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

User-Oriented QoS in Packet Video Delivery

We focus on packet video delivery, with an emphasis on the quality of service perceived by the end-user. A video signal passes through several subsystems, such as the source coder, the network and the decoder. Each of these can impair the information, either by data loss or by introducing delay. We describe how each of the subsystems can be tuned to optimize the quality of the delivered signal, for a given available bit rate in the network. The assessment of end-user quality is not trivial. We present recent research results, which rely on a model of the human visual system

DSL-based triple-play services

This research examines the triple play service based on the ADSL technology. The voice over IP will be checked and combined with the internet data by two monitoring programs in order to examine the performance that this service offers and then will be compared with the usual method of internet connection.This research examines the triple play service based on the ADSL technology. The voice over IP will be checked and combined with the internet data by two monitoring programs in order to examine the performance that this service offers and then will be compared with the usual method of internet connection.

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