Designing new network adaptation and ATM adaptation layers for interactive multimedia applications

Multimedia services, audiovisual applications composed of a combination of discrete and continuous data streams, will be a major part of the traffic flowing in the next generation of high speed networks. The cornerstones for multimedia are Asynchronous Transfer Mode (ATM) foreseen as the technology for the future Broadband Integrated Services Digital Network (B-ISDN) and audio and video compression algorithms such as MPEG-2 that reduce applications bandwidth requirements. Powerful desktop computers available today can integrate seamlessly the network access and the applications and thus bring the new multimedia services to home and business users. Among these services, those based on multipoint capabilities are expected to play a major role.    Interactive multimedia applications unlike traditional data transfer applications have stringent simultaneous requirements in terms of loss and delay jitter due to the nature of audiovisual information. In addition, such stream-based applications deliver data at a variable rate, in particular if a constant quality is required.    ATM, is able to integrate traffic of different nature within a single network creating interactions of different types that translate into delay jitter and loss. Traditional protocol layers do not have the appropriate mechanisms to provide the required network quality of service (QoS) for such interactive variable bit rate (VBR) multimedia multipoint applications. This lack of functionalities calls for the design of protocol layers with the appropriate functions to handle the stringent requirements of multimedia.    This thesis contributes to the solution of this problem by proposing new Network Adaptation and ATM Adaptation Layers for interactive VBR multimedia multipoint services.    The foundations to build these new multimedia protocol layers are twofold; the requirements of real-time multimedia applications and the nature of compressed audiovisual data.    On this basis, we present a set of design principles we consider as mandatory for a generic Multimedia AAL capable of handling interactive VBR multimedia applications in point-to-point as well as multicast environments. These design principles are then used as a foundation to derive a first set of functions for the MAAL, namely; cell loss detection via sequence numbering, packet delineation, dummy cell insertion and cell loss correction via RSE FEC techniques.    The proposed functions, partly based on some theoretical studies, are implemented and evaluated in a simulated environment. Performances are evaluated from the network point of view using classic metrics such as cell and packet loss. We also study the behavior of the cell loss process in order to evaluate the efficiency to be expected from the proposed cell loss correction method. We also discuss the difficulties to map network QoS parameters to user QoS parameters for multimedia applications and especially for video information. In order to present a complete performance evaluation that is also meaningful to the end-user, we make use of the MPQM metric to map the obtained network performance results to a user level. We evaluate the impact that cell loss has onto video and also the improvements achieved with the MAAL.    All performance results are compared to an equivalent implementation based on AAL5, as specified by the current ITU-T and ATM Forum standards.    An AAL has to be by definition generic. But to fully exploit the functionalities of the AAL layer, it is necessary to have a protocol layer that will efficiently interface the network and the applications. This role is devoted to the Network Adaptation Layer.    The network adaptation layer (NAL) we propose, aims at efficiently interface the applications to the underlying network to achieve a reliable but low overhead transmission of video streams. Since this requires an a priori knowledge of the information structure to be transmitted, we propose the NAL to be codec specific.    The NAL targets interactive multimedia applications. These applications share a set of common requirements independent of the encoding scheme used. This calls for the definition of a set of design principles that should be shared by any NAL even if the implementation of the functions themselves is codec specific. On the basis of the design principles, we derive the common functions that NALs have to perform which are mainly two; the segmentation and reassembly of data packets and the selective data protection.    On this basis, we develop an MPEG-2 specific NAL. It provides a perceptual syntactic information protection, the PSIP, which results in an intelligent and minimum overhead protection of video information. The PSIP takes advantage of the hierarchical organization of the compressed video data, common to the majority of the compression algorithms, to perform a selective data protection based on the perceptual relevance of the syntactic information.    The transmission over the combined NAL-MAAL layers shows significant improvement in terms of CLR and perceptual quality compared to equivalent transmissions over AAL5 with the same overhead.    The usage of the MPQM as a performance metric, which is one of the main contributions of this thesis, leads to a very interesting observation. The experimental results show that for unexpectedly high CLRs, the average perceptual quality remains close to the original value. The economical potential of such an observation is very important. Given that the data flows are VBR, it is possible to improve network utilization by means of statistical multiplexing. It is therefore possible to reduce the cost per communication by increasing the number of connections with a minimal loss in quality.    This conclusion could not have been derived without the combined usage of perceptual and network QoS metrics, which have been able to unveil the economic potential of perceptually protected streams.    The proposed concepts are finally tested in a real environment where a proof-of-concept implementation of the MAAL has shown a behavior close to the simulated results therefore validating the proposed multimedia protocol layers

Novel techniques in large scaleable ATM switches

Bibliography: p. 172-178.This dissertation explores the research area of large scale ATM switches. The requirements for an ATM switch are determined by overviewing the ATM network architecture. These requirements lead to the discussion of an abstract ATM switch which illustrates the components of an ATM switch that automatically scale with increasing switch size (the Input Modules and Output Modules) and those that do not (the Connection Admission Control and Switch Management systems as well as the Cell Switch Fabric). An architecture is suggested which may result in a scalable Switch Management and Connection Admission Control function. However, the main thrust of the dissertation is confined to the cell switch fabric. The fundamental mathematical limits of ATM switches and buffer placement is presented next emphasising the desirability of output buffering. This is followed by an overview of the possible routing strategies in a multi-stage interconnection network. A variety of space division switches are then considered which leads to a discussion of the hypercube fabric, (a novel switching technique). The hypercube fabric achieves good performance with an O(N.log₂N)²) scaling. The output module, resequencing, cell scheduling and output buffering technique is presented leading to a complete description of the proposed ATM switch. Various traffic models are used to quantify the switch's performance. These include a simple exponential inter-arrival time model, a locality of reference model and a self-similar, bursty, multiplexed Variable Bit Rate (VBR) model. FIFO queueing is simple to implement in an ATNI switch, however, more responsive queueing strategies can result in an improved performance. An associative memory is presented which allows the separate queues in the ATM switch to be effectively logically combined into a single FIFO queue. The associative memory is described in detail and its feasibility is shown by laying out the Integrated Circuit masks and performing an analogue simulation of the IC's performance is SPICE3. Although optimisations were required to the original design, the feasibility of the approach is shown with a 15Ƞs write time and a 160Ƞs read time for a 32 row, 8 priority bit, 10 routing bit version of the memory. This is achieved with 2µm technology, more advanced technologies may result in even better performance. The various traffic models and switch models are simulated in a number of runs. This shows the performance of the hypercube which outperforms a Clos network of equivalent technology and approaches the performance of an ideal reference fabric. The associative memory leverages a significant performance advantage in the hypercube network and a modest advantage in the Clos network. The performance of the switches is shown to degrade with increasing traffic density, increasing locality of reference, increasing variance in the cell rate and increasing burst length. Interestingly, the fabrics show no real degradation in response to increasing self similarity in the fabric. Lastly, the appendices present suggestions on how redundancy, reliability and multicasting can be achieved in the hypercube fabric. An overview of integrated circuits is provided. A brief description of commercial ATM switching products is given. Lastly, a road map to the simulation code is provided in the form of descriptions of the functionality found in all of the files within the source tree. This is intended to provide the starting ground for anyone wishing to modify or extend the simulation system developed for this thesis

Satellite Networks: Architectures, Applications, and Technologies

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

Multimedia in mobile networks: Streaming techniques, optimization and User Experience

1.UMTS overview and User Experience 2.Streaming Service & Streaming Platform 3.Quality of Service 4.Mpeg-4 5.Test Methodology & testing architecture 6.Conclusion

Analysis of the impact of impulse noise in digital subscriber line systems

In recent years, Digital subscriber line (DSL) technology has been gaining popularity as a high speed network access technology, capable of the delivery of multimedia services. A major impairment for DSL is impulse noise in the telephone line. However, evaluating the data errors caused by this noise is not trivial due to its complex statistical nature, which until recently had not been well understood, and the complicated error mitigation and framing techniques used in DSL systems. This thesis presents a novel analysis of the impact of impulse noise and the DSL framing parameters on transmission errors, building on a recently proposed impulse noise model. It focuses on errors at higher protocol layers, such as asynchronous transfer mode (ATM), in the most widely used DSL version, namely Asymmetric DSL (ADSL). The impulse noise is characterised statistically through its amplitudes, duration, inter-arrival times, and frequency spectrum, using the British Telecom / University of Edinburgh / Deutsche Telekom (BT/UE/DT) model. This model is broadband, considers both the time and the frequency domains, and accounts for the impulse clustering. It is based on recent measurements in two different telephone networks (the UK and Germany) and therefore is the most complete model available to date and suited for DSL analysis. A new statistical analysis of impulse noise spectra from DT measurements shows that impulse spectra can be modelled with three spectral components with similar bandwidth statistical distributions. Also, a novel distribution of the impulse powers is derived from the impulse amplitude statistics. The performance of a generic ADSL modem is investigated in an impulse noise and crosstalk environment for different bit rates and framing parameters. ATM cell and ADSL frame error rates, and subjective MPEG2 video quality are used as performance metrics. A new modification of a bit loading algorithm is developed to enable stable convergence of the algorithm with trellis coding and restricted subtone constellation size. It is shown that while interleaving brings improvement if set at its maximum depth, at intermediate depths it actually worsens the performance of all considered metrics in comparison with no interleaving. No such performance degradation is caused by combining several symbols in a forward error correction (FEC) codeword, but this burst error mitigation technique is only viable at low bit rates. Performance improvement can also be achieved by increasing the strength of FEC, especially if combined with interleaving. In contrast, trellis coding is ineffective against the long impulse noise error bursts. Alien as opposed to kindred crosstalk degrades the error rates and this is an important issue in an unbundled network environment. It is also argued that error free data units is a better performance measure from a user perspective than the commonly used error free seconds. The impact of impulse noise on the errors in DSL systems has also been considered analytically. A new Bernoulli-Weibull impulse noise model at symbol level is proposed and it is shown that other models which assume Gaussian distributed impulse amplitudes or Rayleigh distributed impulse powers give overly optimistic error estimates in DSL systems. A novel bivariate extension of the Weibull impulse amplitudes is introduced to enable the analysis of orthogonal signals. Since an exact closed-form expression for the symbol error probability of multi-carrierQAM assuming Bernoulli-Weibull noise model does not exist, this problem has been solved numerically. Multi-carrier QAM is shown to perform better at high signal-to-noise ratio (SNR), but worse at low SNR than single carrier QAM, in both cases because of the spreading of noise power between subcarriers. Analytical expressions for errors up to frame level in the specific case of ADSL are then derived from the impulse noise model, with good agreement with simulation results. The Bernoulli-Weibull model is applied to study the errors in single-pair highspeed DSL (SHDSL). The performance of ADSL is found to be better when the burst error mitigation techniques are used, but SHDSL has advantages if low bit error rate and low latency are required

A slotted-CDMA based wireless-ATM link layer : guaranteeing QoS over a wireless link.

Thesis (M.Sc.)-University of Natal, Durban, 2000.Future wireless networks will have to handle varying combinations of multimedia traffic that present the network with numerous quality of service (QoS) requirements. The continuously growing demand for mobile phones has resulted in radio spectrum becoming a precious resource that cannot be wasted. The current second-generation mobile networks are designed for voice communication and, even with the enhancements being implemented to accommodate data, they cannot efficiently handle the multimedia traffic demands that will be introduced in the near future. This thesis begins with a survey of existing wireless ATM (WATM) protocols, followed by an examination of some medium access control (MAC) protocols, supporting multimedia traffic, and based on code division multiple access (CDMA) physical layers. A WATM link layer protocol based on a CDMA physical layer, and incorporating techniques from some of the surveyed protocols, is then proposed. The MAC protocol supports a wide range of service requirements by utilising a flexible scheduling algorithm that takes advantage of the graceful degradation of CDMA with increasing user interference to schedule cells for transmission according to their maximum bit error rate (BER) requirements. The data link control (DLC) accommodates the various traffic types by allowing virtual channels (VCs) to make use of forward error correction (FEc) or retransmission techniques. The proposed link layer protocol has been implemented on a Blue Wave Systems DSP board that forms part of Alcatel Altech Telecoms' software radio platform. The details and practicality of the implementation are presented. A simulation model for the protocol has been developed using MIL3 's Opnet Modeler. Hence, both simulated and measured performance results are presented before the thesis concludes with suggestions for improvements and future work

Contribución al diseño de arquitecturas distribuidas de nodos de red programable

Hoy en día, los nodos de red que forman Internet son complejos sistemas hardware/software que soportan un gran número de protocolos, servicios de red, o funcionalidades avanzadas como rewall o NAT. Sin embargo el proceso para añadir un nuevo protocolo o servicio es extremadamente largo y costoso, debido a múltiples causas, pero especialmente a que los routers siguen siendo sistemas propietarios, integrados verticalmente por los fabricantes. En este sentido, la investigación en redes programables intenta simpli car el desarrollo y el despliegue de los servicios de red mediante la de nición de interfaces abiertos entre todos los elementos que forman el router. Sin embargo hasta que los primeros diseños de nodos de red totalmente programables lleguen a comercializarse, es necesario aportar soluciones a corto y medio plazo que permitan ampliar las capacidades y servicios de los routers de alto rendimiento actuales. Esta tesis presenta una arquitectura de nodo de red programable de transici ón y bajo coste, denominada Simple Assistant-Router Architecture (SARA), que permite extender las capacidades de un router comercial delegando el procesamiento avanzado de los paquetes a un cluster de asistentes , lo que simpli ca el desarrollo y despliegue dinámico de los nuevos servicios de red. Un aspecto fundamental de esta arquitectura distribuida es la de nición de mecanismos de coordinación de los asistentes entre sí y con el router legado. Para ello se propone la utilización del Router-Assistant Protocol (RAP), un protocolo de control que permite a los asistentes con gurar el plano de datos del router, recibir eventos, así como desviar paquetes de señalización y ujos de datos para su procesamiento en los asistentes. Dada la heterogeneidad de los requisitos de las aplicaciones de red es necesario proporcionar varios mecanismos para asegurar un reparto de carga efectivo en el cluster de asistentes. Esta Tesis Doctoral propone dos algoritmos de Fast Robust Hashing que permiten la asignación equitativa y persistente de ujos a asistentes, mejorando el rendimiento de las técnicas de Robust Hashing actuales, por lo que son lo su cientemente e cientes como para ser implementados en el plano de datos de un router comercial. Además, este trabajo especi ca el eXtensible Service Discovery Framework (XSDF), un marco de trabajo sencillo y escalable, que integra en un único proceso el descubrimiento de servicios y el reparto de carga entre servidores desacoplados.Nowadays, the network nodes that build Internet are complex hardware/ software systems, that support many signalling protocols, network services, and complex functionalities such as rewalling or NAT. However adding a new capability is a long, complex and costly process, due to many causes, but specially because routers are still proprietary systems, vertically integrated by the vendors. In this sense, the research in programmable networks tries to simplify the development and deployment of network services by specifying open interfaces among all the elements that make up a router. However, before the rst programmable network nodes start being deployed, it is necessary to provide short and medium term solutions that allow current high-performance routers to add advanced capabilities and new network services. This PhD. Thesis presents a low-cost transition architecture for programmable network nodes named Simple Assistant-Router Architecture (SARA), that allows a commercial router to easily extend its capabilities by delegating the advanced packet processing to a cluster of assistants , which greatly simpli es the development and dynamic deployment of new network services. A key aspect of this distributed architecture is the need of several coordination mechanisms between the router and the assistants, and among assistant themselves. Therefore, the Router-Assistant Protocol (RAP) has been proposed, which is a control protocol based on ForCES, that allows assistants to con gure the router's data plane, to notify events, as well as to divert signalling packets and data ows to the assistants. As network application requirements could be very heterogeneous, it is necessary to provide several mechanisms in order to load-balance the assistant cluster. Thus, this Thesis presents two novel Fast Robust Hashing algorithms that provides a permanent and fair mapping of ows to assistants, and improves existing Robust Hash techniques as it is e cient enough to be implemented in the data plane of a commercial router. Moreover this research work also de - nes the eXtensible Service Discovery Framework (XSDF), which integrates in a single process: scalable service location, and load-sharing among lightly-coupled servers