This paper compares the performances of optimal routing as yielded by the pipe, hose, and intermediate models. The pipe model, which is specified by the exact traffic matrix, provides the best routing performance, but the traffic matrix is difficult to measure and predict accurately. On the other hand, the hose model is specified by just the total outgoing/incoming traffic from/to each node, but it has a problem in that its routing performance is degraded compared to the pipe model, due to insufficient traffic information. The intermediate model, where the upper and lower bounds of traffic demands for source-destination pairs are added as constraints, is a construction that lies between the pipe and hose models. The intermediate model, which lightens the difficulty of the pipe model, but narrows the range of traffic conditions specified by the hose model, offers better routing performance than the hose model. An optimal-routing formulation extended from the pipe model to the intermediate model can not be solved as a regular linear programming (LP) problem. Our solution, the introduction of a duality theorem, turns our problem into an LP formulation that can be easily solved. Numerical results show that the network congestion ratio for the pipe model is much lower than that of hose model. The differences in network congestion ratios between the pipe and hose models lie in the range from 27% to 45% for the various network topologies examined. The intermediate model offers better routing performance than the hose model. The intermediate model reduces the network congestion ratio by 34% compared to the hose model in an experimental network, when the upper-bound and lower-bound margins are set to 25% and 20%, respectively.

With the emergence of bandwidth-greedy application services, high-speed transport protocols are expected to effectively and aggressively use large amounts of bandwidth in current broadband and multimedia networks. However, when high-speed transport protocols compete with other standard TCP flows, they can occupy most of the available bandwidth leading to disruption of service. To deploy high-speed transport protocols on the Internet, such unfair situations must be improved. In this paper, therefore, we propose a method to improve fairness, called Kyushu-TCP (KTCP), which introduces a non-aggressive period in the congestion avoidance phase to give other standard TCP flows more chances of increasing their transmission rates. This method improves fairness in terms of the throughput by estimating the stably available bandwidth-delay product and adjusting its transmission rate based on this estimation. We show the effectiveness of the proposed method through simulations.

We present a congestion control algorithm for the Internet and assess its stability. The algorithm has low operation complexity and exercises control over sources without keeping per-flow information. Given the lack of support for explicit-rate feedback in the Internet, we discuss an implementation where feedback is based on explicit binary indications. We assess the stability through a discrete-time model and present simulation results showing the efficacy of the algorithm. The obtained results indicate that when the algorithm is used to control sources that support explicit binary feedback, its stability is not affected and its performance is close to that obtained with sources that support explicit-rate feedback.

This letter proposes a dynamic bandwidth allocation algorithm for access networks based PON (Passive Optical Network). It considers the mixture of transport layer protocols when responding to traffic congestion at the SNI (Service Node Interface). Simulations on a mixture of TCP (Transmission Control Protocol), and UDP (User Datagram Protocol) traffic flows show that the algorithm increases the throughput of TCP, improves the fairness between the two protocols, and solves the congestion problem at the SNI.

We consider a single-link multi-source network with FAST TCP sources. We adopt a continuous-time dynamic model for FAST TCP sources, and propose a static model to adequately describe the queuing delay dynamics at the link. The proposed model turns out to have a structure that reveals the time-varying network feedback delay, which allows us to analyze FAST TCP with due consideration of the time-varying network feedback delay. Based on the proposed model, we establish sufficient conditions for the boundedness of congestion window of each source and for the global asymptotic stability. The asymptotic stability condition shows that the stability property of each source is affected by all other sources sharing the link. Simulation results illustrate the validity of the sufficient condition for the global asymptotic stability.

In this paper, an effective congestion control algorithm is proposed to increase the end-to-end delivery success ratio of upstream traffic by reduction of buffer drop probabilities and their deviation in wireless sensor networks. According to the queue length of parent and child nodes, each child node chooses one of the parents as the next hop to the sink and controls the delay before transmission begins. It balances traffics among parents and mitigates congestion based on congestion level of a node. Simulation results show that the proposed algorithm reduces buffer drop probabilities and their deviation and increases the end-to-end delivery success ratio in wireless sensor networks.

Traditional TCP has a good congestion control strategy that adapts its sending rate in accordance with network congestion. In addition, a fast recovery algorithm can help TCP achieve better throughput by responding to temporary network congestion well. However, if multiple packet losses occur, the time to enter congestion avoidance phase would be delayed due to the long recovery time. Moreover, during the recovery phase, TCP freezes congestion window size until all lost packets are recovered, and this can make recovery time much longer resulting in performance degradation. To mitigate such recovery overhead, we propose Momentary recovery algorithm that recovers packet loss without an extra recovery phase. As other TCP and variants, our algorithm also halves the congestion window size when packet drop is detected, but it performs congestion avoidance phase immediately as if loss recovery is completed. For lost packets, TCP sender transmits them along with normal packets as long as congestion window permits rather than performs fast retransmission. In this manner, we can eliminate recovery overhead efficiently and reach steady state momentarily after network congestion. Finally, we provide a simulation based study on TCP recovery behaviors and confirm that our Momentary recovery algorithm always shows better performance compared with NewReno, SACK, and FACK.

In Low Earth Orbit (LEO) satellite networks, the user distributions are unbalanced due to the geography and the population dispersion. As a result, some satellites have few traffic loads, while others have heavy traffic loads which often lead to congestion events. In this paper, we propose a novel load balancing method based on congestion prediction. In the proposed method, each satellite detects areas where congestion often occurs and conveys their positions to its adjacent satellites. In those areas, the concerned satellites perform load balancing algorithms to prevent congestion. The performance of the proposed method is evaluated through a number of simulations. The simulation results demonstrate that the proposed scheme improves packet drop rate, end-to-end delay, and throughput.

This letter presents the optimal routing by the intermediate model; a construction that lies between the pipe and hose models. We show that it is a practical way of realizing optimal routing. A formulation extended from the pipe model to the intermediate model can not be solved as a regular linear programming (LP) problem. Our solution, the introduction of a duality theorem, successfully turns our problem into an LP formulation that can be easily solved. Numerical results show that the intermediate model has better routing performance than the hose model.

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.

For successful data collection in wireless sensor networks, it is important to ensure that the required delivery ratio is maintained while keeping a fair rate for every sensor. Furthermore, emerging high-rate applications might require complete reliability and the transfer of large volume of data, where persistent congestion might occur. These requirements demand a complete but efficient solution for data transport in sensor networks which reliably transports data from many sources to one or more sinks, avoids congestion and maintains fairness. In this paper, we propose congestion-aware and rate-controlled reliable transport (CRRT), an efficient and low-overhead data transport mechanism for sensor networks. CRRT uses efficient MAC retransmission to increase one-hop reliability and end-to-end retransmission for loss recovery. It also controls the total rate of the sources centrally, avoids the congestion in the bottleneck based on congestion notifications from intermediate nodes and centrally assigns the rate to the sources based on rate assignment policy of the applications. Performance of CRRT is evaluated in NS-2 and simulation results demonstrate the effectiveness of CRRT.

Recently, real-time media delivery services such as video streaming and VoIP have rapidly become popular. For these applications requiring high-level QoS guarantee, our research group has proposed a transport-layer approach to provide predictable throughput for upper-layer applications. In the present paper, we propose a congestion control mechanism of TCP for achieving predictable throughput. It does not mean we can guarantee the throughput, while we can provide the throughput required by an upper-layer application at high probability when network congestion level is not so high by using the inline network measurement technique for available bandwidth of the network path. We present the evaluation results for the proposed mechanism obtained in simulation and implementation experiments, and confirm that the proposed mechanism can assure a TCP throughput if the required bandwidth is not so high compared to the physical bandwidth, even when other ordinary TCP (e.g., TCP Reno) connections occupy the link.

Wireless technology has become widely popular and an important means of communication. A key issue in delivering wireless services is the problem of congestion which has an adverse impact on the Quality of Service (QoS), especially timeliness. Although a lot of work has been done in the context of RRM (Radio Resource Management), the deliverance of quality service to the end user still remains a challenge. Therefore there is need for a system that provides real-time services to the users through high assurance. We propose an intelligent agent-based approach to guarantee a predefined Service Level Agreement (SLA) with heterogeneous user requirements for appropriate bandwidth allocation in QoS sensitive cellular networks. The proposed system architecture exploits Case Based Reasoning (CBR) technique to handle RRM process of congestion management. The system accomplishes predefined SLA through the use of Retrieval and Adaptation Algorithm based on CBR case library. The proposed intelligent agent architecture gives autonomy to Radio Network Controller (RNC) or Base Station (BS) in accepting, rejecting or buffering a connection request to manage system bandwidth. Instead of simply blocking the connection request as congestion hits the system, different buffering durations are allocated to diverse classes of users based on their SLA. This increases the opportunity of connection establishment and reduces the call blocking rate extensively in changing environment. We carry out simulation of the proposed system that verifies efficient performance for congestion handling. The results also show built-in dynamism of our system to cater for variety of SLA requirements.

It is known that TCP data transfer in a wireless multihop network experiences a degradation in inter-connection fairness and throughput. This is because TCP is designed for use in wired networks, and the wireless multihop network has characteristics of sharing of the medium resources among nodes, which wired networks do not have. In particular, in wireless multihop networks where wireless nodes widely exist, hidden/exposed terminal problems are caused even if an RTS/CTS handshake is used. In this paper, two methods are proposed to improve fairness and throughput, without any feedback information from the intermediate nodes or cross-layer information. One method restricts the transfer period, while the other restrains the TCP congestion window. We evaluated these methods using computer simulations.

Vertical handoff is a new type of handoff that is triggered when a mobile node moves over heterogeneous wireless networks with each proving different access bandwidth, transmission latency, and coverage. A mobile node can achieve higher throughput by accessing a higher bandwidth providing wireless network. However, TCP has to experience drastic changes of the bandwidth and the latency due to the vertical handoff which must be recognized as a network congestion, and this degrades end-to-end performance. In this paper, we propose a TCP context switching scheme, named Context-Switching TCP, that maintains TCP variables separately for different types of wireless networks. Through simulations, Context-Switching TCP shows higher performance than TCP SACK for vertical handoff. Especially, it shows much higher performance gain when vertical handoff occurs frequently.

An important issue in designing a TCP congestion control algorithm is that it should allow the protocol to quickly adjust the end-to-end communication rate to the bandwidth on the bottleneck link. However, the TCP congestion control may function poorly in high bandwidth-delay product networks because of its slow response with large congestion windows. In this paper, we propose an enhanced version of TCP Vegas called Quick Vegas, in which we present an efficient congestion window control algorithm for a TCP source. Our algorithm improves the slow-start and congestion avoidance techniques of original Vegas. Simulation results show that Quick Vegas significantly improves the performance of connections as well as remaining fair when the bandwidth-delay product increases.

In this paper, we propose new methods which detect tampered-TCP connections at edge routers and protect well-behaved TCP connections from tampered-TCP connections, which results in fairness among TCP connections. The proposed methods monitor the TCP packets at an edge router and estimate the window size or the throughput for each TCP connection. By using estimation results, the proposed methods assess whether each TCP connection is tampered or not and drop packets intentionally if necessary to improve the fairness amongst TCP connections. From the results of simulation experiments, we confirm that the proposed methods can accurately identify tampered-TCP connections and regulate throughput ratio between tampered-TCP connections and competing TCP Reno connections to about 1.

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.

This proposes a scalable QoS control scheme, called Elephant Flow Control Scheme (EFCS) for high-speed large-capacity networks; it controls congestion and provides appropriate bandwidth to normal users' flows by controlling just the elephant flows. EFCS introduces a sampling packet threshold and drops packets considering flow size. EFCS also adopts a compensation parameter to control elephant flows to an appropriate level. Numerical results show that the sampling threshold increases control accuracy by 20% while reducing the amount of memory needed for packet sampling by 60% amount of memory by packet sampling; the elephant flows are controlled as intended by the compensation parameter. As a result, EFCS provides sufficient bandwidth to normal TCP flows in a scalable manner.

With the rapid development of wireless network technologies, heterogeneous networks with wired and wireless links are becoming common. However, the performance of TCP data transmission deteriorates significantly when a TCP connection traverses such networks, mainly because of packet losses caused by the high bit error rate of wireless links. Many solutions for this problem have been proposed in the past literature. However, most of them have various drawbacks, such as difficulties in their deployment by the wireless access network provider and end users, violation of TCP's end-to-end principle by splitting the TCP connection, or inapplicability to IP-level encrypted traffic because the base station needs to access the TCP header. In this paper, we propose a new mechanism without such drawbacks to improve the performance of TCP over wired and wireless heterogeneous networks. Our mechanism employs a receiver-based approach, which does not need modifications to be made to the sender TCP or the base station. It uses the ACK-splitting method for increasing the congestion window size quickly in order to restrain the throughput degradation caused by packet losses due to the high bit error rate of wireless links. We evaluate the performance of our mechanism and show that our mechanism can increase throughput by up to 94% in a UMTS network. The simulation results also show that our mechanism does not significantly deteriorate even when the receiver cannot perfectly distinguish whether packet losses are due to network congestion or bit errors on the wireless links.