A new burst assembly scheme based on the average packet delay and its performance for TCP traffic

We propose and evaluate a new burst assembly algorithm based on the average delay of the packets comprising a burst. This method fixes the average delay of the packets belonging to an assembled burst to a desired value $T_{\mathrm{AVE}}$ that may be different for each forwarding equivalence class (FEC). We show that the proposed method significantly improves the delay jitter experienced by the packets during the burst assembly process, when compared to that of timer-based and burst length-based assembly policies. Minimizing packet delay jitter is important in a number of applications, such as real-audio and streaming-video applications. We also find that the improvement in the packet delay jitter yields a corresponding significant improvement in the performance of TCP, whose operation depends critically on the ability to obtain accurate estimates of the round-trip times (RTT).     

Selective packet dropping for VoIP and TCP flows

The adoption of the IP protocol for serving diverse applications arises the need for mechanisms to prevent network congestion in scenarios with different traffic types (responsive and unresponsive) sharing limited network resources. To deal with this issue, a number of algorithms for active queue management (AQM) have been proposed. However, most of them do not observe the traffic type and usually disregard this knowledge. In this way, the provided service could not comply with the distinctive requirements of the different type of traffic, such as VoIP services, which demand bounded packet latency and loss rate.     

一种面向分组业务的新型OADM结构

分析了一种面向分组业务的新型光分插复用器(OADM)结构,采用光正交码的自相关性与互相关性,给出了全光地址的识别机理,并将其原理应用于新型的光网元节点中,说明了设计方法、应用示例和仿真结果。     

A new model for scheduling packet radio networks

Packet radio networks are modeled as arbitrary graphs by most researchers. In this paper we show that an arbitrary graph is an inaccurate model of the radio networks. This is true because there exists a large class of graphs which will not model the radio networks. Radio networks can be modeled accurately by a restricted class of graphs called the planar point graphs. Since the radio networks can accurately be modeled only by a restricted class of graphs, the NP-completeness results for scheduling using an arbitrary graph as the model, do not correctly reflect the complexity of the problem. In this paper we study the broadcast scheduling problem using the restricted class as the model. We show that the problem remains NP-complete even in this restricted domain. We give an O(n log n) algorithm when all the transceivers are located on a line. This revised version was published online in July 2006 with corrections to the Cover Date.     

A new loss recovery architecture for wireless TCP

The loss recovery architecture of TCP under wireless environment is considered. We propose sent-time ordered two lists architecture as an alternative to sequence number ordered single list architecture. By keeping the sent-time order, the recovery mechanism can be more efficient and-simpler and transmission decision is decoupled from loss recovery using the second list. Simulation results show the superiority of our mechanism.     

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.     

多目标控制系统中TCP粘包问题的解决方法

测控设备采用传输控制协议/网际协议(TCP/IP)作为基础协议,进行遥控多通道并行工作时易造成网络数据粘包.针对此问题,在对TCP协议特点进行分析的基础上,通过讨论TCP编程模型和流交付模型,研究了发生TCP数据流粘包时的网络数据特征,并分析了发生网络粘包的原因,提出了局域网环境下采用短连接、发送端及格式数据组包3种解决网络粘包的方法.实际测试结果证明了这3种方法的适应性及格式数据方法还原TCP网络数据包的有效性,可为基于TCP应用软件开发提供借鉴.     

光分组交换网的TCP性能研究

与传统光网络相比,光分组交换(OPS)网络具有高速、大吞吐量、低时延和能高效地承载IP业务等突出优点.而作为支撑下一代Internet发展的最有希望的骨干光网络,OPS网与传输控制协议/互联网协议(TCP/IP)的兼容性和支持度是一个值得深入研究的课题.文章以光突发交换(OBS)网为模型,对OPS网络中的TCP传输性能进行了研究.     

数据包大小对TCP流性能的影响

在互联网时代,如何提高网络通信质量一直是研究热点,当前网络通信大部分数据流都采用TCP流。对于TCP流性能的影响因素如丢包、时延已有大量研究,但是数据包大小对其的影响却罕有涉及。另一方面,网络通信的质量很大程度上取决于路由器的性能,而路由器的性能又很大程度上取决于路由器的缓存设置。因此本文从数据包大小这一新的切入点入手,研究在不同的路由器缓存下,其对TCP流性能的影响。论文介绍了TCP协议、路由器缓存、数据包等相关概念,提出了实验方案的网络拓扑模型,选择了NS2平台进行网络仿真。在实验中,我们通过改变数据包大小和路由器缓存容量,得到了对应的实验数据并进行了分析整理,总结出了TCP流吞吐量和丢包率在数据包大小改变时的变化规律,即丢包率随数据包大小的增加呈线性增长,路由器缓存越小丢包率越大;TCP流的吞吐量随数据包大小的增加缓慢变大,路由器缓存达到一定值后不再成为瓶颈因素。     

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.     

TCP performance over wireless MIMO channels with ARQ and packet combining

Multiple-input multiple-output (MIMO) wireless communication systems that employ multiple transmit and receive antennas can provide very high-rate data transmissions without increase in bandwidth or transmit power. For this reason, MIMO technologies are considered as a key ingredient in the next generation wireless systems, where provision of reliable data services for TCP/IP applications such as wireless multimedia or Internet is of extreme importance. However, while the performance of TCP has been extensively studied over different wireless links, little attention has been paid to the impact of MIMO systems on TCP. This paper provides an investigation on the performance of modern TCP systems when used over wireless channels that employ MIMO technologies. In particular, we focus on two representative categories of MIMO systems, namely, the BLAST systems and the space-time block coding (STBC) systems, and how the ARQ and packet combining techniques impact on the overall TCP performance. We show that, from the TCP throughput standpoint, a more reliable channel may be preferred over a higher spectral efficient but less reliable channel, especially under low SNR conditions. We also study the effect of antenna correlation on the TCP throughput under various conditions.     

基于HASH表和SYN计算的TCP包重组方法

本文提出一种基于HASH表和SYN计算的TCP会话重组方法,它利用 HASH表快速定位的特征和TCP包头中的SYN号进行多连接TCP会话重组,可解决IP数据包非按序到达和 TCP数据包重传问题。     

Average-value analysis of 802.11 WLANs with persistent TCP flows

In this letter we develop an average-value analysis of the TCP performance in 802.11 WLANs. Our model characterizes the equilibrium conditions of the network, and this method yields a simple, yet precise, estimate of the throughput obtained by persistent TCP flows. Via simulations we study the accuracy of the model predictions.     

A packet delay analysis for cellular digital packet data

We formulate and analyze a model of voice and data burst traffic for cellular digital packet data (CDPD). To develop a tractible model, we make some reasonable simplifying assumptions to model the voice traffic and one of the logical CDPD channels as an M/M/1 queue in a random environment. Rather than explicitly solve the resulting matrix quadratic equation, we develop an asymptotic analysis that gives us simple approximate formulas (with error bounds) for many performance measures of interest. In particular, one metric that we highlight comes from using our approximate mean data burst delay formula to characterize the performance of a set of interacting logical CDPD channels. We do so by describing it in terms of an equivalent number of independent, dedicated CDPD channels, We can use this metric to suggest what the optimal number of logical CDPD channels for a given system should be     

一种新的VANET网络链路丢包率估计算法

在车载自组织网络（VANET）协议体系中,网络链路层丢包率的估计和预测十分重要,因为它不仅决定了信息传输的效果,并且其大小将直接影响对上层服务协议的执行。因此,提出了一种基于后验期望估计（PEE）算法来预测和估算链路丢包率。根据对实验数据结果的分析,使用很少的探测包就可以快速准确地对链路丢包率进行估算,而且PEE算法优于最大似然估计（MLE）算法和期望传输次数（EXT）算法。     

TCP over OBS: Impact of consecutive multiple packet losses and improvements

In TCP over OBS networks, consecutive multiple packet losses are common since an optical burst usually contains a number of consecutive packets from the same TCP sender. In this paper, we first present a new theoretical method to analyze the behavior of Reno when consecutive multiple packet losses occur. Results of the analysis indicate that even a small number of consecutive multiple packet losses can force Reno to timeout. Then we propose B-Reno, a newly designed TCP implementation that can overcome Reno’s inefficiency in dealing with consecutive multiple packet losses over OBS networks and can avoid the shortcomings of New-Reno and SACK. Results of comprehensive simulations indicate that B-Reno over OBS networks can achieve a performance better than Reno and New-Reno, and that it can also achieve a performance similar to that of SACK. Moreover, B-Reno only needs some simple modifications to New-Reno at the sender’s protocol stack, and thus has less difficulty in deployment and less protocol complexity than that of SACK.

光分组交换网中的一种新模型

目前常用的网络理论模型如曼哈顿网络、洗牌网等均不能适应光分组网络研究的需要.文章提出了一种新的光网络模型———自相似环流网状网.该模型特别适合于光分组网络的研究.文章还研究了该模型的许多特性.     

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.     

Performance evaluation of TCP algorithms in multi‐hop wireless packet networks

Wireless packet ad hoc networks are characterized by multi‐hop wireless connectivity and limited bandwidth competed among neighboring nodes. In this paper, we investigate and evaluate the performance of several prevalent TCP algorithms in this kind of network over the wireless LAN standard IEEE 802.11 MAC layer. After extensively comparing the existing TCP versions (including Tahoe, Reno, New Reno, Sack and Vegas) in simulations, we show that, in most cases, the Vegas version works best. We reveal the reason why other TCP versions perform worse than Vegas and show a method to avoid this by tuning a TCP parameter— maximum window size. Furthermore, we investigate the performance of these TCP algorithms when they run with the delayed acknowledgment (DA) option defined in IETF RFC 1122, which allows the TCP receiver to transmit an ACK for every two incoming packets. We show that the TCP connection can gain 15 to 32 per cent good‐put improvement by using the DA option. For all the TCP versions investigated in this work, the simulation results show that with the maximum window size set at approximately 4, TCP connections perform best and then all these TCP variants differ little in performance. Copyright © 2001 John Wiley & Sons, Ltd.     

Wireless video streaming with TCP and simultaneous MAC packet transmission (SMPT)

Video streaming is expected to account for a large portion of the traffic in future networks, including wireless networks. It is widely accepted that the user datagram protocol (UDP) is the preferred transport protocol for video streaming and that the transmission control protocol (TCP) is unsuitable for streaming. The widespread use of UDP, however, has a number of drawbacks, such as unfairness and possible congestion collapse, which are avoided by TCP. In this paper we investigate the use of TCP as the transport layer protocol for streaming video in a multi‐code CDMA cellular wireless system. Our approach is to stabilize the TCP throughput over the wireless links by employing a recently developed simultaneous MAC packet transmission (SMPT) approach at the link layer. We study the capacity, i.e. the number of customers per cell, and the quality of service for streaming video in the uplink direction. Our extensive simulations indicate that streaming over TCP in conjunction with SMPT gives good performance for video encoded in a closed loop, i.e. with rate control. We have also found that TCP is unsuitable (even in conjunction with SMPT) for streaming the more variable open‐loop encoded video. Copyright © 2004 John Wiley & Sons, Ltd.