On the Internet, end hosts and network nodes interdependently work to smoothly transfer traffic. Observed traffic dynamics are the result of those interactions among those entities. To manage Internet traffic to provide satisfactory quality services, such dynamics need to be well understood to predict traffic patterns. In particular, some nodes have a function that sends back-pressure signals to backward nodes to reduce their sending rate and mitigate congestion. Transmission Control Protocol (TCP) congestion control in end-hosts also mitigates traffic deviation to eliminate temporary congestion by reducing the TCP sending rate. How these congestion controls mitigate congestion has been extensively investigated. However, these controls only throttle their sending rate but do not reduce traffic volume. Such congestion control fails if congestion is persistent, e.g., for hours, because unsent traffic demand will infinitely accumulate. However, on the actual Internet, even with persistent congestion, such accumulation does not seem to occur. During congestion, users and/or applications tend to reduce their traffic demand in reaction to quality of service (QoS) degradation to avoid negative service experience. We previously estimated that 2% packet loss results in 23% traffic reduction because of this upper-layer reaction [1]. We view this reduction as an upper-layer congestion-avoidance mechanism and construct a closed-loop model of this mechanism, which we call the Upper-Layer Closed-Loop (ULCL) model. We also show that by using ULCL, we can predict the degree of QoS degradation and traffic reduction as an equilibrium of the feedback loop. We applied our model to traffic and packet-loss ratio time series data gathered in an actual network and demonstrate that it effectively estimates actual traffic and packet-loss ratio.

In this paper, we investigate the problems of the established congestion solution and then introduce a self-adjustable rate control that supports quality of service assurances over multi-hop wireless mesh networks. This scheme eliminates two phases of the established congestion solution and works on the MAC layer for congestion control. Each node performs rate control by itself so network congestion is eliminated after it independently collects its vector parameters and network status parameters for rate control. It decides its transmission rate based on a predication model which uses a rate function including a congestion risk level and a passing function. We prove that our scheme works efficiently without any negative effects between the network layer and the data link layer. Simulation results show that the proposed scheme is more effective and has better performance than the existing method.

This paper presents an autonomous traffic engineering framework, named ATE, a highly efficient data dissemination mechanism for multipath data forwarding in Wireless Sensor Networks (WSNs). The proposed ATE has several salient features. First, ATE utilizes three coordinating schemes: an incipient congestion inference scheme, an accurate link quality estimation scheme and a dynamic traffic diversion scheme. It significantly minimizes packet drops due to congestion by dynamically and adaptively controlling the data traffic over congested nodes and/or poorer quality links, and by opportunistically exploiting under-utilized nodes for traffic diversion, while minimizing the estimation and measurement overhead. Second, ATE can provide with high application fidelity of the network even for increasing values of bit error rates and node failures. The proposed link quality estimation and congestion inference schemes are light weight and distributed, improving the energy efficiency of the network. Autonomous Traffic Engineering has been evaluated extensively via NS-2 simulations, and the results have shown that ATE provides a better performance with minimum overhead than those of existing approaches.

In addition to high bit error rates, large and sudden variations in delay often occur in wireless cellular networks. The delay can be several times the typical round-trip time, which can cause the spurious timeout. In this letter, we propose a new window control algorithm to improve TCP performance in wireless cellular networks with large delay variation and high bit error rates. Simulation results illustrate that our proposal improves the performance of TCP in terms of fairness and link utilization.

Congestion in WSN increases the energy dissipation rates of sensor nodes as well as the loss of packets and thereby hinders fair and reliable event detection. We find that one of the key reasons of congestion in WSN is allowing sensing nodes to transfer as many packets as possible. This is due to the use of CSMA/CA that gives opportunistic medium access control. In this paper, we propose an energy efficient congestion avoidance protocol that includes source count based hierarchical and load adaptive medium access control and weighted round robin packet forwarding. We also propose in-node fair packet scheduling to achieve fair event detection. The results of simulation show our scheme exhibits more than 90% delivery ratio even under bursty traffic condition which is good enough for reliable event perception.

A critical design issue of Transmission Control Protocol (TCP) is its congestion control that allows the protocol to adjust the end-to-end communication rate based on the detection of packet loss. However, TCP congestion control may function poorly during its slow start and congestion avoidance phases. This is because TCP sends bursts of packets with the fast window increase and the ACK-clock based transmission in slow start, and respond slowly with large congestion windows especially in high bandwidth-delay product (BDP) networks during congestion avoidance. In this article, we propose an improved version of TCP, TCP-Ho, that uses an efficient congestion window control algorithm for a TCP source. According to the estimated available bandwidth and measured round-trip times (RTTs), the proposed algorithm adjusts the congestion window size with a rate between exponential growth and linear growth intelligently. Our extensive simulation results show that TCP-Ho significantly improves the performance of connections as well as remaining fair and stable when the BDP increases. Furthermore, it is feasible to implement because only sending part needs to be modified.

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.

This paper shows that a network supplying variable bit rate services can be prevented from becoming congested if each terminal limits the capacity of its connection in terms of its rate of increase. Variable bit rate sources are adequately assessed with two new concepts: the bit rate increase per unit time (acceleration-rate=αbit/sec2) or the bit rate increase ratio (acceleration-ratio=exp (β) ). The dimension of the acceleration-ratio coefficient βis seconds-1. The upper limits α and β are regulated to guarantee the network's QoS. The proposed concepts allow the network state to be accurately estimated and avoid congestion. The proposed method can be applied to ATM networks, Frame Relay networks, Fast Reservation Protocol systems and so on.