In this paper, we propose a nonlinear control model to characterize the AQM algorithm-GREEN. Based on this model, we analyze its performance and prove that there exists a stable oscillation when in equilibrium. Furthermore, we also investigate the effects of the factors such as bandwidth, round trip time, and load level on the amplitude and frequency of the oscillation. Theoretical analysis and simulation results indicate that GREEN algorithm is insensitive to the network conditions when the link rate and the round trip time are relatively small and becomes more sensitive to the change of network conditions when the bandwidth delay product is relatively high. For GREEN the adaptability to a wide range of network conditions is based on the compromising of the efficiency.

In the Internet, flow analysis and network monitoring have been studied by various methods. Some methods try to make TCP (Transport Control Protocol) traces more readable by showing them graphically. Others such as MRTG, NetScope, and NetFlow read the traffic counters of the routers and record the data for traffic engineering. Even if all of the above methods are useful, they are made only to perform a single task. This paper describes an improved TCP Protocol Machine, a multipurpose tool that can be used for flow analysis, intrusion detection and link congestion monitoring. It is developed based on a finite state machine (automaton). The machine separates the flows into two main groups. If a flow can be mapped to a set of input symbols of the automaton, it is valid, otherwise it is invalid. It can be observed that intruders' attacks are easily detected by the use of the protocol machine. Also link congestion can be monitored, by measuring the percentage of valid flows to the total number of flows. We demonstrate the capability of this tool through measurement and working examples.

Explicit Congestion Notification (ECN) supports the binary congestion information of the network for adjusting the window size. However, this results in the oscillation of the window size and the queue length due to the insufficient congestion information. In this paper, we propose the window-based congestion control mechanism with the modified ECN mechanism. The proposed scheme is based on extracting the network status from the consecutive binary congestion information provided by ECN. From the explicit network information, we estimate the allowable window size to achieve better performance. Through the simulations, the effectiveness of the proposed algorithm is shown as compared with the ECN algorithm.

This paper proposes a stable rate allocation algorithm for ABR service in ATM networks. The main goals in designing this algorithm are to speed up the convergence according to the max-min fairness criterion and to maximize the network utilization while the switch queue length can be properly controlled. Importantly, the set goals should be achieved in a wide range of network conditions without the need for adjusting the algorithm parameters. The algorithm is targeted to work in various networking environments with additional criteria as extended from the work of E-FMMRA (Enhanced Fast Max-Min Rate Allocation) and ERICA+ (Explicit Rate Indication for Congestion Avoidance) . The additional design criteria include the ability to enhance a large number of ABR connections and staggered TCP connections as well as to perform an accurate traffic averaging. The algorithm is analytically proved to be convergent. Simulation results indicate that the proposed algorithm achieves the goals in all evaluated configurations. However, it has some limitations when working in the large-scale network due to its per-connection accounting. It is not recommended to implement the algorithm with a switch that has a small buffer size due to its relatively long averaging interval.

Call demand suddenly and greatly increases in the aftermath of a major disaster, because people want to check on their families and friends in the stricken area. Many call attempts in mobile cellular systems are blocked due to the limited radio frequency resources. In this paper, as a solution to this problem, limiting the holding time of calls is investigated and a dynamic holding time limit (DHTL) method, which varies the holding time limit dynamically based on the number of call attempts, is proposed. The effect of limiting the holding time is investigated first using a computer simulation with a constant and heavy traffic load model. This simulation shows that the average holding time of calls is decreased as the holding time limit is reduced. But it also shows limiting the holding time decreases the number of calls blocked and forced call terminations at handover considerably. Next, a simple estimation method for the holding time limit, which reduces the blocking rate to the normal rate for increasing call demand, is described. Finally, results are given of a simulation, which show that the DHTL method keeps good performance for a sudden and great traffic load fluctuation condition.

We developed a distributed control algorithm to solve the problem of a trade-off between transient response and stability. We applied it to a congestion control algorithm for transmitting best-effort packets such as transmission control protocol (TCP) packets over the Internet. A new transmission power control algorithm suitable for transmitting best-effort packets over the wireless Internet was also developed using the distributed control algorithm. We showed that in a steady state, TCP connections can use the bandwidth efficiently over both wired and wireless Internet when the proposed control algorithms are used. The transient response was also evaluated and it was found that the packet transmission rate and the transmission power adjusted by the proposed control algorithms converge to a steady state faster than when adjusted by conventional control algorithms while maintaining the stability of network systems.

We derived the design requirements that wireless systems and congestion control algorithms must satisfy to transmit best-effort Internet protocol (IP) packets over wireless systems. We proved that, if these requirements are satisfied, congestion control algorithms are robust against unfairness in the systems and can provide near-maximum throughputs in various environments. From the viewpoint of the design requirements, we investigated the effect of automatic repeat request (ARQ) on the throughputs of best-effort IP connections, and showed why ARQ can improve the throughputs while too large a number of retransmissions degrade them. We also investigated the effect of variance in packet transmission rates and clarified what kind of congestion control algorithm degrades the throughputs.

In this paper, we propose a new consolidation algorithm called the Selective Backward Resource Management (BRM) cell Feedback (SBF) algorithm. It achieves a fast response and low consolidation noise by selectively forwarding BRM cell from the most congested branch to the source instead of waiting from all branches. Mathematical models are derived to quantitatively characterize the performance, i.e. the response time and ACR of the source, of SBF and previously proposed algorithms. The interoperation of consolidation algorithms in point-to-multipoint available bit rate (ABR) is investigated. We address response time, consolidation noise and the effect of asymmetrical round trip delay (RTD) from branch point to destinations aspects. All combinations of four different consolidation algorithms are interoperated in both local/metropolitan area network (LAN/MAN) and wide area network (WAN) configuration. By a simulation method, we found that the consolidation algorithm used at the uppermost stream branch point, especially in WAN configuration, plays an important role in determining the performance of the network. However, consolidation algorithm used at the lower stream branch point affects the network performance insignificantly. Hence, in order to achieve a good and effective performance of the consolidation algorithms interoperated network, a fast response with low consolidation noise algorithm should be used at the uppermost stream branch point and a simple and easy to implement algorithm should be used at the lower stream branch point.

It is a fundamental problem to construct a virtual path layout minimizing the hop number as a function of the congestion for a communication network. It is known that we can construct a virtual path layout with asymptotically optimal hop number for a mesh of trees network, butterfly network, cube-connected-cycles network, de Bruijn network, shuffle-exchange network, and complete binary tree network. The paper shows a virtual path layout with minimum hop number for a complete binary tree network. A generalization to complete k-ary tree networks is also mentioned.

This paper proposes a credit-based congestion control scheme for multicast communication which employs application-specific processing at intermediate network nodes. The control scheme was designed not only to take advantage of credit-based flow control for unicast communication, but also to achieve flexibility supported by active network technology. The resultant active multicast congestion control scheme is able to meet the different requirements of various multicast applications in terms of reliability and end-to-end latency. The performance of the proposed control scheme was evaluated using both discrete-event simulations and experiments on a prototype active network implementation. The results show that the proposed scheme performs very well in terms of fairness, responsiveness, and scalability. The implementation experiences also confirmed the feasibility of the scheme in practice.

This paper describes IP traffic, especially the control of VoIP traffic, on the carrier-scale, and proposes algorithms for it. It examines a case that has already been introduced in the United States and discusses the trend of standardization for this control. Control techniques that will be introduced into the IP network in the future are considered from the viewpoints of both "quality" that users receive and the "control" that carriers perform.

In the paper, we propose a congestion control scheme for reliable multicast communication which enables TCP fairness and prevents a drop-to-zero problem. The proposed congestion control scheme is rate-based one based on NAKs from receivers and cooperatively works with a flow control scheme. The congestion control scheme consists of two components of a rate-based controller and a selection mechanism of a representative. The rate-based controller runs between the sender and the representative and achieves TCP fairness and fast response to losses at the representative. The selection mechanism of the representative allows the sender to select the representative in a scalable manner, in which the sender makes use of NAKs from receivers to select it. In the paper, we also propose the switchover mechanism of the flow and congestion control schemes which enables the sender to use either of them adaptively based on network situations. When the network is congested, the congestion control scheme works to share network resources fairly with competing TCP flows. Otherwise, the flow control scheme works to adapt the transmission rate to the slowest receiver. We verify the performance of our proposed schemes by using computer simulation.

Layered multicast approach enables IP multicast to adapt to heterogeneous networks. In layered multicast, each layer of a session is sent to separate multicast groups. These layers will be transmitted on the same route, or on different routes. However, traditional congestion control schemes of layered multicast do not consider the case when layers of a session are transmitted on different routes. In this paper, at first we show that in sparse-mode routing protocols like PIM-SM and CBT, layers of a session can be mapped to different Rendezvous Points or cores due to the bootstrap mechanism. It means that layers of a session can be transmitted on different routes. We then show that traditional congestion control schemes of layered multicast do not work properly in sparse-mode routing regions. At last we introduce Rendezvous Point based Layered Multicast (RPLM), a novel congestion control scheme suitable for sparse-mode routing regions, and show that RPLM works efficiently in regions using sparse mode routing protocols. RPLM uses per-RP packet loss rate instead of the overall one to detect congestion on each route, and can react to congestion quickly by dropping the highest layer on the congested route. In addition, RPLM simultaneously drops all the layers those are useless in quality's improvement to prevent bandwidth waste.

In this paper, we survey the fairness issues in the congestion control mechanisms of TCP, which is one of most important service aspects in the current and future Internet. We first summarize the problems from a perspective of the fair service among connections. Several solution methods are next surveyed. Those are modifications of TCP congestion control mechanism and router support for achieving the fair service among TCP connections. We finally investigate the fair share of resources at endhosts.

This paper proposes two modes of the congestion control scheme to improve its behavior during the start-up period of networks in current TCP over ATM-UBR implementation. The proposed two modes are a single packet loss mode and a multiple packet losses mode. The proposed algorithm is to minimize the number of cell losses in the ATM switch during specially the start-up period. During the start-up period, multiple packet losses often happens because a TCP sender starts with default parameters. It often ends up sending too many packets and too fast, leading to multiple losses is packet burstiness which occurs right after fast recovery ends. We analyze the transition behavior during fast recovery algorithm and estimate the number of new packets sent when multiple packet losses detected. We present a simple simulation model and numerical results to investigate its performance of the proposed algorithms.

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 order to ease the impact of the packet fragmentation problem and to avoid network congestion in TCP over UBR, packet discard schemes in ATM layer (such as PPD and EPD) have been proposed. These schemes drop packets before they reach their intended destinations if the network is congested and the packets are to be partially discarded. On the other hand, TCP also regulates data flow with its own flow control method. Due to restriction of data flow at the TCP layer, buffer space is not fully used in an ATM switch. In order to make use of more buffer resources, this paper generalizes the PPD and EDP schemes. From this generalization, an optimistic packet discard scheme named the "Probability-Based Delayed Packet Discard" (PDPD) scheme is proposed. Depending on a particular probability, this scheme sets a discard flag to delay actual discard operation. This paper presents the results of several simulated models to find out the potential of improvement of goodput by PDPD. The results of these simulations indicate that PDPD obtains higher goodput than ordinary schemes when the packet size is large and the input load is not light. This author concludes that a PDPD scheme should achieve effective goodput and link utilization while using more buffer resources effectively.

In the future Internet, hierarchically classified Quality of Service (QOS) controls will be effective because various connections requiring different QOS are mixed. However, even in such an environment, among the same class connections, performance protection to harmful impact from the other connections and quality differentiation between connections will be required furthermore. In this paper, from this point of view, we focus on the active connections succession time (age of active connections) as a new dimensional criterion for buffer controls. To be concrete, the packet discarding control of congested router's buffer based on active connections is proposed. Moreover, its performance is evaluated through TCP/IP level simulation from the viewpoint of file transfer time. Conventional Internet can be regarded as the environment where only one class traffic exists (unit class environment). The proposed control scheme can provide powerful differentiation capability to avoid the performance disruption of total connections even in the conventional Internet.

In this article, we first discuss QoS metrics of the data networks, followed by raising the challenging problems for the next-generation Internet with high-performance and high-quality. We then discuss how the WDM technology can be incorporated for resolving those problems. Several research issues for the IP over WDM networks are also identified.

Flow control algorithm in high speed networks is a resource-sharing policy implemented in a distributed manner. This paper introduces a novel concept of backlog balancing and demonstrates its application to network flow control and congestion control by presenting a rate-based flow control algorithm for ATM networks. The aim of flow control is to maximize the network utilization for achieving high throughput with tolerable delay for each virtual circuit (VC). In a resource-sharing environment, this objective may also cause network congestion when a cluster of aggressive VC's are contending for the same resource at a particular node. The basic idea of our algorithm is to adjust the service rate of each node along a VC according to backlog discrepancies between neighboring nodes (i.e., to reduce the backlog discrepancy). The handshaking procedure between any two consecutive nodes is carried out by a link-by-link binary feedback protocol. Each node will update its service rate periodically based on a linear projection model of the flow dynamics. The updated service rate per VC at a node indicates its explicit demand of bandwidth, so a service policy implementing dynamic bandwidth allocation is introduced to enforce such demands. Simulation study has validated the concept and its significance in achieving the goal of flow control and yet preventing network congestion at the same time.