End-to-end Internet packet dynamics

We discuss findings from a large-scale study of Internet packet dynamics conducted by tracing 20000 TCP bulk transfers between 35 Internet sites. Because we traced each 100-kbyte transfer at both the sender and the receiver, the measurements allow us to distinguish between the end-to-end behavior due to the different directions of the Internet paths, which often exhibit asymmetries. We: (1) characterize the prevalence of unusual network events such as out-of-order delivery and packet replication; (2) discuss a robust receiver-based algorithm for estimating “bottleneck bandwidth” that addresses deficiencies discovered in techniques based on “packet pair;” (3) investigate patterns of packet loss, finding that loss events are not well modeled as independent and, furthermore, that the distribution of the duration of loss events exhibits infinite variance; and (4) analyze variations in packet transit delays as indicators of congestion periods, finding that congestion periods also span a wide range of time scales     

A new TCP for persistent packet reordering

Most standard implementations of TCP perform poorly when packets are reordered. In this paper, we propose a new version of TCP that maintains high throughput when reordering occurs and yet, when packet reordering does not occur, is friendly to other versions of TCP. The proposed TCP variant, or TCP-PR, does not rely on duplicate acknowledgments to detect a packet loss. Instead, timers are maintained to keep track of how long ago a packet was transmitted. In case the corresponding acknowledgment has not yet arrived and the elapsed time since the packet was sent is larger than a given threshold, the packet is assumed lost. Because TCP-PR does not rely on duplicate acknowledgments, packet reordering (including out-or-order acknowledgments) has no effect on TCP-PR's performance. Through extensive simulations, we show that TCP-PR performs consistently better than existing mechanisms that try to make TCP more robust to packet reordering. In the case that packets are not reordered, we verify that TCP-PR maintains the same throughput as typical implementations of TCP (specifically, TCP-SACK) and shares network resources fairly. Furthermore, TCP-PR only requires changes to the TCP sender side making it easier to deploy.     

Reorder notifying TCP (RN‐TCP) with explicit packet drop notification (EPDN)

Numerous studies have shown that packet reordering is common, especially in networks where there is high degree of parallelism and different link speeds. Reordering of packets decrease the TCP performance of a network, mainly because it leads to overestimation of the congestion in the network. In this paper, we analyse the performance of networks when reordering of packets occur. We propose a proactive solution that could significantly improve the performance of the network when reordering of packets occurs. We report results of our simulation experiments, which support this claim. Our solution is based on enabling the senders to distinguish between dropped packets and reordered packets. Copyright © 2005 John Wiley & Sons, Ltd.     

The effect of packet reordering in a backbone link on application throughput

Packet reordering in the Internet is a well-known phenomenon. As the delay and speed of backbone links continue to increase, what used to be a negligible amount of packet reordering may now, combined with some level of dropped packets, cause multiple invocations of fast recovery within a TCP window. This may result in a significant drop in link utilization and hence in application throughput. What adds to the difficulty is that packet reordering is a silent problem. It may result in significant application throughput degradation while leaving little to no trace. In this article we try to measure and quantify the effect of reordering packets in a backbone link that multiplexes multiple TCP flows on application throughput. Different operating systems and delay values as well as various types of flow mixes were tested in a laboratory setup. The results show that only a small percentage of reordered packets, by at least three packet locations, in a backbone link can cause significant degradation of application throughput. Long flows are affected most. Due to the potential impact of this phenomenon, minimization of packet reordering as well as mitigating the effect algorithmically should be considered.     

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.     

A Network Layer Approach to Enable TCP over Multiple Interfaces

The mobile Internet is set to become ubiquitous with the deployment of various wireless technologies. When heterogeneous wireless networks overlap in coverage, a mobile terminal can potentially use multiple wireless interfaces simultaneously. In this paper, we motivate the advantages of simultaneous use of multiple interfaces and present a network layer architecture that supports diverse multi-access services. Our main focus is on one such service provided by the architecture: Bandwidth Aggregation (BAG), specifically for TCP applications.While aggregating bandwidth across multiple interfaces can improve raw throughput, it introduces challenges in the form of packet reordering for TCP applications. When packets are reordered, TCP misinterprets the duplicate ACKS received as indicative of packet loss and invokes congestion control. This can significantly lower TCP throughput and counter any gains that can be had through bandwidth aggregation. To improve overall performance of TCP, we take a two-pronged approach: (1) We propose a scheduling algorithm that partitions traffic onto the different paths (corresponding to each interface) such that reordering is minimized. The algorithm estimates available bandwidth and thereby minimizes reordering by sending packet pairs on the path that introduces the least amount of delay. (2) A buffer management policy is introduced at the client to hide any residual reordering from TCP. We show through simulations that our network-layer approach can achieve good bandwidth aggregation under a variety of network conditions.Kameswari Chebrolu is an assistant professor in the electrical department of Indian Institute of Technology, Kanpur. Her research interests are in the areas of wireless network architecture, protocol design and analysis. Kameswari obtained her M.S. and Ph.D. degree in Electrical and Computer Engineering from University of California at San Diego, in 2001 and 2004 respectively.Bhaskaran Raman received his B.Tech in Computer Science and Engineering from Indian Institute of Technology, Madras in May 1997. He received his M.S. and Ph.D. in Computer Science from University of California, Berkeley, in 1999 and 2002 respectively. He joined as faculty in the CSE department at Indian Institute of Technology, Kanpur (India) starting June 2003. His research interests and expertise are in communication networks, wireless/mobile networks, large-scale Internet-based systems, and Internet middleware services.Ramesh R. Rao is a professor at the University of California, San Diego (UCSD). He is a member of the faculty of Irwin and Joan Jacobs School of Engineering, since 1984. Professor Rao is the former director of UCSD’s Center for Wireless Communications (CWC), and the current Director of the San Diego Division of the California Institute of Telecommunications and Information Technology [Cal-(IT)²]. As Director of the San Diego Division of Cal-(IT)², he leads several interdisciplinary, collaborative projects. His research interests include architectures, protocols and performance analysis of computer and communication networks, and he has published extensively on these topics. He received his B.E. from the University of Madras and his M.S. and Ph.D. from the University of Maryland at College Park. Most recently, Dr. Rao was honored by being appointed the first holder of the Qualcomm Endowed Chair in Telecommunications and Information Technologies.     

无线Mesh网中基于网络编码的多路径TCP研究

在无线多跳网络中,本地重传和网络编码已经被成功地应用到多路径技术上以增加吞吐量并减少丢包。然而,在提高UDP传输性能的同时,也产生了数据包重排序和延迟等副作用,严重影响了TCP性能。针对此问题,主要提出一种基于网络编码的多路径传输方案NC-MPTCP,即在无线mesh网络的多条路径中引入网络编码、执行拥塞控制以及使用一个基于信用的方法控制节点的传输速率,提高网络的吞吐量以及增加网络传输的可靠性。该方案使用一个简单的算法,评估丢包率以及发送线性组合数据包的速率,用来降低目的节点的数据包解码延迟和防止TCP的超时重传。仿真结果表明设计的NC-MPTCP有效。     

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.     

Transport protocols for Internet-compatible satellite networks

We address the question of how well end-to-end transport connections perform in a satellite environment composed of one or more satellites in geostationary orbit (GEO) or low-altitude Earth orbit (LEO), in which the connection may traverse a portion of the wired Internet. We first summarize the various ways in which latency and asymmetry can impair the performance of the Internet's transmission control protocol (TCP), and discuss extensions to standard TCP that alleviate some of these performance problems. Through analysis, simulation, and experiments, we quantify the performance of state-of-the-art TCP implementations in a satellite environment. A key part of the experimental method is the use of traffic models empirically derived from Internet traffic traces. We identify those TCP implementations that can be expected to perform reasonably well, and those that can suffer serious performance degradation. An important result is that, even with the best satellite-optimized TCP implementations, moderate levels of congestion in the wide-area Internet can seriously degrade performance for satellite connections. For scenarios in which TCP performance is poor, we investigate the potential improvement of using a satellite gateway, proxy, or Web cache to “split” transport connections in a manner transparent to end users. Finally, we describe a new transport protocol for use internally within a satellite network or as part of a split connection. This protocol, which we call the satellite transport protocol (STP), is optimized for challenging network impairments such as high latency, asymmetry, and high error rates. Among its chief benefits are up to an order of magnitude reduction in the bandwidth used in the reverse path, as compared to standard TCP, when conducting large file transfers. This is a particularly important attribute for the kind of asymmetric connectivity likely to dominate satellite-based Internet access     

Round-Robin分配方式下多径传输的重排序性能分析

基于包分配的多径传输在接收端所引起的数据包乱序严重影响TCP的传输性能。针对此问题,从排队论的角度出发对Round-Robin分配方式下多径传输的重排序问题进行了分析。从数据包的乱序率、重排序时间和端到端总时间3个方面考察了路径差异与路径数目对多径传输的性能的影响。结果表明,在采用两条路径传输时,应使得两条路径的传输速率近似相同;在路径传输速率相同的条件下,为明显地提升多径传输的重排序性能,路径数目应不超过4条。     

Potential performance bottleneck in Linux TCP

Transmission control protocol (TCP) is the most widely used transport protocol on the Internet today. Over the years, especially recently, due to requirements of high bandwidth transmission, various approaches have been proposed to improve TCP performance. The Linux 2.6 kernel is now preemptible. It can be interrupted mid‐task, making the system more responsive and interactive. However, we have noticed that Linux kernel preemption can interact badly with the performance of the networking subsystem. In this paper, we investigate the performance bottleneck in Linux TCP. We systematically describe the trip of a TCP packet from its ingress into a Linux network end system to its final delivery to the application; we study the performance bottleneck in Linux TCP through mathematical modelling and practical experiments; finally, we propose and test one possible solution to resolve this performance bottleneck in Linux TCP. Copyright © 2007 John Wiley & Sons, Ltd.     

The impact of multihop wireless channel on TCP performance

This paper studies TCP performance in a stationary multihop wireless network using IEEE 802.11 for channel access control. We first show that, given a specific network topology and flow patterns, there exists an optimal window size W* at which TCP achieves the highest throughput via maximum spatial reuse of the shared wireless channel. However, TCP grows its window size much larger than W* leading to throughput reduction. We then explain the TCP throughput decrease using our observations and analysis of the packet loss in an overloaded multihop wireless network. We find out that the network overload is typically first signified by packet drops due to wireless link-layer contention, rather than buffer overflow-induced losses observed in the wired Internet. As the offered load increases, the probability of packet drops due to link contention also increases, and eventually saturates. Unfortunately the link-layer drop probability is insufficient to keep the TCP window size around W'*. We model and analyze the link contention behavior, based on which we propose link RED that fine-tunes the link-layer packet dropping probability to stabilize the TCP window size around W*. We further devise adaptive pacing to better coordinate channel access along the packet forwarding path. Our simulations demonstrate 5 to 30 percent improvement of TCP throughput using the proposed two techniques.     

Performance Enhancement of Fast Handover for MIPv6 by Reducing Out-of-Sequence Packets

FMIPv6 can reduce packet loss using a tunnel-based handover mechanism which relies on L2 triggers, such as transmitting a packet from a previous access router (PAR) to a new access router (NAR). However, this mechanism may result in decreasing the performance of TCP due to out-of-sequence packets arriving between the tunneled packets from the Home Agent and PAR, and the directly transmitted packets from the correspondent node (CN). In this paper, we propose a new scheme called EF-MIPv6 that uses a modified snoop protocol to prevent the packet reordering problem. This new scheme can prevent sequence reordering of data packets and improve the performance of TCP using enhanced fast binding update (EF-BU). This approach requires modification of the TCP header to execute the last packet expression from the PAR, include a new polling data packet, and use the modified access point system. Simulation results demonstrate that managing the packet sequence in our proposed scheme greatly increases the overall TCP performance in a Mobile IPv6 and FMIPv6 networks.

Adjusting the TCP Sending Rate and Retransmissions after Retransmission Timeouts Based on One-Way Queuing Delay in Wireless Mesh Network

Wireless Mesh Network (WMN) is regarded as a viable solution to provide broadband Internet access flexibly and cost efficiently. Improving the performance of Transmission Control Protocol (TCP) in WMNs is an active research area in the networking community. The existing solutions proposed for improving the TCP performance has concentrated on differentiating the DATA packet drops in the forward direction induced by both network congestion as well as transmission errors. However, the recent studies show that in WMNs packet drops occur not only in the forward direction but also in the reverse direction particularly due to hidden terminal, hidden capture terminal, link asymmetry etc. The loss of ACK packets in the reverse direction cause frequent retransmission timeouts subject to needless retransmissions and unnecessary slowing down the growth of congestion window, which causes the performance degradation of TCP. In this paper, we introduce a sender side TCP algorithm, called detection of packet loss (DPL), which is capable to distinguish the type of packet drops either DATA or ACKs caused by transmission errors as well as network congestion based on one-way queuing delay and react accordingly. To justify our contributions, we implement DPL in Qualnet simulator and compare its performance against existing TCP solutions via extensive simulations. Our simulation results show that the proposed algorithm can accurately distinguish the type of packet drops whether it is a DATA or ACK caused by transmission error or congestion and can significantly improve the performance under a wide range of scenarios in WMNs.     

A packet-reordering solution to wireless losses in transmission control protocol

The wireless medium may cause substantial packet losses, rendering Transmission Control Protocol (TCP) inefficient. We propose a cross-layer solution by combining link-layer retransmission techniques and a solution for TCP packet reordering. It is costly to conduct link-layer retransmission with the constraint of orderly packet delivery. We require the link layer to provide reliable packet delivery, but without orderly delivery guarantee, thus transforming the problem of high packet error rates to the problem of packet reordering. The latter is dealt with by enhancing TCP with a solution for packet reordering. We justify our design by analyzing both general scenarios and the case of IEEE 802.11n. Our simulation results demonstrate that the proposed method is effective in improving TCP connection goodput in wireless networks.     

Network support for IP traceback

This paper describes a technique for tracing anonymous packet flooding attacks in the Internet back toward their source. This work is motivated by the increased frequency and sophistication of denial-of-service attacks and by the difficulty in tracing packets with incorrect, or “spoofed,” source addresses. We describe a general purpose traceback mechanism based on probabilistic packet marking in the network. Our approach allows a victim to identify the network path(s) traversed by attack traffic without requiring interactive operational support from Internet service providers (ISPs). Moreover, this traceback can be performed “post mortem”-after an attack has completed. We present an implementation of this technology that is incrementally deployable, (mostly) backward compatible, and can be efficiently implemented using conventional technology     

Packet scheduling algorithms and performance of a bufferedshufflenet with deflection routing

In a multihop network, packets go through a number of hops before they are absorbed at their destinations. In routing to its destination using minimum path, a packet at a node may have a preferential output link (the so-called “care” packet) or may not (the so-called “don't care” packet). Since each node in an optical multihop network may have limited buffer, when such buffer runs out, contention among packets for the same output link can be resolved by deflection. In this paper, we study packet scheduling algorithms and their performance in a buffered regular network with deflection routing. Using shufflenet as an example, we show that high performance (in terms of throughput and delay) can he achieved if “care” packets can be scheduled with higher priority than “don't care” packets. We then analyze the performance of a shufflenet with this priority scheduling given the buffer size per node. Traditionally, the deflection probability of a packet at a node is solved from a transcendental equation by numerical methods which quickly becomes very cumbersome when the buffer size is greater than one packet per node. By exploiting the special topological properties of the shufflenet, we are able to simplify the analysis greatly and obtain a simple closed-form approximation of the deflection probability. The expression allows us to extract analytically the performance trend of the shufflenet with respect to its buffer and network sizes. We show that a shufflenet indeed performs very well with only one buffer, and can achieve performance close to the store-and-forward case using a buffer size as small as four packets per node     

Performance Enhancement of On-Board Communication Networks Using Outage Prediction

A research area that has become increasingly important in recent years is that of on-board mobile communication, where users on a vehicle are connected to a local network that attaches to the Internet via a mobile router and a wireless link. In this architecture, link disruptions (e.g., due to signal degradation) may have an immediate impact on a potentially large number of connections. We argue that the advance knowledge of public transport routes, and their repetitive nature, allows a certain degree of prediction of impending link disruptions, which can be used to offset their catastrophic impact. Focusing on the transmission control protocol (TCP) and its extension known as Freeze-TCP, we present a detailed analysis of the performance improvement of TCP connections in the presence of disruption prediction. In particular, we propose a Markov model of Freeze-TCP that captures both the TCP behavior and the prediction+“freezing” feature and, using simulations, show that it accurately predicts the performance of the protocol. Our results demonstrate the significant throughput improvement that can be gained by disruption prediction, even with random packet losses or imperfect timing of the predicted disruptions.     

Understanding and improving TCP performance over networks withminimum rate guarantees

A large number of Internet applications are sensitive to overload conditions in the network. While these applications have been designed to adapt somewhat to the varying conditions in the Internet, they can benefit greatly from an increased level of predictability in network services. We propose minor extensions to the packet queueing and discard mechanisms used in routers, coupled with simple control mechanisms at the source that enable the network to guarantee minimal levels of throughput to different sessions while sharing the residual network capacity in a cooperative manner. The service realized by the proposed mechanisms is an interpretation of the controlled-load service being standardized by the Internet Engineering Task Force. Although controlled-load service can be used in conjunction with any transport protocol, our focus in this paper is on understanding its interaction with Transmission Control Protocol (TCP). Specifically, we study the dynamics of TCP traffic in an integrated services network that simultaneously supports both best-effort and controlled-load sessions. In light of this study, we propose and experiment with modifications to TCP's congestion control mechanisms in order to improve its performance in networks where a minimum transmission rate is guaranteed. We then investigate the effect of network transients, such as changes in traffic load and in service levels, on the performance of controlled-load as well as best-effort connections. To capture the evolution of integrated services in the Internet, we also consider situations where only a selective set of routers are capable of providing service differentiation between best-effort and controlled-load traffic. Finally, we show how the service mechanisms proposed here can be embedded within other packet and link scheduling frameworks in a fully evolved integrated services Internet     

Network security via reverse engineering of TCP code: vulnerabilityanalysis and proposed solutions

The Transmission Control Protocol/Internet Protocol (TCP/IP) suite is widely employed to interconnect computing facilities in today's network environments. However, there exist several security vulnerabilities in the TCP specification and additional weaknesses in a number of its implementations. These vulnerabilities may allow an intruder to “attack” TCP-based systems, enabling him/her to “hijack” a TCP connection or cause denial of service to legitimate users. The authors analyze the TCP code via a “reverse engineering” technique called “program slicing” to identify several of these vulnerabilities, especially those that are related to the TCP state-transition diagram. They discuss many of the flaws present in the TCP implementation of many widely used operating systems, such as SUNOS 4.1.3, SVR4, and ULTRIX 4.3. The corresponding TCP attack “signatures” (including the well-known 1994 Christmas Day Mitnick Attack) are described, and recommendations are provided to improve the security state of a TCP-based system (e.g., incorporation of a “timer escape route” from every TCP state). Also, it is anticipated that wide dissemination of this article's results may not only lead to vendor patches to TCP code to plug security holes, but also raise awareness of how program slicing may be used to analyze other networking software and how future designs of TCP and other software can be improved