This paper proposes an idea for modeling aggregated TCP/IP traffic arriving at a bottleneck link by focusing on its scaling behavior. Here, the aggregated TCP/IP traffic means the IP packet traffic from many TCP connections sharing the bottleneck link. The model is constructed based on the outcomes of our previous works investigating how the TCP/IP networking mechanism affects the self-similar scaling behavior of the aggregated TCP/IP traffic in a LAN/WAN environment. The proposed traffic model has been examined from the perspective of application to network performance estimation. The examinations have shown that it models the scaling behavior and queueing behavior of actual traffic, though it neglects the interaction among TCP connections that compete with each other for the single bottleneck link bandwidth.

This paper proposes a bandwidth allocation algorithm and a demand accommodation algorithm guaranteeing utility max-min fairness under bandwidth constraints. We prove that the proposed algorithms can fairly split network resources among connections and achieve call admission control considering the fairness among different types of applications. We then formulate three different network design problems to maximize the total utility of all customers, the number of users accommodated in the network, and the average utility of the customers accommodated in the network. To solve the problems, we extend the conventional network design algorithms considering utility max-min fair share, and numerically evaluate and compare their performance. Finally, we summarize the best algorithms to design the utility max-min fair share networks considering the operation policy of network providers.

For real-time services, such as VoIP and videoconferencing supplied through a multi-domain MPLS network, it is vital to guarantee end-to-end QoS of the inter-domain paths. Thus, it is important to allocate an appropriate QoS class to the inter-domain paths in each transit domain. Because each domain has its own policy for QoS class allocation, each domain must then allocate an appropriate QoS class adaptively based on the estimation of the QoS class allocation policies adopted in other domains. This paper proposes an adaptive method for acquiring a QoS class allocation policy through the use of reinforcement learning. This method learns the appropriate policy through experience in the actual QoS class allocation process. Thus, the method can adapt to a complex environment where the arrival of inter-domain path requests does not follow a simple Poisson process and where the various QoS class allocation policies are adopted in other domains. The proposed method updates the allocation policy whenever a QoS class is actually allocated to an inter-domain path. Moreover, some of the allocation policies often utilized in the real operational environment can be updated and refined more frequently. For these reasons, the proposed method is designed to adapt rapidly to variances in the surrounding environment. Simulation results verify that the proposed method can quickly adapt to variations in the arrival process of inter-domain path requests and the QoS class allocation policies in other domains.

This paper outlines the perspectives on Software Defined Radio (SDR) technology in viewpoint of the standardization of the future mobile communication systems. The activities of ITU-R SG8 Working Party 8F (WP8F) and mITF (mobile IT Forum) of Japan for systems beyond IMT-2000 (B3G) or 4-th generation mobile systems are firstly summarized. The latest discussions relating to SDR technology in the both parties are reported. It is followed by consideration on both expectations and technical issues on SDR in order to realize the technology in the future mobile communication systems. They are clarified in the viewpoint of standardization activity on B3G. Also some regulation issues are lastly summarized.

After a bend structure has been obtained through application of an appropriate voltage, a pi-cell exhibits one of two stable structures: a bend structure if the applied bias is above a certain threshold ( 2 V), and a twist structure at a lower voltage. For use in an optical device, the pi-cell must be operated with the bend structure. It was found, however, that when the bias is switched quickly to a level lower than the threshold, a metastable structure exists for a few hundreds of milliseconds before relaxing to a twist structure. From dynamic optical transmittance analysis, this structure is considered to have a bend configuration. The temporary bend structure persists at lower voltages because it takes a while to initiate an energy redistribution from the bend structure to the twist. This is considered as a novel physical state, and is called a "dynamic bend structure. " It enables the pi-cell to be operated even if the bias is below the threshold voltage, provided that the device is biased at higher voltages for a fraction of each cycle to retain the bend structure.

Packet loss is a serious problem due to the shortage of optical buffers in all-optical packet switched networks. In order to reduce packet losses, a dynamic routing method called 'deflection routing' has been proposed. Deflection routing, however, requires an optical switch to modify routing tables and packet labels for overflowing packets, so this routing method may also lead to other implementation problems in packet routing and forwarding. This paper proposes a simple routing method called 'reflection routing' which utilizes optical transport links as optical buffers to improve the quality of service in optical packet switched networks in terms of packet loss ratio. We numerically demonstrate the effectiveness and applicability of reflection routing.

Telecommunications networks have become an important social infrastructure, and their robustness is considered to be a matter of social significance. Conventional network planning methods are generally based on the maximum volume of ordinary traffic and only assume explicitly specified failure scenarios. Therefore, present networks have marginal survivability against multiple failures induced by an extraordinarily high volume of traffic generated during times of natural disasters or popular social events. This paper proposes a telecommunications network planning method based on probabilistic risk assessment. In this method, risk criterion reflecting the degree of risk due to extraordinarily large traffic loads is predefined and estimated using probabilistic risk assessment. The probabilistic risk assessment can efficiently calculate the small but non-negligible probability that a series of multiple failures will occur in the considered network. Detailed procedures for the proposed planning method are explained using a district mobile network in terms of the extraordinarily large traffic volume resulting from earthquakes. As an application example of the proposed method, capacity dimensioning for the local session servers within the district mobile network is executed to reduce the risk criterion most effectively. Moreover, the optimum traffic-rerouting scheme that minimizes the estimated risk criterion is ascertained simultaneously. From the application example, the proposed planning method is verified to realize a telecommunications network with sufficient robustness against the extraordinarily high volume of traffic caused by the earthquakes.

This paper is concerned with bandwidth reservation for circuit groups which handle calls requesting asymmetric forward and backward multi-connections. A model of circuit group with sub-group configuration is treated, and two types of the bandwidth reservation schemes for the model are studied in this paper. One is a global scheme with monitoring the whole circuit group, and the other is a local scheme with monitoring each sub-group independently. The problems of optimizing the reservation parameters are formulated, and optimization methods for the problems are proposed. Numerical example are presented, and effectiveness of the reservation schemes with using the optimized parameters is numerically examined.

This paper is concerned with two important planning problems for transport network planning; circuit routing problems and facility planning problems. We treated these optimization problems by taking into account survivability requirements. In the circuit routing problem tackled in this paper, therefore, optimization of circuit restoration plans, namely allocation of spare capacity for assumed failure scenarios is considered together with optimization of circuit routing in a no failure case. In the facility planning problems, failure scenarios of new facilities whose installation is yet to be determined are considered. In this paper, we present a formulation of these two optimization problems, and give 1) optimization algorithms based on the IA (Increment Assignment) method for routing problems and 2) optimization algorithms based on a combination of the GA (Genetic Algorithm) and the IA method for facility planning problems. The IA based routing algorithm can cope flexibly with various constraints on practical network operations and is applicable to large-scale complicated network models without causing a rapid increase in computation time. The GA based facility planning algorithm includes the IA based algorithm as a function for evaluating objective function values. Taking advantage of the important features of the IA based algorithm, we propose an acceleration technique for the GA based facility planning algorithm. In this paper, several numerical examples are provided and the effectiveness of the proposed algorithms is numerically evaluated.

Priority control and call admission control are indispensable traffic management methods to guarantee each QoS requirement of connections in ATM networks. The key technique of call admission control under priority control is to estimate required bandwidth of each connection to satisfy all QoSs of calls in progress. In this paper, we propose a novel approximation method to calculate the required bandwidth of ATM connections through priority queues and show a practical call admission control scheme using the proposed method. The essence of the approximation method is to model prioritized parallel queues as a series of queues in tandem with no priority control by focusing on the number of cells in queues. The tandem queue approximation method enables us to model each queue under priority control as a single non-priority FIFO queue in terms of its queue length. This results in that effective bandwidth techniques are applicable to priority queues. The effectiveness of the proposed scheme is evaluated by some numerical examples.

Providing mobile services that ensure user satisfaction is one of the most crucial issues for telecommunication carriers and service providers. Traditionally, user satisfaction with the service, i.e., the Quality of Experience (QoE), has been assessed in a laboratory by using elaborate network systems and customized terminals. However, reliable QoE for a target mobile service in the context of actual use cannot be obtained by laboratory experiment, since QoE can be affected by a variety of factors such as environmental conditions and the mental state of the user. This paper proposes a novel Web script-based field evaluation method designed to ascertain user satisfaction with mobile services. The proposed method has the following practical advantages. Since experimental conditions, especially communication conditions, can be simulated with a Web script, such as FlashLite, a subjective assessment can be conducted anywhere using the mobile terminal of the participant. Thus, QoE for a target mobile service in the field can easily be obtained.

In future systems beyond IMT-2000, macrocell cellular systems such as the 3G systems and high bandwidth microcell wireless systems such as Wireless LAN will complement one another. Routing in the systems beyond IMT-2000 will support seamless inter- and intra-system handover among the cellular and WLAN systems by maintaining active connections. Under such environments, the time scales of mobility and bandwidth-sharing behavior cannot be easily separated. It is not obvious what fraction of traffic is accommodated by each cellular and WLAN system, i.e. the traffic distribution is unknown. This paper shows the considerable impacts the mobility of users has on the capacities of the systems beyond IMT-2000 with roaming capability between different bit rate systems. Especially, this paper demonstrates that the traffic distribution among different systems is a major factor in defining total network throughput. We also provide an analytical method to determine the traffic distribution based on the theory of queueing networks.

This paper describes the implementation of an optical switch control system for best-effort multi-wavelength path assignment in DWDM networks based on the SLAMNet (Statistical Lambda Multiplexing Network) that provides dynamic path assignment capabilities without signaling between nodes. The platform of the prototype system consists of a hardware and operation system that are commercially available in the market. We developed the control functions of the SLAMNet as application software to run on the platform. The control function monitors the instantaneous bit rate of the traffic in the wavelength paths and autonomously sets up or releases optical channels in an independent and distributed manner. We demonstrate that the prototype system achieves a switching time of less than 28 milliseconds, which can respond to the burst traffic that emerges and disappears within a fraction of a second in backbone networks. This result indicates that the network architecture SLAMNet is applicable for best-effort multi-wavelength path assignment as an overlay network built on a legacy infrastructure that has no signaling capabilities. The design description and detail hardware configuration are presented. The control mechanisms and performance analysis are also included.

MPLS-based path technology shows promise as a means of realizing reliable IP networks. Real-time services such as VoIP and video-conference supplied through a multi-domain MPLS network must be able to guarantee end-to-end QoS of the inter-domain paths. Thus, it is important to allocate an appropriate QoS class to the inter-domain paths in each domain traversed by the inter-domain paths. Because each domain has its own policy for QoS class allocation, it is necessary to adaptively allocate the optimum QoS class based on estimation of the QoS class allocation policies in other domains. This paper proposes two kinds of adaptive QoS class allocation schemes, assuming that the arriving inter-domain path requests include the number of downstream domains traversed by the inter-domain paths and the remaining QoS value toward the destination nodes. First, a measurement-based scheme, based on measurement of the loss rates of inter-domain paths in the downstream domains, is proposed. This scheme estimates the QoS class allocation policies in the downstream domains, using the measured loss rates of path requests. Second, a state-dependent type scheme, based on measurement of the arrival rates of path requests in addition to the loss rates of paths in the downstream domains, is also proposed. This scheme allows an appropriate QoS class to be allocated according to the domain state. This paper proposes an application of the Markov decision theory to the modeling of state-dependent type scheme. The performances of the proposed schemes are evaluated and compared with those of the other less complicated non-adaptive schemes using a computer simulation. The results of the comparison reveal that the proposed schemes can adaptively increase the number of inter-domain paths accommodated in the considered domain, even when the QoS class allocation policies change in the other domains and the arrival pattern of path requests varies in the considered domain.

To reduce the cost of fault management in all-optical networks, it is a promising approach to detect the degradation of optical signal quality solely at the terminal points of all-optical monitoring paths. The all-optical monitoring paths must be routed so that all single-link failures can be localized using route information of monitoring paths where signal quality degradation is detected. However, route computation for the all-optical monitoring paths that satisfy the above condition is time consuming. This paper proposes a procedure for deriving the lower bounds of the required number of monitoring paths to localize all single-link failures, and proposes an efficient monitoring path computation method based on the derived lower bounds. The proposed method repeats the route computation for the monitoring paths until feasible routes can be found, while the assumed number of monitoring paths increases, starting from the lower bounds. With the proposed method, the minimum number of monitoring paths with the overall shortest routes can be obtained quickly by solving several small-scale integer linear programming problems when the possible terminal nodes of monitoring paths are arbitrarily given. Thus, the proposed method can minimize the required number of monitors for detecting the degradation of signal quality and the total overhead traffic volume transferred through the monitoring paths.

A feature of services provided in a Next Generation Network (NGN) is that the end-to-end quality is guaranteed. This is quite a challenging issue, given the considerable fluctuation in network conditions within a Fixed Mobile Convergence (FMC) network. Therefore, a novel approach, whereby a network node and a mobile terminal such as a cellular phone cooperate with each other to control service quality is essential. In order to achieve such cooperation, the mobile terminal needs to become more intelligent so it can estimate the service quality, including the user's perceptual quality, and notify the measurement result to the network node. Subsequently, the network node implements some kind of service control function, such as a resource and admission control function, based on the notification from the mobile terminal. In this paper, the role of the mobile terminal in such collaborative system is focused on. As a part of a QoS/QoE measurement system, we describe an objective speech quality assessment with payload discrimination of lost packets to measure the user's perceptual quality of VoIP. The proposed assessment is so simple that it can be implemented on a cellular phone. We therefore did this as part of the QoS/QoE measurement system. By using the implemented system, we can measure the user's perceptual quality of VoIP as well as the network QoS metrics, in terms of criteria such as packet loss rate, jitter and burstiness in real time.

As a network evolves following initial deployment, its service functions remain diversified through the openness of the network functions. This indicates that appropriate simplification of the service functions is essential if the evolving network is to achieve the required scalability of service processing and service management. While the screening of service functions is basically performed by network users and the market, several service functions will be automatically simplified based on the growth of the evolving network. This paper verifies the simplification of service functions resulting from the evolution of the network itself. First, the principles that serve as the basis for simplifying the service functions are explained using several practical examples. Next, a simulation model is proposed to verify the simplification of service functions in terms of the priority control function for path routing and load balancing among multiple paths. From the results of the simulation, this study clarifies that the anticipated simplification of service functions is actually realizable and the service performance requirements can be reduced as the network evolves after deployment. When the simplification of service functions can improve network quality, it accelerates the evolution of the network and increases the operator's revenue.

Secret sharing schemes have been proposed to protect content by dividing it into many pieces securely and distributing them over different locations. Secret sharing schemes can also be used for the secure delivery of content. The original content cannot be reconstructed by the attacker if the attacker cannot eavesdrop on all the pieces delivered from multiple content servers. This paper aims to obtain secure delivery routes for the pieces, which minimizes the probability that all the pieces can be stolen on the links composing the delivery routes. Although such a route optimization problem can be formulated using an ILP (Integer Linear Programming) model, optimum route computation based on the ILP model requires large amounts of computational resources. Thus, this paper proposes a lightweight route computation method for obtaining suboptimum delivery routes that achieve a sufficiently small probability of all the pieces being stolen. The proposed method computes the delivery routes successively by using the conventional shortest route algorithm repeatedly. The distance of the links accommodating the routes that have already been calculated is adjusted iteratively and utilized for calculation of the new shortest route. The results of a performance evaluation clarify that sufficiently optimum routes can be computed instantly even in practical large-scale networks by the proposed method, which adjusts the link distance strictly based on the risk level at the considered link.

This paper studies the local Poisson property of aggregated IP traffic. First, it describes the scenario where IP traffic presents a Poisson-like characteristic within some limited range of time scales when packets from independent traffic streams are aggregated. Each of the independent traffic streams corresponds to a series of correlated IP packets such as those of a transport connection. Since the Poisson-like characteristic is observed only within some limited range of time scales, we call this characteristic the local Poisson property. The limited range of time scales of the local Poisson property can be estimated from a network configuration and characteristics of transport connections. Second, based on these observations, we seek the possibility to apply an ordinary Poisson process to evaluation of the packet loss probability in IP networks. The analytical investigation, where IP traffic is modeled by a superposition of independent branching Poisson processes that presents the local Poisson property, suggests that the packet loss probability can be estimated by a finite-buffer queue with a Poisson process when the buffer size is within a certain range. The investigation is verified by simulations. These findings expand the applicability of conventional Poisson-based approaches to IP network design issues.

Optical burst switching (OBS) is a promising approach for the realization of future flexible high-speed optical networks. In particular, a multicast forwarding method for optical bursts is important if an efficient high-speed grid computing network is to be realized. In OBS networks, the number of burst replicas generated at each node is strongly restricted due to optical power impairment of multicast bursts. Moreover, unrestricted replication of multicast bursts at each OBS node may not be advantageous because an increase in the number of multicast bursts within the network causes more frequent deflection forwarding of both multicast and unicast bursts. This paper proposes an efficient hop-by-hop multicast forwarding method for optical bursts, where idle output ports are selected based on scores simply calculated using a routing table that each OBS node holds. This method can mitigate increases in loss rate and transfer delay of multicast bursts, even if the number of burst replicas generated at each OBS node is strongly restricted. Moreover, this method can efficiently mitigate an increase in the number of multicast bursts within the network by avoiding unnecessary replication of multicast bursts at each OBS node. Simulation results show that the proposed method can actually mitigate degradation of the loss rate and transfer delay for multicast bursts under the restricted number of burst replicas at each OBS node. Moreover, when the arrival rate of multicast bursts is large relative to that of unicast bursts, the proposed method is able to improve the loss rates of both multicast and unicast bursts by switching the forwarding method for the multicast bursts to the simple unicast forwarding method without burst replication.