Dynamics of TCP traffic over ATM networks

Investigates the performance of transport control protocol (TCP) connections over ATM networks without ATM-level congestion control and compares it to the performance of TCP over packet-based networks. For simulations of congested networks, the effective throughput of TCP over ATM can be quite low when cells are dropped at the congested ATM switch. The low throughput is due to wasted bandwidth as the congested link transmits cells from “corrupted” packets, i.e., packets in which at least one cell is dropped by the switch. The authors investigate two packet-discard strategies that alleviate the effects of fragmentation. Partial packet discard, in which remaining cells are discarded after one cell has been dropped from a packet, somewhat improves throughput. They introduce early packet discard, a strategy in which the switch drops whole packets prior to buffer overflow. This mechanism prevents fragmentation and restores throughput to maximal levels     

ATM交换机EPD性能分析

利用ＡＴＭ网络传送ＴＣＰ分组时的一个重要问题是分组中任何一个信元丢失都半导致分组的重传,为解决这一问题,一个方法是发生拥塞时交换机有选择丢弃信元,在有限的存储器容量下,交换机智能化的信元集合源状态与缓存器占用的联合分布,基于这一结果,本文推导出的ＥＰＤ系统的分组丢失率的上界和下界,通过分析比较发现,使用ＥＰＤ的系统,当门限设为缓存器大小时其性能将优于无控制系统。     

Packet loss performance of selective cell discard schemes in ATMswitches

In transport-layer protocols such as TCP over ATM networks, a packet is discarded when one or more cells of that packet are lost, and the destination node then requires its source to retransmit the corrupted packet. Therefore, once one of the cells constituting a packet is lost, its subsequent cells of the corrupted packet waste network resources. Thus, discarding those cells will enable us to efficiently utilize network resources, and will improve the packet loss probability. We focus on tail dropping (TD) and early packet discard (EPD) as selective cell discard schemes which enforce the switches to discard some of the arriving cells instead of relaying them. We exactly analyze the packet loss probability in a system applying these schemes. Their advantages and limits are then discussed based on numerical results derived through the analysis     

SWFA: a new buffer management mechanism for TCP over ATM-GFR

We investigate packet discarding schemes for transport control protocol (TCP) over asynchronous transfer mode with guaranteed frame rate service. In this letter, we propose the selective weighted fair allocation (SWFA) scheme, which discards packets from selected sessions. 15 TCP connections with equal minimum cell rate (MCR) and unequal MCRs are simulated. The SWFA scheme with per-virtual connection (VC) queuing is compared with early packet discard (EPD) with first-in, first-out queuing, EPD with per-VC queuing, and differential fair buffer allocation with per-VC queuing. Our results show that SWFA with per-VC queuing achieves significant enhancement on throughputs, goodputs, and fairness.     

Performance and buffering requirements of Internet protocols overATM ABR and UBR services

ATM networks are quickly being adopted as backbones over various parts of the Internet. This article studies the dynamics and performance of the TCP/IP protocol over the ABR and UBR services of ATM networks. Specifically the buffering requirements in the ATM switches as well as the ATM edge devices. It is shown that with a good switch algorithm, ABR pushes congestion to the edges of the ATM network while UBR leaves it inside the ATM portion. As a result, the switch ABR buffer requirement for zero-packet-loss high-throughput TCP transmission is a sublinear function of the number of TCP connections     

Improving TCP performance in ad hoc networks using signal strength based link management

Mobility in ad hoc networks causes frequent link failures, which in turn causes packet losses. TCP attributes these packet losses to congestion. This incorrect inference results in frequent TCP re-transmission time-outs and therefore a degradation in TCP performance even at light loads. We propose mechanisms that are based on signal strength measurements to alleviate such packet losses due to mobility. Our key ideas are (a) if the signal strength measurements indicate that a link failure is most likely due to a neighbor moving out of range, in reaction, facilitate the use of temporary higher transmission power to keep the link alive and, (b) if the signal strength measurements indicate that a link is likely to fail, initiate a route re-discovery proactively before the link actually fails. We make changes at the MAC and the routing layers to predict link failures and estimate if a link failure is due to mobility. We also propose a simple mechanism at the MAC layer that can help alleviate false link failures, which occur due to congestion when the IEEE 802.11 MAC protocol is used. We compare the above proactive and reactive schemes and also demonstrate the benefits of using them together and along with our MAC layer extension. We show that, in high mobility, the goodput of a TCP session can be improved by as much as 75% at light loads (when there is only one TCP session in the network) when our methods are incorporated. When the network is heavily loaded (i.e., there are multiple TCP sessions in the network), the proposed schemes can improve the aggregate goodput of the TCP sessions by about 14–30%, on average.     

TCP smart framing: a segmentation algorithm to reduce TCP latency

TCP Smart Framing, or TCP-SF for short, enables the Fast Retransmit/Recovery algorithms even when the congestion window is small. Without modifying the TCP congestion control based on the additive-increase/multiplicative-decrease paradigm, TCP-SF adopts a novel segmentation algorithm: while Classic TCP always tries to send full-sized segments, a TCP-SF source adopts a more flexible segmentation algorithm to try and always have a number of in-flight segments larger than 3 so as to enable Fast Recovery. We motivate this choice by real traffic measurements, which indicate that today's traffic is populated by short-lived flows, whose only means to recover from a packet loss is by triggering a Retransmission Timeout. The key idea of TCP-SF can be implemented on top of any TCP flavor, from Tahoe to SACK, and requires modifications to the server TCP stack only, and can be easily coupled with recent TCP enhancements. The performance of the proposed TCP modification were studied by means of simulations, live measurements and an analytical model. In addition, the analytical model we have devised has a general scope, making it a valid tool for TCP performance evaluation in the small window region. Improvements are remarkable under several buffer management schemes, and maximized by byte-oriented schemes.     

MANET中TCP性能改进研究

TCP是为有线网络所设计的,其拥塞控制机制中,假设丢包是由网络拥塞造成的这一结论在MANET中不再适用。MANET中的高信道误码率,路由频繁中断等因素都会造成丢包。TCP错误的将所有丢包事件都当作拥塞处理,造成了传输性能的极大下降。文中首先总结了MANET中导致TCP传输性能下降的主要原因,然后对现有的一些典型TCP改进方案进行了讨论,最后对这些技术方案进行了比较,并指出今后研究的重点。     

Delay analysis for CBR traffic under static-priority scheduling

We examine the delay performance of packets from constant-bit-rate (CBR) traffic whose delay is affected by non-real-time traffic. The delay performance is analyzed by solving the nD/D/1 queue with vacations. We obtain an exact and closed form solution, hence obviating the need of any approximations or numerical Laplace inversions. We then provide various numerical results for low-bit-rate transmission links, in which packets can experience large delay. From our quantitative evaluation, we conclude that there exists an optimum packet size for a given delay bound. In extremely slow links, such as modem links, transmission control protocol (TCP) packets should be segmented to reduce the CBR delay. We therefore investigate the delay impact of TCP packet sizes as well     

Traffic management for TCP/IP over satellite ATM networks

Several Ka-band satellite systems have been proposed that will use ATM technology to seamlessly transport Internet traffic. The ATM UBR, GFR, and ABR service categories have been designed for data. However, several studies have reported poor TCP performance over satellite ATM networks. We describe techniques to improve TCP performance over satellite ATM networks. We first discuss the various design options available for TCP end systems, IP-ATM edge devices, as well as ATM switches for long-latency connections. We discuss buffer management policies, guaranteed rate services, and the virtual source/virtual destination option in ATM. We present a comparison of ATM service categories for TCP transport over satellite links. The main goal of this article is to discuss design and performance issues for the transport of TCP over UBR, GFR, and ABR services for satellite ATM networks     

A new ATM adaptation layer for TCP/IP over wireless ATM networks

This paper describes the design and performance of a new ATM adaptation layer protocol (AAL‐T) for improving TCP performance over wireless ATM networks. The wireless links are characterized by higher error rates and burstier error patterns in comparison with the fiber links for which ATM was introduced in the beginning. Since the low performance of TCP over wireless ATM networks is mainly due to the fact that TCP always responds to all packet losses by congestion control, the key idea in the design is to push the error control portion of TCP to the AAL layer so that TCP is only responsible for congestion control. The AAL‐T is based on a novel and reliable ARQ mechanism to support quality‐critical TCP traffic over wireless ATM networks. The proposed AAL protocol has been validated using the OPNET tool with the simulated wireless ATM network. The simulation results show that the AAL‐T provides higher throughput for TCP over wireless ATM networks compared to the existing approach of TCP with AAL 5. This revised version was published online in July 2006 with corrections to the Cover Date.     

Buffer management and scheduling schemes for TCP/IP over ATM‐GFR

Today ATM technology is facing challenges from integrated service IP, IP switching, gigabit IP router and gigabit ethernet. Although ATM is approved by ITU‐T as the standard technology in B‐ISDN, its survivability is still in question. Since ATM‐UBR (unspecified bit rate) provides no service guarantee and ATM‐ABR (available bit rate) is still unattainable for most users, many existing users have little or no incentives to migrate to ATM technology. The guaranteed frame rate (GFR) service is introduced to deal with this dilemma. The GFR can guarantee the minimum cell rate (MCR) with fair access to excess bandwidth. This paper studies various schemes to support the GFR. We have studied different discarding and scheduling schemes, and compared their throughput and fairness when TCP/IP traffic is carried. Through simulations, it is shown that only per‐VC queueing with weighted round robin (WRR) can guarantee minimum cell rate. Among all the schemes that have been explored, we recommend dynamic threshold–early packet discard (DT–EPD) integrated with MCR+ (a WRR variant) to support the GFR service. Copyright © 2001 John Wiley & Sons, Ltd.     

Guaranteed frame rate: a better service for TCP/IP in ATM networks

Guaranteed frame rate, approved by the ATM Forum, is expected to become an important service category to efficiently support TCP/IP traffic in ATM networks. We first describe the GFR traffic contract in detail. We then present different types of switch implementations that have been proposed to support GFR. We analyze the performance of three of these switch implementations by simulations in two different network environments. These simulations show that the scheduler-based implementations provide a much better performance than the simple switch implementation. However, we also show that coupling an active packet discard mechanism to a scheduler-based switch implementation does not produce a performance gain when many TCP connections are multiplexed inside one ATM VC     

Packet Recycling and Delayed ACK for Improving the Performance of TCP over MANETs

Most of the schemes that were proposed to improve the performance of transmission control protocol (TCP) over mobile ad hoc networks (MANETs) are based on a feedback from the network, which can be expensive (require extra bandwidth) and unreliable. Moreover, most of these schemes consider only one cause of packet loss. They also resume operation based on the same stand-by parameters that might vary in the new route. Therefore, we propose two techniques for improving the performance of TCP over MANETs. The first one, called TCP with packet recycling (TCP-PR), allows the nodes to recycle the packets instead of dropping them after reaching the retransmission limit at the MAC layer. In the second technique, which is called TCP with adaptive delay window (TCP-ADW), the receiver delays sending TCP ACK for a certain time that is dynamically changed according to the congestion window and the trip time of the received packet. TCP-PR and TCP-ADW are simple, easy to implement, do not require network feedback, compatible with the standard TCP, and do not require distinguishing between the causes of packet loss. Our thorough simulations show that the integration of our two techniques improves the performance of TCP over MANETs.     

ATM cell delay and loss for best-effort TCP in the presence ofisochronous traffic

This paper reports the findings of a simulation study of the queueing behavior of “best-effort” traffic in the presence of constant bit-rate and variable bit-rate isochronous traffic. In this study, best-effort traffic refers to ATM cells that support communications between host end systems executing various applications and exchanging information using TCP/IP. The performance measures considered are TCP cell loss, TCP packet loss, mean cell queueing delay, and mean cell queue length. Our simulation results show that, under certain conditions, best-effort TCP traffic may experience as much as 2% cell loss. Our results also show that the probability of cell and packet loss decreases logarithmically with increased buffer size     

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     

Adding network-layer intelligence to mobile receivers for solving spurious TCP timeout during vertical handoff

With the help of mobile IP/IPv6 and soft handoff, ongoing TCP sessions can remain active and handoff packet loss can be avoided. However, TCP still faces several performance degradation issues due to the disparities in bandwidth and propagation delay between different access networks. Particularly, during vertical handoffs, some undesirable phenomena may erroneously trigger TCP congestion-control actions and thus degrade TCP performance. In this article we tackle the spurious timeout problem frequently associated with handovers from fast to slow links. We propose three network-layer schemes: fast ACK, slow ACK, and ACK delaying. These schemes require only minor modifications to the network layer of mobile receivers and no change to the TCP protocol and the TCP sender. The simulation results show that these schemes can effectively improve TCP performance during soft vertical handoffs     

Flow Routing and its Performance Analysis in Optical IP Networks

Optical packet-switching networks deploying buffering, wavelength conversion and multi-path routing have been extensively studied in recent years to provide high capacity transport for Internet traffic. However due to packet-based routing and switching, such a network could result in significant disorder and delay variation of packets when they are received by end users, thus increasing the burstiness of the Internet traffic and causing higher-layer protocol to malfunction. This paper addresses a novel routing and switching method for optical IP networks — flow routing, and its facilitating protocol. Flow routing deals with packet-flows to reduce flow corruption due to packet out-of-order, delay variation and packet loss, without using complicate control mechanism. Detailed performance analysis is given for output-buffered optical routers adopting flow routing. Two flow-oriented discarding techniques, i.e., flow discard (FD) and early flow discard (EFD), are discussed. Compared with optical packet-switching routers, a remarkable improvement of good-throughput is obtained in the optical flow-routers, especially under high congestion periods. We conclude that EFD behaves as a robust technique, which is more tolerant than FD to the change of traffic and transmission system factors.     

Novel Fastest Retransmission and Rate Control Schemes for Improving TCP Performance in Wireless Ad Hoc Networks

A number of different authors have considered the problem of performance degradation of transmission control protocol (TCP) in wireless ad hoc networks. We herein show that pauses in packet transmission due to packet losses are the fundamental cause of performance degradation of TCP in wireless ad hoc networks. To minimize the duration of packet transmission pauses, we propose a fast retransmission scheme for improving TCP performance in consideration of the inter-operability of previously deployed TCPs in wireless ad hoc networks. We also propose an additional rate control scheme for TCPs to reduce the probability of packet contention. Using OPNET and NS2 simulations, we show that our proposed schemes can provide a much better performance than conventional TCPs.     

Size-based scheduling to improve the performance of short TCP flows

The Internet today carries different types of traffic that have different service requirements. A large fraction of the traffic is either Web traffic requiring low response time or peer-to-peer traffic requiring high throughput. Meeting both performance requirements in a network where routers use droptail or RED for buffer management and FIFO as the service policy is an elusive goal. It is therefore worthwhile to investigate alternative scheduling and buffer management policies for bottleneck links. We propose to use the least attained service (LAS) policy to improve the response time of Web traffic. Under LAS, the next packet to be served is the one belonging to the flow that has received the least amount of service. When the buffer is full, the packet dropped belongs to the flow that has received the most service. We show that under LAS, as compared to FIFO with droptail, the transmission time and loss rate for short TCP flows are significantly reduced, with only a negligible increase in transmission time for the largest flows. The improvement seen by short TCP flows under LAS is mainly due to the way LAS interacts with the TCP protocol in the slow start phase, which results in shorter round-trip times and zero loss rates for short flows.