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1
Dogan, Safak. "Video transcoding for multimedia communication networks." Thesis, University of Surrey, 2001. http://epubs.surrey.ac.uk/843006/.
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Video transcoding is a generic name for a video gateway structure whereby the tandeming process does not involve any high complexity cascaded decoding and re-encoding operations in contrast to the existing conventional solutions. Diverse multimedia communication network characteristics, such as bandwidth limitations and varying congestion conditions, incur quality degradation in video transmissions. The matching of input and output network constraints and characteristics is possible with video transcoding at a centralised unit within the network. Moreover, video transcoding also provides a suitable translation mechanism for different video compression standards achieving a transparent interconnection between diverse network topologies. In addition, video transcoding offers a method for providing robustness against transmission error effects which occur during the transmission of compressed video streams over highly bandwidth- restricted communication media, such as popular mobile-wireless networks. Due to the severe bandwidth restrictions of such networks, the video signals require low bit rate coding which in turn renders video streams highly susceptible to radio channel errors. Therefore, error-resilient operations also need to be provided together with the syntax and transmission rate translation features of video transcoders. The unique features of video transcoding provide flexible and efficient ways to alleviate the previously addressed three major issues for various requirements. These requirements can be imposed by the diversity of networks on which numerous applications are running or by different standards themselves as well as by a wide range of users. This is strictly related to a universal interoperability issue of heterogeneous characteristics and requirements which demand effective end-to-end solutions. Thus, the aim of this research presented in this thesis is to develop algorithms for the provision of such remedial solutions to the interoperability problem. In the light of these facts, the research work focuses on the design of various video transcoding algorithms. The objectives of these algorithms are to ease network congestion and/or user bandwidth limitation conditions, support essential standard conversions between different compression schemes and provide necessary error robustness over highly error-prone transmission media, such as mobile radio networks. The ultimate target is to establish a common platform where all the above three aims are successfully satisfied. Extensive computer simulations demonstrate the effectiveness of the proposed and designed systems, throughout the course of the research work. These simulations are assessed with the use of objective and subjective performance measures.
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Nazir, Sajid. "Multimedia communication over mobile IP wireless networks." Thesis, University of Strathclyde, 2012. http://oleg.lib.strath.ac.uk:80/R/?func=dbin-jump-full&object_id=17816.
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The use of Internet Protocol (IP) based mobile wireless transmission is increasing as novel multimedia applications are being deployed. Mobile wireless channels and IP based communications are inherently prone to errors and packet losses. Error resilience features and Forward Error Correction (FEC) at the application layer (AL) are often used to protect the video data against losses. The amount of redundancy added by the FEC attempts to counter the worst channel Signal-to-noise-ratio (SNR) but the protection generally comes at a high complexity and overhead. It is thus imperative to design FEC solutions which are adaptive to the varying wireless channel conditions, i.e., bandwidth and packet loss rate. This adaptive behaviour becomes even more important for transmission to heterogeneous receivers. Fountain codes are rateless codes which can be used to potentially generate an unlimited number of encoded packets from a limited number of source packets. The decoding is possi ble if the number of received encoded packets at the receiver is just a little more than the source packets. As each portion of encoded video data does not have equal importance for the video re-construction, this characteristic can also be advantageously exploited while designing FEC solutions by providing more protection to important portions. Random linear codes (RLC) based schemes have been compared with Raptor codes, and RLC solution is proposed for the mobile television broadcasting standards like Digital Video Broadcasting-Handheld (DVB-H) and DVB-T2 (Second Generation Terrestrial). A reliable unicast video communication solution based on Luby Transform (LT) codes is proposed by exploiting unequal error protection (UEP) for encoded video data partitioned with the Data partitioning (DP) and slicing feature of H264/AVC. A comparison of layered video data transmission with Amplify-and-Forward (AF) and Decode-and-Forward (DF) relay collaboration strategies is provided. A novel scheme for multiple description coding (MDC) has been proposed and its advantages highlighted through simulations over relay based multi hop channels, like Long Term Evolution- Advanced (LTE-A). An algorithm has been proposed which takes into account the PSNR contribution and temporal significance of each slice to prioritize H.264/AVC sliced video data. The simulation results with systematic RLC show the usefulness of the proposed scheme for applications such as video-on-demand (VoD).
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Lee, Chung-Wei. "Altruistic QoS routing and multi-path multimedia communication." [Gainesville, Fla.] : University of Florida, 2001. http://etd.fcla.edu/etd/uf/2001/anp4085/dissertation6.pdf.
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Thesis (Ph. D.)--University of Florida, 2001.Title from first page of PDF file. Document formatted into pages; contains xi, 86 p.; also contains graphics. Vita. Includes bibliographical references (p. 82-85).
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Dai, Rui. "Correlation-based communication in wireless multimedia sensor networks." Diss., Georgia Institute of Technology, 2011. http://hdl.handle.net/1853/42736.
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Wireless multimedia sensor networks (WMSNs) are networks of interconnected devices that allow retrieving video and audio streams, still images, and scalar data from the environment. In a densely deployed WMSN, there exists correlation among the observations of camera sensors with overlapped coverage areas, which introduces substantial data redundancy in the network. In this dissertation, efficient communication schemes are designed for WMSNs by leveraging the correlation of visual information observed by camera sensors. First, a spatial correlation model is developed to estimate the correlation of visual information and the joint entropy of multiple correlated camera sensors. The compression performance of correlated visual information is then studied. An entropy-based divergence measure is proposed to predict the compression efficiency of performing joint coding on the images from correlated cameras. Based on the predicted compression efficiency, a clustered coding technique is proposed that maximizes the overall compression gain of the visual information gathered in WMSNs. The correlation of visual information is then utilized to design a network scheduling scheme to maximize the lifetime of WMSNs. Furthermore, as many WMSN applications require QoS support, a correlation-aware QoS routing algorithm is introduced that can efficiently deliver visual information under QoS constraints. Evaluation results show that, by utilizing the correlation of visual information in the communication process, the energy efficiency and networking performance of WMSNs could be improved significantly.
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Ganesh, Babu Thimma V. J. "Performance analysis of broadband multimedia wireless communication networks." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 2001. http://www.collectionscanada.ca/obj/s4/f2/dsk3/ftp04/NQ63989.pdf.
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Adas, Abdelnaser M. "Supporting multimedia traffic in broadband networks." Diss., Georgia Institute of Technology, 1997. http://hdl.handle.net/1853/14849.
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Kusetoğulları, Hüseyin. "Network routing optimisation and effective multimedia transmission to enhance QoS in communication networks." Thesis, University of Warwick, 2012. http://wrap.warwick.ac.uk/46802/.
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With the increased usage of communication services in networks, finding routes for reliable transmission and providing effective multimedia communication have become very challenging problems. This has been a strong motivation to examine and develop methods and techniques to find routing paths efficiently and to provide effective multimedia communication. This thesis is mainly concerned with designing, implementing and adapting intelligent algorithms to solve the computational complexity of network routing problems and testing the performance of intelligent algorithms’ applications. It also introduces hybrid algorithms which are developed by using the similarities of genetic algorithm (GA) and particle swarm optimization (PSO) intelligent systems algorithms. Furthermore, it examines the design of a new encoding/decoding method to offer a solution for the problem of unachievable multimedia information in multimedia multicast networks. The techniques presented and developed within the thesis aim to provide maximum utilization of network resources for handling communication problems. This thesis first proposes GA and PSO implementations which are adapted to solve the single and multi-objective functions in network routing problems. To offer solutions for network routing problems, binary variable-length and priority based encoding methods are used in intelligent algorithms to construct valid paths or potential solutions. The performance of generation operators in GA and PSO is examined and analyzed by solving the various shortest path routing problems and it is shown that the performance of algorithms varies based on the operators selected. Moreover, a hybrid algorithm is developed based on the lack of search capability of intelligent algorithms and implemented to solve the single objective function. The proposed method uses a strategy of sharing information between GA and PSO to achieve significant performance enhancement to solve routing optimization problems. The simulation results demonstrate the efficiency of the hybrid algorithm by optimizing the shortest path routing problem. Furthermore, intelligent algorithms are implemented to solve a multi-objective function which involves more constraints of resources in communication networks. The algorithms are adapted to find the multi-optimal paths to provide effective multimedia communication in lossy networks. The simulation results verify that the implemented algorithms are shown as efficient and accurate methods to solve the multi-objective function and find multi-optimal paths to deliver multimedia packets in lossy networks. Furthermore, the thesis proposes a new encoding/decoding method to maximize throughput in multimedia multicast networks. The proposed method is combined with two most used Multiple Description Coding (MDC) methods. The utilization of the proposed method is discussed by comparing two the MDC methods. Through analyzing the simulation results using these intelligent systems algorithms, it has been shown that feasible solutions can be obtained by optimizing complex network problems. Moreover, the methods proposed and developed, which are hybrid algorithms and the encoding/decoding method also demonstrate their efficiency and effectiveness as compared with other techniques.
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Jiyapanichkul, Prasit, and jiyapanichkul@yahoo com. "Resource management in broadband multimedia networks." Swinburne University of Technology. Laboratory for Telecommunication Research, 1999. http://adt.lib.swin.edu.au./public/adt-VSWT20050610.100950.
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This research deals with optimal resource management in an overloaded broadband
multimedia network. Optimisation is with respect to user satisfaction, where user
satisfaction reflects both the quality of service experienced by connected users and the
dissatisfaction of users blocked from access to the network.
The research focuses on Asynchronous Transfer Mode (ATM) networks and the Internet, because these are the dominant emerging broadband networks which present some fundamental unsolved problems, related to the sharing of resources between mixed traffic types. ATM networks use conservative admission control, which protects
network resources and ensures a high level of service for those admitted to the network, but results in low network efficiency because of low utilisation of resources due to blocking of many potential users. The Internet does not use admission control, with the result that performance degrades progressively as load increases. This causes frustration among users, and lowers the network efficiency due to high levels of congestion.
We propose an optimisation model for each network (ATM networks and the Internet)which is intended to represent the distribution and consumption of key network resources by different traffic types. The model is aimed at maximising performance such that users admitted to the network are offered no less than some minimum acceptable level of quality of service (QoS). The solution is a set of traffic flow rates on each path which results in maximising an objective function value (revenue based on network operator interest or throughput based on customer interest) for a given network
configuration with given user demand. As an example using the ATM network model, we illustrate the application of the model to an ATM network carrying both connection oriented and connectionless traffic. We explore the optimal response to a link failure which in turn causes node overload. As an example using the Internet model, we consider an overloaded network with link bottlenecks and an overloaded Web server, and explore the effect of transferring some server capacity to a mirror site and a proxy server.
For real-time traffic control, the optimisation model is used to assign quotas for bandwidth or connections to selected paths. A control algorithm is implemented to provide maximum performance by admitting requests within the quotas which are obtained from the optimisation model. In an ATM network simulation, the algorithm is used to manage the virtual path (VP) pool in a network which suffers a link failure. A comparison is made between fixed virtual path management (FVPM) and dynamic
virtual path management (DVPM), comparing the revenue achieved by each. This illustrates how DVPM adapts the VP pool in a robust fashion to achieve maximum revenue in the face of a link failure. However, the transient response suggests that benefit could be obtained using non-steady-state solutions. The model is extended by taking network state and traffic parameters into account to control changes in the VP pool to recognise limits to the rate at which traffic can be moved (through the natural
birth-death processes). This scheme is called state dependent virtual path management (SDVPM). Performance evaluation of the new model shows that SDVPM achieves higher revenue than DVPM when the network suffers a link failure that requires a major change to the VP pool. In an Internet simulation, two algorithms are compared for control of access to a proxy server and a set of primary servers. An algorithm based on
optimal flow solutions provides substantially better network performance than a localised heuristic algorithm. In each simulation case (ATM and Internet examples), the performance using a control system based on the steady state optimum flow model is close to the ideal optimal result.
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Cellatoglu, A. "Adaptive header compression techniques for mobile multimedia networks." Thesis, University of Surrey, 2003. http://epubs.surrey.ac.uk/800046/.
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Mertzanis, Ioannis. "QOS provisioning for broadband satellite-ATM multimedia networks." Thesis, University of Surrey, 1999. http://epubs.surrey.ac.uk/773030/.
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This work is motivated by the current trends in future broadband communication networks. On the one hand, the latest developments and experimentation with the Asynchronous Transfer mode (ATM) technology shows that that ATM is going to be the future transport mechanism in many private and public networks. This is driven by the need to efficiently support a large population of widebandlbroadband users with different data traffic characteristics and certain Quality of Service (QoS) guarantees. On the other hand, the interest in satellites and their role in the future broadband multimedia communications systems, has grown considerably since they can very quickly and economically extend the boundaries of the terrestrial mobile and fixed networks coverage. On-board satellite signal regeneration and' ATM-like' switching is part of the latest experimental developments by many payload manufacturers. In this thesis, the focus is on the Grade of Service (GoS) and QoS provisioning in future broadband satellite multimedia systems by introducing new means for their performance evaluation. The investigation includes modelling techniques for both Geostationary (GEO) and non-GEO systems. An extensive set of representative results derived analytically and by simulation are presented assuming different mixed traffic scenarios. A new methodology for the Available Bit Rate (ABR) service class capacity estimation and the CAC strategy that needs to be adopted in S-ATM systems is developed. Moreover, the performance evaluation of bandwidth reservation techniques for non-GEO satellite constellations is investigated and new rules for maintaining specified GoS performance are proposed. This work contributes towards the definition of a satellite network infrastructure that best satisfies the requirements of an integrated solution with Broadband-Integrated Services Digital Network (B-ISDN).
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Melodia, Tommaso. "Communication and coordination in wireless multimedia sensor and actor networks." Diss., Available online, Georgia Institute of Technology, 2007, 2007. http://etd.gatech.edu/theses/available/etd-07032007-014958/.
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Thesis (Ph. D.)--Electrical and Computer Engineering, Georgia Institute of Technology, 2008.Fujimoto, Richard, Committee Member ; Ma, Xiaoli, Committee Member ; Fekri, Faramarz, Committee Member ; Copeland, John, Committee Member ; Akyildiz, Ian, Committee Chair.
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Kim, Sungwook Varshney Pramod K. "Adaptive bandwidth management for QoS sensitive multimedia cellular/communication networks." Related Electronic Resource: Current Research at SU : database of SU dissertations, recent titles available full text, 2003. http://wwwlib.umi.com/cr/syr/main.
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Zheng, Hai. "QoS concerned efficient video communications over wireless networks." Diss., Georgia Institute of Technology, 2000. http://hdl.handle.net/1853/13869.
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Yerima, Suleiman Y. "Quality of service optimization of multimedia traffic in mobile networks." Thesis, University of South Wales, 2009. https://pure.southwales.ac.uk/en/studentthesis/quality-of-service-optimization-of-multimedia-traffic-in-mobile-networks(975989e3-30f0-450b-9c6f-c2c51362f380).html.
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Mobile communication systems have continued to evolve beyond the currently deployed Third Generation (3G) systems with the main goal of providing higher capacity. Systems beyond 3G are expected to cater for a wide variety of services such as speech, data, image transmission, video, as well as multimedia services consisting of a combination of these. With the air interface being the bottleneck in mobile networks, recent enhancing technologies such as the High Speed Downlink Packet Access (HSDPA), incorporate major changes to the radio access segment of 3G Universal Mobile Telecommunications System (UMTS). HSDPA introduces new features such as fast link adaptation mechanisms, fast packet scheduling, and physical layer retransmissions in the base stations, necessitating buffering of data at the air interface which presents a bottleneck to end-to-end communication. Hence, in order to provide end-to-end Quality of Service (QoS) guarantees to multimedia services in wireless networks such as HSDPA, efficient buffer management schemes are required at the air interface. The main objective of this thesis is to propose and evaluate solutions that will address the QoS optimization of multimedia traffic at the radio link interface of HSDPA systems. In the thesis, a novel queuing system known as the Time-Space Priority (TSP) scheme is proposed for multimedia traffic QoS control. TSP provides customized preferential treatment to the constituent flows in the multimedia traffic to suit their diverse QoS requirements. With TSP queuing, the real-time component of the multimedia traffic, being delay sensitive and loss tolerant, is given transmission priority; while the non-real-time component, being loss sensitive and delay tolerant, enjoys space priority. Hence, based on the TSP queuing paradigm, new buffer managementalgorithms are designed for joint QoS control of the diverse components in a multimedia session of the same HSDPA user. In the thesis, a TSP based buffer management algorithm known as the Enhanced Time Space Priority (E-TSP) is proposed for HSDPA. E-TSP incorporates flow control mechanisms to mitigate congestion in the air interface buffer of a user with multimedia session comprising real-time and non-real-time flows. Thus, E-TSP is designed to provide efficient network and radio resource utilization to improve end-to-end multimedia traffic performance. In order to allow real-time optimization of the QoS control between the real-time and non-real-time flows of the HSDPA multimedia session, another TSP based buffer management algorithm known as the Dynamic Time Space Priority (D-TSP) is proposed. D-TSP incorporates dynamic priority switching between the real-time and non-real-time flows. D-TSP is designed to allow optimum QoS trade-off between the flows whilst still guaranteeing the stringent real-time component’s QoS requirements. The thesis presents results of extensive performance studies undertaken via analytical modelling and dynamic network-level HSDPA simulations demonstrating the effectiveness of the proposed TSP queuing system and the TSP based buffer management schemes.
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Mander, Ranjeet Singh. "Integrated multiple access protocols for personal/mobile communication networks supporting multimedia applications." Thesis, University of Ottawa (Canada), 1997. http://hdl.handle.net/10393/4257.
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The success of personal communication networks will depend on their ability to accommodate the diverse traffic which will be generated from the numerous and diverse future multimedia applications and services. Towards this end, multimedia Medium access control (MAC) algorithms capable of accommodating this integrated traffic will play a major role. In this thesis, novel MAC protocols are proposed and their ability to efficiently utilize the available bandwidth is evaluated. The proposed multimedia MAC protocols are based on random access algorithms and reservation policies using TDM and CDMA technologies. The MAC protocols also support multiple-priority mechanisms: an essential feature in communication systems supporting integrated services with diverse Quality of Service (QoS) requirements. One of the major characteristics of the priority mechanism is its simplicity of implementation. This adds to the practical value of the proposed MAC protocols. The TDMA and the CDMA versions of the protocols are evaluated for voice and data traffic and compared to other MACs proposed in the literature. The CDMA version of the protocol is also investigated for data, voice and video with multi-slot reservation for video. Network configurations servicing CBR (variable quality) and VBR (constant quality) video users are evaluated. The results verified the superiority of the proposed protocol: high school utilization and ability to service effectively high volumes of integrated traffic. The new MACs are found to be stable, robust and easy to implement multimedia services in personal communication networks. The simplicity and high performance level of the proposed MACs makes it valuable for supporting integrated traffic wireless applications.
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Wang, Xiaoxiao Sherry. "Qos management for video delivery over mobile wireless networks." Diss., Georgia Institute of Technology, 2000. http://hdl.handle.net/1853/13856.
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Argyriou, Antonios D. "Transport Layer Optimizations for Heterogeneous Wireless Multimedia Networks." Diss., Georgia Institute of Technology, 2005. http://hdl.handle.net/1853/7466.
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The explosive growth of the Internet during the last few years, has been propelled by the TCP/IP protocol suite and the best effort packet forwarding service. However, quality of service (QoS) is far from being a reality especially for multimedia services like video streaming and video conferencing. In the case of wireless and mobile networks, the problem becomes even worse due to the physics of the medium, resulting into further deterioration of the system performance.
Goal of this dissertation is the systematic development of comprehensive models that jointly characterize the performance of transport protocols and media delivery in heterogeneous wireless networks. At the core of our novel methodology, is the use of analytical models for driving the design of media transport algorithms, so that the delivery of conversational and non-interactive multimedia data is enhanced in terms of throughput, delay, and jitter. More speciffically, we develop analytical models that characterize the throughput and goodput of the transmission control protocol (TCP) and the transmission friendly rate control (TFRC) protocol, when CBR and VBR multimedia workloads are considered. Subsequently, we enhance the transport protocol models with new parameters that capture the playback buffer performance and the expected video distortion at the receiver. In this way a complete end-to-end model for media streaming is obtained. This model is used as a basis for a new algorithm for rate-distortion optimized mode selection in video streaming appli-
cations. As a next step, we extend the developed models for the aforementioned protocols, so that heterogeneous wireless networks can be accommodated. Subsequently, new algorithms are proposed in order to enhance the developed media streaming algorithms when heterogeneous wireless networks are also included. Finally, the aforementioned models and algorithms are extended for the case of concurrent multipath media transport over several hybrid wired/wireless links.
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Yu, Sam Shaokai. "Performance analysis and call control procedures in high-speed multimedia personal wireless communications /." Title page, abstract and contents only, 1999. http://web4.library.adelaide.edu.au/theses/09PH/09phy936.pdf.
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Chavan, Rohit. "JAVA synchronized collaborative multimedia toolkit: A collaborative communication tool." CSUSB ScholarWorks, 2004. https://scholarworks.lib.csusb.edu/etd-project/2549.
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In this project a collaboration multimedia toolkit, JSCMT (Java Synchronized Collaborative Multimedia Toolkit) was developed which is intended to connect a group of people located in different geographical locations who are working on the same project.
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Dua, Aditya. "Packet scheduling and congestion control in communication networks : applications to multimedia streaming and switching /." May be available electronically:, 2007. http://proquest.umi.com/login?COPT=REJTPTU1MTUmSU5UPTAmVkVSPTI=&clientId=12498.
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Levine, David A. "Resource allocation, call admission, and media access control protocols for wireless multimedia networks." Diss., Georgia Institute of Technology, 1996. http://hdl.handle.net/1853/14766.
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Fredholm, Kenth, and Kristian Nilsson. "Implementing an application for communication and quality measurements over UMTS networks." Thesis, Linköping University, Department of Electrical Engineering, 2003. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-1666.
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The interest for various multimedia services accessed via the Internet has been growing immensely along with the bandwidth available. A similar development has emerged in the 3G mobile network. The focus of this master thesis is on the speech/audio part of a 3G multimedia application. The purpose has been to implement a traffic generating tool that can measure QoS (Quality of Service) in 3G networks. The application is compliant to the 3G standards, i.e. it uses AMR (Adaptive Multi Rate), SIP (Session Initiation Protocol) and RTP (Real Time Transport Protocol). AMR is a speech compression algorithm with the special feature that it can compress speech into several different bitrates. SIP signalling is used so that different applications can agree on how to communicate. RTP carries the speech frames over the network, in order to provide features that are necessary for media/multimedia applications. Issues like perception of audio and QoS related parameters is also discussed, from the perspective of users and developers.
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Dash, Trivikram. "Performance Analysis of Wireless Networks with QoS Adaptations." Thesis, University of North Texas, 2003. https://digital.library.unt.edu/ark:/67531/metadc4336/.
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The explosive demand for multimedia and fast transmission of continuous media on wireless networks means the simultaneous existence of traffic requiring different qualities of service (QoS). In this thesis, several efficient algorithms have been developed which offer several QoS to the end-user. We first look at a request TDMA/CDMA protocol for supporting wireless multimedia traffic, where CDMA is laid over TDMA. Then we look at a hybrid push-pull algorithm for wireless networks, and present a generalized performance analysis of the proposed protocol. Some of the QoS factors considered include customer retrial rates due to user impatience and system timeouts and different levels of priority and weights for mobile hosts. We have also looked at how customer impatience and system timeouts affect the QoS provided by several queuing and scheduling schemes such as FIFO, priority, weighted fair queuing, and the application of the stretch-optimal algorithm to scheduling.
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Hong, SungBum. "Solutions for Dynamic Channel Assignment and Synchronization Problem for Distributed Wireless Multimedia System." Thesis, University of North Texas, 2002. https://digital.library.unt.edu/ark:/67531/metadc3249/.
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The recent advances in mobile computing and distributed multimedia systems allow mobile hosts (clients) to access wireless multimedia Data at anywhere and at anytime. In accessing multimedia information on the distributed multimedia servers from wireless personal communication service systems, a channel assignment problem and synchronization problems should be solved efficiently. Recent demand for mobile telephone service have been growing rapidly while the electro-magnetic spectrum of frequencies allocated for this purpose remain limited. Any solution to the channel assignment problem is subject to this limitation, as well as the interference constraint between adjacent channels in the spectrum. Channel allocation schemes provide a flexible and efficient access to bandwidth in wireless and mobile communication systems. In this dissertation, both an efficient distributed algorithm for dynamic channel allocation based upon mutual exclusion model, and an efficient distributed synchronization algorithm using Quasi-sink for wireless and mobile multimedia systems to ensure and facilitate mobile client access to multimedia objects are proposed. Algorithm's performance with several channel systems using different types of call arrival patterns is determined analytically. A set of simulation experiments to evaluate the performance of our scheme using message complexity and buffer usage at each frame arrival time.
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Sekercioglu, Ahmet, and ahmet@hyperion ctie monash edu au. "Fuzzy logic control techniques and structures for Asynchronous Transfer Mode (ATM) based multimedia networks." Swinburne University of Technology, 1999. http://adt.lib.swin.edu.au./public/adt-VSWT20050411.130014.
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The research presented in this thesis aims to demonstrate that fuzzy logic is a useful tool for developing mechanisms for controlling traffc flow in ATM based multimedia networks to maintain quality of service (QoS) requirements and maximize resource utilization. The study first proposes a hierarchical, multilevel control
structure for ATM networks to exploit the reported strengths of fuzzy logic at various
control levels. Then, an extensive development and evaluation is presented for a subset of the proposed control architecture at the congestion control level.
An ATM based multimedia network must have quite sophisticated traffc control capabilities to effectively handle the requirements of a dynamically varying mixture of voice, video and data services while meeting the required levels of performance.
Feedback control techniques have an essential role for the effective and efficient management of the resources of ATM networks. However, development of conventional feedback control techniques relies on the availability of analytical system models. The characteristics of ATM networks and the complexity of service
requirements cause the analytical modeling to be very difficult, if not impossible.
The lack of realistic dynamic explicit models leads to substantial problems in developing
control solutions for B-ISDN networks. This limits the ability of conventional techniques to directly address the control objectives for ATM networks. In the literature, several connection admission and congestion control methods for B-ISDN networks have been reported, and these have achieved mixed success. Usually they either assume heavily simplified models, or they are too complicated to implement, mainly derived using probabilistic (steady-state) models.
Fuzzy logic controllers, on the other hand, have been applied successfully to the task of controlling systems for which analytical models are not easily obtainable.
Fuzzy logic control is a knowledge-based control strategy that can be utilized when an explicit model of a system is not available or, the model itself, if available, is highly complex and nonlinear. In this case, the problem of control system design is based on qualitative and/or empirically acquired knowledge regarding the operation of the system. Representation of qualitative or empirically acquired knowledge in a fuzzy logic controller is achieved by linguistic expressions in the form of fuzzy relational equations. By using fuzzy relational equations, classifications related to system parameters can be derived without explicit description.
The thesis presents a new predictive congestion control scheme, Fuzzy Explicit Rate Marking (FERM), which aims to avoid congestion, and by doing so minimize the cell losses, attain high server utilization, and maintain the fair use of links.
The performance of the FERM scheme is extremely competitive with that of control schemes developed using traditional methods over a considerable period of time.
The results of the study demonstrate that fuzzy logic control is a highly effective design tool for this type of problems, relative to the traditional methods. When controlled systems are highly nonlinear and complex, it keeps the human insight alive and accessible at the lower levels of the control hierarchy, and so higher levels
can be built on this understanding.
Additionally, the FERM scheme has been extended
to adaptively tune (A-FERM) so that continuous automatic tuning of the parameters can be achieved, and thus be more adaptive to system changes leading to better utilization of network bandwidth. This achieves a level of robustness that is not exhibited by other congestion control schemes reported in the literature.
In this work, the focus is on ATM networks rather than IP based networks. For historical reasons, and due to fundamental philosophical differences in the (earlier) approach to congestion control, the research for control of TCP/IP and ATM based networks proceeded separately. However, some convergence between them has recently become evident. In the TCP/IP literature proposals have appeared on active queue management in routers, and Explicit Congestion Notication (ECN) for IP. It is reasonably expected that, the algorithms developed in this study will be applicable to IP based multimedia networks as well.
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Díaz, Santos Juan Ramón. "Design and Implementation of a Communication Protocol to Improve Multimedia QoS and QoE in Wireless Ad Hoc Networks." Doctoral thesis, Universitat Politècnica de València, 2016. http://hdl.handle.net/10251/62162.
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[EN] This dissertation addresses the problem of multimedia delivery over multi-hop ad hoc wireless networks, and especially over wireless sensor networks. Due to their characteristics of low power consumption, low processing capacity and low memory capacity, they have major difficulties in achieving optimal quality levels demanded by end users in such communications.
In the first part of this work, it has been carried out a study to determine the behavior of a variety of multimedia streams and how they are affected by the network conditions when they are transmitted over topologies formed by devices of different technologies in multi hop wireless ad hoc mode. To achieve this goal, we have performed experimental tests using a test bench, which combine the main codecs used in audio and video streaming over IP networks with different sound and video captures representing the characteristic patterns of multimedia services such as phone calls, video communications, IPTV and video on demand (VOD). With the information gathered in the laboratory, we have been able to establish the correlation between the induced changes in the physical and logical topology and the network parameters that measure the quality of service (QoS) of a multimedia transmission, such as latency, jitter or packet loss. At this stage of the investigation, a study was performed to determine the state of the art of the proposed protocols, algorithms, and practical implementations that have been explicitly developed to optimize the multimedia transmission over wireless ad hoc networks, especially in ad hoc networks using clusters of nodes distributed over a geographic area and wireless sensor networks.
Next step of this research was the development of an algorithm focused on the logical organization of clusters formed by nodes capable of adapting to the circumstances of real-time traffic. The stated goal was to achieve the maximum utilization of the resources offered by the set of nodes that forms the network, allowing simultaneously sending reliably and efficiently all types of content through them, and mixing conventional IP data traffic with multimedia traffic with stringent QoS and QoE requirements. Using the information gathered in the previous phase, we have developed a network architecture that improves overall network performance and multimedia streaming. In parallel, it has been designed and programmed a communication protocol that allows implementing the proposal and testing its operation on real network infrastructures.
In the last phase of this thesis we have focused our work on sending multimedia in wireless sensor networks (WSN). Based on the above results, we have adapted both the architecture and the communication protocol for this particular type of network, whose use has been growing hugely in recent years.[ES] Esta tesis doctoral aborda el problema de la distribución de contenidos multimedia a través de redes inalámbricas ad hoc multisalto, especialmente las redes inalámbricas de sensores que, debido a sus características de bajo consumo energético, baja capacidad de procesamiento y baja capacidad de memoria, plantean grandes dificultades para alcanzar los niveles de calidad óptimos que exigen los usuarios finales en dicho tipo de comunicaciones. En la primera parte de este trabajo se ha llevado a cabo un estudio para determinar el comportamiento de una gran variedad de flujos multimedia y como se ven afectados por las condiciones de la red cuando son transmitidos a través topologías formadas por dispositivos de diferentes tecnologías que se comunican en modo ad hoc multisalto inalámbrico. Para ello, se han realizado pruebas experimentales sobre una maqueta de laboratorio, combinando los principales códecs empleados en la transmisión de audio y video a través de redes IP con diversas capturas de sonido y video que representan patrones característicos de servicios multimedia tales como las llamadas telefónicas, videoconferencias, IPTV o video bajo demanda (VOD). Con la información reunida en el laboratorio se ha podido establecer la correlación entre los cambios inducidos en la topología física y lógica de la red con los parámetros que miden la calidad de servicio (QoS) de una transmisión multimedia, tales como la latencia el jitter o la pérdida de paquetes. En esta fase de la investigación se realiza un estudio para determinar el estado del arte de las propuestas de desarrollo e implementación de protocolos y algoritmos que se han generado de forma explícita para optimizar la transmisión de tráfico multimedia sobre redes ad hoc inalámbricas, especialmente en las redes inalámbricas de sensores y redes ad hoc utilizando clústeres de nodos distribuidos en un espacio geográfico. El siguiente paso en la investigación ha consistido en el desarrollo de un algoritmo propio para la organización lógica de clústeres formados por nodos capaces de adaptarse a las circunstancias del tráfico en tiempo real. El objetivo planteado es conseguir un aprovechamiento máximo de los recursos ofrecidos por el conjunto de nodos que forman la red, permitiendo de forma simultánea el envío de todo tipo de contenidos a través de ellos de forma confiable y eficiente, permitiendo la convivencia de tráfico de datos IP convencional con tráfico multimedia con requisitos exigentes de QoS y QoE. A partir de la información conseguida en la fase anterior, se ha desarrollado una arquitectura de red que mejora el rendimiento general de la red y el de las transmisiones multimedia de audio y video en particular. De forma paralela, se ha diseñado y programado un protocolo de comunicación que permite implementar el modelo y testear su funcionamiento sobre infraestructuras de red reales. En la última fase de esta tesis se ha dirigido la atención hacia la transmisión multimedia en las redes de sensores inalámbricos (WSN). Partiendo de los resultados anteriores, se ha adaptado tanto la arquitectura como el protocolo de comunicaciones para este tipo concreto de red, cuyo uso se ha extendido en los últimos años de forma considerable
[CAT] Esta tesi doctoral aborda el problema de la distribució de continguts multimèdia a través de xarxes sense fil ad hoc multi salt, especialment les xarxes sense fil de sensors que, a causa de les seues característiques de baix consum energètic, baixa capacitat de processament i baixa capacitat de memòria, plantegen grans dificultats per a aconseguir els nivells de qualitat òptims que exigixen els usuaris finals en eixos tipus de comunicacions. En la primera part d'este treball s'ha dut a terme un estudi per a determinar el comportament d'una gran varietat de fluxos multimèdia i com es veuen afectats per les condicions de la xarxa quan són transmesos a través topologies formades per dispositius de diferents tecnologies que es comuniquen en mode ad hoc multi salt sense fil. Per a això, s'han realitzat proves experimentals sobre una maqueta de laboratori, combinant els principals códecs empleats en la transmissió d'àudio i vídeo a través de xarxes IP amb diverses captures de so i vídeo que representen patrons característics de serveis multimèdia com son les cridades telefòniques, videoconferències, IPTV o vídeo baix demanda (VOD). Amb la informació reunida en el laboratori s'ha pogut establir la correlació entre els canvis induïts en la topologia física i lògica de la xarxa amb els paràmetres que mesuren la qualitat de servei (QoS) d'una transmissió multimèdia, com la latència el jitter o la pèrdua de paquets. En esta fase de la investigació es realitza un estudi per a determinar l'estat de l'art de les propostes de desenvolupament i implementació de protocols i algoritmes que s'han generat de forma explícita per a optimitzar la transmissió de tràfic multimèdia sobre xarxes ad hoc sense fil, especialment en les xarxes sense fil de sensors and xarxes ad hoc utilitzant clusters de nodes distribuïts en un espai geogràfic. El següent pas en la investigació ha consistit en el desenvolupament d'un algoritme propi per a l'organització lògica de clusters formats per nodes capaços d'adaptar-se a les circumstàncies del tràfic en temps real. L'objectiu plantejat és aconseguir un aprofitament màxim dels recursos oferits pel conjunt de nodes que formen la xarxa, permetent de forma simultània l'enviament de qualsevol tipus de continguts a través d'ells de forma confiable i eficient, permetent la convivència de tràfic de dades IP convencional amb tràfic multimèdia amb requisits exigents de QoS i QoE. A partir de la informació aconseguida en la fase anterior, s'ha desenvolupat una arquitectura de xarxa que millora el rendiment general de la xarxa i el de les transmissions multimèdia d'àudio i vídeo en particular. De forma paral¿lela, s'ha dissenyat i programat un protocol de comunicació que permet implementar el model i testejar el seu funcionament sobre infraestructures de xarxa reals. En l'última fase d'esta tesi s'ha dirigit l'atenció cap a la transmissió multimèdia en les xarxes de sensors sense fil (WSN). Partint dels resultats anteriors, s'ha adaptat tant l'arquitectura com el protocol de comunicacions per a aquest tipus concret de xarxa, l'ús del qual s'ha estés en els últims anys de forma considerable.
Díaz Santos, JR. (2016). Design and Implementation of a Communication Protocol to Improve Multimedia QoS and QoE in Wireless Ad Hoc Networks [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/62162
TESIS
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Paliwal, Gaurav. "Convergence : the next big step /." Online version of thesis, 2006. https://ritdml.rit.edu/dspace/handle/1850/1316.
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Mustafa, Elmabrook B. M. "Some new localized quality of service models and algorithms for communication networks : the development and evaluation of new localized quality of service routing algorithms and path selection methods for both flat and hierarchical communication networks." Thesis, University of Bradford, 2009. http://hdl.handle.net/10454/4288.
Full textAbstract:
The Quality of Service (QoS) routing approach is gaining an increasing interest in the Internet community due to the new emerging Internet applications such as real-time multimedia applications. These applications require better levels of quality of services than those supported by best effort networks. Therefore providing such services is crucial to many real time and multimedia applications which have strict quality of service requirements regarding bandwidth and timeliness of delivery. QoS routing is a major component in any QoS architecture and thus has been studied extensively in the literature. Scalability is considered one of the major issues in designing efficient QoS routing algorithms due to the high cost of QoS routing both in terms of computational effort and communication overhead. Localized quality of service routing is a promising approach to overcome the scalability problem of the conventional quality of service routing approach. The localized quality of service approach eliminates the communication overhead because it does not need the global network state information. The main aim of this thesis is to contribute towards the localised routing area by proposing and developing some new models and algorithms. Toward this goal we make the following major contributions. First, a scalable and efficient QoS routing algorithm based on a localised approach to QoS routing has been developed and evaluated. Second, we have developed a path selection technique that can be used with existing localized QoS routing algorithms to enhance their scalability and performance. Third, a scalable and efficient hierarchical QoS routing algorithm based on a localised approach to QoS routing has been developed and evaluated.
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Haddd, Rami J. "Feed-Forward Bandwidth Indication: An Accurate Approach to Multimedia Bandwidth Forecasting and its Application in Ethernet Passive Optical Networks." University of Akron / OhioLINK, 2011. http://rave.ohiolink.edu/etdc/view?acc_num=akron1312558599.
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Choi, Jee W. "Reducing Communication Through Buffers on a SIMD Architecture." Thesis, Georgia Institute of Technology, 2004. http://hdl.handle.net/1853/4970.
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Advances in wireless technology and the growing popularity of multimedia applications have brought about a need for energy efficient and cost effective portable supercomputers capable of delivering performance beyond the capabilities of current microprocessors and DSP chips. The SIMPil architecture currently being developed at Georgia Institute of Technology is a promising candidate for this task. In order to develop applications for SIMPil, a high level language and an optimizing compiler for the language are essential. However, with the recent trend of interconnect latency becoming a major bottleneck on computer systems, optimizations focusing on reducing latency are becoming more important, especially with SIMPil, as it is highly scalable. The compiler tracks the path of data through the network and buffers data in each processor to eliminate redundant communication. With a buffer size of 5, the compiler was able to eliminate 96 percent of the redundant communication for a 9x9 convolution and 8x8 DCT algorithms. With 5x5 convolution, only 89 percent elimination was observed. In terms of performance, 106 percent speedup was observed with 9x9 convolution at buffer size of 5 while 5x5 convolution and 8x8 DCT which have a much lower number of communication showed only 101 percent speedup.
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Pillai, Anju. "A Connection Admission Control Framework for UMTS based Satellite Systems.An Adaptive Admission Control algorithm with pre-emption control mechanism for unicast and multicast communications in satellite UMTS." Thesis, University of Bradford, 2011. http://hdl.handle.net/10454/5487.
Full textAbstract:
In recent years, there has been an exponential growth in the use of
multimedia applications. A satellite system offers great potential for
multimedia applications with its ability to broadcast and multicast a large
amount of data over a very large area as compared to a terrestrial system.
However, the limited transmission capacity along with the dynamically
varying channel conditions impedes the delivery of good quality multimedia
service in a satellite system which has resulted in research efforts for deriving
efficient radio resource management techniques. This issue is addressed in
this thesis, where the main emphasis is to design a CAC framework which
maximizes the utilization of the scarce radio resources available in the
satellite and at the same time increases the performance of the system for a
UMTS based satellite system supporting unicast and multicast traffic.
The design of the system architecture for a UMTS based satellite system is
presented. Based on this architecture, a CAC framework is designed
consisting of three different functionalities: the admission control procedure,
the retune procedure and the pre-emption procedure. The joint use of these
functionalities is proposed to allow the performance of the system to be
maintained under congestion. Different algorithms are proposed for different
functionalities; an adaptive admission control algorithm, a greedy retune
algorithm and three pre-emption algorithms (Greedy, SubSetSum, and
Fuzzy).
A MATLAB simulation model is developed to study the performance of the
proposed CAC framework. A GUI is created to provide the user with the
flexibility to configure the system settings before starting a simulation. The
configuration settings allow the system to be analysed under different
conditions.
The performance of the system is measured under different simulation
settings such as enabling and disabling of the two functionalities of the CAC
framework; retune procedure and the pre-emption procedure. The simulation
results indicate the CAC framework as a whole with all the functionalities
performs better than the other simulation settings.
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Pillai, Anju. "A connection admission control framework for UMTS based satellite systems : an adaptive admission control algorithm with pre-emption control mechanism for unicast and multicast communications in satellite UMTS." Thesis, University of Bradford, 2011. http://hdl.handle.net/10454/5487.
Full textAbstract:
In recent years, there has been an exponential growth in the use of multimedia applications. A satellite system offers great potential for multimedia applications with its ability to broadcast and multicast a large amount of data over a very large area as compared to a terrestrial system. However, the limited transmission capacity along with the dynamically varying channel conditions impedes the delivery of good quality multimedia service in a satellite system which has resulted in research efforts for deriving efficient radio resource management techniques. This issue is addressed in this thesis, where the main emphasis is to design a CAC framework which maximizes the utilization of the scarce radio resources available in the satellite and at the same time increases the performance of the system for a UMTS based satellite system supporting unicast and multicast traffic. The design of the system architecture for a UMTS based satellite system is presented. Based on this architecture, a CAC framework is designed consisting of three different functionalities: the admission control procedure, the retune procedure and the pre-emption procedure. The joint use of these functionalities is proposed to allow the performance of the system to be maintained under congestion. Different algorithms are proposed for different functionalities; an adaptive admission control algorithm, a greedy retune algorithm and three pre-emption algorithms (Greedy, SubSetSum, and Fuzzy). A MATLAB simulation model is developed to study the performance of the proposed CAC framework. A GUI is created to provide the user with the flexibility to configure the system settings before starting a simulation. The configuration settings allow the system to be analysed under different conditions. The performance of the system is measured under different simulation settings such as enabling and disabling of the two functionalities of the CAC framework; retune procedure and the pre-emption procedure. The simulation results indicate the CAC framework as a whole with all the functionalities performs better than the other simulation settings.
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Zhang, Guoqiang. "Robust Multimedia Communications over Packet Networks." Doctoral thesis, KTH, Ljud- och bildbehandling, 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-24223.
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Multimedia communications over packet networks, and in particular the voice over IP (VoIP) application, have become an integral part of society. However, the unreliable and heterogeneous nature of packet networks has led to a best-effort delivery of services. Delay, limitation of bandwidth, and packet-loss rate all affect the quality of service (QoS). In this thesis, we address two important network impairments in the design of robust multimedia communication systems: packet delay-variation and packet-loss. Paper A considers the mitigation of the effect of packet delay-variation for audio communications by introducing a buffer at the receiver side. A new adaptive playout scheduling approach is proposed to control the buffering length, or, equivalently, the packet playout deadlines, in response to varying network conditions. A Wiener process is used to model the fluctuation of the buffering length without any playout adjustment. The playout scheduling problem is then reformulated as a stochastic impulse control problem by taking the playout adjustment as the control signal. The proposed approach is shown to be the optimal solution to the new control problem. It is demonstrated experimentally that the proposed approach provides improved perceived conversional quality. Papers B, C and D address the packet-loss issue. Paper B focuses on the design of a low-complexity packet-loss concealment (PLC) method that is compatible with existing speech codecs for VoIP application. The new method is rigorously motivated based on the autoregressive (AR) speech model and the minimum mean squared error (MMSE) criterion. The effect of model estimation error on the prediction of the missing speech segment is also considered and an upper bound for the prediction error is derived. Both the theoretical and experimental results provide insight in the performance of the heuristically designed PLC methods. On the other hand, Paper C and D consider an active packet-loss-resilient coding scheme, namely multiple description coding (MDC). In general, MDC can be used for the transmission of any media data. Paper C derives a simple and accurate approximation of the rate-distortion lower bound of a particular multiple- description scenario and then demonstrates that the performance loss of some practical MD systems can be evaluated easily with the new approximation. Paper D studies the performance limit of a vector Gaussian multiple description scenario. An outer bound to the rate-distortion region is derived, and the outer bound is tight when the problem specializes to the scalar Gaussian case.QC20100830
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Fabri, S. N. "Multimedia communications over mobile packet networks." Thesis, University of Surrey, 2001. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.343461.
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This thesis describes several concepts associated with the transmission of multimedia services over mobile radio access networks. The error performance and traffic requirements of real-time video transmission over the General Packet Radio Services (GPRS) access network and its successor Enhanced-GPRS is examined. In view of this. video error resilience techniques which exploit channel prioritisation mechanisms are introduced with a view to increasing the robustness of received video sequences encoded with MPEG-4 to channel errors. These include stream prioritisation using unequal error protection and region-of-interest prioritisation for use in multiparty communications and streaming applications. A new forward-error correction scheme for EGPRS which uses iterative serially-concatenated convolutional-Reed Solomon codes is designed and is shown to significantly enhance the error performance for real-time services. A study of (he use of backward error correction mechanisms when transmitting streaming multimedia services is carried out, and a new retransmission scheme is introduced to increase the capacity of the radio access network when supporting streaming services
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Holzbock, Matthias. "Mobile multimedia service provisioning with collective terminals in broadband satellite networks : an approach for systematic satellite communication system design for service provisioning to collective mobile terminals, including mobile satellite channel modelling, antenna pointing, hierarchical multi-service dimensioning and aeronautical system dimensioning." Thesis, University of Bradford, 2011. http://hdl.handle.net/10454/5657.
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This work deals with provisioning of communication services via satellites for collectively mobile user groups in a heterogeneous network with several radio access technologies. The extended use of personalised user equipment beyond the coverage of one single terrestrial network by means of a satellite transport link, represents an increasingly important trend in mobile satellite communication. This trend is confirmed by the commercial introduction of broadband satellite communication to mobile terminals mounted on vehicles, trains, ships or aircraft. This work provides a consequent and structured approach for provisioning of services to broadband satellite terminals for mobile user groups and addresses: -- a systematic satellite communication system design process for collective mobile terminals; -- mobile satellite modelling at a wide range of frequencies, including current and potential frequencies; -- an optimised Pointing Acquisition and Tracking (PAT) system design including characterisation of moments for vehicle types of all mobile scenarios; -- a general hierarchical multi-service dimensioning methodology for collectively mobile user groups, including voice, data, and multimedia services; -- an aeronautical system dimensioning scheme with (capacity and handover) requirements analysis and evaluation of results for different satellite scenarios.
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Chigwamba, Nyasha. "An investigation of parameter relationships in a high-speed digital multimedia environment." Thesis, Rhodes University, 2014. http://hdl.handle.net/10962/d1021153.
Full textAbstract:
With the rapid adoption of multimedia network technologies, a number of companies and standards bodies are introducing technologies that enhance user experience in networked multimedia environments. These technologies focus on device discovery, connection management, control, and monitoring. This study focused on control and monitoring. Multimedia networks make it possible for devices that are part of the same network to reside in different physical locations. These devices contain parameters that are used to control particular features, such as speaker volume, bass, amplifier gain, and video resolution. It is often necessary for changes in one parameter to affect other parameters, such as a synchronised change between volume and bass parameters, or collective control of multiple parameters. Thus, relationships are required between the parameters. In addition, some devices contain parameters, such as voltage, temperature, and audio level, that require constant monitoring to enable corrective action when thresholds are exceeded. Therefore, a mechanism for monitoring networked devices is required. This thesis proposes relationships that are essential for the proper functioning of a multimedia network and that should, therefore, be incorporated in standard form into a protocol, such that all devices can depend on them. Implementation mechanisms for these relationships were created. Parameter grouping and monitoring capabilities within mixing console implementations and existing control protocols were reviewed. A number of requirements for parameter grouping and monitoring were derived from this review. These requirements include a formal classification of relationship types, the ability to create relationships between parameters with different underlying value units, the ability to create relationships between parameters residing on different devices on a network, and the use of an event-driven mechanism for parameter monitoring. These requirements were the criteria used to govern the implementation mechanisms that were created as part of this study. Parameter grouping and monitoring mechanisms were implemented for the XFN protocol. The mechanisms implemented fulfil the requirements derived from the review of capabilities of mixing consoles and existing control protocols. The formal classification of relationship types was implemented within XFN parameters using lists that keep track of the relationships between each XFN parameter and other XFN parameters that reside on the same device or on other devices on the network. A common value unit, known as the global unit, was defined for use as the value format within value update messages between XFN parameters that have relationships. Mapping tables were used to translate the global unit values to application-specific (universal) units, such as decibels (dB). A mechanism for bulk parameter retrieval within the XFN protocol was augmented to produce an event-driven mechanism for parameter monitoring. These implementation mechanisms were applied to an XFN-protocol-compliant graphical control application to demonstrate their usage within an end user context. At the time of this study, the XFN protocol was undergoing standardisation within the Audio Engineering Society. The AES-64 standard has now been approved. Most of the implementation mechanisms resulting from this study have been incorporated into this standard.
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Wang, Ju. "Multimedia communication with cluster computing and wireless wcdma network." [Gainesville, Fla.] : University of Florida, 2003. http://purl.fcla.edu/fcla/etd/UFE0000944.
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Liang, Lei. "Peformance evaluations of IP multiparty multimedia communications over GEO satellite networks." Thesis, University of Surrey, 2005. http://epubs.surrey.ac.uk/844577/.
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With the rapid development of the Internet, new theories and technologies are blooming and boosting the associated applications. One group of important applications is the multiparty multimedia communications. Satellites, which have played an important role in telephony communication and TV broadcasting services, could also play an important role to provide multiparty multimedia communications with their global coverage and on-board processing ability over IP networks. IP multiparty multimedia conferencing is one of these communication applications. With the support of satellites, one can provide an IP conferencing service globally to anywhere, even the place does not have the access to terrestrial networks. This thesis introduces the VoIP technologies that underpin IP conferencing services. It describes protocols, architectures, network entities, and network performance. In addition, this thesis presents how a multicast routing architecture was designed for an IP conferencing system with consideration to the new features introduced by an integrated GEO satellite network. An associated conferencing model will be presented as well to accompany this routing architecture. All of the technologies used in the design were implemented in a demonstrator. To test and evaluate the system, efforts have been put into the IP traffic measurement technologies and a measurement regime was developed to evaluate the system with consideration to multicast routing and the system architecture. New relative QoS requirements of multiparty communications are identified in this thesis. A set of parameters to present these new requirements are proposed as derivations of the IPPM (IP Performance Metrics) end-to-end parameters. A new adaptive QoS optimisation algorithm is proposed that is based on the measurement of these new parameters to satisfy the relative QoS requirements of the multiparty multimedia communications. Simulations were carried out to verify this algorithm and the results prove that it can optimize the relative QoS for multiparty multimedia communications.
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Ni, Jian. "Content management and admission control in multimedia content delivery networks /." View abstract or full-text, 2003. http://library.ust.hk/cgi/db/thesis.pl?ELEC%202003%20NI.
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Kim, Kicheon. "QoS supporting mechanisms for a global packet switching network." Thesis, Lancaster University, 1997. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.387467.
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Kompella, Sastry Venkata Subrahmanya. "Video Communications over Dynamic Ad Hoc Networks." Diss., Virginia Tech, 2006. http://hdl.handle.net/10919/28281.
Full textAbstract:
Video communications play a vital role in present and future wireless ad hoc networks. One of the key requirements for a successful deployment of multimedia applications in multihop wireless networks is the ability to provide an acceptable video quality, even under a highly dynamic and perhaps unfriendly (or hostile) environment (e.g., in the presence of frequent node/link failure, interference, shadowing, fading, and so forth). Existing ad hoc routing protocols work well for data communications, but are not optimized for video, which is sensitive to latency and packet loss. Moreover, traditional end system based error control mechanisms alone cannot guarantee a sustainable video quality. Conventional QoS approaches typically optimize one or more network layer metrics, but they are usually agnostic to any kind of application layer performance. Consequently, new methodologies must be explored to improve the performance of video applications in multihop wireless networks.
This dissertation directly addresses this important problem area by leveraging recent advances in video coding techniques along with novel cross-layer formulations and powerful optimization techniques. We follow an application centric cross-layer approach to address multimedia service provisioning over ad hoc networks. Our research efforts show that video communications over multihop wireless networks can substantially benefit from a cross-layer design principle by factoring in application layer video quality into routing algorithmic designs at the network layer. There are three components in this investigation, namely, (1) concurrent routing, (2) path selection and rate allocation, and (3) multipath routing for multiple description video. Each component addresses one or more unique challenges that hinder video communications in multihop wireless networks. Although we expect that a cross-layer approach will be more effective than a network centric (single-layer) approach in addressing application performance, it also brings in complex problems that cannot be effectively solved using traditional methods, and thus, calls for the design of customized algorithms.
In concurrent routing, we focus on issues that arise while supporting multiple concurrent video communication sessions in an ad hoc network. These sessions compete for limited network resources (such as bandwidth) while interacting with each other. Such inter-session interactions couple the performance of an individual flow with that of other flows. Applying a video centric cross-layer design principle, we model the end-to-end video distortion as a function of network layer behavior, and formulate a network-wide optimal routing problem that minimizes the total video distortion. Results based on computational experiments performed using randomly generated network topologies establish the relative efficacy and robustness of the proposed genetic algorithm based solution approach. Specifically, we demonstrate that our approach outperforms other trajectory based metaheuristic approaches as well as with conventional network centric routing algorithms such as shortest path and disjoint shortest path routing.
The joint path selection and rate allocation problem considers not only selecting the best set of paths for video communication, but also, computing the optimal video encoding rate and partitioning it among the chosen set of paths. The end-to-end video distortion is modeled as a function of network layer resources by capturing the tight coupling that exists between the optimal encoding rate for each video session, the selection of paths for video transmission, and the allocation of traffic among these selected paths. This problem is formulated as a nonlinear nonconvex programming problem, for which a tight linear programming relaxation is constructed via the Reformulation-Linearization Technique (RLT). This construct is embedded within a specialized branch-and-bound algorithm to achieve global optimality. Computational experience is reported for various problem instances, and the results validate the robustness of the proposed algorithmic procedure. The results exhibit the advantage of the solution approach over the popularly used max-min rate allocation scheme.
The emergence of Multiple Description (MD) coding technique offers great potential for multipath routing of video in multihop wireless networks. In studying multipath routing for MD coding, we show that MD coded video, when used in combination with multipath routing in wireless networks, has tremendous advantages over traditional layered video coding techniques. We discuss how to implement an MD video codec and formulate a cross-layer optimization problem that can find a set of optimal paths, (one for each description) such that the overall video quality at the receiver is maximized. We further devise a specialized RLT-based branch-and-bound solution procedure for the ensuing 0-1 mixed integer nonconvex optimization problem. Convergence behavior of the proposed solution procedure is observed for various network topologies and the results further demonstrate the performance advantage of the proposed cross-layer approach over non-cross-layer approaches.
The scope of this research is highly interdisciplinary. It intersects video communication, networking, optimization, and algorithm design. We expect that the theoretical and algorithmic results of this investigation will serve as important building blocks in developing a comprehensive methodology for addressing complex cross-layer problems in the area of wireless ad hoc networks.Ph. D.
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Foster, Mark. "A delay-efficient satellite network for multimedia communication a pilot study /." [Gainesville, Fla.] : University of Florida, 2002. http://purl.fcla.edu/fcla/etd/UFE1000123.
Full textAbstract:
Thesis (M.S.)--University of Florida, 2002.Title from title page of source document. Document formatted into pages; contains viii, 100 p.; also contains graphics. Includes vita. Includes bibliographical references.
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Lee, Myounghwan. "Quality of service with DiffServ architecture in hybrid mesh/relay networks." Diss., Georgia Institute of Technology, 2010. http://hdl.handle.net/1853/34694.
Full textAbstract:
The objective of this research is to develop an optimized quality of service (QoS) assurance algorithm with the differentiated services (DiffServ) architecture, and a differentiated polling algorithm with efficient bandwidth allocation for QoS assurance in the hybrid multi-hop mesh/relay networks. These wide area networks (WANs), which will employ a connection-based MAC protocol, along with QoS-enabled wireless local area networks (WLANs) that use a contention-based MAC protocol, need to provide an end-to-end QoS guarantee for data communications, particularly QoS-sensitive multimedia communications.
Due to the high cost of construction and maintenance of infrastructure in wireless networks, engineers and researchers have focused their investigations on wireless mesh/relay networks with lower cost and high scalability. For current wireless multi-hop networks, an end-to-end QoS guarantee is an important functionality to add, because the demand for real-time multimedia communications has recently been increasing. For real-time multimedia communication in heterogeneous networks, hybrid multi-hop mesh/relay networks using a connection-based MAC protocol, along with QoS-enabled WLANs that use a contention-based MAC protocol can be an effective multi-hop network model , as opposed to multi-hop networks with a contention-based MAC protocol without a QoS mechanism. To provide integrated QoS support for different QoS mechanisms, the design of the cross-layer DiffServ architecture that can be applied in wireless multi-hop mesh/relay networks with WLANs is desirable.
For parameterized QoS that requires a specific set of QoS parameters in hybrid multi-hop networks, an optimized QoS assurance algorithm with the DiffServ architecture is proposed here that supports end-to-end QoS through a QoS enhanced WAN for multimedia communications.
For a QoS assurance algorithm that requires a minimum per-hop delay, the proper bandwidth to allow the per-hop delay constraint needs to be allocated. Therefore, a polling algorithm with a differentiated strategy at multi-hop routers is proposed here. The proposed polling algorithm at a router differentially computes and distributes the polling rates for routers according to the ratio of multimedia traffic to overall traffic, the number of traffic connections, and the type of polling service.
By simulating the architecture and the algorithms proposed in this thesis and by analyzing traffic with the differentiated QoS requirement, it is shown here that the architecture and the algorithms produce an excellent end-to-end QoS guarantee.
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Katar, Srinivas. "Quality of service design issues in multimedia communications over power line networks." [Gainesville, Fla.] : University of Florida, 2006. http://purl.fcla.edu/fcla/etd/UFE0013881.
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Zhou, Liang. "Research on Key Techniques for Multimedia communications in Wireless Multi-Hop Networks." Cachan, Ecole normale supérieure, 2009. http://tel.archives-ouvertes.fr/tel-00506045/fr/.
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Dans cette thèse, nous étudions certains points techniques clé sur ce sujet et proposons une résolution pratique et efficace des aspects suivants: Dans un cadre de design inter-couche, nous avons d'abord réglé le problème du contrôle de débit et du routage. Nous étudions la combinaison de l'acheminement et le contrôle du taux avec une formulation optimisation convexe afin de proposer une solution distribuée commune basée sur la conception des inter-couche. Nous avons d'abord développé un modèle de distorsion qui couvre à la fois l'impact de la quantification du codeur et la perte de paquets en raison de la congestion du réseau sur la qualité vidéo globale. Ensuite, le contrôle du taux d'émission optimal commun et le choix du routage sont réalisés en adaptant la vitesse variable d'émission au cours du temps et en minimisant la congestion du réseau global. Ensuite, nous abordons le problème du mécanisme d'authentification du codage conjoint. Dans cette partie, nous proposons un cadre pour une collaboration entre authentification et codage multimédia transmis sur des réseaux sans fil. Premièrement, nous fournissons un système d'authentification basé sur des méthodes graphiques qui peut non seulement construire une authentification flexible, mais aussi un compromis parmi certaines exigences pratiques telles que la robustesse et le retard. Puis, un taux de distorsion conjoint Source-Canal (JSCC) est optimisé ; une approche résistante aux erreurs vidéo encodée adpatative est présentée, dans laquelle la vidéo est encodée en plusieurs flux indépendants et à chaque flux est attribué une correction (Forward Error Correction-FEC) afin d'éviter la propagation des erreurs. En outre, nous considérons l'authentification avec le régime spécifique JSCC afin de parvenir à un résultat satisfaisant en termes de qualité de reconstruction de bout en bout. Ensuite, nous proposons d'appliquer conjointement la résilience aux erreurs (ER) et la super-résolution (SR) afin d'améliorer glabalement la résolution des flux d'images transmises sur les réseaux sans fil. Afin de lutter contre la propagation d'erreurs, une description multiple, méthode flexible de codage basée sur SPIHT-3-D (partitionnement 3-D dans des arbres hiérarchiques) est présentée pour générer des descriptions variables indépendantes en fonction de l'état du réseau. Puis, une stratégie originale de protection inégale contre les erreurs suivant le niveau de priorité est prévu afin d'attribuer un niveau supérieur de protection contre les erreurs des parties les plus importantes des flux. En outre, un algorithme robuste SR est proposé, en présence de différents types de taux de perte de paquets pour améliorer la résolution de l'image. Enfin, nous décrivons le problème du contrôle du taux pour le multimédia sur des réseaux hétérogènes. Dans cette partie, nous développons et évaluons un cadre d'allocation des taux optimaux sur plusieurs réseaux hétérogènes toujours basés sur le cadre de la conception inter-couches. Au début, nous développons et évaluons un modèle de distorsion en fonction de taux disponibles observés (ABR) et du temps d'aller-retour (RTT) dans chaque réseau d'accès, ainsi que des caractéristiques de chaque application. Ensuite, la répartition du taux est formulée comme un problème d'optimisation convexe qui minimise la somme de la distorsion attendue sur tous les flux applicatifs. Afin d'obtenir une résolution satisfaisante et simple à ce problème, nous proposons un théorème d'approximation par morceaux afin de simplifier le problème convexe et de prouver la validité de la théorie. En outre, la réalisation de l'algorithme rapide heuristique pour parvenir à une répartition optimale de qualité de service ou quasi-optimale de bout en bout dans le cadre du budget global des ressources limitées est le point le plus important de cette partie de thèse. En fin de compte, un bref résumé de tous les sujets abordés dans cette thèse est donné. Les principales contributions de cette thèse sont alors rappelésWith the latest developments in multimedia coding technology and fast deployment of end-user wireless connections, real-time media applications become increasing interesting for both private users and businesses. However, most of current wireless networks remain a best-effort service network unable to guarantee the stringent requirements of the media application, in terms of high, constant bandwidth, low packet loss rate and transmission delay. Therefore, efficient scheduling mechanism must be derived in order to bridge the application requirements with the transport medium characteristics. Up to now, multimedia communications over wireless multi-hop networks is still an open problem. In this dissertation, we study some key technical points about this topic and propose some practical and efficient resolution from the following aspects: Within a cross-layer design framework, we first address the problem of joint rate control and routing. Here, we invest the combination of the routing and rate control in a united convex optimization formulation, and propose a distributed joint solution based on cross-layer design. We first develop a distortion model which captures both the impact of encoder quantization and packet loss due to network congestion on the overall video quality. Then, the optimal joint rate control and routing scheme is realized by adapting its rate to the time-varying traffic and minimizing the overall network congestion. Next, we address the problem of joint authentication-coding mechanism. In this part, we propose a framework for jointly authenticating and coding multimedia to be transmitted over wireless networks. We firstly provide a novel graph-based authentication scheme which can not only construct the authentication graph flexibly but also trade-off well among some practical requirements such as overhead, robustness and delay. And then, a rate-distortion optimized joint source-channel coding (JSCC) approach for error-resilient scalable encoded video is presented, in which the video is encoded into multiple independent streams and each stream is assigned forward error correction (FEC) codes to avoid error propagation. Furthermore, we consider integrating authentication with the specific JSCC scheme to achieve a satisfactory authentication results and end-to-end reconstruction quality by optimally applying the appropriate authentication and coding rate. Robust resolution-enhancement scheme problem is addressed next in our thesis. We propose a coordinated application of ER (Error-Resilient) and SR (Super-Resolution) to enhance the resolution of image transmitted over wireless networks. In order to combat error propagation, a flexible multiple description coding method based on shifted 3-D SPIHT (3-D Set Partitioning In Hierarchical Trees) algorithm is presented to generate variable independent descriptions (sub-streams) according to the network condition. And then, a novel unequal error protection strategy based on the priority level is provided to assign a higher level of error protection to more important parts of bit-stream. Moreover, a robust SR algorithm is proposed in the presence of different kinds of packet loss rate to enhance the image resolution. Finally, we describe the problem of rate control for multimedia over heterogeneous networks. In this part, we develop and evaluate a framework for optimal rate allocation over multiple heterogeneous networks based on cross-layer design framework. At first, we develop and evaluate a distortion model according to the observed Available Bit Rate (ABR) and the Round Trip Time (RTT) in each access network, as well as each application's rate-distortion characteristic. Then, the rate allocation is formulated as a convex optimization problem that minimizes the sum of expected distortion of all application streams. In order to get a satisfying and simple resolution for this problem, we propose a piecewise-approximate theorem to simplify the convex optimal problem and prove its validity in theory. Furthermore, the realization of the fast heuristic rate allocation algorithm for achieving an optimal or close-to-optimal end-to-end QoS under the overall limited resource budget is the highlight of this paper. In the end, a brief summary of all discussed topics in this dissertation is given. The main contributions of this dissertation and several further studies or worth studies are pointed out at this part
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Curtin, Patrick. "A realistic, survivable packet radio network design for mobile multimedia communications." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1998. http://www.collectionscanada.ca/obj/s4/f2/dsk2/tape17/PQDD_0034/MQ38741.pdf.
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Samanta, Swadesh Kumar. "Factors affecting the growth of multimedia communications services in a network." Thesis, University of Essex, 2009. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.510506.
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Curtin, Patrick Rowland. "A realistic, survivable packet radio network design for mobile multimedia communications." Thesis, University of Ottawa (Canada), 1998. http://hdl.handle.net/10393/4543.
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The provision of multimedia services to a wireless environment presents several technical challenges. The extension of these services to a survivable, distributed system adds a significant level of complexity. Packet radio network technology offers one possible starting point for a solution. This thesis approaches the problem of delivering integrated voice, low grade video and low rate data services to a wireless network operating in a distributed environment through the use of a packet radio architecture. An overview of the technical specifications to be addressed in the network design is used to define the operational requirement. A radio channel model based on the Hata-Okumara approach is then developed in order to provide a realistic simulation framework for the system and a basis for network connectivity maps in subsequent chapters. The issues related to the operation of a distributed network are then explored, essentially illustrating the design implications of survivability. Several different multiple access techniques are also evaluated for suitability in this network design. Based upon the criteria provided at the onset, a slotted CDMA methodology is recommended. This leads to the establishment of network control parameters including the data structure, routing and scheduling schemes based on slotted CDMA operation in a distributed environment. Finally, issues related to network integration and enhancements are discussed, including CDMA code orientation, control channel modifications and acknowledgements. Recommendations for future work are also provided.
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Hsiao, Hsu-Feng. "Multimedia streaming congestion control over heterogeneous networks : from distributed computation and end-to-end perspectives /." Thesis, Connect to this title online; UW restricted, 2005. http://hdl.handle.net/1773/5946.
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Shen, Ji. "Efficient 3G multimedia communication control protocol (H.245) and implementations /." access full-text access abstract and table of contents, 2005. http://libweb.cityu.edu.hk/cgi-bin/ezdb/thesis.pl?mphil-cs-b19886093a.pdf.
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Thesis (M.Phil.)--City University of Hong Kong, 2005."Submitted to Department of Computer Science in partial fulfillment of the requirements for the degree of Master of Philosophy" Includes bibliographical references (leaves 110-112)