Recently, as multimedia and high-speed traffic become more popular on the Internet, the various traffic requiring different qualities of service (QoS) will co-exist. In addition, classified service based on Diff-Serv (Differentiated Service), MPLS (Multi-Protocol Label Switching),etc., have come into wide use. Today's Internet environment requires routers to perform control mechanisms in order to guarantee various QoSs. In this paper, we propose a buffer management scheme for the Internet router that uses class-based priority control. This paper focuses on per-flow queueing, and evaluates the performance of the proposed buffer management scheme. Realization of differentiated services and dissolution of buffer occupation by specific flow is expected by the proposed control.

A new objective speech quality measure, Bark Coherence Function is presented. The Coherence Function was used for evaluating the non-linear distortion of low-to-medium rate speech coders. However, it is not well suited for quality estimation in modern speech transmission, especially, CDMA mobile communication system. In the proposed method, Coherence Function is newly defined in psycho-acoustic domain as the cognition module of perceptual speech quality measure and evaluates the perceptual non-linear distortion of mobile system. The experimental results showed that the proposed method has good performance over CDMA PCS and digital cellular system.

A quality-of-service based link control scheme to counteract correlated channel errors for wireless multimedia communications is proposed in this paper. Both the medium access (MAC) and data link control (DLC) layers are treated. The performance of the proposed scheme is evaluated using both analysis and simulation. The delay and jitter behaviors are examined for both the constant bit rate (CBR) traffic and variable bit rate (VBR) traffic. The throughput performance is also obtained for the available bit rate (ABR) traffic. Through numerical experiments, the proposed scheme is demonstrated to be not only robust against channel impairments but also capable of providing the desired QoS for wireless multimedia communications.

The IETF Differentiated Services (DiffServ) framework achieves scalability by (1) aggregating traffic flows with coarse grain QoS on the data plane, and (2) allocating network resources with a bandwidth broker (BB) on the control plane. However, there are many issues that need to be addressed under such framework. First, it has been shown that the concatenation of strict priority (SP) scheduler of class-based queues (CBQ) can cause delay jitter unbounded under certain utilization, which is not acceptable to support the premium service (PS). Furthermore, it is not clear how such a DiffServ network can support traffic flows requiring the guaranteed service (GS), which is a desirable feature of the future Internet. This paper presents architecture and mechanisms to support multiple QoS under the DiffServ paradigm. On the data plane, we present a node architecture based on the virtual time reference system (VTRS). The key building block of our node architecture is the core-stateless virtual clock (CSVC) scheduling algorithm, which, in terms of providing delay guarantee, has the same expressive power as a stateful weighted fair queueing (WFQ) scheduler. With the CSVC scheduler as our building block, we design a node architecture that is capable of supporting integrated transport of the GS, the PS, the assured service (AS), and the traditional best effort (BE) service. On the control plane, we present a BB architecture to provide flexible resource allocation and QoS provisioning. Simulation results demonstrate that our architecture and mechanisms can provide scalable and flexible transport of integrated traffic of the GS, the PS, the AS, and the BE services.

In the past, a number of scheduling algorithms that approximate GPS, such as WFQ, have been proposed and have received much attention. This class of algorithms provides per-flow QoS guarantees in terms of the bounded delay and minimum bandwidth guarantee. However, with O(log N) computational cost for each new arrival scheduling, where N is the number of backlogged flows, these algorithms are expensive to implement (e.g., in terms of scalability). Moreover, none of them addresses the issues of delay distribution and jitter. In this paper, we propose a new traffic scheduling discipline called Jitter Control Frame-based Queueing (JCFQ) that provides an upper bound for delay jitter in the case of rate-controlled connections, such as packet video streams and IP telephony, while guaranteeing bounded delay and worst-case fair weighted fairness, such as in the WF2Q algorithm, but with O(1) complexity in selecting the next packet to serve, assuming that the number of flows is fixed. Three different algorithms for slot or service order assignment between flows are proposed: Earliest Jitter Deadline First (EJDF), Rate Monotonic (RM) and Maximum Jitter First (MJF). In these algorithms, delay jitter is formulated into the virtual finish time calculation. We compare the fairness, delay and jitter performance of the JCFQ with that of the MJF algorithm with WF2Q via simulation. The results show that with proper choice of the slot size, JCFQ can achieve better flow isolation in delay distribution than can WF2Q.

In this paper we study the scalability issue in the design of a centralized bandwidth broker model for dynamic control and management of QoS provisioning. We propose and develop a path-oriented, quota-based dynamic bandwidth allocation mechanism for efficient admission control operations under the centralized bandwidth broker model. We demonstrate that this dynamic bandwidth allocation mechanism can significantly reduce the overall number of QoS state accesses/updates, thereby increasing the overall call processing capability of the bandwidth broker. Based on the proposed dynamic bandwidth allocation mechanism, we also extend the centralized architecture with a single bandwidth broker to a hierarchically distributed architecture with multiple bandwidth brokers to further improve its scalability. Our study demonstrates that the bandwidth broker architecture can be designed in such a manner that it scales with the increase in the network capacity.

Emerging multimedia technologies introduce the prevalent multicast transmission, and the multicast tree is determined using the time-invariant network parameters. This paper addresses the time-varying multicast tree problem and presents path selection heuristics for multicast routing to determine an alternative path for real-time applications. A network is partitioned into the optimal region, the disjoint region, and the edge cutset if a branch of the multicast tree meets the un-guaranteed QoS condition. The path selection heuristics operate during the multicast session phase to efficiently select an alternative routing path containing an edge in the edge cutset to connect the multicast tree again. The source-based heuristics PS-SPT finds the path for minimal source-to-destination delay and the sharing-based heuristics PS-DDMC for minimal total cost. These path selection heuristics can efficiently provide solutions to keep the multicast transmission reliable. Simulation results also show that the proposed heuristics can provide effective good solutions for real-time multicast transmission. PS-SPT can select a path with optimal source-to-destination delay and PS-DDMC can select a path with optimal total cost.

A completely symmetric dual-level spiral inductor structure is proposed. The symmetry, area efficiency, the size dependence of the coupling factor, and the quality factors of the dual-level inductors are evaluated and compared with that of the single-level. This work demonstrates that, with most RF applications, the dual-level inductors are the better choice than the single-level.

One drawback of Integrated service architecture is the scaling problem. Therefore, flow aggregation is an important solution for supporting quality of service in large-scale network. In advance reservation, priori information of advance-reserved requests can be used for flow aggregation before their initiation time. However, an impolitic aggregation can lead to violate admission control. In this paper, we propose an effective algorithm to aggregate advance-reserved requests with guaranteed delay. The proposed algorithm not only can reduce the amount of state in core network but also minimize the bandwidth consumption. The simulation result indicates that the state in the core network can be reduced as low as 17.3% even in the worst case.

This paper suggests the way to perform the handover by predicting the movement route of the mobile terminal by considering the movement pattern of the user. By considering the fact that the most users has the constant movement pattern, the channels needed for the handover can be reserved, and the required quality of service (QoS) is maintained during handover. The suggested algorithm makes the channel allocation schemes more efficient.

The differentiated services (diffserv) architecture has been proposed for implementing scalable service differentiation in the Internet. Expedited forwarding and assured forwarding have been standardized as Per-Hop Behaviors (PHB) in diffserv. Assured forwarding can be utilized to realize the service, which provides each user with a minimum guaranteed rate and a fair share of the residual bandwidth. We call it guaranteed rate (GR) service. With GR service, each packet for flow i is marked in or out based on comparison between the sending rate and the minimum guaranteed rate. When congestion occurs in networks, out packets are dropped more aggressively than in packets. Recently, several fair queuing schemes have been proposed for core stateless networks. They can achieve fairer bandwidth allocation than random early detection (RED). However, there have not been any studies that consider in/out bit usage to support GR service. This paper proposes how to extend the schemes that have been proposed for core stateless networks to allow the support of in/out bit usage. We present the performance of one of the extended schemes and compare the scheme to RED with in/out bit (RIO) in terms of fair bandwidth allocation.

We have developed a new scheme to provide Diffserv-based QoS over ATM access networks. Well-known Diffserv over ATM scheme requires some extension for conventional routers with ATM interfaces. The routers must map their Diffserv classes of services into ATM QoS classes and forward IP packets into prioritized VCs based on DSCP (DiffServ Code Point). The purpose of this work is to provide Diffserv-based QoS over ATM network using conventional IP over ATM interfaces on routers. We propose DSCP snooping at ATM edge nodes, which differentiates services over a single VC between two IP domains. A prototype circuit was used to evaluate this scheme.

We propose middleware which works on widely-used commercial off-the-shelf platforms (UDP/IP, FastEthernet, and Windows NT or commercial real-time kernels) to realize real-time distributed services for plant monitoring and control systems. It is not suitable to use TCP/IP for the systems because of its unpredictable re-transmission, while, as well known, UDP/IP does not guarantee certain arrivals of packets and it is also not acceptable for the systems. With UDP/IP, packets are lost mainly because of collisions in a network and buffer overflows. To avoid these packet losses, the middleware controls scheduling of all the packets transmitted between the nodes in a distributed system and prevents excessive collisions and buffer overflows. The middleware provides a necessary set of functions for plant monitoring and control applications. The middleware on each node in a distributed system consists of library functions and run-time modules. An application program on the node is required to use these library functions according to the rules the middleware provides. In this way the middleware can manage all the traffic among the nodes in the system. Receiving requests from the application via library functions, the run-time module of the middleware schedules transmission of messages to other nodes, avoiding unexpected delivery delays or buffer overflows. The module also guarantees application-to-application quality of service (QoS), such as transmission period and delivery deadline, required by the applications. This is achieved by assigning the resources not shared by other services to each distributed service and scheduling these resources so as not to violate the assignment. Here, resources include maximum numbers of packets which a node can receive or send in a specific period (20 msec, for example). We show implementation of the middleware to make it clear how to guarantee application-to-application QoS with some application examples.

Call admission control (CAC) is becoming vital for multimedia services in the ability of wireless/mobile networks to guarantee Quality of Service (QoS) partially due to the network's limited capacity. In this paper, we propose an optimal call admission control scheme with bandwidth reallocation algorithm (multi-class-CAC-BRA) for multi-classes of adaptive multimedia services in wireless/mobile networks. The multi-class-CAC-BRA approach optimizes revenue for service providers and satisfies QoS requirements for service users. The proposed approach adopts semi-Markov Decision Process to model both call admission control and bandwidth reallocation algorithm. In other words, whenever decisions are made, decisions are made for both call admission control and bandwidth reallocation. Since the non-adaptive multimedia traffic is a special case of the adaptive multimedia traffic, the non-adaptive optimal CAC scheme is a special case of our optimal multi-class-CAC-BRA scheme. Furthermore, the Interior-point Method in linear programming is used to solve the optimal decision problem. Simulation results reveal that the proposed multi-class-CAC-BRA scheme adapts itself well to adaptive multi-class multimedia traffic, achieves optimal revenue, and satisfies QoS requirements that are the upper bounds of handoff dropping probabilities. Our approach solves the optimal adaptive multimedia CAC problem. We believe that this work has both theoretical and practical significance.

Automatic repeat request (ARQ) for wireless ATM (WATM) operating at 20 Mbit/s or higher is required to achieve high throughput performance as well as high transmission quality, i.e., low CLR (cell loss ratio). Selective Repeat (SR) and Go-Back-N (GBN) are typical ARQ schemes. Though SR-ARQ is superior to GBN-ARQ in throughput performance, the implementation complexity of SR-ARQ's control procedures is disadvantageous to its application to high-speed wireless systems. In addition, when PDU (protocol data unit) length on wireless link is short, the capacity for ARQ control messages can be significantly large. GBN-ARQ, on the other hand, cannot avoid serious throughput degradation due to fairly high BER caused by multipath fading and shadowing, though its implementation is simple. To solve the above-mentioned problems, this paper proposes a novel ARQ scheme named PRIME-ARQ (Partial selective Repeat superIMposEd on gbn ARQ). PRIME-ARQ achieves high throughput performance, almost equal to selective repeat ARQ, with a simple algorithm resulting in reduced implementation complexity for high speed operation. This paper describes the design, implementation, and performance of the proposed PRIME-ARQ. In addition, it shows the experimental results using an experimental PRIME-ARQ hardware processor and proto-type AWA equipment.

It is always difficult to monitor stringent cell loss ratio (CLR), e.g. 10-9, in asynchronous transfer mode (ATM) networks, because its measurement period is too long for real-time measurement. In this paper, we propose new performance monitoring mechanisms for stringent CLR. We consider virtual ATM switches whose resources are much smaller than the real system and thus much higher CLRs will be obtained within a relatively short measurement period. By applying some regression methods in the CLRs obtained from the virtual system, we can estimate the actual CLR of the real system quite accurately and our performance monitoring mechanisms will be operated based on the estimation. Through the numerical examples, our mechanisms are not only more accurate than the traditional methods, but also have shorter measurement periods compared with direct measurement.

Several types of coplanar waveguide (CPW) resonator are analyzed by use of the field decay method based on the FDTD algorithm in this paper. Quality (Q) factors of a CPW resonator, including radiation Q, conductor Q, dielectric Q and unloaded Q, are investigated thoroughly. A new procedure to calculate conductor Q is proposed. Some CPW resonators are fabricated and measured in order to validate the analysis method. It is shown that radiation from a CPW resonator can be reduced by means of the stepped impedance technique. It is also seen that miniaturization of CPW resonators can be realized if the stepped impedance structure is adopted.

In this paper, we describe a method of determining meat quality using the concepts of "marbling score" and texture analysis. The marbling score is a measure of the density distribution of fat in the rib-eye region. Based on the results of an investigation carried out by handing out questionnaires to graders, we consider the marbling of meat to be a texture pattern and propose a method for the implementation of a grading system using a texture feature. In this system, we use a gray level co-occurrence matrix as the texture feature, which is a typical second-order statistic of gray levels of a texture image, and determine standard texture-feature vectors for each grade based on it. The grade of an unevaluated image is determined by comparing the texture-feature vector of this unevaluated image with the standard texture-feature vectors. Experimental results show the proposed method to be effective.

We present several challenging gridding problems for multi-dimensional device and process simulation and discuss how new strategies might contribute to their solution. Formulating grid quality requirements for the standard Scharfetter-Gummel box method discretization in device simulation, we demonstrate how the offsetting techniques compares with quadtree grid generation methods and how they apply to modern device designs. Further we present a grid adaptation approach which respects the grid quality criteria and touch upon the main adaptation difficulties within device simulation. For the 3D moving boundary grids in process simulation we present a new algorithm.

Since mobile communication systems using optical rays (optical mobile communication systems) do not radiate radio waves from the mobile terminals, they are expected to be used in environments containing sensitive electronic equipment. However, the placement and direction of the optical receivers must be suitably determined for mobile communication because light has high directivity. In optical mobile communication systems, the communication quality varies with the direction of the mobile terminal. Therefore, we examined the angle over which communication is possible at various measurement points and defined it as the communication angle. The mean opinion score (MOS) was obtained to assess the communication quality using the communication angle as a parameter. In this paper, the two situations, walking and sitting down, was considered the way optical mobile communication systems actually used. We found that for walking, when the communication angle was over 180 degrees, the MOS was over 3 and over 50% of users could communicate usefully. When used sitting down, the communication quality did not depend on the communication angle, but only on whether or not the user could communicate in the direction he/she was facing. Thus, if the communication angle in the service area is over 180 degrees, it is possible to communicate in practical situations, even while walking.