We propose a method of IP traffic management where the TCP performance at a bottleneck link is estimated from monitored data about the behavior of the number of active flows versus link utilization, which are both easy to measure. This method is based on our findings that (i) TCP performance remains constant as long as the link utilization is below some threshold value, but becomes degraded when it exceeds this value and (ii) the number of active flows increases linearly with link utilization up to the same value, and the increase becomes nonlinear above it. Though this threshold may vary depending on traffic/network conditions, our method requires neither predetermination of a threshold on the basis of assumed traffic conditions nor direct measurement of TCP performance.

Hierarchical Mobile IPv6 (HMIPv6) has been proposed to improve the performance capability of Mobile IPv6 at handover. In HMIPv6, local entities named Mobility Anchor Points (MAPs) are distributed throughout a network to localize the management of intra-domain mobility. In particular, multi-layered MAP has been proposed to improve performance. MAPs reduce the number of Binding Updates to the Home Agent and improve the communication quality at handover. These conventional methods that manage a multi-layered MAP cannot, however, select an appropriate MAP because they use the virtual mobility speed. As a result, they increase the signaling traffic in a multi-layered MAP. Moreover, they may cause the load to concentrate at a specific MAP. In this paper, we propose a location management method for Hierarchical Mobile IPv6 using the MN's mobile history. In this method, when a MN performs a handover, the Access Router calculates the area-covered rate of each upper MAP from the MN's mobile history and selects the MAP that best manages the MN in accordance with its rate. Thus, the proposed method reduces both the number of Binding Updates to the Home Agent and the signaling traffic because it reduces the frequency of changing the MAP. We evaluate the performance of the proposed method by simulation.

User Generated Content (UGC) VoD services such as YouTube are becoming more and more popular, and their maintenance costs are growing as well. Many P2P solutions have been proposed to reduce server load in such systems, but almost all of them focus on the single-video approach, which only has limited effect on the systems serving short videos such as UGC. The purpose of this paper is to investigate the potential of an alternative approach, the multi-video approach, and we use a very simple method called collaborative caching to show that methods using the multi-video approach are generally more suitable for current UGC VoD systems. We also study the influence of the major design factors through simulations and provide guidelines for efficiently building systems with this method.

Measuring the duration of flow of a TCP connection in an end-to-end path is important in the management of network performance, and when this is done, an administrator can manage the quality of the networks using the α percentile of the distribution. We propose a method of estimating the distribution of flow duration in an end-to-end path through active measurement using a small degree of traffic. This method of estimation is based on traffic characteristics that are observed in measuring traffic in actual networks. It imposes little additional load on networks and the time in computation required to estimate the distribution is also short. The distribution to be estimated is assumed as a log-normal distribution, and the mean and variance of the distribution of a target file size is estimated by using results of active measurements. Means and variances in various file sizes are estimated through the linear relationships between these values (or their logarithms) and file size. We also provide numerical examples using actual traffic data.

With conventional dynamic routing algorithms, many query messages are required in a distributed environment for efficient multicast routing of any traffic volume. We have developed a dynamic routing algorithm that uses a predetermined path search in which an appropriate multicast path is dynamically constructed by searching only a few nodes. This algorithm can construct an efficient multicast tree for any traffic volume. Simulation has shown that the proposed algorithm is advantageous compared with conventional dynamic routing algorithms when nodes are added to or removed from the multicast group during steady-state simulation.

An architecture is presented for efficient and reliable delivery of multimedia contents from a primary center (PC) to secondary centers (SCs). Requested contents are delivered from the PC to the SCs through a satellite broadcast channel, or from one SC to another SC through a terrestrial channel. Cycling methods are presented that enable sharing of the contents directory of each SC. Several fundamental models and algorithms are introduced for possible consideration during the planning and design of a contents-delivery system. Simulation has shown that using both satellite broadcast and terrestrial channels for contents delivery is superior in terms of cost to the conventional use of only a satellite network.

In recent years elephant flows are increasing by expansion of peer-to-peer (P2P) applications on the Internet. As a result, bandwidth is occupied by specific users triggering unfair resource allocation. The main packet-scheduling mechanism currently employed is first-in first-out (FIFO) where the available bandwidth of short flows is limited by elephant flows. Least attained service (LAS), which decides transfer priority of packets by the total amount of transferred data in all flows, was proposed to solve this problem. However, routers with LAS limit flows with large amount of transferred data even if they are low-rate. Therefore, it is necessary to improve the quality of low-rate flows with long holding times such as voice over Internet protocol (VoIP) applications. This paper proposes rate-based priority control (RBPC), which calculates the flow rate and control the priority by using it. Our proposed method can transfer short flows and low-rate flows in advance. Moreover, its fair performance is shown through simulations.

Distributed Web caching allows multiple clients to quickly access a pool of popular Web pages. Conventional distributed Web caching schemes, e. g. , the Internet cache protocol and hash routing, require the sending of many query messages among cache servers and/or impose a large load on the cache servers when they are widely dispersed. To overcome these problems, we propose a hash-based query caching method using both a hash function and a query caching method. This method can find cached objects among several cache servers by using only one query message, enabling the construction of an efficient large-scale distributed Web cache server. Compared to conventional methods, this method reduces cache server overhead and object retrieval latency.

This paper reports an overload control method for the Intelligent Network (IN). The IN, which is being investigated as a future communication network, facilitates both rapid introduction of new services and easy modification of existing services. In the IN, the call processing functions and data needed to achieve IN services are distributed over several nodes. Therefore, traffic demand for the various services may cause varying patterns of node overloads. It is therefore important to develop effective overload control methods and to evaluate their characteristics. We propose an overload control method and evaluate its characteristics in comparison with other methods under various overload traffic patterns with a network simulator that models all nodes and their relationships in the IN. In particular, we focus on three aspects of overload control: how can high throughput be maintained, how can an overloaded node be stabilized, and how can fair access be guaranteed.

In ad hoc networks, broadcast forwarding protocols called OR (opportunistic routing) have been proposed to gain path diversity for higher packet delivery rates and shorter end-to-end delays. In general backoff-based OR protocols, each receiver autonomously makes a forwarding decision by using certain metrics to determine if a random backoff time is to be applied. However, each forwarder candidate must wait for the expiration of the backoff timer before forwarding a packet. Moreover, they cannot gain path diversity if the forwarding path includes local sparse areas, and this degrades performance as it strongly depends on the terminal density. In this paper, we propose a novel OR protocol called PRIOR (prioritized forwarding for opportunistic routing). In PRIOR, a terminal, called a prioritized forwarder and which forwards packets without using a backoff time, is selected from among the neighbours. In addition, PRIOR uses lightweight hop-by-hop retransmission control to mitigate the effect of terminal density. Moreover, we introduce an enhancement to PRIOR to reduce unnecessary forwarding by using an explicit acknowledgement. We evaluate PRIOR in comparison with conventional protocols in computer simulations.

This paper proposes ABdis, a new active measurement method for estimating the available bandwidth on a communication network path. Due to the recent explosion in multimedia applications, it is becoming increasingly important to manage network QoS, and tools that can measure network quality precisely are necessary to do this. Many conventional active measurement methods/tools, however, can measure/estimate only the average of the available bandwidth in a given period but cannot measure its distribution. If the distribution of the bandwidth over short intervals can be measured, the information would be useful for network management, proxy selection, and end-to-end admission control. We propose an end-to-end active measurement method called ABdis, which can estimate the distribution of the available bandwidth in a network path. ABdis employs multiple probes having different rates and a parameter-matching technique for estimating distribution. Furthermore, we present the results of simulations and verify ABdis's performance under various conditions by changing probe parameters, amount of cross traffic, and network models.

A leaky-bucket-with-gate algorithm is proposed to control connection-setup congestion in telecommunication networks providing multimedia services, in place of the call-gapping algorithm used in telephone networks. Multimedia services may use more than one connection simultaneously, while standard telephone services use only one connection at a time. A set of connections used to construct a multimedia service is called a correlated connection group, and the setup requests of such a group form correlated request group. A correlated request group is assumed to be accepted into the network only when all the connection-setup requests for the group are accepted. In this paper, the proposed leaky-bucket-with-gate algorithm, a pure leaky-bucket algorithm, and a call-gapping algorithm are evaluated by simulating traffic with a mix of correlated and uncorrelated connection-setup requests, which models setup requests for video conferencing and telephone services. The simulation results show that the proposed algorithm accepts correlated request groups more efficiently than the pure leaky-bucket and call-gapping algorithms under the simulated traffic conditions, except when the interarrival time in a correlated request group is longer than the acceptance interval. We also present queueing analysis for determining the control parameters in the proposed algorithm. Implementation of this algorithm will facilitate the handling of both setup request traffic for correlated connection groups and for uncorrelated connections in multimedia networks.

Satellite-terrestrial (ST) networks, in which many nodes are interconnected by both satellite and terrestrial networks, can efficiently support multicast services. This is because satellite broadcasting is suitable for a large multicast group and a terrestrial network is suitable for a small multicast group. An ST network requires a multicast routing algorithm that can select the appropriate satellite and terrestrial routes. Conventional dynamic routing algorithms for terrestrial networks cannot construct an efficient multicast routing tree because they basically select a less-expensive route when a node is added. We have developed a dynamic routing algorithm, a virtual-cost-based algorithm, for ST networks that selects the route to use according to the multicast group size when a node is added to the group. Simulation showed that the proposed algorithm is advantageous when nodes are added to or removed from the multicast group during steady-state simulation.

Measuring traffic dynamics during intervals of a few seconds is important in the management of network performance. If the distribution of average traffic volume during a few seconds is measured, an administrator can manage the quality of the networks using the α percentile of the distribution. We propose a method of estimating the distribution of traffic volume during short intervals, such as a few seconds, by using only traffic information from the management information base (MIB) of routers or switches. This estimation method is based on traffic characteristics that are observed in traffic measurements in actual networks. It imposes little additional load on routers or switches and the computation time required to estimate the distribution is also short. Numerical examples using actual traffic data are also given.

Analyzing short-interval-traffic behaviors is important for network performance management to realize high quality multimedia applications. However, it is difficult to measure short-interval-traffic volumes because there are complications in collecting short-interval-traffic data from routers. An example is a heavy load on routers or inaccurate measurement by the short-polling interval; it even demands expensive measurement tools. To resolve these disadvantages, an estimating method of short-interval-traffic distribution (EMSIT) has been proposed. This method estimates short-interval-traffic distributions using MIB (Management Information Base) data, which collects traffic volumes in cycles of several minutes. In this paper, we propose a new estimation method (EMSIT-LD) based on EMSIT, which applies to the case of long-term-traffic dynamics. We evaluate it using computer simulations and actual traffic data.

A huge amount of content exists on the Internet, and contents from mobile devices are also present. Growth of the Internet of Things (IoT) is further accelerating this trend. Content oriented networks have been proposed as a new network architecture that conducts routing using the content's ID instead of an IP address. Content queries are routed on the content name itself instead of a destination address in these content oriented networks. When the content from a mobile device moves somewhere else, all the routing tables are generally re-created with the movement information that the mobile device sends. However, a routing scheme that uses ant colony optimization has attracted attention for supporting this process, but this optimization has a problem in that it cannot cope with moving contents and users sufficiently. In this paper, we propose a scheme that can cope with moving contents sources and users that require contents by using pheromones that are laid by these moving mobile devices. This proposed scheme can be applied to case of not only moving content sources but also the moving request users. Moreover, we conduct simulations to evaluate the performance of the proposed scheme.

Searching mechanisms employed in unstructured overlay networks typically hit multiple peers for the desired content. We propose the use of a simple method that raises the hit rates of unpopular contents and balances the loads by choosing the peer holding the least contents as the provider when multiple candidates exist.

Distributed web caching reduces retrieval latency of World Wide Web (WWW) objects such as text and graphics. Conventional distributed web caching methods, however, require many query messages among cache servers, which limits their scalability and reliability. To overcome these problems, we propose a query caching method in which each cache server caches not only WWW objects but also a query history. This method of finding cached objects can reduce the number of query messages among cache servers, making it possible to construct a large-scale distributed web cache server. We also propose an algorithm for constructing efficient query relationships among cache servers.

We develop a method of dimensioning and managing the bandwidth of a link on which TCP flows from access links are aggregated. To do this, we extend the application of the processor-sharing queue model to TCP performance evaluation by using flow statistics. To handle various factors that affect actual TCP behavior, such as round-trip time, window-size, and restrictions other than access-link bandwidth, we extend the model by replacing the access-link bandwidth with the actual file-transfer speed of a flow when the aggregation link is not congested. We only use the number of active flows and the link utilization to estimate the file-transfer speed. Unlike previous studies, the extended model based on the actual transfer speed does not require any assumptions/predeterminations about file-size, packet-size, and round-trip times, etc. Using the extended model, we predict the TCP performance when the link utilization increases. We also show a method of dimensioning the bandwidth needed to maintain TCP performance. We show the effectiveness of our method through simulation analysis.

In this paper, we propose a method of detecting TCP performance degradation using only bottleneck-link utilization statistics: mean and variance. The variance of link utilization normally increases as the mean link-utilization increases. However, because link-utilization has a maximum of 100%, as the mean approaches 100%, the possible range of fluctuation becomes narrow and the variance decreases to zero. In this paper, using the M/G/R processor sharing model, we relate this phenomenon to the behavior of flows. We also show that by using this relationship, we can detect TCP performance degradation using the mean and variance of link utilization. In particular, this method enables a network operator to determine whether or not the degradation originates from the congestion of his/her own network. Because our method requires us to measure only link utilization, the cost of performance management can be greatly decreased compared with the conventional method, which requires dedicated functions for directly measuring the TCP performance.