The emergence of nomadic applications have recently generated a lot of interest in wireless network infrastructures which support multimedia services. In this paper, we propose a bandwidth routing algorithm for multimedia support in a multihop wireless network. This network can be interconnected to wired networks (e. g. ATM or Internet) or stand alone. Our bandwidth routing includes bandwidth calculation and reservation schemes. Under such a routing algorithm, we can derive a route to satisfy bandwidth requirement for quality-of-service (QoS) constraint. At a source node, the bandwidth information can be used to decide to accept a new call or not immediately. This is specially important to carry out a fast handoff when interconnecting to an ATM backbone infrastructure. It enables an efficient call admission control. The simulation results show that the bandwidth routing algorithm is very useful in extending the ATM virtual circuit service to the wireless network. Different types of QoS traffic can be integrated in such a dynamic radio network with high performance.

A per-connection end-to-end call admission control (CAC) problem is solved in this paper to allocate network resources to an input session to guarantee its quality of service (Qos) requirements. In conjunction with the solution of the CAC problem, a traffic descriptor is proposed to describe the loss rate and the delay bound Qos requirements of the connection to be set up as well as the statistical characteristics of the associated input traffic which is modeled as a linear mean function plus a (zero-mean) fractional Brownian motion. The information in the traffic descriptor is sufficient to determine the allocation of channel bandwidth and buffer space to the input traffic in a network which employs leaky bucket shapers and scheduling algorithms to guarantee the Qos requirements. The CAC problem is solved by an iterative algorithm of which there are two stages in each iteration: one is responsible for the search of a candidate end-to-end routing path and the other for the verification of the legitimacy of this candidate path to meet the Qos requirements and for the allocation of resources in such a legitimate path.

In this paper, we analytically study the effects of overlap and overlay structure on the quality of service (QoS) of Low Earth-Orbital Satellite (LEOS) communication systems. We consider two-layered overlay of cells and intentional overlap of neighboring small cells. In order to measure the QoS, the probabilities of rejection of a newly arrived call (blocking) and forced termination due to failure of a handover (call dropping) are derived. In addition to these measures, the largest traffic intensity which guarantees the required blocking and dropping probabilities is also used.

Future high-speed switches and routers will be expected to support a large number of ports at high line rates carrying traffic with diverse statistical properties. Accordingly, scheduling mechanisms will be required to handle Tbit/sec aggregated capacity while providing quality of service (QoS) guarantees. In this paper a novel high-capacity switching scheme for ATM/WDM networks is presented. The proposed architecture is contention-free, scalable, easy to implement and requires no internal "speedup. " Non-uniform destination distribution and bursty cell arrivals are examined when studying the switching performance. Simulation results show that at an aggregated throughput of 1 Tbit/sec, low latency is achieved, yielding a powerful solution for high-performance packet-switch networks.

Combined input output queued (CIOQ) architecture such as crossbar with speedup has recently been proposed to build a large capacity switch for broadband integrated services networks. It was shown that, for a speedup factor of 2, a CIOQ switch can achieve 100% throughput with a simple maximal matching algorithm. Achieving 100% throughput, however, is not sufficient for per-connection quality of service (QoS) guarantee. In [2],[3], it is proved that a CIOQ switch with a speedup factor of 2 can exactly emulate an output queued (OQ) switch if stable matching is adopted. Unfortunately, the complexity of currently known algorithms makes stable matching impractical for high-speed switches. In this paper, we propose a new matching algorithm called the least cushion first/most urgent first (LCF/MUF) algorithm and formally prove that a CIOQ switch with a speedup factor of 2 can exactly emulate an OQ switch which adopts any service discipline for cell transmission. A potential implementation of our proposed matching algorithm for strict priority service discipline is also presented.

This paper presents a new distributed scheduling architecture for wireless ATM networks. Usually, in WATM scheduling architectures, a fixed order is defined among the different connections through their ATM service category (CBR VBR ABR UBR). We argue that although this static priority is easier to implement, this type of precedence is not necessarily a good choice for the MAC layer. The MAC layer scheduling should define an order such that it uses efficiently the resources while providing quality of service (QoS) guarantees. In this spirit, our architecture delays (without violating their QoS) the real time connections in order to improve the performance of non real time connections.

This paper describes the effects of traffic distributions on uplink and downlink communications qualities in CDMA cellular systems. Many researches have been done from the viewpoint of the system capacity under ideal conditions in both uplink and downlink. However, there are few studies regarding traffic distributions that concurrently affect the uplink and downlink quality. The characteristics in both links are different even in a spatially uniform traffic distribution because the system structures are not symmetric between both links. When non-uniform radio environments are assumed, both link qualities become very different from each other. It is therefore important to design systems in consideration of link-specific characteristics in whole service area. This paper clarifies the difference in both link characteristics in CDMA systems regarding traffic distributions.

In this paper, we present a traffic scheduling algorithm, called the Delay-Bound Monotonic with Average Rate Reservation (DM/ARR), which generates minimum output burstiness streams. We assume that connection i is policed by the leaky bucket algorithm with parameters (σi,ρi) where σi is the bucket size (or burstiness) and ρi is the leaky rate. Compared with the totally isolated scheme where connection i is allocated a bandwidth ri=max{σi/di,ρi} (di is the delay bound requirement of connection i), the DM/ARR algorithm has a better performance in the sense that it has a larger admission region. We prove that, among all possible scheduling algorithms that satisfy the delay bound requirements of established connections, DM/ARR results in the minimum output burstiness. This is important because a smaller burstiness implies a smoother traffic and thus the receiver (or next switch node in a multihop network) can handle it more easily. Numerical results show that the admission region of the DM/ARR algorithm is close to that of the earliest deadline first algorithm. A packetized version is studied for ATM networks.

We study some sufficient conditions of codeword lengths for the existence of a fix-free code. Ahlswede et al. proposed the 3/4 conjecture that Σi=1n a-li 3/4 implies the existence of a fix-free code with lengths li when a=2 i. e. the alphabet is binary. We propose a more general conjecture, and prove that the upper bound of our conjecture is not greater than 3/4 for any finite alphabet. Moreover, we show that for any a2 our conjecture is true if codeword lengths l1,l2,. . . consist of only two kinds of lengths.

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.

After a brief discussion of the demands in meshing for semiconductor process and device simulation, we present a three-dimensional Delaunay refinement technique combined with a modified advancing front algorithm.

In this paper, we propose and evaluate the performance of Wireless ATM MAC layer protocol to support efficiently various ATM traffics, such as CBR, VBR, ABR and UBR, in wireless ATM network environments for reverse and forward link. The proposed MAC protocol could extend efficiently the service discipline of ATM traffics from wired network to wireless ATM network environments. Thus, available bandwidth, which is remained except the bandwidth for CBR and VBR traffics, could be effectively allocated to ABR and UBR traffics. Especially, in view of reverse link, two-phase scheduling algorithm supports successfully variable characteristics of VBR traffic. And, in view of forward link, 'Wireless Dynamic Weighted Earliest Deadline First' scheduling algorithm minimizes the mean cell delay and required buffer size. Simulation results show that proposed method provides effective performance in wireless ATM environments.

In this paper we present two traffic control approaches, a circuit emulation traffic control (CETC) and an adaptive priority traffic control (APTC) for supporting voice services in ATM networks. Most voice services can be handled as CBR traffic, this causes a lot of wasted bandwidth. Sending voice through VBR (variable bit rate) may be a better alternative, because it allows the network to allocate voice bandwidth on demand. In CETC, the service discipline guarantees the quality of service (QOS) for voice circuits. Through mathematical analysis, we show that CETC features an adequate performance in delay-jitter. Moreover, it is feasible in implementation. We also present an APTC approach which uses a dynamic buffer allocation scheme to adjust the buffer size based on the real traffic need, as well as employs an adaptive priority queuing technique to handle various delay requirements for VBR voice traffic. It provides an adequate QOS for voice circuits in addition to improving the multiplexing gain. Simulation results show that voice traffic get satisfied delay performance using our approaches. It may fulfill the emerging needs of voice service over ATM networks.

With the introduction of ATM technology, service providers around the world have actively engaged in offering high bandwidth services. Currently, services, such as T1/E1, T3/E3 circuit emulation, are made available to large-volume account users. However, more advanced services, such as multimedia applications, have demanded not just high bandwidth but also flexible rate adaptation with quality-of-service (QoS) guarantee. To support the above service requirements, sophisticated network planning and engineering procedures should be taken. In the past few years, we have conducted various researches on developing the engineering strategies for resource control and management to support multi-rate service offering. We have also looked into the design details of connection control and management for achieving the QoS requirement. We considered the service quality of the underlying transport in regard with the QoS management. In this paper, we will outline those results and give an overview description about the proposed framework.

As the current Internet becomes popular in information access, demands for real-time display and playback of continuous media are ever increasing. The applications include real-time audio/video clips embedded in WWW, electronic commerce, and video-on-demand. In this paper, we present a new control protocol R3CP for real-time applications that transmit stored MPEG video stream over a lossy and best-effort based network environment like the Internet. Several control mechanisms are used: a) packet framing based on the meta data; b) adaptive queue-length based rate control scheme; c) data preloading; and d) look-ahead pre-retransmission for lost packet recovery. Different from many adaptive rate control schemes proposed in the past, the proposed flow control is to ensure continuous, periodic playback of video frames by keeping the receiver buffer queue length at a target value to minimize the probability that player finds an empty buffer. Contrary to the widespread belief that "Retransmission of lost packets is unnecessary for real-time applications," we show the effective use of combining look-ahead pre-retransmission control with proper data preloading and adaptive rate control scheme to improve the real-time playback performance. The performance of the proposed protocol is studied via simulation using actual video traces and actual delay traces collected from the Internet. The simulation results show that R3CP can significantly improve frame playback performance especially for transmission paths with poor packet delivery condition.

The performance of AC plasma displays has been improved in the area of brightness and contrast, while significant advances in image quality are still required for the HDTV quality. In particular, in full color motion video, motion artifacts and lack of color depth are still visible in some situations. These motional artifacts are mitigated as the number of the subfields increases, usually at the cost of losing brightness or increasing driving circuitry. Therefore, it is still one of our great concerns to find out the optimized subfield configuration through weighting and order of each subfield, and their coding of combination. For evaluation and improvement of motion picture disturbance, we have established a procedure that fully simulates the image quality of displays which utilize the subfield driving scheme. The simulation features virtually located sensor pixels on human retina, eye-tracking sensor windows, and a built-in spatial low pass filter. The model pixelizes the observers retina like a sensor chip in a CCD camera. An eye-tracking sensor window is assigned to every light emission from the display, to calculate the emissions from one to four adjoining pixels along the trajectory of motion. Through this model, a scene from original motion picture without disturbance is transformed into the still image with simulated disturbance. The integration of the light emission from adjoining pixels through the window, also functions as a built-in spatial low pass filter to secure the robust output, considering the MTF of the human eye. Both simulation and actual 42-in-diagonal PDPs showed close results under various conditions, showing that the model is simple, but reasonable. Through the simulation, general properties of the subfield driving scheme for gray scale have been elucidated. For example, a PWM-like coding offers a better performance than an MSB-split coding in many cases. The simulation also exemplifies the motion picture disturbance as a non-linear filter process caused by the dislocation of bit weightings, suggesting that tradeoffs between disturbance and resolution in motion area are mandatory.

Low bit-rate speech and audio codings are key technologies for multimedia communications. A number of coding scheme have been developed for various applications. In Internet application, good speech and audio quality at very low bit-rate (8-16 kb/s) is valuable. Two recently proposed speech and audio-coding schemes, CS-ACELP (Conjugate Structure Algebraic Code Excited Linear Prediction, standardized by the ITU-T in Recommendation G. 729) and TwinVQ (Transform-domain Weighted INterleave Vector Quantization, one of the candidates for MPEG-4 audio) were compared from the viewpoints of coding schemes and quality. Although there are significant differences in their basic structures and frame lengths, this paper describes that both use the same compression techniques, such as LPC (Linear Predictive Coding)-analysis pitch-period estimation and vector quantization. While CS-ACELP provides toll quality for speech at 8 kb/s, the quality it provides for music signals is insufficient. The TwinVQ transform coder is based on LPC and vector quantization and is also capable of operating at 8 kb/s. Evaluation of these two schemes in terms of their fundamental technologies, quality, delay, and complexity showed that the quality of TwinVQ for music signals is better than that of CS-ACELP, and that the quality of CS-ACELP is better for speech signals. Therefore, TwinVQ may be better suited for one-directional Internet applications, and CS-ACELP may be better for two-directional communication.

Packet scheduling is one of the key mechanisms that will be employed in the network nodes (routers and switches) for supporting multiple quality of services. In this paper we propose a new packet scheduling algorithm called Urgency-based Round Robin (URR) which computes an index for flows in order to keep track of instantaneous bursts. Basically the index is employed as a measure of the time-dependent service necessity for each flow thus making it possible to detect those flows which might be in need of momentary service. Also, we propose a novel weight allocation scheme to be used together with the scheduler with the aim of preventing network underutilization. Our algorithm can be considered as a version of Weighted Round Robin (WRR) with improved delay characteristics. We show analytically that URR has the desired capability of upper-bounding unfairness. We also show, by simulation, that URR can improve delay performance even under extremely bursty traffic conditions without bandwidth overprovisioning. We also give simulation results for network traffic which exhibits long range dependency (self-similarity) and show that URR is again more effective than a plain round robin multiplexer.

This paper presents the subjective speech quality evaluation in terms of the Mean Opinion Score (MOS) and the relationship between BER and subjective speech quality in a GSM-based radio system. The results show that in certain environments (hilly terrain and rural areas), a SNR (or C/I) higher than 12 dB is required for acceptable speech quality. For an acceptable speech quality, a BER(c1) better than 10-2 is needed in a GSM-based system.

In the design of nonlinear reliable controllers, one major issue is to solve for the solutions of the Hamilton-Jacobi inequality. In general, it is hard to obtain a closed form solutions due to the nonlinear nature of the inequality. In this paper, we seek for the existence conditions of quadratic type positive semidefinite solutions of Hamilton-Jacobi inequality. This is achieved by taking Taylor's series expansion of system dynamics and investigating the negative definiteness of the associated Hamilton up to fourth order. An algorithm is proposed to seek for possible solutions. The candidate of solution is firstly determined from the associated algebraic Riccati inequality. The solution is then obtained from the candidate which makes the truncated fourth order polynomial of the inequality to be locally negative definite. Existence conditions of the solution are explicitly attained for the cases of which system linearization possesses one uncontrollable zero eigenvalue and a pair of pure imaginary uncontrollable eigenvalues. An example is given to demonstrate the application to reliable control design problem.