新世代匯流網路架構及其應用服務研究

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(1)⊕ 國立中山大學企業管理學系英語 MBA 碩士碩士論文. A Study on Converged Network Architecture and Application Services for Next Generation Network 新世代匯流網路架構及其應用服務研究. 研究生：林玉章撰指導教授：陳得發博士趙善中博士. 中華民國九十六年六月.

(2) 論文提要學年度：95 學期：2 校院名稱：國立中山大學系所名稱：企業管理學系論文名稱(中)：新世代匯流網路架構及其應用服務研究論文名稱(英)：A Study on Converged Network Architecture and Application Services for Next Generation Network 學位類別：碩士語言別：英文學號：N934110002 論文頁數：75 研究生(中)姓名：林玉章研究生(英)姓名：LIN﹐YU-CHANG DAVID 指導教授(中)姓名：陳得發指導教授(英)姓名：CHEN﹐DER-FA ROBERT 指導教授(中)姓名：趙善中指導教授(英)姓名：CHAO﹐S WILLIAM. i.

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(5) 致謝辭一篇論文的完成，並非僅是個人的心血和努力，而是一群人共同付出的貢獻。通訊科技的發展一日千里，目前由第三代行動通信及光通訊技術引領電信產業邁向數位匯流的時代，Baldwin & McVoy在《大匯流》一書中曾提到（Baldwin, McVoy, and Steinfeld, 1996）：「寬頻通訊系統」（Broadband Communication）已經形成，它將聲音、影像及資料整合起來，儲存龐大訊息供隨時選取（On demand），並集互動功能於一身。不久之前，電話、有線電視、無線及電腦工業都還是壁壘分明，現在這些產業卻逐漸匯流（convergence）成一個整合寬頻系統（integrated broadband system），儼然為新世代的資通訊網路。新世代的資通訊網路除傳統語音服務外，也可進一步提供影像資訊傳輸服務；藉由網際網路也可收聽廣播及收視節目，由此可知數位科技整合已成趨勢，因此，選定這個論文題目加以研究，相信在電信通訊、電腦網路、有線電視廣播和電子商務等產業的數位匯流趨勢下，對數位電信傳播產業發展、閱聽人/消費者權益及政府的規管機制與政策等將產生深切影響，應有值得探討的問題。在研提論文的過程中，首先要感謝的是陳得發老師及趙善中老師，指導架構理論及統計分析。本文可以說完全在他們的鞭策下，千錘百鍊打造出來的。李清潭老師是市場與法律議題的專家，對於當代經濟議題有獨到的見解，另外是郭倉義老師在資訊管理及應用服務的深切指導，使本篇論文在科技層面，能夠更臻完整。同班同學對本論文所提出的意見和觀點，提供了寶貴的資料，豐富了本文的內容。幫助者太多，無法一一在此明列致謝，特致歉意。最後謹向內人章瓈文及吾兒暐恩致上深深感激，感謝他們給予我的支持及愛護，讓我能全心全力完成論文的撰寫。林玉章謹識于中山企研所中華民國九十六年六月. iv.

(6) 中文摘要在二十一世紀資通訊科技（Information Communication Technology，ICT）議題中，對於決定一個國家競爭力之要素而言，主導知識經濟的電信產業與電信部門之發展及相關科技政策與管理是關鍵的重點。展望未來，電信、廣播及資訊三者之技術與服務匯流將日趨明顯，扮演整合通訊與傳播市場的監理角色，能有效排除跨業經營障礙，使得電信、廣播及資訊等三大領域之產業均能在公平的基礎上從事良性競爭與互動，從而加速相關產業的整合發展與應用，使得消費者、政府及業者皆能從技術精進及產業整合中獲取最大的實益。在廣播和電信服務的匯流（Convergence）、固網和行動服務的整合應用，以及網際網路成為所有應用服務的傳遞平台，亦即所謂的「數位匯流」趨勢下，不論是全球化、市場與客戶導向，自傳統技術至互連性（Interconnectivity）與互通性（Interoperability）等技術之精進，或是傳統上各自獨立的產業包括電信、傳播與資訊整合成所謂的「資通訊產業」（Info-communication Industry），其語音、影像、數據、多媒體傳播均將因網際網路的發展統合彙整在單一共通的網路平台上。這個全球通訊電信的變遷，促使大部分國家之電信部門為因應此通訊環境之快速變化，大多已進行組織變革及技術的提升，或是處於此變革改造的發展之中。在ICT發展潮流下，不論是「新世代網路」（Next Generation Network; NGN）、「多媒體子系統」（IP Multimedia Subsystem; IMS）或是「固定業務和移動業務融合」（Fixed-Mobile Convergence; FMC），均帶動一個數位匯流之新世紀。然而在此因技術所推動的變遷下，原有政策組織與法律已不敷時代之變化，衍生出創新應用的經營模式，同時其所產生的「數位落差」（Digital Divide）現象如富國與窮國、城市與鄉村之差距等，均需藉由調整新政策與法規來順應時代的變化。解決之方法為：開放寬頻資通訊市場競爭；允許私人企業之參與；以及建立監理法規。目前以VoIP技術為基礎的網路電信業務，開始挑戰傳統電信服務市場；各家業者均積極開始規劃新世代的通訊技術與服務。新的網路技術與新的應用服務，勾勒出一幅新的應用藍圖，引領新一波的企業投資與競爭！本研究以新世代匯流網路之架構及應用服務VoIP為案例，提出Inter-Exchange Center of Converged Network Architecture (IEXCNA,匯流互連交換中心)的架構，應用teletraffic theory，證明在傳輸頻寬的利用上，能更有效率。以提供在邁向新世代匯流型網路的變革中，解決電信網通業者的互連障礙及降低建置成本，使民眾的利益及服務品質獲得應有的保障。【關鍵詞彙】資通訊科技、匯流、互連性、互通性、新世代網路、多媒體子系統、固定業務和移動業務融合、網路電信、匯流互連交換中心. v.

(7) Abstract In 21st century, the development strategy and the management policy of Information Communication Technology (ICT) is a key issue to a nation’s competitiveness. Particularly, the development of telecommunication liberalization, governance in the telecommunications regime and the relevant science and technology policies of the telecommunication industry is the most important factor for creating a leading knowledge-based economy for Information Communication Technology related industries. Looking forward to the future, the trends and developments of the convergence of telecommunication, broadcasting and internet services will drive the demand for the telecommunication and broadcasting industries to look for cost-efficient provisioning of converged multimedia services. The emerging technology of next generation network (NGN) infrastructure enables the convergence of multi-access networks to deliver the multimedia contents and application services seamlessly. Therefore, the consumers, government, and industry can obtain the benefits because of technology development and industrial cooperation. In addition, the converged broadcasting and telecommunication services as well as Fixed-Mobile Converged applications over the internets results in the needs for the improvement in network interconnectivity and service interoperability. No matter it is telecommunication, broadcasting or information, to converge as common info-communication services, such as telephony, data and multimedia will eventually run over the all IP networks. Therefore, IP Multimedia Subsystem (IMS) becomes one of the promising technologies to drive it happen. However, existing policy and regulations must cope with the changes of this development. The liberalization of broadband and communication market allows the participation of the private companies to create new investment and revenue. This research studies the possible evolution of an IP-based communication infrastructure from today's networks toward a converged next generation network and proposes an Inter-Exchange Center of Converged Network Architecture (IEXCNA), applying to VoIP service. The IEXCNA model shows more efficient in the transmission, interconnectivity and reducing implementation cost of network infrastructure. Thus the consumers are able to enjoy better quality of service and cheaper rate of use for multimedia services. 【Key Words】 Information Communication Technology、convergence、next generation network、 interconnectivity、interoperability、Fixed-Mobile Converged、 IP Multimedia Subsystem、VoIP、Inter-Exchange Center of Converged Network Architecture. vi.

(8) Table of Contents 論文提要 ......................................................................................................................... i 致謝辭............................................................................................................................ iv Abstract ............................................................................................................................. v I、INTRODUCTION ....................................................................................................... 1 1.1、Research Motivation ........................................................................................ 2 1.2、Research Purpose ............................................................................................. 3 II、REVIEW OF LITERATURE ..................................................................................... 4 2.1、Wireless and Wireline Technology Development Trend ................................. 4 2.2、NGN Converged Network Architecture and IMS.......................................... 16 2.3、Voice over Internet Protocol (VoIP) Service.................................................. 25 2.4、Traffic Dimensioning Theory for Telephony Service .................................... 28 III、RESEARCH METHODOLOGY............................................................................ 31 3.1、Research Strategy and Expected Outcome .................................................... 31 3.2、Thesis Structure ............................................................................................. 32 IV、IEXCNA MODEL FOR VOIP SERVICE .............................................................. 33 4.1、System Functionality of IEXCNA for VoIP Service...................................... 33 4.2、Call Flow of VoIP Service in IEXCNA Model .............................................. 37 V、RESEARCH FINDINGS AND DISCUSSION ....................................................... 46 5.1、Traffic Load Calculation and Bandwidth Dimensioning ............................... 46 5.2、Overall IEXCNA Model for Commercial VoIP Service ................................ 54 VI、CONCLUSIONS .................................................................................................... 58 REFERENCES ............................................................................................................... 63. vii.

(9) List of Figures Figure 2.1-1 Network Architecture of NGN Layer Modeling .......................................... 5 Figure 2.1-2 the current architecture of communication networking ............................... 7 Figure 2.1-3 Next Generation Wireless Network Architecture......................................... 9 Figure 2.1-4 Current coverage of FGNGN Release 1 defined by ITU-T ....................... 11 Figure 2.1-5 Impacts of NGN to the Telecommunications Industry............................... 14 Figure 2.2-1 NGN IMS Network Architecture ............................................................... 17 Figure 2.2-2 IMS Network Components ........................................................................ 17 Figure 2.2-3 IMS Core Network..................................................................................... 18 Figure 2.2-4 CSCF control in IMS ................................................................................. 20 Figure 2.2-5 IMS with VXML control of media servers ................................................ 22 Figure 2.3-1 Basic functions of a VoIP system ............................................................... 26 Figure 3-1 Research Plan and Procedures ...................................................................... 31 Figure 4.2-1 System Architecture of IEXCNA............................................................... 37 Figure 4.2-2 IEXCNA model of Network Inter-connections for VoIP ........................... 39 Figure 4.2-3 The National Numbering Plan and Routing Table Database ..................... 40 Figure 4.2-4 Type 2-A to Type 2-B Call via IEXCNA ................................................... 40 Figure 4.2-5 Type 2 A to Type 1 PSTN Call via IEXCNA ............................................. 41 Figure 4.2-6 Type 1 PSTN to Type 2 A Call via IEXCNA ............................................. 41 Figure 4.2-7 Signaling Message Flow Type 2 A to Type 2 B via IEXCNA ................... 42 Figure 4.2-8 SIP Signaling Message Flow Type 2 A to Type 1 PSTN/PLMN via IEXCNA ......................................................................................................................................... 42 Figure 4.2-9 SIP Signaling Message flow Type 1-PSTN/PLMN to Type 2 A via IEXCNA ......................................................................................................................................... 43 Figure 4.2-10 Internet & SIP Peering for VoIP service................................................... 44 Figure 4.2-11 The functional blocks of IEXCNA model for VoIP service ..................... 45. viii.

(10) Figure 5.1-1 The bandwidth requirement for NON-IEXCNA model of VoIP service ... 47 Figure 5.1-2 The bandwidth requirement for IEXCNA model of VoIP service ............. 48 Figure 5.1-3 BSP of IEXCNA model related to N of Type 2 Operators......................... 51 Figure 5.1-4 Non-IEXCNA model operational cost compared to IEXCNA model operational cost ............................................................................................................... 51 Figure 5.2-1 The practical example of network architecture based on IEXCNA model for VoIP service .................................................................................................................... 54 Figure 5.2-2 Business Service Market for VoIP Revenues ............................................. 55 Figure 5.2-3 VoIP Impact on Net Industry Business....................................................... 56 Figure 6-1 Non-IEXCNA model for Point to Point network inter-connections ............. 58 Figure 6-2 The IEXCNA model of network inter-connections....................................... 59. ix.

(11) List of Tables Table 2.2-1 NGN IMS Interface description................................................................... 23 Table 5.1-1 BSP γ related to Number of Type 2 Operators ........................................... 50 Table 5.1-2 Erlang B Table ............................................................................................. 52 Table 5.1-3 Overall traffic load and bandwidth dimensioning ....................................... 53. x.

(12) I. Introduction. The liberalization in the telecommunications and the renovations in regulation have changed the telecommunications industry trend dramatically in Taiwan as well as worldwide. The regulatory authority – DGT (Directorate General Telecommunications) under the MOTC (Ministry of Transportations and Communications) guidance and supervision, has gradually opened the telecom market, persistently created an environment for fair competition, and sure-footedly safeguarded consumers' rights and interests. It has also continued its effort to promote the infrastructure construction for broadband networks, laying a firm foundation for an information-based society. As of now, the telecom market has been fully opened to fair competition, and all the competition mechanisms have been put on the track of the world. As a result, consumers have been able to enjoy the benefits of the sharp fall of telecom rates, all forms of high-quality and incessantly improving services brought about by competition. Envisioning the future, people can clearly see the increasing convergence of the telecommunications, communications and information technologies. The integration and renovation of regulation will effectively remove the barriers between the three sectors so that the three can engage in benign competition and interaction on a level ground and accelerate the broadband communication development and applications. The sustained value of a next generation network (NGN) application to customers depends critically upon how economically it is delivered (ES 282 001, 2005). The application Go-to-Market (GTM) deployment strategy adopted by the service provider addresses the location and ownership of the application platform, as well as its financing. It ranges from hosting the applications through a third party for a fixed fee per subscriber, through pay-as-you-grow arrangements that involve varying degrees of revenue share, to outright purchase of the hardware and application platform. Each strategy offers a unique mix of returns and risk mitigation. This study investigates the problems of inter-connection and interoperability of the VoIP service for the VoIP service operators (May, 2005) and proposes an Inter-Exchange Center of Converged Network Architecture model (IEXCNA) to solve the problem of inter-connections when evolving toward NGN for both fixed and mobile service providers. In recent years, the IP Multimedia Subsystem (IMS) is envisioned as the solution for the next generation multimedia rich communication (May, 2005). Based on an open IP infrastructure, it enables convergence of data, speech, video and mobile network technology. The Session Initiation Protocol (SIP) is the signaling protocol chosen by the 3GPP consortium for IMS (Poikselka and Mayer, 2006). This research proposes a network connection model for different operators to provide VoIP service when evolving toward NGN. It analyzes the network inter-connection based on the bandwidth utilization and SIP signaling delay to predict performance trends of for the network inter-connection efficiency, that allow us to solve the inter-connection problems existing in current non-IMS VoIP service. The research focuses on the Inter-Exchange Center of Converged Network Architecture model (IEXCNA) for VoIP network and traffic-bandwidth utilization, and on the design of such network for optimal performance. Our analysis is based on a careful study of live VoIP network traffic. 1.

(13) 1.1. Research Motivation. The rapid change in technology for microelectronics, software, photonics and wireless are disrupting the fundamental nature of networks. Other contributing factors include the deregulation in telecommunications markets on a global scale, the public’s insatiable appetite for telecommunication services, decreasing terminal costs, migration of analog networks to digital networks, the increase competition among service providers and the resulting need for differentiated services. These forces for high speed, high capacity data services are driving the next generation network architecture toward a packet network. These next generation networks (NGN) are basically of two types: wired networks and wireless networks (ES 282 007, 2006). Today’s wired networks can be divided into two groups: the public switched telephone network (PSTN) and the public switched data network (PSDN). The PSTN consists of large, centralized, proprietary switches with remote switching modules and digital loop carriers. The PSTN is a low delay, fixed bandwidth network. In contrast, the PSDN is based mainly on packet switches that provide very flexible data services. It is a variable delay, variable bandwidth network that provides no guarantees on the quality of service (QoS) that the traffic it supports will experience. The projection is that data traffic volumes will overtake voice traffic volumes around the world over the next few years. This is based on the explosive growth of the Internet and related intranets and extranets. As data traffic surpasses voice traffic, it will be desirable to converge the multiple networks around a single packet-based core network (TR 181 003, 2006). Such convergence will support emerging multimedia services, increase the ability of a carrier to support the multiple needs of its customers, and reduce the cost of network operations. This convergence around a packet based core network will also allow many networks to collaborate. As a result, customers will think that they are working with a single, integrated network. Continuing exponentially increasing advances in microelectronics, photonics, and wireless technology are supporting the market drive to converged networks. These technologies provide the cost-effective foundation for the network elements making up the next generation of converged networks. The next generation networks will likely share a high level architecture. The next generation fixed networks will contain the following two major components: 1.1.1. Packet-based core network:. Also referred to as the backbone network, the core transport network will be built around dense wave division multiplexing (DWDM) transport systems. Systems carrying 400 Gbps utilizing 80 wavelengths are now available. They will rapidly evolve to carry significantly more information using an ever increasing number of wavelengths. Today, building a converged network core that provides QoS requires accommodating four protocols—IP, STM, ATM and DWDM. Each protocol provides important network capabilities. Internet protocol (IP) provides data interoperability and emerging support for QoS; asynchronous transfer mode (ATM) provides guaranteed QoS support for multiple services; synchronous transfer mode (STM) provides reliable transport; and DWDM provides high-speed transport (ES 282 001, 2005). 2.

(14) 1.1.2. Broadband access for wireline and wireless:. There will be multiple broadband access solutions for residences and businesses. Access systems include ADSL and FSO for wire-lined networks and 3G and LTE for wireless networks. Asymmetric digital subscriber line (ADSL) protocols do allow a few megabits to be carried over the already existing copper wiring in a cost effective manner. Free space optical (FSO) networking technologies provide an effective and economical solution to the “ last mile” problem of connecting to fiber infrastructure in metropolitan areas. These protocols support both traditional voice services and data oriented Internet connections. It is useful and necessary to pay attention to the development of NGN in regards to broadband access and IP based packet network. The existing mesh-connected network topology has caused inter-working and interoperability problems among different operators. It is important to analyze these problems and find out solutions to provide better QoS and improve service interoperability and network inter-connection.. 1.2. Research Purpose. This research is aimed to propose a network connection model (Inter-Exchange Center of Converged Network Architecture model, IEXCNA) for VoIP service implemented in IP based NGN networks. The IEXCNA model expects to provide efficient routing for the VoIP calls between ISP operators (known as Type 2 operators according to Taiwan Telecommunications Act) and PSTN or PLMN (known as Type 1 operators with requirements of national coverage of service provision). Also IEXCNA model provides the possibility of number portability service and emergency call service for VoIP subscribers to insure the quality of service without downgrading. Through the calculation of the input data of the traffic model and compared the result with the existing mesh connected model of current networks, it shows a better efficiency of the traffic throughput and bandwidth utilization as well as improve service interoperability and network inter-connection for the application services running over IP based NGN networks.. 3.

(15) II. Review of Literature. 2.1. Wireless and Wireline Technology Development Trend. The evolution of mobile networking technologies from the early personal communications systems of the 1980’s to the present day voice over IP and mobile broadband multimedia services. The research will compare some of the parallel developments of wireless and wireline broadband networks and examine the extent of their convergence. Future trends in ubiquitous network technologies and some of the potential applications will also be discussed. 2.1.1. Current Status of Telecommunications Technology Trend The last 10 years or more have seen an increasingly fast integration of computers and telephony, both equipment and networks. Traditional public network operators (PNOs) have seen a decrease in telephony traffic on their public switched telecommunications networks (PSTNs), due in part to the increasing popularity of mobile telephones and the movement of services from telephone networks to the public Internet. Telephone network customers’ demands have moved away from the all-embracing “one-stop shop” for communications provided by their network provider, preferring the unregulated but huge content and communications possibilities offered by the public Internet. The so-called “fixed” network operators’ response has been to meet that demand by deploying broadband. While this solution satisfies the customer demands, it has done little to ensure the continued development of global communications networks, as the network operator is left merely providing access to the public Internet (or worse, access to an Internet service provider, ISP) while content and services are provided without any association with networking costs. Customers buy services and not technology, so it is the ability to offer services that can take advantage of broadband which is important from the network operators’ point of view.. 4.

(16) Figure 2.1-1 Network Architecture of NGN Layer Modeling. 5.

(17) The concept of a new, integrated broadband network has developed over the last few years and has been labeled next-generation network (NGN). The basic characteristics of an NGN can be determined from the problems faced by the network operators: the need to provide services over broadband accesses (to increase revenue); the need to merge diverse network services, such as data (Web browsing), voice, telephony, multimedia, and emerging “popular” Internet services such as instant messaging and presence (IMPS)、 voice over IP (VoIP) and broadcast type services like IPTV and MBMS; and the desire of customers to be able to access their services from anywhere (inherent mobility). Rather than a network to provide a specific solution (e.g., the PSTN), what was needed for the 21st century was a series of networks that could support a flexible platform for service delivery. One of the most important features of IP is the independence of protocol layers (upper or lower) (ES 282 004, 2006). This feature has greatly impacted global connectivity networks, which provide connections independent of any kind of sub-layered networks such as PSTN, asynchronous transfer mode (ATM), and frame relay. Broadband access, such as asymmetrical digital subscriber line (ADSL), has enabled global connectivity coupled with various online applications, making a huge impact and creating a kind of online global village. An NGN therefore aims to combine the best of both worlds from the PSTN and the Internet.. 6.

(18) Figure 2.1-2 the current architecture of communication networking. 2.1.2. Emerging next generation network NGN technology has been discussed in standards since at least 2003, and the commonest question asked has been “What is an NGN?” The commercial needs, as outlined in the introduction of this paper, provided the starting point in determining the requirements to answer the question. International Telecommunication Union — Telecommunication Standardization Sector (ITU-T) Study Group 13 defined an NGN in Recommendation Y.2001 as “A packet-based network able to provide telecommunication services and able to make use of multiple broadband, QoS-enabled transport technologies, and in which service-related functions are independent from underlying transport-related technologies. It enables unfettered access for users to networks and to competing service providers and/or services of their choice. It supports generalized mobility which will allow consistent and ubiquitous provision of services to users.” Recommendation Y.2001 further defines the NGN by the following fundamental characteristics (Lee, 2005):. 7.

(19) – Packet-based transfer – Separation of control functions among bearer capabilities, call/session, and application/service – Decoupling of service provision from transport, and provision of open interfaces – Support for a wide range of services, applications, and mechanisms based on service building blocks (including real-time/streaming/non-real-time and multimedia services) – Broadband capabilities with end-to-end quality of service (QoS) – Inter-working with legacy networks via open interfaces – Generalized mobility – Unrestricted access by users to different service providers – A variety of identification schemes – Unified service characteristics for the same service as perceived by the user – Converged services between fixed/mobile – Independence of service-related functions from underlying transport technologies – Support of multiple last mile technologies – Compliance with all regulatory requirements, for example, concerning emergency communications, security, privacy, and lawful interception Recommendation Y.2001 decomposed the NGN into a number of areas to be studied for requirements and solutions. These areas still largely form the basis of standardization activities in ITU-T and other standards development organizations (SDOs): – – – – – – – – – – – –. General framework and architectural principles Service capabilities and service architecture Interoperability of services and network in the NGN Telecommunications capabilities for disaster relief Architecture models for the NGN End-to-end QoS Service platforms Network management Security Generalized mobility Network control architecture(s) and protocols Numbering, naming, and addressing. This research addresses general framework and architectural principles, and service capabilities and service architecture in NGN. The technical requirements are still developing, mostly because NGN covers such a large area. Delivery of voice services leads the complexity of PSTN control and management; generalized mobility introduces fixed-mobile convergence (FMC) and the decoupling of service provision from transport, while provision of open interfaces adds the complexity of adapting public Internet approaches to provide the same safe, secure, and reliable networking as does the PSTN.. 8.

(20) Figure 2.1-3 Next Generation Wireless Network Architecture. A further commercial need from some PNOs was to use the NGN to replace their aging PSTN (in whole or in part). This requirement was rather more than inter-working and interoperation with legacy networks and terminals. An NGN used to replace a PSTN must provide all of the services the individual operator’s PSTN provides — and in an exactly equivalent manner, from the customers’ point of view. This is because a customer with legacy equipment that chooses not to take up the new services an NGN offers does not have a new contract with a service provider (SP) and therefore should not be affected by any changes to the network or its technology. The overall requirements for an NGN can be derived from the needs and characterization in this section. This was clearly a large task, and the demanding timescales required cooperation between standards bodies and the organizations attending them. Two fundamental principles dominated the organization of the work. 2.1.3. Actions in standards organization bodies In July 2003, ITU-T organized the NGN workshop with the title “Next Generation Networks: What, When and How?” at its headquarters in Geneva, Switzerland. The workshop participants covered most of the telecommunications area, with regulators, industries, carriers, and user groups all represented. One of the difficulties during the workshop was that people used the same term for the future network, NGN, but in somewhat different ways, causing some confusion. Nevertheless, the necessity for NGN 9.

(21) global standards was clearly expressed, and was sufficient to make ITU-T members take serious note of these needs. In considering the results of this workshop, ITU-T SG13 launched an NGN Joint Rapporteur Group (JRG) initiative almost immediately in 2003. The NGN-JRG was mandated to identify key issues and develop fundamental standards for building the frameworks of an NGN, including a definition within an ITU-T context, and continued until June 2004. Recommendations Y.2001 (Lee, 2005) and Y.2011 (TR 181 003, 2006) are the results, and are now the basis for NGN studies in ITU-T. 2.1.3.1 ETSI TISPAN In parallel with the ITU-T initiatives, a European initiative began in the regional standards body, the European Telecommunications Standards Institute (ETSI). Two existing committees within ETSI were combined in the summer of 2003. ETSI TIPHON had been investigating the requirements for inter-connecting voice over IP (VoIP) and existing PSTN network with some success. ETSI Signaling Protocols and Networks (SPAN) had a long history of providing the European flavor of telecommunications standards. Combining these two provided a Technical Body (TB) with a focus on Internet protocols, reuse of services, and skill in developing well used standards. Although ETSI is a European regional SDO, it drew membership from beyond Europe and is probably best known for developing the second generation (2G) Global Systems for Mobile Communications (GSM). To develop access network independence, and to promote Fixed-Mobile Convergence (FMC), TISPAN chose to support existing fixed broadband access networks and required the IP-CAN (IP Connectivity Access Network) to be supported.. 10.

(22) Figure 2.1-4 Current coverage of FGNGN Release 1 defined by ITU-T. TISPAN started slowly, but after much discussion and vital strategic input from a small number of network operators, it derived a simple plan to meet the immediate needs of the market: – To provide all of the services enabled by the Third Generation Partnership Project (3GPP) Internet multimedia subsystem (IMS) to broadband customers, and selected IMS services to PSTN/ISDN customers connected to an NGN. – Providing most of a network operator’s existing PSTN/ISDN services to legacy equipment and interfaces to support PSTN/ISDN replacement scenarios. – Extending the 3GPP IMS to cover those regulatory areas 3GPP may not have covered: specifically, emergency calling and lawful intercept (LI), possibly malicious call indication (MCI). TISPAN was also required to show that the regulatory requirements for privacy can be met (a calling line identification presentation/restriction, CLIP/CLIR-like, service) and overruled when necessary (emergency calling, LI, and MCI, at least). The 3GPP IMS was selected as it provided several of the fundamental characteristics of an NGN. The IMS is predicated on packet-based transfer; it supports the separation of control functions among bearer capabilities, call/session, and application/ service; it decouples service provision from transport; provides open interfaces; and supports a wide range of services, applications, and mechanisms based on service building blocks. The IMS incorporates generalized mobility since it was defined in a mobile based SDO. PSTN replacement, as has been shown, was a key factor for some network operators, but not all. Those that did not have this requirement were more interested in providing PSTN-like services over broadband accesses (e.g., ADSL). This led to the definition of two types of PSTN services: emulation and simulation. These similar terms were meant to differentiate between the need to support customers’ legacy equipment and interfaces in an identical manner to the PSTN, and the need to supply similar types of services over broadband accesses to new or enhanced customer equipment. The elements of the TISPAN NGN were therefore derived to be: – A service control plane supporting differing service subsystems (initially PSTN emulation and an adapted IMS) – A separate applications plane – A core transport plane based on IP technology – Integration with existing (diverse) broadband access networks – Security, QoS, and network management For the purposes of TISPAN, an access network was regarded as the network component between the customer equipment and the first network element to support service control interactions. To develop access network independence and promote FMC, TISPAN chose to support existing fixed broadband access networks and required the IP connectivity access network (IP-CAN) to be supported. This would also reduce the temptation to change the 3GPP IMS except by addition for FMC. TISPAN could not 11.

(23) provide a set of global, only regional, standards, and therefore needed somewhere to globalize its results, and the ITU-T was the obvious place to start. 2.1.3.2 ITU-T As several standards organizations initiated their plans for NGN standards at the beginning of 2004, industry and carriers expressed concerns about possible overlaps, delays, and incompatibilities among future NGN standards if they were produced by different organizations with different contexts. These concerns were discussed among regional standards bodies from Europe, North America. and Asia, and resulted in setting up one special group in ITU-T to initiate a single coordinated NGN standards activity. This would also bridge the gap between study periods. The ITU-T is organized into four-year study periods, and the period between 2000–2004 and 2005–2008 was potentially extremely disruptive as considerable reorganization of the ITU-T was proposed by a number of organizations and members. After consulting ETSI, the Alliance for Telecommunications Industry Solutions (ATIS), China Communications Standards Association (CCSA), Telecommunication Technology Association (TTA), and Telecommunication Technology Committee (TTC), and also receiving some high-level consultation from the ninthth Global Standard Collaboration (GSC), ITU-T launched its Focus Group on NGN (FGNGN) under the responsibility of the ITU-T Director in June 2004. This Focus Group was required to coordinate all aspects of NGN studies and specifically included a requirement to provide the globalization of ETSI TISPAN results. 2.1.3.3 ITU-T Focus Group on Next Generation Networks (FGNGN) One of the urgent issues at the beginning of the FGNGN was to define its terms of reference (ToR), because of its position as a special group with a specific timeframe inside the ITU-T. The ToR on the FGNGN group were developed based on ITU-T Recommendation A.7 (Lee, 2005), which specifies focus group activities. After consultation, including with many ITUT Study Group chairpeople, and serious discussion during the ninth GSC meeting at May 2004, mandates were given to the FGNGN to create its deliverables within 12 months based on ITUT Recommendation Y.2001 (Lee, 2005) and Y.2011 (TR 181 003, 2006), and specifically concerning the following topics: – NGN functional architecture (e.g., based on 3GPP/3GPP2 IMS, but including support for broadband, e.g., xDSL access) – Generalized mobility – QoS – NGN control and signaling – Security capabilities, including authentication capabilities – Evolution from existing networks to NGN The firstst meeting of FGNGN was held at the ITU premises in Geneva, June 2004 to approve the proposed ToR, and set up management teams and working groups to meet the missions. 2.1.3.4 FGNGN Release Plan and Future ITU-T FGNGN works on a release basis with clear objectives and a target date. A release is a method of prioritizing by identifying a set of services to be addressed in a. 12.

(24) specific timeframe. The FGNGN is progressing the work to define the service requirements, and the capabilities needed to realize those services as well as to define other associated capabilities to facilitate an NGN in its first release. 2.1.3.5 Components of the FGNGN Release To use this release concept as a method of prioritizing the work, various harmonized components cooperating to provide key features were needed. The FGNGN uses the following two component groups. A. Functional components: These are technical components that represent the configuration of NGN from both vertical and horizontal perspectives: – Vertical perspective: covering transport stratum and service stratum or layers 1–7 of the open systems interconnection (OSI) model – Horizontal perspective: end to end (user terminal to user terminal, including all network and service platforms) B. Structural components: These are operational components useful in managing the development of each release. The following parameters were used in ITU-T during the development of important telecommunications infrastructure, such as ISDN based on ITU-T Rec.I.130 (Lee, 2005). – Timing (release) : Dates for publication of deliverables – Stages: 1–31 based on Recommendation I.130 (Lee, 2005) – Depth: level of descriptions 2.1.4 Status of FGNGN“RELEASE 1” Development The ITU-T FGNGN is developing its deliverables based on the seven working groups outlined earlier. An overview of the Release 1 service aspects has formed the basis and overall derived in conjunction with ETSI TISPAN and depicts the Release 1 development. This figure 5 shows several key aspects of the release 1 approach to NGN. Horizontally across the figure 5, the NGN is broken down into three sections: customer equipment, network equipment, and interconnection with other networks (including other peer NGNs). Vertically, the NGN is separated into two areas: the service stratum and the transport stratum. The service stratum can be further separated into services/control functions and application functions. The transport stratum is divided both horizontally and vertically. As the transport stratum is completely separated from the service stratum, it requires its own transport control plane to control bearer functions, such as the required QoS mechanism for a given session, and implement policy and admission. The transport network can also be divided into access and core. The linkage between the two strata is provided by the resource and admission control functions (RACF) for bearers, and by the network access and attachment functions in association with the user profile. The ITU-T FGNGN adopted the results and reused the 3GPP IMS as the basis for. 13.

(25) call/session control of real-time conversational communications in NGN. Standards meetings are never as smooth as the outputs would sometimes suggest, and during the FGNGN meetings there were many debates about the appropriateness of the IMS, especially from an architectural viewpoint. To satisfy all the participants, the fifth FGNGN meeting decided to take two different approaches to architectural modeling. One approach defines a general architecture model, while the other provides a specific case-oriented architecture using the IMS. 2.1.5 The Impact of NGN One of the important key features of the NGN is the separation between different functionalities that have an impact on the business models as well as regulatory implications. The separation between the service and transport strata will have many impacts from various points of view, but the largest impact is likely to be a change in regulatory direction. Today, most services are tightly coupled with a specific transport network and signaling protocol, so regulation has been applied mainly in a vertical direction (e.g., regulation for service always also applies to the transport network). This will change with NGN to the horizontal direction, so there will be different regulations between services and transport networks. One example of this is that service could be regulated to encourage competition through flexible development, and transport networks could be derived from building infrastructure and resources. This is shown in the left side of Fig. 2.1-5.. Figure 2.1-5 Impacts of NGN to the Telecommunications Industry. The second important impact of NGN is the separation of access capabilities with core transport capabilities, as shown on the right of Fig. 2.1-5. This feature may influence changing business environments. The business of an access network provider domain will be dynamically expanded according to the various access technologies, and users may. 14.

(26) have much more freedom to choose access capabilities based on their specific requirements. Furthermore, another important aspect will be to stimulate convergence between fixed and mobile communications. Thus, users will choose some fixed and some mobile access capabilities, and combine either, or both, with core transport capabilities, using a single (or at least minimum) user subscription identification. In the future NGN is likely to include telecommunications and broadcasting, which will provide convergence between telecommunications and broadcasting. 2.1.6 The Trend and Future Direction of Telecommunications Technology During the ITU-T FGNGN activities, a common question has been “What is the difference between an NGN and the Internet?” The question arises since both use IP as one important protocol. One clear difference of the NGN is that it does not restrict service delivery to best effort. The NGN will support various contractual services to meet users’ dynamic requirements. The NGN will be a secure, trustworthy managed network. The NGN will provide an opportunity, not only to service providers building business based on specific capabilities, but also to industries developing systems. In addition to this, the NGN covers more than data communications, providing the migration and integration of traditional telephone networks. The evolution of current networks into NGN is an important aspect. The future direction of NGN is undoubtedly the convergence of fixed and mobile networks and customer equipment. Currently, NGN standards groups are becoming tightly coupled with mobile groups (FGNGN and SG13 with SG19 in ITU-T; ETSI TISPAN with 3GPP, etc.). As fixed-mobile convergence develops, the distinction between fixed and mobile may disappear entirely, even in the last mile technologies. The other major trend for the future is likely to be tele-broadcasting convergence. The NGN is no longer a next generation objective, but is becoming a present generation reality.. 15.

(27) 2.2. NGN Converged Network Architecture and IMS. As part of the 3GPP specifications, the UMTS Release 5 was completed in March 2002. This release is well known, for example, for the HSDPA enhancements; however, important work was presented as well in the core network domain. UMTS Release 5 contributes with very important standard, the IP Multimedia Subsystem - IMS (Camarillo and Garcia-Martin, 2005). IMS defines a standard framework for the deployment of next generation IP-based application services. 2.2.1 Introduction of IP Multimedia Subsystem - IMS IMS defines how the services connect and communicate with the underlying telecommunication networks and how they integrate with the network provider's back-end systems. The driving force behind the IMS is that mobile networks and PSTN have a common requirement. It is the need for a flexible and standard platform to simplify the deployment of new services, regardless of the type of access network. 3GPP UMTS Release 6 extends the IMS standard toward an access independent network infrastructure. Most of the IMS system components are independent of access networks and so, with the appropriate gateways in place, IMS can meet the general requirement. The IMS standard encourages development of many interesting features as for example Push-to-talk over Cellular (PoC), presence and instant messaging, multimedia conferencing, and voice and video streaming services. They can be offered, in low costs, simultaneously to users over multiple access technologies from GPRS to UMTS, other wireless or even fix-line technologies. IMS is a broad and evolving standard and it will continue to evolve over the next few years. 3GPP UMTS Release 7, supported by ETSI TISPAN's activities (Telecoms & Internet converged Services & Protocols for Advanced Networks), will ameliorate and even extend the IMS standard towards principles of full fixed-mobile network convergence. The integration with other alternative radio technologies such as WLAN or WiMAX is envisaged, too. This paper gives an overview of the IMS architecture and it is organized as follows. The next section describes how the core IMS network separates from the underlying telecommunications networks. Section 3 introduces IMS system components and it provides an understanding of the most important system functions (Poikselka and Mayer, 2006).. 16.

(28) 2.2.2 NGN IMS Network Architecture The IMS provides a platform for communicating between all kinds of terminals. These terminals range from plain old telephones through mobile handsets and PDAs all the way to desktop PCs and laptops. The IMS Network Architecture is described as Fig. 2.2-1.. Figure 2.2-1 NGN IMS Network Architecture. The IMS supports the inter-working with mobile networks, PSTN and other circuit-switched networks for voice, and with corporate intranets, ISP networks and the Internet for support of packet oriented services including VoIP. The simplified view of the IMS and connected networks is shown on the Fig. 2.2-2.. Figure 2.2-2 IMS Network Components. 17.

(29) The IMS works together with any packet-based access network. It can be integrated with UMTS as well as with GPRS, EDGE, CDMA2000 and license-free hotspot radio technologies such as WLAN 802.11x. This flexibility is accomplished via a separation of access, transport, and control. The control is further separated in distinct entities into media control, session control and application control. For IP networks, IMS supports SIP calls from endpoints across either public or private networks. For calls across a public network, the Interconnect Border Control Function (IBCF) provides mainly NAPT (Network Address and Port Translation) and routing of signaling and media traffic through the provider's firewalls. These functionalities are usually implemented in devices called Session Border Controllers – SBC (Cumming, 2005; TS 182 009, 2006). Since SIP/RTP traffic is fundamental to the IMS, SS7/TDM calls from legacy networks have to be converted in media gateways. Media Gateway Control Function maps the SS7 signaling to SIP protocol and Media Gateway converts the TDM slots to RTP streams. The negotiation of media encoding is done directly between the gateways and the endpoints. Media Gateway Control Function controls the media gateways using H.248/Megaco interface. Media gateways are already widely used to bridge the TDM to VoIP domains. 2.2.3 IMS Core Network This section provides closer look at the core IMS network and describes the functions of the key IMS logical system components. The core IMS network is shown in Fig. 2.2-3.. Figure 2.2-3 IMS Core Network. 18.

(30) 2.2.3.1 Common Session Control Function Application services control. A central system component of the IMS network infrastructure is the Common Session Control Function (CSCF). One of the main purposes of the CSCF is the signaling routing function. CSCF proxies all SIP signaling traffic and provides the following network services: – – – – –. Session control services including subscription, registration, routing and roaming Combination of several different media bearer per session Central service based charging Secure authentication and confidentiality based on the ISIM/USIM Quality of service control (Policy Decision Function - PDF). Upon arrival of a new SIP call, the CSCF first authenticates the subscriber in the Home Subscriber Server (HSS). Then it passes the SIP signaling over the IP Multimedia Service Control interface – ISC (TS 181 010, 2005) to the application services. The ISC defines a set of filters that can be obtained from the HSS and assigned to each subscriber. In the ISC, each SIP message is compared to the appropriate filter and thus the CSCF decides to which application services to route the call and in what sequence. Additionally, each called application service may interrogate HSS. A decision is made how the call shall be handled according to the service-specific requirements (e.g. the called party's presence, the called/caller's preferences, terminal capabilities, or the called/caller's current credit, etc.). The IMS defines interfaces towards application services; nevertheless the behavior of applications is beyond the scope of IMS specifications. Although CSCF forwards SIP messages to the ISC interface to the applications, sometimes it needs to remain in control of the call. This is accomplished by adding some additional header information in the standard SIP messages (REGISTER, INVITE, etc.). If the application itself can fully handle the call request, then it informs CSCF and no further SIP routing is applied by CSCF. On the other hand, if it is required that more services be involved in the call handling, the scenario is as follows. The application of the first service returns the SIP message to the CSCF. The CSCF successively inspects the ISC filters again to check which is the next service in the chain, and finally it passes the message on. Applications independence. Another important role of CSCF is to provide independence between application services. The ISC interface with standard filters and SIP signaling assures independent development and deployment of multiple applications with no cross-dependences. Moreover, legacy application services might be integrated, too, as it should be possible to write these in Parlay interface and to place a Parlay gateway between CSCF and the legacy applications. CSCF components. In UMTS Release 6 of the 3GPP specifications, the IMS has been specified to be access independent. Consequently, it is of vital importance that the IMS-enabled network providers offer connections to their services from any foreign. 19.

(31) network. Full roaming scenarios have to be considered. Going deeper into IMS architecture, the CSCF splits into three controlling functions (TS 182 006, 2006) as depicted in Fig. 2.2-4. They are the Proxy-CSCF (PCSCF), the Interrogating-CSCF (I-CSCF) and the Serving-CSCF (S-SCSF).. Figure 2.2-4 CSCF control in IMS. The P-CSCF is the first point of contact within the IMS for a user. The P-CSCF may be located in the home or visited network. The P-CSCF ensures that registration of the user is passed to the correct home network and that SIP session messages are passed to the correct S-CSCF once registration has occurred. Contact with the home network during registration is through the home network I-CSCF and initial SIP session set-up is through called party I-CSCF. The P-CSCF provides even more interesting network features, for example it can detect services, which are hosted in the visitor's network. This may be wanted for access to content services, but very important for emergency services and lawful intercept. The P-CSCF also takes care of allocation of resources for the media flows, and it can provide defense against SIP signaling attacks. The Interrogation-CSCF (I-CSCF) is important for sessions from peered networks. This function determines the S-CSCF with which a user should register. This is achieved by querying the Home Subscriber Server (HSS), which checks that the user is allowed to register in the originating network and returns an S-CSCF name and capability. The I-SCSF is then able to contact the S-SCSF with the register. Once it has been determined which S-CSCF is in use, the I-CSCF might be removed from the signaling path. However, the I-SCSF may host the Topology Hiding Inter-network Gateway (THIG) function,. 20.

(32) which can act to hide the operator's topology from the peer networks. If the THIG is active, the I-CSCF remains involved in the signaling of the call. The Serving-CSCF (S-CSCF) is the central brain of the CSCF. The operation of the S-CSCF is controlled by the policy stored in the HSS. This logic is responsible for users' authentications, registrations and authorizations, for call processing including retrievals of service triggering information and user profiles from HSS, and finally for routing of calls to applications. The S-CSCF maintains session's counters and provides billing information to billing mediation systems. 2.2.3.2 Home Subscriber Server Functionality and interface. IMS specifications include a definition of subscriber profile database, a central repository for subscriber information called Home Subscriber Server – HSS (TS 181 010, 2005). The HSS maintains all subscriber information necessary for establishing sessions between users and for providing them with services. The HSS is an equivalent of HLR in GSM, however, it goes even further in its standardization. The interface to the HSS is denoted as Sh and it includes standards for protocol for communicating with HSS, the data stored in the HSS and even an XML-based mechanism for storing a subscriber's service-specific data. The HSS includes information about: – – – –. Subscriber registration (name, address, subscribed services, etc.); Subscriber preferences (baring information, forwarding settings, etc.); Subscriber location; Service-specific information.. The Sh interface is build upon the existing DIAMETER standard used for accessing subscriber information. Note that DIAMETER itself is an extension of RADIUS standard. Integration with existing databases. The IMS concept takes into account previous operators' investments in their network infrastructures and moderates future technology deployments in that extent. The HSS shall be viewed as a server that provides a consolidated interface to multiple sources of information, better than another big repository for subscriber data. By this, IMS can rely on subscriber information from multiple sources like HSS, network HLRs, location servers, business repositories, etc. The HSS can pull the data together under one interface, rather than replacing already existing repositories. 2.2.3.3 Media Resource Function Control and Media Server The majority of next-generation services require media processing that can be delivered by general purpose media servers. For example, these media servers are able to play audio prompts, convert text to speech, mix audio for conference calls, etc. They are often delivered with VoiceXML (VXML) technology. The IMS standard includes a system component called Media Resource Function Control (MRFC), which accepts instructions from an application service and directs the Media Server to handle the media stream.. 21.

(33) The MRFC controls the Media Server using the standard H.248/Megaco protocol. The separation of the MRCF from the Media Server is illustrated in Fig. 2.2-4. In practice the Media Server and the Media Gateway could merge into one component, as long as it is permitted by a system design for intended services (ES 282 004, 2006). The IMS specifications give MRFC also other responsibilities like charging of media server, conference call control and control of subscriber roaming. These can be equally accomplished by application services themselves or by CSCF. Moreover, since VXML-enabled media servers and application platforms often facilitate development and integration of new services, the future IMS architecture could evolve toward VXML-based interface standard between media servers and application services. This approach, i.e. without MRFC involvement, is shown in Fig. 2.2-5.. Figure 2.2-5 IMS with VXML control of media servers. 2.2.3.4 Charging Collection Function There are two interfaces specified by IMS for charging for next-generation services. The interface Rf allows for an off-line charging, whereas Ro interface provides on-line charging capabilities to support real-time charging models. Both interfaces use a common format of DIAMETER message, the ACCOUNTING REQUEST. Either both or one of the interfaces should be always supported by an IMS compliant application service. The main function of the IMS charging entity is to adopt a flexible structure of the call data records and to integrate with existing back-end business systems and on-line charging technologies. Offline charging is applied to users who pay for their services periodically (e.g., at the end of the month). Online charging, also known as credit-based charging, is used for 22.

(34) prepaid services, or real-time credit control of postpaid services. Both may be applied to the same session. Offline Charging : All the SIP network entities (P-CSCF, I-CSCF, S-CSCF, BGCF, MRFC, MGCF, AS) involved in the session use the DIAMETER Rf interface to send accounting information to a CCF (Charging Collector Function) located in the same domain. The CCF will collect all this information, and build a CDR (Call Detail Record), which is sent to the billing system (BS) of the domain. Each session carries an ICID (IMS Charging Identifier) as a unique identifier. IOI (Inter Operator Identifier) parameters define the originating and terminating networks. Each domain has its own charging network. Billing systems in different domains will also exchange information, so that roaming charges can be applied. Online charging : The S-CSCF talks to an SCF (Session Charging Function) which looks like a regular SIP application server. The SCF can signal the S-CSCF to terminate the session when the user runs out of credits during a session. The AS and MRFC use the DIAMETER Ro interface towards an ECF (Event Charging Function). When IEC (Immediate Event Charging) is used, a number of credit units is immediately deducted from the user's account by the ECF and the MRFC or AS is then authorized to provide the service. The service is not authorized when not enough credit units are available. When ECUR (Event Charging with Unit Reservation) is used, the ECF first reserves a number of credit units in the user's account and then authorizes the MRFC or the AS. After the service is over, the number of spent credit units is reported and deducted from the account; the reserved credit units are then cleared. Interfaces description Table 2.2-1 NGN IMS Interface description. Interface Name. IMS entities. Description. Protocol HTTP over dedicated TCP/SCTP channels. Cr. MRFC, AS. Used by MRFC to fetch documents (scripts and other resources) from an AS. Cx. I-CSCF, S-CSCF, HSS. Used to communicate between I-CSCF/S-CSCF and HSS. DIAMETER. Dh. SIP AS, OSA, SCF, IM-SSF, HSS. Used by AS to find a correct HSS in a multi-HSS environment. DIAMETER. Dx. I-CSCF, S-CSCF, SLF. Used by I-CSCF/S-CSCF to find a correct HSS in a multi-HSS. DIAMETER. 23.

(35) environment UE, P-CSCF. Used to exchange messages between UE and CSCFs. SIP. Go. PDF, GGSN. Allows operators to control QoS in a user plane and exchange charging correlation information between IMS and GPRS network. COPS (Rel5), DIAMETER (Rel6+). Gq. P-CSCF, PDF. Used to exchange policy decisions-related information between P-CSCF and PDF. DIAMETER. ISC. S-CSCF, I-CSCF, AS. Used to exchange messages between CSCF and AS. SIP. Mg. MGCF -> I-CSCF. MGCF converts ISUP signalling to SIP signalling and forwards SIP signalling SIP to I-CSCF. Mi. S-CSCF -> BGCF. Used to exchange messages between S-CSCF and BGCF. SIP. Mj. BGCF -> MGCF. Used to exchange messages between BGCF and MGCF in the same IMS network. SIP. Mk. BGCF -> BGCF. Used to exchange messages between BGCFs in different IMS networks. SIP. Mm. I-CSCF, S-CSCF, external IP network. Used for exchanging messages between Not specified IMS and external IP networks. Mn. MGCF, IM-MGW. Allows control of user-plane resources. H.248. Mp. MRFC, MRFP. Used to exchange messages between MRFC and MRFP. H.248. Mr. S-CSCF, MRFC. Used to exchange messages between S-CSCF and MRFC. SIP. Mw. P-CSCF, I-CSCF, S-CSCF. Used to exchange messages between CSCFs. SIP. Sh. SIP AS, OSA SCS, HSS. Used to exchange information between DIAMETER SIP AS/OSA SCS and HSS. Si. IM-SSF, HSS. Used to exchange information between MAP IM-SSF and HSS. Sr. MRFC, AS. Used by MRFC to fetch documents. Gm. 24. HTTP.

(36) (scripts and other resources) from an AS Ut. 2.3. UE, AS (SIP Enables UE to manage information AS, OSA SCS, related to his services IM-SSF). HTTP(s). Voice over Internet Protocol (VoIP) Service. Voice over Internet Protocol (VoIP), is a technology that allows users to make voice calls using a broadband Internet connection instead of a regular (or analog) phone line. Some VoIP services may only allow users to call other people using the same service, but others may allow users to call anyone who has a telephone number - including local, long distance, mobile, and international numbers. Also, while some VoIP services only work over desktop computer or a special VoIP phone, other services allow users to use a traditional phone connected to a VoIP adapter. 2.3.1 How VoIP Works VoIP converts the voice signal from the telephone into a digital signal that travels over the Internet. If users are calling a regular telephone number, the signal is then converted back at the other end. Users can make a VoIP call from a computer, a special VoIP phone, or a traditional phone using an adapter. In addition, new wireless "hot spots" in public locations such as airports, parks, and cafes allow users to connect to the Internet, and may enable them to use VoIP service wirelessly. If users make a call using a telephone with an adapter, they’ll be able to dial just as always have. If the VoIP service provider assigns a regular telephone number, then users can receive calls from regular telephones that don’t need special equipment. Here is one example of how VoIP service works:. 25.

(37) Figure 2.3-1 Basic functions of a VoIP system. 2.3.2 The Advantages of VoIP Service VoIP may offer features and services that are not available with more traditional telephone services. If people use VoIP, they can decide whether to pay the cost of keeping the regular telephone service. They can also use desktop computers and VoIP service at the same time. Assuming users have an Internet connection and the necessary equipment, they can also take the VoIP service with them when travelling. Some VoIP providers do not charge for calls to other subscribers to the service. Some VoIP providers charge for a long distance call to a number outside the calling area, similar to existing, traditional wireline telephone service. Other VoIP providers permit users to call anywhere at a flat rate for a fixed number of minutes. The VoIP service provider may permit users to select an area code for the VoIP service that is different from the area code in which they live. Calls within the VoIP area code may not be billed as long distance calls. People calling the VoIP area code from another area code, however, may incur long distance charges. VoIP services allow users generally to receive calls from, and terminate calls to, the Public Switched Telephone Network (PSTN), including wireless networks. 2.3.3 The Top 10 VoIP Service Providers in Y2006 Point Topic Ltd is a UK-based company founded in 1998. The company mission is to provide focused information on broadband communications services. Now the www.point-topic.com website is internationally recognised as one of the best sources on broadband. It provides subscribers with regularly updated online databases and reports about broadband services around the world. Summary information is available for free to any visitor to the site.. 26.

(38) The Top 10 VoIP Service Providers in the world by subscription count is listed below by the end of 2006 which provided by Point Topic. 1. Skype - 5,300,000 2. Yahoo! Japan - 4,517,000 3. VoiceGlo - 605,000 4. Free (France) - 600,000 5. Vonage - 535,000. 6. FastWeb (Italy) - 528,000 7. Cox - 413,000 8. Time Warner - 372,000 9. Cablevision - 364,000 10. Neuf (France) - 297,000 According to Point Topic, there were over 11 million people using retail VoIP service for at least some of their telephone calls at the end of March 2005. That’s an increase from just over 5 million at mid-2004 according to Point Topic research published in December 2004. This number is based on a total of publicly available figures for VoIP subscribers, combined with estimates where subscriber numbers were not available. Many of these estimates are based on a conservative 10 percent growth from Q4 2004 to Q1 2005. The total does not include PC-based ‘soft-client’ services like Skype and VoiceGlo. The release goes on to say that more than half of that number is in Japan. Yahoo! Softbank provides the majority of the services to the 7.2 million Japanese VoIP subscribers. After Japan, the American cable sector is numerically the most important VoIP sector, with around 2.1 million subscribers. France is the largest market for VoIP in Europe, with 1.2 million subscribers by the end of the first quarter. Most of these lines are provisioned by Free and Neuf. 2.3.4 Retail VoIP subscribers market information during 2006 Retail voice subscriber numbers more than doubled during 2006, from 19 million to 40 million. Point Topic has been tracking the phone-to-phone VoIP market for several years and is the only research company to publish detailed operator-by-operator figures for this market worldwide. Overview 2006 was the year when VoIP (Voice over Internet Protocol) really began to gain traction in many consumer markets. Japan was the leading VoIP country during 2006 with 13.75 million subscribers. The USA had 8.9 million subscribers, and France reported 6.6 million. In regulatory terms, VoIP is legal and regulated in a variety of ways in the developed world. Many countries, such as Japan, the USA, France and Sweden have adopted a liberal approach to VoIP. Some countries continue to consult on how best to regulate VoIP. VoIP is illegal in some developing countries. Bundling VoIP with broadband and television is the most common way of selling VoIP services, although standalone providers such as Vonage and Telio provide an alternative route to VoIP service.. 27.

(39) Asia Pacific Japan has a liberal regime towards VoIP. It also has Softbank, a broadband service provider with its own IP network that pioneered mass VoIP sales. The rapid success of Softbank encouraged NTT to develop its own VoIP offerings. In South Korea, VoIP services officially started in 2005, but subscriber numbers are not as significant as those in Japan. In China, VoIP is still officially still in a trial stage, with PC-to-PC services illegal. Given the relatively high long distance calling costs in China, it is likely that there is considerable unofficial VoIP activity, although PC-to-PSTN numbers are probably relatively low. Americas In the USA and Canada, cable companies (sometimes called MSOs, or multiple systems operators) continue to make the running in VoIP. VoIP or ‘digital voice’ services, as they are often marketed, enable cable companies to offer triple-play to their subscribers. This helps cable companies to protect against customer poaching by telcos that are now starting to offer video services (such as Verizon’s service over its FiOS fibre network). In some cases, the growth rate is high. Comcast added 1.5 million voice customers during 2006, almost quadrupling its total voice subscriber count. Europe Within France, VoIP was responsible for almost a quarter of voice traffic originating from fixed phones. French regulator ARCEP reported 6.6 million VoIP subscriptions at the end of 2006. Of these, 4.3 million were subscriptions in addition to an existing PSTN subscription. But 2.3 million VoIP subscriptions were over a fully unbundled (or ‘naked’) line, with no PSTN service on that line. The growth in VoIP has stopped the decline in fixed subscriptions (caused by people abandoning fixed telephony altogether in favour of mobile). Germany reported 3.5 million VoIP subscriptions, whilst KPN estimates that there are 1.4 million VoIP subscriptions in the Netherlands. The situation in the UK is more confused. BT reported it had over 1 million VoIP subscriptions by Q3 2006, whereas regulator Ofcom estimated 300,000 in August 2006, with around 150,000 subscriptions with other operators. It is possible that Ofcom’s survey is underestimating VoIP usage, or it is possible that some of the BT reported lines are not being regularly used addition to a BT PSTN service. It is one example of the difficulty of compiling reliable figures on VoIP.. 2.4. Traffic Dimensioning Theory for Telephony Service. The traffic dimensioning unit named the Erlang is used in telecommunication management as a statistical measure of the volume of telecommunications traffic. It is named after the Danish telephone engineer A. K. Erlang, the originator of traffic engineering and queuing theory (Mina, 1974). Traffic of one Erlang refers to a single resource being in continuous use, or two channels being at fifty percent use, and so on. For example, if an office had two telephone operators who are both busy all the time, that. 28.

(40) would represent two Erlangs of traffic. Alternatively, an Erlang may be regarded as a "use multiplier" per unit time, so 100% use is 1 Erlang, 200% use is 2 Erlangs, and so on. For example, if total cell phone use in a given area per hour is 180 minutes, this represents 180/60 = 3 Erlangs. In general, if the mean arrival rate of new calls is λ per unit time and the mean call holding time is h, then the traffic in Erlangs A is: A = λh This may be used to determine if a system is over-provisioned or under-provisioned (has too many or too few resources allocated). For example, the traffic measured over many busy hours might be used for a T1 or E1 circuit group to determine how many voice lines are likely to be used during the busiest hours. If no more than 12 out of 24 channels are likely to be used at any given time, the other 12 might be made available as data channels. Traffic measured in Erlangs is used to calculate grade of service (GOS) or quality of service (QoS) (Martin, 1972). The Erlang B formula can be derived by means of a special case of continuous-time Markov processes known as a birth-death process. The Erlang B formula assumes an infinite population of sources (such as telephone subscribers), which jointly offer traffic to N servers (such as links in a trunk group). The rate of arrival of new calls (birth rate) is equal to λ and is constant, not depending on the number of active sources, because the total number of sources is assumed to be infinite. The rate of call departure (death rate) is equal to the number of calls in progress divided by h, the mean call holding time. The formula calculates blocking probability in a loss system, where if a request is not served immediately when it tries to use a resource, it is aborted. Requests are therefore not queued. Blocking occurs when there is a new request from a source, but all the servers are already busy. The formula assumes that blocked traffic is immediately cleared.This may be expressed recursively as follows, in a form that is used to calculate tables of the Erlang B formula:. where:. B is the probability of blocking N is the number of resources such as servers or circuits in a group A = λh is the total amount of traffic offered in Erlangs The Erlang B formula applies to loss systems, such as telephone systems on both fixed and mobile networks, which do not provide traffic buffering, and are not intended to 29.

(41) do so. It assumes that the call arrivals may be modeled by a Poisson process, but is valid for any statistical distribution of call holding times.. 30.