Business model with discount incentive in a P2P-cloud multimedia streaming system

Today P2P faces two important challenges: design of mechanisms to encourage users' collaboration in multimedia live streaming services; design of reliable algorithms with QoS provision, to encourage the multimedia providers employ the P2P topology in commercial live streaming systems. We believe that these two challenges are tightly-related and there is much to be done with respect. This paper analyzes the effect of user behavior in a multi-tree P2P overlay and describes a business model based on monetary discount as incentive in a P2P-Cloud multimedia streaming system. We believe a discount model can boost up users' cooperation and loyalty and enhance the overall system integrity and performance. Moreover the model bounds the constraints for a provider's revenue and cost if the P2P system is leveraged on a cloud infrastructure. Our case study shows that a streaming system provider can establish or adapt his business model by applying the described bounds to achieve a good discount-revenue trade-off and promote the system to the users.

Motion vector encryption in multimedia streaming

Streaming video application requires high security as well as high computational performance. In video encryption, traditional selective algorithms have been used to partially encrypt the relatively important data in order to satisfy the streaming performance requirement. Most video selective encryption algorithms are inherited from still image encryption algorithms, the encryption on motion vector data is not considered. The assumption is that motion vector data are not as important as pixel image data. Unfortunately, in some cases, motion vector itself may be sufficient enough to leak out useful video information. Normally motion vector data consume over half of the whole video stream bandwidth, neglecting their security may be unwise. In this paper, we target this security problem and illustrate attacks at two different levels that can restore useful video information using motion vectors only. Further, an information analysis is made and a motion vector information model is built. Based on this model, we describe a new motion vector encryption algorithm called MVEA. We show the experimental results of MVEA. The security strength and performance of the algorithm are also evaluated.

Opportunities and Challenges of Implementing P2P Streaming Applications in the Web

P2P applications are increasingly present on the web. We have identified a gap in current proposals when it comes to the use of traditional P2P overlays for real-time multimedia streaming. We analyze the possibilities and challenges to extend WebRTC in order to implement JavaScript APIs for P2P streaming algorithms.

A novel monetary incentive for commercial P2P streaming

Today P2P faces two important challenges: design of mechanisms to encourage users’ collaboration in multimedia live streaming services; design of reliable algorithms with QoS provision, to encourage multimedia providers employ the P2P topology in commercial streaming services. We believe that these two challenges are tightly-related and there is much to be done with respect. This paper proposes a novel monetary incentive for P2P multimedia streaming. The incentive model classifies the users in groups according to the perceived video quality. We apply the model to a streaming system’s billing model in order to evaluate its feasibility and visualize its quantitative effect on the users’ motivation and the provider’s profit. We conclude that monetary incentive can boost up users’ cooperation, loyalty and enhance the overall system integrity and performance. Moreover the model defines the constraints for the provider’s cost and profit when the system is leveraged on the cloud. Considering those constraints, a multimedia content provider can adapt the billing model of his streaming service and achieve desirable discount-profit trade-off. This will moreover contribute to better promotion of the service, across the users on the Internet.

Um mecanismo para realçar a conectividade de roteamento geográfico na transmissão multimídia em VANTs

As redes de Veículos Aéreos Não Tripulados (VANTs) têm potencializado o uso de aplicações multimídia devido ao seu elevado grau de mobilidade e versatilidade. Esse crescimento ocorre pela facilidade de acesso e utilização dos VANTs quadricópteros em diversas aplicações civis. As aplicações civis têm como característica principal o uso de dados multimídia, como vídeos, por oferecerem uma perspectiva visual mais detalhada do ambiente. A transmissão desse tipo de dado nas redes VANTs por meio de protocolos geográ cos melhora a taxa de entrega de dados. Entretanto, ela ainda não é su ciente para prover qualidade de experiência (QoE). Isso ocorre devido ao elevado grau de mobilidade dos quadricópteros que ocasiona quebras de enlace durante a transmissão multimídia. Por consequência, ela prejudica a conectividade e induz a altas perdas de pacotes e atrasos. Então, este trabalho propõe um mecanismo, chamado RCRV, com base em técnicas de predição de mobilidade em termos de posicionamento e da estimativa do tempo de enlace para realçar a tomada de decisão de roteamento em protocolos geográ cos. Além disso, ele utiliza critérios que consideram os comportamentos de mobilidade dos VANTs. Assim, o RCRV possibilita o prolongamento da conectividade nas redes VANTs. O RCRV é um mecanismo que complementa as estratégias de roteamento dos protocolos geográ cos. Desse modo, adicionou-se o RCRV ao protocolo GPSR a m de avaliar os ganhos e benefícios do mecanismo em um protocolo de roteamento geográ co. Simulações foram realizadas com diferentes velocidades e números de VANTs. Os resultados mostraram que o RCRV aumenta a conectividade da transmissão, melhorando a entrega do conteúdo multimídia e a qualidade do vídeo observado pelo usuário.

Investigation on probing schemes in probe-based multicast admission control

Multicast is an efficient approach to save network bandwidth for multimedia streaming services. To provide Quality of Services (QoS) for the multimedia services while maintain the advantage of multicast in bandwidth efficiency, admission control for multicast sessions are expected. Probe-based multicast admission control (PBMAC) schemes are of a sort of scalable and simple admission control for multicast. Probing scheme is the essence of PBMAC. In this paper, after a detailed survey on three existing probing schemes, we evaluate these schemes using simulation and analysis approaches in two aspects: admission correctness and group scalability. Admission correctness of the schemes is compared by simulation investigation. Analytical models for group scalability are derived, and validated by simulation results. The evaluation results illustrate the advantages and weaknesses of each scheme, which are helpful for people to choose proper probing scheme for network.

Reliable Broadcasting and Streaming of Multimedia Content

This thesis is focused on the study of techniques that allow to have reliable transmission of multimedia content in streaming and broadcasting applications, targeting in particular video content. The design of efficient error-control mechanisms, to enhance video transmission systems reliability, has been addressed considering cross-layer and multi-layer/multi-dimensional channel coding techniques to cope with bit errors as well as packet erasures. Mechanisms for unequal time interleaving have been designed as a viable solution to reduce the impact of errors and erasures by acting on the time diversity of the data flow, thus enhancing robustness against correlated channel impairments. In order to account for the nature of the factors which affect the physical layer channel in the evaluation of FEC schemes performances, an ad-hoc error-event modeling has been devised. In addition, the impact of error correction/protection techniques on the quality perceived by the consumers of video services applications and techniques for objective/subjective quality evaluation have been studied. The applicability and value of the proposed techniques have been tested by considering practical constraints and requirements of real system implementations.

Mobile multimedia : user and technology perspectives

As multimedia-enabled mobile devices such as smart phones and tablets are becoming the day-to-day computing device of choice for users of all ages, everyone expects that all mobile multimedia applications and services should be as smooth and as high-quality as the desktop experience. The grand challenge in delivering multimedia to mobile devices using the Internet is to ensure the quality of experience that meets the users' expectations, within reasonable costs, while supporting heterogeneous platforms and wireless network conditions. This book aims to provide a holistic overview of the current and future technologies used for delivering high-quality mobile multimedia applications, while focusing on user experience as the key requirement. The book opens with a section dealing with the challenges in mobile video delivery as one of the most bandwidth-intensive media that requires smooth streaming and a user-centric strategy to ensure quality of experience. The second section addresses this challenge by introducing some important concepts for future mobile multimedia coding and the network technologies to deliver quality services. The last section combines the user and technology perspectives by demonstrating how user experience can be measured using case studies on urban community interfaces and Internet telephones.

An analytical model for performance evaluation of multimedia applications over EDCA in an IEEE 802.11e WLAN

We extend the modeling heuristic of (Harsha et al. 2006. In IEEE IWQoS 06, pp 178 - 187) to evaluate the performance of an IEEE 802.11e infrastructure network carrying packet telephone calls, streaming video sessions and TCP controlled file downloads, using Enhanced Distributed Channel Access (EDCA). We identify the time boundaries of activities on the channel (called channel slot boundaries) and derive a Markov Renewal Process of the contending nodes on these epochs. This is achieved by the use of attempt probabilities of the contending nodes as those obtained from the saturation fixed point analysis of (Ramaiyan et al. 2005. In Proceedings ACM Sigmetrics, `05. Journal version accepted for publication in IEEE TON). Regenerative analysis on this MRP yields the desired steady state performance measures. We then use the MRP model to develop an effective bandwidth approach for obtaining a bound on the size of the buffer required at the video queue of the AP, such that the streaming video packet loss probability is kept to less than 1%. The results obtained match well with simulations using the network simulator, ns-2. We find that, with the default IEEE 802.11e EDCA parameters for access categories AC 1, AC 2 and AC 3, the voice call capacity decreases if even one streaming video session and one TCP file download are initiated by some wireless station. Subsequently, reducing the voice calls increases the video downlink stream throughput by 0.38 Mbps and file download capacity by 0.14 Mbps, for every voice call (for the 11 Mbps PHY). We find that a buffer size of 75KB is sufficient to ensure that the video packet loss probability at the QAP is within 1%.

Stochastic models of IEEE 802.11e wireless net works with multimedia applications

We provide a survey of some of our recent results ([9], [13], [4], [6], [7]) on the analytical performance modeling of IEEE 802.11 wireless local area networks (WLANs). We first present extensions of the decoupling approach of Bianchi ([1]) to the saturation analysis of IEEE 802.11e networks with multiple traffic classes. We have found that even when analysing WLANs with unsaturated nodes the following state dependent service model works well: when a certain set of nodes is nonempty, their channel attempt behaviour is obtained from the corresponding fixed point analysis of the saturated system. We will present our experiences in using this approximation to model multimedia traffic over an IEEE 802.11e network using the enhanced DCF channel access (EDCA) mechanism. We have found that we can model TCP controlled file transfers, VoIP packet telephony, and streaming video in the IEEE802.11e setting by this simple approximation.

SeaMo plus : a virtual real-time multimedia service framework on handhelds to enable remote real-time patient monitoring for mobile doctors

The key requirements for enabling real-time remote healthcare service on a mobile platform, in the present day heterogeneous wireless access network environment, are uninterrupted and continuous access to the online patient vital medical data, monitor the physical condition of the patient through video streaming, and so on. For an application, this continuity has to be sufficiently transparent both from a performance perspective as well as a Quality of Experience (QoE) perspective. While mobility protocols (MIPv6, HIP, SCTP, DSMIP, PMIP, and SIP) strive to provide both and do so, limited or non-availability (deployment) of these protocols on provider networks and server side infrastructure has impeded adoption of mobility on end user platforms. Add to this, the cumbersome OS configuration procedures required to enable mobility protocol support on end user devices and the user's enthusiasm to add this support is lost. Considering the lack of proper mobility implementations that meet the remote healthcare requirements above, we propose SeaMo+ that comprises a light-weight application layer framework, termed as the Virtual Real-time Multimedia Service (VRMS) for mobile devices to provide an uninterrupted real-time multimedia information access to the mobile user. VRMS is easy to configure, platform independent, and does not require additional network infrastructure unlike other existing schemes. We illustrate the working of SeaMo+ in two realistic remote patient monitoring application scenarios.

Dynamic Window-Constrained Scheduling for Real-Time Media Streaming

This paper describes an algorithm for scheduling packets in real-time multimedia data streams. Common to these classes of data streams are service constraints in terms of bandwidth and delay. However, it is typical for real-time multimedia streams to tolerate bounded delay variations and, in some cases, finite losses of packets. We have therefore developed a scheduling algorithm that assumes streams have window-constraints on groups of consecutive packet deadlines. A window-constraint defines the number of packet deadlines that can be missed in a window of deadlines for consecutive packets in a stream. Our algorithm, called Dynamic Window-Constrained Scheduling (DWCS), attempts to guarantee no more than x out of a window of y deadlines are missed for consecutive packets in real-time and multimedia streams. Using DWCS, the delay of service to real-time streams is bounded even when the scheduler is overloaded. Moreover, DWCS is capable of ensuring independent delay bounds on streams, while at the same time guaranteeing minimum bandwidth utilizations over tunable and finite windows of time. We show the conditions under which the total demand for link bandwidth by a set of real-time (i.e., window-constrained) streams can exceed 100% and still ensure all window-constraints are met. In fact, we show how it is possible to guarantee worst-case per-stream bandwidth and delay constraints while utilizing all available link capacity. Finally, we show how best-effort packets can be serviced with fast response time, in the presence of window-constrained traffic.

A programmable architecture for layered multimedia streams in IPv6 networks

A new configurable architecture is presented that offers multiple levels of video playback by accommodating variable levels of network utilization and bandwidth. By utilizing scalable MPEG-4 encoding at the network edge and using specific video delivery protocols, media streaming components are merged to fully optimize video playback for IPv6 networks, thus improving QoS. This is achieved by introducing “programmable network functionality” (PNF) which splits layered video transmission and distributes it evenly over available bandwidth, reducing packet loss and delay caused by out-of-profile DiffServ classes. An FPGA design is given which gives improved performance, e.g. link utilization, end-to-end delay, and that during congestion, improves on-time delivery of video frames by up to 80% when compared to current “static” DiffServ.