The simple data link (SDL) is a new framing protocol for variable/fixed length packets over a general-purpose point-to-point communications channel. SDL extends the HEC-like framing mechanism used in ATM to variable-length data. Its low implementation complexity makes it particularly suitable for high-speed transport links in wavelength channels and dark fiber applications. SDL is a very attractive alternative to conventional solutions such as ATM and PPP-over-SONET (POS) for high data rate environments. SDL has also been designed to facilitate many OAM&P functions needed in next generation multiservice optical packet networks including multiprotocol encapsulation, virtual links, quality-of-service (QoS) differentiation, link management, and control. We describe framing and data link synchronization procedures in SDL, and evaluate its performance over octet and bit synchronous transport facilities.

Asynchronous transfer mode (ATM) is now well recognized as the fundamental switching and multiplexing technique for future broadband ISDN. As these networks will be increasingly relied upon for providing a multitude of integrated voice, data, and video services, network reliability is a key concern. There are several intrinsic features of ATM networks that could potentially be exploited to provide improved restoration techniques, beyond those established for synchronous transfer mode (STM) networks, such as digital cross-connect restoration or self-healing rings. These features include ATM cell level error detection, inherent rate adaptation and nonhierarchical multiplexing. The authors explore the use of these features in developing fast restoration strategies for ATM networks. In particular, they address: (1) ATM error detection capabilities for enhanced failure detection, (2) network rerouting strategies, (3) spare capacity allocation, and (4) network control architecture and related implementation aspects. Their findings suggest that fast network span failure detection and bandwidth-efficient rerouting capabilities can be combined to develop restoration strategies for ATM networks with significantly greater performance-cost ratios when compared to existing STM network restoration strategies.>

This paper investigates performance and engineering issues concerning a multiplexer scheme that has been implemented in AT&T's Integrated Access Terminal (IAT) to transport packetized voice and data traffic on shared facilities. The multiplexer serves voice and data traffic according to a dynamic bandwidth allocation scheme in order to simultaneously meet their performance requirements. A bit-dropping procedure is employed for voice packets to provide a graceful degradation of voice quality under overload conditions. An analytical model is developed for the multiplexer service scheme that estimates performance parameters given the voice and data offered loads. The model is used to demonstrate the capacity advantages of dynamic bandwidth allocation, and to generate load-service curves that illustrate the tradeoffs of carrying different combinations of voice and data traffic on the multiplexer. Sensitivity of voice and data performance to the multiplexer time-slice parameters is also investigated. The model is readily embedded in a design approach that determines the bandwidth required to carry the voice and data traffic demands while satisfying all desired performance objectives.>

Error control problems arising in the broadband ISDN environment are analyzed. The first problem relates to end-to-end integrity. The second problem is related to synchronization. The third problem is the performance of the cell-based cyclic redundancy check (CRC) in the presence of a self-synchronous scrambler to avoid attacks by malicious users. However, the scrambler creates error multiplication whereby a single error on the physical link is translated into two errors in the cell exactly 43 bits apart. In this context, it is proved that the cell-based CRC is capable of correcting correlated double errors produced by the self-asynchronous scrambler. The last problem studied is parallel implementation of CRC coders and decoders.>

The authors investigate error detection and correction options for data services in the broadband integrated services digital network (B-ISDN). They discuss and analyze different options for error detection and correction by considering the various alternatives ranging from no error protection to only error detection. Based on the analysis and results presented, a per-cell cyclic redundancy check (CRC) has been adopted in the adaptation layer for all data services. The authors focus on the ATM (asynchronous transfer mode) adaptation layer (AAL) error protection alternatives available within the framework of a per-cell CRC. The ATM cell header and AAL fields are described. A 4 bit cell sequence number for detecting cell misordering is analyzed. Based on the analysis, it is concluded that the 4 bit sequence number provides a powerful capability for detection of cell misordering.>

Error performance issues concerning IEEE Standard 802.6 metropolitan area networks (MANs) which will be based on fiber transmission systems are investigated. The error protection requirements for MAN cell headers are studied, and a simple cyclic redundancy check (CRC)-based scheme to provide single-bit correction and multiple-bit detection is proposed. The functions of the adaptation layer for data end points are discussed. A critical function of the data adaptation layer is to prevent corruption of information carried in 802.6 cells. Alternative ways of providing error protection are discussed and analyzed. It is shown that the alternative based on a CRC check per segment provides robust error protection and is amenable to very-high-speed implementation.>

An analytical model is presented for computing the end-to-end voice call performance in a packet network which drops the less significant bits in voice packets during periods of congestion. The authors first modify an existing single-node bit-dropping model to include the situation when the overall arrival process at a node consists of a mix of packets of different sizes. Variability in packet size is caused by bit dropping at previous nodes. A detailed model is presented to capture bit dropping effects in a tandem connection of nodes. The model includes the effects of load fluctuations at each node, and also takes into account the dependencies in bit dropping experienced by a voice packet at successive nodes in a tandem connection. These dependencies occur because a packet that has experienced bit dropping at a node in congestion state Q would not be further bit-dropped at a subsequent node if its current congestion state is no worse than Q. The model also incorporates the reduction in packet service times at intermediate nodes due to dropping at previous nodes. In addition to the detailed model, the authors present two approximation procedures which also serve as upper and lower bounds. In particular, the upper bound is shown to be very tight for practical range of loads, and hence serves as a good approximation with significant computational simplicity.>

Analytical models are presented for computing the end-to-end voice call performance in a packet network that drops the less significant bits in voice packets during periods of congestion. These models provide information about the end-to-end quality likely to be experienced in future packet-switched integrated services networks. An existing single-node bit-dropping model is modified to include the situation resulting when the overall arrival process at an internal node consists of a mix of packets of different sizes due to bit dropping at previous nodes. A detailed model to capture bit-dropping effects in a tandem connection of nodes is presented. The model includes the effect of load fluctuations at each node, and also takes into account the dependencies in bit dropping experienced by a voice packet at successive nodes in a tandem connection. The model also incorporates the internodal dependence when reductions in packet service times occur at intermediate nodes due to bit dropping at previous nodes. Two approximation procedures are discussed that serve as upper and lower bounds. In particular, the upper bound is shown to be very tight for a practical range of loads, and hence serves as a good approximation with significant computational simplicity.>

A comprehensive performance analysis method that models, at bit level, the error performance of individual links in an end-to-end connection is presented. The link model accounts for the burst-error behaviour of each individual link. A method to concatenate several individual links and extract a model for the end-to-end connection is given. This resulting end-to-end model can be used to calculate performance measures such as bit error rate and block error rate for any given block size. A procedure to compute the probability distribution of errors within a specific block is also developed. Finally, a method to compute the probability distribution of blocks having a certain error rate over a given period of time is presented. The utility and power of the model are illustrated with the help of an example connection.>

The two-dimensional (2-D) doubly stochastic Gaussian (DSG) model was introduced by one of the authors to provide a complete model for spatial filters which adapt to the local structure in an image signal. Here we present the optimal estimator and 2-D fixed-lag smoother for this DSG model extending earlier work of Ackerson and Fu. As the optimal estimator has an exponentially growing state space, we investigate suboptimal estimators using both a tree and a decision-directed method. Experimental results are presented.

This paper compares four image filtering algorithms on common data sets for various signal to noise ratios and white Gaussian noise. The algorithms are the median filter, the Wallis filter, the reduced update Kalman filter, and a multiple model, decision-directed filter. This comparison produces some surprising results from both a subjective (visual error) and a mean square error (MSE) viewpoint.

We present a recursive estimation algorithm for autoregressive moving average (ARMA) random field models. The estimator is an extension to ARMA models of the efficient reduced update Kalman filter (RUKF). We also discuss the identification of the ARMA model parameters from noise-free image data. The ARMA estimator is run on several sets of random field data as well as on real images. The experimental results are compared to those achievable using an AR model and RUKF on the same data fields. We find no significant improvement for comparable model orders.