Distributed rate allocation in switch-based multiparty videoconference

Multiparty videoconferences, or more generally multiparty video calls, are gaining a lot of popularity as they offer a rich communication experience. These applications have however, large requirements in terms of both network and computational resources and have to deal with sets of heterogenous clients. The multiparty videoconferencing systems can be grouped in two classes. They are based either on expensive central nodes, called multipoint control units (MCU), with transcoding capabilities, or, on a peer-to-peer strategy where users help each other to distribute the different video streams. Whereas the first one requires an expensive central hardware, the second one depends completely on the redistribution capacity of the users, which sometimes might neither provide sufficient bandwidth nor be reliable enough. In this work we propose an alternative solution where we use a central node to distribute the video streams but at the same time we maintain the hardware complexity and the computational requirements of this node as low as possible. The proposed solution uses a distributed algorithm to allocate the users' rates in a Quality of Service (QoS) aware manner. The allocation algorithm is also extremely fast and is able to quickly reallocate the rates in case the conditions change. We have further implemented our solution in a network simulator where we show that our rate allocation algorithm is able to properly optimize users' QoS and adapt to dynamic changes in the system. We also illustrate the benefits of our solution in terms network usage and average utility when compared to a baseline heuristic method operating on the same system architecture

Distributed Rate Allocation in Switch-Based Multiparty Videoconferencing System

Multiparty videoconferences, or more generally multiparty video calls, are gaining a lot of popularity as they offer a rich communication experience. These applications have, however, large requirements in terms of both network and computational resources and have to deal with sets of heterogenous clients. The multiparty videoconferencing systems are usually either based on expensive central nodes, called Multipoint Control Units (MCU), with transcoding capabilities, or on a peer-to-peer architecture where users cooperate to distribute more efficiently the different video streams. Whereas the first class of systems requires an expensive central hardware, the second one depends completely on the redistribution capacity of the users, which sometimes might neither provide sufficient bandwidth nor be reliable enough. In this work we propose an alternative solution where we use a central node to distribute the video streams, but at the same time we maintain the hardware complexity and the computational requirements of this node as low as possible, e.g. it has no video decoding capabilities. We formulate the rate allocation problem as an optimization problem that aims at maximizing the Quality of Service (QoS) of the videoconference. We propose two different distributed algorithms for solving the optimization problem: the first algorithm is able to find an approximate solution of the problem in a one-shot execution, whereas the second algorithm, based on Lagrangian relaxation, performs iterative updates of the optimization variables in order to gradually increase the value of the objective function. The two algorithms, though being disjointed, nicely complement each other. If executed in sequence, they allow to achieve both, a quick approximate rate reallocation in case of a sudden change of the system conditions, and a precise refinement of the variables which avoids problems caused by possible faulty approximate solutions. We have further implemented our solution in a network simulator where we show that our rate allocation algorithm is able to properly optimize users' QoS. We also illustrate the benefits of our solution in terms of network usage and overall utility when compared to a baseline heuristic method operating on the same system architecture

Network utility maximization for delay-sensitive applications in unknown communication settings

In the last decades the Internet traffic has greatly evolved. The advent of new Internet services and applications has, in fact, led to a significant growth of the amount of data transmitted, as well as to a transformation of the data type. As a matter of fact, nowadays, the largest amount of traffic share consists of multimedia data, which do not represent classical Internet data. Due to the increasing amount of traffic, the network resources might be scarce, and in such cases it becomes extremely important to optimize network transmission in order to provide a satisfying service to the users. Although methods for maximizing the network utility in scenarios with limited resources have been studied extensively, the evolution of the Internet services poses continuously new challenges that require novel solution methods to meet the transmission requirements. In this thesis we propose novel solutions methods to network utility maximization problems that arise in the context of nowadays network communications. In particular we analyze problems related to delay-sensitive Internet applications and rate allocation in unknown network settings. In the first problem we study how to effectively allocate the transmission rates in a multiparty videoconference system. The main contribution of this chapter is an approximate fast rate rate allocation method that is able to adapt quickly to changes in the videoconference conditions. This fast adaptation cannot be achieved with classical network utility maximization solving methods, as they are usually based on iterative approaches. In this case we leverage the particular structure of the problem to design a novel distributed solving method which proves to be very effective when compared to baseline solutions. The next problem that we address is the design of a congestion control algorithm for delay-sensitive applications. One of the main problems of existing delay-based congestion control algorithms is that they tend to achieve an extremely low throughput when competing against loss-based algorithms. In order to overcome this difficulty we propose a novel adaptive controller based on a bandit problem approach. The adaptive controller tries to infer how the network responds, in terms of rate-delay pair at equilibrium, when changing the delay sensitivity of an underlying delay-based congestion control. Once the network response is inferred, the controller selects the sensitivity that leads to the best trade-off between the transmitting rate and the experienced delay. In the final problem, we analyze the design of an overlay rate allocation systems to be used when: the amount of available network resources is not known, and the user congestion feedback cannot be used as valid signal to reach the optimal rate allocation. Such a scenario appears when an Internet application wants to maximize a certain utility metric, but, at the same time, it must operate using a specific congestion control algorithm that is completely unaware of the application utility. To solve this problem we design a distributed system that coordinates the users in order to perform active learning on the amount of network resource. Adopting such a method reveals to be the key to an effective maximization of the long term application utility for the entire system

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

Applications of satellite technology to broadband ISDN networks

Two satellite architectures for delivering broadband integrated services digital network (B-ISDN) service are evaluated. The first is assumed integral to an existing terrestrial network, and provides complementary services such as interconnects to remote nodes as well as high-rate multicast and broadcast service. The interconnects are at a 155 Mbs rate and are shown as being met with a nonregenerative multibeam satellite having 10-1.5 degree spots. The second satellite architecture focuses on providing private B-ISDN networks as well as acting as a gateway to the public network. This is conceived as being provided by a regenerative multibeam satellite with on-board ATM (asynchronous transfer mode) processing payload. With up to 800 Mbs offered, higher satellite EIRP is required. This is accomplished with 12-0.4 degree hopping beams, covering a total of 110 dwell positions. It is estimated the space segment capital cost for architecture one would be about $190M whereas the second architecture would be about$250M. The net user cost is given for a variety of scenarios, but the cost for 155 Mbs services is shown to be about $15-22/minute for 25 percent system utilization

A Cloud Infrastructure for Multimedia Conferencing Applications

Conferencing enables the conversational exchange of media between several parties. Conferencing applications are among important enterprise applications nowadays. However, fine grained scalability and elasticity remain quite elusive for multimedia conferencing applications, although they are key to efficiency in the resource usage. Cloud computing is an emerging paradigm for provisioning network, storage, and computing resources on demand using a pay-per-use model. Cloud-based conferencing services can inherent several benefits such as resource usage efficiency, scalability and easy introduction of different types of conferences. This thesis relies on a recently proposed business model for cloud-based conferencing. The model has the following roles: conferencing substrate provider, conferencing infrastructure provider, conferencing platform provider, conferencing service provider, and broker. Conferencing substrates are generally atomic and served as elementary building blocks (e.g. signalling, mixing) of conferencing applications. They can be virtualized and shared among several conferencing applications for resource efficiency purposes. Multiple conferencing substrates provided by different conferencing substrate providers can be combined to build a conferencing service (e.g. a dial-out signalling substrate and an audio mixer substrate can be composed to build a dial-out audio conference service). This thesis focuses on the conferencing infrastructure provider and conferencing substrate provider roles. It proposes a virtualized cloud infrastructure for multimedia conferencing applications. This infrastructure relies on fine grained conferencing substrates (e.g. dial-out signalling, dial-in signalling, audio mixer, video mixer, floor control, etc.) and offers several advantages in addition to fine grained scalability and elasticity (e.g. assembling substrates on the fly to build new conferencing applications). An architecture is proposed to realize the roles of conferencing infrastructure provider, conferencing substrate provider and their interactions. A resource allocation mechanism for conferencing substrates is also proposed. We have also built a prototype with Xen as the virtualization platform and validated the architecture. Performance has also been evaluated

Seeing Eye to I? The Influence of Self-Video Display Size on Visual Attention and Collaborative Performance in Peer-to-Peer Video Chat

This thesis examines the influence of self-video size in video chat conversations on visual attention, collaborative performance, grounding, comfort and distraction during a brainstorming task. Twenty pairs of female university students were randomly assigned to either a large or small self-video condition. Two eye tracking systems were used to simultaneously record pairs of participants' gaze across 4 areas-of-interest spanning a 15-minute task. Participants with larger self-video gazed at themselves longer but did not spend a significantly different percentage of the conversation gazing at their partner. Participants sufficiently estimated how long they looked at each other, but significantly overestimated how long they, and their partners, gazed at their own self-video. A majority of participants found their self-video to be comforting, and participants with larger displays found it to be more distracting than those with smaller displays. Over a third of participants would prefer to chat without their self-video visible