A cooperative cellular and broadcast conditional access system for Pay-TV systems

This is the author's accepted manuscript. The final published article is available from the link below. Copyright @ 2009 IEEE. Personal use of this material is permitted. Permission from IEEE must be obtained for all other users, including reprinting/ republishing this material for advertising or promotional purposes, creating new collective works for resale or redistribution to servers or lists, or reuse of any copyrighted components of this work in other works.The lack of interoperability between Pay-TV service providers and a horizontally integrated business transaction model have compromised the competition in the Pay-TV market. In addition, the lack of interactivity with customers has resulted in high churn rate and improper security measures have contributed into considerable business loss. These issues are the main cause of high operational costs and subscription fees in the Pay-TV systems. As a result, this paper presents the Mobile Conditional Access System (MICAS) as an end-to-end access control solution for Pay-TV systems. It incorporates the mobile and broadcasting systems and provides a platform whereby service providers can effectively interact with their customers, personalize their services and adopt appropriate security measurements. This would result in the decrease of operating expenses and increase of customers' satisfaction in the system. The paper provides an overview of state-of-the-art conditional access solutions followed by detailed description of design, reference model implementation and analysis of possible MICAS security architectures.Strategy & Technology (S&T) Lt

Optimizing IETF multimedia signaling protocols and architectures in 3GPP networks : an evolutionary approach

Signaling in Next Generation IP-based networks heavily relies in the family of multimedia signaling protocols defined by IETF. Two of these signaling protocols are RTSP and SIP, which are text-based, client-server, request-response signaling protocols aimed at enabling multimedia sessions over IP networks. RTSP was conceived to set up streaming sessions from a Content / Streaming Server to a Streaming Client, while SIP was conceived to set up media (e.g.: voice, video, chat, file sharing, …) sessions among users. However, their scope has evolved and expanded over time to cover virtually any type of content and media session. As mobile networks progressively evolved towards an IP-only (All-IP) concept, particularly in 4G and 5G networks, 3GPP had to select IP-based signaling protocols for core mobile services, as opposed to traditional SS7-based protocols used in the circuit-switched domain in use in 2G and 3G networks. In that context, rather than reinventing the wheel, 3GPP decided to leverage Internet protocols and the work carried on by the IETF. Hence, it was not surprise that when 3GPP defined the so-called Packet-switched Streaming Service (PSS) for real-time continuous media delivery, it selected RTSP as its signaling protocol and, more importantly, SIP was eventually selected as the core signaling protocol for all multimedia core services in the mobile (All-)IP domain. This 3GPP decision to use off-the-shelf IETF-standardized signaling protocols has been a key cornerstone for the future of All-IP fixed / mobile networks convergence and Next Generation Networks (NGN) in general. In this context, the main goal of our work has been analyzing how such general purpose IP multimedia signaling protocols are deployed and behave over 3GPP mobile networks. Effectively, usage of IP protocols is key to enable cross-vendor interoperability. On the other hand, due to the specific nature of the mobile domain, there are scenarios where it might be possible to leverage some additional “context” to enhance the performance of such protocols in the particular case of mobile networks. With this idea in mind, the bulk of this thesis work has consisted on analyzing and optimizing the performance of SIP and RTSP multimedia signaling protocols and defining optimized deployment architectures, with particular focus on the 3GPP PSS and the 3GPP Mission Critical Push-to-Talk (MCPTT) service. This work was preceded by a detailed analysis work of the performance of underlying IP, UDP and TCP protocol performance over 3GPP networks, which provided the best baseline for the future work around IP multimedia signaling protocols. Our contributions include the proposal of new optimizations to enhance multimedia streaming session setup procedures, detailed analysis and optimizations of a SIP-based Presence service and, finally, the definition of new use cases and optimized deployment architectures for the 3GPP MCPTT service. All this work has been published in the form of one book, three papers published in JCR cited International Journals, 5 articles published in International Conferences, one paper published in a National Conference and one awarded patent. This thesis work provides a detailed description of all contributions plus a comprehensive overview of their context, the guiding principles beneath all contributions, their applicability to different network deployment technologies (from 2.5G to 5G), a detailed overview of the related OMA and 3GPP architectures, services and design principles. Last but not least, the potential evolution of this research work into the 5G domain is also outlined as well.Els mecanismes de Senyalització en xarxes de nova generació es fonamenten en protocols de senyalització definits per IETF. En particular, SIP i RTSP són dos protocols extensibles basats en missatges de text i paradigma petició-resposta. RTSP va ser concebut per a establir sessions de streaming de continguts, mentre SIP va ser creat inicialment per a facilitar l’establiment de sessions multimèdia (veu, vídeo, xat, compartició) entre usuaris. Tot i així, el seu àmbit d’aplicació s’ha anat expandint i evolucionant fins a cobrir virtualment qualsevol tipus de contingut i sessió multimèdia. A mesura que les xarxes mòbils han anat evolucionant cap a un paradigma “All-IP”, particularment en xarxes 4G i 5G, 3GPP va seleccionar els protocols i arquitectures destinats a gestionar la senyalització dels serveis mòbils presents i futurs. En un moment determinat 3GPP decideix que, a diferència dels sistemes 2G i 3G que fan servir protocols basats en SS7, els sistemes de nova generació farien servir protocols estandarditzats per IETF. Quan 3GPP va començar a estandarditzar el servei de Streaming sobre xarxes mòbils PSS (Packet-switched Streaming Service) va escollir el protocol RTSP com a mecanisme de senyalització. Encara més significatiu, el protocol SIP va ser escollit com a mecanisme de senyalització per a IMS (IP Multimedia Subsystem), l’arquitectura de nova generació que substituirà la xarxa telefònica tradicional i permetrà el desplegament de nous serveis multimèdia. La decisió per part de 3GPP de seleccionar protocols estàndards definits per IETF ha representat una fita cabdal per a la convergència del sistemes All-IP fixes i mòbils, i per al desenvolupament de xarxes NGN (Next Generation Networks) en general. En aquest context, el nostre objectiu inicial ha estat analitzar com aquests protocols de senyalització multimèdia, dissenyats per a xarxes IP genèriques, es comporten sobre xarxes mòbils 3GPP. Efectivament, l’ús de protocols IP és fonamental de cara a facilitar la interoperabilitat de solucions diferents. Per altra banda, hi ha escenaris a on és possible aprofitar informació de “context” addicional per a millorar el comportament d’aquests protocols en al cas particular de xarxes mòbils. El cos principal del treball de la tesi ha consistit en l’anàlisi i optimització del rendiment dels protocols de senyalització multimèdia SIP i RTSP, i la definició d’arquitectures de desplegament, amb èmfasi en els serveis 3GPP PSS i 3GPP Mission Critical Push-to-Talk (MCPTT). Aquest treball ha estat precedit per una feina d’anàlisi detallada del comportament dels protocols IP, TCP i UDP sobre xarxes 3GPP, que va proporcionar els fonaments adequats per a la posterior tasca d’anàlisi de protocols de senyalització sobre xarxes mòbils. Les contribucions inclouen la proposta de noves optimitzacions per a millorar els procediments d’establiment de sessions de streaming multimèdia, l’anàlisi detallat i optimització del servei de Presència basat en SIP i la definició de nous casos d’ús i exemples de desplegament d’arquitectures optimitzades per al servei 3GPP MCPTT. Aquestes contribucions ha quedat reflectides en un llibre, tres articles publicats en Revistes Internacionals amb índex JCR, 5 articles publicats en Conferències Internacionals, un article publicat en Congrés Nacional i l’adjudicació d’una patent. La tesi proporciona una descripció detallada de totes les contribucions, així com un exhaustiu repàs del seu context, dels principis fonamentals subjacents a totes les contribucions, la seva aplicabilitat a diferents tipus de desplegaments de xarxa (des de 2.5G a 5G), així una presentació detallada de les arquitectures associades definides per organismes com OMA o 3GPP. Finalment també es presenta l’evolució potencial de la tasca de recerca cap a sistemes 5G.Postprint (published version

Integrating Context-Awareness in the IP Multimedia Subsystem for Enhanced Session Control and Service Provisioning Capabilities

The 3GPP-defined IP Multimedia Subsystem (IMS) is becoming the de-facto standard for IP-based multimedia communication services. It consists of an overlay control and service layer that is deployed on top of IP-based mobile and fixed networks. This layer encompasses a set of common functions (e.g. session control functions allowing the initiation/modification/termination of sessions) and service logics that are needed for the seamless provisioning of IP multimedia services to users, via different access technologies. As it continues to evolve, the IMS still faces several challenges including: the enabling of innovative and personalized services that would appeal to users and increase network operators' revenues; its interaction with other types of networks (e.g. wireless sensor networks) as means to enhance its capabilities; and the support of advanced QoS schemes that would manage the network resources in an efficient and adaptive manner. The context-awareness concept, which comes from the pervasive computing field, signifies the ability to use situational information (or context) in support to operations and decision making and for the provision of relevant services to the user. Context-awareness is considered to enhance users' experience and is seen as an enabler to adaptability and service personalization - two capabilities that could play important roles in telecommunication environments. This thesis focuses on the introduction of the context-awareness technology in the IMS, as means to enhance its session control and service provisioning capabilities. It starts by presenting the necessary background information, followed by a derivation of requirements and a review of the related work. To ensure the availability of contextual information within the network, we then propose an architecture for context information acquisition and management in the IMS. This architecture leverages and extends the 3GPP presence framework. Building on the capabilities of this architecture, we demonstrate how the managed information could be integrated in IMS operations, at the control and service levels. Showcasing control level integration, we propose a novel context-aware call differentiation framework as means to offer enhanced QoS support (for sessions/calls) in IMS-based networks. This framework enables the differentiation between different categories of calls at the IMS session control level, via dynamic and adaptive resource allocation, in addition to supporting a specialized charging model. Furthermore, we also propose a framework for enhanced IMS emergency communication services. This framework addresses the limitations of existing IP-based emergency solutions, by offering three main improvements: a QoS-enhanced emergency service; a context-aware personalized emergency service; and a conferencing-enhanced emergency service. We demonstrate the use of context awareness at the IMS service level using two new context-aware IMS applications. Finally, to validate our solutions and evaluate their performance, we build various proof-of-concept prototypes and OPNET simulation model

The evolution of public safety communications in Europe: the results from the FP7 HELP project

This paper describes the results from the FP7 HELP project, which investigated the potential of emerging wireless communication technologies and potential synergies between mobile public safety and commercial networks to enhance the communication capabilities of public safety users. The paper identifies the current trends and challenges for wireless communications in the public safety domain, the potential solution frameworks identified in the HELP project and an economic analysis to show the benefits in terms of cost saving. The paper also describes the related research activities of the European Commission in this area.Peer ReviewedPostprint (published version

Designing and prototyping WebRTC and IMS integration using open source tools

WebRTC, or Web Real-time Communications, is a collection of web standards that detail the mechanisms, architectures and protocols that work together to deliver real-time multimedia services to the web browser. It represents a significant shift from the historical approach of using browser plugins, which over time, have proven cumbersome and problematic. Furthermore, it adopts various Internet standards in areas such as identity management, peer-to-peer connectivity, data exchange and media encoding, to provide a system that is truly open and interoperable. Given that WebRTC enables the delivery of multimedia content to any Internet Protocol (IP)-enabled device capable of hosting a web browser, this technology could potentially be used and deployed over millions of smartphones, tablets and personal computers worldwide. This service and device convergence remains an important goal of telecommunication network operators who seek to enable it through a converged network that is based on the IP Multimedia Subsystem (IMS). IMS is an IP-based subsystem that sits at the core of a modern telecommunication network and acts as the main routing substrate for media services and applications such as those that WebRTC realises. The combination of WebRTC and IMS represents an attractive coupling, and as such, a protracted investigation could help to answer important questions around the technical challenges that are involved in their integration, and the merits of various design alternatives that present themselves. This thesis is the result of such an investigation and culminates in the presentation of a detailed architectural model that is validated with a prototypical implementation in an open source testbed. The model is built on six requirements which emerge from an analysis of the literature, including previous interventions in IMS networks and a key technical report on design alternatives. Furthermore, this thesis argues that the client architecture requires support for web-oriented signalling, identity and call handling techniques leading to a potential for IMS networks to natively support these techniques as operator networks continue to grow and develop. The proposed model advocates the use of SIP over WebSockets for signalling and DTLS-SRTP for media to enable one-to-one communication and can be extended through additional functions resulting in a modular architecture. The model was implemented using open source tools which were assembled to create an experimental network testbed, and tests were conducted demonstrating successful cross domain communications under various conditions. The thesis has a strong focus on enabling ordinary software developers to assemble a prototypical network such as the one that was assembled and aims to enable experimentation in application use cases for integrated environments

A Floor Control Server in a Distributed Conference Service

The conferencing systems in IP Multimedia (IM) networks are going through restructuring, accomplished in the near future. One of the changes introduced is the concept of floors and floor control in its current form with matching entity roles. The Binary Floor Control Protocol (BFCP) is a novelty to be exploited in distributed tightly coupled conferencing services. The protocol defines the floor control server (FCS), which implements floor control giving access to shared resources. As the newest tendency is to distribute the conferencing services, the locations of different functionality units play an important role in developing the standards. The floor control server location is not yet single-mindedly fixed in different standardization bodies, and the debate goes on where to place it within the media server, providing the conferencing service. The thesis main objective is to evaluate two distinctive alternatives in respect the Mp interface protocol between the respective nodes, as the interface in relation to floor control is under standardization work at the moment. The thesis gives a straightforward preamble in IMS network, nodes of interest including floor control server and conferencing. Knowledge on several protocols – BFCP, SDP, SIP and H.248 provides an important background for understanding the functionality changes introduced in the Mp interface and therefore introductions on those protocols and how they are connected to the full picture is given. The actual analysis on the impact of the floor control server into the Mp reference point is concluded in relation to the locations, giving basic flows, requirements analysis including a limited implementation proposal on supporting protocol parameters. The overall conclusion of the thesis is that even if both choices are seemingly useful, not one of the locations is clearly the most suitable in the light of this work. The thesis suggests a solution having both possibilities available to be chosen from in separate circumstances, realized with consistent standardization. It is evident, that if the preliminary assumption for the analysis is kept regarding to only one right place for the floor control server, more work is to be done in connected areas to discover the one most appropriate location

Software defined wireless network (sdwn) for industrial environment: case of underground mine

Avec le développement continu des industries minières canadiennes, l’établissement des réseaux de communications souterrains avancés et sans fil est devenu un élément essentiel du processus industriel minier et ceci pour améliorer la productivité et assurer la communication entre les mineurs. Cette étude vise à proposer un système de communication minier en procurant une architecture SDWN (Software Defined Wireless Network) basée sur la technologie de communication LTE. Dans cette étude, les plateformes les plus importantes de réseau mobile 4G ont été étudiées, configurées et testées dans deux zones différentes : un tunnel de mine souterrain et un couloir intérieur étroit. Également, une architecture mobile combinant SDWN et NFV (Network Functions Virtualization) a été réalisée

Optimizing IETF multimedia signaling protocols and architectures in 3GPP networks : an evolutionary approach

Signaling in Next Generation IP-based networks heavily relies in the family of multimedia signaling protocols defined by IETF. Two of these signaling protocols are RTSP and SIP, which are text-based, client-server, request-response signaling protocols aimed at enabling multimedia sessions over IP networks. RTSP was conceived to set up streaming sessions from a Content / Streaming Server to a Streaming Client, while SIP was conceived to set up media (e.g.: voice, video, chat, file sharing, …) sessions among users. However, their scope has evolved and expanded over time to cover virtually any type of content and media session. As mobile networks progressively evolved towards an IP-only (All-IP) concept, particularly in 4G and 5G networks, 3GPP had to select IP-based signaling protocols for core mobile services, as opposed to traditional SS7-based protocols used in the circuit-switched domain in use in 2G and 3G networks. In that context, rather than reinventing the wheel, 3GPP decided to leverage Internet protocols and the work carried on by the IETF. Hence, it was not surprise that when 3GPP defined the so-called Packet-switched Streaming Service (PSS) for real-time continuous media delivery, it selected RTSP as its signaling protocol and, more importantly, SIP was eventually selected as the core signaling protocol for all multimedia core services in the mobile (All-)IP domain. This 3GPP decision to use off-the-shelf IETF-standardized signaling protocols has been a key cornerstone for the future of All-IP fixed / mobile networks convergence and Next Generation Networks (NGN) in general. In this context, the main goal of our work has been analyzing how such general purpose IP multimedia signaling protocols are deployed and behave over 3GPP mobile networks. Effectively, usage of IP protocols is key to enable cross-vendor interoperability. On the other hand, due to the specific nature of the mobile domain, there are scenarios where it might be possible to leverage some additional “context” to enhance the performance of such protocols in the particular case of mobile networks. With this idea in mind, the bulk of this thesis work has consisted on analyzing and optimizing the performance of SIP and RTSP multimedia signaling protocols and defining optimized deployment architectures, with particular focus on the 3GPP PSS and the 3GPP Mission Critical Push-to-Talk (MCPTT) service. This work was preceded by a detailed analysis work of the performance of underlying IP, UDP and TCP protocol performance over 3GPP networks, which provided the best baseline for the future work around IP multimedia signaling protocols. Our contributions include the proposal of new optimizations to enhance multimedia streaming session setup procedures, detailed analysis and optimizations of a SIP-based Presence service and, finally, the definition of new use cases and optimized deployment architectures for the 3GPP MCPTT service. All this work has been published in the form of one book, three papers published in JCR cited International Journals, 5 articles published in International Conferences, one paper published in a National Conference and one awarded patent. This thesis work provides a detailed description of all contributions plus a comprehensive overview of their context, the guiding principles beneath all contributions, their applicability to different network deployment technologies (from 2.5G to 5G), a detailed overview of the related OMA and 3GPP architectures, services and design principles. Last but not least, the potential evolution of this research work into the 5G domain is also outlined as well.Els mecanismes de Senyalització en xarxes de nova generació es fonamenten en protocols de senyalització definits per IETF. En particular, SIP i RTSP són dos protocols extensibles basats en missatges de text i paradigma petició-resposta. RTSP va ser concebut per a establir sessions de streaming de continguts, mentre SIP va ser creat inicialment per a facilitar l’establiment de sessions multimèdia (veu, vídeo, xat, compartició) entre usuaris. Tot i així, el seu àmbit d’aplicació s’ha anat expandint i evolucionant fins a cobrir virtualment qualsevol tipus de contingut i sessió multimèdia. A mesura que les xarxes mòbils han anat evolucionant cap a un paradigma “All-IP”, particularment en xarxes 4G i 5G, 3GPP va seleccionar els protocols i arquitectures destinats a gestionar la senyalització dels serveis mòbils presents i futurs. En un moment determinat 3GPP decideix que, a diferència dels sistemes 2G i 3G que fan servir protocols basats en SS7, els sistemes de nova generació farien servir protocols estandarditzats per IETF. Quan 3GPP va començar a estandarditzar el servei de Streaming sobre xarxes mòbils PSS (Packet-switched Streaming Service) va escollir el protocol RTSP com a mecanisme de senyalització. Encara més significatiu, el protocol SIP va ser escollit com a mecanisme de senyalització per a IMS (IP Multimedia Subsystem), l’arquitectura de nova generació que substituirà la xarxa telefònica tradicional i permetrà el desplegament de nous serveis multimèdia. La decisió per part de 3GPP de seleccionar protocols estàndards definits per IETF ha representat una fita cabdal per a la convergència del sistemes All-IP fixes i mòbils, i per al desenvolupament de xarxes NGN (Next Generation Networks) en general. En aquest context, el nostre objectiu inicial ha estat analitzar com aquests protocols de senyalització multimèdia, dissenyats per a xarxes IP genèriques, es comporten sobre xarxes mòbils 3GPP. Efectivament, l’ús de protocols IP és fonamental de cara a facilitar la interoperabilitat de solucions diferents. Per altra banda, hi ha escenaris a on és possible aprofitar informació de “context” addicional per a millorar el comportament d’aquests protocols en al cas particular de xarxes mòbils. El cos principal del treball de la tesi ha consistit en l’anàlisi i optimització del rendiment dels protocols de senyalització multimèdia SIP i RTSP, i la definició d’arquitectures de desplegament, amb èmfasi en els serveis 3GPP PSS i 3GPP Mission Critical Push-to-Talk (MCPTT). Aquest treball ha estat precedit per una feina d’anàlisi detallada del comportament dels protocols IP, TCP i UDP sobre xarxes 3GPP, que va proporcionar els fonaments adequats per a la posterior tasca d’anàlisi de protocols de senyalització sobre xarxes mòbils. Les contribucions inclouen la proposta de noves optimitzacions per a millorar els procediments d’establiment de sessions de streaming multimèdia, l’anàlisi detallat i optimització del servei de Presència basat en SIP i la definició de nous casos d’ús i exemples de desplegament d’arquitectures optimitzades per al servei 3GPP MCPTT. Aquestes contribucions ha quedat reflectides en un llibre, tres articles publicats en Revistes Internacionals amb índex JCR, 5 articles publicats en Conferències Internacionals, un article publicat en Congrés Nacional i l’adjudicació d’una patent. La tesi proporciona una descripció detallada de totes les contribucions, així com un exhaustiu repàs del seu context, dels principis fonamentals subjacents a totes les contribucions, la seva aplicabilitat a diferents tipus de desplegaments de xarxa (des de 2.5G a 5G), així una presentació detallada de les arquitectures associades definides per organismes com OMA o 3GPP. Finalment també es presenta l’evolució potencial de la tasca de recerca cap a sistemes 5G

Architecture and Remote Interaction Techniques for Digital Media Exchange across 3G Mobile Devices

For users away from the office or home, there is an increasing demand for mobile solutions that offer effective collaborative facilities on the move. The mobile cellular device, or “smart phone”, can offer a ubiquitous platform to deliver such services, provided that its many physical and technological constraints can be overcome. In an effort to better support mobile collaboration, this thesis presents a contributing Mobile Exchange Architecture (MEA) designed to improve upon the capabilities provided by mobile devices to enable synchronous exchange of digital media during a phone conversation using wireless networks and cellular devices. This research includes the design and development of one such MEA in the form of a fully functional Photo-conferencing service, supporting shared remote interaction techniques, simultaneous voice communication and seamless digital media exchange between remote and collocated mobile users. Furthermore, through systematic design, experimental evaluations and field studies we evaluate the effects of different shared remote interaction techniques – 'pointing', 'scaling', 'mixed' and 'hybrid' – assessing the task effort required by users when interacting around shared images across resource constrained mobile devices. This thesis presents a direction for the future development of technologies and methods to enable a new era of scalable always-to-hand mobile collaborative environments.EThOS - Electronic Theses Online ServiceGBUnited Kingdo