Energy efficiency is an important requirement to forth-coming NDN (Named Data Networking) networks and caching inherent to NDN is a main driver of energy reduction in such networks. This paper addresses the research question “Does caching really reduce the energy consumption of the entire network?”. To answer the question, we precisely estimate how caching reduces energy consumption of forth-coming commercial NDN networks by carefully considering configurations of NDN routers. This estimation reveals that energy reduction due to caching depends on energy-proportionality of NDN routers.

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.

In the Mobile IP handoff procedure, mobile node movement is detected from advertisements of foreign agents that differ from previously received advertisement and the new "care-of" address is registered with the home agent. However, user packets are not forwarded to the new foreign agent until a registration is completed and this interruption may degrade the quality of service especially in real-time applications such as audio and video, or may lower the TCP throughput due to retransmission timeout. To tackle these issues, we propose a new low latency handoff method, where access points used in a wireless LAN environment and a dedicated MAC bridge are jointly used to alleviate packet loss without altering the Mobile IP specifications. In this paper, we present design architecture of the proposed method and evaluate its performance in an actual network environment to verify the effectiveness of our approach.

Although TCP is widely used in the Internet, new specifications are still proposed and implemented. In the circumstance above, it is highly possible that some errors are detected on the communication between new and old implementations. Several test tools were developed so far. However, they do not have enough functions to allow test operators to modify test sequences suitable for their test purposes. We have developed a TCP tester which generates test sequence using test scenario. The tester performs exceptional TCP protocol behavior only when the condition specified in the test scenario is satisfied. Otherwise, it performs normal TCP behavior. The tester is implemented by modifying TCP module of NetBSD with SACK code developed by Pittsburgh Supercomputing Center. We have also evaluated implementations of congestion control and SACK algorithms using the tester.

This paper presents a protocol verification method which verifies that the behaviors of a protocol meet requirements. In this method, a protocol specification is expressed as Extended Finite State Machines (EFSM's) that can handle variables, and requirements are expressed using a branching-time temporal logic for a concise and unambiguous description. Using the acyclic expansion algorithm extended such that it can deal with EFSM's, the verification method first generates a state transition graph consisting of executable transitions for each process. Then a branching-time temporal logic formula representing a requirement is evaluated on one of the generated graphs which is relevant to the requirement. An executable state transition graph for each process is much smaller than a global state transition graph which has been used in the conventional verification techniques to represent the behaviors of the whole protocol system consisting of all processes. The computation for generating the graphs is also reduced to much extent for a large complex protocol. As a result, the presented method achieves efficient verification for requirements regarding a state of a process, transmission and reception of messages by a process, varibales of a process and sequences that interact among processes. The validity of the method is illustrated in the paper by the verification of a path-updating protocol for requirements such as process state reachability or fair termination among processes.

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.

It has become more important to reduce the protocol implementation costs as the functions of protocols have become more abundant. The protocol implementation tools which automatically generate a protocol program from a specification described by an FDT (Formal Description Technique) are very promising. Selecting SDL as a target FDT, we have developed an SDL-based protocol implementation tool which consists of a process scheduler and a compiler. Since the efficient SDL process execution is a key to generating the high-speed program, the scheduler is introduced. It provides the mechanism which executes SDL processes concurrently as light-weight-processes. It optimizes so that as few context switches take places as possible. The compiler converts as many kinds of SDL functions whose behaviors can be determined at compile time into programming language statements as possible. These elaborations are so successful that the tool can generate an efficient program. The OSI Transport protocol class 0 program generated by the compiler can process more than 500 packets per second on a 6MIPS workstation.

Internet of Things (IoT) devices, which have different characteristics in mobility and communication patterns from traditional mobile devices such as cellular phones, have come into existence as a new type of mobile devices. A strict mobility management scheme for providing highly mobile devices with seamless access is over-engineered for IoT devices' mobility management. We revisit current mobility management schemes for wireless mobile networks based on identifier/locator separation. In this paper, we focus on IoT communication patterns, and propose a new routing-based mobility scheme for them. Our scheme adopts routing information aggregation scheme using the Bloom Filter as a data structure to store routing information. We clarify the effectiveness of our scheme in IoT environments with a large number of IoT devices, and discuss its deployment issues.

Most of current real time video retrieval systems use video transfer protocols such that servers simply transmit video packets in the same rate as clients play them. If any packets are corrupted during transmission, they will be lost and cannot be recovered by retransmission. In video retrieval systems, however, teh video data are stored in servers and clients can prefetch them prior to playing. So, it might be possible for the video retrieval systems to make corrupted video packets retransmitted before the play-out dead line. But the application of existing reliable protocols causes problems such that, if a packet does not arrive before the dead line due to retransmission, the packets following it will not be delivered to the upper layer even if they have already arrived. In this paper, we discuss how to apply reliable protocols to real time video retrieval systems and propose an new real time video transfer protocol over ATM network, which provides the video data prefetch, the flow control for video buffer, the selective retransmission with skipping function for video packets late for the play-out dead line, and the resynchronization function for video buffer. We have implemented an experimental system using our protocol and evaluated the performance. The results of performance evaluation shows that the proposed protocol decreases the number of unplayed video data largely when transmission errors are inserted in an ATM network.

The current interdomain routing protocol, BGP, is not resilient to a path failure due to its single-path and slowly-converging route calculation. This paper proposes a novel approach to improve the resilience of the interdomain communication by enabling a set of ASes to form an alliance for themselves. The alliance members cooperatively discover a set of disjoint paths using not only the best routes advertised via BGP but also the ones unadvertised. Since such a set of disjoint paths are unlikely to share a link failure, a member AS can provide a pair of the other members with a transit to circumvent the failure. We evaluate how many disjoint paths we could discover from both advertised and hidden (unadvertised) routes by analyzing publicly available BGP route data. Our feasibility study indicates that an alliance of ASes can establish a set of disjoint paths between arbitrary pair of its alliance members with high probability to improve the resilience of interdomain routing among the members.

Nowadays, many IP based communication systems are connected by high-speed networks to realize high-speed TCP/IP communication. However, since the mechanism of TCP/IP protocols is based on a best effort service, the quality of the communication may change by the time or the route of the traffic. Therefore, it is important for network providers to investigate the quality of their users' communication. In order to analyze the traffic on an Internet backbone, the realtime analysis is one of the important factors. So far, several tools were developed for the purpose of the traffic measurement. However, none of them can analyze sufficient statistics to evaluate the quality for each end user's communication in realtime. Therefore, we have designed and implemented a performance monitor, which collects the statistics representing the performance such as TCP throughput while capturing the traffic. The statistics are collected by every pair of IP addresses and by every application. The monitor also provides the function to analyze effectively for the statistics records such as sorting and filtering of the records and the graphical user interface to operate the software tool. This paper describes the design and implementation of the performance monitor.

The Internet was designed for academic use more than 40 years ago. After having been used commercially, many unpredictable requirements have emerged, including mobility, security and content distribution. In addition, the Internet has become so ossified that fulfilling new requirements is difficult. Instead of developing ad-hoc solutions, re-designing clean-slate Internet architectures has become a key research challenge in networking communities. This survey paper addresses key research issues and then introduces ongoing research projects from Japan, the United States and the European Union.

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.

This paper presents a realization of our IP based realtime VLBI (Very Long Baseline Interferometer) observation testbed with the highest sensitivity in the world. Today's rapid deployment of high-speed wide area networks will give a major breakthrough in VLBI astronomy in terms of its observational sensitivity and immediateness. VLBI requires huge amount of data transfer from several radio telescopes located separately each other for calculating cross-correlation. High-speed networks can be applied to such data transfer instead of conventional magnetical tape recording and physical transportation, which cause a serious performance bottleneck. We have newly designed and implemented a special component named gigabit network access node, which can exchange 2.048 Gbps telescope data through a 2.488 Gbps OC-48c/STM-16c SONET/SDH link. We have also constructed the world's first multi-gigabit-rate VLBI observation testbed using actual high-speed wide area optical networks and successfully conducted several real observations.

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.

In forthcoming sensor networks, a multitude of sensor nodes deployed over a large geographical area for monitoring traffic, climate, etc. are expected to become an inevitable infrastructure. Clustering algorithms play an important role in aggregating a large volume of data that are produced continuously by the huge number of sensor nodes. In such networks, equal-sized multi-hop clusters which include an equal number of nodes are useful for efficiency and resiliency. In addition, scalability is important in such large-scale networks. In this paper, we mathematically design a decentralized equal-sized clustering algorithm using a partial differential equation based on the Fourier transform technique, and then design its protocol by discretizing the equation. We evaluated through simulations the equality of cluster sizes and the resiliency against packet loss and node failure in two-dimensional perturbed grid topologies.

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.

In the 5G era, centralized mobility management raises the issue of traffic concentration on the mobility anchor. Distributed mobility management is expected to be a solution for this issue, as it moves mobility anchor functions to multiple edge routers. However, it incurs path stretch and redundant traffic on the backhaul links. Although these issues were not considered important in the 3G/4G era, they are expected to be a serious problem in the 5G era. In this paper, we design a routing-based mobility management mechanism to address the above problems. The mechanism integrates distributed routing with Bloom Filters and an anchor-less scheme where edge routers work as mobility anchors. Simulations show that the proposed mechanism achieves a good balance between redundant traffic on the backhaul links and routing overhead.

Internet of Things (IoT) devices deployed in urban areas are seen as data sources for urban sensing IoT applications. Since installing cellular interfaces on a huge number of IoT devices is expensive, we propose to use a user equipment (UE) device with a local wireless interface as a mobile IoT gateway for fixed IoT devices. In this paper, we design a new mobile architecture based on cellular networks to accommodate non-cellular fixed IoT devices by UE devices working as IoT gateways. One key feature is that our architecture leverages proximity services (ProSe) to discover relay UE devices with low overhead in terms of discovery messages. Through simulation studies, we clarify the feasibility of our architecture including the relay UE discovery mechanism in urban areas.

In the gauge of IMT-2000, Mobile IP based location management is standardized. The cellular network of IMT-2000 holds many users and covers a large area. In large-scale Mobile IP networks, such as IMT-2000, a foreign agent may manage many mobile nodes by the visitor list and its forwarding performance degrades owing to the list search overload. In this paper, we propose a fast data transfer method in Mobile IP networks by reducing the load of the search processes on the foreign agent using cooperation between mobility agents. In our experiments, we measured the packet forwarding performance and the packet forwarding process time. The results indicate the possibility of the performance degradation owing to the part of the registered mobile nodes in the visitor list. We verify that the foreign agent using our method can forward all the received packets on Fast Ethernet LAN when one million entries are recorded in the visitor list.