多个TCP连接的拥塞丢包模型

本文提出了有关TCP连接的拥塞丢包分析模型.网络瓶颈一般承载许多TCP连接,瓶颈处不可避免的拥塞和缓存溢出,是导致网上丢包的主要原因.网络瓶颈处的行为很大程度上左右了网络性能.本文的模型估计了存在大量持续TCP连接时,网络瓶颈的丢包概率和网络传输中断概率,给出了对实际网络的良好近似.这对于研究TCP对网络性能的影响,提出改善网络性能的新算法,以及分析(从长远来看)TCP还应做哪些改进,都是非常有用的.     

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.     

Optimal TCP segment size for mobile server access over wireless links of cellular networks

Internet access from mobile phones over cellular networks suffers from severe bandwidth limitations and high bit error rates over wireless access links. Tailoring TCP connections to best fit the characteristics of this bottleneck link is thus very important for overall performance improvement. In this work, we propose a simple algorithm in deciding the optimal TCP segment size to maximize the utilization of the bottleneck wireless TCP connection for mobile contents server access, taking the dynamic TCP window variation into account. The proposed algorithm can be used when the product of the access rate of the wireless link and the propagation time to mobile contents servers is not large. With some numerical examples, it is shown that the optimal TCP segment size becomes a constant value when the TCP window size (WS) exceeds a threshold. One can set the maximum segment size (MSS) of a wireless TCP connection to this optimal segment size for mobile contents server access for maximum efficiency on the expensive wireless link. Copyright © 2007 John Wiley & Sons, Ltd.     

Performance of TCP congestion control with explicit rate feedback

We consider a modification of TCP congestion control in which the congestion window is adapted to explicit bottleneck rate feedback; we call this RATCP (Rate Adaptive TCP). Our goal in this paper is to study and compare the performance of RATCP and TCP in various network scenarios with a view to understanding the possibilities and limits of providing better feedback to TCP than just implicit feedback via packet loss. To understand the dynamics of rate feedback and window control, we develop and analyze a model for a long-lived RATCP (and TCP) session that gets a time-varying rate on a bottleneck link. We also conduct experiments on a Linux based test-bed to study issues such as fairness, random losses, and randomly arriving short file transfers. We find that the analysis matches well with the results from the test-bed. For large file transfers, under low background load, ideal fair rate feedback improves the performance of TCP by 15%-20%. For small randomly arriving file transfers, though RATCP performs only slightly better than TCP it reduces losses and variability of throughputs across sessions. RATCP distinguishes between congestion and corruption losses, and ensures fairness for sessions with different round trip times sharing the bottleneck link. We believe that rate feedback mechanisms can be implemented using distributed flow control and recently proposed REM in which case, ECN bit itself can be used to provide the rate feedback.     

Active queue management with flow proportional buffering

Most 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 study on a receiver‐based management scheme of access link resources for QoS‐controllable TCP connections

Although the bandwidth of access networks is rapidly increasing with the latest techniques such as DSL and FTTH, the access link bandwidth remains a bottleneck, especially when users activate multiple network applications simultaneously. Furthermore, since the throughput of a standard TCP connection is dependent on various network parameters, including round‐trip time and packet loss ratio, the access link bandwidth is not shared among the network applications according to the user's demands. In this thesis, we present a new management scheme of access link resources for effective utilization of the access link bandwidth and control of the TCP connection's throughput. Our proposed scheme adjusts the total amount of the receive socket buffer assigned to TCP connections to avoid congestion at the access network, and assigns it to each TCP connection according to characteristics in consideration of QoS. The control objectives of our scheme are (1) to protect short‐lived TCP connections from the bandwidth occupation by long‐lived TCP connections, and (2) to differentiate the throughput of the long‐lived TCP connections according to the upper‐layer application's demands. One of the results obtained from the simulation experiments is that our proposed scheme can reduce the delay of short‐lived document transfer perceived by the receiver host by up to about 90%, while a high utilization of access link bandwidth is maintained. Copyright © 2006 John Wiley & Sons, Ltd.     

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.     

The performance of TCP/IP for networks with high bandwidth-delayproducts and random loss

This paper examines the performance of TCP/IP, the Internet data transport protocol, over wide-area networks (WANs) in which data traffic could coexist with real-time traffic such as voice and video. Specifically, we attempt to develop a basic understanding, using analysis and simulation, of the properties of TCP/IP in a regime where: (1) the bandwidth-delay product of the network is high compared to the buffering in the network and (2) packets may incur random loss (e.g., due to transient congestion caused by fluctuations in real-time traffic, or wireless links in the path of the connection). The following key results are obtained. First, random loss leads to significant throughput deterioration when the product of the loss probability and the square of the bandwidth-delay product is larger than one. Second, for multiple connections sharing a bottleneck link, TCP is grossly unfair toward connections with higher round-trip delays. This means that a simple first in first out (FIFO) queueing discipline might not suffice for data traffic in WANs. Finally, while the Reno version of TCP produces less bursty traffic than the original Tahoe version, it is less robust than the latter when successive losses are closely spaced. We conclude by indicating modifications that may be required both at the transport and network layers to provide good end-to-end performance over high-speed WANs     

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     

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.     

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.     

Implicit admission control

Internet protocols currently use packet-level mechanisms to detect and react to congestion. Although these controls are essential to ensure fair sharing of the available resource between multiple flows, in some cases they are insufficient to ensure overall network stability. We believe that it is also necessary to take account of higher level concepts, such as connections, flows, and sessions when controlling network congestion. This becomes of increasing importance as more real-time traffic is carried on the Internet, since this traffic is less elastic in nature than traditional Web traffic. We argue that, in order to achieve better utility of the network as a whole, higher level congestion controls are required. By way of example, we present a simple connection admission control (CAC) scheme which can significantly improve the overall performance. This paper discusses our motivation for the use of admission control in the Internet, focusing specifically on control for TCP flows. The technique is not TCP specific, and can be applied to any type of flow in a modern IP infrastructure. Simulation results are used to show that it can drastically improve the performance of TCP over bottleneck links. We go on to describe an implementation of our algorithm for a router running the Linux 2.2.9 operating system. We show that by giving routers at bottlenecks the ability to intelligently deny admission to TCP connections, the goodput of existing connections can be significantly increased. Furthermore, the fairness of the resource allocation achieved by TCP is improved     

TCP/IP performance with random loss and bidirectional congestion

With the growth in Internet access services over networks with asymmetric links such as asymmetric digital subscriber line (ADSL) and cable-based access networks, it becomes crucial to evaluate the performance of TCP/IP over systems in which the bottleneck link speed on the reverse path is considerably slower than that on the forward path. In this paper, we provide guidelines for designing network control mechanisms for supporting TCP/IP. We determine the throughput as a function of buffering, round-trip times, and normalized asymmetry (defined as the ratio of the transmission time of acknowledgment (ACK) in the reverse path to that of data packets in the forward path). We identify three modes of operation which are dependent on the forward buffer size and the normalized asymmetry, and determine the conditions under which the forward link is fully utilized. We also show that drop-from-front discarding of ACKs on the reverse link provides performance advantages over other drop mechanisms in use. Asymmetry increases the TCP already high sensitivity to random packet losses that occur on a time scale faster than the connection round-trip time. We generalize the by-now well-known relation relating the square root of the random loss probability to obtained TCP throughput, originally derived considering only data path congestion. Specifically, random loss leads to significant throughput deterioration when the product of the loss probability, the normalized asymmetry and the square of the bandwidth delay product is large. Congestion in the reverse path adds considerably to TCP unfairness when multiple connections share the reverse bottleneck link. We show how such problems can be alleviated by per-connection buffer and bandwidth allocation on the reverse path     

TCP Westwood: End-to-End Congestion Control for Wired/Wireless Networks

TCP Westwood (TCPW) is a sender-side modification of the TCP congestion window algorithm that improves upon the performance of TCP Reno in wired as well as wireless networks. The improvement is most significant in wireless networks with lossy links. In fact, TCPW performance is not very sensitive to random errors, while TCP Reno is equally sensitive to random loss and congestion loss and cannot discriminate between them. Hence, the tendency of TCP Reno to overreact to errors. An important distinguishing feature of TCP Westwood with respect to previous wireless TCP extensions is that it does not require inspection and/or interception of TCP packets at intermediate (proxy) nodes. Rather, TCPW fully complies with the end-to-end TCP design principle. The key innovative idea is to continuously measure at the TCP sender side the bandwidth used by the connection via monitoring the rate of returning ACKs. The estimate is then used to compute congestion window and slow start threshold after a congestion episode, that is, after three duplicate acknowledgments or after a timeout. The rationale of this strategy is simple: in contrast with TCP Reno which blindly halves the congestion window after three duplicate ACKs, TCP Westwood attempts to select a slow start threshold and a congestion window which are consistent with the effective bandwidth used at the time congestion is experienced. We call this mechanism faster recovery. The proposed mechanism is particularly effective over wireless links where sporadic losses due to radio channel problems are often misinterpreted as a symptom of congestion by current TCP schemes and thus lead to an unnecessary window reduction. Experimental studies reveal improvements in throughput performance, as well as in fairness. In addition, friendliness with TCP Reno was observed in a set of experiments showing that TCP Reno connections are not starved by TCPW connections. Most importantly, TCPW is extremely effective in mixed wired and wireless networks where throughput improvements of up to 550% are observed. Finally, TCPW performs almost as well as localized link layer approaches such as the popular Snoop scheme, without incurring the overhead of a specialized link layer protocol.     

Fair and efficient Transmission Control Protocol access in the IEEE 802.11 infrastructure basic service set

When the stations in an IEEE 802.11 infrastructure basic service set employ Transmission Control Protocol (TCP), this exacerbates per‐flow unfair access problem. We propose a novel analytical model to approximately calculate the maximum per‐flow TCP congestion window limit that prevents packet losses at the access point buffer and therefore provides fair TCP access both in the downlink and uplink. The proposed analysis is unique in considering the effects of varying number of uplink and downlink TCP flows, differing round trip times among TCP connections and the use of delayed TCP acknowledgment (ACK) mechanism. Motivated by the findings of this theoretical analysis and simulations, we design a link layer access control block to be employed only at the access point in order to resolve the unfair access problem. The proposed link layer access control block uses congestion control and ACK filtering approach by prioritizing the access of TCP data packets of downlink flows over TCP ACK packets of uplink flows. Via simulations, we show that the proposed algorithm can provide both short‐term and long‐term fair accesses while improving channel utilization and access delay. Copyright © 2013 John Wiley & Sons, Ltd.     

Comparison of FEC types with regard to the efficiency of TCP connections over AWGN satellite channels

Optimizing the end-to-end throughput of a TCP connection (goodput) over geostationary satellite links is a challenging research topic. This is because the high delay-bandwidth product, together with a non-negligible random loss of packets, is a condition that considerably differs from the environments TCP was originally designed for. As a result, TCP performance is significantly impaired by the channel bit error rate. The literature is full of suggestions for improving TCP goodput, most based on modifications of the protocol itself. We only investigated the application of different FEC (forward error correction) types for TCP goodput optimization, leaving the end-to-end protocol unaltered. Using a method midway between analysis and simulation to evaluate the goodput of TCP long-lived connections, we first studied the influence of packet loss rate, introduced by errors on the channel, on the TCP goodput. We showed that, in some cases, the packet loss rate does not need to be negligible with respect to that caused by congestion, as it is widely-held opinion. We then applied physical-level FEC techniques, such as convolutional encoding/Viterbi decoding, Reed Solomon, link-level erasure codes and their combinations, over a wide field of signal to noise conditions of the satellite channel. For each FEC type, we found the FEC rate that maximizes the TCP goodput, in each channel condition. We finally compared the results of all FECs used between them, and presented the case of multiple TCP connections sharing the same link as well     

Enhancing satellite broadcasting services using multiresolutionmodulations

In this paper we propose to take advantage of the energy link margin that can exist of satellite connections to enrich the DVB-S services with Web-like interactive services. The exploitation of such a margin is obtained by using multiresolution modulation techniques. The system architecture analysed is asymmetrical, composed of a satellite forward link and a narrowband terrestrial reverse link. ATM is adopted to support different QoS for different types of information delivered. The satellite propagation delay and the traffic and congestion control of ATM suggest to modify the slow start and the congestion avoidance of the TCP. Our approach is based on the combination of a fixed window flow control at the transport layer with the ATM traffic and congestion control. Our analysis shows that the system performance is satisfactory if some bounds of the TCP buffer size are respected and the spacing of the resource management cells is within a given range of values     

Two-way TCP traffic over rate controlled channels: effects andanalysis

We study the performance of bidirectional TCP/IP connections over a network that uses rate-based flow and congestion control. An example of such a network is an asynchronous transfer mode (ATM) network using the available bit rate (ABR) service. The sharing of a common buffer by TCP packets and acknowledgment (acks) has been known to result in an effect called ack compression, where acks of a connection arrive at the source bunched together, resulting in unfairness and degraded throughput. It has been the expectation that maintaining a smooth flow of data using rate-based flow control would mitigate, if not eliminate, the various forms of burstiness experienced with the TCP window flow control. However, we show that the problem of TCP ack compression appears even while operating over a rate-controlled channel. The degradation in throughput due to bidirectional traffic can be significant. For example, even in the simple case of symmetrical connections with adequate window sizes, the throughput of each connection is only 66.67% of that under one-way traffic. By analyzing the periodic bursty behavior of the source IP queue, we derive estimates for the maximum queue size and arrive at a simple predictor for the degraded throughput, for relatively general situations. We validate our analysis using simulation on an ATM network using the explicit rate option of the ABR service. The analysis predicts the behavior of the queue and the throughput degradation in simple configurations and in more general situations     

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.     

REFWA: An Efficient and Fair Congestion Control Scheme for LEO Satellite Networks

This paper examines some issues that affect the efficiency and fairness of the Transmission Control Protocol (TCP), the backbone of Internet protocol communication, in multi-hops satellite network systems. It proposes a scheme that allows satellite systems to automatically adapt to any change in the number of active TCP flows due to handover occurrence, the free buffer size, and the bandwidth–delay product of the network. The proposed scheme has two major design goals: increasing the system efficiency, and improving its fairness. The system efficiency is controlled by matching the aggregate traffic rate to the sum of the link capacity and total buffer size. On the other hand, the system min-max fairness is achieved by allocating bandwidth among individual flows in proportion with their RTTs. The proposed scheme is dubbed Recursive, Explicit, and Fair Window Adjustment (REFWA). Simulation results elucidate that the REFWA scheme substantially improves the system fairness, reduces the number of packet drops, and makes better utilization of the bottleneck link. The results demonstrate also that the proposed scheme works properly in more complicated environments where connections traverse multiple bottlenecks and the available bandwidth may change over data transmission time.