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.

Scalable Overlay Multicast Tree Construction for QoS-Constrained Media Streaming

Overlay networks have become popular in recent times for content distribution and end-system multicasting of media streams. In the latter case, the motivation is based on the lack of widespread deployment of IP multicast and the ability to perform end-host processing. However, constructing routes between various end-hosts, so that data can be streamed from content publishers to many thousands of subscribers, each having their own QoS constraints, is still a challenging problem. First, any routes between end-hosts using trees built on top of overlay networks can increase stress on the underlying physical network, due to multiple instances of the same data traversing a given physical link. Second, because overlay routes between end-hosts may traverse physical network links more than once, they increase the end-to-end latency compared to IP-level routing. Third, algorithms for constructing efficient, large-scale trees that reduce link stress and latency are typically more complex. This paper therefore compares various methods to construct multicast trees between end-systems, that vary in terms of implementation costs and their ability to support per-subscriber QoS constraints. We describe several algorithms that make trade-offs between algorithmic complexity, physical link stress and latency. While no algorithm is best in all three cases we show how it is possible to efficiently build trees for several thousand subscribers with latencies within a factor of two of the optimal, and link stresses comparable to, or better than, existing technologies.

Efficient delivery of scalable media streaming over lossy networks

Recent years have witnessed a rapid growth in the demand for streaming video over the Internet, exposing challenges in coping with heterogeneous device capabilities and varying network throughput. When we couple this rise in streaming with the growing number of portable devices (smart phones, tablets, laptops) we see an ever-increasing demand for high-definition videos online while on the move. Wireless networks are inherently characterised by restricted shared bandwidth and relatively high error loss rates, thus presenting a challenge for the efficient delivery of high quality video. Additionally, mobile devices can support/demand a range of video resolutions and qualities. This demand for mobile streaming highlights the need for adaptive video streaming schemes that can adjust to available bandwidth and heterogeneity, and can provide us with graceful changes in video quality, all while respecting our viewing satisfaction. In this context the use of well-known scalable media streaming techniques, commonly known as scalable coding, is an attractive solution and the focus of this thesis. In this thesis we investigate the transmission of existing scalable video models over a lossy network and determine how the variation in viewable quality is affected by packet loss. This work focuses on leveraging the benefits of scalable media, while reducing the effects of data loss on achievable video quality. The overall approach is focused on the strategic packetisation of the underlying scalable video and how to best utilise error resiliency to maximise viewable quality. In particular, we examine the manner in which scalable video is packetised for transmission over lossy networks and propose new techniques that reduce the impact of packet loss on scalable video by selectively choosing how to packetise the data and which data to transmit. We also exploit redundancy techniques, such as error resiliency, to enhance the stream quality by ensuring a smooth play-out with fewer changes in achievable video quality. The contributions of this thesis are in the creation of new segmentation and encapsulation techniques which increase the viewable quality of existing scalable models by fragmenting and re-allocating the video sub-streams based on user requirements, available bandwidth and variations in loss rates. We offer new packetisation techniques which reduce the effects of packet loss on viewable quality by leveraging the increase in the number of frames per group of pictures (GOP) and by providing equality of data in every packet transmitted per GOP. These provide novel mechanisms for packetizing and error resiliency, as well as providing new applications for existing techniques such as Interleaving and Priority Encoded Transmission. We also introduce three new scalable coding models, which offer a balance between transmission cost and the consistency of viewable quality.

Media Streaming

Hundreds of streamable films and videos for free use in teaching and learning within the University of Southampton.

SDC: Scalable description coding for adaptive streaming media

Video compression techniques enable adaptive media streaming over heterogeneous links to end-devices. Scalable Video Coding (SVC) and Multiple Description Coding (MDC) represent well-known techniques for video compression with distinct characteristics in terms of bandwidth efficiency and resiliency to packet loss. In this paper, we present Scalable Description Coding (SDC), a technique to compromise the tradeoff between bandwidth efficiency and error resiliency without sacrificing user-perceived quality. Additionally, we propose a scheme that combines network coding and SDC to further improve the error resiliency. SDC yields upwards of 25% bandwidth savings over MDC. Additionally, our scheme features higher quality for longer durations even at high packet loss rates.

ALD: adaptive layer distribution for scalable video

Bandwidth constriction and datagram loss are prominent issues that affect the perceived quality of streaming video over lossy networks, such as wireless. The use of layered video coding seems attractive as a means to alleviate these issues, but its adoption has been held back in large part by the inherent priority assigned to the critical lower layers and the consequences for quality that result from their loss. The proposed use of forward error correction (FEC) as a solution only further burdens the bandwidth availability and can negate the perceived benefits of increased stream quality. In this paper, we propose Adaptive Layer Distribution (ALD) as a novel scalable media delivery technique that optimises the tradeoff between the streaming bandwidth and error resiliency. ALD is based on the principle of layer distribution, in which the critical stream data is spread amongst all datagrams thus lessening the impact on quality due to network losses. Additionally, ALD provides a parameterised mechanism for dynamic adaptation of the scalable video, while providing increased resilience to the highest quality layers. Our experimental results show that ALD improves the perceived quality and also reduces the bandwidth demand by up to 36% in comparison to the well-known Multiple Description Coding (MDC) technique.

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.

A Unified Load Generator for Geographically Distributed Generation ofNetwork Traffic

During the last decade, the Internet usage has been growing at an enormous rate which has beenaccompanied by the developments of network applications (e.g., video conference, audio/videostreaming, E-learning, E-Commerce and real-time applications) and allows several types ofinformation including data, voice, picture and media streaming. While end-users are demandingvery high quality of service (QoS) from their service providers, network undergoes a complex trafficwhich leads the transmission bottlenecks. Considerable effort has been made to study thecharacteristics and the behavior of the Internet. Simulation modeling of computer networkcongestion is a profitable and effective technique which fulfills the requirements to evaluate theperformance and QoS of networks. To simulate a single congested link, simulation is run with asingle load generator while for a larger simulation with complex traffic, where the nodes are spreadacross different geographical locations generating distributed artificial loads is indispensable. Onesolution is to elaborate a load generation system based on master/slave architecture.

Um protocolo de comunicação multicast na camada de aplicação com Consciência de Localização

Internet applications such as media streaming, collaborative computing and massive multiplayer are on the rise,. This leads to the need for multicast communication, but unfortunately group communications support based on IP multicast has not been widely adopted due to a combination of technical and non-technical problems. Therefore, a number of different application-layer multicast schemes have been proposed in recent literature to overcome the drawbacks. In addition, these applications often behave as both providers and clients of services, being called peer-topeer applications, and where participants come and go very dynamically. Thus, servercentric architectures for membership management have well-known problems related to scalability and fault-tolerance, and even peer-to-peer traditional solutions need to have some mechanism that takes into account member's volatility. The idea of location awareness distributes the participants in the overlay network according to their proximity in the underlying network allowing a better performance. Given this context, this thesis proposes an application layer multicast protocol, called LAALM, which takes into account the actual network topology in the assembly process of the overlay network. The membership algorithm uses a new metric, IPXY, to provide location awareness through the processing of local information, and it was implemented using a distributed shared and bi-directional tree. The algorithm also has a sub-optimal heuristic to minimize the cost of membership process. The protocol has been evaluated in two ways. First, through an own simulator developed in this work, where we evaluated the quality of distribution tree by metrics such as outdegree and path length. Second, reallife scenarios were built in the ns-3 network simulator where we evaluated the network protocol performance by metrics such as stress, stretch, time to first packet and reconfiguration group time

Adaptación de un reproductor de contenidos multimedia para la recepción de vídeo escalable sobre un protocolo de tiempo real

En los últimos años, debido al notable desarrollo de los terminales portátiles, que han pasado de ser “simples” teléfonos o reproductores a puros ordenadores, ha crecido el número de servicios que ofrecen cada vez mayor cantidad de contenido multimedia a través de internet. Además, la distinta evolución de estos terminales hace que nos encontremos en el mercado con una amplísima gama de productos de diferentes tamaños y capacidades de procesamiento, lo que hace necesario encontrar una fórmula que permita satisfacer la demanda de dichos servicios sea cual sea la naturaleza de nuestro dispositivo. Para poder ofrecer una solución adecuada se ha optado por la integración de un protocolo como RTP y un estándar de video como SVC. RTP (Real-time Transport Protocol), en contraposición a los protocolos de propósito general fue diseñado para aplicaciones de tiempo real por lo que es ideal para el streaming de contenido multimedia. Por su parte, SVC es un estándar de video escalable que permite transmitir en un mismo stream una capa base y múltiples capas de mejora, por lo que podremos adaptar la calidad y tamaño del contenido a la capacidad y tamaño de nuestro dispositivo. El objetivo de este proyecto consiste en integrar y modificar tanto el reproductor MPlayer como la librería RTP live555 de tal forma que sean capaces de soportar el formato SVC sobre el protocolo RTP y montar un sistema servidorcliente para comprobar su funcionamiento. Aunque este proceso esté orientado a llevarse a cabo en un dispositivo móvil, para este proyecto se ha optado por realizarlo en el escenario más sencillo posible, para lo cual, se emitirán secuencias a una máquina virtual alojada en el mismo ordenador que el servidor. ABSTRACT In recent years, due to the remarkable development of mobile devices, which have evolved from "simple" phones or players to computers, the amount of services that offer multimedia content over the internet have shot up. Furthermore, the different evolution of these terminals causes that we can find in the market a wide range of different sizes and processing capabilities, making necessary to find a formula that will satisfy the demand for such services regardless of the nature of our device. In order to provide a suitable solution we have chosen to integrate a protocol as RTP and a video standard as SVC. RTP (Real-time Transport Protocol), in opposition to general purpose protocols was designed for real-time applications making it ideal for media streaming. Meanwhile, SVC is a scalable video standard which can transmit a single stream in a base layer and multiple enhancement layers, so that we can adapt the quality and size of the content to the capacity and size of our device. The objective of this project is to integrate and modify both MPlayer and RTP library live555 so that they support the SVC format over RTP protocol and set up a client-server system to check its behavior. Although this process has been designed to be done on a mobile device, for this project we have chosen to do it in the simplest possible scenario so we will stream to a virtual machine hosted on the same computer where we have the server.

Towards near-threshold server processors

The popularity of cloud computing has led to a dramatic increase in the number of data centers in the world. The ever-increasing computational demands along with the slowdown in technology scaling has ushered an era of power-limited servers. Techniques such as near-threshold computing (NTC) can be used to improve energy efficiency in the post-Dennard scaling era. This paper describes an architecture based on the FD-SOI process technology for near-threshold operation in servers. Our work explores the trade-offs in energy and performance when running a wide range of applications found in private and public clouds, ranging from traditional scale-out applications, such as web search or media streaming, to virtualized banking applications. Our study demonstrates the benefits of near-threshold operation and proposes several directions to synergistically increase the energy proportionality of a near-threshold server.

Connected Viewing: Selling, Streaming and Sharing Media in the Digital Era

As patterns of media use become more integrated with mobile technologies and multiple screens, a new mode of viewer engagement has emerged in the form of connected viewing, which allows for an array of new relationships between audiences and media texts in the digital space. This exciting new collection brings together twelve original essays that critically engage with the socially-networked, multi-platform, and cloud-based world of today, examining the connected viewing phenomenon across television, film, video games, and social media. The result is a wide-ranging analysis of shifting business models, policy matters, technological infrastructure, new forms of user engagement, and other key trends affecting screen media in the digital era. Connected Viewing contextualizes the dramatic transformations taking place across both media industries and national contexts, and offers students and scholars alike a diverse set of methods and perspectives for studying this critical moment in media culture.

Infrastructure Disputes Shape Future of Streaming Media

In November 2010, tension between Internet infrastructure companies boiled over in a dispute between content distribution network (CDN) Level 3 and Internet service provider (ISP) Comcast. Level 3, a distribution partner of Netflix, accused Comcast of violating the principles of net neutrality when the ISP increased distribution fees for carrying high bandwidth services. Comcast justified its actions by stating that the price increase was standard practice and argued Level 3 was trying to avoid paying its fair share. The dispute exemplifies the growing concern over the rising costs of streaming media services. The companies facing these inflated infrastructure costs are CDNs (Level 3, Equinix, Limelight, Akamai, and Voxel), companies that host streaming media content on server farms and distribute the content to a variety of carriers, and ISPs (Comcast, Time Warner, Cox, and AT&T), the cable and phone companies that provide “last mile” service to paying customers. Both CDNs and ISPs are lobbying government regulators to keep their costs at a minimum. The outcome of these disputes will influence the cost, quality, and legal status of streaming media.

Gismo: A Generator of Internet Streaming Media Objects and Workloads

This paper presents a tool called Gismo (Generator of Internet Streaming Media Objects and workloads). Gismo enables the specification of a number of streaming media access characteristics, including object popularity, temporal correlation of request, seasonal access patterns, user session durations, user interactivity times, and variable bit-rate (VBR) self-similarity and marginal distributions. The embodiment of these characteristics in Gismo enables the generation of realistic and scalable request streams for use in the benchmarking and comparative evaluation of Internet streaming media delivery techniques. To demonstrate the usefulness of Gismo, we present a case study that shows the importance of various workload characteristics in determining the effectiveness of proxy caching and server patching techniques in reducing bandwidth requirements.

Accelerating Internet Streaming Media Delivery Using Network-Aware Partial Caching

Internet streaming applications are adversely affected by network conditions such as high packet loss rates and long delays. This paper aims at mitigating such effects by leveraging the availability of client-side caching proxies. We present a novel caching architecture (and associated cache management algorithms) that turn edge caches into accelerators of streaming media delivery. A salient feature of our caching algorithms is that they allow partial caching of streaming media objects and joint delivery of content from caches and origin servers. The caching algorithms we propose are both network-aware and stream-aware; they take into account the popularity of streaming media objects, their bit-rate requirements, and the available bandwidth between clients and servers. Using realistic models of Internet bandwidth (derived from proxy cache logs and measured over real Internet paths), we have conducted extensive simulations to evaluate the performance of various cache management alternatives. Our experiments demonstrate that network-aware caching algorithms can significantly reduce service delay and improve overall stream quality. Also, our experiments show that partial caching is particularly effective when bandwidth variability is not very high.