Fairness in MIMD Congestion Control Algorithms

The Mulitplicative Increase Multiplicative Decrease (MIMD) congestion control algorithm in the form of Scalable TCP has been proposed for high speed networks. We study fairness among sessions sharing a common bottleneck link, where one or more sessions use the MIMD algorithm. Losses, or congestion signals, occur when the capacity is reached but could also be initiated before that. Both synchronous as well as asynchronous losses are considered. In the asynchronous case, only one session suffers a loss at a loss instant. Two models are then considered to determine which source looses a packet: a rate dependent model in which the packet loss probability of a session is proportional to its rate at the congestion instant, and the independent loss rate model. We first study how two MIMD sessions share the capacity in the presence of general combinations of synchronous and asynchronous losses. We show that, in the presence of rate dependent losses, the capacity is fairly shared whereas rate independent losses provide high unfairness. We then study inter protocol fairness: how the capacity is shared in the presence of synchronous losses among sessions some of which use Additive Increase Multiplicative Decrease (AIMD) protocols whereas the others use MIMD protocols.     

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     

基于明确丢失应答的异质网络传输控制

指出链路层策略在提高无线链路可靠性的同时,对传输层带来的影响,并采用明确丢失应答作为反馈信息,根据不同的数据丢失原因采取相应的措施,保证了对有线拥塞的及时检测和处理能力,使源端能在无线链路传输差错率高的情况下保持较高的传输速率,同时避免传输层与链路层之间不必要的重发竞争,从而提高了异质媒体网络上端到端的数据吞吐量。     

Enhancing Fairness and Throughput of TCP in Heterogeneous Wireless Access Networks

Most of the recent research on TCP over heterogeneous wireless networks has concentrated on differentiating between packet drops caused by congestion and link errors, to avoid significant throughput degradations due to the TCP sending window being frequently shut down, in response to packet losses caused not by congestion but by transmission errors over wireless links. However, TCP also exhibits inherent unfairness toward connections with long round-trip times or traversing multiple congested routers. This problem is aggravated by the difference of bit-error rates between wired and wireless links in heterogeneous wireless networks. In this paper, we apply the TCP Bandwidth Allocation (TBA) algorithm, which we have proposed previously, to improve TCP fairness over heterogeneous wireless networks with combined wireless and wireline links. To inform the sender when congestion occurs, we propose to apply Wireless Explicit Congestion Notification (WECN). By controlling the TCP window behavior with TBA and WECN, congestion control and error-loss recovery are effectively separated. Further enhancement is also incorporated to smooth traffic bursts. Simulation results show that not only can the combined TBA and WECN mechanism improve TCP fairness, but it can maintain good throughput performance in the presence of wireless losses as well. A salient feature of TBA is that its main functions are implemented in the access node, thus simplifying the sender-side implementation.     

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.     

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。     

多路径TCP拥塞控制算法研究

首先介绍了多路径传输协议的几种典型的拥塞控制算法,然后对MPTCP协议进行了理论分析,包括MPTCP拥塞控制算法在瓶颈链路TCP公平性、平衡拥塞能力以及flap现象,实验分析表明LINKED INCREASES算法效果最好。最后,指出其存在的问题,并指出了进一步的研究方向。     

卫星TCP/ATM传输中的明确窗口指示拥塞控制策略研究

卫星ATM是近年来卫星通信领域的一个研究热点,TCP业务在卫星ATM中的传输是其中一个重要的研究课题.由于在TCP协议中主要是通过对窗口的控制来实现拥塞控制,而卫星信道传输的长延时特性大大降低了TCP层拥塞控制的效率.本文提出一种基于UBR传输的明确窗口指示拥塞控制策略,仿真结果表明在效率、公平性、VBR背景传输下的性能等各方面,取得了比目前广泛使用的Reno TCP更好的结果.并且该算法实现简单,交换机实现零信元丢失的缓冲区很小并且与TCP源连接的数目无关.     

Rate-adaptive snoop: a TCP enhancement scheme over rate-controlled lossy links

We study TCP performance over the wireless links deploying a wireless rate-control technique, whose link characteristics are identified by variable link rate and bursty transmission error. We present a TCP enhancement scheme, called rate-adaptive snoop (RA-Snoop). RA-Snoop caches TCP packets selectively based on the wireless channel condition and the cached packets are retransmitted locally over the wireless link in case corruption loss is detected. In addition, for effective adaptation to variable bandwidth, RA-Snoop calculates the window feedback based on the bandwidth-delay product estimation and the queue level, then conveys this feedback information on the receiver's advertised window field in the acknowledgements returning to TCP sources. We compare the performance of RA-Snoop with that of existing schemes in the aspect of goodput and fairness. Results from simulations reveal that RA-Snoop achieves significant improvements over the existing schemes for various traffic scenarios.     

Multipath congestion control algorithm based on link capacity

Factors such as link differences and TCP friendliness constraints lead to the problem of unfair bandwidth allocation when the TCP congestion control mechanism was applied directly to multi-path transmission.To address the problem,it was proposed that a multipath congestion control algorithm was based on link capacity.The proposed algorithm which was based on the concept of feedback regulation achieved multipath joint congestion control by establishing the M/M/1 cache queue model to adjust the throughput rate of senders.Experimental results show the proposed algorithm can improve the multipath transmission bandwidth utilization and the multipath congestion control algorithm responsiveness,and ensure the fairness of multipath transmission.     

Basestation flow control for wired to wireless networks

With the exponential growth of the internet, wireless networks such as satellite networks are becoming increasingly popular. The characteristics of satellite networks such as long latency, large delay-bandwidth product, high bit error rate over satellite links and variable round trip time, severely degrade TCP/IP performance. At the conjunction of the satellite link and the fixed link, the basestation, the difference in capacity between the satellite link and the fixed link causes the basestation to experience congestion losses that adversely impact TCP performance. We propose a technique that substantially reduces the congestion at the base station and enforces fairness among the TCP connections that are sharing the satellite link. The technique does not require any change in the TCP sender or the receiver. The stability of our algorithm is analytically proven and its performance is evaluated using ns-2 simulations. Preliminary results yield almost a null congestion loss rate, a 60% decrease in average queue length, and more than 30% increase in the throughput. Fairness is well enforced.     

FLID-DL: congestion control for layered multicast

We describe fair layered increase/decrease with dynamic layering (FLID-DL): a new multirate congestion control algorithm for layered multicast sessions. FLID-DL generalizes the receiver-driven layered congestion control protocol (RLC) introduced by Vicisano et al. (Proc. IEEE INFOCOM, San Francisco, CA, , p.996-1003, Mar. 1998)ameliorating the problems associated with large Internet group management protocol (IGMP) leave latencies and abrupt rate increases. Like RLC, FLID-DL, is a scalable, receiver-driven congestion control mechanism in which receivers add layers at sender-initiated synchronization points and leave layers when they experience congestion. FLID-DL congestion control coexists with transmission control protocol (TCP) flows as well as other FLID-DL sessions and supports general rates on the different multicast layers. We demonstrate via simulations that our congestion control scheme exhibits better fairness properties and provides better throughput than previous methods. A key contribution that enables FLID-DL and may be useful elsewhere is dynamic layering (DL), which mitigates the negative impact of long IGMP leave latencies and eliminates the need for probe intervals present in RLC. We use DL to respond to congestion much faster than IGMP leave operations, which have proven to be a bottleneck in practice for prior work.     

The lightening effect of adaptive window control

We consider adaptive window (e.g., TCP) controlled transfer of http-like traffic (i.e., finite volume file transfers starting at random time instants) over a link, and develop an analysis for obtaining the stationary distribution of the link buffer occupancy. The significance of this work is that it provides a framework for analyzing the "closed-loop" behavior of the link buffer under adaptive window controlled transfer of files with a general file size distribution. We find that the tail of the stationary distribution of the buffer occupancy is lighter than may be expected from an open loop analysis. We also find that the tail of the buffer distribution is sensitive to the distribution of the file sizes, which provides insight into the sensitivity of TCP performance to file size distribution.     

解决FAST TCP缓存溢出相关问题的改进pacing technique算法和α参数调整算法

分析FAST TCP在缓存溢出发生时的性能,发现在缓存溢出场景中,收敛中的FAST TCP流经历严重的报文段丢失。相反,已经收敛了的FAST TCP流维持着高吞吐量和低报文段丢失概率。这种不公平是由FAST TCP缩减其窗口时的零传输率导致的。通过修改FAST TCP pacing算法,可以解决此问题。文中提出的α-adjusting算法,通过动态调整FAST TCP协议中的α参数来避免频繁的缓存溢出。通过分析ns2仿真结果,证明该算法在公平性和稳定性方面可获得令人满意的性能。     

Towards end-host-based identification of competing protocols against TCP in a bottleneck link

Classical Transmission Control Protocol (TCP) designs have never considered the identity of the competing transport protocol as useful information to TCP sources in congestion control mechanisms. When competing against a TCP flow on a bottleneck link, a User Datagram Protocol (UDP) flow can unfairly occupy the entire link bandwidth and suffocate all TCP flows on the link. If it were possible for a TCP source to know the type of transport protocol that deprives it of link access, perhaps it would be better for the TCP source to react in a way which prevents total starvation. In this paper, we use coefficient of variation and power spectral density of throughput traces to identify the presence of UDP transport protocols that compete against TCP flows on bottleneck links. Our results show clear traits that differentiate the presence of competing UDP flows from TCP flows independent of round-trip times variations. Signatures that we identified include an increase in coefficient of variation whenever a competing UDP flow joins the bottleneck link for the first time, noisy spectral density representation of a TCP flow when competing against a UDP flow in the bottleneck link, and a dominant frequency with outstanding power in the presence of TCP competition only. In addition, the results show that signatures for congestion caused by competing UDP flows are different from signatures due to congestion caused by competing TCP flows regardless of their round-trip times. The results in this paper present the first steps towards development of more ’intelligent’ congestion control algorithms with added capability of knowing the identity of aggressor protocols against TCP, and subsequently using this additional information for rate control.     

Steady state and transient state behaviours analyses of TCP connections considering interactions between TCP connections and network

The Internet uses a window‐based congestion control mechanism in transmission control protocol (TCP). In the literature, there have been a great number of analytical studies on TCP. Most of those studies have focused on the statistical behaviour of TCP by assuming a constant packet loss probability in the network. However, the packet loss probability, in reality, changes according to the packet transmission rates from TCP connections. Conversely, the window size of a TCP connection is dependent on the packet loss probability in the network. In this paper, we explicitly model the interaction between the congestion control mechanism of TCP and the network as a feedback system. By using this model, we analyse the steady state and the transient state behaviours of TCP. We derive the throughput and the packet loss probability of TCP, and the number of packets queued in the bottleneck router. We then analyse the transient state behaviour using a control theoretic approach, showing the influence of the number of TCP connections and the propagation delay on the transient state behaviour of TCP. Copyright © 2005 John Wiley & Sons, Ltd.     

Improvement of Transmission Control Protocol for High Bandwidth Applications

Transmission control protocol (TCP) is the widely and dominantly used protocol in today’s internet. A very recent implementation of congestion control algorithm is BBR by Google. Bottleneck bandwidth and round-trip time (BBR) is a congestion control algorithm which is created with the aim of increasing throughput and reducing delay. The congestion control protocols mentioned previously try to determine congestion limits by filling router queues. BBR drains the router queues at the bottleneck by sending exactly at the bottleneck link rate. This is done by the BBR through pacing rate which infers the delivery rate of the receiver and uses this as the estimated bottleneck bandwidth. But when the data rate is high, in the startup phase itself pipe becomes full and leads to some degradation in the Access Point of wireless environments by inducing losses specific to this environment. So the current pacing rate is not suitable for producing higher throughputs. Therefore, in the proposed system named R-BBR, this startup gain should be lower than the current startup gain which eventually would reduce pacing rate to reduce queue pressure in the sink node during the startup phase. The startup phase of BBR is modified to solve the problem of pipe full under high data rate. R-BBR has been evaluated over a wide range of wired as well as wireless networks by varying different factors like startup gain, congestion window, and pacing rate. It is inferred that R-BBR performs better than BBR with significant performance improvement.

     

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     

TCP-friendly flow control of wireless multimedia using ECN marking

In a wireless network packet losses can be caused not only by network congestion but also by unreliable error-prone wireless links. Therefore, flow control schemes which use packet loss as a congestion measure cannot be directly applicable to a wireless network because there is no way to distinguish congestion losses from wireless losses. In this paper, we extend the so-called TCP-friendly flow control scheme, which was originally developed for the flow control of multimedia flows in a wired IP network environment, to a wireless environment. The main idea behind our scheme is that by using explicit congestion notification (ECN) marking in conjunction with random early detection (RED) queue management scheme intelligently, it is possible that not only the degree of network congestion is notified to multimedia sources explicitly in the form of ECN-marked packet probability but also wireless losses are hidden from multimedia sources. We calculate TCP-friendly rate based on ECN-marked packet probability instead of packet loss probability, thereby effectively eliminating the effect of wireless losses in flow control and thus preventing throughput degradation of multimedia flows travelling through wireless links. In addition, we refine the well-known TCP throughput model which establishes TCP-friendliness of multimedia flows in a way that the refined model provides more accurate throughput estimate of a TCP flow particularly when the number of TCP flows sharing a bottleneck link increases. Through extensive simulations, we show that the proposed scheme indeed improves the quality of the delivered video significantly while maintaining TCP-friendliness in a wireless environment for the case of wireless MPEG-4 video.     

An explicit and fair window adjustment method to enhance TCP efficiency and fairness over multihops Satellite networks

Transmission control protocol (TCP) is the most widely used transport protocol in today's Internet. Despite the fact that several mechanisms have been presented in recent literature to improve TCP, there remain some vexing attributes that impair TCPs performance. This paper addresses the issue of the efficiency and fairness of TCP in multihops satellite constellations. It mainly focuses on the effect of the change in flows count on TCP behavior. In case of a handover occurrence, a TCP sender may be forced to be sharing a new set of satellites with other users resulting in a change of flows count. This paper argues that the TCP rate of each flow should be dynamically adjusted to the available bandwidth when the number of flows that are competing for a single link, changes over time. An explicit and fair scheme is developed. The scheme matches the aggregate window size of all active TCP flows to the network pipe. At the same time, it provides all the active connections with feedbacks proportional to their round-trip time values so that the system converges to optimal efficiency and fairness. Feedbacks are signaled to TCP sources through the receiver's advertised window field in the TCP header of acknowledgments. Senders should accordingly regulate their sending rates. The proposed scheme is referred to as explicit and fair window adjustment (XFWA). Extensive simulation results show that the XFWA scheme substantially improves the system fairness, reduces the number of packet drops, and makes better utilization of the bottleneck link.