Serviços multimédia multicast de próxima geração

Mestrado em Engenharia Electrónica e TelecomunicaçõesUma das mais recentes conquistas na evolução móvel foi o 3G, permitindo o acesso a serviços multimédia com qualidade de serviço assegurada. No entanto, a tecnologia UMTS, tal como definida na sua Release ’99, é apenas capaz de transmitir em modo unicast, sendo manifestamente ineficiente para comunicações multimédia almejando grupos de utilizadores. A tecnologia IMS surge na Release 5 do 3GPP que começou a responder já a algumas necessidades, permitindo comunicações sobre IP oferecendo serviços Internet a qualquer momento e em qualquer lugar sobre tecnologias de comunicação móveis fornecendo pela primeira vez sessões multimédia satisfatórias. A Release 6 por sua vez trouxe a tecnologia MBMS que permite transmissões em broadcast e multicast para redes móveis. O MBMS fornece os serviços de aplicações multimédia que todos estavam à espera, tanto para os utilizadores como para os prestadores de serviços. O operador pode agora fazer uso da tecnologia existente aumentando todo o tipo de benefícios no serviço prestado ao cliente. Com a possível integração destas duas tecnologias passa a ser possível desenvolver serviços assentes em redes convergentes em que os conteúdos são entregues usando tecnologias unicast, multicast ou broadcast. Neste contexto, o principal motivo deste trabalho consiste essencialmente em fazer uso dos recursos da rede terminando com o desperdício dos mesmos e aumentando a eficiência dos serviços através da integração das tecnologias IMS e MBMS. O trabalho realizado começa com o estudo do estado da arte das telecomunicações móveis com referência às tecnologias referidas, seguindo-se a apresentação da possível integração IMS-MBMS e terminando com o projecto de uma plataforma de demonstração que no futuro possa ser uma implementação de serviço multimédia multicast. O objectivo principal é mostrar os benefícios de um serviço que era normalmente executado em unicast relativamente ao modo multicast, fazendo uso da nova convergência de tecnologias IMS e MBMS. Na conclusão do trabalho são referidas as vantagens do uso de portadoras multicast e broadcast, tendo como perspectiva de que este trabalho possa ser um ponto de partida para um novo conjunto de serviços poupando recursos de rede e permitindo uma eficiência considerável em serviços inovadores.3G is bang up to date in the mobile phone industry. It allows access to multimedia services and gives a guarantee of quality of service. The UMTS technology, defined in 3GPP Release ’99, provides an unicast transmission, but it is completely inefficient when it comes to multimedia group communications. The IMS technology first appeared in Release 5 that has already started to consider the interests of the clients. It provides communications over IP, offering Internet services anytime, anywhere on mobile communication technologies. Also, it offers for the first time satisfactory multimedia sessions. On the other hand, Release 6 gave rise to the MBMS technology that provides broadcast and multicast transmissions for mobile networks. The MBMS provides multimedia applications services that everyone was waiting, including users and service providers. Now the operator makes use of existing technology in order to provide better costumer services. The possible integration of these two technologies will contribute to develop services based on converged networks in which contents are delivered through the unicast, multicast or broadcast technologies. Therefore, the objective of this work is basically to make use of network resources avoiding wastes and improving customer services through the integration of the IMS and the MBMS technologies. The executed work starts with the mobile telecommunications state of the art with reference to the referred technologies, followed by the IMS-MBMS convergence presentation and finishing with the proposal for implementation of a service platform that can be used for a multimedia multicast service. The main point is to show the benefits of a service that has been normally executed in unicast mode over the multicast mode, making use of the new IMS and MBMS technologies integration. To closure the work it is referred the advantages to use multicast and broadcast bearers, with the perspective that this work could be a starting point to a new set of services, saving network resources and allowing for innovate services a considerable efficency

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

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Optimização de recursos para difusão em redes de próxima geração

Doutoramento em ElectrotecniaEsta tese aborda o problema de optimização de recursos de rede, na entrega de Serviços de Comunicação em Grupo, em Redes de Próxima Geração que suportem tecnologias de difusão. De acordo com esta problemática, são feitas propostas que levam em atenção a evolução espectável das redes 3G em Redes Heterogéneas de Próxima Geração que incluam tecnologias de difusão tais como o DVB. A optimização de recursos em Comunicações em Grupo é apresentada como um desafio vertical que deve cruzar diversas camadas. As optimizações aqui propostas cobrem tanto a interface entre Aplicação e a Plataforma de Serviços para a disponibilização de serviços de comunicação em grupo, como as abstracções e mapeamentos feitos na interface entre a Rede Central e a Rede de Acesso Rádio. As optimizações propostas nesta tese, assumem que o caminho evolutivo na direcção de uma Rede de Próxima Geração é feito através do IP. Em primeiro lugar são endereçadas as optimizações entre a Aplicação e a Plataforma de Serviços que já podem ser integradas nas redes 3G existentes. Estas optimizações podem potenciar o desenvolvimento de novas e inovadoras aplicações, que através do uso de mecanismos de distribuição em difusão podem fazer um uso mais eficiente dos recursos de rede. De seguida são apresentadas optimizações ao nível da interface entre a Rede Central e a Rede de Acesso Rádio que abordam a heterogeneidade das redes futuras assim como a necessidade de suportar tecnologias de difusão. É ainda considerada a possibilidade de aumentar a qualidade de serviço de serviços de difusão através do mapeamento do IP multicast em portadoras unidireccionais. Por forma a validar todas estas optimizações, vários protótipos foram desenvolvidos com base num router avançado para redes de acesso de próxima geração. As funcionalidades e arquitectura de software desse router são também aqui apresentadas.This thesis addresses the problem of optimizing network resource usage, for the delivery of Group Services, in Next Generation Networks featuring broadcast technologies. In this scope, proposals are made according to the expected evolution of 3G networks into Next Generation Heterogeneous Networks that include broadcast technologies such as DVB. Group Communication resource optimization is considered a vertical challenge that must cross several layers. The optimizations here proposed cover both Application to Service Platform interfaces for group communication services, and Core Network to Radio Access Network interface abstractions and mappings. The proposed optimizations are also presented taking into consideration network evolution path towards an All-IP based Next Generation Network. First it is addressed the Application to Service Platform optimization, which can already be deployed over 3G networks. This optimization could potentiate the development of new and innovative applications that through the use of broadcast/multicast service delivery mechanisms could be more efficient network wise. Next proposals are made on the Core Network to Radio Access Network interfaces that address the heterogeneity of future networks and consider the need to support broadcast networks. It is also considered the possibility to increase the Quality of Service of broadcast/multicast services based on the dynamic mapping of IP multicast into unicast radio bearers. In order to validate these optimizations, several prototypes were built based on an advanced access router for next generation networks. Such access router functionalities and software architecture are also presented here

Registry composition in ambient networks

Ambient Networks (AN) is a new networking concept for beyond 3G. It is a product of the European Union's Sixth Framework Program (FP6). Network composition is a core concept of ANs. It allows dynamic, scalable and uniform cooperation between heterogeneous networks. ANs can host various registries. These registries may be of different types (e.g. centralized, distributed), store heterogeneous types of information (e.g. raw data vs. aggregated data), and rely on different interfaces to access the stored information (i.e. protocols or programming interfaces). When ANs compose, the hosted registries need to compose. Registry composition is a sub-process of network composition. It provides seamless and autonomous access to the content of all of the registries in the composed network. This thesis proposes a new architecture for registry composition in ANs. This overall architecture is made up of four components: interface interworking, data interworking, negotiation and signaling. Interface interworking enables dynamic intercommunication between registries with heterogeneous interfaces. Data interworking involves dynamically overcoming data heterogeneity (e.g. format and granularity). Interface and data interworking go beyond static interworking using gateways, as done today. The negotiation component allows the negotiation of the composition agreement. Signaling coordinates and regulates the negotiation and the execution of the composition agreement. Requirements are derived and related work is reviewed. We propose a new functional entity and a new procedure to orchestrate the composition process. We also propose a new architecture for interface interworking, based on a peer to peer overlay network. We have built a proof-of-concept prototype. The interface-interworking component is used as the basis of our new architecture to data interworking. This architecture reuses mechanisms and algorithms from the federated data base area. The thesis proposes as well a new architecture for on-line negotiation. The architecture includes a template for composition agreement proposals, and a negotiation protocol that was validated using SPIN. A new signaling framework is also proposed. It is based on the IETF Next Step in Signaling (NSIS) framework and was validated using OPNET. Most of these contributions are now part of the AN concept, as defined by the European Union's Sixth Framework Progra

The strategies associated with the migration of networks to 4G

The networks need to provide higher speeds than those offered today. For it, considering that in the spectrum radio technologies is the scarcest resource in the development of these technologies and the new developments is essential to maximize the performance of bits per hertz transmitted. Long Term Evolution optimize spectral efficiency modulations with new air interface, and more advanced algorithms radius. These capabilities is the fact that LTE is an IPbased technology that enables end-to-end offer high transmission rates per user and very low latency, ie delay in the response times of the network around only 10 milliseconds, so you can offer any realtime application. LTE is the latest standard in mobile network technology and 3GPP ensure competitiveness in the future, may be considered a technology bridge between 3G networks - current 3.5G and future 4G networks, which are expected to reach speeds of up to 1G . LTE operators provide a simplified architecture but both robust, supporting services on IP technology. The objectives to be achieved through its implementation are ambitious, first users have a wide range of added services like capabilities that currently enjoys with residential broadband access at competitive prices, while the operator will have a network fully IP-based environment, reducing the complexity and cost of the same, which will give operators the opportunity to migrate to LTE directly. A major advantage of LTE is its ability to fuse with existing networks, ensuring interconnection with the same, increasing his current coverage and allowing a data connection established by a user in the environment continue when fade the coverage LTE. Moreover, the operator has the advantage of deploying network gradually, starting initially at areas of high demand for broadband services and expand progressively in line with this. RESUMEN. Las redes necesitan proporcionar velocidades mayores a las ofertadas a día de hoy. Para ello, teniendo en cuenta que en tecnologías radio el espectro es el recurso más escaso, en la evolución de estas tecnologías y en los nuevos desarrollos es esencial maximizar el rendimiento de bits por hercio transmitido. Long Term Evolution optimiza la eficiencia espectral con nuevas modulaciones en la interfaz aire, así como los algoritmos radio más avanzado. A estas capacidades se suma el hecho de que LTE es una tecnología basada en IP de extremo a extremo que permite ofrecer altas velocidades de transmisión por usuario y latencias muy bajas, es decir, retardos en los tiempos de respuesta de la red en torno a sólo 10 milisegundos, por lo que permite ofrecer cualquier tipo de aplicación en tiempo real. LTE es el último estándar en tecnología de redes móviles y asegurará la competitividad de 3GPP en el futuro, pudiendo ser considerada una tecnología puente entre las redes 3G – 3.5G actuales y las futuras redes 4G, de las que se esperan alcanzar velocidades de hasta 1G. LTE proporcionará a las operadoras una arquitectura simplificada pero robusta a la vez, soportando servicios sobre tecnología IP. Los objetivos que se persiguen con su implantación son ambiciosos, por una parte los usuarios dispondrá de una amplia oferta de servicios añadidos con capacidades similares a las que disfruta actualmente con accesos a banda ancha residencial y a precios competitivos, mientras que el operador dispondrá de una red basada en entorno totalmente IP, reduciendo la complejidad y el costo de la misma, lo que dará a las operadoras la oportunidad de migrar a LTE directamente. Una gran ventaja de LTE es su capacidad para fusionarse con las redes existentes, asegurando la interconexión con las mismas, aumentando su actual cobertura y permitiendo que una conexión de datos establecida por un usuario en el entorno LTE continúe cuando la cobertura LTE se desvanezca. Por otra parte el operador tiene la ventaja de desplegar la red LTE de forma gradual, comenzando inicialmente por las áreas de gran demanda de servicios de banda ancha y ampliarla progresivamente en función de ésta

Feasibility study of VoIP in 3GPP UMTS release 5 interworking with fixed networks

Masteroppgave i informasjons- og kommunikasjonsteknologi 2003 - Høgskolen i Agder, GrimstadThe Universal Mobile Telecommunications System (UMTS) is denoted as a 3rd generation cellular system and has been designed with the objective to be a system with global coverage. With improvement of bandwidth capabilities, the UMTS system has the ability to support real time multimedia services. The focus in this thesis is Voice over IP (VoIP) which enables a user to make phone calls in the packet switched network in UMTS. This thesis starts with a presentation of VoIP with the quality requirements related to a voice session. A voice conversation needs a guaranteed quality to satisfy the participants. This thesis focuses on three main aspects; Quality of Service mechanisms (Best Effort, IntServ and DiffServ), VoIP in UMTS with a certain quality and last but not least implementation of Quality of Service (QoS) in a voice call interworking with external networks. Best Effort cannot be used when dealing with real time traffic such as VoIP. IntServ reserves resources from the application itself, and gives opportunity for each application in the terminal to request a certain quality. DiffServ works on a higher level and classifies traffic based on type of traffic, not for a particular request. For UMTS interworking with IP networks, the theoretical results suggest that IntServ over DiffServ should be used in the UMTS gateway node. An evaluation of the UMTS network is done by checking the voice quality attained by the network during a VoIP session in comparison of a traditional circuit switched call setup. Moreover, tests from the Norwegian UMTS network operator NetCom became useful when evaluating how well the VoIP could work when implementing UMTS release 5. The tests were set up with the focus on delay and voice quality in the network, and were meant for disclosing the differences with and without quality parameters during a transmission. Due to network restrictions the test results are limited

Inter-domain interoperability framework based on WebRTC

Nowadays, the communications paradigm is changing with the convergence of communication services to a model based on IP networks. Applications such as messaging or voice over IP are increasing its popularity and Communication Service Providers are focusing on offering this kind of services. Moreover, Web Real Time Communication (WebRTC) has emerged as a technology that eases the creation of web applications featuring Real-Time Communications over IP networks without the need to develop and install any plug-in. It lacks of specifications in the control plane, leaving the possibility to use WebRTC over tailored web signalling solutions or legacy networks such as IP Multimedia Subsystem (IMS). This technology brings a wide range of possibilities for web developers, but Communication Service Providers are adviced to develop solutions based on the WebRTC technology as described in the Eurescom Study P2252. The lack of WebRTC specifications on the signalling platform together with the threats and opportunities that this technology represents for Communication Service Providers, makes evident the need of research on interoperability solutions for the different kind of signalling implementations and experimentation on the best way for Communication Service Providers to obtain the maximum benefit from WebRTC technology. The main goal of this thesis is precisely to develop a WebRTC interoperability framework and perform experiments on whether the Communication Service Providers should use their existing IMS solutions or develop tailored web signalling platforms for WebRTC deployments. In particular, the work developed in this thesis was completed under the framework of the Webrtc interOperability tested in coNtradictive DEployment scenaRios (WONDER) experimentation for the OpenLab project. OpenLab is a Large-scale integrating project (IP) and is part of the European Union Framework Programme 7 for Research and Development (FP7) addressing the work programme topic Future Internet Research and Experimentation.Actualmente, el paradigma de comunicaciones está cambiando gracias a la convergencia de los servicios de comunicaciones hacia un modelo basado en redes IP. Aplicaciones tales como la mensajería y la voz sobre IP están creciendo en popularidad mientras los proveedores de servicios de comunicaciones se centran en ofrecer este tipo de servicios basados en redes IP. Por otra parte, la tecnología WebRTC ha surgido para facilitar la creación de aplicaciones web que incluyan comunicaciones en tiempo real sobre redes IP sin la necesidad de desarrollar o instalar ningún complemento. Esta tecnología no especifica los protocolos o sistemas a utilizar en el plano de control, dejando a los desarrolladores la posibilidad de usar WebRTC sobre soluciones de señalizaci on web específicas o utilizar las redes de señalización existentes, tales como IMS. WebRTC abre un gran abanico de posibilidades a los desarrolladores web, aunque también se recomienda a los proveedores de servicios de comunicaciones que desarrollen soluciones basadas en WebRTC como se describe en el estudio P2252 de Eurescom. La falta de especificaciones en el plano de señalización junto a las oportunidades y amenazas que WebRTC representa para los proveedores de servicios de comunicaciones, hacen evidente la necesidad de investigar soluciones de interoperabilidad para las distintas implementaciones de las plataformas de señalización y de experimentar c omo los proveedores de servicios de comunicaciones pueden obtener el máximo provecho de la tecnología WebRTC. El objetivo principal de este Proyecto Fin de Carrera es desarrollar un marco de interoperabilidad para WebRTC y realizar experimentos que permitan determinar bajo que condiciones los proveedores de servicios de comunicaciones deben utilizar las plataformas de se~nalizaci on existentes (en este caso IMS) o desarrollar plataformas de señalización a medida basadas en tecnologías web para sus despliegues de WebRTC. En particular, el trabajo realizado en este Proyecto Fin de Carrera se llevó a cabo bajo el marco del proyecto WONDER para el programa OpenLab. OpenLab es un proyecto de integración a gran escala en el cual se desarrollan investigaciones y experimentos en el ámbito del futuro Internet y que forma parte del programa FP7 de la Unión Europea.Ingeniería de Telecomunicació

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