An energy efficient http adaptive streaming protocol design for mobile hand-held devices

Internet traffic generated from mobile devices has experienced a huge growth in the last few years. With the increasing popularity of streaming applications in mobile devices, video traffic generated from mobile devices is also increasing. One of the big challenges of streaming applications on mobile devices is the energy intensive behaviour of such applications. Energy management has always been a critical issue for mobile devices. A wireless network interface consumes a significant portion of the total system energy while active. During video streaming, the network interface is kept awake for a long period of time. This causes a large energy drain. There are several research works focused on reducing energy consumption during video streaming on mobile devices. HTTP adaptive streaming is gaining popularity as a method of video delivery because of its significant advantages in terms of both user-perceived quality and resource utilization. By using rate adaptation via changes in the requested video version, it adapts to varying network available capacity. There are several research work that aim to increase the performance of rate adaptation. None of the previous works have focused on reducing energy consumption during HTTP adaptive streaming. In this thesis, an energy efficient HTTP adaptive streaming protocol is designed. The new protocol uses an efficient buffer management approach and a three step bitrate selection mechanism. The proposed protocol is implemented by modifying the Adobe OSMF player version 1.6. Performance evaluation of the new protocol is carried out by running a number of experiments in both a lab environment and three real world environments. The experimental results show that the proposed protocol is able to achieve high amounts of sleep time (by more than an estimated 70% for WiFi and more than 35% for 3G/EDGE) and reduce energy consumption during data transfer. It can also reduce data wastage by 80% in case of playback interruption in the video playback

Performance Analysis in IP-Based Industrial Communication Networks

S rostoucím počtem řídicích systémů a jejich distribuovanosti získávájí komunikační sítě na důležitosti a objevují se nové výzkumné trendy. Hlavní problematikou v této oblasti, narozdíl od dřívějších řídicích systémů využívajících dedikovaných komunikačních obvodů, je časově proměnné zpoždění měřicích a řídicích signálů způsobené paketově orientovanými komunikačními prostředky, jako např. Ethernet. Aspekty komunikace v reálném čase byly v těchto sítích již úspěšně vyřešeny. Nicméně, analýzy trendů trhu předpovídají budoucí využití také IP sítí v průmyslové komunikaci pro časově kritickou procesní vyměnu dat. IP komunikace má ovšem pouze omezenou podporu v instrumentaci pro průmyslovou automatizace. Tato výzva byla nedávno technicky vyřešena v rámci projektu Virtual Automation Networks (virtuální automatizační sítě - VAN) zapojením mechanismů kvality služeb (QoS), které jsou schopny zajistit měkkou úroveň komunikace v reálném čase. Předložená dizertační práce se zaměřuje na aspekty výkonnosti reálného času z analytického hlediska a nabízí prostředek pro hodnocení využitelnosti IP komunikace pro budoucí průmyslové aplikace. Hlavním cílem této dizertační práce je vytvoření vhodného modelovacího rámce založeného na network calculus, který pomůže provést worst-case výkonnostní analýzu časového chování IP komunikačních sítí a jejich prvků určených pro budoucí použití v průmyslové automatizaci. V práci byla použita empirická analýza pro určení dominantních faktorů ovlivňujících časového chování síťových zařízení a identifikaci parametrů modelů těchto zařízení. Empirická analýza využívá nástroj TestQoS vyvinutý pro tyto účely. Byla navržena drobná rozšíření rámce network calculus, která byla nutná pro modelování časového chování používaných zařízení. Bylo vytvořeno několik typových modelů zařízení jako výsledek klasifikace různých architektur síťových zařízení a empiricky zjištěných dominantních faktorů. U modelovaných zařízení byla využita nová metoda identifikace parametrů. Práce je zakončena validací časových modelů dvou síťových zařízení (přepínače a směrovače) oproti empirickým pozorováním.With the growing scale of control systems and their distributed nature, communication networks have been gaining importance and new research challenges have been appearing. The major problem, contrary to previously used control systems with dedicated communication circuits, is time-varying delay of control and measurement signals introduced by packet-switched networks, such as Ethernet. The real-time issues in these networks have been tackled by proper adaptations. Nevertheless, market trend analyses foresee also future adoptions of IP-based communication networks in industrial automation for time-critical run-time data exchange. IP-based communication has only a limited support from the existing instrumentation in industrial automation. This challenge has recently been technically tackled within the Virtual Automation Networks (VAN) project by adopting the quality of service (QoS) architecture delivering soft-real-time communication behaviour. This dissertation focuses on the real-time performance aspects from the analytical point of view and provides means for applicability assessment of IP-based communication for future industrial applications. The main objective of this dissertation is establishment of a relevant modelling framework based on network calculus which will assist worst-case performance analysis of temporal behaviour of IP-based communication networks and networking devices intended for future use in industrial automation. Empirical analysis was used to identify dominant factors influencing the temporal performance of networking devices and for model parameter identification. The empirical analysis makes use of the TestQoS tool developed for this purpose. Minor extensions to the network calculus framework were proposed enabling to model the required temporal behaviour of networking devices. Several exemplary models were inferred as a result of classification of different networking device architectures and empirically identified dominant factors. A novel method for parameter identification was used with the modelled devices. Finally, two temporal models of networking devices (a switch and a router) were validated against empirical observations.

Support infrastructures for multimedia services with guaranteed continuity and QoS

Advances in wireless networking and content delivery systems are enabling new challenging provisioning scenarios where a growing number of users access multimedia services, e.g., audio/video streaming, while moving among different points of attachment to the Internet, possibly with different connectivity technologies, e.g., Wi-Fi, Bluetooth, and cellular 3G. That calls for novel middlewares capable of dynamically personalizing service provisioning to the characteristics of client environments, in particular to discontinuities in wireless resource availability due to handoffs. This dissertation proposes a novel middleware solution, called MUM, that performs effective and context-aware handoff management to transparently avoid service interruptions during both horizontal and vertical handoffs. To achieve the goal, MUM exploits the full visibility of wireless connections available in client localities and their handoff implementations (handoff awareness), of service quality requirements and handoff-related quality degradations (QoS awareness), and of network topology and resources available in current/future localities (location awareness). The design and implementation of the all main MUM components along with extensive on the field trials of the realized middleware architecture confirmed the validity of the proposed full context-aware handoff management approach. In particular, the reported experimental results demonstrate that MUM can effectively maintain service continuity for a wide range of different multimedia services by exploiting handoff prediction mechanisms, adaptive buffering and pre-fetching techniques, and proactive re-addressing/re-binding

A Slotted Ring Test Bed for the Study of ATM Network Congestion Management

This thesis addresses issues raised by the proposed Broadband Integrated Services Digital Network which will provide a flexible combination of integrated services traffic through its cell-based Asynchronbus Transport Mode (ATM). The introduction of a cell-based, connection-oriented, transport mode brings with it new technical challenges for network management. The routing of cells, their service at switching centres, and problems of cell congestion not encountered in the existing network, are some of the key issues. The thesis describes the development of a hardware slotted ring testbed for the investigation of congestion management in an ATM network. The testbed is designed to incorporate a modified form of the ORWELL protocol to control media access. The media access protocol is analysed to give a model for maximum throughput and reset interval under various traffic distributions. The results from the models are compared with measurements carried out on the testbed, where cell arrival statistics are also varied. It is shown that the maximum throughput of the testbed is dependent on both traffic distribution and cell arrival statistics. The testbed is used for investigations in a heterogeneous traffic environment where two classes of traffic with different cell arrival statistics and quality of service requirements are defined. The effect of prioritisation, media access protocol, traffic intensity, and traffic source statistics were investigated by determining an Admissible Load Region (ALR) for a network station. Conclusions drawn from this work suggest that there are many problems associated with the reliable definition of an ALR because of the number of variable parameters which could shift the ALR boundary. A suggested direction for further work is to explore bandwidth reservation and the concept of equivalent capacity of a connection, and how this can be linked to source control parameters

Congestion Control for Streaming Media

The Internet has assumed the role of the underlying communication network for applications such as file transfer, electronic mail, Web browsing and multimedia streaming. Multimedia streaming, in particular, is growing with the growth in power and connectivity of today\u27s computers. These Internet applications have a variety of network service requirements and traffic characteristics, which presents new challenges to the single best-effort service of today\u27s Internet. TCP, the de facto Internet transport protocol, has been successful in satisfying the needs of traditional Internet applications, but fails to satisfy the increasingly popular delay sensitive multimedia applications. Streaming applications often use UDP without a proper congestion avoidance mechanisms, threatening the well-being of the Internet. This dissertation presents an IP router traffic management mechanism, referred to as Crimson, that can be seamlessly deployed in the current Internet to protect well-behaving traffic from misbehaving traffic and support Quality of Service (QoS) requirements of delay sensitive multimedia applications as well as traditional Internet applications. In addition, as a means to enhance Internet support for multimedia streaming, this dissertation report presents design and evaluation of a TCP-Friendly and streaming-friendly transport protocol called the Multimedia Transport Protocol (MTP). Through a simulation study this report shows the Crimson network efficiently handles network congestion and minimizes queuing delay while providing affordable fairness protection from misbehaving flows over a wide range of traffic conditions. In addition, our results show that MTP offers streaming performance comparable to that provided by UDP, while doing so under a TCP-Friendly rate

Satellite Networks: Architectures, Applications, and Technologies

Since global satellite networks are moving to the forefront in enhancing the national and global information infrastructures due to communication satellites' unique networking characteristics, a workshop was organized to assess the progress made to date and chart the future. This workshop provided the forum to assess the current state-of-the-art, identify key issues, and highlight the emerging trends in the next-generation architectures, data protocol development, communication interoperability, and applications. Presentations on overview, state-of-the-art in research, development, deployment and applications and future trends on satellite networks are assembled

Layer-based coding, smoothing, and scheduling of low-bit-rate video for teleconferencing over tactical ATM networks

This work investigates issues related to distribution of low bit rate video within the context of a teleconferencing application deployed over a tactical ATM network. The main objective is to develop mechanisms that support transmission of low bit rate video streams as a series of scalable layers that progressively improve quality. The hierarchical nature of the layered video stream is actively exploited along the transmission path from the sender to the recipients to facilitate transmission. A new layered coder design tailored to video teleconferencing in the tactical environment is proposed. Macroblocks selected due to scene motion are layered via subband decomposition using the fast Haar transform. A generalized layering scheme groups the subbands to form an arbitrary number of layers. As a layering scheme suitable for low motion video is unsuitable for static slides, the coder adapts the layering scheme to the video content. A suboptimal rate control mechanism that reduces the kappa dimensional rate distortion problem resulting from the use of multiple quantizers tailored to each layer to a 1 dimensional problem by creating a single rate distortion curve for the coder in terms of a suboptimal set of kappa dimensional quantizer vectors is investigated. Rate control is thus simplified into a table lookup of a codebook containing the suboptimal quantizer vectors. The rate controller is ideal for real time video and limits fluctuations in the bit stream with no corresponding visible fluctuations in perceptual quality. A traffic smoother prior to network entry is developed to increase queuing and scheduler efficiency. Three levels of smoothing are studied: frame, layer, and cell interarrival. Frame level smoothing occurs via rate control at the application. Interleaving and cell interarrival smoothing are accomplished using a leaky bucket mechanism inserted prior to the adaptation layer or within the adaptation layerhttp://www.archive.org/details/layerbasedcoding00parkLieutenant Commander, United States NavyApproved for public release; distribution is unlimited