The concept of a schedulable region (SR) was introduced to characterize the capacity of a multiplexer and provide a separation between call-level and cell-level phenomena. In this paper, we present a framework and algorithm for real-time estimation of the schedulable region. A major problem associated with online estimation is that the objects of measurement are not fixed in the presence of call arrivals and departures. The invariance property is exploited to carry out measurements in the presence of call arrivals and departures. By virtue of it, the equivalent bandwidth could be defined on the condition of the number of each traffic class call in progress. Another important thing we consider here is that the search algorithm to estimate the effective bandwidth should be chosen depending on the arrival statistics and QOS constraints. The algorithms presented here have been implemented on an ATM switch.

A typical user is concerned only with the quality of service of a network on an end-to-end basis. Therefore, how end-to-end requirements are mapped into the local switching node requirements and maximum network utilization is a function of network internal design. In this paper, we address the problem of QOS allocation. We derived an optimal QOS allocation policy and decided the maximum utilization bound in a deterministic traffic model. We adopted the worst case delay bound as the end-to-end and local QOS requirement. With (σ, ρ) traffic model, we derived a formula for delay bound and the number of connections. We found that with the delay bound as the QOS metric, there is a significant difference in the performance of allocation policies. We also developed an evaluation strategy to analyze allocation policies. The numerical results for two simple network topologies: tandem network model and uneven traffic load model, compare the equal allocation policy with the optimal allocation policy and show the correctness and efficiency of QOS allocation policy.

The MPEG video coding is the most widely used video coding standard which usually generates variable bitrate (VBR) data streams. Although ATM can deliver VBR traffic, the burst traffic still has the possibility to be dropped due to network congestion. The cell loss can be minimized by using an enforced rate control method. However, the quality of the reproduced video may be sacrificed due to insufficient peak rate available. In this work, we propose an end-to-end quality adaptation mechanism for MPEG traffic over ATM. The adaptive quality control (AQC) scheme allocates a certain number of coding bits to each video frame based on the network condition and the type of next frame. More bits may be allocated if the network condition, represented by the connection-level, is good or the next frame is B-frame that usually consumes fewer bits. A high connection-level allows a relatively large number of tagged cells, which are non-guaranteed in delivery, for video frames with high peak rates. The connection-level adjustment unit at the encoder end adjusts the connection-level based on the message of the network condition from the quality monitoring unit at decoder. The simulation results show that the AQC system can effectively utilize the channel bandwidth as well as maintain satisfactory video quality in various network conditions.

A partial buffer sharing scheme is proposed as loss-priority control for a finite buffer with batch Poisson inputs under a whole batch acceptance rule. Customer and batch loss probabilities for high- and low-priority customers are derived under this batch acceptance rule using a supplementary variable method. A comparison of the partial buffer sharing scheme and a system without loss-priority control is made in terms of admissible offered load. Whole batch acceptance and partial batch acceptance rules are also compared in terms of admissible offered load.

A new asynchronous transfer mode adaptation layer (AAL), called AAL2, is being designed mainly for low-bit-rate voice traffic, and nodes that can assemble and disassemble AAL2 cells are being developed to make AAL2 usage efficient. This paper investigates the delay and performance of AAL2 nodes by an analytical method. Then, using the results, it analyzes a network using AAL2 nodes and shows the bandwidth reduction achieved by using AAL2 switching nodes as transit nodes.

Shared multibuffer ATM switches are attractive since they can extend memory bandwidth by the use of multiple independent buffer memories. Although the parallel accessibility allows a considerable improvement in memory bandwidth, a proper assignment of memory addresses to cells is necessary to better utilize the potential bandwidth. In this paper, we present an efficient cell placement strategy for shared multibuffer ATM switches. It is based on a combination of two key concepts, uniform distribution for writes and reference locality for reads. The former is to reduce cell loss ratio due to overflow and write-access conflicts. The latter is to have cells destined for the same output port read from the same buffer memory to minimize read-access conflicts. A single threshold is employed to assign memory locations adaptively depending on the cell distribution among the shared buffer memories. The proposed strategy is shown to outperform the existing ones, in terms of cell loss ratio, cell delay, and throughput. Moreover, the performance gains have been made with a simple control circuit.

This paper presents the generator polynomial matrices and the upper bound on the constraint length of punctured convolutional codes (PCCs), respectively. By virtue of these properties, we provide the puncturing realizations of the good known nonsystematic and systematic high rate CCs.

A novel multiple-access optical network architecture is presented that not only employs the WDM technique but also divides networks. The subnetworks are connected to each other via a wavelength-dependent interconnection network, and pairs of subnetworks are optically linked with different combinations for each wavelength. Through an analysis of the throughput and delay for the slotted ALOHA protocol, the architecture is confirmed to be improved from the conventional one that employs only the WDM technique. For example, the improvement ratio of the throughput for a four-wavelength network is 2.4, and that for an eight-wavelength network is 4.4.

This paper proposes a novel universal line termination scheme for the ONUs (optical network units) of fiber-optic local access systems. Its main feature is that only low cost AD/DA converters for Hi-Fi audio are needed. Because audio AD/DA converters are insufficient for ISDN basic rate access (● 320kbaud) and cause waveform distortion, we develop a simple detection algorithm that does not use any equalizing filter. The algorithm can handle plural channels with one general purpose MPU (micro-processing unit). Based on this, a novel architecture for a fiber-optic local access system is presented that removes the MPUs from each optical network unit (ONU) and places them in the central office (CO). The proposed system yields a small, service-uniform ONU that supports a wide range of narrow-band services (POTS & ISDN) with no distinction. To realize this system at the lowest possible cost, a high-speed code division multiplexing (CDM) scheme with novel code word sets is developed.

In digital modulation for mobile radio telephone services frequency modulation with continuous phase with small modulation indices (MSK/GMSK) is sometimes used. Extension of the synchronization subsystems' pulling band in a coherent receiver and reducing synchronization delay is important for the mobile communication. At this moment there are only two possible synchronization schemes for the coherent MSK/GMSK receiver: Costas and de Buda's. This paper presents a new method (a possible alternative to both of them) where the frequency discriminator with decoupled carrier and bit synchronizing subsystem are combined to handle the task. For comparison, this paper also describes performances of the Costas carrier recovery scheme, which is widely employed for MSK/GMSK coherent demodulation. Discrimination and fluctuation characteristics for frequency, phase, and symbol delay synchronization subsystems are shown and the BER degradation from the conventional Costas scheme is calculated. This paper demonstrates with simulation results that the proposed scheme improves RF carrier acquisition performances, and at the same time, for large signal-to-noise ratios (SNR's) provides similar or better tracking performances than the Costas one. While limited to higher SNR ratios, the proposed synchronization scheme is suitable for many applications and can be implemented with simpler circuitry, well suited to integrated circuit implementation.

A RAKE receiver accomplishing joint blind multipath diversity combining and PN code timing recovery is proposed for direct-sequence spread-spectrum signaling over a frequency-selective fading channel. In this technique, an improved known modulus adaptive algorithm is exploited to perform multipath diversity combining in the blind mode, while a modified PN code timing recovery technique based on the timing error estimator extracts the finger error signals path by path independently. Taking the advantage of inherent diversity, this modified PN code timing recovery technique can efficiently combine finger error signals to avoid the problems with the drift or flutter effects in the timing error signals, and thus provide better code tracking performance as well. Extensive computer simulation results have verified the analysis and indicated very attractive performance of the proposed technique.

To support efficiently the high-speed service and the multimedia service in mobile communication environments, network architecture is generally based on small cell radius structure to increase the wireless channel capacity. In a small cell radius structure based mobile network environment, a mobile station with large mobility should perform frequent handover. Thus, advanced handover mechanism should be provided, and must be done fastly. For these reasons, we suggest the efficient handover mechanism based on forward handover method in this paper. Proposed handover mechanism could support fast handover procedure, and is performed by using optimized path. Also, proposed handover mechanism could support QoS (Quality of Service) attribute of multimedia traffic during handover procedure. We evaluated the performance of proposed handover mechanism, and analysis results show that the suggested forward handover mechanism has less blocking ratio and lower delay than conventional backward handover mechanism.

We propose and analyze a traffic model of a cellular radio communication network with an arbitrary cell connection and arbitrary probabilistic movement of mobiles between the cells. Our analytic model consists of birth-and-death processes for individual cells connected by the numerical adjustment of hand-off rates. This approximation is validated by simulation. We evaluate the probabilities of the immediate loss, the completion, and the forced termination during hand-off for an arbitrary call in the network. Our numerical examples reveal the cases in which the increase in the generation rate of new calls results in the increase in the loss probability without affecting much the probability of forced termination in a limited service area.

Adaptive channel estimation filters are presented for coherent DS-CDMA reverse link using time-multiplexed pilot and parallel pilot structures. Fast transmit power control (TPC) is adopted in the reverse link. Fading statistical properties are not preserved when fast TPC is used. When fading is slow, the channel is similar to non-fading channel, but its starts to vary as fading become faster since fast TPC cannot track fading perfectly. A pragmatic approach is used in this paper to derive adaptive channel estimation filter. The filter coefficients are updated based on the measured autocorrelation function of the instantaneous channel estimate. The bit error rate (BER) performance under frequency selective Rayleigh fading is evaluated by computer simulation to show that the adaptive channel estimation filter provides superior performance to the previously proposed non-adaptive WMSA filter.

In this paper, we propose a polling-based scheduling strategy for multiple access control in wireless ATM networks (POSTMAN). A pure centrally controlled polling scheme is adopted in our protocol, therefore no contention channel is needed. The POSTMAN protocol assumes a TDMA/TDD frame format, by which wireless bandwidth is allocated flexibly both among multiple mobile terminals and between downlink and uplink channels. When polling the uplink traffic, the POSTMAN needs no priori traffic model to predict the cell arrivals. Instead, a token queue and an ABR buffer status table are used in the base station. Simulation results show that the POSTMAN protocol is robust in most cases and can work steadily under very high network load conditions.

For the reduction of the jitter originated from the cell losses in ATM network when CBR traffic is transferred on AAL5, we propose that the receiver maintain a timer whose expiration time is proportional to the cell time of the source traffic plus the standard deviation of the 1-point CDV of the received ATM cells. Moreover, to enhance the granularity of the error or loss detection mechanism in the AAL5 PDUs, we also modified the AAL5 PDU trailer fields so that each cell comprising the AAL5 PDU has a sequence number field. The simulation results show that the peak-to-peak PDV of the AAL5 PDU by the proposed method is less than 69.4% to that by AAL5. Moreover, the AAL5 user receives the same or more error-free transport packets in the proposed algorithm than those in the ITU-T AAL5 for the same network simulation environment.

This letter proposes an integrated approach to multimedia ATM network design. An optimization model that combines the physical layer design with the logical layer design is developed. A key feature of the model is that the objective to be maximized is a profit function. It includes more comprehensive cost functions for the physical and logical layers. A simple heuristic algorithm to solve the model is presented. It should be useful in practice for network designers and operators. Some numerical examples are given to illustrate the application of the model and the algorithm.

We performed 10 Gbit/s optical soliton transmission experiment over 2,000 km with bit error rate of < 10^-9 in a comb-like dispersion profiled fiber (CDPF) loop of 80 km amplifier spacing which corresponds to 1.8 times of dispersion distance. By reducing the average dispersion of the CDPF, error free distance of 3,000 km was obtained.

Comparison of TDM and WDM for backbone ring network application is discussed from the perspective of system economizing. A critical advantage that WDM can provide is the optical pass-through function at a node having a relatively small drop and insert bandwidth. Circumstances where WDM is more advantageous than TDM are frequent especially in center-node type ring networks.

In this paper, performance degradation of satellite broadcasting receiving antenna systems due to weather conditions is examined by measuring their G/T continuously. We show that an offset parabolic reflector antenna of smaller aperture tends to be less affected by weather conditions.