TCP-Jersey for wireless IP communications

Improving the performance of the transmission control protocol (TCP) in wireless Internet protocol (IP) communications has been an active research area. The performance degradation of TCP in wireless and wired-wireless hybrid networks is mainly due to its lack of the ability to differentiate the packet losses caused by network congestions from the losses caused by wireless link errors. In this paper, we propose a new TCP scheme, called TCP-Jersey, which is capable of distinguishing the wireless packet losses from the congestion packet losses, and reacting accordingly. TCP-Jersey consists of two key components, the available bandwidth estimation (ABE) algorithm and the congestion warning (CW) router configuration. ABE is a TCP sender side addition that continuously estimates the bandwidth available to the connection and guides the sender to adjust its transmission rate when the network becomes congested. CW is a configuration of network routers such that routers alert end stations by marking all packets when there is a sign of an incipient congestion. The marking of packets by the CW configured routers helps the sender of the TCP connection to effectively differentiate packet losses caused by network congestion from those caused by wireless link errors. This paper describes the design of TCP-Jersey, and presents results from experiments using the NS-2 network simulator. Results from simulations show that in a congestion free network with 1% of random wireless packet loss rate, TCP-Jersey achieves 17% and 85% improvements in goodput over TCP-Westwood and TCP-Reno, respectively; in a congested network where TCP flow competes with VoIP flows, with 1% of random wireless packet loss rate, TCP-Jersey achieves 9% and 76% improvements in goodput over TCP-Westwood and TCP-Reno, respectively. Our experiments of multiple TCP flows show that TCP-Jersey maintains the fair and friendly behavior with respect to other TCP flows.     

TCP Window Control for Variable Bandwidth in Wireless Cellular Networks

Most of TCP schemes in wireless networks assume that the bandwidth of the bottleneck link remains constant over time. However, in wireless cellular networks, to effectively manage the limited resources, the bandwidth is controlled based on radio condition over time. Such varying bandwidth can cause the networks congestion or underutilization. In this letter, we propose a new window control algorithm to improve TCP performance in wireless cellular networks with variable bandwidth. Simulation results illustrate that our proposal improves the performance of TCP in terms of fairness and link utilization     

An active queue management scheme based on a capture-recapture model

One of the challenges in the design of switches/routers is the efficient and fair use of the shared bottleneck bandwidth among different Internet flows. In particular, various active queue management (AQM) schemes have been developed to regulate transmission control protocol traffic in response to router congestion. In addition, in order to provide fair bandwidth sharing, these AQM must protect the well-behaved flows from the misbehaving flows. However, most of the existing AQM schemes cannot provide accurate fair bandwidth sharing while being scalable. The key to the scalability and fairness of the AQM schemes is the accurate estimation of certain network resources without keeping too much state information. We propose a novel technique to estimate two network resource parameters: the number of flows in the buffer and the data source rate of a flow by using a capture-recapture (CR) model. The CR model depends on simply the random capturing/recapturing of the incoming packets, and as a result, it provides a good approximation tool with low time/space complexity. These network resource parameters are then used to provide fair bandwidth sharing among the Internet flows. Our experiments and analysis will demonstrate that this new technique outperforms the existing mechanisms and closely approximates the "ideal" case, where full state information is needed.     

基于显式速率的TCP友好的UDP拥塞控制策略

本文提出了一种基于显式速率的UDP拥塞控制策略：通过源端和网络中的路由器相互配合，使得实时UDP应用能够根据网络的反馈以瓶颈链路的公平带宽为速率发送数据。此种控制策略对TCP应用是友好的，并且提高了网络的吞吐量和利用率。仿真结果表明：基于显式速率的UDP拥塞控制策略与采用TFRC(TCP—Friendly Rate Control)的UDP拥塞控制策略相比，在吞吐量、TCP友好性等方面性能有较大提高。     

TD-SCDMA网络TCP性能优化技术

3G时代为人们提供了更高的无线业务承载带宽,互联网的发展引入了更多的业务需求,如何最大限度的提高和保证端到端的服务性能,是全系统问题。在E2E解决方案中,仅依赖无线资源管理等传统算法的优化已解决不了所有的问题。而传统TCP协议并不适应无线网络的传输环境,其没有考虑无线空口环境剧烈变化,慢启动机制导致分组丢失时重传延迟大,影响用户吞吐率,严重影响用户感知。     

基于流媒体的实时网络传输系统中若干问题研究

流媒体在Internet上的实时传输常因其要求高带宽、低延迟而造成网络拥塞。探讨了基于Internet的实时流媒体中视频流的QoS控制策略,并主要论述了基于速率的网络拥塞控制方法并给出一种视频流编码速率调整算法。这些控制技术应用于终端系统并不需要路由器和网络QoS支持,可以较好地提高视频质量。     

TCP Veno: TCP enhancement for transmission over wireless access networks

Wireless access networks in the form of wireless local area networks, home networks, and cellular networks are becoming an integral part of the Internet. Unlike wired networks, random packet loss due to bit errors is not negligible in wireless networks, and this causes significant performance degradation of transmission control protocol (TCP). We propose and study a novel end-to-end congestion control mechanism called TCP Veno that is simple and effective for dealing with random packet loss. A key ingredient of Veno is that it monitors the network congestion level and uses that information to decide whether packet losses are likely to be due to congestion or random bit errors. Specifically: (1) it refines the multiplicative decrease algorithm of TCP Reno-the most widely deployed TCP version in practice-by adjusting the slow-start threshold according to the perceived network congestion level rather than a fixed drop factor and (2) it refines the linear increase algorithm so that the connection can stay longer in an operating region in which the network bandwidth is fully utilized. Based on extensive network testbed experiments and live Internet measurements, we show that Veno can achieve significant throughput improvements without adversely affecting other concurrent TCP connections, including other concurrent Reno connections. In typical wireless access networks with 1% random packet loss rate, throughput improvement of up to 80% can be demonstrated. A salient feature of Veno is that it modifies only the sender-side protocol of Reno without changing the receiver-side protocol stack.     

一种异构网络TCP拥塞控制算法

针对互联网中端对端带宽、时延和丢包率等的差异性日益加剧,导致TCP传输性能严重退化,该文提出一种链路自适应TCP拥塞控制算法(INVS)。INVS在拥塞避免阶段初期采用基于指数函数的凸窗口增长函数,以提高链路利用率;在窗口增长函数中引入了自适应增长因子实现窗口增长速率与链路状态相匹配;采用了自适应队列门限的丢包区分策略以提高无线环境下TCP的性能。性能分析和评估表明,INVS提高了TCP拥塞控制算法的吞吐量、公平性、链路利用率和RTT公平性。     

Congestion Detection in the European Internet

In this paper we present a study about the utilization of one-way delay measurements to detect and characterize network congestion in the european Internet. The experiments have been made using the ETOMIC platform that allows one-way delay measurement with high precision timestamps. We have found a peculiar router behaviour in which the bottleneck is not the available bandwidth but it is the packet processing power of the router (backplane and CPU constraints). This router has been characterized with several network parameters. Some of them are the dependency of this limitation with the input data rate in packets per second, the size of burst packet losses measured in packets or time and the absence of specific scheduling algorithms in the router that could affect to larger flows.     

CPCA: An efficient wireless routing algorithm in WiNoC for cross path congestion awareness

Wireless network-on-chip (WiNoC) is a new paradigm to mitigate the long-distance transmission latency for conventional wired network-on-chip. The wireless routers in WiNoC have to handle a large number of packets which could cause data congestion, thus reducing the network performance. In this paper, we propose a novel wireless routing algorithm, called CPCA, which exploits the cross path congestion information as hints to route the packets. Under CPCA, the whole network is partitioned into sub-networks. In each subnet, the congestion information of the wireless router is propagated along the cross path. As a result, the routers in the same dimension can get the congestion degree of wireless router within the subnet. Based on the congestion information, CPCA can compute the suitable path for packets routing, which can prominently avoid the congestion aggravation in the wireless router. Experimental results show that our proposed method can effectively improve performance in terms of packets transmission latency and network throughput.     

Improving Transport Layer Performance in Multihop Ad Hoc Networks by Exploiting MAC Layer Information

The traditional TCP congestion control mechanism encounters a number of new problems and suffers a poor performance when the IEEE 802.11 MAC protocol is used in multihop ad hoc networks. Many of the problems result from medium contention at the MAC layer. In this paper, we first illustrate that severe medium contention and congestion are intimately coupled, and TCP's congestion control algorithm becomes too coarse in its granularity, causing throughput instability and excessively long delay. Further, we illustrate TCP's severe unfairness problem due to the medium contention and the tradeoff between aggregate throughput and fairness. Then, based on the novel use of channel busyness ratio, a more accurate metric to characterize the network utilization and congestion status, we propose a new wireless congestion control protocol (WCCP) to efficiently and fairly support the transport service in multihop ad hoc networks. In this protocol, each forwarding node along a traffic flow exercises the inter-node and intra-node fair resource allocation and determines the MAC layer feedback accordingly. The end-to-end feedback, which is ultimately determined by the bottleneck node along the flow, is carried back to the source to control its sending rate. Extensive simulations show that WCCP significantly outperforms traditional TCP in terms of channel utilization, delay, and fairness, and eliminates the starvation problem     

一种基于链路传输延迟的MPTCP拥塞控制算法*

由于传统TCP拥塞控制算法直接应用到MPTCP(Multipath Transport Control Protocol)中存在公平性问题,以及不能有效地发挥多路径传输的优势,因而从公平性方面对MPTCP现有拥塞控制算法进行研究.研究发现,现有的MPTCP拥塞控制算法均受到相同的回路时间限制.提出一种基于链路延迟的RTT补偿算法(Compensating for RTT mismatch,C-RTT ).该算法通过设置网络带宽占用比参数以及对MPTCP连接的子流设置侵略因子,从而保证瓶颈链路处MPTCP数据流和TCP数据流公平地共享可用带宽.最后通过NS3仿真实验证明,该算法能够有效地补偿链路中因RTT不等引起的公平性问题,并避免链路之间数据的非周期抖动,且保证了多路径传输的优越性.     

无线Mesh网络中基于链路负载估算的拥塞控制策略

针对无线Mesh网络的网络特性,提出了一种基于链路负载估算的拥塞控制策略LLECC。LLECC算法计算有效链路带宽和链路负载估算确定RED算法中的调整因子,通过调整因子调整RED算法中的参数从而实现动态的对无线网络拥塞控制。详细讨论了LLECC算法的实现过程和相关参数的计算方法,通过仿真分析验证了该算法对无线Mesh网络性能的提高。     

Promoting the use of end-to-end congestion control in the Internet

This paper considers the potentially negative impacts of an increasing deployment of non-congestion-controlled best-effort traffic on the Internet. These negative impacts range from extreme unfairness against competing TCP traffic to the potential for congestion collapse. To promote the inclusion of end-to-end congestion control in the design of future protocols using best-effort traffic, we argue that router mechanisms are needed to identify and restrict the bandwidth of selected high-bandwidth best-effort flows in times of congestion. The paper discusses several general approaches for identifying those flows suitable for bandwidth regulation. These approaches are to identify a high-bandwidth flow in times of congestion as unresponsive, “not TCP-friendly”, or simply using disproportionate bandwidth. A flow that is not “TCP-friendly” is one whose long-term arrival rate exceeds that of any conformant TCP in the same circumstances. An unresponsive flow is one failing to reduce its offered load at a router in response to an increased packet drop rate, and a disproportionate-bandwidth flow is one that uses considerably more bandwidth than other flows in a time of congestion     

Congestion control for sudden bandwidth changes in TCP

In this paper, we propose a novel technique to deal with sudden bandwidth changes in transmission control protocol (TCP). In the current Internet, sudden bandwidth changes may occur because of vertical handovers between heterogeneous access networks, routing path changes, cognitive ratio, and multi‐rate wireless local area network. The current implementation of TCP is designed and optimized for stable networks and does not adapt well upon sudden bandwidth changes. Consequently, it might suffer from packet losses in burst upon sudden bandwidth decrement and under‐utilization upon sudden bandwidth increment. To resolve this problem, we propose to modify the current TCP algorithm to include a new phase, called fast adaptation (FA). The FA phase is triggered upon detecting sudden bandwidth changes, and a TCP sender in the FA phase attempts to recover lost packets quickly to avoid spurious timeouts upon sudden bandwidth decrement. Upon sudden bandwidth increment, it increases its window size drastically to realize full utilization. Through extensive simulations, experiments, and analysis, it is shown that the proposed scheme can effectively deal with sudden bandwidth changes. Copyright © 2011 John Wiley & Sons, Ltd.     

无线多媒体网络中自适应拥塞控制算法的研究

提出了一种适用于采用无线接入多媒体流的拥塞控制算法,简称adaptive-MQWB(adaptivemediaQoSandwirelessbandwidth)。该算法依据当前带宽的动态变化率,在满足多媒体传输的最大时延的前提下,以最佳目标队长为实现目标,寻求最优的主动队列管理方案。仿真结果表明,与目前已有的MADR、tuned-RED算法相比,adaptive-MQWB算法在带宽动态变化的自适应性和多媒体传输时延的QoS保证方面都表现出更好的性能优势。     

WiNoC中无线节点和无线链路级拥塞避免的高效路由器设计

无线片上网络中,无线网络拥塞可以分为无线节点级的拥塞和无线链路级的拥塞,这两种拥塞都会造成网络性能下降.针对无线节点和无线链路级拥塞,本文设计了一种拥塞避免的高效无线路由器,首先提出了节点级的拥塞避免机制,无线节点通过广播本地拥塞信息相互感知拥塞程度,避免向拥塞程度较高的无线节点发送数据包;其次提出了链路级拥塞避免机制,在无线接口中设计了并行FIFO,允许无线接口以流水的方式,在单个时钟周期内传输一个拥塞信息数据包和三个数据微片,数学建模证明使用并行FIFO至少降低50%无线信道竞争频率,从而避免了无线链路级拥塞,提高了无线资源利用率.实验表明本文方案相较普通无线路由器增加了少量的面积,但是在网络整体性能、无线路由器性能以及功耗方面都取得了不错的优势.     

A Two-Time-Scale Design for Edge-Based Detection and Rectification of Uncooperative Flows

Existing Internet protocols rely on cooperative behavior of end users. We present a control-theoretic algorithm to counteract uncooperative users which change their congestion control schemes to gain larger bandwidth. This algorithm rectifies uncooperative users; that is, forces them to comply with their fair share, by adjusting the prices fed back to them. It is to be implemented at the edge of the network (e.g., by ISPs), and can be used with any congestion notification policy deployed by the network. Our design achieves a separation of time-scales between the network congestion feedback loop and the price-adjustment loop, thus recovering the fair allocation of bandwidth upon a fast transient phase     

A Prioritization Based Congestion Control Protocol for Healthcare Monitoring Application in Wireless Sensor Networks

Recent developments in biosensor and wireless technology have led to a rapid progress in wearable real time health monitoring. Unlike wired networks, wireless networks are subject to more packet loss and congestion. In this paper, we propose a congestion control and service prioritization protocol for real time monitoring of patients’ vital signs using wireless biomedical sensor networks. The proposed system is able to discriminate between physiological signals and assign them different priorities. Thus, it would be possible to provide a better quality of service for transmitting highly important vital signs. Congestion control is performed by considering both the congestion situation in the parent node and the priority of the child nodes in assigning network bandwidth to signals from different patients. Given the dynamic nature of patients’ health conditions, the proposed system can detect an anomaly in the received vital signs from a patient and hence assign more priority to patients in need. Simulation results confirm the superior performance of the proposed protocol. To our knowledge, this is the first attempt at a special-purpose congestion control protocol specifically designed for wireless biosensor networks.     

Revisiting the fair queuing paradigm for end-to-end congestion control

Today, the dominant paradigm for congestion control in the Internet is based on the notion of TCP friendliness. To be TCP-friendly, a source must behave in such a way as to achieve a bandwidth that is similar to the bandwidth obtained by a TCP flow that would observe the same round-trip time (RTT) and the same loss rate. However, with the success of the Internet comes the deployment of an increasing number of applications that do not use TCP as a transport protocol. These applications can often improve their own performance by not being TCP-friendly, which severely penalizes TCP flows. To design new applications to be TCP-friendly is often a difficult task. The idea of the fair queuing (FQ) paradigm as a means to improve congestion control was first introduced by Keshav (1991). While Keshav made a fundamental step toward a new paradigm for the design of congestion control protocols, he did not formalize his results so that his findings could be extended for the design of new congestion control protocols. We make this step and formally define the FQ paradigm as a paradigm for the design of new end-to-end congestion control protocols. This paradigm relies on FQ scheduling with per-flow scheduling and longest queue drop buffer management in each router. We assume only selfish and noncollaborative end users. Our main contribution is the formal statement of the congestion control problem as a whole, which enables us to demonstrate the validity of the FQ paradigm. We also demonstrate that the FQ paradigm does not adversely impact the throughput of TCP flows and explain how to apply the FQ paradigm for the design of new congestion control protocols. As a pragmatic validation of the FQ paradigm, we discuss a new multicast congestion control protocol called packet pair receiver-driven layered multicast (PLM).