A threshold secret sharing scheme can realize reliable delivery of important content using redundant routes through a network. Furthermore, multicast delivery of threshold secret shared content can achieve efficient resource utilization thanks to the application of multicast and network coding techniques to multiple pieces of the content. Nevertheless, a tradeoff exists between reliability and efficiency if multicast content delivery uses network coding. This paper proposes a flexible multicast delivery scheme for threshold secret shared content that can control the tradeoff between reliability and efficiency. The proposed scheme classifies all the pieces obtained from the original content into multiple groups, and each group is subjected to network coding independently. An optimization procedure is proposed for the multicast delivery scheme, which involves two different heuristic delivery route computation methods applicable to large-scale networks. Evaluation results show that the optimized multicast delivery scheme adopting an appropriate grouping method and classifying the pieces into a suitable number of groups can minimize the required link bandwidth while satisfying a specified content loss probability requirement.

It is important to monitor routing protocols to ensure IP networks and their operations can maintain sufficient level of stability and reliability because IP routing is an essential part of such networks. In this paper, we focus on Open Shortest Path First (OSPF), a widely deployed intra-domain routing protocol. Routers running OSPF advertise their link states on Link State Advertisements (LSAs) as soon as they detect changes in their link states. In IP network operations, it is important for operators to ascertain the location and type of a failure in order to deal with failures adequately. We therefore studied IP network failure identification based on the monitoring of OSPF LSAs. There are three issues to consider in regard to identifying network failures by monitoring LSAs. The first is that multiple LSAs are flooded by a single failure. The second is the LSA delay, and the third is that multiple failures may occur simultaneously. In this paper, we propose a method of network failure identification based on a detailed analysis of OSPF LSA flooding that takes into account the above three issues.

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.

Recently, many kinds of content are being circulated within a great many service-specific overlay networks. When the content is not extremely delay-sensitive, content circulation between wireless terminals can be realized without additional resources by using off-peak periods in wireless access links. In such content circulation, peer-to-peer content multicast is a promising approach to reduce the load on the centralized server. However, to minimize battery drain, each wireless terminal can only forward content to a restricted number of neighboring terminals once it has received the content. This paper proposes an efficient forwarding scheme for peer-to-peer content multicast between the wireless terminals intermittently connected with the backhaul network. In the proposed scheme, a restricted number of terminals with an earlier start time of off-peak periods are selected to forward the content when the number of forwarding hops from the source terminal is less than or equal to a predetermined threshold. In contrast, a restricted number of terminals are selected randomly when the number of forwarding hops exceeds the threshold. This paper clarifies that the proposed hybrid forwarding scheme can multicast the content to many terminals within an arbitrarily restricted period. A guideline to determine the optimum threshold for switching the terminal selection method in the proposed hybrid scheme is derived from simulation results.

As the Internet has become the infrastructure for the global communication, the quality degradation due to network failures and illegal traffic such as DDoS (Distributed Denial of Service) have become a serious problem. In order to solve the problem, a network monitoring system that monitors the traffic of Internet in real time is strongly desired. Traffic monitors that collect the statistics from captured packets play a key roll in the system; however, they are not flexible enough for being used in the rapidly changing Internet. The traditional approach such that a new traffic monitor is developed for a new requirement results in a long turn around time of the development. Therefore, we have proposed a flexible network monitoring system that consists of programmable traffic monitors. Traffic monitors are made programmable by introducing active network techniques; therefore, we call the network monitoring system as the programmable monitor network. This paper describes the implementation of the programmable monitor network and its application to DDoS (Distributed Denial of Service) attack detection.

This paper presents an IP performance management system having the triple frameworks of performance measurement, topology monitoring and data analysis. The system infers the causal location of the performance degradation with a network tomographic approach. Since the Internet is still highly prone to performance deterioration due to congestion, router failure, and so forth, not only detecting performance deterioration, but also monitoring topology and locating the performance-degraded segments in real-time is vital to ensure that Internet Service Providers can mitigate or prevent such performance deterioration. The system is implemented and evaluated through a real-world experiment and its considerable potential for practical network operations is demonstrated.

With the rapid growth of high performance ICT (Information Communication Technologies) devices such as smart phones and tablet PCs, multitasking has become one of the popular ways of using mobile devices. The reasons users have adopted multitask operation are that it reduces the level of dissatisfaction regarding waiting time and makes effective use of time by switching their attention from the waiting process to other content. This is a good solution to the problem of waiting; however, it may cause another problem, which is the increase in traffic volume due to the multiple applications being worked on simultaneously. Thus, an effective method to control throughput adapted to the multitasking situation is required. This paper proposes a transmission rate control method for web browsing that takes multitasking behavior into account and quantitatively demonstrates the effect of service by two different field experiments. The main contribution of this paper is to present a service design process for a new transmission rate control that takes into account human-network interaction based on the human-centered approach. We show that the degree of satisfaction in relation to waiting time did not degrade even when a field trial using a testbed showed that throughput of the background task was reduced by 40%.

This paper proposes a point-to-multipoint (P2MP) Multi-Protocol Label Switching (MPLS) based hierarchical service management system. Traditionally, general management systems deployed in some service providers control MPLS Label Switched Paths (LSPs) (e.g., RSVP-TE and LDP) and services (e.g., L2VPN, L3VPN and IP) separately. In order for dedicated management systems for MPLS LSPs and services to cooperate with each other automatically, a hierarchical service management system has been proposed with the main focus on point-to-point (P2P) TE LSPs in MPLS path management. In the case where P2MP TE LSPs and services are deployed in MPLS networks, the dedicated management systems for P2MP TE LSPs and services must work together automatically. Therefore, this paper proposes a new algorithm that uses a correlation between P2MP TE LSPs and multicast VPN services based on a P2MP MPLS-based hierarchical service management architecture. Also, the capacity and performance of the proposed algorithm are evaluated by simulations, which are actually based on certain real MPLS production networks, and are compared to that of the algorithm for P2P TE LSPs. Results show this system is very scalable within real MPLS production networks. This system, with the automatic correlation, appears to be deployable in real MPLS production networks.

Since congestion is very likely to happen in the Internet, locating congested areas (path segments) along a congested path is vital to appropriate actions by Internet Service Providers to mitigate or prevent network performance degradation. We propose a practical method to locate congested segments by actively measuring one-way end-to-end packet losses on appropriate paths from multiple origins to multiple destinations, using a network tomographic approach. Then we conduct a long-term experiment measuring packet losses on multiple paths over the Japanese commercial Internet. The experimental results indicate that the proposed method is able to precisely locate congested segments. Some findings on congestion over the Japan Internet are also given based on the experiment.

This paper proposes a hierarchical service management system for MPLS network services. Traditionally, general management systems which have been deployed in some service providers control MPLS LSPs (e.g. RSVP-TE, LDP) and services (e.g. L2VPN, L3VPN and IP) separately. If a fault occurs in an MPLS network, the dedicated management system for MPLS LSPs can detect the fault and recognize the state of MPLS LSPs. However, it cannot detect the extent of the impact due to the fault in each service. Furthermore, its own inability to identify the affected customer means it takes some time to identify the affected customers, cooperating manually with the dedicated management system for services. Therefore, this paper proposes a new automatic correlation between MPLS LSPs and each service. In particular, this paper proposes a new algorithm for a correlation between RSVP-TE LSPs and L3VPN services. Simulations are conducted to evaluate the capacity on a correlation table and the performance searching on a correlation table, and results show this system is very scalable within real MPLS production networks. This system, with the automatic correlation, could be sufficiently deployed in real MPLS production networks.

It is important to establish the technology to accommodate best effort TCP/IP traffic over wide area ATM networks. The UBR (Unspecified Bit Rate) service category is the most typical service category for the best effort traffic, especially in the LAN environment. On the other hand, the VBR (Variable Bit Rate) service category with SCD (Selective Cell Discard) option is considered as the service category which is appropriate for wide area networks due to its fairness and minimum guarantee of the cell transmission using not only PCR (Peak Cell Rate) but SCR (Sustainable Cell Rate) and MBS (Maximum Burst Size). However, there is no actual evaluation for such service. We have, therefore, performed the experimental studies on TCP/IP over VBR with SCD along with UBR and VBR without SCD by VC (Virtual Channel) level policing when each TCP connection is mapped to a different VC. Through these experiments, we measured the link utilization of the effective data and the fairness between each obtained TCP throughput during the congestion of the ATM switch. From the results of the link utilization, the value is over 95% under the various conditions. Therefore, even in the case of the cell losses due to SCD or buffer overflow in ATM switch congestion, average throughput is almost the same as the value which equals the trunk line speed divided by the number of the accommodated TCP connections. From the results of the fairness, VBR with SCD per VC is better than UBR and also obtains better TCP throughput than VBR without SCD. Furthermore, to confirm those characteristics more generally, we adopt the accommodated TCP connections not only with the same TCP send/receive socket buffer size but with different sizes. Finally, we discuss the effectiveness between VBR with SCD and the other service categories, such as UBR and ABR (Available Bit Rate) and GFR (Guaranteed Frame Rate), and conclude that VBR with SCD is one of the most suitable ATM service categories for accommodating best effort traffic.

In order to increase the flexibility and the extensibility for packet transmission, the active network approach, that makes network nodes programmable, is very promising. Every packet includes a program specifying behavior of the packet at network nodes, and the network nodes only have the functions to execute programs of the packet. In an active network, network congestion management is achieved more flexibly and intelligently than that in the conventional internet. A number of mechanisms for congestion management are easily developed because various network functions such as QoS routing and congestion detection are easily combined in the same network nodes. By executing different routing protocols that use different levels of the link information, a congestion management scheme can be customized by users. This paper shows how QoS routing and congestion detection achieve a congestion management over an active internetwork system.

Network management is an important issue in maintaining the Internet as an important social infrastructure. Finding excessive consumption of network bandwidth caused by P2P mass flows is especially important. Finding Internet viruses is also an important security issue. Although stream mining techniques seem to be promising techniques to find P2P and Internet viruses, vast network flows prevent the simple application of such techniques. A mining technique which works well with extremely limited memory is required. Also it should have a real-time analysis capability. In this paper, we propose a cache based mining method to realize such a technique. By analyzing the characteristics of the proposed method with real Internet backbone flow data, we show the advantages of the proposed method, i.e. less memory consumption while realizing real-time analysis capability. We also show the fact that we can use the proposed method to find mass flow information from Internet backbone flow data.

Video traffic occupies a major part of current mobile traffic. The characteristics of video traffic are dominated by the behavior of the video application users. This paper uses a state transition diagram to analyze the behavior of video application users on smart phones. Video application users are divided into two categories; keyword search users and initial screen users. They take different first action in video viewing. The result of our analysis shows that the patience of video application users depends on whether they have a specific purpose when they launch a video application or not. Mobile network operators can improve the QoE of video application users by utilizing the results of this study.

In ATM Network, the VBR (Variable Bit Rate) service category is used to accommodate TCP/IP traffic. In an international ATM network with large propagation delay, higher TCP throughput can be obtained by use of window scale option. In order to accommodate TCP traffic with window scale option effectively, it is required to select appropriate values of VBR parameters, i. e. SCR (Sustainable Cell Rate) and MBS (Maximum Burst Size), and to evaluate the impact of UPC (Usage Parameter Control) function on TCP throughput. We have studied those technical issues for the conventional TCP, but the results cannot be applied to TCP traffic with the window scale option due to the TCP terminal performance and the large window size. In this paper, we proposed VBR parameter determination method for TCP with the window scale option and evaluated the values in each condition. These results show that the determined MBS is much smaller than the burst length of TCP segments especially using low performance TCP terminals. Furthermore, we also discuss some experimental results of TCP throughput degradation due to UPC function. It shows that the throughput of TCP with large window size is degraded when the SCR and MBS values used in ATM switch are smaller than the determined values.