Number Of TCP Connections Research Articles

Design and analysis of a stable AQM controller for network congestion control

In this work, a stable active queue management (AQM) controller named SAQM is proposed to control the congestion in the network and improve the stability of the queue length at internet router. It is an integrated rate and queue-based AQM technique which is stable and robust under dynamic environment where a number of TCP connections, bottleneck bandwidth, round trip time (RTT), and target queue length keeps changing. The proposed AQM controller is implemented in NS2 network simulator. In this paper, the control theory is used for stability analysis of TCP/SAQM system and finally validated through MATLAB Simulink model. The simulation results show that the SAQM outperforms other existing AQM controllers in terms of achieving queue stability, less oscillatory with faster response and at the same time it responds quickly to traffic change.

Design and analysis of a stable AQM controller for network congestion control

In this work, a stable active queue management (AQM) controller named SAQM is proposed to control the congestion in the network and improve the stability of the queue length at internet router. It is an integrated rate and queue-based AQM technique which is stable and robust under dynamic environment where a number of TCP connections, bottleneck bandwidth, round trip time (RTT), and target queue length keeps changing. The proposed AQM controller is implemented in NS2 network simulator. In this paper, the control theory is used for stability analysis of TCP/SAQM system and finally validated through MATLAB Simulink model. The simulation results show that the SAQM outperforms other existing AQM controllers in terms of achieving queue stability, less oscillatory with faster response and at the same time it responds quickly to traffic change.

Design of feedback controller for TCP/AQM networks

In this paper, we propose a novel proportional-differential-type feedback controller called Novel-PD as new active queue management (AQM) to regulate the queue length with small oscillation. It measures the current queue length and uses the current queue length and differential error signals to adjust packet drop probability dynamically. We provide control theoretic analysis of system stability and develop guidelines to select control gain parameters of Novel-PD. The design of Novel-PD for TCP/AQM system is given in details. NS2 is used for conducting extensive simulation. The proposed controller is compared with random early detection (RED), random exponential marking (REM), proportional integrator (PI) and proportional derivative (PD) controller. Result shows that, Novel-PD is stable and achieves faster response in dynamic environments where number of TCP connections, bottleneck capacity, round trip time (RTT) keeps changing. The proposed controller outperforms other AQM schemes.

Study on Migration from IPv4 to IPv6 of a Large Scale Network

This work mainly addresses the design a large scale network using dual stack mechanisms. We concentrated on the most imperative theoretical notions of the IPv6 protocol, such as IP addressing, address allocation, routing with the OSPF and BGP protocols and routing protocols performance in dual stack network using GNS3 simulations and Wireshark Network protocol analyzer. It is evaluate a real large-scale network environment utilizing accessible end-to-end estimation methods that focuses on a large-scale IPv4 and IPv6 backbone and made performance the IPv4 and IPv6 network. In this paper, we compiled IPv6 address planning in a large scale network, performance metrics of each network in terms of time sequence graph, round trip time, TCP throughput, TCP connection time and the number of TCP connections per second that a client can establish with a remote server. It is found that, a minor degradation in the throughput of the TCP, TCP response time and a lower packet loss rate are arise in a real large scale dual stack network. We also showed a concise case study on relationship between RTT and network topology, which is cooperative to develop the performance of IPv6 networks. The result shows the proposed scheme for network migration from IPv4 to IPv6 is more reliable than other existing schemes.

Broadcasting on Large Scale Heterogeneous Platforms under the Bounded Multi-Port Model

We consider the classical problem of broadcasting a large message at an optimal rate in a large scale distributed network under the multi-port communication model. In this context, we are interested in both building an overlay network and providing an explicit algorithm for scheduling the communications. From an optimization point of view, we aim both at maximizing the throughput (ie the rate at which nodes receive the message) and minimizing the degree of the participating nodes, ie the number of TCP connections they must handle simultaneously. The main novelties of our approach are the introduction of this degree constraint and the classification of the set of participating nodes into two parts: open nodes that stay in the open-Internet and guarded'' nodes that lie behind firewalls or NATs. Two guarded nodes cannot communicate directly, but rather need to use an open node as a gateway for transmitting a message. In the case without guarded nodes, we prove that it is possible to reach the optimal throughput, at the price of a quasi-optimal (up to a small additive increase) degree of the participating nodes. In presence of guarded nodes, our main contributions are a closed form formula for the optimal cyclic throughput and the proof that the optimal solution may require arbitrarily large degrees. In the acyclic case, we propose an algorithm that reaches the optimal throughput with low degree. Then, we prove a worst case ratio between the optimal acyclic and cyclic throughput and show through simulations that this ratio is on average very close to 1, what makes acyclic solutions efficient both in terms of throughput maximization and degree minimization.

CloudProxy: A NAPT Proxy for Vulnerability Scanners based on Cloud Computing

Security-as-a-service (SaaS) is an outsourcing model for security management in cloud computing. Vulnerability scanners based on cloud computing is becoming one of the killer applications in SaaS due to the pay-per-use manner and powerful scanning capability. When performing vulnerability scanning through network, the scanner needs to establish a large number of TCP connections with the target host. To deal with the problem of IPv4 address shortening and to protect the hosts within the organization, the target hosts are almost always deployed behind a NAPT(Network Address and Port Translation) device, TCP packets sent by the scanner outside the network isolated by the NAPT device will be blocked, thus unable to complete the vulnerability scanning task when the scanners are deployed in the cloud. While there exists NAPT traversal methods, they support TCP poorly and therefore is not ready for the vulnerability scanning scenario where a large number of TCP connections needs to be established. In this paper we proposed a NAPT proxy named CloudProxy for adopting vulnerability scanners in cloud computing by combining the TURN extension protocol and the Socks5 protocol. We integrated function of Socks5 into the TURN client, so that the destination port of all scanning packets will be aggregated before passing through the TURN server, lessen the burden of the TURN server. The experimental results show that CloudProxy can relay packets for the vulnerability scanner based on cloud computing in a transparent way and its scalability is sufficient for practical use.

Analyzing Buffer Requirement for Wireless Enhancement of TCP in Network Mobility

On-board TCP (obTCP) has been shown to address the dual wireless link related issues in network mobility (NEMO) effectively. obTCP uses agents at both base station (BS) and mobile router (MR). The agents store incoming TCP packet in a buffer for future possible retransmissions over the wireless links. Since the number of TCP connections passing through the MR could be very large, the amount of memory space required for the buffers may become very high. This may put the deployment of obTCP in question. So, in this paper, the authors investigate the buffer requirement problem of obTCP at MR for each TCP connection. For this purpose, they describe a Markov model to keep track of the packet transmission process of MR. The buffer size for each TCP connection is represented as a function of loss probability of the wireless links. Interestingly, from the numerical results, the authors find that the buffer size requirement at MR is significantly low for each TCP connections. This observation claims possible implementation of obTCP in NEMO.

GridFTP-APT: Automatic Parallelism Tuning Mechanism for GridFTP in Long-Fat Networks

In this paper, we propose an extension to GridFTP that optimizes its performance by dynamically adjusting the number of parallel TCP connections. GridFTP has been used as a data transfer protocol to effectively transfer a large volume of data in Grid computing. GridFTP supports a feature called parallel data transfer that improves throughput by establishing multiple TCP connections in parallel. However, for achieving high GridFTP throughput, the number of TCP connections should be optimized based on the network status. In this paper, we propose an automatic parallelism tuning mechanism called GridFTP-APT (GridFTP with Automatic Parallelism Tuning) that adjusts the number of parallel TCP connections according to information available to the Grid middleware. Through simulations, we demonstrate that GridFTP-APT significantly improves the performance of GridFTP in various network environments.

Energy Efficiency and Capacity for TCP Traffic in Multi-Hop Wireless Networks

We study the performance metrics associated with TCP-regulated traffic in multi-hop, wireless networks that use a common physical channel (e.g., IEEE 802.11). In contrast to earlier analyses, we focus simultaneously on two key operating metrics--the energy efficiency and the transport-layer (TCP) throughput. Using analysis and simulations, we show how these metrics are strongly influenced by the radio transmission range of individual nodes. Due to tradeoffs between the individual packet transmission energy and the likelihood of retransmissions, the total energy consumption is a convex function of the number of hops (and hence, of the transmission range). On the other hand, the throughput of a single TCP session decreases with a decrease in the transmission range. The overall achievable TCP throughput in an ad-hoc network thus involves a tradeoff between the reduced throughput of an individual flow and the greater degree of spatial reuse possible. As a consequence of this tradeoff, the overall network capacity turns out to be a concave function of the transmission range. We analyze how parameters such as the node density and the radio transmission range affect the overall network capacity under different operating conditions. Our analysis shows that capacity metrics at the TCP layer behave quite differently from the capacity results previously presented in literature. We then extend the work and examine the sensitivity of the TCP-layer capacity to the speed of the nodes and the number of TCP connections in an ad hoc network. By incorporating the notion of a minimal acceptable QoS metric (loss) for an individual session, we show why the QoS-compliant capacity is a more accurate metric for studying the capacity of TCP traffic in an ad hoc network. Finally, we study the dependence of capacity on the source application (Telnet or FTP traffic) and on the choice of the ad hoc routing protocol (AODV, DSR or DSDV).

High Data Rate Video Transmission Using Parallel TCP Connections: Approaches and Performance Evaluation

In our study, we investigate a packet-level protocol parallelization approach, which works by parallel multithreading the protocol execution such that packets within and among connections are processed in parallel using distinct processors/threads. The major advantage of this approach is its high scalability--with proper scheduling, more protocol connections, and hence more requests, can be supported by using more threads. In this paper, we present results of our detailed simulations using the NS-2 platform for reliably transferring video stream data in a client-server system. Different types of parameters are used to measure the performance. The parameters include available network bandwidth, different number of TCP connections, and different video sources. Our results show that a parallel approach can indeed significantly enhance the playback quality.

A TCP-Aware Link Layer Protocol Based on Selective-repeat ARQ with No Resequencing

A link layer protocol based on SR (Selective-Repeat) ARQ (Automatic Repeat reQuest) is required to achieve high performance over a lossy and large delay link. In spite of its effectiveness, SR ARQ has problems of large resequencing delay and bursty packet output due to the resequencing function. To mitigate these problems, this paper proposes a scheme where the resequencing is not performed by a receiver of SR ARQ, but by TCP end to end. The proposed scheme has advantages of scalability with regard to link bandwidth, the number of TCP connections on the link, and the size of the TCP window. It also preserves the end-to-end semantics of TCP and requires no modification to the existing TCP. To suppress unnecessary retransmissions by TCP, a sender of SR ARQ is aware of retransmitted TCP data packets and drops duplicate ACKs due to out-of-order packets, which are caused by the lack of resequencing by SR ARQ. The effectiveness of this proposed scheme is evaluated by simulations, which show that it attains high performance in comparison with other schemes.

Analysis of link-level hybrid FEC/ARQ-SR for wireless links and long-lived TCP traffic

Since the TCP protocol uses the loss of packets as an indication of network congestion, its performance degrades over wireless links, which are characterized by a high bit error rate. Different solutions have been proposed to improve the performance of TCP over wireless links, the most promising one being the use of a hybrid model at the link level combining Forward Error Correction (FEC), Automatic Repeat Request with Selective Repeat (ARQ-SR), and an in-order delivery of packets to IP. The drawback of FEC is that it consumes some extra bandwidth to transmit the redundant information. ARQ-SR consumes extra bandwidth only when packets are lost, its drawback is that it increases the round-trip time (RTT), which may deteriorate the performance of TCP. Another drawback of ARQ-SR is that a complete packet can be retransmitted to correct a small piece of errored data. We study in this paper the performance of TCP over a wireless link implementing hybrid FEC/ARQ-SQ. The study is done by simulating and modeling long-lived TCP transfers over wireless links showing Bernoulli errors. We are motivated by how to tune link-level error recovery, e.g. amount of FEC, persistency of ARQ, so as to maximize the performance of TCP. We provide results for different physical characteristics of the wireless link (delay, error rate) and for different traffic loads (number of TCP connections).

A variable weighted least-connection algorithm for multimedia transmission

Under high loads, a multimedia cluster server can serve many hundreds of connections concurrently, where a load balancer distributes the incoming connection request to each node according to a preset algorithm. Among existing scheduling algorithms, round-Robin and least-connection do not take into account the difference of service capability of each node and improved algorithms such as weighted round-Robin and weighted least-connection. They also do not consider the fact that the ratio of number of TCP connections and the fixed weight does not reflect die real load of node. In this paper we introduce our attempts in improving the scheduling algorithms and propose a variable weighted least-connection algorithm, which assigns variable weight, instead of fixed weight, to each node according to its real time resource. A validating trial has been performed and the results show that the proposed algorithm has effective load balancing in one central control node scenario.

Active queue management with flow proportional buffering

AbstractMost active queue management schemes maintain an average of the queue length which they use together with a number of queue thresholds to detect congestion. However, the setting of the queue thresholds is problematic because the required buffer size for good sharing among TCP connections is dependent on the number of TCP connections using the buffer. This paper describes an improved active queue management scheme which dynamically changes its threshold settings as the number of connections and system load changes. This technique allows network devices to effectively control packet losses and TCP timeouts while maintaining high link utilization. Copyright © 2003 John Wiley &Sons, Ltd.

A comparison of active queue management algorithms using the OPNET Modeler

A number of active queue management algorithms for TCP/IP networks such as random early detection (RED), stabilized RED (SRED), BLUE, and dynamic RED (DRED) have been proposed in the past few years. This article presents a comparative study of these algorithms using simulations. The evaluation is done using the OPNET Modeler, which provides a convenient and easy-to-use platform for simulating large-scale networks. The performance metrics used in the study are queue size, packet drop probability, and packet loss rate. The study shows that, among the four algorithms, SIZED and DRED are more effective at stabilizing the queue size and controlling the packet loss rate while maintaining high link utilization. The benefits of stabilized queues in a network are high resource utilization, bounded delays, more certain buffer provisioning, and,traffic-load-independent network performance in terms of traffic intensity and number of TCP connections.

A load adaptive mechanism for buffer management

The basic idea behind active queue management schemes such as random early detection (RED) is to detect incipient congestion early and to convey congestion notification to the end-systems, allowing them to reduce their transmission rates before queues in the network overflow and packets are dropped. The basic RED scheme (and its newer variants) maintains an average of the queue length which it uses together with a number of queue thresholds to detect congestion. RED schemes drop incoming packets in a random probabilistic manner where the probability is a function of recent buffer fill history. The objective is to provide a more equitable distribution of packet loss, avoid the synchronization of flows, and at the same time improve the utilization of the network. The setting of the queue thresholds in RED schemes is problematic because the required buffer size for good sharing among TCP connections is dependent on the number of TCP connections using the buffer. This paper describes a technique for enhancing the effectiveness of RED schemes by dynamically changing the threshold settings as the number of connections (and system load) changes. Using this technique, routers and switches can effectively control packet losses and TCP timeouts while maintaining high link utilization.

Improving end-to-end performance of the Web using server volumes and proxy filters

The rapid growth of the World Wide Web has caused serious performance degradation on the Internet. This paper offers an end-to-end approach to improving Web performance by collectively examining the Web components --- clients, proxies, servers, and the network. Our goal is to reduce user-perceived latency and the number of TCP connections, improve cache coherency and cache replacement, and enable prefetching of resources that are likely to be accessed in the near future. In our scheme, server response messages include piggybacked information customized to the requesting proxy. Our enhancement to the existing request-response protocol does not require per-proxy state at a server, and a very small amount of transient per-server state at the proxy, and can be implemented without changes to HTTP 1.1. The server groups related resources into volumes (based on access patterns and the file system's directory structure) and applies a proxy-generated filter (indicating the type of information of interest to the proxy) to tailor the piggyback information. We present efficient data structures for constructing server volumes and applying proxy filters, and a transparent way to perform volume maintenance and piggyback generation at a router along the path between the proxy and the server. We demonstrate the effectiveness of our end-to-end approach by evaluating various volume construction and filtering techniques across a collection of large client and server logs.