Creation of value with open source software in the telecommunications field

Tese de doutoramento. Engenharia Electrotécnica e de Computadores. Faculdade de Engenharia. Universidade do Porto. 200

A Unified Mobility Management Architecture for Interworked Heterogeneous Mobile Networks

The buzzword of this decade has been convergence: the convergence of telecommunications, Internet, entertainment, and information technologies for the seamless provisioning of multimedia services across different network types. Thus the future Next Generation Mobile Network (NGMN) can be envisioned as a group of co-existing heterogeneous mobile data networking technologies sharing a common Internet Protocol (IP) based backbone. In such all-IP based heterogeneous networking environments, ongoing sessions from roaming users are subjected to frequent vertical handoffs across network boundaries. Therefore, ensuring uninterrupted service continuity during session handoffs requires successful mobility and session management mechanisms to be implemented in these participating access networks. Therefore, it is essential for a common interworking framework to be in place for ensuring seamless service continuity over dissimilar networks to enable a potential user to freely roam from one network to another. For the best of our knowledge, the need for a suitable unified mobility and session management framework for the NGMN has not been successfully addressed as yet. This can be seen as the primary motivation of this research. Therefore, the key objectives of this thesis can be stated as: To propose a mobility-aware novel architecture for interworking between heterogeneous mobile data networks To propose a framework for facilitating unified real-time session management (inclusive of session establishment and seamless session handoff) across these different networks. In order to achieve the above goals, an interworking architecture is designed by incorporating the IP Multimedia Subsystem (IMS) as the coupling mediator between dissipate mobile data networking technologies. Subsequently, two different mobility management frameworks are proposed and implemented over the initial interworking architectural design. The first mobility management framework is fully handled by the IMS at the Application Layer. This framework is primarily dependant on the IMS’s default session management protocol, which is the Session Initiation Protocol (SIP). The second framework is a combined method based on SIP and the Mobile IP (MIP) protocols, which is essentially operated at the Network Layer. An analytical model is derived for evaluating the proposed scheme for analyzing the network Quality of Service (QoS) metrics and measures involved in session mobility management for the proposed mobility management frameworks. More precisely, these analyzed QoS metrics include vertical handoff delay, transient packet loss, jitter, and signaling overhead/cost. The results of the QoS analysis indicates that a MIP-SIP based mobility management framework performs better than its predecessor, the Pure-SIP based mobility management method. Also, the analysis results indicate that the QoS performances for the investigated parameters are within acceptable levels for real-time VoIP conversations. An OPNET based simulation platform is also used for modeling the proposed mobility management frameworks. All simulated scenarios prove to be capable of performing successful VoIP session handoffs between dissimilar networks whilst maintaining acceptable QoS levels. Lastly, based on the findings, the contributions made by this thesis can be summarized as: The development of a novel framework for interworked heterogeneous mobile data networks in a NGMN environment. The final design conveniently enables 3G cellular technologies (such as the Universal Mobile Telecommunications Systems (UMTS) or Code Division Multiple Access 2000 (CDMA2000) type systems), Wireless Local Area Networking (WLAN) technologies, and Wireless Metropolitan Area Networking (WMAN) technologies (e.g., Broadband Wireless Access (BWA) systems such as WiMAX) to interwork under a common signaling platform. The introduction of a novel unified/centralized mobility and session management platform by exploiting the IMS as a universal coupling mediator for real-time session negotiation and management. This enables a roaming user to seamlessly handoff sessions between different heterogeneous networks. As secondary outcomes of this thesis, an analytical framework and an OPNET simulation framework are developed for analyzing vertical handoff performance. This OPNET simulation platform is suitable for commercial use

Wireless triple play system

Dissertação para obtenção do Grau de Mestre em Engenharia Electrotécnica e ComputadoresTriple play is a service that combines three types of services: voice, data and multimedia over a single communication channel for a price that is less than the total price of the individual services. However there is no standard for provisioning the Triple play services, rather they are provisioned individually, since the requirements are quite different for each service. The digital revolution helped to create and deliver a high quality media solutions. One of the most demanding services is the Video on Demand (VoD). This implicates a dedicated streaming channel for each user in order to provide normal media player commands (as pause, fast forward). Most of the multimedia companies that develops personalized products does not always fulfil the users needs and are far from being cheap solutions. The goal of the project was to create a reliable and scalable triple play solution that works via Wireless Local Area Network (WLAN), fully capable of dealing with the existing state of the art multimedia technologies only resorting to open-source tools. This project was design to be a transparent web environment using only web technologies to maximize the potential of the services. HyperText Markup Language (HTML),Cascading Style Sheets (CSS) and JavaScript were the used technologies for the development of the applications. Both a administration and user interfaces were developed to fully manage all video contents and properly view it in a rich and appealing application, providing the proof of concept. The developed prototype was tested in a WLAN with up to four clients and the Quality of Service (QoS) and Quality of Experience (QoE) was measured for several combinations of active services. In the end it is possible to acknowledge that the developed prototype was capable of dealing with all the problems of WLAN technologies and successfully delivery all the proposed services with high QoE

Study, Analysis and Modeling of IP Multimedia systems on next generation networks providing mobile and fixed multimedia services

Not availabl

Mobile Networks

The growth in the use of mobile networks has come mainly with the third generation systems and voice traffic. With the current third generation and the arrival of the 4G, the number of mobile users in the world will exceed the number of landlines users. Audio and video streaming have had a significant increase, parallel to the requirements of bandwidth and quality of service demanded by those applications. Mobile networks require that the applications and protocols that have worked successfully in fixed networks can be used with the same level of quality in mobile scenarios. Until the third generation of mobile networks, the need to ensure reliable handovers was still an important issue. On the eve of a new generation of access networks (4G) and increased connectivity between networks of different characteristics commonly called hybrid (satellite, ad-hoc, sensors, wired, WIMAX, LAN, etc.), it is necessary to transfer mechanisms of mobility to future generations of networks. In order to achieve this, it is essential to carry out a comprehensive evaluation of the performance of current protocols and the diverse topologies to suit the new mobility conditions

Investigation of an intelligent personalised service recommendation system in an IMS based cellular mobile network

Success or failure of future information and communication services in general and mobile communications in particular is greatly dependent on the level of personalisations they can offer. While the provision of anytime, anywhere, anyhow services has been the focus of wireless telecommunications in recent years, personalisation however has gained more and more attention as the unique selling point of mobile devices. Smart phones should be intelligent enough to match user’s unique needs and preferences to provide a truly personalised service tailored for the individual user. In the first part of this thesis, the importance and role of personalisation in future mobile networks is studied. This is followed, by an agent based futuristic user scenario that addresses the provision of rich data services independent of location. Scenario analysis identifies the requirements and challenges to be solved for the realisation of a personalised service. An architecture based on IP Multimedia Subsystem is proposed for mobility and to provide service continuity whilst roaming between two different access standards. Another aspect of personalisation, which is user preference modelling, is investigated in the context of service selection in a multi 3rd party service provider environment. A model is proposed for the automatic acquisition of user preferences to assist in service selection decision-making. User preferences are modelled based on a two-level Bayesian Metanetwork. Personal agents incorporating the proposed model provide answers to preference related queries such as cost, QoS and service provider reputation. This allows users to have their preferences considered automatically

Extending AES with DH Key-Exchange to Enhance VoIP Encryption in Mobile Networks

Recently,the evolution and progress have become significant in the field of information technology and mobile technology, especially inSmartphone applications that are currently widely spreading. Due to the huge developments in mobile and smartphone technologies in recent years, more attention is given to voice data transmission such as VoIP (Voice overIP) technologies– e.g. (WhatsApp, Skype, and Face Book Messenger). When using VoIP services over smartphones, there are always security and privacy concerns like the eavesdropping of calls between the communicating parties. Therefore, there is a pressing need to address these risks by enhancing the security level and encryption methods. In this work, we use scheme to encrypt VoIP channels using (128, 192 & 256-bit) enhanced encryption based on the Advanced Encryption Standard (AES) algorithm, by extending it with the well-known Diffie-Hellman (DH) key exchange method. We have performed a series of real tests on the enhanced (AES-DH) algorithm and compared its performance with the generic AES algorithm. The results have shown that we can get a significant increase in the encryption strength at a very small overhead between 4% and 7% of execution timebetween AES and AEScombine withDH for all scenario which was incurred by added time of encryption and decryption. Our approach uses high security and speed and reduces the voice delay.In dealing with sound transfer process via the internet, we use the SIP server to authenticate the communication process between the two parties. The implementation is done on a mobile device (Which is operated by (Android) system) because it has recently been widely used among different people around the world.اصبحت الثورة والتطور كبيرة حديثاً في حقول تكنولوجيا االتصاالت واليواتف النقالة، وخصوصا في تطبيقات اليواتف الذكية التي تنتشر حاليا بشكل واسع. وتم اعطاء المزيد من االىتمام لنقل البيانات الصوتية مثل تكنولوجيا االتصال عبر بروتكول االنترنت، عمى سبيل المثال: )الواتساب، السكايب، الفيس بوك، والماسنجر(. ويعزى ذلك لمتطور الكبير في تكنولوجيا اليواتف النقالة والذكية في السنوات االخيرة. عند استخدام خدمات االتصال عبر بروتكول االنترنت VoIP ،ىناك مخاوف دائمة حول الحماية والخصوصية كالتجسس عمى المكالمات بين جيات االتصال. ولذلك ىنالك حاجة ماسة لمعالجة ىذه المخاطر عن طريق تحسين مستوى الحماية وطرق التشفير. في ىذا العمل، نستخدم/ نجمع بين اثنتين من الخوارزميات لتشفير قنوات االتصال عبر بروتوكول االنترنت )128 ، 192 ،و 256 بت( عبر خوارزمية AESوتمديدىا عبر طريقة تبادل ديفي ىيممان الرئيسية المعروفة. وقمنا باداء العديد من التجارب الحقيقية عمى DH-AES ، وقمنا بمقارنة ادائيا مع اداء خوارزمية معيار التشفير المتقدم العامة. اظيرت النتائج انو بامكاننا الحصول عمى زيادة كبيرة في قوة التشفير بنسبة صغيرة جدا بين 4 %و7 %من وقت التنفيذ بين AESو DH/AES لجميع السيناريو والتي تم تكبدىا من قبل الوقت المضاف لمتشفير وفك التشفير. يستخدم نيجنا درجة عالية من الحماية والسرعة ويقمل من تأخير الصوت، ونستخدم في التعامل مع عممية نقل الصوت عبر االنترنت Server SIPلتوثيق عممية االتصال بين الجيتين. وتم التنفيذ عمى ىاتف نقال يعمل عمى نظام اندرويد؛ النو استخدم بشكل واسع مؤخرا بين مختمف الناس حول العالم

Multimedia delivery in the future internet

The term “Networked Media” implies that all kinds of media including text, image, 3D graphics, audio and video are produced, distributed, shared, managed and consumed on-line through various networks, like the Internet, Fiber, WiFi, WiMAX, GPRS, 3G and so on, in a convergent manner [1]. This white paper is the contribution of the Media Delivery Platform (MDP) cluster and aims to cover the Networked challenges of the Networked Media in the transition to the Future of the Internet. Internet has evolved and changed the way we work and live. End users of the Internet have been confronted with a bewildering range of media, services and applications and of technological innovations concerning media formats, wireless networks, terminal types and capabilities. And there is little evidence that the pace of this innovation is slowing. Today, over one billion of users access the Internet on regular basis, more than 100 million users have downloaded at least one (multi)media file and over 47 millions of them do so regularly, searching in more than 160 Exabytes1 of content. In the near future these numbers are expected to exponentially rise. It is expected that the Internet content will be increased by at least a factor of 6, rising to more than 990 Exabytes before 2012, fuelled mainly by the users themselves. Moreover, it is envisaged that in a near- to mid-term future, the Internet will provide the means to share and distribute (new) multimedia content and services with superior quality and striking flexibility, in a trusted and personalized way, improving citizens’ quality of life, working conditions, edutainment and safety. In this evolving environment, new transport protocols, new multimedia encoding schemes, cross-layer inthe network adaptation, machine-to-machine communication (including RFIDs), rich 3D content as well as community networks and the use of peer-to-peer (P2P) overlays are expected to generate new models of interaction and cooperation, and be able to support enhanced perceived quality-of-experience (PQoE) and innovative applications “on the move”, like virtual collaboration environments, personalised services/ media, virtual sport groups, on-line gaming, edutainment. In this context, the interaction with content combined with interactive/multimedia search capabilities across distributed repositories, opportunistic P2P networks and the dynamic adaptation to the characteristics of diverse mobile terminals are expected to contribute towards such a vision. Based on work that has taken place in a number of EC co-funded projects, in Framework Program 6 (FP6) and Framework Program 7 (FP7), a group of experts and technology visionaries have voluntarily contributed in this white paper aiming to describe the status, the state-of-the art, the challenges and the way ahead in the area of Content Aware media delivery platforms

Integrating Voice over IP Solution in IPv6 and IPv4 Networks to Increase Employee Productivity: A Case Study of Cameroon Telecommunications (Camtel), North-West

Telecommunications organizations have to follow the rapid innovation of technology if they want to face challenges raised by competition. The challenge to respond to the huge market demand of updated products and services from customers requires that the organization‘s working environment be equipped with tools and communication facilities that contribute to ameliorating productivity. Cameroon Telecommunications (Camtel) is facing a digital telephony and Internet Protocol strategic management challenge. Successful implementation cannot be achieved if the employees are still depending on the ageing public switched telephone network (PSTN) as their primary communication system, despite the frequent loss of dial tone experience in a day which can last up to a week, with serious repercussions on business activities and revenues. This study is designed to provide a solution to the telecommunications challenge. The fundamental question is how to integrate a digital telephony system that will provide telephony services in the existing IPv4 data network while prioritizing IPv6 traffic forwarding. This study proposes and implements solutions that integrate a Voice over IP solution with IPv6 as an alternative communication system that relies on the existing IPv4 data network. VoIP is deemed as one of the driving forces behind the adoption of IPv6. The purpose is to offer to workers an option that will free them from the poor Quality of Service (QoS) of their existing PSTN based solution, hopefully enhancing the overall productivity. This paper follows two research methodologies: Qualitative Research in Applied Situations and Engineering design process. The first part of this study reports the results of the evaluation of how much such a solution can enhance workers’ productivity. As it is important to provide an environment where IPv4 and IPv6 networks and applications/devices can interoperate in the context of VoIP; the second part describes practically a simulation environment where various configurations of network entities are done following a Dual-Stack transition approach. Document and records were used to gather information related to the structure, operations, and topological update of the Camtel’s existing IP data network. The findings demonstrated that VoIP can be an effective communication solution for Camtel and its implementation with IPv6 will be preferable. However, for this to be efficient there must be a provision of sufficient bandwidth and usage of types of equipment and transmission mediums that minimizes processing and propagation delays. Findings also reveal that better productivity will be achieved if workers are fully trained for the exploitation. This research article tries to highlight, discuss a required transition roadmap and extend the local knowledge and practice on IPv6. Future expansion of this research work will consist of deploying Dual-Stack VoIP in the remaining 9 regional offices for full integration in the corporate communication system of Camtel