Internet protocol television (IPTV): The Killer application for the next-generation internet

Internet Protocol Television (IPTV) will be the killer application for the next-generation Internet and will provide exciting new revenue opportunities for service providers. However, to deploy IPTV services with a full quality of service (QoS) guarantee, many underlying technologies must be further studied. This article serves as a survey of IPTV services and the underlying technologies. Technical challenges also are identified

Joint in-network video rate adaptation and measurement-based admission control: algorithm design and evaluation

The important new revenue opportunities that multimedia services offer to network and service providers come with important management challenges. For providers, it is important to control the video quality that is offered and perceived by the user, typically known as the quality of experience (QoE). Both admission control and scalable video coding techniques can control the QoE by blocking connections or adapting the video rate but influence each other's performance. In this article, we propose an in-network video rate adaptation mechanism that enables a provider to define a policy on how the video rate adaptation should be performed to maximize the provider's objective (e.g., a maximization of revenue or QoE). We discuss the need for a close interaction of the video rate adaptation algorithm with a measurement based admission control system, allowing to effectively orchestrate both algorithms and timely switch from video rate adaptation to the blocking of connections. We propose two different rate adaptation decision algorithms that calculate which videos need to be adapted: an optimal one in terms of the provider's policy and a heuristic based on the utility of each connection. Through an extensive performance evaluation, we show the impact of both algorithms on the rate adaptation, network utilisation and the stability of the video rate adaptation. We show that both algorithms outperform other configurations with at least 10 %. Moreover, we show that the proposed heuristic is about 500 times faster than the optimal algorithm and experiences only a performance drop of approximately 2 %, given the investigated video delivery scenario

Comparación de técnicas de QoS en una red IP para aplicaciones de videostreaming

En el siguiente documento se realiza un estudio comparativo e introductorio de las estrategias de calidad de servicio utilizadas para la transmisión de video. En primera instancia se ilustra al lector sobre los conceptos básicos de QoS y de técnicas de videostreaming. Se definen conceptos, técnicas y estrategias para el manejo de congestión en redes ip y lo concerniente a la codificación de los archivos antes de transmitir para hacer más eficiente procesos como partición de información, serialización y transmisión de los datos a través de la red. En segundo lugar, se define el tráfico a usar en la aplicación, y al cual se le aplican las políticas de calidad de servicio, las cuales son el objeto de comparación en el siguiente trabajo de investigación. Para realizar el análisis correspondiente, se configura en el laboratorio un escenario apto para evaluar y comparar las técnicas de calidad de servicio escogidas, este está conformado por dos redes LAN de dos computadores cada una, comunicadas entre sí por dos dispositivos enrutadores que emulan una WAN y que mediante un enlace de 1544 kbps generan un cuello de botella que facilita la generación de congestión en el circuito en los momentos de transmisión. La transmisión es hecha en una sola dirección utilizando como protocolo de transporte RTP y con los requerimientos de ancho de banda definidos previamente. El contenido está configurado para que en el momento en que los servidores transmitan al tiempo encuentren que el enlace no es suficiente para los dos y se produzcan errores y pérdidas de paquetes. Basados en esta congestión se aplican tres estrategias de encolamiento para evaluar su funcionamiento ante el problema de congestión. FIFO, WFQ y CB-WFQ son las técnicas escogidas. Se realizan capturas mediante un software de tipo sniffer en las interfaces de los clientes y se evalúan los parámetros de medida de calidad de servicio como lo son: pérdida de paquetes, Jitter, OWD (One Way Delay) y caudal. Teniendo los resultados de cada estrategia, se compara el desempeño de cada una en el momento de cogestión y se concluye con base en esto.Incluye bibliografía, anexo

Measuring And Improving Internet Video Quality Of Experience

Streaming multimedia content over the IP-network is poised to be the dominant Internet traffic for the coming decade, predicted to account for more than 91% of all consumer traffic in the coming years. Streaming multimedia content ranges from Internet television (IPTV), video on demand (VoD), peer-to-peer streaming, and 3D television over IP to name a few. Widespread acceptance, growth, and subscriber retention are contingent upon network providers assuring superior Quality of Experience (QoE) on top of todays Internet. This work presents the first empirical understanding of Internet’s video-QoE capabilities, and tools and protocols to efficiently infer and improve them. To infer video-QoE at arbitrary nodes in the Internet, we design and implement MintMOS: a lightweight, real-time, noreference framework for capturing perceptual quality. We demonstrate that MintMOS’s projections closely match with subjective surveys in accessing perceptual quality. We use MintMOS to characterize Internet video-QoE both at the link level and end-to-end path level. As an input to our study, we use extensive measurements from a large number of Internet paths obtained from various measurement overlays deployed using PlanetLab. Link level degradations of intra– and inter–ISP Internet links are studied to create an empirical understanding of their shortcomings and ways to overcome them. Our studies show that intra–ISP links are often poorly engineered compared to peering links, and that iii degradations are induced due to transient network load imbalance within an ISP. Initial results also indicate that overlay networks could be a promising way to avoid such ISPs in times of degradations. A large number of end-to-end Internet paths are probed and we measure delay, jitter, and loss rates. The measurement data is analyzed offline to identify ways to enable a source to select alternate paths in an overlay network to improve video-QoE, without the need for background monitoring or apriori knowledge of path characteristics. We establish that for any unstructured overlay of N nodes, it is sufficient to reroute key frames using a random subset of k nodes in the overlay, where k is bounded by O(lnN). We analyze various properties of such random subsets to derive simple, scalable, and an efficient path selection strategy that results in a k-fold increase in path options for any source-destination pair; options that consistently outperform Internet path selection. Finally, we design a prototype called source initiated frame restoration (SIFR) that employs random subsets to derive alternate paths and demonstrate its effectiveness in improving Internet video-QoE

PRAM: Penalized Resource Allocation Method for Video Services

The human visual system response to picture quality degradation due to packet loss is very different from the responses of objective quality measures. While video quality due to packet loss may be impaired by at most for one Group of Pictures (GOP), its subjective quality degradation may last for several GOPs. This has a great impact on resource allocation strategies, which normally make decisions on instantaneous conditions of multiplexing buffer. This is because, when the perceptual impact of degraded video quality is much longer than its objective degradation period, any assigned resources to the degraded flow is wasted. This paper, through both simulations and analysis shows that, during resource allocation, if the quality of a video stream is significantly degraded, it is better to penalize this degraded flow from getting its full bandwidth share and instead assign the remaining share to other flows preventing them from undergoing quality degradation

Architecture and Protocol to Optimize Videoconference in Wireless Networks

[EN] In the past years, videoconferencing (VC) has become an essential means of communications. VC allows people to communicate face to face regardless of their location, and it can be used for different purposes such as business meetings, medical assistance, commercial meetings, and military operations. There are a lot of factors in real-time video transmission that can affect to the quality of service (QoS) and the quality of experience (QoE). The application that is used (Adobe Connect, Cisco Webex, and Skype), the internet connection, or the network used for the communication can affect to the QoE. Users want communication to be as good as possible in terms of QoE. In this paper, we propose an architecture for videoconferencing that provides better quality of experience than other existing applications such as Adobe Connect, Cisco Webex, and Skype. We will test how these three applications work in terms of bandwidth, packets per second, and delay using WiFi and 3G/4G connections. Finally, these applications are compared to our prototype in the same scenarios as they were tested, and also in an SDN, in order to improve the advantages of the prototype.This work has been supported by the "Ministerio de Economia y Competitividad" in the "Programa Estatal de Fomento de la Investigacion Cientifica y Tecnica de Excelencia, Subprograma Estatal de Generacion de Conocimiento" within the project under Grant TIN2017-84802-C2-1-P.Jimenez, JM.; García-Navas, JL.; Lloret, J.; Romero Martínez, JO. (2020). Architecture and Protocol to Optimize Videoconference in Wireless Networks. Wireless Communications and Mobile Computing. 2020:1-22. https://doi.org/10.1155/2020/4903420S122202

QoS Provisioning in Converged Satellite and Terrestrial Networks: A Survey of the State-of-the-Art

It has been widely acknowledged that future networks will need to provide significantly more capacity than current ones in order to deal with the increasing traffic demands of the users. Particularly in regions where optical fibers are unlikely to be deployed due to economical constraints, this is a major challenge. One option to address this issue is to complement existing narrow-band terrestrial networks with additional satellite connections. Satellites cover huge areas, and recent developments have considerably increased the available capacity while decreasing the cost. However, geostationary satellite links have significantly different link characteristics than most terrestrial links, mainly due to the higher signal propagation time, which often renders them not suitable for delay intolerant traffic. This paper surveys the current state-of-the-art of satellite and terrestrial network convergence. We mainly focus on scenarios in which satellite networks complement existing terrestrial infrastructures, i.e., parallel satellite and terrestrial links exist, in order to provide high bandwidth connections while ideally achieving a similar end user quality-of-experience as in high bandwidth terrestrial networks. Thus, we identify the technical challenges associated with the convergence of satellite and terrestrial networks and analyze the related work. Based on this, we identify four key functional building blocks, which are essential to distribute traffic optimally between the terrestrial and the satellite networks. These are the traffic requirement identification function, the link characteristics identification function, as well as the traffic engineering function and the execution function. Afterwards, we survey current network architectures with respect to these key functional building blocks and perform a gap analysis, which shows that all analyzed network architectures require adaptations to effectively support converged satellite and terrestrial networks. Hence, we conclude by formulating several open research questions with respect to satellite and terrestrial network convergence.This work was supported by the BATS Research Project through the European Union Seventh Framework Programme under Contract 317533

Quality of Service improvements for real time multimedia applications using next generation network architectures and blockchain in Internet Service Provider cooperative scenario

Real time communications are becoming part of our daily life, requiring constrained requisites with the purpose of being enjoyed in harmony by end users. The factors ruling these requisites are Quality of Service parameters of the users' Internet connections. Achieving a satisfactory QoS level for real time communications depends on parameters that are strongly influenced by the quality of the network connections among the Internet Service Providers, which are located in the path between final users and Over The Top service providers that are supplying them with real time services. Final users can be: business people having real time videoconferences, or adopting crytpocurrencies in their exchanges, videogamers playing online games together with others residing in other countries, migrants talking with their relatives or watching their children growing up in their home countries, people with disabilities adopting tecnologies to help them, doctors performing remote surgeries, manufacturers adopting augmented reality devices to perform dangerous tasks. Each of them performing their daily activities are requiring specific QoS parameters to their ISPs, that nowadays seem to be unable to provide them with a satisfactory QoS level for these kinds of real time services. Through the adoption of next generation networks, such as the Information Centric Networking, it would be possible to overcome the QoS problems that nowadays are experienced. By adopting Blockchain technologies, in several use cases, it would be possible to improve those security aspects related to the non-temperability of information and privacy. I started this thesis analyzing next generation architectures enabling real time multimedia communications. In Software Defined Networking, Named Data Networking and Community Information Centric Networking, I highlighted potential approaches to solve QoS problems that are affecting real time multimedia applications. During my experiments I found that applications able to transmit high quality videos, such as 4k or 8k videos, or to directly interact with devices AR/VR enabled are missing for both ICN approaches. Then I proposed a REST interface for the enforcing of a specific QoS parameter, the round trip time (RTT) taking into consideration the specific use case of a game company that connects with the same telecommunication company of the final user. Supposing that the proposed REST APIs have been deployed in the game company and in the ISP, when one or more users are experiencing lag, the game company will try to ask the ISP to reduce the RTT for that specific user or that group of users. This request can be done by performing a call to a method where IP address(es) and the maximum RTT desired are passed. I also proposed other methods, through which it would be possible to retrieve information about the QoS parameters, and exchange, if necessary, an exceeding parameter in change of another one. The proposed REST APIs can also be used in more complex scenarios, where ISPs along the path are chained together, in order to improve the end to end QoS among Over The Top service provider and final users. To store the information exchanged by using the proposed REST APIs, I proposed to adopt a permissioned blockchain, analizying the ISPs cooperative use case with Hyperledger Fabric, where I proposed the adoption of the Proof of Authority consensus algorithm, to increase the throughput in terms of transactions per second. In a specific case that I examined, I am proposing a combination of Information Centric Networking and Blockchain, in an architecture where ISPs are exchanging valuable information regarding final Users, to improve their QoS parameters. I also proposed my smart contract for the gaming delay use case, that can be used to rule the communication among those ISPs that are along the path among OTT and final users. An extension of this work can be done, by defining billing costs for the QoS improvements