A rate control scheme for adaptive real-time applications in IP networks with lossy links and long round trip times

Currently there is no control for real-time traffic sources in IP networks. This is a serious problem because real-time traffic can not only congest the network but can also cause unfairness and starvation of TCP traffic. However, it is not possible to apply current solutions for Internet to the networks with high bandwidth-delay products and high bit error rates. The channel errors may result in inaccurate congestion control decisions and unnecessary rate throttles leading to severe performance degradation. This problem is amplified in the links with high bandwidth-delay products, since the link is inefficiently utilized for a very long time until the unnecessary rate throttle is recovered. In this paper, a new Rate Control Scheme, RCS, is introduced for real-time interactive applications in networks with high bandwidth-delay products and high bit error rates. RCS is based on the concept of using dummy packets to probe the availability of network resources. Dummy packets are treated as low priority packets and consequently they do not affect the throughput of actual data traffic. Therefore, RCS requires all the routers in the connection path to support some priority policy. A new algorithm is also proposed to improve the robustness of the RCS to temporal signal loss conditions. The delay-bound considerations for real-time traffic sources using RCS rate control scheme are also investigated. Simulation experiments show that in environments with high bandwidth-delay products and high bit error rates, RCS achieves high throughput performance without penalizing TCP connections.     

Explicit window adaptation: a method to enhance TCP performance

We study the performance of TCP in an internetwork consisting of both rate-controlled and nonrate-controlled segments. A common example of such an environment occurs when the end systems are part of IP datagram networks interconnected by a rate-controlled segment, such as an ATM network using the available bit rate (ABR) service. In the absence of congestive losses in either segment, TCP keeps increasing its window to its maximum size. Mismatch between the TCP window and the bandwidth-delay product of the network results in accumulation of large queues and possibly buffer overflows in the devices at the edges of the rate-controlled segment, causing degraded throughput and unfairness. We develop an explicit feedback scheme, called explicit window adaptation, based on modifying the receiver's advertised window in TCP acknowledgments returning to the source. The window size indicated to TCP is a function of the free buffer in the edge device. Results from simulations with a wide range of traffic scenarios show that this explicit window adaptation scheme can control the buffer occupancy efficiently at the edge device, and results in significant improvements in packet loss rate, fairness, and throughput over a packet discard policy such as random early detection (RED)     

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     

Performance evaluation of high speed TCP over optical burst switching networks

Study of TCP performance over OBS networks has been an important problem of research lately and it was found that due to the congestion control mechanism of TCP and the inherent bursty losses in the Optical Burst Switching (OBS) network, the throughput of TCP connections degrade. On the other hand, High Speed TCP (HSTCP) was proposed as an alternative to the use of TCP in high bandwidth-delay product networks. HSTCP aggressively increases the congestion window used in TCP, when the available bandwidth is high and decreases the window cautiously in response to a congestion event. In this work, we make a thorough simulation study of HSTCP over OBS networks. While the earlier works in the literature used a linear chain of nodes as the network topology for the simulation, we use the popular 14-node NSFNET topology that represents an arbitrary mesh network in our study. We also study the performance of HSTCP over OBS for different bandwidths of access networks. We use two different cases for simulations where in the first HSTCP connections are routed on disjoint paths while in the second they contend for resources in the network links. These cases of simulations along with the mesh topology help us clearly distinguish between the congestion and contention losses in the OBS network and their effect on HSTCP throughput. For completeness of study, we also simulate TCP traffic over OBS networks in all these cases and compare its throughput with that of HSTCP. We observe that irrespective of the access network bandwidth and the burst loss rate in the network, HSTCP outperforms TCP in terms of the throughput and robustness against multiple burst losses up to the expected theoretical burst loss probability of 10⁻³.     

Modeling the performance of HTTP over several transport protocols

This paper considers the interaction of HTTP with several transport protocols, including TCP, Transaction TCP, a UDP-based request-response protocol, and HTTP with persistent TCP connections. We present an analytic model for each of these protocols and use that model to evaluate network overhead carrying HTTP traffic across a variety of network characteristics. This model includes an analysis of the transient effects of TCP slow-start. We validate this model by comparing it to network packet traces measured with two protocols (HTTP and persistent HTTP) over local and wide-area networks. We show that the model is accurate within 5% of measured performance for wide-area networks, but can underestimate latency when the bandwidth is high and delay is low. We use the model to compare the connection-setup costs of these protocols, bounding the possible performance improvement. We evaluate these costs for a range of network characteristics, finding that setup optimizations are relatively unimportant for current modem, ISDN, and LAN users but can provide moderate to substantial performance improvement over high-speed WANs. We also use the model to predict performance over future network characteristics     

A comparison of active queue management algorithms using the OPNETModeler

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     

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 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.     

Comparative study of various TCP versions over a wireless link with correlated losses

We investigate the behavior of the various transmission control protocol (TCP) algorithms over wireless links with correlated packet losses. For such a scenario, we show that the performance of NewReno is worse than the performance of Tahoe in many situations and even OldTahoe in a few situations because of the inefficient fast recovery method of NewReno. We also show that random loss leads to significant throughput deterioration when either the product of the square of the bandwidth-delay ratio and the loss probability when in the good state exceeds one, or the product of the bandwidth-delay ratio and the packet success probability when in the bad state is less than two. The performance of Sack is always seen to be the best and the most robust, thereby arguing for the implementation of TCP-Sack over the wireless channel. We also show that, under certain conditions, the performance depends not only on the bandwidth-delay product but also on the nature of timeout, coarse or fine. We have also investigated the effects of reducing the fast retransmit threshold.     

FRED-fair random early detection algorithm for TCP over ATMnetworks

The authors present the FRED (fair random early detection) algorithm as a congestion control mechanism for TCP over ATM networks. The FRED algorithm enhances the RED gateway algorithm, by using the fact that TCP connections should be allocated buffer space in proportion to their bandwidth-delay products. Through simulation, the effectiveness of the proposed FRED algorithm is shown as compared with the drop-tail and the original RED algorithms     

Experimental TCP Performance Evaluation on Diffserv AF PHBs over ATM SBR Service

In these days, IP network customers become sensitive to the QoS guarantee provided by the network. Amongst many QoS guarantee schemes, Diffserv (Differentiated Services) is the one of the most practical candidates for the next generation IP networks. Under these circumstances, the commercial network venders have already provided the Diffserv routers that support EF (Expedited Forwarding) PHB (Per Hop Behavior). However, AF (Assured Forwarding) PHB in Diffserv still has not been provided by commercial routers. Therefore, it is a realistic solution that SBR3 (Statistical Bit Rate 3) of ATM emulates AF PHB, but it is not clear whether TCP traffic over AF PHB emulated by ATM is differentiated from the best effort TCP traffic over DF (Default Forwarding) PHB. To confirm the differentiation, we have experimentally studied TCP performance through the link into which TCP connections over AF PHB and DF PHB are aggregated. This paper describes the experimental results and discusses the possibility of the TCP performance differentiation between AF PHB and DF PHB over ATM.     

One More Bit is Enough

Achieving efficient and fair bandwidth allocation while minimizing packet loss and bottleneck queue in high bandwidth-delay product networks has long been a daunting challenge. Existing end-to-end congestion control (e.g., TCP) and traditional congestion notification schemes (e.g., ${hbox {TCP+AQM/ECN}}$) have significant limitations in achieving this goal. While the XCP protocol addresses this challenge, it requires multiple bits to encode the congestion-related information exchanged between routers and end-hosts. Unfortunately, there is no space in the IP header for these bits, and solving this problem involves a non-trivial and time-consuming standardization process.      

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     

A Simulation Study of Scalable TCP and HighSpeed TCP in Geostationary Satellite Networks

This paper investigates the performance of two TCP enhancements (i.e., Scalable TCP and HighSpeed TCP), recently proposed for high-speed backbone networks with a very large bandwidth-delay product, in a geostationary satellite environment. Both persistent and elastic traffic patterns are considered, performance being evaluated in terms of TCP throughput and mean session delay, respectively. The impact of channel characteristics (packet error rate, correlation between successive losses) is widely discussed. Fairness issues are also accounted for, together with the impact of some system parameters, such as the satellite link bandwidth. Extensive comparisons are carried out among Scalable TCP, HighSpeed TCP and other congestion control schemes. The obtained results show the soundness for the use of such protocols in geostationary satellite networks. Indeed, both protocols permit to improve the performance of TCP connections in a wide range of channel conditions, showing (especially Scalable TCP) to be able to cope well with rainy conditions and to exploit a future increase in the satellite link capacity. This work was carried out within the framework of the SatNex Network of Excellence, http://www.satnex.org     

Multi-layer loss recovery in TCP over optical burst-switched networks

It is well-known that the bufferless nature of optical burst-switching (OBS) networks cause random burst loss even at low traffic loads. When TCP is used over OBS, these random losses make the TCP sender decrease its congestion window even though the network may not be congested. This results in significant TCP throughput degradation. In this paper, we propose a multi-layer loss-recovery approach with automatic retransmission request (ARQ) and Snoop for OBS networks given that TCP is used at the transport layer. We evaluate the performance of Snoop and ARQ at the lower layer over a hybrid IP-OBS network. Based on the simulation results, the proposed multi-layer hybrid ARQ + Snoop approach outperforms all other approaches even at high loss probability. We developed an analytical model for end-to-end TCP throughput and verified the model with simulation results.     

Analysis of Internet services in IP over ATM networks

This article presents a trace-driven analysis of IP over ATM services from an ATM-centric standpoint. We provide a characterization of TCP flows as VBR streams with burstiness (MBS) and throughput (SCR). On the other hand, a macroscopic analysis comprising percentage of flows and bytes per service, TCP average transaction duration, and bytes transferred both ways is also presented. The ATM link under analysis concentrates traffic from a large population of 1500 hosts from the Public University of Navarra campus network, that produce 1,700,000 TCP connections approximately in the measurement period of one week. The results obtained from such a wealth of data indicate that the sustained throughput of Web connections does not grow beyond 80 kb/s with 70 percent probability in the data transfer phase (i.e., in the established state), and we observe a strong influence of the connection establishment phase on the user-perceived throughput. On the other hand, the burstiness of individual TCP connections is rather small; namely, TCP connections do not produce bursts according to the geometric law given by slow start and commonly assumed in previously published studies     

Cross-layer speculative architecture for end systems and gateways in computer networks with lossy links

The throughput degradation of Transport Control Protocol (TCP)/Internet Protocol (IP) networks over lossy links due to the coexistence of congestion losses and link corruption losses is very similar to the degradation of processor performance (i.e., cycle per instruction) due to control hazards in computer design. First, two types of loss events in networks with lossy links are analogous to two possibilities of a branching result in computers (taken vs. not taken). Secondly, both problems result in performance degradations in their applications, i.e., penalties (in clock cycles) in a processor, and throughput degradation (in bits per second) in a TCP/IP network. This has motivated us to apply speculative techniques (i.e., speculating on the outcome of branch predictions), used to overcome control dependencies in a processor, for throughput improvements when lossy links are involved in TCP/IP connections. The objective of this paper is to propose a cross-layer network architecture to improve the network throughput over lossy links. The system consists of protocol-level speculation based algorithms at transport layer, and protocol enhancements at middleware and network layers that provide control and performance parameters to transport layer functions. Simulation results show that, compared with prior research, our proposed system is effective in improving network throughput over lossy links, capable of handling incorrect speculations, fair for other competing flows, backward compatible with legacy networks, and relatively easy to implement.     

门禁系统中通信平台的应用

传统门禁控制系统并不支持出入口控制器系统的网络化,本文通过引入通信平台,给出了一种实现TCP/IP的新方法,通信平台在操作软件客户端和控制器之间架起了TCP通道,通过对串口数据和客户端TCP/IP数据包进行转换可以任意客户端访问网络节点上的硬件服务器,实现TCP/IP通信,减少系统施工布线.     

基于视频序列的交通检测系统

基于人工电视监视的交通检测方法存在检测效率低、实时性差的缺点,提出了基于视频序列的交通参数和交通事件检测系统。将采集和预处理后的视频信号通过DSP处理,检测视频交通参数和交通事件,提取的交通参数和交通事件等分析结果通过TCP/IP网络传输协议传给视频分析识别终端,在视频分析识别终端上存储、显示交通参数与交通事件和视频信息,设置系统参数,同时可以进行查询、检索以及管理交通参数与交通事件。该系统实现了对车流量、车速、抛落物、行人和停车等交通参数与事件的实时性检测。