Next generation wireless networks are expected to support multimedia applications (audio phone, video on demand, video conference, file transfer, etc.). Multimedia applications make a great demand for bandwidth and impose stringent quality of service (QoS) requirements on the wireless networks. In order to provide mobile hosts with high QoS, efficient and better bandwidth reservation is necessary in multimedia wireless networks. This paper presents a novel hybrid dynamic-grouping bandwidth reservation scheme to support QoS guarantees in the next generation wireless networks. The proposed scheme is based on probabilistic resource estimation to provide QoS guarantees for multimedia traffic in cellular networks. We establish several reservation time-sections, called groups, according to the mobility information of mobile hosts (MHs) of each base station (BS). The amount of reserved bandwidth for each BS is dynamically adjusted for each reservation group. We use the hybrid dynamic-grouping bandwidth reservation scheme to decrease the connection-dropping probability (CDP) and connection-blocking probability (CBP), while increasing the bandwidth utilization. The simulation results show that the hybrid dynamic-grouping bandwidth reservation scheme provides less CDP and less CBP, and achieves high bandwidth utilization.

The block-based fast transmission scheme, which is one of typical stored video delivery schemes, is reasonable in terms of its bandwidth efficiency and tolerance to the delay jitter, etc. However, it causes packet loss because of its burst data transmission method. Thus, we propose a slotted multicast scheme for MPEG video based on the block transmission scheme to maintain a higher quality and to include time constraints. We define two delivery units, the "GoPs Group" and the "Frame Type," on the basis of the MPEG characteristics with periodical NACK feedback from the clients. The former is tolerant to burst packet loss, and the latter gives priority to important frames. Our block multicast has two phases: a "Transmission Phase" and a "Retransmission Phase." In the former, a server multicasts a block, and in the latter, a server retransmits lost packets using multicast according to the proper delivery unit. We evaluate our proposal from some viewpoints with a computer simulation. We also measure the quality of the video reflected the result of a computer simulation. From these results, we confirm performance effectiveness of our proposal.

Supporting quality of service (QoS) capabilities for multi-media applications is one of the major issues in medium access control (MAC) research. In distributed contention-based MAC algorithms, it is a challenging task to support the desired QoS because of the inherent random access characteristics. In this paper, we propose an efficient prioritized fast collision resolution algorithm. The MAC protocol with this new algorithm attempts to provide significantly high throughput performance for data services and support QoS for real-time services. We incorporate the priority algorithm based on service differentiations with the fast collision resolution algorithm, and show that this algorithm can simultaneously achieve high throughput and good QoS support for real-time and data services.

In this letter, we analyze blocking probabilities for prioritized multi-classes in optical burst switching (OBS) networks. The blocking probability of each traffic class can be analytically evaluated by means of class aggregation and iteration method. The analytic results are validated with results garnered from simulation tests.

A principal attraction of ATM networks, in both wired and wireless realizations, is that the key quality of service (QoS) parameters of every call, including end-to-end delay, jitter, and loss are guaranteed by the network when appropriate cell-level traffic controls are imposed at the user network interface (UNI) on a per call basis, utilizing the peak cell rate (PCR) and the sustainable cell rate (SCR) values for the multimedia--voice, video, and data, traffic sources. There are three practical difficulties with these guarantees. First, while PCR and SCR values are, in general, difficult to obtain for traffic sources, the typical user-provided parameter is a combination of the PCR, SCR, and the maximum burstiness over the entire duration of the traffic. Second, the difficulty in accurately defining PCR arises from the requirement that the smallest time interval must be specified over which the PCR is computed which, in the limit, will approach zero or the network's resolution of time. Third, the literature does not contain any reference to a scientific principle underlying these guarantees. Under these circumstances, the issue of providing QoS guarantees in the real world, through traffic controls applied on a per call basis, is rendered uncertain. This paper adopts a radically different, high level approach to the issue of QoS guarantees. It aims at uncovering through systematic experimentation a relationship, if any exists, between the key high level user traffic characteristics and the resulting QoS measures in a realistic operational environment. It may be observed that while each user is solely interested in the QoS of his/her own traffic, the network provider cares for two factors: (1) Maximize the link utilization in the network since links constitute a significant investment, and (2) ensure the QoS guarantees for every user traffic, thereby maintaining customer satisfaction. Based on the observations, this paper proposes a two-phase strategy. Under the first phase, the average "link utilization" computed over all the links in a network is maintained within a range, specified by the underlying network provider, through high level call admission control, i.e. by limiting the volume of the incident traffic on the network, at any time. The second phase is based on the hypothesis that the number of traffic sources, their nature--audio, video, or data, and the bandwidth distribution of the source traffic, admitted subject to a specific chosen value of "link utilization" in the network, will exert a unique influence on the cumulative delay distribution at the buffers of the representative nodes and, hence, on the QoS guarantees of each call. The underlying thinking is as follows. The cumulative buffer delay distribution, at any given node and at any time instant, will clearly reflect the cumulative effect of the traffic distributions of the multiple connections that are currently active on the input links. Any bounds imposed on the cumulative buffer delay distribution at the nodes of the network will also dominate the QoS bounds of each of the constituent user traffic. Thus, for each individual traffic source, the buffer delay distributions at the nodes of the network, obtained for different traffic distributions, may serve as its QoS measure. If the hypothesis is proven true, in essence, the number of traffic sources and their bandwidth distribution will serve asa practically realizable high level traffic control in providing realistic QoS guarantees for every call. To verify the correctness of the hypothesis, an experiment is designed that consists of a representative ATM network, traffic sources that are characterized through representative and realistic user-provided parameters, and a given set of input traffic volumes appropriate for a network provider approved link utilization measure. The key source traffic parameters include the number of sources that are incident on the network and the constituent links at any given time, the bandwidth requirement of the sources, and their nature. For each call, the constituent cells are generated stochastically, utilizing the typical user-provided parameter as an estimate of the bandwidth requirement. Extensive simulations reveal that, for a given link utilization level held uniform throughout the network, while the QoS metrics--end-to-end cell delay, jitter, and loss, are superior in the presence of many calls each with low bandwidth requirement, they are significantly worse when the network carries fewer calls of very high bandwidths. The findings demonstrate the feasibility of guaranteeing QoS for each and every call through high level traffic controls. As for practicality, call durations are relatively long, ranging from ms to even minutes, thereby enabling network management to exercise realistic controls over them, even in a geographically widely dispersed ATM network. In contrast, current traffic controls that act on ATM cells at the UNI face formidable challenge from high bandwidth traffic where cell lifetimes may be extremely short, in the range of µs. The findings also underscore two additional important contributions of this paper. First, the network provider may collect data on the high level user traffic characteristics, compute the corresponding average link utilization in the network, and measure the cumulative buffer delay distributions at the nodes, in an operational network. The provider may then determine, based on all relevant criteria, a range of input and system parameters over which the network may be permitted to operate, the intersection of all of which may yield a realistic network operating point (NOP). During subsequent operation of the network, the network provider may guide and maintain the network at a desired NOP by exercising control over the input and system parameters including link utilization, call admittance based on the requested bandwidth, etc. Second, the finding constitutes a vulnerability of ATM networks which a perpetrator may exploit to launch a performance attack.

Call Admission Control (CAC) is a very important issue in CDMA systems to guarantee a required quality of service (QoS) and to increase system capacity. In this paper, we proposed and analyzed the CAC scheme using multiple criterions (MCAC), which can provide a quicker processing time and better performance. One is based on the number of active users with the minimum/maximum threshold by considering the spillover ratio, and the other is based on the signal to interference ratio (SIR). If active users are lower/higher than the minimum/maximum number of users threshold (N_min )/(N_max ), we accept/reject the new call without any other considerations based on the first criterion. And if the number of active users is between the N_min and N_max, we consider the current SIR to guarantee QoS based on the second criterion. Then the system accepts the new call when the SIR satisfies the system requirements, otherwise, the call is rejected. The multiple criterions scheme is investigated and its performance is compared with the number of user based CAC and power based CAC.

The issue of guaranteeing Quality of Services (QoS) in a network has emerged in recent years. The Proportional Delay Differentiated Model has been presented to provide the predictable and controllable queueing delay differentiation for different classes of connections. However, most related works have focused on providing this model for a wired network. This study proposes a novel scheduler to provide proportional delay differentiation in a wireless network that includes a multi-state link. This scheduler, Look-ahead Waiting-Time Priority (LWTP), offers proportional delay differentiation and a low queueing delay, by adapting to the location-dependent capacity of the wireless link and solving the head-of-line (HOL) blocking problem. The simulation results demonstrate that the LWTP scheduler actually achieves delay ratios much closer to the target delay proportion between classes and yields smaller queueing delays than past schedulers.

This paper investigates the relations between IP network performances and the speech quality of the Voice over IP (VoIP) service through extensive experiments on a test bed network. The aim is to establish an effective and practical methodology for telecommunications operators to manage the quality of VoIP service via the management of IP network performances under their control. As IP network performances, utilization of the bottleneck link in the test bed and the following statistical factors of VoIP packets are examined: the standard deviation of delay variations (jitters), the standard deviation of packet interarrival times, and the packet loss ratio. On the other hand, VoIP speech quality is monitored as the Perceptual Evaluation of Speech Quality (PESQ). To investigate the relations under various network conditions, the experiments are performed by varying the following network related parameters of the test bed: the bandwidth of the bottleneck link, the size of the bottleneck buffer, the propagation delay, and the average of the data sizes transmitted as background data traffic. Statistical analyses of the experimental results suggest that managing the standard deviation of jitters in a network serves as a promising methodology, because its close relation to VoIP speech quality possesses robustness to changes in the network conditions. The robustness makes it practically useful since telecommunications operators can apply it to their networks, which are subject to change. The findings in this paper have opened up new visions for telecommunications operators to manage the Quality of Service (QoS) of VoIP service.

This paper considers the problem of providing relative service differentiation in IEEE 802.11 Wireless LAN by using different Medium Access Control (MAC) parameters for different service classes. We present an analytical model which predicts the saturation throughput of IEEE 802.11 Distributed Coordination Function with multiple classes of service. This model allows us to show that relative service differentiation can be achieved by varying the initial contention window alone. In this case, the saturation throughput of a station can be shown to be approximately inversely proportional to the initial contention window size being used by that station. The simulation results validate our analytical model.

Modern video compression usually consists of ME/MC (Motion Estimation/Motion Compensation), transform, and quantization of the transform coefficients. Efficient bit allocation technique to distribute available bits to motion parameters and quantized coefficients is an important part of the whole system. A method that is very complex and/or needs buffering of many future frames is not suitable for real time application. We develop an efficient bit allocation technique that utilizes the estimated effect of allocated bits to motion parameter and quantization on the overall quality. We also propose an hierarchical block based ME/MC technique that requires less computations than classical BMA (Block Matching Algorithm) while offering better motion estimation.

This paper presents a methodology for optimizing the layout of on-chip spiral inductors using structural parameters and design frequency in a response surface method. The proposed method uses scattering parameters (S-parameter) to express inductor characteristics, and hence is independent of spiral geometries and equivalent circuit models. The procedure of inductor optimization is described, and a design example is presented.

VoIP is one of the key technologies for recent telecommunication services. The quality of its services should be discussed in subjective terms. Since subjective quality assessment is time-consuming and expensive, however, objective quality assessment which estimates subjective quality without carrying out subjective quality experiments is desirable. This paper discusses the performance of the objective quality measure that was standardized as ITU-T Recommendation P.862 and clarifies the quality factors that can be evaluated with satisfactory accuracy based on it. We found that P.862 can be applied to the evaluation of coding distortion, tandeming of codecs, transmission bit-errors, packet loss, and silence compression in a codec, at least for clean Japanese speech. In addition, we propose a method of estimating the subjective quality evaluation value from objective measurement results and show the validity of this method. We also evaluate the uniqueness of objective quality assessment based on P.862 from the viewpoints of the effect of measurement noise and the variation of test speech samples, and propose how to improve the reproducibility of objective quality assessment.

Multiuser diversity, identified by recent information theoretic results, is a form of diversity inherent in a wireless network. The diversity gain is obtained from independent time-varying fading channels across different users. The main practical issue in multiuser diversity is lack of Quality of Service (QoS) guarantees. This study proposes a wireless scheduling algorithm named MUDSEQ for downlink channels exploiting multiuser diversity under explicit QoS constraints. The numerical results demonstrate that the novel algorithm can yield non-negligible diversity gain even under tight QoS constraints and little scattering or slow fading environments. Additionally, a system framework for dynamic resource allocation based on the proposed algorithm is developed.

This paper describes an adaptive fingerprint-sensing scheme for a user authentication system with a fingerprint sensor LSI to obtain high-quality fingerprint images suitable for identification. The scheme is based on novel evaluation indexes of fingerprint-image quality and adjustable analog-to-digital (A/D) conversion. The scheme adjusts dynamically an A/D conversion range of the fingerprint sensor LSI while evaluating the image quality during real-time fingerprint-sensing operation. The evaluation indexes pertain to the contrast and the ridgelines of a fingerprint image. The A/D conversion range is adjusted by changing quantization resolution and offset. We developed a fingerprint sensor LSI and a user authentication system to evaluate the adaptive fingerprint-sensing scheme. The scheme obtained a fingerprint image suitable for identification and the system achieved an accurate identification rate with 0.36% of the false rejection rate (FRR) at 0.075% of the false acceptance rate (FAR). This confirms that the scheme is very effective in achieving accurate identification.

Future mobile communication systems are expected to support multimedia applications (audio phone, video on demand, video conference, file transfer, etc.). Multimedia applications make a great demand for bandwidth and impose stringent quality of service requirements on the mobile wireless networks. In order to provide mobile hosts with high quality of service in the next generation mobile multimedia wireless networks, efficient and better bandwidth reservation schemes must be developed. A novel traffic-based bandwidth reservation scheme is proposed in this paper as a solution to support quality of service guarantees in the mobile multimedia wireless networks. Based on the existing network conditions, the proposed scheme makes an adaptive decision for bandwidth reservation and call admission by employing fuzzy inference mechanism, timing based reservation strategy, and round-borrowing strategy in each base station. The amount of reserved bandwidth for each base station is dynamically adjusted, according to the on-line traffic information of each base station. We use the dynamically adaptive approach to reduce the connection-blocking probability and connection-dropping probability, while increasing the bandwidth utilization for quality of service sensitive mobile multimedia wireless networks. Simulation results show that our traffic-based bandwidth reservation scheme outperforms the previously known schemes in terms of connection-blocking probability, connection-dropping probability, and bandwidth utilization.

Future IP networks will provide multi-class-services that have multiple levels of Quality of Services (QoS) at different prices. One of the issues for the network service provider (NSP) will be how to profit by providing them. This paper proposes a scheme that maximizes the profit of the NSP by controlling the service-list under the constraint of the available network resources. We introduce a model in which the users' selection from among the multiple classes is influenced not just by the price and QoS of one class, but the prices and QoS levels of all classes. In short, the user's selection involves a balance between the price and QoS levels of all classes. To model the users' class choice, we adopt discrete choice analysis; it can estimate the model parameters such that the model fits actual choice data. This paper proposes a functional framework that consists of User Choice Model Function, Original Demand Forecast Function, and Service-list Determination Function. The proposed model has the advantage of following actual changes adaptively. Effectiveness of the proposed scheme is evaluated by computer simulation for a multiple class service; even if the real parameters are changed, the proposal can follow the change and provide the optimal service-list that maximizes profit adaptively.

Quality of Service (QoS) provisioning is a new but challenging research area in the field of Mobile Ad hoc Network (MANET) to support multimedia data communication. However, the existing QoS routing protocols in ad hoc network did not consider a major aspect of wireless environment, i.e., mutual interference. Interference between nodes belonging to two or more routes within the proximity of one another causes Route Coupling. This can be avoided by using zone-disjoint routes. Two routes are said to be zone disjoint if data communication over one path does not interfere with the data communication along the other path. In this paper, we have proposed a scheme for supporting priority-based QoS in MANET by classifying the traffic flows in the network into different priority classes and giving different treatment to the flows belonging to different classes during routing so that the high priority flows will achieve best possible throughput. Our objective is to reduce the effect of coupling between routes used by high and low priority traffic by reserving zone of communication. The part of the network, used for high priority data communication, i.e, high priority zone, will be avoided by low priority data through the selection of a different route that is maximally zone-disjoint with respect to high priority zones and which consequently allows contention-free transmission of high priority traffic. The suggested protocol in our paper selects shortest path for high priority traffic and diverse routes for low priority traffic that will minimally interfere with high priority flows, thus reducing the effect of coupling between high and low priority routes. This adaptive, priority-based routing protocol is implemented on Qualnet Simulator using directional antenna to prove the effectiveness of our proposal. The use of directional antenna in our protocol largely reduces the probability of radio interference between communicating hosts compared to omni-directional antenna and improves the overall utilization of the wireless medium in the context of ad hoc wireless network through Space Division Multiple Access (SDMA).

The issue of scalable Differentiated Services (DiffServ) admission control now is still an open research problem. We propose a new admission control model that can not only provide coarse grain Quality of Services (QoS), but also guarantee end-to-end QoS for assured service without per-flow state management at core routers within DiffServ domain. Associated with flow aggregation model, a hybrid signaling protocol is proposed to select the route satisfying the end-to-end QoS requirements. Simulation result shows that the proposed model can accurately manage resource, leading to much better performance when compared to other schemes.

By means of the three-dimensional (3D) finite-difference time domain (FDTD) method, we have investigated in detail the optical properties of a two-dimensional photonic crystal (PC) surface-emitting laser having a square-lattice structure. The 3D-FDTD calculation is carried out for the finite size PC slab structure. The device is based on band-edge resonance, and plural band edges are present at the corresponding band edge point. For these band edges, we calculate the mode profile in the PC slab, far field pattern (FFP) and polarization mode of the surface-emitted component, and photon lifetime. FFPs are shown to be influenced by the finiteness of the structure. Quality (Q) factor, which is a dimensionless quantity representing photon lifetime, is introduced. The out-plane radiation loss in the direction normal to the PC plane greatly influences the total Q factor of resonant mode and is closely related with the band structure. As a result, Q factors clearly differ among these band edges. These results suggest that these band edges include resonant modes that are easy to lase and resonant modes that are difficult to lase.

FEC (Forward Error Correction) can repair the damage to communication quality due to packet loss. The growing requirement of FEC for high-quality video transmission is inevitable on broadband networks. We have designed and implemented FEC, and integrated it to our developed video transmission system named "mpeg2ts." Our goal is to make it possible to deploy this system on the broadband Internet. However, the problem with constant redundancy of FEC is that weakness to fluctuation of network condition. To resolve this problem, in this paper, we propose and evaluate an efficient FEC method for high-quality video transmission. The proposed mechanisms can provide robustness as well as saving of processing load and optimization of bandwidth consumption. Moreover, we integrate it into a system to deploy it on the real broadband Internet. Transmission experiment demonstrates availability of developed system deployed on the network.