Video Quality Prediction for Video over Wireless Access Networks (UMTS and WLAN)

Transmission of video content over wireless access networks (in particular, Wireless Local Area Networks (WLAN) and Third Generation Universal Mobile Telecommunication System (3G UMTS)) is growing exponentially and gaining popularity, and is predicted to expose new revenue streams for mobile network operators. However, the success of these video applications over wireless access networks very much depend on meeting the user’s Quality of Service (QoS) requirements. Thus, it is highly desirable to be able to predict and, if appropriate, to control video quality to meet user’s QoS requirements. Video quality is affected by distortions caused by the encoder and the wireless access network. The impact of these distortions is content dependent, but this feature has not been widely used in existing video quality prediction models. The main aim of the project is the development of novel and efficient models for video quality prediction in a non-intrusive way for low bitrate and resolution videos and to demonstrate their application in QoS-driven adaptation schemes for mobile video streaming applications. This led to five main contributions of the thesis as follows:(1) A thorough understanding of the relationships between video quality, wireless access network (UMTS and WLAN) parameters (e.g. packet/block loss, mean burst length and link bandwidth), encoder parameters (e.g. sender bitrate, frame rate) and content type is provided. An understanding of the relationships and interactions between them and their impact on video quality is important as it provides a basis for the development of non-intrusive video quality prediction models.(2) A new content classification method was proposed based on statistical tools as content type was found to be the most important parameter. (3) Efficient regression-based and artificial neural network-based learning models were developed for video quality prediction over WLAN and UMTS access networks. The models are light weight (can be implemented in real time monitoring), provide a measure for user perceived quality, without time consuming subjective tests. The models have potential applications in several other areas, including QoS control and optimization in network planning and content provisioning for network/service providers.(4) The applications of the proposed regression-based models were investigated in (i) optimization of content provisioning and network resource utilization and (ii) A new fuzzy sender bitrate adaptation scheme was presented at the sender side over WLAN and UMTS access networks. (5) Finally, Internet-based subjective tests that captured distortions caused by the encoder and the wireless access network for different types of contents were designed. The database of subjective results has been made available to research community as there is a lack of subjective video quality assessment databases.Partially sponsored by EU FP7 ADAMANTIUM Project (EU Contract 214751

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

Virtualisation and Thin Client : A Survey of Virtual Desktop environments

This survey examines some of the leading commercial Virtualisation and Thin Client technologies. Reference is made to a number of academic research sources and to prominent industry specialists and commentators. A basic virtualisation Laboratory model is assembled to demonstrate fundamental Thin Client operations and to clarify potential problem areas

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

A More Realistic RTP/RTCP-Based Simulation Platform for Video Streaming QoS Evaluation

[EN] Over the last few years, the demand for real-time multimedia services has been growing progressively so that video streaming applications are expected to be dominant in future communications systems, and most of them using RTP/RTCP protocols. This paper presents an evolved tool-set for video streaming QoS evaluation in simulated environments using such protocols. We have designed a new NS-2 module with a full RTP/RTCP implementation (following strictly the RFC 3550) and we propose to combine it with additional multimedia tools to obtain an advanced simulation framework that allows the measurement of network-level QoS metrics (such as throughput, delay, jitter or loss rate) in simulation time. Besides, as the transmitted video files can be reconstructed and played out at the receiver side, the measurement of the quality of the delivered video streams, by employing the most common objective quality metrics (such as PSNR, SSIM or VQM) or subjective metrics (MOS), is also supported. By using this tool-set, researchers and practitioners can assess their novel designs (such as network protocols, routing strategies or video coding mechanisms) for such applications in heterogeneous scenarios over different network conditions. As RTCP feedback capabilities have been added, source based control techniques (such as rate adaptability or Multiple Description Coding) could be included and tested using this more realistic and powerful simulation platform.This work has been financed, partially, by Polytechnics University of Valencia (UPV), under its R&D Support Program in PAID-06-08-002-585 Project and in PAID-01-10, and by Generalitat Valenciana, under its R&D Support Program in GV 2010/009 Project.Boronat Segui, F.; Montagud Aguar, M.; Vidal Gimeno, VE. (2011). A More Realistic RTP/RTCP-Based Simulation Platform for Video Streaming QoS Evaluation. Journal of Mobile Multimedia. 7(1):66-88. http://hdl.handle.net/10251/45964S66887

Handoff management for infotainment services over vehicular networks

Intelligent Transportation Systems (ITS) has impulsed the vehicular communications at the present time. The vehicular communications field is a hot research topic and is attracting a great interest in the automotive industry and telecommunications. There are essentially two main lines of work: (1) communication services related to road safety and traffic information; and (2) information and entertainment services, also named infotainment services. These latter services include both transmitting multimedia (voice over IP, streaming, on-line gaming, etc.) and classic data services (e-mail, access to private networks, web browsing, file sharing, etc.). In this thesis we will focus on these infotainment services because further research in this immature research field is necessary and, until nowadays, the main effort of the research community regarding vehicular communication has been focused on road safety and traffic information. Vehicular nodes need to be reached from the Internet and vice versa to be able to access to infotainment services. While vehicles move along the road infrastructure, they change their wireless point of attachment to the network. During this process, connectivity breaks down until the vehicle is connected again to a new road side unit in its area. This disconnection causes a disruption in the communications. Fast handoffs are a crucial requirement for vehicular networks to avoid long disruption times, since the high speed of vehicular nodes involves suffering a lot of handoffs during an Internet connection. This thesis is focused on Vehicular-to-Infrastructure (V2I) real-time infotainment services. The main contributions of this thesis are: i) a new testing framework for V2I communications to be able to test infotainment services in an easy way; ii) the analysis of the deployability of infotainment video services in vehicular networks using mobility protocols; and iii) the development of a new TCP architecture that will provide a better performance for all TCP-based infotainment services in a vehicular scenario with handoffs. In this thesis, firstly, we propose a new testing framework for vehicular infotainment applications. This framework is a vehicular emulation platform that allows testing real applications installed on Linux virtual machines. Using emulation, we are able to evaluate the performance of real applications with real-time requirements, so we can test multimedia applications used to offer infotainment services in vehicular scenarios in a straightforward way. Secondly, using the testing framework implemented in the first part of the thesis, we have done a performance evaluation of an infotainment service. Among these services, we think that video on demand services on highways will be interesting for users, and generate revenue to network operators. So we evaluated how network-layer handoffs can limit the deployment of a video streaming service. According to the results obtained, driving at high speeds will be an issue for a correct playback of video content, even using fast handoffs techniques. Finally, we developed a new TCP architecture to enhance performance during handoffs. Most of the non-safety services on ITS rely on the Transport Control Protocol (TCP), one of the core protocols of the Internet Protocol Suite. However there exists several issues related to TCP and mobility that can affect to TCP performance, and these issues are particularly important in vehicular networks due to its high mobility. Using new IEEE 802.21 MIH services, we propose a new TCP architecture that is able to anticipate handoffs, permitting to resume the communication after a handoff, avoiding long delays caused by TCP issues and adapting the TCP parameters to the new characteristics of the network. Using the architecture proposed, the performance of TCP is enhanced, getting a higher overall throughput and avoiding TCP fairness issues between users.Els Sistemes de Transport Intel·ligents (ITS) han impulsat les comunicacions vehiculars en l'actualitat. Les comunicacions vehiculars és un camp d'investigació de moda, i està atraient un gran interès en la indústria automobilística i de les telecomunicacions. En el camp de les comunicacions vehiculars, hi ha principalment dues línies de treball: (1) serveis de comunicacions relacionats amb la seguretat viària i la informació del trànsit; i (2) serveis d'informació i entreteniment, també anomenats serveis d'infotainment. Aquests últims inclouen tant serveis multimèdia (veu sobre IP, streaming, jocs on-line, etc.), com serveis clàssics de dades (correu electrònic, accés a xarxes privades, navegació web, compartir arxius, etc.). En aquesta tesi ens centrarem en aquests serveis d'infotainment ja que és necessari aprofundir en la investigació per aquests tipus de serveis, ja que, fins avui, els esforços de la comunitat científica en el camp de les comunicacions vehiculars s'ha centrat en els serveis relacionats amb la seguretat viària i la informació del trànsit. Els nodes vehiculars necessiten tenir connexió a Internet per a poder tenir accés als serveis d'infotainment. Mentre els vehicles estan en moviment a través de la xarxa viària, els vehicles han d'anar canviant el punt de connexió sense fils amb la xarxa. Durant aquest procés de canvi de punt de connexió, anomenat handoff, es perd la connectivitat fins que el vehicle es reconnecta a un altre punt de connexió viària prop de la seva àrea. Aquesta desconnexió causa interrupcions en les comunicacions. Uns handoffs ràpids són bàsics a les xarxes vehiculars per a evitar llargs períodes d'interrupció durant les comunicacions, ja que la gran velocitat a la que es mouen els nodes vehiculars significa un gran nombre de handoffs durant una connexió a Internet. Aquesta tesi es centra en serveis d'infotaiment en temps real per a comunicacions Vehicle-a-Infraestructura (V2I). Les principals contribucions d'aquesta tesi son: i) un nou marc de proves per a les comunicacions (V2I) per a poder provar serveis d'infotainment d'una manera fàcil; ii) l'anàlisi de la viabilitat del desplegament de serveis d'infotainment de vídeo en xarxes vehiculars utilitzant protocols de mobilitat IP; i iii) el desenvolupament d'una nova arquitectura TCP que proporciona un millor funcionament per a tots aquells serveis d'infotainment basats en el protocol TCP en un escenari vehicular amb handoffs. En aquesta tesi, primer proposem un nou marc de proves per a aplicacions vehiculars d'infotainment. Aquest marc és una plataforma d'emulació vehicular que permet provar aplicacions reals instal·lades en màquines virtuals Linux. Utilitzant l'emulació, som capaços d'avaluar el rendiment d'aplicacions reals amb característiques de temps real. D'aquesta manera es poden avaluar aplicacions multimèdia utilitzades per oferir serveis d'infotainment d'una forma senzilla en escenaris vehiculars. Segon, utilitzant el marc de prova implementat en la primera part de la tesi, hem avaluat el rendiment d'un servei d'infotainment. Entre aquest tipus de servei, creem que els serveis de vídeo sota demanda en autopistes/autovies serà interessant pels usuaris i generarà beneficis per als operadors de la xarxa. Per tant, hem avaluat com els handoffs a nivell de la capa de xarxa poden limitar el desplegament d'un servei de streaming de vídeo sota demanda. D'acord amb els resultats obtinguts, conduir a grans velocitats podria ser un problema per a poder reproduir un vídeo correctament, tot i utilitzar tècniques de handoffs ràpids. Finalment, hem desenvolupat una nova arquitectura TCP per a millorar el rendiment del protocol durant els handoffs. La majoria dels serveis d'infotainment utilitzen el Protocol de Control de Transport (TCP), un dels principals protocols de la pila de protocols d'Internet. Però existeixen forces problemes relacionats amb l'ús de TCP i la mobilitat que n'afecta el rendiment, i aquests problemes són particular

Quality-Oriented Mobility Management for Multimedia Content Delivery to Mobile Users

The heterogeneous wireless networking environment determined by the latest developments in wireless access technologies promises a high level of communication resources for mobile computational devices. Although the communication resources provided, especially referring to bandwidth, enable multimedia streaming to mobile users, maintaining a high user perceived quality is still a challenging task. The main factors which affect quality in multimedia streaming over wireless networks are mainly the error-prone nature of the wireless channels and the user mobility. These factors determine a high level of dynamics of wireless communication resources, namely variations in throughput and packet loss as well as network availability and delays in delivering the data packets. Under these conditions maintaining a high level of quality, as perceived by the user, requires a quality oriented mobility management scheme. Consequently we propose the Smooth Adaptive Soft-Handover Algorithm, a novel quality oriented handover management scheme which unlike other similar solutions, smoothly transfer the data traffic from one network to another using multiple simultaneous connections. To estimate the capacity of each connection the novel Quality of Multimedia Streaming (QMS) metric is proposed. The QMS metric aims at offering maximum flexibility and efficiency allowing the applications to fine tune the behavior of the handover algorithm. The current simulation-based performance evaluation clearly shows the better performance of the proposed Smooth Adaptive Soft-Handover Algorithm as compared with other handover solutions. The evaluation was performed in various scenarios including multiple mobile hosts performing handover simultaneously, wireless networks with variable overlapping areas, and various network congestion levels

Energy-aware adaptive solutions for multimedia delivery to wireless devices

The functionality of smart mobile devices is improving rapidly but these devices are limited in terms of practical use because of battery-life. This situation cannot be remedied by simply installing batteries with higher capacities in the devices. There are strict limitations in the design of a smartphone, in terms of physical space, that prohibit this “quick-fix” from being possible. The solution instead lies with the creation of an intelligent, dynamic mechanism for utilizing the hardware components on a device in an energy-efficient manner, while also maintaining the Quality of Service (QoS) requirements of the applications running on the device. This thesis proposes the following Energy-aware Adaptive Solutions (EASE): 1. BaSe-AMy: the Battery and Stream-aware Adaptive Multimedia Delivery (BaSe-AMy) algorithm assesses battery-life, network characteristics, video-stream properties and device hardware information, in order to dynamically reduce the power consumption of the device while streaming video. The algorithm computes the most efficient strategy for altering the characteristics of the stream, the playback of the video, and the hardware utilization of the device, dynamically, while meeting application’s QoS requirements. 2. PowerHop: an algorithm which assesses network conditions, device power consumption, neighboring node devices and QoS requirements to decide whether to adapt the transmission power or the number of hops that a device uses for communication. PowerHop’s ability to dynamically reduce the transmission power of the device’s Wireless Network Interface Card (WNIC) provides scope for reducing the power consumption of the device. In this case shorter transmission distances with multiple hops can be utilized to maintain network range. 3. A comprehensive survey of adaptive energy optimizations in multimedia-centric wireless devices is also provided. Additional contributions: 1. A custom video comparison tool was developed to facilitate objective assessment of streamed videos. 2. A new solution for high-accuracy mobile power logging was designed and implemented

Performance Evaluation of H.264/AVC encoded Video over TETRA Enhanced Data Service (TEDS)

Public Safety Systems (PSS) are communication networks oriented towards supporting activities of public safety actors (police, medical, fire-fighters, etc.). TErrestrial Trunked RAdio (TETRA) is a Professional Mobile Radio (PMR) standard designed to meet PSS requirements with specialized voice communication features and reliable, secure communication links. TETRA Release 2 introduces TETRA Enhanced Data Service (TEDS), to support emerging data-intensive applications such as online navigation and tele-medicine by providing higher, scalable data rates. This thesis studies the feasibility of streaming video over a wideband TEDS link using the H.264/AVC codec, a video compression standard that manages to retain high decoded video quality while dramatically reducing streaming bit rate. A bandwidth limiter is used to emulate a link that supports data rates equivalent to those specified in the TEDS standard. Effects of video streaming parameters such as codec rate and play-out buffer size coupled with link-induced delay variation on decoded video quality are investigated. Visual quality is rated using objective quality metrics to quantify results with some measure of reliability. The overall aim is to identify the technical requirements needed to support an acceptable quality of video transmission over TEDS. To this end, we measure decoded video quality in different channel loss conditions, varying video streaming parameters and at different channel bandwidths, plus enhancements such as data traffic prioritisation as defined in the TEDS specification

Quality of service differentiation for multimedia delivery in wireless LANs

Delivering multimedia content to heterogeneous devices over a variable networking environment while maintaining high quality levels involves many technical challenges. The research reported in this thesis presents a solution for Quality of Service (QoS)-based service differentiation when delivering multimedia content over the wireless LANs. This thesis has three major contributions outlined below: 1. A Model-based Bandwidth Estimation algorithm (MBE), which estimates the available bandwidth based on novel TCP and UDP throughput models over IEEE 802.11 WLANs. MBE has been modelled, implemented, and tested through simulations and real life testing. In comparison with other bandwidth estimation techniques, MBE shows better performance in terms of error rate, overhead, and loss. 2. An intelligent Prioritized Adaptive Scheme (iPAS), which provides QoS service differentiation for multimedia delivery in wireless networks. iPAS assigns dynamic priorities to various streams and determines their bandwidth share by employing a probabilistic approach-which makes use of stereotypes. The total bandwidth to be allocated is estimated using MBE. The priority level of individual stream is variable and dependent on stream-related characteristics and delivery QoS parameters. iPAS can be deployed seamlessly over the original IEEE 802.11 protocols and can be included in the IEEE 802.21 framework in order to optimize the control signal communication. iPAS has been modelled, implemented, and evaluated via simulations. The results demonstrate that iPAS achieves better performance than the equal channel access mechanism over IEEE 802.11 DCF and a service differentiation scheme on top of IEEE 802.11e EDCA, in terms of fairness, throughput, delay, loss, and estimated PSNR. Additionally, both objective and subjective video quality assessment have been performed using a prototype system. 3. A QoS-based Downlink/Uplink Fairness Scheme, which uses the stereotypes-based structure to balance the QoS parameters (i.e. throughput, delay, and loss) between downlink and uplink VoIP traffic. The proposed scheme has been modelled and tested through simulations. The results show that, in comparison with other downlink/uplink fairness-oriented solutions, the proposed scheme performs better in terms of VoIP capacity and fairness level between downlink and uplink traffic