Many ATM switching modules with high performance have been proposed and analyzed. A development of a large scale ATM switching system (e.g., used as a central switch) is the key to realization of the broadband ISDN. However, the dimension of ATM switching ICs is limited by the technological and physical constraints on VLSI. A multistage switching configuration is one of the promising configurations for a large scale ATM switch. In this paper, we treat a 3-stage switching configuration with no internal bufferes; i.e., bufferless switches are employed at the first and second stages, and output buffered switches at the third stage. A short-term cell loss probability is analyzed in order to examine the influence of bursty traffic on performance of the bufferless switch used at the first two stages. Furthermore, we propose a 4-stage switching configuration with traffic distributors added at the first stage. This switch provides more paths between a pair of input and output ports than the 3-stage switching configuration mentioned above. A few schemes to distribute cells are compared. It is shown that the distributor successfully reduces the deterioration of cell loss probability due to bursty traffic by splitting incoming cells into several switching modules.

It is now becoming important for mobile cellular networks to accommodate all kinds of Internet of Things (IoT) communications. However, the contention-based random access and radio resource allocation used in traditional cellular networks, which are optimized mainly for human communications, cannot efficiently handle large-scale IoT communications. For this reason, standardization activities have emerged to serve IoT devices such as Cellular-IoT (C-IoT). However, few studies have been directed at evaluating the performance of C-IoT communications with periodic data transmissions, despite this being a common characteristic of many IoT communications. In this paper, we give the performance analysis results of mobile cellular networks supporting periodic C-IoT communications, focusing on the performance differences between LTE and Narrowband-IoT (NB-IoT) networks. To achieve this, we first construct an analysis model for end-to-end performance of both the control plane and data plane, including random access procedures, radio resource allocation, establishing bearers in the Evolved Packet Core network, and user-data transmissions. In addition, we include the impact of the immediate release of the radio resources proposed in 3GPP. Numerical evaluations show that NB-IoT can support more IoT devices than LTE, up to 8.7 times more, but imposes a significant delay in data transmissions. We also confirm that the immediate release of radio resources increases the network capacity by up to 17.7 times.

In CDMA mobile cellular systems, wireless quality is improved by soft handoff techniques. However, it requires to hold multiple channels of cells, which is likely to increase call blocking at wired channels. It is therefore necessary to consider the entire system including the wired and wireless portions of systems for investigating an effectiveness of the soft handoff. In this paper, we also clarify the effect of interference power from mobile stations that are not in the soft handoff because of lack of wired channels. In the analysis, we model three-way soft handoff which has not been considered in past researches. We also show the effect of a call admission control to wireless quality.

Rate based control is a promising way to achieve an efficient packet transmission especially in high speed packet switching networks where round trip delay is much larger than packet transmission time. Although inappropriate tuning for the parameters, increasing and decreasing factors, of the rate control function causes the performance degradation, most of the previous works so far have not studied the effect of the parameters on the performance. In this paper, we investigate the effect of the rate control parameters on the throughput under the condition that the packet loss probability is kept below a specific value, say 10-6. For this purpose, we build a queueing model and carry out a transient analysis to examine the dynamic behavior of the queue length at an intermediate node in a high speed network suffering from large propagation delay. Numerical examples exploit the optimal value of the parameters when one or two source-destination pairs transmit packets. We also discuss the effect of the propagation delay on the performance. Our model can be applicable to investigate the performance of various kinds of rate-based congestion control when the relation between the congestion measure and the rate control mechanism is given explicitly.

In the present paper, ImTCP-bg, a new background TCP data transfer mechanism that uses an inline network measurement technique, is proposed. ImTCP-bg sets the upper limit of the congestion window size of the sender TCP based on the results of the inline network measurement, which measures the available bandwidth of the network path between the sender and receiver hosts. ImTCP-bg can provide background data transfer without affecting the foreground traffic, whereas previous methods cannot avoid network congestion. ImTCP-bg also employs an enhanced RTT-based mechanism so that ImTCP-bg can detect and resolve network congestion, even when reliable measurement results cannot be obtained. The performance of ImTCP-bg is investigated through simulations, and the effectiveness of ImTCP-bg in terms of the degree of interference with foreground traffic and the link bandwidth utilization is also investigated.

Overlay networks are expected to be a promising technology for the realization of QoS (Quality of Service) control. Overlay networks have recently attracted considerable attention due to the following advantages: a new service can be developed in a short duration and it can be started with a low cost. The definition and necessity of the overlay network is described, and the classification of various current and future overlay networks, particularly according to the QoS feature, is attempted. In order to realize QoS control, it is considered that routing overlay and session overlay are promising solutions. In particular, session and overlay networks are explained in detail since new TCP protocols for QoS instead of current TCP protocols that control congestion in the Internet can be used within overlay networks. However, many open issues such as scalability still need further research and development although overlay networks have many attractive features and possess the potential to become a platform for the deployment of new services.

When computing resources are consolidated in a few huge data centers, a massive amount of data is transferred to each data center over a wide area network (WAN). This results in increased power consumption in the WAN. A distributed computing network (DCN), such as a content delivery network, can reduce the traffic from/to the data center, thereby decreasing the power consumed in the WAN. In this paper, we focus on the energy-saving aspect of the DCN and evaluate its effectiveness, especially considering traffic locality, i.e., the amount of traffic related to the geographical vicinity. We first formulate the problem of optimizing the DCN power consumption and describe the DCN in detail. Then, numerical evaluations show that, when there is strong traffic locality and the router has ideal energy proportionality, the system's power consumption is reduced to about 50% of the power consumed in the case where a DCN is not used; moreover, this advantage becomes even larger (up to about 30%) when the data center is located farthest from the center of the network topology.

We investigate performance of TCP protocol over ATM networks by using a simulation technique. As the ATM layer, we consider (1) rate-based control of the ABR service class and (2) an EPD (Early Packet Discard) technique applied to the UBR service class and (3) and EPD with per-VC accounting for fairness enhancement applied to the UBR service class. In comparison, we adopt a multi-hop network model where the multiple ATM switches are interconnected. In such a network, unfairness among connections is a possible cause of the problem due to differences of the number of hops and/or the round trip times among connections. Simulation results show that the rate-based control method of ABR achieves highest throughput and best fairness in most circumstances. However, the performance of TCP over ABR is degraded once the cell loss takes place due to the inappropriate control parameter setting. To avoid this performance degradation, we investigate the appropriate parameter set suitable to TCP on ABR service. As a result, parameter tuning can improve the performance of TCP over ABR, but limited. We therefore consider TCP over ABR with EPD enhancement where the EPD technique is incorporated into ABR. We last consider the multimedia network environment, where the VBR traffic exists in the network in addition to the ABR/UBR traffic. By this, we investigate an applicability of the above observations to a more generic model. Through simulation experiments, we find that the similar results can be obtained, but it is also shown that parameters of the rate-based congestion control must be chosen carefully by taking into account the existence of VBR traffic. For this, we discuss the method to determine the appropriate control parameters.

In this paper, we propose a new multicast routing algorithm for constructing the delay-constrained minimal spanning tree in the VP-based ATM networks, in which we consider the efficiency even in the case where the destination dynamically joins/departs the multicast connection. For constructing the delay-constrained spanning tree, we first generate a reduced network consisting of only VCX nodes from a given ATM network, originally consisting of VPX/VCX nodes. Then, we obtain the delay-constrained spanning tree with a minimal tree cost on the reduced network by using our proposed heuristic algorithm. Through numerical examples, we show that our dynamic multicast routing algorithm can provide an efficient usage of network resources when the membership nodes frequently changes during the lifetime of a multicast connection, while the existing multicast routing algorithm may be useful for constructing the multicast tree with a static nature of destination nodes. We also demonstrate that more cost-saving can be expected in dense networks when applying our proposed algorithm.

This article introduces a self-organizing model which builds the topology of a DHT mapping system for ICN. Due to its self-organizing operation and low average degree of maintenance, the management overhead of the system is reduced dramatically, which yields inherent scalability. The proposed model can improve latency by around 10% compared to an existing approach which has a near optimal average distance when the number of nodes and degree are given. In particular, its operation is simple which eases maintenance concerns. Moreover, we analyze the model theoretically to provide a deeper understanding of the proposal.

Content-centric networking (CCN) is an innovative network architecture that is being considered as a successor to the Internet. In recent years, CCN has received increasing attention from all over the world because its novel technologies (e.g., caching, multicast, aggregating requests) and communication based on names that act as addresses for content have the potential to resolve various problems facing the Internet. To implement these technologies, however, requires routers with performance far superior to that offered by today's Internet routers. Although many researchers have proposed various router components, such as caching and name lookup mechanisms, there are few router-level designs incorporating all the necessary components. The design and evaluation of a complete router is the primary contribution of this paper. We provide a concrete hardware design for a router model that uses three basic tables — forwarding information base (FIB), pending interest table (PIT), and content store (CS) — and incorporates two entities that we propose. One of these entities is the name lookup entity, which looks up a name address within a few cycles from content-addressable memory by use of a Bloom filter; the other is the interest count entity, which counts interest packets that require certain content and selects content worth caching. Our contributions are (1) presenting a proper algorithm for looking up and matching name addresses in CCN communication, (2) proposing a method to process CCN packets in a way that achieves high throughput and very low latency, and (3) demonstrating feasible performance and cost on the basis of a concrete hardware design using distributed content-addressable memory.

In a multi-tenant data center, nodes and links of tenants' virtual networks (VNs) share a single component of the physical substrate network (SN). The failure of a single SN component can thereby cause the simultaneous failures of multiple nodes and links in a single VN; this complex of failures must significantly disrupt the services offered on the VN. In the present paper, we clarify how the fault tolerance of each VN is affected by a single SN failure, especially from the perspective of VN allocation in the SN. We propose a VN allocation model for multi-tenant data centers and formulate a problem that deals with the bandwidth loss in a single VN due a single SN failure. We conduct numerical simulations (with the setting that has 1.7×108bit/s bandwidth demand on each VN, (denoted by Ci)). When each node in each VN is scattered and mapped to an individual physical server, each VN can have the minimum bandwidth loss (5.3×102bit/s (3.0×10-6×Ci)) but the maximum required bandwidth between physical servers (1.0×109bit/s (5.7×Ci)). The balance between the bandwidth loss and the required physical resources can be optimized by assigning every four nodes of each VN to an individual physical server, meaning that we minimize the bandwidth loss without over-provisioning of core switches.

In the big data era, messaging systems are required to process large volumes of message traffic with high scalability and availability. However, conventional systems have two issues regarding availability. The first issue is that failover processing itself has a risk of failure. The second issue is to find a trade-off between consistency and availability. We propose a resilient messaging system based on a distributed in-memory key-value store (KVS). Its servers are interconnected with each other and messages are distributed to multiple servers in normal processing state. This architecture can continue messaging services wherever in the messaging system server/process failures occur without using failover processing. Furthermore, we propose two methods for improved resilience: the round-robin method with a slowdown KVS exclusion and the two logical KVS counter-rotating rings to provide short-term-availability in the messaging system. Evaluation results demonstrate that the proposed system can continue service without failover processing. Compared with the conventional method, our proposed distribution method reduced 92% of error responses to clients caused by server failures.

By deploying hundreds or thousands of microsensors and organizing a network of them, one can monitor and obtain information of environments or objects for use by users, applications, or systems. Since sensor nodes are usually powered by batteries, an energy-efficient data gathering scheme is needed to prolong the lifetime of the sensor network. In this paper, we propose a novel scheme for data gathering where sensor information periodically propagates from the edge of a sensor network to a base station as the propagation forms a concentric circle. Since it is unrealistic to assume any type of centralized control in a sensor network whose nodes are deployed in an uncontrolled way, a sensor node independently determines the cycle and the timing at which it emits sensor information in synchrony by observing the radio signals emitted by sensor nodes in its vicinity. For this purpose, we adopt a pulse-coupled oscillator model based on biological mutual synchronization such as that used by flashing fireflies, chirping crickets, and pacemaker cells. We conducted simulation experiments, and verified that our scheme could gather sensor information in a fully-distributed, self-organizing, robust, adaptive, scalable, and energy-efficient manner.

Continuous and explosive growth of the Internet has shown that current TCP mechanisms can obstruct efficient use of high-speed, long-delay networks. To address this problem we propose an enhanced transport-layer protocol called gHSTCP, based on HighSpeed TCP proposed by Sally Floyd. It uses two modes in the congestion avoidance phase based on the changing trend of RTT. Simulation results show gHSTCP can significantly improve performance in mixed environments, in terms of throughput and fairness against the traditional TCP Reno flows. However, the performance improvement is limited due to the nature of TailDrop router, and the RED/ARED routers can not alleviate the problem completely. Therefore, we present a modified version of Adaptive RED, called gARED, directed at the problem of simultaneous packet drops by multiple flows in high speed networks. gARED can eliminate weaknesses found in Adaptive RED by monitoring the trend in variation of the average queue length of the router buffer. Our approach, combining gARED and gHSTCP, is quite effective and fair to competing traffic than Adaptive RED with HighSpeed TCP.

Proportional fair bandwidth allocation in packet switches is a fundamental issue to provide quality of service (QoS) support in IP networks. In input-queued switches, packet-mode scheduling delivers all the segments of a packet contiguously from the input port to the output port, thus greatly simplifying the design of packet reassembly modules and yielding performance advantage over cell-mode scheduling under certain conditions [1]. One of the important issues of packet-mode scheduling is how to achieve fair bandwidth allocation among flows with different packet sizes. This paper presents an algorithm called packet-mode fair scheduling (pFS) that guarantees each flow a bandwidth proportional to its reservation regardless of the packet size distribution and the system load. Simulations show that our approach achieves good fairness as well as high throughput and low packet delay. Compared to algorithms without fairness mechanism, pFS yields significant performance improvement in terms of average packet delay when the traffic is heterogeneous. A hardware implementation is presented to show that the proposed algorithm has low complexity and the computation can be completed in a single clock cycle, which makes pFS applicable to high-speed switches.

Although the use of software-defined networking (SDN) enables routes of packets to be controlled with finer granularity (down to the individual flow level) by using traffic engineering (TE) and thereby enables better balancing of the link loads, the corresponding increase in the number of states that need to be managed at routers and controller is problematic in large-scale networks. Aggregating flows into macro flows and assigning routes by macro flow should be an effective approach to solving this problem. However, when macro flows are constructed as TE targets, variations of traffic rates in each macro flow should be minimized to improve route stability. We propose two methods for generating macro flows: one is based on a greedy algorithm that minimizes the variation in rates, and the other clusters micro flows with similar traffic variation patterns into groups and optimizes the traffic ratio of extracted from each cluster to aggregate into each macro flow. Evaluation using traffic demand matrixes for 48 hours of Internet2 traffic demonstrated that the proposed methods can reduce the number of TE targets to about 1/50 ∼ 1/400 without degrading the link-load balancing effect of TE.

Robustness is one of the significant properties in wireless sensor networks because sensor nodes and wireless links are subjected to frequent failures. Once these failures occur, system performance falls into critical condition due to increases in traffic and losses of connectivity and reachability. Most of the existing studies on sensor networks, however, do not conduct quantitative evaluation on robustness and do not discuss what brings in robustness. In this paper, we define and evaluate robustness of wireless sensor networks and show how to improve them. By computer simulation, we show that receiver-initiated MAC protocols are more robust than sender-initiated ones and a simple detour-routing algorithm has more than tripled robustness than the simple minimum-hop routing algorithm.

The delay/disruption tolerant network (DTN) has been researched actively in the last years because of its high applicability to ubiquitous network services such as sensor networks and intelligent transport system (ITS) networks. An efficient data forwarding method for those network services is one of the key components in DTN due to the limitation of wireless network resources. This paper proposes a new DTN scheme for vehicle network systems by introducing the parameter, “approach ratio”, which represents node movement history. The proposal utilizes passive copy strategy, where nodes within one hop area of packet forwarders receive, copy and store packets (namely, passive copies) for future forwarding, in order to obtain higher delivery rate and lower delivery delay whilst suppressing the network resource consumption. Depending on its approach ratio, a node with passive copy decides whether it forwards the passive copy or not by referring to the approach ratio threshold. The approach ratio allows our proposal to adjust the property of both single-copy type scheme, that can lower network resource consumption, and multi-copy type scheme, that can enhance the performance of delivery rate and delay time. In simulation evaluation, the proposal is compared with three typical existing schemes with respect to network consumption, delivery rate and delivery delay. Our proposal shows the superior performance regarding the targeted purpose. It is shown that the approach ratio plays the significant role to obtain the higher delivery rate and lower delay time, while keeping network resource consumption lower.

As a result of the rapid development of the Internet in recent years, network security has become an urgent issue. Distributed denial of service (DDoS) attacks are one of the most serious security issues. In particular, 60 percent of the DDoS attacks found on the Internet are TCP attacks, including SYN flood attacks. In this paper, we propose adaptive timer-based countermeasures against SYN flood attacks. Our proposal utilizes the concept of soft-state protocols that are widely used for resource management on the Internet. In order to avoid deadlock, a server releases resources using a time-out mechanism without any explicit requests from its clients. If we change the value of the timer in accordance with the network conditions, we can add more flexibility to the soft-state protocols. The timer is used to manage the resources assigned to half-open connections in a TCP 3-way handshake mechanism, and its value is determined adaptively according to the network conditions. In addition, we report our simulation results to show the effectiveness of our approach.