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The problem of digital finite communication bandwidth (DFCB) control has come to the attention of the research community in connection with a growing interest in the development of distributed and/or networked control systems. In these systems, actuators, sensors, and other components are connected via data-rate constrained links such as wireless radio, etc. In this paper, we consider a scalar model of DFCB control that accommodates time-varying data-rate constraints, such as might occur with intermittent network congestion, and asynchronism of sampling and control actuation. Because of the possibly unpredictable fluctuation of the data-rate, we are interested in feedback control designs that will tolerate significantly constrained data-rates on feedback loops, while providing acceptable performance when such data-rate constraints are not in force. In light of a very basic notion of acceptable performance, we show that control designs with different number of quantization levels tolerate constrained data-rates differently. This leads to the conclusion that binary control represents the most robust control quantization under data-rate constraints imposed by time-varying congestion on the feedback communication channel. The advantage margin of binary control is further investigated numerically with and without the sampling-control asynchronism being considered. We show that the advantage margin is more substantial when the sampling-control asynchronism is significant. A design of quantized (binary) feedback with side channel information is proposed, and stability properties are discussed. We conclude the paper by examining performance limitations of our binary coding in the presence of noise. Index Terms--Coding, communication protocols, quantized control, sampling-control asynchronism.

The effects of the asynchronous transfer mode (ATM) and frame relay (FR) physical and data-link layers and their interworking on Internet protocol (IP) performance is examined and reported.

Protocols are proposed to achieve mutual anonymity in a peer to peer (P2P) file-sharing environment.

SEQUIN, a simple network management protocol (SNMP)-based monitoring system capable of monitoring quality of service (QoS) metrics, is described and discussed.

Session initiation protocol (SIP) elements are described, and services they can supply are discussed.

The session initiation protocol (SIP) is described, and the various ways it can be used are discussed.

The results of tests conducted on a 16 by 16 switch chip design based on the switch chip component of the highly scaleable crossbar system are presented. The successful demonstration of multi-Gigabit per second transmission through the superconducting switch was achieved with packet destination addressing having been decoded from the header of the data packet. The data were transmitted to a separate superconducting amplifier chip that is mounted on a superconducting multi-chip module.

This correspondence address two critical problems in designing new full-duplex fast training algorithms for simultaneously estimating echo and channel responses. Although algorithms of this type have been discussed and evaluated in previous work, the proposed algorithms concentrate on coping with the following two problems that were not solved in the previous approaches: 1) the symbol rate difference between the local transmitter and the remote transmitter, and 2) the frequency offsets in both far echo and far signal that are caused by the analog carrier network. The performance of the new methods is analyzed in terms of mean-square error. Simulation results are presented to confirm the analysis. Index Terms - Data transmission, digital communications, echo cancellation, equalization.

The increasing need for high data-rate transmissions over time- or frequency-selective fading channels has drawn attention to modulation schemes with high spectral efficiency such as QAM. With the aim of increasing the "diversity order" of the signal set we consider multidimensional rotated QAM constellations. Very high diversity orders can be achieved and this results in an almost Gaussian performance over the fading channel. This multidimensional modulation scheme is essentially uncoded and enables one to trade diversity for system complexity, at no power or bandwidth expense. Index Terms - Diversity, fading, lattices, number fields, QAM modulation, rotation.

This paper presents an adaptive-speed/CAV algorithm which regulates the disc velocity in a CD-ROM system to accomplish high data throughput (bit rate) during continuous playback and low power consumption during continuous data access. The adaptive-speed profile is based upon quasi-CLV control [1] which allows drive optimization towards both data throughput and access time. The presented algorithm can automatically distinguish between two operating modes - continuous access and continuous playback. By detecting the state of continuous access, the CD-ROM drive switches its spindle-motor loop into a constant angular velocity (CAV) mode, avoiding therefore the acceleration/braking of the motor, and thus reducing the power consumption. By detecting the state of continuous playback, the spindle motor will be set to follow an adaptive-speed profile which is highly optimized for data transfer rate. The software and hardware implementation of the CAV/quasi-CLV switching technique in a CD-ROM system is also described. Keywords - access time, adaptive-speed CD-ROM drive, bit rate, CAV control, CD-ROM, CLV control, data throughput, quasi-CLV control, seek time.

We examine the design of a communication link involving the data transfer from a small, low-orbit satellite to a ground station, but through a geostationary satellite. The advantage of this approach is that a single ground station, which tracks only the geostationary satellite, may be shared by a multiplicity of small satellites. Our goal is to select certain small satellite parameters - signaling rate, small satellite antenna beamwidth, modulation scheme and coding scheme - which maximize the data throughput in bits/day. Our approach uses orbital simulations of this scenario together with theoretical analysis based on the channel capacity C and cutoff rate [R.sub.0]. The throughput achievable by several practical coding schemes is also examined. Index Terms - Capacity, cutoff rate, GEO satellites, LEO satellites, time-varying channels, time-varying codes.

Researchers examined methods of minimizing subtask execution times and communication times in a heterogeneous computer network. The subtasks in question are the routines which enable an application program to run on the different machines in the system, and the duration of communication times between machines can have a significant effect on overall system performance. The researchers developed a mathematical framework to represent the processes involved, and derived an algorithm to give the best scheduling and data relocation times for any subtask/machine matching.

Two media fabrication methods are developed in order to improve the data-transfer rate of optical disks. One is a beam-scanning method for multispiral pregroove recording. A sixfold spiral pregroove with a 1.6-micrometer spacing can be recorded with this method. The other is a method for fabrication of phase-change optical-disk media that provides signal reading at 670 nm, writing at 830 nm, and high erasability at high linear velocity. The linear recording density of disks fabricated in this way is 1.2 times higher than that of optical disks read at 830 nm, and the erasability is more than 25 dB at 40 m/s. These methods offer the possibility of a data transfer rate 10 times higher than with conventional methods.

High Definition Television (HDTV) is expected to be one of the major services offered in future Broadband Integrated Services Digital Networks (B-ISDN). Transmitting HDTV signals over digital networks requires sophisticated compression techniques to limit bit-rate requirements and to provide high-quality, reliable services to customers. To transmit image signals within a studio and between studios or broadcasting stations, multiple encoding and decoding of signals without visible degradation is required. In this paper, we describe a high-quality HDTV codec that can encode original 1.2 Gb/s HDTV signals at 600 Mb/s. A subband DPCM (Differential Pulse-Code Modulation) coding algorithm capable of distortion-free encoding is used. Input signals are decomposed into four bands in the horizontal and vertical directions. A new architecture of perfect reconstructive short-kernel subband filters that reduces subband signal entropy has been adopted. Three bands are coded with adaptive DPCM and one band is coded with PCM. The performance of the codec is evaluated using true video signals. Experimental results show that a very high percentage of samples of component signals can be transmitted distortion free. A Signal-to-Noise Ratio (SNR) of over 45 dB is obtained with signals transmitted after the quantization process. Degradation after nine encoding and decoding processes is only 2 dB in terms of SNR.

A flexible layered coding scheme using block-based DCT domain decimation has been simulated in software. A conditional replacement scheme which switches adaptively between the pyramid and subband approaches at the coefficient level was studied. Experimental results comparing the adaptive scheme against the pyramid approach show that this new approach increases the overall coding efficiency of the scalable coder by up to 1 dB.

A noiseless ideal low-pass filter, followed by a limiter, is normally used as a binary data channel by sampling the output once per Nyquist interval. Detectors that sample more often encounter intersymbol interference, but can be used in ways that increase the information rate.

Traditional packet switching networks have typically employed window-based congestion control schemes in order to regulate traffic flow. In broadband networks, the high speed of the communication links and the varied nature of the carried traffic make such schemes inappropriate. Therefore, simpler and more efficient schemes have to be proposed to fully exploit the large available bandwidth. These schemes usually operate through input rate regulation. Typically, they force the information sources to limit their average input rate below some predefined rate while still allowing for a certain degree of burstiness. This ensures that no source will exceed for an extensive period of time the rate provided by the network during the call-setup procedure. The 'leaky bucket' scheme is an example of an input rate regulation. In this paper, input rate regulation schemes are extensively studied from the viewpoint of smoothing and regulating effects of the incoming traffic. The smoothing effect is characterized by the variance of the interdeparture time of the packet departure process from the input rate regulation mechanism. Under the assumption of Poisson arrivals the characteristics of this departure process are explicitly derived in terms of scheme's parameters and the tradeoff between the smoothness of the departure process and packets waiting time is studied. We present results for both finite and infinite buffer pool sizes.

Under a TDMA (time-division multiple-access) scheme, a station shares a multiple-access communications channel by transmitting its messages during its dedicated time slots. In many networking (multiple-access, multiplexing, time-slot interchange switching, etc.) situations, each station is allocated a number of contiguous time slots during each time frame. In this paper, we present an exact result concerning the queue-size and message delay analysis of TDMA systems in which a station is allocated multiple consecutive slots per frame. We obtain the generating function of the system queue-size for a general independent arrival message process. Messages consist of a random number of packets, following a geometric distribution. We then derive an exact result for the generating function of the message delay for various common message arrival processes, as well as tight bounds for the mean message queue-size and delay. Our results are compared to previously derived approximations. We also prove that a slot allocation scheme which distributes station slots uniformly over the frame yields a message-delay lower bounds. Our results also apply to the analysis of time-shared reservation schemes.

A rate-distortion theorem for arbitrary (not necessarily stationary or ergodic) discrete-time finite-alphabet sources is given. This result, which provides the expression of the minimum [Epsilon]-achievable fixed-length coding rate subject to a fidelity criterion, extends a recent data compression theorem by Steinberg and Verdu. Index Terms - Arbitrary discrete sources, data compression, rate-distortion theory, Shannon theory.

To improve the data rate of embedded-servo disk drives, data head switching at a high rate is required. To meet this requirement, a control method of two-degree-of-freedom that compensates for the computation time delay and phase lag of filters has been developed. This method has reduced the head switching time by 50%.

Some of the recent research in the area of uniform verification of parameterized systems in 2004 is reviewed. The method of counter abstraction that offers simple liveness proofs for many parameterized systems, and novel methodologies of using counter abstraction to automatically verify that probabilistic parameterized systems satisfy their temporal specifications and probability are discussed.

This issue of the Bell Labs Technical Journal is devoted to Internet protocol (IP) quality of service (QoS) research having to do with tethered and wireless service provider networks.