The notion of effective bandwidth provides an elegant and powerful mathematical basis for the provision of QoS-assured services over IP networks. In this paper, we propose a semi-parametric estimator of effective bandwidth, called Gaussian estimator using buffer masurement, for superposition of sources in IP networks. In contrast to most existing proposals concerning the effective bandwidth estimator, our proposal works based on a small set of measurements of the workload in the buffer of a router. We analytically show the property of the proposed estimator with respect to the dependence on the service rate. We provide numerical results to show that our proposed estimator is more accurate than estimators that rely only on the amount of traffic from sources.

We propose a packet sampling strategy called fixed-period sampling, which selects at most one packet in every fixed-length period. Under the fixed-period sampling, the number of flow-cache lookups during a unit of time or the number of entries in a flow cache is bounded by a constant, which is simply expressed by a few tuning parameters. As an application of the fixed-period sampling, we also focus on the flow-rate estimation from fixed-period sampled packet streams. In particular, we propose a simple estimator based solely on the sampling frequency. We have conducted simulation experiments using two real traces to show basic characteristics of the fixed-period sampling for the comparison with the fixed-period sampling. We also show the accuracy of the proposed flow-rate estimator through simulations.

We describe a parameter estimation method for a target object in an area that sensors monitor. The parameters to be estimated are the perimeter length, size, and parameter determined by the interior angles of the target object. The estimation method does not use sensor location information, only the binary information on whether each sensor detects the target object. First, the sensing area of each sensor is assumed to be line-segment-shaped, which is a model of an infrared distance measurement sensor. Second, based on the analytical results of assuming line-segment-shaped sensing areas, we developed a unified equation that works with general sensing areas and general target-object shapes to estimate the parameters of the target objects. Numerical examples using computer simulation show that our method yields accurate results.

This paper discusses the problems in designing virtual-path (VP) networks and underlying transmission-path (TP) networks using the "self-sizing" capability. Self-sizing implies an autonomous adjustment mechanism for VP bandwidths based on traffic conditions observed in real time. The notion of "bandwidth demand" has been introduced to overcome some of the problems with VP bandwidth sizing, e.g., complex traffic statistics and diverse quality of service requirements. Using the bandwidth demand concept, a VP-bandwidth-sizing procedure is proposed in which real-time estimates of VP bandwidth demand and successive VP bandwidth allocation are jointly utilized. Next, TP bandwidth demand, including extra capacity to cover single-link failures, is defined and used to measure the congestion level of the TP. Finally, a TP provisioning method is proposed that uses TP "lifetime" analysis.

The spatial relations between sensors placed for target detection can be inferred from the responses of individual sensors to the target objects. Motivated by this fact, this paper proposes a method for estimating the location of sensors by using their responses to target objects. The key idea of the proposal is that when two or more sensors simultaneously detect an object, the distances between these sensors are assumed to be equal to a constant called the basic range. Thus, new pieces of proximity information are obtained whenever an object passes over the area in which the sensors are deployed. This information is then be aggregated and transformed into a two dimensional map of sensors by solving a nonlinear optimization problem, where the value of the basic range is estimated together. Simulation experiments verify that the proposed algorithm yields accurate estimates of the locations of sensors.

QoS (Quality of Service) control in WLAN (IEEE802.11 Wireless LAN) is becoming increasingly important because WLAN is widely deployed as an access network and also plays a key role in providing seamless QoS communication between wired networks and wireless terminals. Although previous research has attempted to increase total throughput or available bandwidth in WLAN, few studies have treated individual TCP/UDP flow QoS. EDCA in IEEE802.11e might provide prioritized QoS functions that would partially address this problem. However, in uplink flow, which is defined as data moving from a terminal toward an Access Point, EDCA has limitations. These manifest themselves both across classes and in differentiated QoS control between terminals in the same class. Furthermore, 802.11e requires modification of terminals as well as other alterations proposed by other researchers. Instead of 802.11e or other modifications of 802.11, we propose an approach to controlling QoS that requires no terminal modifications or installation of additional software/hardware. The proposed idea is MAC-frame Receiving-Opportunity Control (ROC), in which a MAC (Media Access Control) frame is completely delivered only if it has sufficiently high priority; otherwise either the MAC frame is discarded or an Acknowledge (ACK) to the frame is not sent. The frame that was discarded is forced to accept a back-off waiting time for retransmission, consistent with 802.11DCF. This results in QoS degradation for low priority flows and eventually results in QoS improvements for high priority flows. Performance evaluation shows that the ROC causes some performance degradation in total WLAN throughput but can achieve not only QoS priority control but also arbitrary throughput performance. In particular, the ROC (in the MAC layer) can also permit different throughputs for high priority and low priority flows, conditioned on control processes in other layers. These may include rate adaptation (in the MAC layer) and TCP congestion control (in the TCP layer).

This paper proposes two algorithms for defining a routing domain in multiclass-of-service networks. One an off-line-based method, whose objective is to optimize dynamic routing performance by using precise knowledge on the traffic levels. The algorithm of the proposed method takes into account the random nature of the traffic flow, which is not considered in the network flow approach. The proposed method inherits the conceptual simplicity of the network flow approach and remains applicable to large and complex networks. In simulation experiments, the proposed off-line-based method performs better than the method based on the network flow approach, but has a similar the computation time requirement. The other method proposed here is an on-line-based method for application to B-ISDNs, where precise traffic data is not expected to be available. In this method, the routing domain is defined adaptively according to the network performance (call-blocking probability) measured in real-time. In simulation experiments, the performance of this method is comparable to that of the off-line-based method--especially when highly efficient dynamic routing is used. This paper also derives and describes methods for approximating the implied costs for multiclass-of-service networks. The approximations are very useful not only for off-line-based routing domain definition (RDD) methods but also for other kinds of network controls or optimal network dimensioning based on the concept of revenue optimization.

This paper presents an analytical model for network throughput of WLANs, taking into account heterogeneous conditions, namely network nodes transmit different length frames with various offered load individually. The airtime concept, which is often used in multi-hop network analyses, is firstly applied for WLAN analysis. The proposed analytical model can cover the situation that there are saturation and non-saturation nodes in the same network simultaneously, which is the first success in the WLAN analyses. This paper shows the network throughput characteristics of four scenarios. Scenario 1 considers the saturation throughputs for the case that one or two length frames are transmitted at the identical offered load. Scenarios 2 and 3 are prepared for investigating the cases that all network nodes transmit different length frames at the identical offered load and identical length frames at the different offered loads, respectively. The heterogeneous conditions for not only frame length but also offered load are investigated in Scenario 4.

Randomly distributed wireless sensors used to monitor and detect a moving object were investigated, and performance measures such as the expected time/space detection ratio were theoretically analyzed. In particular, the insensitivities (robustness) of the performance measures to the conditions of the distributed wireless sensors and the target object were analyzed. Robust explicit equations for these performance measures were derived, and these equations can be used to calculate them without knowing the sensing area shape or the target object trajectory. These equations were applied to the following two applications. (1) They were used to estimate the impact of active/sleeping state schedule algorithms of sensors on the expected ratio of the time that the sensors detect the target object during its movement. The results were used to identify the active state schedule that increases the expected time ratio. (2) They were also applied to a sensor density design method that uses a test object. This method can be used to ensure that the expected time ratio that at least one sensor can detect the target satisfies the target value without knowing the sensing area size or the movement of the target object.

The new algorithm for VP bandwidth control described and analyzed in this paper is a revised version of the Successive Modification Method. Its operation is based only on call-level performance (call blocking probabilities) measured in real time, without explicitly taking the cell-level performance into account. This algorithm does not need to predict future traffic demand and to perform network-wide optimization according to the predicted traffic. These features are well suited for a B-ISDN environment, with the variety of ATM bearer services and the uncertainty of their traffic demand and other characteristics. This paper describes the relationship between the proposed control and other traffic controls in ATM networks, such as CAC and VP shaping/policing. It also offers a solution to the problem of the competition that arises when several VPs in the same transmission path need increased bandwidth. Evaluation of the transient behavior of the VP bandwidth occupied by VCs shows that there is a lower limit in the control cycle and that this limit can be estimated as the longest average holding time of VCs among all services. Numerical results obtained using a call-by-call simulator show that proposed control is effective in preventing the performance degradation caused by a large traffic imbalance in communications networks. Comparison of the proposed control with a dynamical alternate routing for VC reveals that the VP bandwidth control is effective in relieving only the areas showing serious performance degradation, but that it is not so effective in improving the overall network performance.

Asynchronous Transfer Mode (ATM) technology is expected to be used in constructing a B-ISDN. ATM networks must support a variety of services, e.g., voice, data, and image communications with different grade of service requirements. The demand for these services and their traffic characteristics, however, are not yet clear. To implement B-ISDN under this situation, it is necessary to establish a network control scheme that can' absorb the difference between the estimated traffic and the traffic that is actually offered. In ATM networks, virtual path bandwidth control is a key control scheme for absorbing this traffic estimation error, and several control algorithms have already been proposed. When we try to further utilize the VP resource by dynamically reallocating the bandwidth according to the short-term traffic variation, however, we need control schemes that are highly responsive. This is achieved by using control intervals that are shorter than the intervals over which traffic fluctuates. Control algorithms based on central controllers generally need to collect a large amount of information from geographically widespread network facilities and solve a large optimization problem. This can make them difficult to use with short control intervals in large networks. An alternative enabling the shorter control periods is to use multiple distributed controllers that use only local information. This paper proposes two new VP bandwidth control algorithms suitable for this distributed implementation. In these algorithms, decentralized controllers are located at network nodes including ATM switch (ATM-SW) or ATM cross connect (ATM-XC) function, and each controller observes the quality of the VPs relevant to it. The bandwidth is modified successively as these distributed controllers communicate with each other. We therefore call this method "successive modification method" (SMM). Numerical evaluation using a model network shows the effectiveness of these algorithms for preventing the performance degradation caused by large-scale traffic imbalance within a network. Comparison with the batch modification method (BMM), which has no feedback effect, shows that the proposed SMM with approprate control intervals can be more responsive to traffic variation over time, but is slightly inferior when network conditions are static.

We consider a multirate loss system that accommodates several service-classes with cooperative and non-cooperative users, which generate random or quasi-random arriving calls (depending on the users population). The term “ cooperative” refers to users that can retry (with a certain probability, when the total occupied bandwidth of the system is below a predefined threshold) to be connected in the system with a reduced bandwidth, if they are blocked with their initial peak-bandwidth requirement. This behavior increases the quality of service (QoS) perceived by other users. Due to the retries, the system model does not have a product form solution for the steady state distribution. However, we propose an efficient calculation of system's occupancy distribution, congestion probabilities and system's utilization, while avoiding complex state space enumeration and processing. As it is shown through simulation, the proposed recursive formulas are consistent and quite accurate. For evaluation, we use the conventional trunk/bandwidth reservation control to equalize the congestion probabilities and compare them with the results of the proposed models. Besides, we generalize the proposed models by considering the coexistence of random and quasi-random arrivals. Furthermore, we propose (a) a heuristic method for the determination of desired retry probabilities according to a fairness index and (b) an optimization procedure whereby we assess the retry threshold so that the QoS of the least speed non-cooperative calls is guaranteed for certain retry probabilities of cooperative calls.

A connection admission control (CAC) that guarantees a negotiated cell-loss ratio for all cell-streams passing through the usage parameter control (UPC) in ATM networks is proposed. In particular, the cases in which a jumping-window, sliding-window, or continuous-leaky-bucket scheme are used for peak-cell-rate policing are discussued, and the upper bound for cell-loss ratio of the cell-streams passing through each type of UPC is derived. The CACs based on the derived cell-loss-ratio upper bounds ensure the quality of service in all cases by combining the relevant UPCs. There are three possible combinations of CAC and UPC, depending on the UPC mechanism used. The impact of the choice of CAC and UPC combination on bandwidth utilization is discussed using several numerical examples.

This paper deals with a network architecture based on a backbone network, using ATM switches (ATM-SW) and ATM Cross-Connect Systems (ATM-XC). The backbone network is efficiently utilized by multiple-routing scheme. The performance of the network is controlled, exploiting the concept of Virtual Paths (VP) in ATM technology. The network is controlled by allocating the bandwidth of VPs so as to minimize the worst call blocking probability of all ATM-SW pairs, under the constraints of the ATM-SW capacities and the bandwidths of transmission paths in the backbone network. To improve network performance, we use a trunk reservation scheme among service classes. We propose a heuristic approach to solve the problem of non-linear integer programming. Evaluation of the proposed optimization scheme, in comparison to other optimal methods, shows the efficiency of the present scheme.

In wireless single-hop networks, IEEE 802.11e Enhanced Distributed Channel Access (EDCA) is the standard for Quality of Service (QoS) control. However, it is necessary for controlling QoS to modify the currently used IEEE 802.11 Distributed Coordination Function (DCF)-compliant terminals as well as Access Points (APs). In addition, it is necessary to modify the parameter of IEEE 802.11e EDCA when the traffic is heavy. This paper proposes a novel scheme to guarantee QoS of high-priority flow with Receiving Opportunity Control in MAC Frame (ROC) employed adaptive flow control in wireless multi-hop network. In the proposed scheme, the edge APs which are directly connected to user terminals estimate the network capacity, and calculate appropriate ACK prevention probability against low-priority flow according to traffic load. Simulation evaluation results show that the proposed scheme guarantees QoS.

We investigated performance of routing schemes in B-ISDN, for heterogeneous traffic flows under various bandwidths. In particular, we compared the simulated performance of these schemes by evaluating their blocking probabilities. To achieve high performance, these schemes use special kinds of routing algorithm, one which is pre-selection algorithm and one which is cyclic algorithm. We investigated the efficiency of the pre-selection algorithm and the robustness of the cyclic algorithm for nonuniform traffic and network resources. We found that these routing algorithm schemes can compensate for errors in resource design.

We propose an analytical model for IEEE 802.11 wireless local area networks (WLANs). The analytical model uses macroscopic descriptions of the distributed coordination function (DCF): the backoff process is described by a few macroscopic states (medium-idle, transmission, and medium-busy), which obviates the need to track the specific backoff counter/backoff stages. We further assume that the transitions between the macroscopic states can be characterized as a continuous-time Markov chain under the assumption that state persistent times are exponentially distributed. This macroscopic description of DCF allows us to utilize a two-dimensional continuous-time Markov chain for simplifying DCF performance analysis and queueing processes. By comparison with simulation results, we show that the proposed model accurately estimates the throughput performance and average queue length under light, heavy, or asymmetric traffic.