since the
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TS is the same also, the content of the bursts are the same, which means that the
handover will naturally be seamless.
1.3.5.2 MPE-FEC
The objective of the MPE-FEC is to improve the C/N- and Doppler performance in
mobile channels and to improve the tolerance to impulse interference. This is
accomplished through the introduction of an additional level of error correction at the
MPE layer. By adding parity information calculated from the datagrams and sending
this parity data in separate MPE-FEC sections, error-free datagrams can be output
after MPE-FEC decoding despite a very bad reception condition. The use of MPE-
FEC is optional.
With MPE-FEC a flexible amount of the transmission capacity is allocated to parity
overhead. The IP datagrams of each time sliced burst are protected by Reed–
Solomon parity data (RS data), calculated from the IP datagrams of the burst. The RS
data are encapsulated into MPE-FEC sections, which are also part of the burst and
are sent immediately after the last MPE section of the burst, in the same ES, but with
different table_id than the MPE sections, which enables the receiver to discriminate
between the two types of sections in the ES. For the calculation of the RS data an
MPE-FEC frame is used. The MPE-FEC frame consists of an application data table
(ADT), which hosts the IP datagrams (and possible padding), and an RS data table,
which hosts the RS data. This is shown in Figure 16.
Figure 16: MPE-FEC Frame
Technical Overview
50
For a given set of transmission parameters providing 25 % of parity overhead, the
MPE-FEC may require about the same C/N as a receiver with antenna diversity. The
MPE-FEC overhead can be fully compensated by choosing a slightly weaker
transmission code rate, while still providing far better performance than DVB-T
(without MPE-FEC) for the same throughput. This MPE-FEC scheme should allow
high-speed single antenna DVB-T reception using 8K/16-QAM or even 8K/64-QAM
signals. In addition MPE-FEC provides good immunity to impulse interference.
The MPE-FEC, as standardized, works in such a way that MPE-FEC ignorant (but
MPE capable) receivers will be able to receive the data stream in a fully backwards-
compatible way, provided it does not reject the used stream_type.
1.3.6 Physical Layer:
1.3.6.1 4K Mode and In-depth Interleavers
The objective of the 4K mode is to improve network planning flexibility by trading
off mobility and SFN size. To further improve robustness of the DVB-T 2K and 4K
modes in a mobile environment and impulse noise reception conditions, an in-depth
symbol interleaver is also standardized. The additional 4K transmission mode is a
scaled set of the parameters defined for the 2K and 8K transmission modes. It aims
to offer an additional trade-off between Single Frequency Network (SFN) cell size
and mobile reception performance, providing an additional degree of flexibility for
network planning.
Terms of the trade-off can be expressed as follows:
• The DVB-T 8K mode can be used both for single transmitter operation and
for small, medium and large SFNs. It provides a Doppler tolerance allowing
high speed reception.
• The DVB-T 4K mode can be used both for single transmitter operation and
for small and medium SFNs. It provides a Doppler tolerance allowing very
high speed reception.
• The DVB-T 2K mode is suitable for single transmitter operation and for
small SFNs with limited transmitter distances. It provides a Doppler tolerance
allowing extremely high speed reception.
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For 2K and 4K modes the in-depth interleavers increase the flexibility of the symbol
interleaving, by decoupling the choice of the inner interleaver from the transmission
mode used. This flexibility allows a 2K or 4K signal to take benefit of the memory of
the 8K symbol interleaver to effectively quadruple (for 2K) or double (for 4K) the
symbol interleaver depth to improve reception in fading channels. This provides also
an extra level of protection against short noise impulses caused by, e.g. ignition
interference and interference from various electrical appliances. 4K and in-depth
interleavers affect the physical layer, however their implementations do not imply
large increase in equipment (i.e. logic gates and memory) over the version 1.4.1 of
DVB-T standard for either transmitters or receivers. A typical mobile demodulator
already incorporates enough RAM and logic for the management of 8K signals,
which exceed that required for 4K operation.
The emitted spectrum of the 4K mode is similar to the 2K and 8K modes thus no
changes in transmitter filters are envisaged.
1.3.6.2 DVB-H Signaling: TPS-bit Signaling
The objective of DVB-H signaling is to provide a robust and easy-to-access signaling
to the DVB-H receivers, thus enhancing and speeding up service discovery. So, TPS-
bit signaling provides robust multiplex level signaling capability to the DVB-T
transmission system.
TCP is a very robust signaling channel allowing TCP-lock in a demodulator with
very low C/N-values. TPS provides also a faster way to access signaling than
demodulating and decoding the Service Information (SI) or the MPE-section header.
The DVB-H system uses two TPS bits to indicate the presence of time-slicing and
optional MPE-FEC. Besides these, the signaling of the 4K mode and the use of in-
depth symbol interleavers are also standardized.
1.3.7 DVB-H Networks
1.3.7.1 The IP-DataCasting System
A typical application for DVB-H is IP Datacasting service to handheld terminals like
mobile phones. Figure 17 shows a full IP-DC system with the various components
and elements included.
Technical Overview
52
First the service system is used to produce the various IP streams (like video streams)
to the network. They are then distributed over the multicast intranet to the IP
Encapsulators, which will output the DVB-H TS with time slicing and MPE-FEC
included. This TS is then distributed to the DVB-T/H transmitters of the broadcasting
network. The IP Datacast (IP-DC) system may include other functions via cellular
networks like General Packet Radio Service (GPRS) or Universal Mobile
Telecommunications System (UMTS).
Figure 17: A Typical IP-DC System
1.3.7.2 Broadcasting Spectrum
DVB-H is intended to use the same broadcasting spectrum, which DVB-T is
currently using. The physical layer of DVB-H is in fact DVB-T and therefore there is
a full spectrum compatibility with other DVB-T services.
DVB-H can be introduced either in a dedicated DVB-H network or by sharing an
existing DVB-T multiplex between DVB-H and DVB-T services. When the final
selection of the DVB-H concept was made, the capability to share a multiplex with
DVB-T was indeed one of the decisive factors, as it was seen that this would enhance
the commercial introduction possibilities of the service in the crowded UHF
broadcasting spectrum. Technically almost any DVB-T frequency allotment or
assignment can be used also for DVB-H; the only limitations come from
Technical Overview
53
interoperability with GSM900 cellular transmitter in the DVB-H terminal. If
simultaneous operation is required, the frequencies below about 700–750 MHz are
favored.
For broadcasters DVB-H can be seen just as a new means to provide broadcast
services for a new, interesting group of customers, namely, the mobile phone users.
If this is seen as interesting enough, spectrum will be available. It is in any case
expected that the situation will be more relaxed after the analog TV services will
start to close.
It should also be noted that DVB-H is very spectrum efficient when compared with
the traditional TV-services. One 8-MHz channel can deliver 30–50 video streaming
services to the small screen terminals. This is ten times more than standard-definition
TV (SDTV) with MPEG-2 or 20 times more than high-definition TV (HDTV) with
AVC.
1.3.7.3 DVB-H Network Architectures
As said in the previous paragraph, there three possible architectures for a DVB-H
network:
• DVB-H Standalone
• Shared network DVB-H/T by multiplexing
• Shared network DVB-H/T by hierarchical modulation
1.3.7.3.1 DVB-H Standalone (Dedicated DVB-H Networks)
When a full multiplex can be reserved for DVB-H, the freedom in planning is
increased. If needed, now it is possible to select the new 4k mode or in-depth
interleavers introduced in the latest DVB-T standard for DVB-H. A dedicated DVB-
H network is shown in Figure 18.
Technical Overview
54
Figure 18: A DVB-H Standalone Network
A typical network is composed of several SFN areas, each using its own frequency
allotment. The maximum size of one SFN area depends on the FFT size, guard
interval, and geographical properties in the network, but can typically be in the order
of tens of kilometers. Each SFN area has probably several GPS-synchronized
transmitters supported by a number of on-channel repeaters to cover some smaller
holes. As the required field strength in a DVB-H network is fairly high and the
allowed total interfering power from an allotment is limited by the coordinated plan,
the number of synchronized main transmitters should be higher and the transmitter
powers and antenna heights lower than in a traditional DVB-T network. The network
can be called dense SFN. Obviously the cost of the network is higher than in
conventional DVB-T network, but also the number of services in one multiplex is ten
times higher.
1.3.7.3.2 Sharing with DVB-T
Shared Network DVB-H/T by Multiplexing:
In this scenario the DVB-H IP services are inserted to the transport stream at the
multiplexer level in parallel with the MPEG-2 services.
Remultiplexing issues fall into two main themes:
• Jitter: how does jitter in the MUX and modulator chain affect time-slicing?
Technical Overview
55
• PSI/SI management (ID harmonization requirements; private descriptors)
According to a survey that was done in 2nd quarter 2003 in order to evaluate these
two themes, the existing multiplexers of most vendors can be used for multiplexing
both DVB-H services and MPEG-2 services. Should first deployments within pilot
networks show any problems, these can be expected to be minor.
Figure 19: DVB-H Introduction example in existing DVB-T Networks with Multiplexing
The following two minor issues can make existing multiplexers more suitable for
DVB-H:
• Smooth reinsertion of managed PSI/SI sections
• Support for ID management in INT table
By smoothing the reinsertion of PSI/SI sections, a stable amount of bitrate will be
used for PSI/SI, leading to even less jitter on elementary streams carrying timesliced
IP services.
Multiplexers usually manage the IDs contained in the PSI/SI tables (PAT, PMT,
NIT, SDT, etc.). The goal is to re-allocate PIDs, service IDs, transport stream IDs etc
in order to resolve collisions between incoming transport streams. The INT table is
currently a private section and does not participate in this ID management.
Therefore, if the multiplexer changes e.g. the service ID of an incoming service that
Technical Overview
56
carries encapsulated IP streams, the INT table that contains the IP-to-ES mappings
for this service is destroyed.
The DVB-T network already being in place, the time to market depends only on the
availability of a timeslice-capable IP encapsulator and of timeslice-capable receivers.
Shared Network DVB-H/T by Hierarchical Modulation:
In this scenario the DVB-H IP services are inserted in the High Priority stream of the
DVB-T modulator. The modulators are the existing 2K or 8K ones. A new TS
distribution network for the IP stream is needed as well as the IP-encapsulator with
DVB-H capability.
Figure 20: DVB-H Introduction example in existing DVB-T Network with Hierarchical
Modulation
There are several advantages of using hierarchical modulation instead of
multiplexing:
• There can be separate sets (though mutually dependent) of modulation
parameters for fixed (DVB-T) and mobile (DVB-H) reception, leading to
more optimal bandwidth usage
Technical Overview
57
• No multiplexer being involved, the jitter and ID management concerns that
apply to the multiplexing scenario do not apply to the hierarchical modulation
scenario
The disadvantage is that a fixed amount of bandwidth has to be used for DVB-H, so
there is no flexibility in there.
The time to market depends only on the availability of a timeslice-capable IP
encapsulator and of timeslice-capable receivers, and on the deployment of
modulators and SFN-areas that support hierarchical mode.
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CCHHAAPPTTEERR TTWWOO
2. ACCESS NETWORK ANALYSIS AND
COMPARISON FOR UMTS, WiMAX AND DVB-H
2.1 Introduction
UMTS, WiMAX and DVB-H technologies are now introduced and it is clear why
they are chosen as access solutions thanks to their features.
In this chapter their Access Networks are analyzed in order to comprehend how far
the convergence can advance. This study is done assuming that an all-IP backbone is
the basis of a reachable convergence between different Access Networks.
In the following paragraphs the main functionalities of the different elements
constituting the three Access Networks are described to be able to create a unique
network model.
According to this scope, all the devices composing the different Access Network are
compared on a functional level. This comparison is needed to individuate which
functions and features they perform in common and to highlight their own
peculiarities and specializations.
Resuming, the method used to achieve a convergence is composed of three phases:
1. Analysis: it is needed to understand which specific functions every physical
element of each access network performs
2. Comparison: it is a functional comparison between different access networks’
devices looking for common functionalities based on the previous analysis
3. Unique Architectural Block Model: it is just a logical access network model
because a physical convergence is unreachable, according to what comes
from the analysis and comparison.
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2.2 Access Networks Analysis: UMTS UTRAN
The Access Network in the UMTS system is the UTRAN (UMTS Terrestrial Radio
Access Network).
The UTRAN consists of a set of Radio Network Subsystems connected to the Core
Network through the Iu.
A RNS consists of a Radio Network Controller one or more Node Bs and optionally
one SAS. It can be either full UTRAN or only a part of a UTRAN. The RNS offers
the allocation and release of specific radio resources to establish means of connection
in between an UE and the UTRAN, it is also responsible for the resources and
transmission/reception in a set of cells.
A Node B is a logical node in the RNS responsible for radio transmission/reception
in one or more cells to/from the UE. It is connected to the RNC through the Iub
interface.
A Node B can support FDD mode, TDD mode or dual-mode operation.
There are two chip-rate options in the TDD mode:
• 3.84 Mcps TDD
• 1.28 Mcps TDD
Each TDD cell supports either of these options.
A Node B which supports TDD cells can support one chip-rate option only, or both
options.
Alike, a RNC which supports TDD cells can support one chip-rate option only, or
both options. A RNC is a logical node in the RNS in charge of controlling the use
and the integrity of the radio resources. It is responsible for the Handover decisions
that require signaling to the UE. A RNC may include a combining/splitting function
to support combination/splitting of information streams.
Inside the UTRAN, the RNCs of the Radio Network Subsystems can be
interconnected together through the Iur. Iu(s) and Iur are logical interfaces. Iur can be
conveyed over direct physical connection between RNCs or virtual networks using
any suitable transport network.
The UTRAN architecture is shown in Figure 21.
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Figure 21: UTRAN Architecture
As said before, the RNS is responsible for the resources of its set of cells. For each
connection between User Equipment and the UTRAN, One RNS is the Serving RNS.
The Serving RNS is in charge of the radio connection between a UE and the
UTRAN. When required, Drift RNSs support the Serving RNS by providing radio
resources. The role of an RNS (Serving or Drift) is on a per connection basis
between a UE and the UTRAN.
To support UE mobility between UTRAN and GERAN Iu mode, the Serving RNS
may be connected to the DBSS and vice versa. The role of an RNS or BSS (Serving
or Drift) is on a per connection basis between an UE and the UTRAN/GERAN Iu
mode.
The UTRAN is layered into a Radio Network Layer and a Transport Network Layer.
The UTRAN architecture, i.e. the UTRAN logical nodes and interfaces between
them, are defined as part of the Radio Network Layer.
For each UTRAN interface (Iu, Iur, Iub, Iupc) the related transport network layer
protocol and functionality is specified.
The transport network layer provides services for user plane transport, signaling
transport and transport of implementation specific O&M. An implementation of
equipment compliant with the specifications of a certain interface shall support the
Radio Network Layer protocols specified for that interface. It shall also as a
minimum, for interoperability, support the transport network layer protocols
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according to the transport network layer specifications for that interface. The
network architecture of the transport network layer is not specified by 3GPP and is
left as an operator issue.
The equipment compliant to 3GPP standards shall at least be able to act as endpoints
in the transport network layer, and may also act as a switch/router within the
transport network layer.
For implementation specific O&M signaling to the Node B, only the transport
network layer protocols are in the scope of UTRAN specifications.
In summary, UTRAN is composed of three entities:
1. Node B
2. RNC
3. RNS
2.2.1 Node B functionality
The Node B is responsible for Spreading and Modulation, that is:
• code generation
• supports FDD, TDD or both, and CDMA
It terminates physical channels and transport channels, while logical channels
terminate at RNC.
The Node B supports Fast Power Control with an “inner loop”, in effect it measures
strength of received signals and informs UE if it needs to adjust.
Moreover, it measures connection quality and strength.
2.2.2 RNC functionality
The RNC, Radio Resource Management, guarantees stability and QoS of radio
connection by means of a radio bearer. It supports Power Control with an “outer
loop”.
The RNC performs the Handover Control, in particular it decides if should there be a
handover, based on measurements of UE and Node B.
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Other functionalities are Admission Control and Packet Scheduling, it decides if a
new session can be established on the UTRAN without compromising the quality of
existing sessions:
• Plan channel use, calculate interference and utilization levels
• Configure radio resources accordingly
The RNC is responsible for Code Management and Macrodiversity Management. In
UMTS one UE can communicate via up to 6 antennas simultaneously, this is
Macrodiversity.
The RNC can also provide the UTRAN Control function:
• Setup, maintenance, and release of a radio connection (radio bearer)
• System information broadcasting (e.g., radio measurement criteria, etc.)
• Initial access and signaling connection setup and management
(synchronization, broadcast of initial scrambling code, etc.)
• UTRAN security functions: it protects user and control data by encryption
and integrity protection
• UTRAN level mobility management, informing new cell (Node B) and UE
about handover, new channel, etc. and serving RNS relocation
• Database handling: it stores cell information and sends it to corresponding
Node Bs and Ues, that is cell identity, power levels, connections quality,
neighboring cells information (needed for handover)
• UE positioning: it selects and controls UE positioning method, using cell ID,
RTT, etc.
For each UE, one RNC is responsible and it is called SRNC, Serving RNC: typically
this is the RNC controlling the cell in which the UE is located. If UE moves to a cell
controlled by a different RNC this becomes the Drift RNC, DRNC and the control
stays with SRNC.
Also macrodiversity may introduce DRNCs.
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Figure 22: Serving RNC and Drift RNC
2.2.3 UTRAN Functions
The main element in the access network is the RNC, its function is to manage all the
functionalities in the radio interface (user side) and to allow the service transport in a
transparent mode to the Core Network. So, the user mobility, the handover function
and the macro diversity are completely controlled by the UTRAN.
The Core Network is completely separated from the service transport functions,
while control and signaling functions end in the RNC. RNC converts them into the
radio protocol format necessary for the user.
The main functions performed by the UTRAN are the following:
• Transfer of User Data: This function provides user data transfer capability
across the UTRAN between the Iu and Uu interfaces points.
• Functions related to Overall System Access Control: System access is the
means by which a UMTS user is connected to the UTRAN in order to use
UMTS services and/or facilities. User system access may be initiated from
either the mobile side, e.g. a mobile originated call, or the network side, e.g. a
mobile terminated call.
- Admission Control: The purpose of the admission control is to admit or
deny new users, new radio access bearers or new radio links (for
example due to handover). The admission control should try to avoid
overload situations and base its decisions on interference and resource
measurements. The admission control is employed at for example
initial UE access, RAB assignment/reconfiguration and at handover.
These cases may give different answers depending on priority and
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situation. The Admission Control function based on UL (Up Link)
interference and DL (Down Link) power is located in the Controlling
RNC. The Serving RNC is performing admission Control towards the
Iu interface.
- Congestion Control: The task of congestion control is to monitor,
detect and handle situations when the system is reaching a near
overload or an overload situation with the already connected users.
This means that some part of the network has run out, or will soon run
out of resources. The congestion control should then bring the system
back to a stable state as seamless as possible. Congestion Control is
performed within the UTRAN.
- System Information Broadcasting: This function provides the mobile
station with the Access Stratum and Non Access Stratum information
which are needed by the UE for its operation within the network. The
basic control and synchronization of this function is located in UTR.
- MOCN and GWCN Configuration Support: The MOCN is the Multi
Operator Core Network and the GWCN is the GateWay Core
Network. So, in the MOCN configuration only the radio access part of
the network is shared. For the MOCN configuration it is required that
the rerouting function is supported. In the GWCN configuration,
besides shared radio access network, the core network operators also
share part of the core network, at least MSC and/or SGSN. For both
the GWCN and MOCN configurations, the RNC carries the selected
PLMN-id between network sharing supporting UEs and the
corresponding CN.
• Radio Channel Ciphering and Deciphering: This function is a pure
computation function whereby the radio transmitted data can be protected
against a no authorized third-party. Ciphering and deciphering may be based
on the usage of a session-dependent key, derived through signaling and/or
session dependent information. This function is located in the UE and in the
UTRAN.
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• Functions related to Mobility:
- Handover: This function manages the mobility of the radio interface. It
is based on radio measurements and it is used to maintain the Quality
of Service requested by the Core Network. Handover may be directed
to/from another system (e.g. UMTS to GSM handover). The Handover
function may be either controlled by the network, or independently by
the UE. Therefore, this function may be located in the SRNC, the UE,
or both.
- SRNS Relocation: The SRNS (Serving RNS) Relocation function
coordinates the activities when the SRNS role is to be taken over by
another RNS/BSS. The SRNS relocation function manages the Iu
interface connection mobility from an RNS to another RNS/BSS. The
SRNS Relocation is initiated by the SRNC (Serving RNC). This
function is located in the RNC and the CN.
- Paging Support: This function provides the capability to request a UE
to contact the UTRAN/GERAN Iu mode when the UE is in Idle. This
function also encompasses a coordination function between the
different Core Network Domains onto a single RRC connection. A
RRC connection is a point-to-point bi-directional connection between
RRC peer entities on the UE and the UTRAN sides, respectively.
- Positioning: This function provides the capability to determine the
geographic position of a UE.
- NAS Node Selection Function: The optional NAS (Non Access
Stratum) Node Selection Function (NNSF) enables the RNC to
initially assign Core Network resources to serve a UE and
subsequently setup a signaling connection to the assigned Core
Network resource.
• Functions related to Radio Resource Management and Control: Radio
Resource Management is concerned with the allocation and maintenance of
radio communication resources. UMTS radio resources must be shared
between circuit transfer mode services and packet transfer modes services
(i.e. Connection-oriented and/or connectionless-oriented services).
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- Radio Resource Configuration and Operation: This function performs
configures the radio network resources, i.e. cells and common
transport channels, and takes the resources into or out of operation.
- Radio Environment Survey: This function performs measurements on
radio channels (current and surrounding cells) and translates these
measurements into radio channel quality estimates. Measurements may
include:
1. Received signal strengths (current and surrounding cells);
2. Estimated bit error ratios, (current and surrounding cells);
3. Estimation of propagation environments (e.g. high-speed, low-
speed, satellite, etc.);
4. Transmission range (e.g. through timing information);
5. Doppler shift;
6. Synchronization status;
7. Received interference level;
8. Total DL transmission power per cell.
This function is located in the UE and in the UTRAN.
- Combining/Splitting Control: This function controls the
combining/splitting of information streams to receive/ transmit the
same information through multiple physical channels (possibly in
different cells) from/towards a single mobile terminal. In some cases,
depending on physical network configuration, there may be several
entities which combine the different information streams, i.e. there
may be combining/splitting at the SRNC, DRNC or Node B level. This
function is located in the UTRAN.
- Connection Set-Up and Release: This function is responsible for the
control of connection element set-up and release in the radio access
sub network. The purpose of this function is:
1. To participate in the processing of the end-to-end connection set-
up and release;
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2. And to manage and maintain the element of the end-to-end
connection, which is located in the radio access sub network.
This function is located both in the UE and in the RNC.
- Allocation and Deallocation of Radio Bearers: This function consists
of translating the connection element set-up requests into physical
radio channel allocation accordingly to the QoS of the Radio Access
Bearer. This function is located in the CRNC (Controlling Radio
Network Controller) and SRNC.
- TDD – Dynamic Channel Allocation (DCA): Dynamic Channel
Allocation (DCA) is an automatic process for assigning traffic
channels in a frequency reuse wireless system. The base station
continuously monitors the interference in all idle channels and makes
an assignment using an algorithm that determines the channel that
will produce the least amount of additional interference. DCA is used
in the TDD mode. It includes Fast DCA and Slow DCA. Slow DCA
is the process of assigning radio resources, including time slots, to
different TDD cells according to the varying cell load. Fast DCA is
the process of assigning resources to Radio Bearers, and is related to
Admission Control.
- Radio Protocol Functions: This function provides user data and
signaling transfer capability across the UMTS radio interface by
adapting the services (according to the QoS of the Radio Access
Bearer) to the Radio transmission.
- RF Power Control: This group of functions controls the level of the
transmitted power in order to minimize interference and keep the
quality of the connections. It consists of the following functions:
UL/DL Outer/Inner Loop Power Control.
- Radio Channel Coding: This function introduces redundancy into the
source data flow, increasing its rate by adding information calculated
from the source data, in order to allow the detection or correction of
signal errors introduced by the transmission medium. The channel
coding algorithm used and the amount of redundancy introduced may
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be different for the different types of logical channels and different
types of data. This function is located in both the UE and in the
UTRAN.
- Radio Channel Decoding: This function tries to reconstruct the
source information using the redundancy added by the channel
coding function to detect or correct possible errors in the received
data flow. The channel decoding function may also employ a priori
error likelihood information generated by the demodulation function
to increase the efficiency of the decoding operation. The channel
decoding function is the complement function to the channel coding
function. This function is located in both the UE and in the UTRAN.
- Channel Coding Control: This function generates control information
required by the channel coding/decoding execution functions. This
may include channel coding scheme, code rate, etc. This function is
located in both the UE and in the UTRAN.
- Initial (random) Access Detection and Handling: This function will
have the ability to detect an initial access attempt from a mobile
station and will respond appropriately. The handling of the initial
access may include procedures for a possible resolution of colliding
attempts, etc. The successful result will be the request for allocation
of appropriate resources for the requesting mobile station. This
function is located in the UTRAN.
• Functions related to Broadcast and Multicast Services:
- Broadcast/Multicast Information Distribution: The
broadcast/multicast information distribution function distributes
received CBS (Cell Broadcast Service) messages towards the BMC
(Broadcast/Multicast Control) entities configured per cell for further
processing. The distribution of broadcast/multicast information relate
on the mapping between service area and cells controlled by the
RNC. The provision of this mapping information is an O&M
function.
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- Broadcast/Multicast Flow Control: When processing units of the
RNC becomes congested, the Broadcast/Multicast Flow Control
function informs the data source about this congestion situation and
takes means to resolve the congestion.
• Functions related to MBMS (Multimedia Broadcast Multicast Service):
- MBMS Provision: The MBMS provision enables the RNC to provide
a multicast service via an optimized transmission of the MBMS
bearer service in UTRAN via techniques such as PTM (Point-To-
Multipoint) transmission, selective combining, Soft Combining and
transmission mode selection between PTM and PTP (Point-To-Point)
bearer. The MBMS provision enables the RNC to provide a
broadcast service via a PTM transmission bearer.
- MBMS Notification Coordination: The characteristic of MBMS
implies a need for MBMS notification co-ordination i.e. specific
handling of MBMS Notification when UE is in Cell-DCH state.
MBMS notification co-ordination is performed by UTRAN when the
session is ongoing. The TMGI (Temporary Mobile Group Identity) is
used for coordination.
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The scope of this section is to analyze each entity functionalities in the UTRAN, so
in the Figure 23 it is shown and summarized which device performs the functions
explained before.
Figure 23: UTRAN Entities and Functions
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2.3 Access Networks Analysis: WiMAX
The Mobile WiMAX End-to-End Network Architecture is based on All-IP platform,
all packet technology with no legacy circuit telephony. It offers the advantage of
reduced total cost of ownership during the lifecycle of a WiMAX network
deployment. The use of All-IP means that a common network core can be used,
without the need to maintain both packet and circuit core networks, with all the
overhead that goes with it.
The Mobile WiMAX End-To-End Network Architecture is composed of three main
logical entities: MS, ASN, CSN. Each of these entities represents a grouping of
functional entities. Each of these functions may be realized in a single physical
device or may be distributed over multiple physical devices. The grouping and
distribution of functions into physical devices within a functional entity (such as
ASN) is an implementation choice; a manufacturer may choose any physical
implementation of functions, either individually or in combination, as long as the
implementation meets the functional and interoperability requirements.
This section will focus on the WiMAX Access Network that can be individuated in
the ASN. The ASN represents a boundary for functional interoperability with
WiMAX clients, WiMAX connectivity service functions and aggregation of
functions embodied by different vendors.
As the ASN is just a logical entity, it is composed of two different physical devices:
• Base Station
• ASN Gateway
This basic view of the many entities within the functional groupings of ASN is
shown in Figure 24.
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Figure 24: Network IP-Based Architecture
Mapping of functional entities to logical entities may be performed in different ways,
so the purpose of the analysis performed in this section is understanding how is
composed the logical entity and the correspondence between the physical and
functional entities.
2.3.1 Base Station
The main task of the Mobile WiMAX Base Station is 802.16 Interface Handling, the
interface (R1) between the MS and the ASN is via air interface (PHY and MAC)
specifications (IEEE 802.16d/e). This interface (R1) may include additional
protocols related to the management plane.
Another important task is Processes Handling like:
• Handover
• Power Control
• Network Entry
Base Station also includes QoS PEP (Policy Enforcement Points) for traffic via air
interface.
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Furthermore, the Base Station is responsible of Micro Mobility HO triggering for
mobility tunnel establishment, supporting Tunnelling Protocol toward ASN GW EP
and Traffic Classification.
Other main functionalities are:
• Radio Resource Management Update
• MS Activity Status Update (Active, Idle)
• Session Management (RSVP Proxy)
• DCHCP Proxy
• Multicast Group Association Management (IGMP Proxy)
• Key Management
• TEK/KEK (Traffic Encryption Key/Key Encryption Key) Generation and
Delivery to the BS/MS
2.3.2 ASN GW (Access Service Network Gateway)
The ASN GW is the gateway between the Access Network (ASN) and the Core
Network (CSN). So it must hold the interface between the ASN and the CSN (R3)
which supports AAA, Policy Enforcement and Mobility Management capabilities.
This interface (R3) also encompasses the bearer plane methods (e.g. tunnelling) to
transfer IP data between the ASN and CSN.
The main functions provided by the ASN GW are:
• Intra ASN Location Management & Paging
• Network Session/Mobility Management (server)
• Regional Radio Resource Management & Admission control
• ASN Temporary Cashing subscriber profile and encryption keys (ASN like-
VLR)
• AAA Client/Proxy
• delivery Radius/Diameter messaging to selected CSN AAA
• Mobility Tunneling establishment and management with BSs
• Session/mobility management (client)
• QoS and Policy Enforcement
• Foreign Agent (FA) (with Proxy MIP)
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• Routing to selected CSN
Finally, the interface between the Base Station and the ASN GW is standardized and
is called R6. It consists of a set of control and bearer plane protocols for
communication between the BS and the ASN GW. The bearer plane consists of intra-
ASN data path or inter-ASN tunnels between the BS and ASN GW. The control
plane includes protocols for IP tunnel management (establish, modify, and release) in
accordance with the MS mobility events. R6 may also serve as a conduit for
exchange of MAC states information between neighboring BSs.
2.3.3 Functions
Some general tenets have guided the development of Mobile WiMAX Network
Architecture include the following features:
• Provision of logical separation between such procedures and IP addressing,
routing and connectivity management procedures and protocols to enable use
of the access architecture primitives in standalone and interworking
deployment scenarios,
• Support for sharing of ASN(s) of a Network Access Provider (NAP) among
multiple NSPs,
• Support of a single NSP providing service over multiple ASN(s) – managed
by one or more NAPs,
• Support for the discovery and selection of accessible NSPs by an MS or SS
• Support of NAPs that employ one or more ASN topologies,
• Support of access to incumbent operator services through internetworking
functions as needed,
• Specification of open and well-defined reference points between various
groups of network functional entities (within an ASN, between ASNs,
between an ASN and a CSN, and between CSNs), and in particular between
an MS, ASN and CSN to enable multi-vendor interoperability
• Support for evolution paths between the various usage models subject to
reasonable technical assumptions and constraints,
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• Enabling different vendor implementations based on different combinations
of functional entities on physical network entities, as long as these
implementations comply with the normative protocols and procedures across
applicable reference points, as defined in the network specifications
• Support for the simplest scenario of a single operator deploying an ASN
together with a limited set of CSN functions, so that the operator can offer
basic Internet access service without consideration for roaming or
interworking.
The WIMAX architecture also allows both IP and Ethernet services, in a standard
mobile IP compliant network. The flexibility and interoperability supported by the
WiMAX network provides operators with a multi-vendor low cost implementation of
a WiMAX network even with a mixed deployment of distributed and centralized
ASN’s in the network.
So, the major functions performed by the WiMAX Access Network can be
summarized as:
• Support for Service and Applications: The WiMAX Architecture includes the
support for:
- Voice, multimedia services and other mandated regulatory services
such as emergency services and lawful interception
- Access to a variety of independent Application Service Provider (ASP)
networks in an agnostic manner
- Mobile telephony communications using VoIP
- Support interfacing with various interworking and media gateways
permitting delivery of incumbent/inherit services translated over IP (for
example, SMS over IP, MMS, WAP) to WiMAX access networks
- Support delivery of IP Broadcast and Multicast services over WiMAX
access networks
• Security: The end-to-end WiMAX Network Architecture is based on a
security framework that is agnostic to the operator type and ASN topology
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and applies consistently internetworking deployment models and usage
scenarios. In particular there is support for:
- Strong mutual device authentication between an MS and the WiMAX
network, based on the IEEE 802.16 security framework,
- All commonly deployed authentication mechanisms and authentication
in home and visited operator network scenarios based on a consistent
and extensible authentication framework,
- Data integrity, replay protection, confidentiality and non-repudiation
using applicable key lengths,
- Use of MS initiated/terminated security mechanisms such as Virtual
Private Networks (VPNs),
- Standard secure IP address management mechanisms between the
MS/SS and its home or visited NSP.
• Mobility and Handover: The end-to-end WiMAX Network Architecture has
extensive capability to support mobility and handovers. It will:
- Include vertical or inter-technology handovers— e.g., to Wi-Fi, 3GPP,
3GPP2, DSL, or MSO – when such capability is enabled in multi-mode
MS
- Support IPv4 or IPv6 based mobility management. Within this
framework, and as applicable, the architecture will accommodate MS
with multiple IP addresses and simultaneous IPv4 and IPv6
connections,
- Support roaming between NSPs,
- Utilize mechanisms to support seamless handovers at up to vehicular
speeds— satisfying well defined.
Some of the additional capabilities in support of mobility include the support of:
- Dynamic and static home address configurations,
- Dynamic assignment of the Home Agent in the service provider
network as a form of route optimization, as well as in the home IP
network as a form of load balancing,
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- Dynamic assignment of the Home Agent based on policies.
• Scalability, Extensibility, Coverage and Operator Selection: The end-to-end
WiMAX Network Architecture has useful support for scalable, extensible
operation and flexibility in operator selection. In particular, it will:
- Enable a user to manually or automatically select from available NAPs
and NSPs,
- Enable ASN and CSN system designs that easily scale upward and
downward – in terms of coverage, range or capacity
- Accommodate a variety of ASN topologies – including hub-and-spoke,
hierarchical, and/or multi-hop interconnects
- Accommodate a variety of backhaul links, both wireline and wireless
with different latency and throughput characteristics
- Support incremental infrastructure deployment
- Support phased introduction of IP services that in turn scale with
increasing number of active users and concurrent IP services per user
- Support the integration of base stations of varying coverage and
capacity - for example, pico, micro, and macro base stations
- Support flexible decomposition and integration of ASN functions in
ASN network deployments in order to enable use of load balancing
schemes for efficient use of radio spectrum and network resources
Additional features pertaining to manageability and performance of WiMAX
Network Architecture include:
- Support a variety of online and offline client provisioning, enrollment,
and management schemes based on open, broadly deployable, IP-based,
industry standards
- Accommodation of Over-The-Air (OTA) services for MS terminal
provisioning and software upgrades
- Accommodation of use of header compression/suppression and/or
payload compression for efficient use of the WiMAX radio resources.
• Multi-Vendor Interoperability: Another key aspect of the WiMAX Network
Architecture is the support of interoperability between equipment from
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different manufacturers within an ASN and across ASNs. Such
interoperability will include interoperability between:
- BS and backhaul equipment within an ASN,
- Various ASN elements (possibly from different vendors) and CSN,
with minimal or no degradation in functionality or capability of the
ASN.
The IEEE 802.16 standard defines multiple convergence sub-layers. The
WiMAX Network Architecture framework supports a variety of CS types
including: Ethernet CS, IPv4 CS and IPv6 CS.
• Quality of Service: The WiMAX Network Architecture has provisions for
support of QoS mechanisms. In particular, it enables flexible support of
simultaneous use of a diverse set of IP services. The architecture supports:
- Differentiated levels of QoS: coarse-grained (per user/terminal) and/or
fine-grained (per service flow per user/terminal),
- Admission control,
- Bandwidth management
- Implementation of policies as defined by various operators for QoS-
based on their SLAs (including policy enforcement per user and user
group as well as factors such as location, time of day, etc.). Extensive
use is made of standard IETF mechanisms for managing policy
definition and policy enforcement between operators. The flexible
WiMAX network specifications allows different implementations of
Access Service Network (ASN) configurations namely ASN profiles,
including both distributed/collapsed as well as centralized architectures.
Furthermore, the WiMAX forum is developing an interoperability
framework in which intra-ASN and inter-ASN interoperability across
different vendors is ensured.
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2.4 Access Networks Analysis: DVB-H
DVB-H is principally a transmission system allowing reception of broadcast
information on single antenna handheld mobile devices. It provides an efficient way
of carrying multimedia services over digital terrestrial broadcasting networks to
handheld terminals (DVB-H). So, the DVB-H network can not be considered as an
Access Network in the way that is usually meant. DVB-H is a really delivering
network. No channel access method is used to share a communications channel or
physical communications medium between multiple users.
According to the scope of this thesis, in this section the whole DVB-H network is
considered as the Access Network, always keeping in mind that the communication
is unidirectional (only in downlink).
Because DVB-H is a broadcasting technology, the functionalities performed by
DVB-H devices are regarding exclusively on the signal transmission and on the
carousel information synchronization. In addition to these (based on DVB-T
transmission), a DVB-H network provides some functionalities related to the
mobility. It is very important to underline that these functionalities just allow the
mobility but they do not manage it. The Mobility Management is not possible
because there is not a return channel and, most of all, User Localization and
Identification are not broadcast transmission requirements.
There are three possible architectures for a DVB-H network: DVB-H Standalone and
Shared network DVB-H/T by multiplexing or hierarchical modulation.
In the first one the DVB-H system has a dedicated multiplexer and the elements
composing this kind of architecture are shown in Figure 25.
Figure 25: Headend Construction for Dedicated Multiplex
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The IP Encapsulator is assumed to take responsibility for generating MPE sections
from incoming IP datagrams, as well as to add the required PSI/SI data. Also, MPE-
FEC Frames, when used, are generated in the IP Encapsulator. The output stream of
the IP Encapsulator is composed of MPEG-2 transport packets.
Instead, a shared network is a network of DVB-T transmitters is serving both DVB-
H and DVB-T terminals. The existing DVB-T network has to be, however, designed
for portable indoor reception so that it can provide high enough field strength for the
hand-held terminals inside the wanted service area. The only required modification
in the transmitters is an update so that the DVB-H signaling bits and Cell ID bits are
added to the TPS information of the transmitter.
The actual sharing is done at the multiplex level. DVB-H offers a full flexibility to
select the wanted portion of the multiplex to DVB-H services.
The key DVB-H component in the network is the IP-Encapsulator, where the MPE
of IP data, time slicing, and MPE-FEC are implemented.
Another possibility to share the network is to use the DVB-T hierarchical
modulation. In that case the MPEG-2 and DVB-H IP services will have their own
independent TS inputs in the DVB-T transmitters. The DVB-H services would use
the high-priority part, which would offer increased robustness over the low-priority
input, which is then used for the normal digital TV services.
In Figure 26 a shared network by multiplexing is illustrated and its components are
highlighted.
Figure 26: Headend Construction for Mixed Multiplex
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In order to introduce DVB-H services into an existing DVB-T network using
multiplexing, the following steps are required, in any order:
• Timeslice-capable IP Encapsulators are connected to the last-hop multiplexer,
which is ideally located in each coverage area (MFN or SFN), and a fixed
amount of bitrate is reserved for DVB-H services
• The last-hop-multiplexers are upgraded for better DVB-H support (smoothing
of reinserted PSI/SI tables, management of INT table)
• If necessary, improve the coverage of the DVB-T network (more cells,
upgrade of single-transmitter cells to SFN-areas, addition of radio frequency
repeaters)
The possibility to have global and local IP services is the same as in the case of a
dedicated DVB-H network, and the properties of the IP backbone network are the
same. The number of last-hop-multiplexers determines the granularity of service
coverage areas. This is why these multiplexers (and with them the IP encapsulators)
are ideally located locally in each coverage area (MFN or SFN).
For network-wide distribution of IP streams, there is now an additional option: the IP
streams can be encapsulated centrally, and distributed to the sites within a centrally
produced transport stream, which is then re-multiplexed by the last-hop-multiplexer
to produce the final transport stream that is broadcast.
Whether or not this is a good option depends on many factors. IP networks can be
expected to be cheaper, more scalable, and simpler to manage than transport stream
distribution networks. But if there is capacity available in an existing transport
stream distribution network, why not use it, especially if there is no IP network
available.
In this case, the centrally encapsulated IP streams should not be timesliced, but
simply embedded in the transport stream using normal multi-protocol encapsulation.
The local IP Encapsulator can then decapsulate these IP streams, and timeslice them
as any other IP stream that is received over the IP backbone network.
It would be technically possible and allowed by the standard to timeslice also the
centrally encapsulated IP streams, and to add locally another set of timesliced IP
streams. However, this would not be optimal from power-saving perspective.
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As timeslicing is a technology for reduction of power consumption of a mobile
handheld terminal, there is no need for central timeslicing.
A shared network by hierarchical modulation is quite different as it is shown in
Figure 27.
Figure 27: Headend Construction for Hierarchical Tranmission
In order to introduce DVB-H services into an existing DVB-T network using
hierarchical modulation, the following steps are required:
1. If necessary, replace modulators with models that support hierarchical mode and
put a 2nd synchronized transport stream distribution system in place for
modulators in SFN-areas
2. Timeslice-capable IP Encapsulators are connected to the modulators, or, in case
of SFN-areas, to the SFN timestamp inserter
If necessary, improve the coverage of the DVB-T network (more cells, upgrade of
single-transmitter cells to SFN-areas, addition of radio frequency repeaters).
From DVB-H perspective, this case is identical to having a dedicated DVB-H
network, so all the comments on how to construct an IP backbone network and how
to mix global and local IP streams are the same.
Hence, the main elements of a DVB-H network are the IP Encapsulator, the
Multiplexer and the DVB-H Modulator.
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2.4.1 IP Encapsulator
The IP Encapsulator puts the IP packets into TS packets using the MPE protocol. It is
composed by the Time-Slicing Module and MPE-FEC Module.
The Time-Slicing Module controls the receiver/transmitter to decode the requested
service and shut off during the other service bits. It aims to reduce power
consumption while also enabling a smooth and seamless frequency handover. The
MPE-FEC module offers, in addition to the error correction in the physical layer
transmission, a complementary FEC function that allows the receiver/transmitter to
cope with particularly difficult reception situations. As we said before, DVB has
introduced multiprotocol encapsulation (MPE) for encoding OSI-model layer 3
(Network Layer) datagrams into TS packets. Each IP datagram is encoded into a
single MPE section. A Single Elementary Stream may contain multiple MPE section
streams. The IPDC DVB-H Receiver may differentiate MPE encoded IP streams by
checking the IP source and/or destination address in the IP datagram carried within
an MPE section. In such case, the Receiver does not differentiate MPE section
streams, but is directly filtering IP streams. For such a Receiver, there is no need to
differentiate MPE section streams within an Elementary Stream.
2.4.2 Multiplexer
A DVB network consists of one or more Transport Streams (TS) each carrying a
multiplex and being transmitted by one or more DVB signals.
A MUX multiplexes a set of DVB services together, and then these services are
carried over a Transport Stream. A TS carries exactly one multiplex (set of services).
If one multiplex is transmitted on two different radio signals (i.e. DVB signals)
within a DVB network, the DVB signals carry the same TS. However, if the DVB
signals belong to different DVB networks, the TSs are different. In both cases, the set
of DVB signals, PSI information and multiplex identifiers are identical. However, in
later case, SI information (particularly the information about the actual DVB
network) is different. In such a case, only one multiplex occurs, even though the
information was carried on two different TSs.
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Therefore, a multiplex is a set of DVB services, while a TS is a bitstream carrying a
multiplex and related PSI/SI information. A multiplex may be delivered on multiple
DVB networks, while a TS belongs to exactly one DVB network.
Within a DVB network, a TS may be carried on multiple DVB signals. A DVB
signal using non-hierarchical modulation carries one TS, while a DVB signal using
hierarchical modulation carries two TSs.
A DVB service is a sequence of program events, each of which groups together a set
of components, each carried in its own Elementary Stream.
2.4.3 Modulator
The Modulator works on the physical layer. It performs the modulation of the signal
using OFDM symbols. In compliance with the DVB-T, the DVB-H Modulator can
work in 2k and 8k mode. Furthermore it can operate in the 4k mode. The additional
4k transmission mode is a scaled set of the parameters defined for the 2k and 8k
modes. In order to further improve robustness of the DVB-T 2k and 4k modes in a
mobile environment and impulse noise reception conditions, an in-depth symbol
interleaver is also standardized. In the Modulator, the TPS-bit signaling provides
robust multiplex level signaling capability to the DVB-T transmission system.
2.4.4 Features
The main features of a DVB-H system are summarized in Figure 28.
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Figure 28: Main Features of a DVB-H System
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2.5 Access Networks Comparison
After studying the three access networks considered in the previous paragraphs, this
section will be focused on the different functionalities of each element, within each
network, in order to find similarities and differences between them. Finding common
functionalities of the different Access Networks devices is needed to reach a
Convergent Access Network between different architectures, in particular the UMTS
Release 6, Mobile WiMAX and DVB-H ones.
It is important to underline that DVB-H is a broadcast technology, so it does not
really have an access network. Because the transmission is not bidirectional and there
is not the need of knowing who and where the users are, it is more difficult to find
similarities with the other two technologies. Hence, the parallelism is less obvious.
Figure 29: DVB-H “Access Network”
Figure 29 shows which are the components of a DVB-H Access Network, described
in the previous paragraph.
On the other side, UMTS and WiMAX are point-multipoint technologies. They can
be matched easier because of their comparable features, although they work in quite
different ways. For example, they use different multiple access schemes, modulation
modes, channel management, etc...
The UMTS and WiMAX Access Network are respectively illustrated in Figure 30
and in Figure 31.
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Figure 30: UMTS Access Network
As these pictures show, I have supposed an IP Core Network not specified yet in this
section, in order to analyze these three Access Networks and to study their
functionalities comparing their devices. The Core Network will be introduced in the
next chapter.
Figure 31: WiMAX Access Network
For the reasons listed before, the result of the Access Network Analysis is that it is
not possible to reach a physical convergence for the access networks. In fact, kinds
of features like frequency, bandwidth, modulation mode, channel management,
transport protocol, policy rules, etc are peculiar to each technology.
Thus, I have looked for a unique architectural block model, that although every
technology has its own specific features, it simplifies their analogous functionalities.
This effort is summarized and shown in Table 8.
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Table 8: Devices Functionality Comparison
The Table 8 shows that the DVB-H Modulator, the UMTS Node B and the WiMAX
Base Station play similar roles, as does the DVB-H Encapsulator, the UMTS RNC
and the WiMAX ASN-GW. However, it is evident that this association is forced
because we can see that some functions are performed by the RNC and the Base
Station, relative to the UMTS and WiMAX.
Some of the functionalities needed for the comparison are not performed by any
DVB-H device. The reason for this is that DVB-H is broadcast, so it does not provide
all the functions related to the Location and Radio Resource Management, Channel
Allocation, Admission Control, Ciphering, Scheduling and QoS Management.
Considering that each technology (DVB-H, UMTS and WiMAX) has a mobile user
terminal, the user mobility is always possible but the Mobility Management is
supported only by the UMTS and WiMAX.
From this study results also that the DVB-H MUX is not involved in the comparison
because it does not perform any functionality related to the signal processing,
furthermore its role depends on the DVB-H network architecture used.
Let’s now consider each function in detail. The Handover is a function allowed in the
DVB-H thanks to the time-slicing performed inside the Encapsulator, while in the
UMTS and WiMAX handover is based on radio measurements performed by Node B
and the Base Station together with UE. On the other side, the Handover Control is
managed by the RNC and the ASN-GW. It is the same situation concerning the
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Power Control, in fact in the DVB-H there is not a dynamic power management
because in transmission the power is determined in advance. However, the Time-
Slicing Module in the DVB-H Modulator allows power saving in the UE
(requirement needed for mobile terminals). In the UMTS (WCDMA system) the
Power Control plays a fundamental role because the mobile terminals work at the
same frequency and time inside a cell8. So, the power in both downlink and uplink
needs to be the minimum to ensure the Signal-to-Ratio (SIR) required by the service.
During a communication, UE and UTRAN exchange power control messages.
The Air Interface is distinguished for each technology [Chapter 1]. Mainly the DVB-
H Air Interface works only in transmission while the UMTS and WiMAX ones can
act also as receivers.
The Modulation/Demodulation are functions performed by the DVB-H Modulator,
UMTS Node B and WiMAX Base Station but the modulation mode used changes. In
the DVB-H system the most usable modulation scheme for a mobile and portable
reception is 16-QAM with a code rate of ½ or 2/3 (requiring a moderate C/N)9. This
modulation scheme is only in DL (downlink), in fact the DVB-H Modulator is able
to perform only the Modulation while the Demodulation is a own function of the
User Terminal (DVB-H Receiver). On the contrary, WiMAX and UMTS support
modulation schemes both for DL (Modulation) and UL (Demodulation). WiMAX
provides different modulation: QPSK, 16-QAM and 64-QAM (mandatory in DL and
optional in UL) with 1/2, 2/3, 3/4 and 5/6 convolutional code or convolutional turbo
code. The modulation used by the UMTS is QPSK and spreading and scrambling
codes are applied.
Other kinds of functions like Ciphering, Scheduling, Channel Allocation and so on
are performed only by UMTS and WiMAX, because they are bidirectional point-
multipoint technologies. Anyway, these functions are performed by UMTS and
WiMAX in different ways [Chapter 1].
8 The near-far problem makes the Power Control in UMTS a critical issue 9 Providing enough capacity to meet commercial requirements, the available constellations for a DVB-H system are QPSK, 16-QAM and eventually, although not recommended, 64-QAM. The FEC code rate can be 1/2, 2/3, 3/4, 5/6 and 7/8.
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2.6 A UNIQUE ARCHITECTURAL BLOCK MODEL FOR
ACCESS NETWORK ANALYSIS AND COMPARISON
Starting from the considerations made in the previous paragraph I have built a model
of the access network. Table 9 proves that the Access Networks can be modeled as a
unique logical access network, but the physical convergence is unachievable.
Table 9: Logical Access Network
As different blocks could be joined into one physical device or a physical device
could be split into several functional blocks, a one-to-one correspondence with the
real architectures is not always possible.
Figure 32: Architectural Block Model
Figure 32 shows the unique architectural block model that I have proposed in order
to represent the access network. It is composed of three main functional blocks
which are the Air Interface Control Module, Enforcement Module and
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Decision&Control Module. Each function of DVB-H, WiMAX and UMTS Access
Network elements can be mapped into these blocks. As said before, some elements
are split up and others are grouped together.
This block model could be seen as a tree where the Decision&Control Module
controls more Enforcement Modules and each Enforcement Module is responsible of
many Air Interfaces Control Modules.
The logical blocks that I have identified are explained in the following paragraphs.
2.6.1 Air Interface Control Module
This module is the access network end-point and it manages the physical channel
with the user. It is related to just one cell.
Some of its most important functions are:
• Handover
• Air Interface Transmission/Reception
• Power Control
• Modulation/Demodulation
The devices that compose the Air Interface Control Module are the Modulator for the
DVB-H, the Node B for the UMTS and the Base Station for the WiMAX.
2.6.2 Enforcement Module
The Enforcement Module is responsible for the resource allocation and distribution.
It mainly enforces functions controlled by the Decision&Control Module.
Some of the major functions are:
• Radio Resource Management Update
• Active Set Update
• Ciphering
• Scheduling
• Channel Allocation
This module does not hold any DVB-H device because the functions it performs are
not needed for the broadcasting transmission. This module is the one that makes the
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parallelism between UMTS and WiMAX devices less clear, considering that it
comprises the RNC and the Base Station. So, the Enforcement Module is the critical
issue in order to reach a unique Access Network.
2.6.3 Decision & Control Module
It is the module that manages all of the functions in the access network. It is the
supervisor of a set of cells.
It is responsible for:
• Power Control
• UE Location Management
• Radio Resource Control
• Handover Control
• Admission Control
• Mobility Management
• QoS Management
• Interface to the Core Network
The UMTS RNC and WiMAX ASN GW are placed inside the Decision&Control
Module. The DVB-H Encapsulator also belongs to this module but it does not
performs all the functions related to the Location, Radio Resource, Mobility and
Admission Control for the reasons explained before. Handover Control is signed as
function provided by the DVB-H Encapsulator but it is performed in a quite different
way compared with the UMTS or WIMAX one.
The Interface to the Core Network is the only one functionality provided by all three
of these devices and it distinguishes each technology.
Core Network Description
93
CCHHAAPPTTEERR TTHHRREEEE
3. CORE NETWORK: IMS TECHNICAL
DESCRIPTION
3.1 Introduction
In the previous chapter the Access Network was described and the accomplishment
was a unique architectural block model that needs to converge to one single Core
Network. Thus, in this section the Core Network is analyzed. The IMS is the best
choice for the Core Network because is the standard one that gathers all the features
needed to the convergence.
The IMS (IP Multimedia Subsystem) is a standard that defines a generic architecture
for multimedia services. IMS standard supporting multiple access types has the goal
of offering Internet services everywhere and at any time on different devices, so it is
the key for implementing fixed-mobile convergence.
As the cellular networks already provide a wide range of services (in fact any
cellular user can access the Internet using a data connection), all the power of the
Internet is already available for 3G users through the packet-switched domain, but
IMS is necessary to achieve three purposes: QoS (Quality of Service), charging, and
integration of different services. So, the reason for creating the IMS was to provide
the QoS required for real time multimedia sessions (packet-switched domain
provides a best effort service without QoS), to be able to charge multimedia sessions
appropriately (a multimedia session in the packet-switched domain usually transfers
a large amount of data that may generate large expenses to the user) and, finally, to
provide integrated services to users.
Operators want to be able to use services developed by third parties, combine them,
integrate them with services they already have, and provide the user with a
completely new service. Furthermore, the aim of the IMS is not only to provide new
services but to provide all the services that Internet provides. IMS achieves this by
using a layered and horizontal architecture where service enablers and common
function can be reused for multiple applications. This simplifies the interoperability
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and roaming, and providing bearer control, charging, and security. From an
architectural point of view IMS uses SIP protocol for signaling.
The 3GPP IMS, that is a part of 3GPP Release 5, is a standard implemented by the
Third Generation Partnership Project. It uses IPv6 and other advanced IP
technologies. Many of the capabilities of these technologies have been incorporated
into the design of the IMS architecture. This includes nearly limitless addresses, QoS
control, access independence (WLAN etc), IPSec and IPv6 routing efficiency.
The migration to an all IP context lowers maintenance costs (this is partly because
the maintenance platforms are based on open standards). It also establishes the
operation of new services that can be accessed by different types of terminal devices.
Today IP implementations are almost all IPv4 based and many of the advanced IP
technologies that have been assumed in the design of the IMS architecture are
currently not deployed on a large scale. Introducing IMS services into existing 3GPP
networks therefore requires upgrades in many parts of the system. New or enhanced
features have to be available in the different components that are involved in the end-
to-end service provisioning.
Implementing IMS will have a large impact on network structures, terminals, packet-
switched domain nodes, nodes for IP support (e.g. DNS and AAA servers), IMS
servers and application server, therefore there is doubt on the wide-spread
availability of the required critical components for 3GPP compliant IMS deployment.
3.1.1 Why IMS?
As said before, IMS provides QoS benefits, cost savings and the deployment of new
applications. The main problem with using the packet-switched domain is that it
performs a “best effort” service, which is inadequate for real-time communication,
because of the bandwidth and delay variations. But, IMS fixes this problem.
Furthermore, IMS enables the operator to know the data type and the best way to
charge the data stream (not only based on the number of byte transferred). For
example, a video conference in the packet-switched domain needs a lot of audio-
video data and this can be very expensive if the operator charges the service
proportionally to the transferred bytes. Instead, knowing the type of data thanks to
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IMS, the operator can select a different charging criteria based on time, QoS or other
service features.
IMS allows the operators to use services developed by others and combine them,
which provides innovative services for users. For example, if an operator has voice-
mail service, it can buy a test-speech conversion service and combine them to give a
speech version of the messages to blind people.
Because IMS uses Internet technology and protocols it can provide all of the Internet
services also to mobile users. Users can enjoy these services in roaming as well as in
their home network.
IMS based on packet switching can provide more efficient services than UMTS
based on circuit switching. Every service, thanks to the SIP protocol, knows all the
changes of the session, and therefore can provide new functionality. For example it
can give off an alert when the presence state of a colleague changes from busy to free
and allows the user to invite the colleague to take part in a video conference. Because
the service can know all the session features, it can make a lot of operations without
sending data which saves bandwidth. So, it can provide better QoS to the user or
serve more users with the same QoS.
IMS employs devices that don’t access to the circuit switched domain so the number
of IMS users who can connect increases considerably.
3.1.2 The Introduction of IMS
The introduction of IMS is an innovation regarding the whole network: both the core
network and the radio access network, creating an integration of different core
networks into one based on packet switching. It will be able to use just one platform
to provide multimedia services. The objective of the introduction of IMS is the
unification of the platforms that supply the services. This allows the definitions of
the services to be the same quality for all users, independent of the technology that
they use to access the network. As signaling is needed for multimedia sessions
control, it is placed on the application layer. Signaling goes through the same path as
the user traffic, instead of separated from it, as it was before. That has a considerable
effect on channel configuration because the channels also have to administrate a
signaling filter.
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In order to manage communication session and packet-switched media, assuring
traditional network compatibility (as ISDN or PSTN), new nodes need to be
introduced.
The impact that this new technology has on the network is tied to the introduction of
new protocols and network nodes. In order to realize and manage the sessions
(SIP/SDP protocols), an evolution of the signaling gateway is necessary. This will
allow the realization of all the service configurations that are based on both the
terminals and the access network.
We have to also consider that we will need to introduce the protocols that are
necessary to manage the real-time services (i.e. RTP/RTCP). They will allow
appropriate packet switching that meets the requirements of the imperceptible delays
(delays not perceived by the user).
The migration from a circuit-switched platform to a packet-switched one also needs
to take into consideration the management of the additional overload that is created
by the transmission. In fact, in circuit switching the voice, or otherwise the coded
stream of the voice, is transported by the network in a transparent mode. The
migration to the packet switching requires both payload (real content and voice
communication) and header (additional data). Because of the lower bitrate of this
service (12.2 kbps for the voice), the overload (the additional load of transport data)
can become enormous—up to almost 100%.
It is therefore necessary to determine the optimal configuration of the radio channels
in order to assure the quality of service for the flow of the signaling (SIP/SDP),
which is something that does not exist in circuit switching, and also for the protocols
that serve to transport voice and control data.
3.2 The Architecture of IMS
When one thinks of IMS architecture, it is important to keep in mind that the Third
Generation Partnership Project (3GPP) does not standardize roles, but instead
functions. The importance of the requirements is in the standardized interfaces. This
creates a flexible architecture where implementers can combine different functions
into a single node or split that function into multiple nodes. In order to access the
IMS network, the end user needs a terminal which is referred to as a User Equipment
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(UE). This terminal could be a mobile device, Personal Digital Assistant (PDA), or
even a computer. In addition, the modes of access to the network are variable. The
terminals can not only connect using a radio link, but can also use WLAN or ADSL.
The nodes included in the IP Multimedia Core Network Subsystem are:
• User Databases, called HSS (Home Subscriber Service) and SLF (Subscriber
Location Function)
• SIP servers, called CSCF (Call Section Control Function)
• AS (Application Service)
• MRF (Media Resource Functions) each one further divided into:
- MRFC (Media Resource Function Controller)
- MRFP (Media Resource Function Processor)
• BGCF (Breakout Gateway Control Function)
• PSTN gateway, each divided into:
- SGW (Signaling Gateway)
- MGCF (Media Gateway Controller Function)
- MGW (Media Gateway)
Figure 33: Architecture of IMS
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3.2.1 The HSS and SLF Databases
The Home Subscriber Subsystem (HSS) holds all of the user’s personal information.
This allows the user to log on to the network by holding his or her subscription data
that is required to run the multimedia sessions.
Possible Data:
• Location Information
• Security Information (Authentication and Authorization)
• User Profile Information (Service that User is Subscribed to)
• Serving-CSCF allocated to user
A network may contain more than one HSS, but the information for a single user is
stored in a single HSS. If a network has more than one HSS it needs a SLF, which is
a database that maps the users’ addresses. When a node queries the SLF to obtain
information related to a particular user, the SLF locates the correct HSS containing
the user’s information.
3.2.2 The CSCF
The CSCF (Call/Session Control Function) is a SIP server that is essential to IMS.
The three types of this server are:
• P-CSCF (Proxy-CSCF)
• I-CSCF (Interrogating-CSCF)
• S-CSCF (Serving-CSCF)
3.2.2.1 P-CSCF
P-CSCF is the first point of contact (in the signaling place) between the IMS terminal
and the IMS network. This means that all requests initiated by the IMS terminal or
destined to the IMS terminal traverse the P-CSCF. P-CSCF is necessary to receive
the requests and forward them in the appropriate direction, either towards the IMS
terminal or towards the IMS network. During IMS registration, a particular P-CSCF
is allocated to each IMS terminal. That P-CSCF never changes for the duration of the
registration.
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The following are some of the most important functions of the P-CSCF:
• Security: One of the functions of P-CSCF is security. It first establishes some
IPsec security associations toward the IMS terminal. The IPsec is a security
standard that allows the encryption and authentication of IP packets. P-CSCF
authenticates the user and asserts his or her identity to the other nodes in the
network. This means it is not necessary for the other nodes to further
authenticate the user, because they trust the P-CSCF.
• SIP Verification: Another function of the P-CSCF is to verify the SIP
requests to the terminal in order verify that they are appropriate for the SIP
rules that govern that terminal.
• SIP Message Compressor and Decompressor: This compresses the SIP
message in order to gain bandwidth, and then decompresses it at the receiving
end.
• PDF: The P-CSCF may include a PDF (Policy Decision Function) that
authorizes and media planes and manages QoS over the media plane.
• Charging: Generates charging information toward a charging collection node.
The P-CSCF may be located either in the visited network or in the home network.
When the packet network is based on GPRS, the P-CSCF is always located in the
same network where the GGSN (Gateway GPRS Support Node) is located.
3.2.2.2 The I-CSCF
The I-CSCF is a SIP proxy server located at the edge of an administrative domain.
The DSN (Domain Name System) records of the domain hold the address of the I-
CSCF. When a SIP server looks for the next hop for a message it obtains the address
of an I-CSCF belonging to the destination domain. I-CSCF interfaces with HSS and
SLF. It finds the user location information and routes the SIP request to the
appropriate destination.
Optionally I-CSCF may encrypt part of the messages if it has sensitive information
about the domain.
Typically, I-CSCF is located in the home network.
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3.2.2.3 The S-CSCF
Although the S-CSCF is basically a SIP server it also performs session control. It
works as a SIP registrar as well so it can connect the user location (IP address of the
user’s terminal) with Public User Identity (the user’s SIP address of record).
When a user is connecting to the IMS network, S-CSCF HSS’s interface informs
HSS that it is the allocated S-CSCF for the registered user and gets user
authentication vector.
S-CSCF has a central role in the network: all the SIP signaling to or from the IMS
terminal passes through the allocated S-CSCF. It analyzes the SIP messages to
determinate the application servers that they have to cross. Each of these servers will
be able to provide its services to the user. If the user dials a telephone number instead
of a SIP URI the S-CSCF provides translation the service.
S-CSCF has a policy role which inhibits the user from performing an unauthorized
session.
S-CSCF is located in the home network.
3.2.3 The AS
The AS (Application Server) is a SIP entity interfaced with S-CSCF that hosts and
executes services. It can operate like the following:
• SIP proxy
• SIP UA (User Agent): endpoint
• SIP B2BUA (Back-to-Back User Agent): concatenation of two SIP User
Agents
There are three types of AS:
• SIP AS (Application Server): the native AS that hosts and executes IP
Multimedia Services based on SIP
• OSA-SCS (Open Service Access – Service Capability Server): this
application server provides an interface to the OSA framework Application
Server. It has all the OSA capabilities, including the ability to access securely
to IMS from external network.
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• IM-SSF (IP Multimedia Switching Function): this server permits the reuse of
CAMEL (Customized Application for Mobile network Enhanced Logic)
services developed for GSM and it allows a gsmSCF (GSM Service Control
Function).
The AS may optionally interface to the HSS but only when the AS is located in the
Home Network.
3.2.4 The MRF
The MRF (Media Resource Network) is located in the home network and provides it
with a source of media. This allows the home network to play announcements, mixed
media streams (e.g. in centralized conference bridge), transcode between different
codes, obtain statistics, and perform analyses.
It is divided into:
• MRFC (MRF Controller): a signaling plane node that acts as a User Agent. It
contains a SIP interface towards the S-CSCF and controls the resources in
MSFP.
• MRFP (MRF Processor): a media plane node
3.2.5 The BGCF
The BGCF is a SIP server with routing functionality based on telephone numbers. It
is used in sessions that are initiated by an IMS terminal and addressed to a user in the
circuit-switched network (like PSTN or PLMN).
Depending on where the destination user BGCF is located:
• if this network isn’t the one where the BGCF is located, it selects an
appropriate network where interworking with the circuit-switched domain is
to occur;
• if interworking is to occur in the same network where the BGCF is located, it
selects an appropriate PSTN/CS gateway.
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3.2.6 The PSTN/CS Gateway
The PSTN/CS gateway provides an interface toward a circuit-switched network,
allowing IMS terminals to make and receive calls to and from the PSTN.
The PSTN/CS gateway is further divided into three functional elements:
• SGW (Signaling Gateway): interfaces with the signaling plane of the CS
network and performs lower layer protocol conversion.
• MGCF (Media Gateway Control Function): central node of the PSTN/CS
gateway, witch implements a state machine that does protocol conversion and
maps SIP (the call control protocol on the IMS side) to either ISUP over IP or
BICC over IP (the call control protocol in circuit-switched networks). In
addition it controls the resources in an MGW.
• MGW (Media Gateway): interfaces the media plane of the PSTN or CS
network and:
- It is able to send and receive IMS media over the Real-Time Protocol
(RTP),
- It uses PCM (Pulse Code Modulation) time slots to connect to the CS
network,
- It performs transcodification when the IMS terminal does not support
the codec used by the CS side.
3.2.7 Home and Visited Networks
IMS inherits the concepts of home networks (our operator’s network) and visited
networks (the network of another operator who manage our operator’s no covered
area). If there is a roaming agreement between the home and visited network, where
some aspects like costs or quality of services are defined, the user benefits from the
same service that is provided by his home network.
3.2.8 Identification in the IMS
As in the PSTN (Public Switched Telephone Network) users are identified with their
telephone numbers and the service with some special number like 800, in IMS we
need some identification method too.
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3.2.8.1 Public User Identification
An IMS user is allocated with one or more Public User Identities that identify him in
a deterministic way like the MSISDN in GSM.
A public User Identity is either a SIP URI (e.g. sip: [email protected]) or a
TEL URL (+39-06-123456). Additionally, it is possible to include a telephone
number in a SIP URL (sip: [email protected]; user=phone). The Public
User Identities are needful to route the SIP signaling.
We need all this type of format for the Public User Identities in order to allow the
interaction between different type of terminals like computers and telephones.
There are reasons for allocating more than one Public User Identity to a user, such us
having the ability to differentiate personal identities like the private and the business
one or for triggering a different set of services.
3.2.8.2 Private User Identities
Each IMS subscriber is assigned a Private User Identity that isn’t SIP URI or TEL
URL but takes the format of NAI (Network Access Identifier). Private User Identities
are exclusively used for subscription purposes and not for routing SIP requests so the
users don’t need to know it. They are stored in a smart card like IMSI in SIM for
GSM.
3.2.8.3 The relation between Public and Private User Identities
Every user has a Private User Identity and one or more Public User Identity, this data
are stored in the HSS.
3.2.8.4 Public Service Identities
The Public Service Identity is an identity allocated to a service that can take the
format of a SIP URI or a TEL URL.
3.2.9 SIM, USIM and ISIM in 3GPP
The UICC (Universal Integrated Circuit Card) is a removable card that contains a
limited storage of data, including subscription information, authentication keys, a
phone book, and messages. Removing the smart card from a terminal and inserting it
into another one the user can easily move his subscription.
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UICC is a generic term that defines the physical characteristics of the smart card, like
the number and the positions of the pins or the voltage values that define a standard
interface between the UICC and the terminal.
The UICC may contain several applications:
• SIM (Subscriber Identity Module): provides storage for a collection of
parameters for the identification of the user and the service, which are
essential for operating in a GSM network. Sometime it is used to refer to the
physical card whereas in this case UICC is more appropriated.
• USIM (Universal Subscriber Identity Module): is an application that resides
in third-generation UICCs. It provide another set of parameters, similar in
nature but different from those provided by SIM, useful for the identification
of the user and the service in a UMTS (Universal Mobile
Telecommunication) network.
• ISIM (IP Multimedia Service Identity Module): contains the collection of
parameters that are used for user identification, user authentication, and
terminal configuration when the terminal operates in the IMS. Among other
things it will contain Private and Public User Identity, Home Network
Domain URL (used to find the home network during the registration
procedure) and Long-term secret (a key for authentication, encrypt/decrypt or
integrity purpose). The user can not modify these parameters.
ISIM, USIM and SIM can co-exist in the same UICC.
Access to the 3GPP IMS network relies on presence of either an ISIM or a USIM
application in the UICC although ISIM is preferred because it is tailored to the ISIM.
Non-3GPP IMS networks that do not support UICC in the IMS terminals store the
parameters contained in the ISIM as a part of terminal’s configuration or in the
terminal’s build-in memory.
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3.3 Protocols
The protocols that are defined for IMS are classified into three extensive categories:
1. protocols needed for signaling and session control
2. protocols used in the “media plane”
3. authentication and security protocols
3.3.1 Session Control Protocol
A session is an exchange of data in a communication. Many Internet applications
need the creation and the management of a session. The protocol which is dedicated
to the session control in IMS is SIP [§ 3.4].
3.3.2 Media Plane Protocol
IMS employs RTP (Real Time Protocol) and RTCP (Real Time Control Protocol,
which provides statistics and information about the media stream) for delivering
multimedia contents.
Multimedia contents can be transported through an IP network, so it is possible for
packets to accumulate delays. Due to this, packets may arrive at the destination late
or out of sequence because of the IP network jitter. RTP timestamps are put inside
the packets to allow the receiver to get the correct content. All the packets are also
numbered in order to verify those that were misplaced during the transmission. If the
number of lost packets increases considerably, we can choose different codes for
improving the quality of service.
RTCP gives statistics about QoS and permits synchronization among the media.
One of the most important features of RTCP is that it establishes an association
between the RTP timestamps and the reference clock. So, is possible to achieve
synchronization among the media. The synchronization is of vital importance in
applications like video conferences.
3.3.3 Security and Authentication Protocol
In the IMS architecture there are three interfaces where authentication functions are
performed. The authentication protocol used within this interfaces is DIAMETER. It
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works over safe protocols as TCP and STCP and is an evolution of the previous
RADIUS protocol.
3.4 SIP Protocol
SIP (Session Initiation Protocol) is a signaling and control protocol on the
application layer. This protocol needs a transport network to be IP based.
SIP guarantees establishing, changing and ending communication sessions. Sessions
can be single or multi user, regardless of the session communication type.
Furthermore, SIP allows the achievement of a complete integration between data and
voice in real-time communication, regardless of the device type.
SIP is also independent from the transport layer, so the transport protocol can be
UDP, TCP or STCP. Because of this, SIP has got its own dependability and
protection mechanism.
SIP signaling messages are similar to the HTTP ones, so we can say that SIP is a
textual protocol.
In summary, SIP is distinguished by these main features:
• Based on the Client/Server model: every transaction is originated from a node
of the network able to act as client and it is directed to another entity that
involves from server;
• Simple: signaling is composed by a sequence of textual message (header and
body);
• Internet-oriented: it provides a complete integration with all the Internet open
standards like HTTP, URI (Uniform Resource Indicators), DNS (Domain
Name System) and MIME (Multipurpose Internet Mail Extension);
• Terminal-based: user terminals have the complete control of the session (or of
the call), while in the telephonic network (circuit-switched) the control is
decentralized;
• Text-based: the protocol is managed through text-based messages, so it
allows the adaptation the different situations;
• Based on a Request/Response model: every transaction consists of a request
and one or more answers (provisional or final).
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3.4.1 SIP functionality
As said before, the main goal of SIP is to deliver a session description to a user at
their current location. Once the user has been located and the initial session
description delivered, SIP can deliver new descriptions to modify the characteristics
of the outgoing sessions and terminate the session whenever the user wants. A
session description is, as its name indicates, a description of the session to be
established. It contains enough information for the remote user to join the session. In
multimedia sessions over the Internet this information includes the IP address and
port number where the media needs to be sent and the codes used to encode the voice
and the images of the participants.
In addition, in order to establish and close multimedia communication, the SIP
protocol supports five fundamental functions:
• User location: it determinates which terminal system we have to use for the
multimedia communication,
• User Availability: it determinates if a part wants to be involved in the
communication,
• User Capability: it determinates which kind of the media are involved in the
communication and its descriptive parameters,
• Session Set Up: it establishes the session, both on the originating and
terminating side,
• Session Management: session management foresees the changing of the
parameters of a session in course, the invocation of services on behalf of the
user and the closing of the session.
Finally, we have to consider that SIP is a layered protocol, that is, each functionality
is realized like independent elaboration stages.
SIP is used combined with other protocols:
1. RSTVP (Resource Reservation Protocol), it reserves network resources
2. RTP (Real Time Protocol) and RTCP (real Time Transfer Control Protocol), they
transport data in real time and provide QoS using feedback
3. RTSP (Real Time Streaming Protocol), it controls the media flow transmission
4. SAP (Session Announcement Protocol), it delivers multicast communication
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5. SDP (Session Description Protocol), it describes multimedia session
3.4.2 SIP Entities
A SIP network is composed of five types of logical SIP entities. Each entity has
specific functions and participates in SIP communication as a client (initiates
requests), as a server (responds to requests), or as both.
One physical device can have the functionality of more than one logical SIP entity.
The logical SIP entities are:
• User Agent (UA): it is an endpoint entity which initiates and terminates
sessions by exchanging requests and responses. UA is defined as an
application which contains both a User Agent Client (UAC, a client
application that initiates SIP requests) and a User Agent Server (UAS, is a
server application that contacts the user when a SIP request is received and
that returns a response on behalf of the user. During a session these entities
can dynamically interchange their roles while preserving them during a
transaction.
• Proxy Server: it is an intermediary entity that acts as both a server and a
client, for the purpose of making requests on behalf of other clients. Requests
are serviced either internally or by passing them on, possibly after translation,
to other servers. A Proxy interprets, and, if necessary, rewrites a request
message before forwarding it. Proxy Servers can be Stateless (logical entity
which does not keep memory of the transaction) or Stateful (proxy which
keeps the record of the transaction).
• Redirected Server: it is a server that accepts a SIP request, maps the SIP
address of the called party into zero (if there is no known address) or more
new addresses and returns them to the client. Unlike Proxy servers, Redirect
Servers do not pass the request onto other servers. It uses the Location
service.
• Registrar: it is a server that accepts REGISTER requests for the purpose of
updating a location database with the contact information of the user
specified in the request.
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• B2BUA (Back-to-Back User Agent): it is a logical entity that receives a
request, processes it as a User Agent Server (UAS) and, in order to determine
how the request should be answered, acts as a User Agent Client (UAC) and
generates requests. A B2BUA must maintain call state and actively
participate in sending requests and responses for dialogs in which it is
involved. The B2BUA has tighter control of the call than a Proxy. For
example, a Proxy cannot disconnect a call or alter the messages.
3.4.3 Messages
SIP is based on HTTP and, so, is a textual request-response protocol. There are two
types of SIP messages:
• Requests: they are sent from the client to the server. They contain a Request
Line, a Header, and a Message Body.
• Responses: they are sent from the server to the client. They contain a Status
Line, a Header, and a Message Body.
In general, SIP messages are composed of three parts: Start line, Headers and
Message Body.
3.4.3.1 Start Line
Every SIP message begins with a Start Line. The Start Line conveys the message
type (method type in requests, and response code in responses) and the protocol
version. Table 10 shows the methods that are currently defined in SIP and their
meaning.
The Start Line may be either a Request-line (requests) or a Status-line (responses), as
follows:
• The Request-line includes a Request-URI, which indicates the user or service
to which this request is being addressed.
• The Status-line holds the numeric Status-code and its associated textual
phrase.
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Table 10: SIP Methods
Method name Meaning
ACK
BYE
CANCEL
INFO
INVITE
NOTIFY
OPTIONS
PRACK
PUBLISH
REGISTER
SUBSCRIBE
UPDATE
MESSAGE
REFER
Acknowledges the establishment of a session
Terminates a session
Cancels a pending request
Transports PSTN telephony signaling
Establishes a session
Notifies the user agent about a particular event
Queries a server about its capabilities
Acknowledges the reception of a provisional response
Uploads information to a server
Maps a public URI with the current location of the user
Requests to be notified about a particular event
Modifies some characteristics of a session
Carries an instant message
Instructs a server to send a request
3.4.3.2 Headers
SIP header fields convey message attributes that provide additional information
about the message. They are similar in syntax and semantics to HTTP header fields
(in fact, some headers are borrowed from HTTP) and thus always take the format:
<name>:<value>.
Headers can span multiple lines. Some SIP headers such as Via, Contact, Route and
Record-Route can appear multiple times in a message or, alternatively, can take
multiple comma-separated values in a single header occurrence.
3.4.3.3 Message Body
A message body is used to describe the session to be initiated (for example, in a
multimedia session this may include audio and video codec types and sampling
rates), or alternatively it may be used to contain opaque textual or binary data of any
type which relates in some way to the session. Message bodies can appear both in
request and in response messages. SIP makes a clear distinction between signaling
information, conveyed in the SIP Start Line and headers, and the session description
information, which is outside the scope of SIP.
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Possible body types include:
• SDP (Session Description Protocol)
• Multipurpose Internet Mail Extensions (MIME)
• Others, to be defined in the IETF and in specific implementations.
3.4.4 Management of a SIP session
3.4.4.1 Session Establishment and Termination
Figure 34 shows the interaction between two user agents during trivial session
establishment and termination.
FFIIGGUURREE 3344:: SSIIPP SSEESSSSIIOONN EESSTTAABBLLIISSHHMMEENNTT AANNDD CCAALLLL TTEERRMMIINNAATTIIOONN
The call flow describing the session establishment shown in the figure is explained
next:
1. UA1 sends an INVITE message to Bob’s SIP address: sip:[email protected].
This message also contains an SDP packet describing the media capabilities
of the calling terminal.
2. UA2 receives the request and immediately responds with a 100-Trying
response message.
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3. UA2 starts “ringing” to inform Bob of the new call. Simultaneously a 180
(Ringing) message is sent to the UAC.
4. UA2 sends a 182 (Queued) call status message to report that the call is behind
two other calls in the queue.
5. UA2 sends a 182 (Queued) call status message to report that the call is behind
one other call in the queue.
6. Bob picks up the call and the UA2 sends a 200 (OK) message to the calling
UA. This message also contains an SDP packet describing the media
capabilities of Bob’s terminal.
7. UA1 sends an ACK request to confirm the 200 (OK) response was received.
Thus, the session termination call flow proceeds as follows:
1. The caller decides to end the call and “hangs-up”. This results in a BYE
request being sent to UA2.
2. UA2 responds with 200 (OK) message and notifies Bob that the conversation
has ended.
3.4.4.2 Call Redirection
Figure 35 shows a simple call redirection scenario.
FFIIGGUURREE 3355:: SSIIMMPPLLEE CCAALLLL RREEDDIIRREECCTTIIOONN UUSSIINNGG AA RREEDDIIRREECCTT SSEERRVVEERR
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The call redirection call flow is:
1. First a SIP INVITE message is sent to [email protected], but finds the Redirect
server sip.acme.com along the signaling path.
2. The Redirect server looks up Bob’s current location in a Location Service
using a non-SIP protocol (for example, LDAP).
3. The Location Service returns Bob’s current location: SIP address
4. The Redirect Server returns this information to the calling UA using a 302
(Moved Temporarily) response. In the response message it enters a contact
header and sets the value to Bob’s current location, [email protected].
5. The calling UA acknowledges the response by sending an ACK message.
6. The calling UAC then continues by sending a new INVITE directly to
gw.telco.com.
7. gw.telco.com is able to notify Bob’s terminal of the call and Bob “picks up”
the call. A 200 (OK) response is sent back to the calling UA.
8. The calling UA acknowledges with an ACK message.
3.4.4.3 Call Proxing
Figure 36 shows call set-up between two User Agents with the assistance of an
intermediate Proxy server.
The call flow is the following:
1. An INVITE message is sent to bob@ acme.com, but finds the Proxy server
sip.acme.com along the signaling path.
2. The Proxy server immediately responds with a 100 (Trying) provisional
response.
3. The Proxy server looks-up Bob’s current location in a Location Service using
a non-SIP protocol (For example, LDAP).
4. The Location Service returns Bob’s current location: SIP address
5. The Proxy server decides to Proxy the call and creates a new INVITE
transaction based on the original INVITE message, but with the Request-URI
in the start line changed to [email protected]. The Proxy server sends this
request to the UA2.
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6. The UA2 responds first with a 100 (Trying).
7. The UA2 responds with a 180 (Ringing) response.
8. The Proxy server forwards the 180 (Ringing) response back to the UA1.
9. When the call is accepted by the user (for example, by picking up the
handset) UA2 sends a 200 (OK) response. In this example, UA2 inserts a
Contact header into the response with the value [email protected]. Further
SIP communication will be sent directly to it and not via the Proxy server.
10. The Proxy forwards the 200 (OK) response back to the calling UAC.
11. The calling UA sends an ACK directly to UA2 at the lab (according to the
Contact header it found in the 200 (OK) response).
FFIIGGUURREE 3366:: CCAALLLL PPRROOXXIINNGG SSCCEENNAARRIIOO
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3.5 IPv6
IMS uses IPv6, which is a new version of IP that is designed to be an evolutionary
step from IPv4. It is a natural increment to IPv4. It was created to answer to some
problem of the actual standard like the followings:
• The growth of the Internet may cause exhaustion of the IPv4 address space
• The need for simpler configuration
• The requirement for security at the IP level
• The need to control the Quality of Service (QoS)
To solve this question IPv6 introduces:
• New Header Format
• Larger Address Space
• Efficient and Hierarchical Addressing and Routing Infrastructure
• Stateless and Stateful Address Configuration
• Built-in Security
• Better Support for QoS
• New Protocol for Neighboring Node Interaction
• Extensibility
3.5.1 A new IP standard
3.5.1.1 A New Header
The new header format of IPv6 is in place to minimize the header overhead.
This is achieved by removing non-essential fields and optional fields and putting
them in the extension headers that are placed after the IPv6 header. This technique
also simplifies the router’s work. The IPv4 header is 24 bytes large, 8 used for
addresses and 16 for twelve additional fields. The new IPv6 header is 40 bytes large,
32 used for the addresses and 8 for 6 additional fields.
The header structure is shown in Figure 37, as we can see some fields are missing:
Extension Header capacities are used to provide these lost functions. In this way the
great part of the packets can go quickly through the network and just packets having
particular requests receive a different treatment by analyzing the extension header.
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A lot of the extension headers require an end-to-end functionality so they don’t need
intermediate router elaboration.
Figure 37: IPv6 header
The IPv6 header fields are:
• Version (4 bits): This field simply identifies the IP version number.
• Traffic class (8 bits): It looks like TOS in IPv4. It allows traffic
differentiation:
- (0-7) traffic with congestion control (like TCP),
- (8-15) traffic without congestion control.
• Flow label (20 bits): It allows the routers to recognize the packets that belong
to the same data flow. With flow we mean a sequence of packets that need
the same processing by the network. This field is useful for the QoS
management. Because the traffic is identified in the IPv6 header, support for
QoS can be achieved even when the packet payload is encrypted through
IPsec.
• Payload length (16 bits): This is an unsigned integer that specifies the length
of the payload measured in bytes. The payload can be up to 64k in size in
standard mode, or larger with a "jumbo payload" option (payload==0).
• Next header (8 bits): The optional information are placed in separated headers
that are put between IPv6 header and the upper level’s header. IPv6 can
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easily be extended for new features by adding extension headers after the
IPv6 header. Unlike options in the IPv4 header, which can only support 40
bytes of options, the size of IPv6 extension headers is only constrained by the
size of the IPv6 packet. Some possible value are the following:
- 0: Hop-by-Hop Option Header
- 44: Fragment Header
- 50: Encapsulation Security Payload (IPsec)
- 51: Authentication Header (IPsec)
- 60: Destination Header
- 135: Mobility Header
- 59: No text Header
As we can see, IPv6 supports IPsec. This provides a standards-based solution
for network security needs and promotes interoperability between different
IPv6 implementations.
• Hop limit (8 bits): The field is decreased any time the packet crosses a router.
It has the same role of the IPv4 time to live
• Both source and destination addresses (128 bits each): see below
From IPv4’s header are removed:
• Checksum: The error check is assigned to the upper levels
• Fragment offset: The fragment management in assigned to optional headers.
Only the sender host can fragment the packets.
3.5.1.2 Addressing
IPv6 has 128-bit (16-byte) source and destination IP addresses. IPv4 provides
232≈4,3*109 addresses and considering that the number of people is 7*109 we
understand that a larger address space is needful. With IPv6 we can use
2128≈3,4*1038, they say that with IPv6 every grain of sand will have an IP address.
This great number of addresses is useful to exploit the hierarchical organization. The
IPv4 address penury has forced to use NAT (Network Address Translator) to map a
number of private addresses to a single public one. This affects the network’s
security level and hinders the diffusion of client-service applications. The IPv6
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address syntax requires eight blocks of four hexadecimal numbers (each one made of
four bits). It allows autoconfiguration (plug and play). Usually IPv6 has a 64 bit
subnet prefix. There are address subspaces allocated to Unicast and Multicast
addresses. It also introduces the concept of anycast (the closer node) useful for the
mobility communications. It is syntactically indistinguishable from the Unicast
address.
3.5.1.2.1 Unicast Addresses
It is a single interface identifier. A packet sent to a unicast address is consigned to
the interface identified by this address. There are different unicast addresses
classified by the scope:
• Link-Local address: It is visible only in the site. It is automatically configured
by the MAC address (EUI-64 standard)
• Site-Local address: It is visible only in the link.
• Aggregatable Global Unicast Address: It is a public address so it is visible in
the entire network. They are hierarchically organized to facilitate the address
aggregation.
There also some special use unicast addresses:
• Unspecified Address: (0:0:0:0:0:0:0:0) It is not allocated to a node but
represents a virtual interface. It means the absence of address. It is useful
during the address negotiation.
• Loopback Address: (0:0:0:0:0:0:0:1) It is used by a node that sends an IPv6
packet to himself. It is useful for diagnostic purposes. It can’t be allocated to
a node but represents a node’s virtual interface.
• IPv6 Address with Embedded IPv4 Address: It is an IPv6 address that hides
an IPv4 address. There are two type of IPv6 addresses that hide an IPv4
address:
- IPv4-compatible IPv6 address: It is assigned to a host that has a double
stack (IPv4 and IPv6). It is used during the transition and allows the
creation of an automatic tunnel IPv6 on IPv4.
- IPv4-mapped IP6 address: It is used to communicate with only IPv4
address nodes that don’t support IPv6 address.
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3.5.1.2.2 Anycast Addresses
It is a set of interface identifiers belonging to different nodes. A packet sent to an
anycast address is consigned to one of these interfaces, typically the closest one. It is
syntactically indistinguishable from the Unicast address. The node that has an
anycast address knows his nature (upper layers). It is useful to identify a set of
routers that provide the same service.
• A router can’t uses can anycast address like source address
• Only a router can have an anycast address so it can not be assigned to a host.
3.5.1.2.3 Multicast Address
It is a set of interface identifiers belonging to different nodes. A packet sent to a
multicast address is consigned to all these interfaces. A node can belonging to more
than one multicast group. It can have a different scope (link, site, global...).
• Multicast addresses can’t be used like source addresses
• They can’t appear in routing protocol headers
3.5.1.3 Mobile in IPv6
Terminal mobility in the next future will reach the same proportions as mobile
telephone ones. We can distinguish between terminal portability and mobility. In the
first case the user can connect himself from different access points but doesn’t need
to remain connected during the movement. This is possible also with IPv4 standard
but the terminal needs a different IP address every time and the configuration
procedure may be quite complex. It is simplified with the IPv6 autoconfiguration
procedure.
The mobility ensures the communication during the terminal movement. The mobile
node needs a different IP address every time it arrives at a new link, but this doesn’t
assure the transport level connection because by modifying the address the TCP
session may fall down. To assure the communication, a different protocol is useful.
A host needs more than one address, in base to the visited network:
It needs the following addresses:
• a “permanent address” independent of the access point,
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• a “variable address” to assure the achievement anytime, independently from
the visited network.
The permanent address, called home address, is a static address, registered in the
DNS home network and so globally reachable from Internet. When the terminal
connects to a network different from the home one (foreign network) it earns a new
address called care-of-address. The mobility matter can be solved by managing the
appropriate use of the home address or the care-of-address one, according to the
context. A home network router, called home agent, acts for the mobile host
connected to a foreign network. It will send the messages to the mobile host by using
a memory called binding cache in which it maintains the association between the
home address and the care-of-address. Every time the host changes network it earns a
new care-of-address called primary care-of-address and communicates it to the home
agent by a Binding Update message. The Binding Update message is also sent to
nodes that have communicated with the mobile host. All the addresses receiving the
Binding Update message are stored in a structure created by the mobile terminal
called Binding Update List. Previous care-of-addresses are maintained to allow the
host to receive packets sent to the old care-of-address. The care-of-address and home
address link is called binding. Figure 38 shows a possible IPv6 mobile situation.
There are three possible situations:
1. Mobile node is connected to the home network: the packets are routed to the
home network default gateway
2. Mobile node is connected to a foreign network: The packets are sent to the home
agent who knows the mobile host primary care-of-address thanks to the Binding
Update messages and the binding cache. The home agent will send the messages
to the mobile terminal by tunnel. When the message reaches the mobile terminal
it learns the sender address and sends it a Binding Update. The sender learns the
new primary care-of-address and associates it to the mobile terminal home
address in the Binding Cache; in this way it will send the messages directly to the
primary care-of-address.
3. Mobile node isn’t connected to any network. Home network default gateway
sends a message to the sender to notify the mobile host that it is not achievable.
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Figure 38: IPv6 mobile
3.5.2 Differences between IPv4 and IPv6
In conclusion some of the key differences between IPv4 and IPv6 are highlighted:
• Source and destination addresses are 128 bits (16 bytes) in length.
• IPsec support is required.
• Packet flow identification for QoS handling by routers is included in the IPv6
header using the Flow Label field.
• Fragmentation is not done by routers, only by the sending host.
• Header does not include a checksum.
• All optional data is moved to IPv6 extension headers.
• ARP Request frames are replaced with multicast Neighbor Solicitation
messages.
• IGMP is replaced with Multicast Listener Discovery (MLD) messages.
• ICMP Router Discovery is replaced with ICMPv6 Router Solicitation and
Router Advertisement messages and is required.
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122
CCHHAAPPTTEERR FFOOUURR
4. EDGE NETWORK PROPOSAL FOR UMTS,
WiMAX AND DVB-H, AND CONVERGENT
ARCHITECTURE
4.1 Introduction
A convergent architecture between WiMAX, DVB-H and UMTS is the scope of this
thesis. In order to do this, in the Chapter two I have defined a unique architectural
block model for the different Access Networks, grouping common logical
functionalities. As the analysis in the Chapter three has shown, IMS is the best
choice for the Core Network of a convergent architecture because it allows the
convergence between different technologies. The IMS is based on the SIP protocol
for control signaling, and on other standardized IETF protocols (like RTP in case of
VoIP) for user application traffic. It is an access independent platform providing
services in a standardized way. So, in this section the way to connect each different
Access Network to IMS Core Network will be studied. Since there is a gap between the Access Network and the Core Network, an Edge
Network filling this gap is needed. According to this point of view, the Edge
Network could be considered as an extension of the Core Network.
In Figure 39 is shown a logical representation of the Edge Network.
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123
Figure 39: Schematic Representation of a Network
In the next paragraph I am going to explain the reasons of why the Edge Network is
required for each access technology. Moreover, I am going to describe which
features the Edge Network must provide to allow the connection between any Access
Network and the IMS Core Network. As the Access Networks taken into
consideration are UMTS Release 6, Mobile WiMAX and DVB-H, the Edge Network
devices refer only to these three technologies [§ 4.3-4.6].
Finally, I am going to expose the way the Edge Network allows a complete
convergent network architecture clarifying all the stages it goes through. The
resulting proposal is a whole integrated network where the different access
technologies are not separated.
4.2 What is the Edge Network?
The Access Networks analyzed previous [Chapter 2] can not be connected directly to
the IMS Core Network. This issue is caused because every technology has different
requirements. In particular, regarding to the IMS features the major problems are that
DVB-H is broadcast and it does not provide session establishing. IMS is based on the
SIP protocol for signaling and consequently it is based on a Client/Server model.
This means that every transaction is originated from a node of the network able to act
as client and it is directed to another entity that involves from server. These
conditions can not be achieved in a broadcast network.
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124
Furthermore, UMTS Access Network elements do not communicate by IP but by
ATM protocol. In the UMTS Release 6 the interaction with the IMS is standardized
but the UTRAN is still not IP-based.
Finally, SIP protocol is not mandatory for WiMAX standard, even though it is all-IP.
The integration between WiMAX and IMS is supposed to be possible, but it is not
standardized yet.
So, assuming the Access Network represented as the unique architectural access
model described in the section 2.6, the difficulties are mainly concentrated on the
Decision&Control Module and IMS link. It is shown in Figure 40.
Keeping in mind which are the main Decision&Control Module functionalities (all
the functions in the Access Network management), in this case it is important to
underline that it is responsible for the interface to the Core Network. The physical
devices included in this module are the UMTS RNC, the WiMAX ASN-GW and the
DVB-H IP-Encapsulator.
For the reasons listed above, there is a ‘connection failure’ between these entities and
the first point of contact in the IMS (that is the P-CSCF in the signaling plane). Thus,
some additional elements need to be considered to allow the connection between the
Access Networks and Core Network. This set of elements is called Edge Network in
my proposal.
The Edge Network links the Decision&Control Module of the Access Network to the
IMS Core Network, as it is illustrated in the Figure 41.
Figure 40: Connection Failure
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125
Figure 41: The Introduction of the Edge Network
The main features that an Edge Network must provide are related to the connectivity
of different access network to IMS Core Network. The SIP protocol is mandatory for
IMS then the Edge Network uses SIP protocol to allow establishing sessions.
Furthermore, the Core Network is All-IP and the Edge Network has to ‘translate’ the
transfer mode wherever the IP protocol is not used. Therefore the Edge Network is
responsible of IP addresses management.
Finally, it holds DataBases and servers for AAA. It also holds the media resources
for the DVB-H network.
The different technologies of the access networks and the devices needed to the core
network connection will be presented in the next paragraphs.
The Access and Edge Network protocol stacks together with the Core Network
boundary devices ones have been object of my study.
In fact the way of connecting the Access and IMS Core Networks has been deeply
influenced by the protocol analysis. As the interaction between IMS and the Access
Networks is not standardized yet, the protocol analysis has been helpful to define
some interconnecting devices.
For each device will be highlighted the most important protocols, and the analysis
will be concluded once the protocol stacks reaches the common neighbouring device
layer. Consequently, it is possible to identify the type of header that each element has
to analyze or to add to the packet that it receives.
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126
4.3 The Edge Network for UMTS Release 6 (PS Domain)
Figure 42: Edge Network in UMTS
With the ever increasing penetration of IP technologies and the tremendous growth
in wireless data traffic, the wireless industry is evolving the mobile Core Networks
towards all IP technology. The 3rd Generation Partnership Project (3GPP) is
specifying an IP Multimedia Sub-system (IMS) in UMTS Release 6, which is adjunct
to the UMTS Packet Switched (PS) GPRS CN. This IP-based network will allow
mobile operators to provide commonly used Internet applications to wireless user.
The UMTS IMS uses the Internet Engineering Task Force (IETF) defined text-based
Session Initiation Protocol (SIP), to control a wide range of anticipated IP-based
services offering new services such as multimedia calls, chat, presence services.
Hence, due to the fact that I want to reach one Core Network independent of the kind
of Access Network the UMTS PS domain is moved out the Core Network. In this
way, the PS domain is placed in the Edge Network and it allows the UTRAN to be
connected to the only IMS Core Network. It can be seen in Figure 42.
Indeed, the UMTS PS domain (or GPRS) Support Nodes (GSN) are the Gateway
GSN (GGSN) and the Serving GSN (SGSN). They constitute the interface between
the Radio System and the Fixed Networks for packet switched services (IMS).
The GSN performs all necessary functions in order to handle the packet transmission
to and from the mobile stations.
Thus, the PS domain refers to the set of all the CN entities offering “PS type of
connection” for user traffic as well as the entities supporting the related signaling. A
“PS type of connection” transports the user information using autonomous
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127
concatenation of bits called packets: each packet can be routed independently from
the previous one. While the IM subsystem (IMS) comprises all the CN elements for
provision of IP multimedia services comprising audio, video, text, chat, etc. and a
combination of them delivered over the PS-Domain.
In the PS domain is performed the Location Register. The Location Register is a
function that stores information regarding where the mobile station is located. This is
the function that enables communication to a mobile station.
The Location Register is handled by four different entities:
• HLR
• VLR
• SGSN
• GGSN
4.3.1 Serving GPRS Support Node (SGSN)
This node coordinates:
• virtual channels (PDP Context) establishing / closing
• virtual channel maintenance
It manages:
• user traffic
• user mobility
SGSN also performs the functionalities related to security.
UMTS cells are grouped in Routing Areas, which are contained in Location Areas.
When a mobile that is registered in a SGSN moves from one Routing Area to
another, it informs the SGSN in order to update its location.
Thus, the location register function in the SGSN stores two types of subscriber data
needed to handle originating and terminating packet data transfer (in order to manage
user mobility and traffic):
• Subscription Information
- The IMSI
- One or more temporary identities
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128
- Zero or more IP addresses (or PDP addresses)
• Location Information
- The cell or the routing area where the MS is registered, depending on
the operating mode and state of the MS
- The VLR number of the associated VLR (if the Gs interface to the
MSC/VLR in the CS-Domain is implemented)
- The GGSN address of each GGSN for which an IP virtual channel ism
active (an active PDP context exists).
4.3.2 Gateway GPRS Support Node (GGSN)
The GGSN is the node that interfaces the external PS-Domain networks, as Internet,
and IMS.
When a mobile wants to transfer some packet data, it asks to the SGSN to establish a
PDP Context. The SGSN, together with the GGSN, creates a virtual channel which
connects the UE to the desired network. The GGSN is the gateway which gives the
access to the external network. So the GGSN stores user data needed to manage the
user traffic.
The location register function in the GGSN stores subscriber data received from the
HLR and the SGSN. There are two types of subscriber data needed to handle
originating and terminating packet data transfer:
• Subscription Information
- The IMSI
- Zero or more IP addresses (PDP addresses)
• Location Information
- The SGSN address for the SGSN where the MS is registered
4.3.3 Data Bases
4.3.3.1 Home Subscriber Server (HSS)
The HSS is the master database for a given user. It is the entity containing the
subscription-related information to support the network entities actually handling
calls/sessions.
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129
A Home Network may contain one or several HSSs: it depends on the number of
mobile subscribers, on the capacity of the equipment and on the organization of the
network.
For example, the HSS provides support to the call control servers in order to
complete the routing/roaming procedures by solving authentication, authorization,
naming/addressing resolution, location dependencies, etc.
The HSS is responsible for holding the following user-related information:
• User Identification, Numbering and addressing information
• User Security information: Network access control information for
authentication and authorization
• User Location information at inter-system level: the HSS supports the user
registration, and stores inter-system location information, etc.
• User profile information.
The HSS also generates User Security information for mutual authentication,
communication integrity check and ciphering.
Based on this information, the HSS also is responsible for supporting the call control
and session management entities of the different Domains and Subsystems of the
operator as shown in Figure 43.
Rp Gr Gc
Location information
Subscription information
HSS
Cx D C
GUP Server CSCF GGSN SGSN MSC Server GMSC Server
Figure 43: Example of a Generic HSS structure and basic interfaces
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130
The HSS may integrate heterogeneous information, and enable enhanced features in
the core network to be offered to the application & services domain, at the same time
hiding the heterogeneity.
The HSS consists of the following functionalities:
• IP multimedia functionality to provide support to control functions of the IM
subsystem such as the CSCF. It is needed to enable subscriber usage of the
IM CN subsystem services. This IP multimedia functionality is independent
of the access network used to access the IM CN subsystem
• The subset of the HLR/AUC functionality required by the PS Domain
• The subset of the HLR/AUC functionality required by the CS Domain, if it is
desired to enable subscriber access to the CS Domain or to support roaming
to legacy GSM/UMTS CS Domain networks
The HSS is considered as GUP Data Repository for IM CN Subsystem user related
data. The RAF (Repository Access Function) provides the Rp reference point.
4.3.3.2 The Home Location Register (HLR)
The HLR can be considered a subset of the HSS that has the following functions:
• The function required to provide support to PS Domain entities such as the
SGSN and GGSN, through the Gr and Gc interfaces and the 3GPP AAA
Server for the I-WLAN through the D'/Gr' interface. It is needed to enable
subscriber access to the PS Domain services
• The function required to provide support to CS Domain entities such as the
MSC/MSC server and GMSC/GMSC server, through the C and D interfaces.
It is needed to enable subscriber access to the CS Domain services and to
support roaming to legacy GSM/UMTS CS Domain networks.
4.3.3.3 The Authentication Centre (AuC)
The AuC can be considered a subset of the HSS that holds the following
functionality for the CS Domain and PS Domain:
• The AuC is associated with an HLR and stores an identity key for each
mobile subscriber registered with the associated HLR. This key is used to
generate security data for each mobile subscriber:
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131
- data which are used for mutual authentication of the International
Mobile Subscriber Identity (IMSI) and the network
- a key used to check the integrity of the communication over the radio
path between the mobile station and the network
- a key used to cipher communication over the radio path between the
mobile station and the network.
• The AuC communicates only with its associated HLR over a non-
standardized interface denoted the H-interface. The HLR requests the data
needed for authentication and ciphering from the AuC via the H-interface,
stores them and delivers them to the VLR and SGSN which need them to
perform the security functions for a mobile station.
4.3.3.4 HSS logical functions
• Mobility Management: This function supports the user mobility through CS
Domain, PS Domain and IM CN subsystem
• Call and/or session establishment support: The HSS supports the call and/or
session establishment procedures in CS Domain, PS Domain and IM CN
subsystem. For terminating traffic, it provides information on which call
and/or session control entity currently hosts the user
• User security information generation: The HSS generates user authentication,
integrity and ciphering data for the CS and PS Domains and for the IM CN
subsystem user security support. The HSS supports the authentication
procedures to access CS Domain, PS Domain and IM CN subsystem services
by storing the generated data for authentication, integrity and ciphering and
by providing this data to the appropriate entity in the CN (i.e. MSC/VLR,
SGSN, 3GPP AAA Server or CSCF)
• User identification handling: The HSS provides the appropriate relations
among all the identifiers uniquely determining the user in the system: CS
Domain, PS Domain and IM CN subsystem (e.g. IMSI and MSISDNs for CS
Domain; IMSI, MSISDNs and IP addresses for PS Domain, private identity
and public identities for IM CN subsystem)
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132
• Access authorization: The HSS authorizes the user for mobile access when
requested by the MSC/VLR, SGSN, 3GPP AAA Server or CSCF, by
checking that the user is allowed to roam to that visited network
• Service authorization support: The HSS provides basic authorization for MT
call/session establishment and service invocation. Besides that the HSS
updates the appropriate serving entities (i.e., MSC/VLR, SGSN, 3GPP AAA
Server, CSCF) with the relevant information related to the services to be
provided to the user
• Service Provisioning Support: The HSS provides access to the service profile
data for use within the CS Domain, PS Domain and/or IM CN subsystem.
Application Services and CAMEL Services Support
• The HSS communicates with the SIP Application Server and the OSA-SCS to
support Application Services in the IM CN subsystem. It communicates with
the IM-SSF to support the CAMEL Services related to the IM CN subsystem.
It communicates with the gsmSCF to support CAMEL Services in the CS
Domain and PS Domain.
• GUP Data Repository: The HSS supports the storage of IM CN Subsystem
user related data, and provides access to these data through the Rp reference
point.
4.3.4 The Equipment Identity Register (EIR)
The Equipment Identity Register (EIR) in the GSM system is the logical entity which
is responsible for storing in the network the International Mobile Equipment
Identities (IMEIs), used in the GSM system. The equipment is classified as "white
listed", "grey listed", "black listed" or it may be unknown.
This functional entity contains one or several databases which store(s) the IMEIs
used in the GSM system. The mobile equipment may be classified as "white listed",
"grey listed" and "black listed" and therefore may be stored in three separate lists. An
IMEI may also be unknown to the EIR. An EIR shall as a minimum contain a "white
list" (Equipment classified as "white listed").
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4.3.5 Interfaces
4.3.5.1 Interface between SGSN and RNS (Iu_PS-interface)
The RNS-SGSN interface is used to carry information concerning:
• Packet data transmission
• Mobility management
4.3.5.2 Interface between SGSN and HLR (Gr-interface)
This interface is used to exchange the data related to the location of the mobile
station and to the management of the subscriber. The main service provided to the
mobile subscriber is the capability to transfer packet data within the whole service
area. The SGSN informs the HLR of the location of a mobile station managed by the
latter. The HLR sends to the SGSN all the data needed to support the service to the
mobile subscriber. Exchanges of data may occur when the mobile subscriber requires
a particular service, when he wants to change some data attached to his subscription
or when some parameters of the subscription are modified by administrative means.
Signaling on this interface uses the Mobile Application Part (MAP), which in turn
uses the services of Transaction Capabilities (TCAP).
4.3.5.3 Interface between SGSN and GGSN (Gn- and Gp-interface)
These interfaces are used to support mobility between the SGSN and GGSN. The Gn
interface is used when GGSN and SGSN are located inside one PLMN. The Gp-
interface is used if GGSN and SGSN are located in different PLMNs. The Gn/Gp
interface also includes a part which allows SGSNs to communicate subscriber and
user data, when changing SGSN. Signaling on this interface uses the User Datagram
Protocol, UDP/IP.
4.3.5.4 Signaling Path between GGSN and HLR (Gc-interface)
This optional signaling path may be used by the GGSN to retrieve information about
the location and supported services for the mobile subscriber, in order to activate a
packet data network address. There are two alternative ways to implement this
signaling path:
• if an SS7 interface is implemented in the GGSN, signaling between the
GGSN and the HLR uses the Mobile Application Part (MAP), which in turn
uses the services of Transaction Capabilities (TCAP)
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• if there is no SS7 interface in the GGSN, any GSN in the same PLMN and
which has an SS7 interface installed can be used as a GTP to MAP protocol
converter, thus forming a signaling path between the GGSN and the HLR.
4.3.5.5 Interface between SGSN and EIR (Gf-interface)
This interface is used between SGSN and EIR to exchange data, in order for the EIR
to be able to verify the status of the IMEI retrieved from the Mobile Station.
Signaling on this interface uses the Mobile Application Part (MAP), which in turn
uses the services of Transaction Capabilities (TCAP).
4.3.6 Protocol Analysis and Definition
Figure 44: UMTS’s Device Protocols
In UMTS each access network element is a transit node for the UE and the IMS CN.
These transit nodes perform all the functionalities of the physical layer and of the
data link layer. Figure 44 shows that the transfer mode used by Node B and the RNC
is ATM. Because of this, the UTRAN cannot connect directly to the IMS CN but an
Edge Network must be taken into consideration. In the Edge Network the transfer
mode uses the IP protocol. The SGSN is the first element of the Edge Network and it
makes a ‘protocol conversion’. In fact, it has a dual protocol stack: on the access
network side it communicates with the RNC by the ATM protocol, while on the
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other side it is connected to the GGSN by the Gn interface which supports the UDP
protocol (on the transport layer). The GGSN is the gateway to the IMS core network
and it establishes a virtual channel (PDP Context) between the UE and the IMS core
network. The GGSN supports UDP/TCP on the transport layer and SIP on the
application layer (HTTP/FTP are also supported here).
All of these interfaces and connections are standardized in the UMTS
Release 6.
4.4 The Edge Network for WiMAX
WiMAX is a broadband access technology, and so it should have no problems
connecting to IMS as DSL or WiFi technology are also both able to connect without
problems. Nowadays the WiMAX Network Group (NWG) is still working on the
WiMAX architecture, and the way to connect to IMS is an open issue. It is a
requirement for the standardized architecture but it is has not yet been defined.
WiMAX should be used as transparent IP-bearer connecting the IMS-Client with the
IMS-Network.
In this context it is reasonable to suppose a Wireless Access Gateway (WAG) in the
CSN with SIP protocol on the session layer to interface to the IMS Network. The
WAG will also provide Policy Enforcement, which is a functionality implemented to
ensure that packets coming from or going to the WLAN Access Network are allowed
based on unencrypted data within the packets (e.g. source and destination IP address
and port number).
So, the Edge Network for the WiMAX can be individuated in the NSP (Network
Service Provider) which is implemented by one or more CSN (Connectivity Service
Network). This infrastructure is shown in Figure 45.
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Figure 45: Edge Network in WiMAX
The Network Service Provider (NSP) is a business entity that provides IP
connectivity and WiMAX services to WiMAX subscribers consistent with the
Service Level Agreement it establishes with WiMAX subscribers. Thus, the IMS
may be located in the private network of a network operator (NSP) or at a third-
party provider. The IMS could be connected directly to the provider’s backbone
network, connected via a specific gateway or via the internet.
Another element needed in the NSP to enable a Wireless Access Network to
connect to the IMS is an AAA Server.
4.4.1 CSN-WAG
Connectivity Service Network (CSN) is defined as a set of network functions that
provide IP connectivity services to the WiMAX subscriber(s). Thus, a CSN may
comprise network elements such as routers, AAA proxy/servers, user databases and
Interworking gateway devices. In order to allow the WiMAX Access Network
(ASN) to be linked to the IMS Core Network, the device held in the CSN is a WAG.
A CSN may provide the following functions:
• MS IP address and endpoint parameter allocation for user sessions
• IP address management (based on PoA management)
• Internet access, Connectivity to Internet, ASP and other PLMN sand
Corporate Networks
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• AAA proxy or server, User, equipment and services authentication,
authorization and accounting (AAA)
• Policy and Admission Control based on user subscription profiles
• ASN-CSN tunnelling support
• WiMAX subscriber billing and inter-operator settlement
• Inter-CSN tunnelling for roaming
• Location management between ASNs
• Inter-ASN mobility and roaming between ASNs
• WiMAX services such as location based services, connectivity for peer-to-
peer services, provisioning
• authorization and/or connectivity to IP multimedia services and facilities to
support lawful intercept services
• Policy & QoS management based on the SLA/contract with the user
Interworking gateway devices, such as those compliant with Communications
Assistance Law Enforcement Act (CALEA) procedures.
4.4.2 Interfaces
The interfaces involved in the Edge Network are called R3 and R5.
The R3 is the interface between the ASN and the CSN and it is responsible for
supporting AAA, policy enforcement and mobility management capabilities. It also
encompasses the bearer plane methods (e.g. tunnelling) to transfer IP data between
the ASN and the CSN.
On the other side, the R5 interface consists of a set of control plane and bearer plane
protocols for interworking between CSNs operated by either the home or visited
NSP. Thus, this interface can be considered as the interface between the Edge
Network and the IMS Core Network, i.e. between the CSN-WAG and the P-CSCF.
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4.4.3 Protocol Analysis and Definition
Figure 46: WiMAX’s Device Protocols
Figure 46 shows WiMAX’s device protocol stacks. The user equipment has SIP on
the session Layer, this protocol allows the user to establish a session with other users,
even when they access the network using different technologies. Between the
terminal and the BS the communication is at the Link Layer. The BS manages and
controls the MAC Layer, allocating channels (time and bandwidth) to different users.
On the other side, the terminal sends some essential information, like the CQI, to the
BS. This interface is the one standardized with the 802.16 standard. From this point
on, up to the CN the communication is all IP. The different control layers highlighted
in Figure 46 show how these devices manage traffic across the network. In order to
interface the CSN-WAG to the IMS CN we have decided to use SIP as the session
protocol in the WAG.
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4.5 The Edge Network for DVB-H
I have already said that DVB-H is a broadcasting technology so, without using
another technology to implement a return channel, the DVB-H network is
unidirectional.
A DVB-H unidirectional network together with another broadband network which
provides the return channel is standardized as IP Datacasting (IPDC). IPDC is
needed to achieve any interaction with the user. Indeed, IPDC is an end-to-end DVB
system for services over mobile terminals comprising both a broadcast path through
the DVB-H technology and a bidirectional telecommunication part. This
mobile/cellular technology is standardized to be UMTS/GPRS, but I believe that also
WiMAX will be taken into account thanks to the convergence towards one Core
Network. The IPDC convergence means convergence on transport and applications
layer. Instead, my objective is an improved convergent architecture where DVB-H
system is able to connect to IMS Core Network. In this way the return channel could
not be just the UMTS one but could be chosen dynamically between the best
available access technologies.
However, there is no standard regarding how to connect a DVB-H network to the
IMS. So, my effort was to analyzing how to connect the two networks and how to
realize a whole convergent network involving the DVB-H system.
Some of the advantages that this convergence would give are the following:
• Services synchronization: a synchronization of broadcast services with
additional no-broadcast services, which eventually uses interactivity, is
possible, i.e. interactive advertisement during a football match.
• Presence: it is possible to create presence services related to the broadcast
television programs. In this way, it allows the network owners to know who-
sees-what. This information is very important for the advertising market and
program scheduling (“palinsesto”).
• Service reuse: IMS is a horizontal architecture, so it enables the reuse of
services created by third parties.
• Unique network: allows the user to connect to a unique network and above all
with just one service provider.
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At the top of what is considered as the DVB-H Access Network, i.e. before the IP
Encapsulator, I must suppose a server which I call Content Provider. This Content
Provider server is placed into the Edge Network, as Figure 47 can describe.
Figure 47: Edge Network in DVB-H
This device is the one which interfaces with IMS Core Network. In order to make
this integration possible the Content Provider server must hold the SIP protocol on
the session layer, because IMS is based on SIP. This protocol allows the Content
Provider to establish sessions with other elements of the network. In this way the
Content Provider is able to share contents with other servers. This is the reason of
why I chose to integrate the server which allows the connection to IMS with the one
which stores the multimedia contents to be sent by the broadcast network. It is
possible to perform services with the other technologies reusing the same contents
sent by the broadcast network i.e. TV on-demand by UMTS or WiMAX. SIP also
gives the users (connected to the network via another access network) the possibility
to send some kind of content (i.e. SMS, MMS, etc.) to the Content Provider, which
will transmit them in broadcast together with the shows.
Another advantage of the SIP protocol inside the Content Provider server is that
multimedia can be entered just by establishing a session by a remote server. For that
reason the Content Provisioning shown in Figure 47 is not mandatory inside the Edge
Network.
I assume that the Content Provider server is directly connected to the IMS S-CSCF
via the ISC interface. This interface is standardized to connect the S-CSCF to a SIP
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Application Server or to a third party Application Server. There is not need to
connect the Content Provider to the S-CSCF via P-SCSF because it is not a mobile
IMS terminal. However, the Content Provider can be seen by the IMS network as a
user terminal, i.e. IMS client.
It is important to underline that the Content Provider server performs broadcast
transmission on the air interface side, it is able to establish sessions (bidirectional
transmission) on the other side (towards the Core Network).
4.5.1 Protocol Analysis and Definition
The connection of AN DVB-H with the CN IMS was probably the most difficult
point in the research of an architecture that permits the convergence between the
examined technologies. The difficulty comes from the impossibility to create a
session on a Broadcast network. This difficulty was then solved by choosing to
establish the sessions between the Content Provider and the other network terminals.
Figure 48: DVB-H Protocol Stack
Figure 48 shows the typical protocol stack of this technology.
On the Application layer the FLUTE (File Delivery over Unidirectional Transport)
protocol is to be considered if no return channel is available for the DVB-H system.
It is a protocol for the unidirectional delivery of files over the Internet, which is
particularly suited to multicast networks. It is based on a mechanism for signaling
and mapping the properties of files to concepts of ALC (Asynchronous Layered
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Coding) in a way that allows receivers to assign those parameters for received
objects. ALC is a protocol designed for delivery of arbitrary binary objects. It is
especially suitable for massively scalable, unidirectional, multicast or broadcast
distribution.
Figure 49: DVB-H’s Device Protocols
MUX and Modulator work at the Data Link layer. The first, MUX, is for
multiplexing the different IP streams that come from the IP Encapsulator in a unique
Transport Stream and it also adds the SI/PSI and MPE metadata. From this point, the
data is no longer managed at the IP level. The second, the Modulator, is for sending
TS on-air. The Content Provider is an element of the Edge Network that acts as the
interface for the SIP-IMS world and it is the only element that manages data at the
Application layer. Therefore, it is the only one that works in the grey level pictured
in Figure 48. Figure 49 represents this scenario. As you follow the straight path of
information across the broadcast network, the different elements are continually
working at a lower layer than the previous device.
I would like the highlight the necessity of the SIP Protocol in the application layer of
the Content Provider server. This is an indispensable requirement that allows the
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connection to the IMS network, which then offers the benefits and possibilities that
have already been explained in this section.
4.6 CONVERGENT ARCHITECTURE PROPOSAL
All the devices needed to connect the different Access Networks to the IMS Core
Network are now identified. I have called the set of these elements the Edge
Network. The resulting architecture scheme of the whole convergent network is the
one shown in Figure 50.
Figure 50: Convergent Architecture Scheme
Hence, the introduction of the Edge Network consents to realize a convergent
architecture. A convergent architecture is an implementation of technologies aimed
to optimize the network architecture in order to transport video, voice and data on a
single support. The main feature this architecture must provide is access
independency, that is services must be available over different access technologies.
So, a user does not have to take care of which kind of terminal has to be used for a
specific service. In fact there is one network architecture for accommodating all
services. That allows to provide and to require optimized Quality of Service. Such
architecture also consents easier interworking with the Internet.
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As the IP protocol is the reference point for any kind of network transporting video,
voice and data, a convergent architecture is possible assuming an all-IP based
network.
Thus, in my study I have supposed an all-IP backbone and the motivations are
summarized next:
• Enables rich communications combining multiple media or services;
• New IP-based services, easier & faster service creation and execution;
• Smooth evolution from today’s networks and standards:
- Cost efficiency, evolution for current solutions;
• Openness: both specifications and (distributed) architecture.
In order to guarantee one network architecture and access independency with the
existent technologies, the Edge Network is needed. It allows to consider one generic
Access Network which is composed of more access technologies, i.e. UMTS,
WiMAX and DVB-H in this case. Therefore, the Access Network could be any kind
of network because the role of the Edge Network is to fill the gap between Access
and Core Network linking them together.
Figure 51 represents the result of integration between different Access Networks and
one Core Network which is my idea of the convergent network.
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Figure 51: Convergent Architecture
In this figure the Access, Edge and Core Network are highlighted. The Access
Network is intended as unique architectural block model [§ 2.6] which physically
holds the UMTS UTRAN, WiMAX and DVB-H systems.
The Core Network is the basic communications platform IMS which offers easy
integration with other IP protocols and applications and seamless service offering
over various access networks. Thanks to the IMS features (due to the SIP Protocol),
such Convergent Network provides:
• Personalized services aware of desired communication capabilities and
preferences,
• Straightforward integration of voice, images, video and other interactive
communications services,
• Web like service development approach
• Enhanced service inter-working with a client in terminal and a server in
network.
The Edge Network can be considered as the whole of what is needed to connect the
Access Network to the Core Network in order to achieve the convergence. In the
Paragraph 4.2 the Edge Network was introduced, explaining what it is meant for and
which functionality it must hold. While in the successive paragraphs [§ 4.3-4.5] the
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physical elements composing the Edge Network for each technology were described.
In Figure 51 is evident that some elements that were in the Edge Network of each
technology are now not present in the Edge Network of my idea of convergent
architecture. This modification is due to some assessment I have done and it is
another step towards the complete convergence. Everything will be explained in the
next paragraph which gives details about the Edge Network evolutions.
4.6.1 Edge Network Evolutions
The result of my study is a proposal of an Edge Network which can be considered as
the key for achieving a fully convergent architecture. This definition passes through
various phases and the finally convergence, achieved via Edge Network, is
developed. This paragraph is intended to make clear all the stage I have followed to
reach the architecture illustrated in this section.
The starting point of my work is that, currently, there are three distinct Access
Networks standalone. In this environment every access technology needs its own
user terminal, as Figure 52 depicts. Anyway, every technology user equipment needs
to act as IMS Client but the DVB-H Receiver. In the case of DVB-H system the
Content Provider Server is seen as client by the IMS network [§ 4.5]. The main
prerogative an IMS Client must hold is the presence of SIP protocol on the
application layer.
The WiMAX architecture provides unified support of functionality needed in several
usage scenarios ranging from fixed, nomadic, portable, simple mobility to fully
mobile subscribers. In this thesis the Mobile WiMAX is taken into account so the
user terminal is assumed to be a mobile terminal and it is called MS (Mobile
Subscriber). Since the WiMAX network acts as the IP-connectivity Access Network
to IMS based services for the transport of IMS signaling and bearer traffic, the MS
has to run IMS client software to be compatible with IMS at the network side. This
requires Gm interface and Um (3GPP2)10 interfaces between WiMAX client and P-
CSCF, i.e. on the R3/R5 reference point.
10 Gm is the reference point supporting the communication between UE and IMS CN (CSCF), Um is the interface between the Mobile Station and the Base Station
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Summing up, the WiMAX network enables wireless users to access all IP
Multimedia Subsystem (IMS) based services then WiMAX subscribers are able to
establish a WiMAX connection, perform P-CSCF discovery and register to IMS as
defined in the IMS registration procedure.
UMTS UEs are also able to access IMS services, as access to IMS relies on the
presence of either e USIM or an ISIM application in the Universal Integrated Circuit
Card inside the terminal.
Figure 52: Access Networks Standalone
In the first step towards the convergence, the Edge Network is the union of the three
Edge Networks introduced for each Access Network. It is just an interface to the
Core Network. However, it groups all the physical elements and features needed for
the connection to the Core Network by each technology independently. There is not
an interaction between each different technology and the whole network is the result
of three Access Networks put together.
Moreover, some of the functions performed in the Edge Network are the same
performed also by other devices in the IMS Core Network.
Since this issues causes redundancy information and, above all, no scalability a
further effort needs to be made.
Thus, the last evolution is a unique network where the Edge Network is composed
just by the elements strictly needed for connecting each technology to the Core
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Network. In order to optimize the complete convergent network, all the common and
duplicated functions are shifted into the Core Network.
The convergent architecture derived from my considerations is the one in Figure 53.
Figure 53: Convergent Architecture
Some elements that were in the Edge Network of each technology are not present in
the Edge Network of this model of convergent architecture. That is because I have
moved the functionalities those elements perform toward other devices, when I have
found some correspondence between the functions in the Edge Network and the ones
in the IMS network. For example, the HLR in UMTS is a database which stores a
subset of information stored in the HSS within the IMS. So, the same user
information is duplicated and redundant. All the functionalities related to the
Location Register can be centralized into the Core Network, as the IMS devices can
offer them. In addition, the AAA services needed in the WiMAX connection are
performed also by the S-CSCF. I have also centralized the AAA functionalities
performed by WiMAX or UMTS servers and databases dedicated for the same
purpose. These functionalities are performed by IMS CSCF devices which use Core
Network databases. At last, I have chosen to do not put DVB-H content provisioning
inside the Edge Network, it could be in a remote location and it could load contents
by establishing a peer-to-peer SIP session.
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Hence, I have decided to centralize some Edge Network functionalities inside the
IMS Core Network in order to achieve a convergent architecture. In this way, the
Edge Network results to be the bridge between the Access Networks and the Core
Network but not only. Its importance grows up thanks to the fact that it allows a
complete integration between different Access Networks. Similar functions
performed by different technologies (although performed in different ways) are
supposed to be joint together. This means that an interaction between various
elements inside the Edge Network is possible. It follows that the whole network can
be considered as one. Something very important to be taken into account is that the
convergent architecture considered up to now is simple to realize because it holds all
the physical and functional elements currently present, although devices composing
the Edge Network must satisfy some mandatory requirements not needed before
now, regarding each technology. On the contrary, the new opportunity and possible
services introduced by such an architecture are highly developed. All the functions
already existing are able to be combined, due to the Edge Network.
A vertical handover between different technologies can be performed in order to
dynamically choose the best access technology with the respect of the service
requirements (QoS, required bandwidth, coverage area, etc...). For example, the
network can switch the transmission over either UMTS UTRAN or WiMAX Access
Network depending on which kind of service is to be offered.
The same consideration can be done about the User Equipment. Such a network can
be accessed by a unique user terminal regardless of the access technology used.
This User Equipment needs to hold an interchanging radio access interface which is
able to choose either an access mode or another. So the user does not have to care
about which technology is needed to be able to get a specific service. A requirement
the User Equipment must meet is the SIP protocol on the application layer of its
protocol stack. SIP allows session establishing between users, most of all between
users and Content Provider server [Appendix A].
Moreover, a user can access using various kind of terminals and be identified by his
own IP address regardless the kind of terminal. Thanks to the IMS service presence,
the network is able to understand which terminal he is using, then how much
bandwidth is allocable and which bit rate can be reached. Everything is possible
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because the HSS data bases are centralized and the user information are the same for
each devices of every technology.
Some significant functions, like Security, are performed by both Edge Network and
Core Network elements. However, in such case this redundant functionality is not a
bother but a further way to make the system more efficient and robust than the earlier
one. The idea of separate the network into more sections (Access Network, Edge
Network and Core Network) allows a modularization of some functionalities
management. To better understand this assertion, one can think of a protocol stack:
each layer is dedicated to perform some specific functions and it adds other
functionalities to the underlying one.
So, concerning the Handover function:
• it can be performed by the Access Network on a physical level,
• an Handover between different technologies can be considered inside the
Edge Network,
• it can be provided by the one Core Network on a user level.
All these consideration about a developing Edge Network are the basis for an
improved convergent architecture which is supposed to be reached in the future. In
fact, it is assumed that similar management policies for the same functions performed
by different technologies (which work in quite different ways by now) will be
accomplished in the upcoming time. This will enable to integrate various devices in
the Edge Network maintaining one element for different technologies.
4.6.2 Conclusions
In conclusion, it is important to say that a convergent architecture is to be considered
as a development existing networks instead of their revolution. The elements
composing this architecture are the same components as before with improved
features and additional functions, both multiprotocol and multilayer. The
implementation of multiprotocol equipments enables to access to different
technologies.
As a system oriented to the convergence needs a careful valuation of the existing
infrastructures and of the objective to be satisfied, my thesis was aimed to this. So,
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the result of my effort is a Convergent Architecture proposal which takes the
following advantages:
• Integrated and synchronized services are possible: the integration between
broadcast and multicast services is improved as soon as possible;
• All the functions in common between these three different access
technologies (UMTS, WiMAX and DVB-H) are centralized: the Access
Network can be considered as one;
• In the Edge Network are performed just the functionalities related to the need
of connecting different Access Networks: the whole network can develops
with the introduction of new access technologies without changing anything
(Scalability);
• User profiling is allowed because all the Data Bases are centralized in the
HSS.
In order to validate what I have described in this chapter and what is my end result,
the subsequent points have been analyzed:
• Feasibility: the whole system cost is limited because the implementation does
not occur ex novo, but the existing equipments are expanded.
• Flexibility: the whole system can be amplified and adapted introducing new
upcoming access technologies without changing the existing architecture. The
only portion of the network to be modified is the Edge Network.
• Dependability: this network is able to provide required services continuously
due to the centralized Core Network.
• Robustness: critical situations can be supported without failure, in such cases
as traffic increasing, insufficient bandwidth, power lower than a bound in a
coverage area, etc another access technology can be used.
• Maintenance: simple configuration, monitoring and statistics of the system by
software from a remote place because the whole architecture is
interconnected.
• Usability: a user can log on and utilize any service simply and without
knowing which technology is used
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Finally, one of the main issues this kind of convergent architecture introduces is
about Security. As said in the previous paragraph, such convergence makes the entire
system more robust because this feature is performed by different sections of the
network regarding different aspects. Moreover there are centralized AAA server and
HSS. However, it is difficult to ensure protection of information and to save from not
authorized accesses to services because users can establish sessions between other
users and escape somehow authorization or identification. This problem is an open
issue and it must be taken into account.
APPENDIX A: Implementation
153
AAPPPPEENNDDIIXX AA:: IImmpplleemmeennttaattiioonn
A.1 Introduction
The resulting proposal of this thesis is a convergent architecture allowed by the
definition of an Edge Network. The benefits the introduction of the Edge Network
gives are explained in Chapter four. Now I want to present how this convergent
architecture can be implemented. In order to achieve this, a short definition of an
example of a convergent service is illustrated. Then the way this service is possible
thanks to the architecture is proved. The devices involved during the interactive
service are described. The role of the DVB-H Content Provider is especially
underlined because it is most improved innovation. In fact, it is one of the
fundamental devices for the Edge Network and it needs to hold some functionality
that it does not perform currently.
A.2 Case Pilot of a Convergent Service: m-Advertising Game
The m-Advertising Game is an interactive, realistic advertisement that allows the
user to play a “choose your own adventure”-style advertisement inside broadcast
programming over the mobile phone. From the user’s perspective, he or she first has
the option to choose a personalized “virtual reality”-type character that can play
inside the advertisement. Programs for creating these characters are already
available, as can be seen from Figure 54, which is taken from the clothing store
H&M’s website.
Figure 54: m-Advertising Game
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154
This makes the service highly personalized, which is one of the main customer
demands when it comes to next generation services—especially for the defined
“Speen Generation”11 target.
The advertisement appears as a normal break in traditional m-broadcast television
program. The user has the choice to watch a traditional (non-interactive)
advertisement, or participate in the new advertising service. Both advertisements,
regardless of which the user chooses, are made with the audience in mind. That is, if
someone is watching a program on home decoration, the appropriate advertisements
that fit those watchers will be displayed, and if the program is a Formula 1 race, the
advertisement will also be targeted to those users. Please see below for an example of
an advertising match.
Figure 55: Interactive Spot
If the user chooses the interactive spot, he or she has many possibilities that are
inside the short advertisement. The user can choose the background, for example if
Fiat is the main advertiser, the user can choose from a lineup of Fiat cars which one
he or she prefers to have in the ad. The number of choices and real-world products
inside the ad depend on the number of advertisers who pay for the placement,
however there should not be so many products that the user spends the entire
advertising time bombarded with choices and has no time to play the actual game.
Once the personalized choices (background theme, music, clothing, etc.) have been
chosen, the virtual reality character enters into the designed environment.
11 “Speen Generation” target is a target group of age 15-21. This word derives from Speed+Teen=Speen.
APPENDIX A: Implementation
155
Inside that environment, the user has the possibility to download content (example
product demos, music videos, and entertaining advertisements) and interact with
another user. This interaction is a text-based chat, as seen in the figure below. The
innovative idea of having the advertisement also as a game that the user can play, is
in reality not a new idea. Many analysts predict that the mix of advertising and
gaming will be a very probable reality for the future. The prediction for in-game
revenue is that by 2010 it will reach $1.8B, and account for 3% of total media
spending.12 In fact, this forecast comes from a company called Massive, Inc. that is
completely focused on in-game advertising and was just purchased by Microsoft in
May of 2006.
Figure 56: Invite a Friend
A.3 Mapping the convergent service on the convergent architecture
Keeping in mind the service presented in the previous paragraph, I want to explain
here which devices take part in this service delivery. This service is just one of the
many available so this procedure description is a subset of what is reachable due to
the convergence through the Edge Network. I have considered this case pilot because
it involves the Content Provider server which is the mainly effort I had to face.
12 http://www.massiveincorporated.com
APPENDIX A: Implementation
156
Figure 57 displays a high level call flow describing the m-Advertising Game.
Figure 57: High Level Service Call Flow
The DVB-H Content Provider is the server transmitting broadcast contents
continuously. It is the one which stores the program scheduling so it is able to know
when a spot time is occurring. It sends an “Advertisement” message indicating the
beginning and ending time of the advertisement time to the IMS application server in
advance. In this way the IMS AS can inform users that an interactive spot is able. In
order to do this the AS sends a multicast message to all the users authorized to
receive this service, asking if they want to play the interactive spot. In the meantime,
the Content Provider is still transmitting in broadcast. As two of the users accept, the
IMS application server performs an association joining two generic users together.
From now up, a session between these two users is established. They can chat or
share files.
Moreover, the user can choose another background inside the short advertisement,
select a spot, download content or music, send an MMS to be displayed on the
APPENDIX A: Implementation
157
broadcast transmission etc... These additional requests he or she can do are allowed
because the Content Provider server is considered by the IMS network as an
additional user. Thus, in this case the session is established between a user and the
Content Provider server.
Everything described up to now is possible thanks to the convergent architecture
defined in this thesis. Indeed what I have done was to bind the IMS Presence Service
to the DVB-H TV transmission. This is highlighted in this kind of service. This
makes possible Instant Messaging and session establishment between users (Users
Association). Furthermore, the Content Provider is able to know who-see-what.
Due to the SIP protocol layered inside the Content Provider, this server can be
synchronized with the IMS Application Server by messages exchange.
Finally, the most innovative focus of attention is to simulate a session establishment
and a session management between a user and the Content Provider, in order to allow
Content Sharing:
• TV on demand
• Content Downloading
• Display MMS.
A.4 Proxy Simulation
In order to demonstrate everything described before is possible I chose to simulate an
IMS environment using a SIP proxy. In this simulation also the Content Provider
server is involved. This device is an Edge Network element, so it provides all the
functionalities needed to connect a DVB-H system to the IMS network. One of the
functions the Content Provider server performs is session establishment. I want to
underline this function because, at present, it is not offered in a kind of device like
this. To understand better, Figure 58 shows the role played by the Content Provider
server during the synchronization with the S-CSCF in the m-Advertising Game.
APPENDIX A: Implementation
158
Figure 58: Content Provider server
This figure illustrates an example of what the Content Provider does end which SIP
messages are involved in a session establishing (in this case with the SIP Application
Server). In particular, this points out that the Content Provider behaves differently by
the two sides:
• User side: it sends contents in broadcast
• CN Side: it sends an advise to the SIP AS, establishing a session via S-CSCF
In order to establish a session the message needed are INVITE and 200OK (S-CSCF
redirect), in fact the Content Provider server does not need a registration procedure
via P-CSCF because it is not a mobile terminal.
Emphasizing the capability of the Content Provider server to establish a session, i.e.
with a user as well as an IMS Application Server, I aimed to simulate it. So, I have
considered a proxy which functions as a SIP registrar and a SIP presence server,
using source code of a JAVA based SIP proxy built on top of the JAIN-SIP-1.1 API.
Apart from other capabilities, this proxy can act as presence server and be able to
process NOTIFY and SUBSCRIBE requests. If the parameter “Presence Server” is
disabled, the proxy will simply forward those kinds of request following the
appropriate routing decision. The proxy can fork the INVITE requests it receives.
In Figure 59 the SIP proxy diagram class is displayed.
APPENDIX A: Implementation
159
Figure 59: SIP Proxy
This kind of proxy allows Instant Messaging between users, especially thanks to the
presence function. There is, however, an issue to be solved in order to obtain what I
was aimed at. This simulator does not establish sessions between users, so I had to
implement the signaling needed to session.
The IM (Instant Messaging) package containing all the class related to the IMS
clients is shown in Figure 60.
APPENDIX A: Implementation
160
Figure 60: Instant Messaging
The Content Provider server can be simulated as one of this IM client which is IM
User Agent inside the Presence Package.
Moreover, I had to enable an HTTP session between users and between the Content
Provider server and users in order to allow content or file sharing. Due to this, I have
used an HTTPserver class. This class is called inside the processMessage( ) method,
which is invoked inside the IMMessageProcessing class (in the package
instantmessaging.presence). What I have done can be seen in the code below.
APPENDIX A: Implementation
161
APPENDIX A: Implementation
162
The class diagram of what I have described is shown in Figure 61.
Figure 61: HTTPServer
Now is possible to establish sessions between users, in particular also HTTP
sessions, without passing through the proxy compulsory excepting for the initial
phase.
Concentrating the attention to the Content Provider server, it is important to indicate
that it takes the users’ presence from the proxy. In fact, the Content Provider can be
treated by the proxy as a user taking advantage of other users’ presence. It can also
take the users’ IP address from the proxy. This is very important to be able to
establish a session directly with the users without the proxy. The User-Agent header
field of the IP packets exchanged (between the User Agent and the SIP proxy in the
INVITE or SUBSCRIBE procedure) contains information about the UAC (User
Agent Client) originating the request. The interface Address is the one used for this
purpose. It represents methods for manipulating Address object values for any header
that contains an Address value.
APPENDIX A: Implementation
163
In order to obtain everything explained up to now and to verify everything obtained
works in the right way, a signaling trace was very helpful for me. It shows all the
messages exchanged every time two terminals are connected. As the example in
Figure 62 illustrates, the method, address and protocol used are displayed.
Figure 62: Example of Signaling Trace
Summing up, this simulation enables the Content Provider Server to connect directly
to users due to a SIP proxy, establishing HTTP sessions. The Content Provider server
is able to know the users’ presence, in fact a set of registrations that will be uploaded
into the proxy at start-up time can be specified, modifying the "registrations.xml"
file, as follows:
<REGISTRATIONS>
<REGISTRATION displayName="ANDREA" uri="sip:[email protected]">
<BUDDY uri="sip:[email protected]" />
</REGISTRATION>
<REGISTRATION displayName="FRANCESCA" uri="sip:[email protected]">
<BUDDY uri="sip:[email protected]" />
</REGISTRATION>
</REGISTRATIONS>
APPENDIX A: Implementation
164
Finally, the screenshot in Figure 63 highlights the added capability of the Content
Provider server to hold a buddy list of users. The Content Provider is the one
surrounded in green: it has its own SIP URL and the SIP proxy is able to recognize
it. It is clear in this way that the Content Provider can know the presence state of
each user.
Figure 63: SIP Proxy and Content Provider Server
APPENDIX B: Telecommunications Market Overview
165
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In the telecommunication market scenario there is a continuous evolution. In
particular the attention is focused on the new possibilities that come from the
innovative technologies. The telecommunications world market is expected to
continue its double-digit growth from 2004 to reach over $2 trillion by 2008. The
principal drivers for this growth are improving economic conditions throughout the
world, a growth in infrastructure equipment investment, and demand for mobile
devices and wireless services. The number of wireless subscribers is growing quickly
and high-speed broadband access will be a principal driver of equipment revenue in
the next four years, helped by increased government support and a stronger economic
environment. Broadband access revenue will triple between 2004 and 2008, from
$33 billion to $101 billion. As the broadband market expands, the need for
infrastructure to support the traffic will revitalize the network infrastructure
equipment market.
TIA13 expects equipment spending to increase at 8.1%, rising from $238 billion in
2004 to $325 billion in 2008. As the migration to wireless, voice over Internet
protocol (VoIP) and cable telephony continues, the landline market will increase
from $391 billion in 2004 to $422 billion in 2008, averaging only a 1.9 percent of
growth. International wireless revenue will expand at an 11.6% from 2005-08,
reaching $466 billion in 2008. The Wi-Fi and WiMAX market is increasing at a fast
pace and will continue to grow as hot spots proliferate. We expect revenue from
spending on wireless capital expenditures/Wi-Fi/WiMAX to reach an estimated
$22.3 billion in 2005, climbing to $29.3 billion by 2008, a 7.1 percent compound
annual gain. With the Wi-Fi and WiMAX markets expanding rapidly, we will begin
to see more demand for mobile broadband and broadband connectivity. The digital
television market on mobile phones is also expected to grow. It is predicted that 270
million users will subscribe to digital mobile television by the year 2009.
13 Telecommunications Industries Association, January 2006. www.tiaonline.org
APPENDIX B: Telecommunications Market Overview
166
Every market has unique needs and challenges. However, regulators and government
telecommunications officials should strive for common principles, including the
following:
• All people should have access to affordable, highly advanced and secure
communications services.
• Broadband deployment policies should be technology-neutral with respect to
user/service provider choice among multiple broadband technology options.
All broadband access technologies should be given equal consideration, if
technologically feasible. These technologies include, but are not limited to,
DSL, fiber, satellite, and fixed and mobile wireless.
• All governments are strongly encouraged to adopt fair and well-considered
national broadband deployment strategies:
- promote competition as a means of facilitating ubiquitous deployment
of broadband technologies;
- limit regulation to that which addresses a specific, critical problem and
minimize disruption to competitive market forces;
- recognize the national benefits derived from the use of nascent
technologies and not impose legacy regulatory models that would
inhibit broadband technology deployment and present obstacles to
addressing national needs;
- adopt spectrum management policies that include plans for the
provision of radio spectrum for the deployment of advanced
communications services;
One of the first concrete steps made by the Italian Government was the legislation
regarding the management and deployment across the country of Wi-Fi. From now
on it will be possible to cover and offer services between offices and houses.
Operators and providers are now able to create new businesses and ADSL access
also in areas that are difficult to reach with wired connections.
However, the 3G network, and in particular the UMTS technology, makes different
types of services already possible. The objective that the telephone companies have
proposed is to reduce the dependence on voice traffic, therefore increasing the weight
APPENDIX B: Telecommunications Market Overview
167
of the value added services. This new scenario should respond to the ruthless
competition that they find in these months due to the introduction and growth of
VoIP technology. The entry of new players may not destroy the value of this market
– indeed, it may significantly increase it overall – but it will certainly change the
rules of engagement.14
B.1 The International Scenario
In Japan, the operator NTT DoCoMo has introduced 3G-specific services and content
– including video-based entertainment – via its i-mode portal, while maintaining
compatibility with existing 2G services. Usage has been further stimulated by the
adoption of ‘all you can eat’ flat rate data tariffs that encourage user experimentation
and exploration of new services. DoCoMo’s strategy – teaming a friendly user
experience with extensive network coverage (now exceeding 99.9% of the
population) and the availability of affordable, attractive handsets with longer battery
life – has resulted in continuing subscriber growth as well as strengthened usage and
ARPUs. One of the key explanatory variables for the success of i-mode is that
DoCoMo created a viable business model where all actors, including operators,
content providers, terminal manufacturers, portal/search engine providers and
distributors, cooperated and had possibilities to run a profitable business.
Vodafone is an operator with an extensive European presence that has already
targeted consumers with its Live! 2G portal. Vodafone has already introduced 3G
Live! across several of its operations, integrating 3G services (spanning mobile TV
and video on demand) into its existing portal structure.
Orange, in France, meanwhile, has associated the launch of its consumer/residential
3G/UMTS offering with four key services, namely video calling, video MMS,
mobile television and mobile PC card access. Its Orange World portal has also been
enhanced with new content, featuring a range of thematic channels for the French
market and personalized infotainment services in the UK. In France, the operator’s
‘Orange Intense’ offering spans music and video downloads including more than 40
channels of mobile TV and radio plus video telephony and MMS. Orange’s initial
14 “3G/UMTS Towards Mobile Broadband and Personal Internet.” White Paper from the UMTS Forum, October 2005.
APPENDIX B: Telecommunications Market Overview
168
offer has been supported by a range of 3G, 3G/EDGE and EDGE-only handset
choices plus limited-period access to free mobile TV and unlimited video telephony.
In Spain, Telefonica has introduced a mobile TV channel to Movistar 3G subscribers
with services including streamed news.
Overall, the analysts predict in the telecommunications market a continuous growth
in partnerships between companies. There will also be the entrance of new players,
the introduction of new dynamics and the evolution of the business and mobile
services, all reaching the goal of a renewed convergent technologies, that once again
can be the key factor for generating new opportunities.15
B.2 The Italian Scenario
The goal of this section is to evaluate the Italian players and the competitive scenario
in the mobile telecommunications market in Italy, in respect to the fact that the
technological changes could create not only opportunities, but also substitute
products and services, which will create an environment of high competition. Also,
as convergence happens between what were formerly separate businesses, there is
more competition to offer similar services through different modes, putting formerly
separate companies in competition with one another. Therefore, it is helpful to
identify the current mobile players and understand the dispersion of the market.
B.2.1 Players in the Italian Market
TIM (Telecom Italia Mobile), founded in 1990, manages TACS,
GSM and also UMTS networks; it is a division of the fixed
operator Telecom Italia, which is also an Internet Service Provider (ISP), offering a
broadband service called Alice. The TIM Group, listed on the Milan Stock Exchange
in 1995, is one of the main mobile telecommunications operators in the world; leader
in the domestic market with over 26 million lines and a 42% market share as of
December 31, 2004 and present in Europe, the Mediterranean Basin and South
America. Recent news includes making accords with both RAI and Mediaset to offer
15 ”Mobile TV for Marketers: Monetizing the Smallest Screen.” eMarketer, April 2006. http://www.emarketer.com/Report.aspx?mobile_tv_apr06
APPENDIX B: Telecommunications Market Overview
169
broadcasting to the mobile phone. They have also announced that they are teaming
with Motorola and LG to develop Fixed-Mobile Convergence.
Operative since 1995, “Vodafone” (that has bought Omnitel to
enter in the Italian market) is the second-largest mobile operator
in Italy. It manages GSM, GRPS, and UMTS networks. The Italian division is
controlled by Vodafone Group, which is the world’s largest mobile operator. Recent
news involving Vodafone Italy is that it has also announced a DVB-H partnership
with Mediaset.16
Started in 1997, “Wind” is Italy’s third largest Mobile Network
Operator. It owns GSM and UMTS networks. It was the first mobile
operator in Italy to also offer fixed telecommunications (Infostrada), and a broadband
internet service, called Libero. It is a joint venture with France Telecom.
Operative since 2003, “H3G Italia” manages a 3G, GSM Dualband,
and CDMA networks, and it was also the first mobile operator in
Europe to offer UMTS services. It is a part of the internationally operating
Hutchinson Whampoa group. “Tre” has just acquired a television channel and has
therefore become the world’s first mobile operator to own a broadcasting license.
They are launching a DVB-H network, and connected services, in June of 2006.
“Blu” entered the market in 2000 and left in 2002, before the
entrance of H3G (until now there have never been more than four
mobile providers in Italy). It was the smallest of the network operators at the time.
The company was a consortium of international companies, the main share-holders
of which were Autostrade (with 32%) and British Telecom (20%) and Mediaset. It
pulled out of the bidding for the 3G network license, and soon after shut down.
16 “Mediaset, Vodafone Italia team up for mobile TV.” Digital Media News for Europe, April 2006. http://www.dmeurope.com/default.asp?ArticleID=14988
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