SCTP: new transport protocol for TCP/IP

For the past 20 years (1980-2000), applications and end users of the TCP/IP suite have employed one of two protocols: the transmission control protocol or the user datagram protocol. Yet some applications already require greater functionality than what either TCP or UDP has to offer, and future applications might require even more. To extend transport layer functionality, the Internet Engineering Task Force approved the stream control transmission protocol (SCTP) as a proposed standard in October 2000. SUP was spawned from an effort started in the IETF Signaling Transport (Sigtrans) working group to develop a specialized transport protocol for call control signaling in voice-over-IP (VoIP) networks. Recognizing that other applications could use some of the new protocol's capabilities, the IETF now embraces SCTP as a general-purpose transport layer protocol, joining TCP and UDP above the IP layer. Like TCP, STCP offers a point-to-point, connection-oriented, reliable delivery transport service for applications communicating over an IP network     

机会网络中一种提高TCP性能的跨层改进算法

针对机会网络中TCP与传染路由结合时,传染路由的"洪泛"特性和机会网络的间断特性会使得TCP性能变差的问题,提出了一种基于传染路由协议和TCP/Reno协议的跨层改进算法——ACK-EPI.该算法对连接开始阶段的慢开始门限值进行修改,以避免因网络错误进入拥塞避免阶段而导致拥塞窗口增长速度过于缓慢.同时,为了避免网络中已经成功交付但仍存储在网络中其他节点的数据包的复本继续在网络中存在并扩散,造成网络资源浪费,算法还利用ACK作为到达通告来删除这些冗余数据包.通过深入的仿真分析和比较,结果表明ACK-EPI算法能明显改善TCP性能.     

一种TD-HSUPA系统的TCP优化方法

TCP在TD-HSUPA系统中的表现不理想,需要采取一些优化措施,在分析了TD-HSUPA系统的网络协议结构之后,提出了一种适用于TD-HSUPA系统的TCP优化方法--由RNC解析终端TCP连接的状态,在终端可能出现超时之前,由NodeB通过调度和信道分配相结合的方法改变终端物理层网络环境.经仿真表明,对于复杂多变的...     

基于收方参与的移动自组网TCP增强的路由协议

移动自组网多跳、动态拓扑的特性,使得路由信息易老化,源节点在路由失败后不能及时启动路由发现,造成TCP有效吞吐率随移动速度增加而下降。在DSR路由协议的基础上,提出一种新的基于接收方参与启动路由发现的新路由协议RP DSR,它通过接收方的ACK确认包启动路由发现,主动为发送方提供最新的路由信息,从而提高了自组网的TCP的性能。仿真结果表明,在节点快速移动环境中,RP DSR协议的TCP性能明显优于DSR协议。     

A reactive and dependable transport protocol for wireless mesh networks

In the last few years, several different mesh network architectures have been conceived by both industry and academia; however, many issues on the deployment of efficient and fair transport protocols are still open. One of these issues is rate adaptation, that is, how to allocate the network resources among multiple flows, while minimizing the performance overhead. In order to address this problem, in this paper, we first define an analytical framework for a very simple topology. The model allows us to study the performance of an adaptive and responsive transport protocol when the effect of the lower layers are ignored. The mathematical approach alone does not represent a feasible solution, but it contributes to determining the strengths and weaknesses of our proposal. The main novelty of the proposed solution is that the congestion control approach is based on a hop-by-hop mechanism, which allows nodes to adapt their transmitting rates in a distributed way and to keep track of dynamic multi-hop network characteristics in a responsive manner. This is in contrast with classical literature solutions, founded on an end-to-end support. Anyway, to ensure the reliability, a coarse-grained end-to-end algorithm is integrated with the proposed hop-by-hop congestion control mechanism to provide packet level reliability at the transport layer. Performance evaluation, via extensive simulation experiments, shows that the proposed protocol achieves a high performance in terms of network throughput.     

A compression-gathering technique for improving the performance of ISO 11898 networks

ISO 11898 (CAN) networks suffer from serious limitations on both the maximum network extension and the allowable bit rate because of the particular arbitration technique they adopt.

In this paper, a modification of the basic ISO 11898 protocol is described which combines the content compression and data gathering techniques and provides high performance services for periodic data exchanges and message transmissions.

The new protocol yields, on the average, a 10-times performance improvement over conventional CAN without worsening neither the responsiveness nor the bus length, and also features a satisfactory degree of compatibility with the original standard protocol.     

一种自适应改进无线TCP性能的算法

对采用TCP协议传输数据的实现过程及其在无线网络中可能遇到的问题进行了描述，针对这些问题提出了一种算法对无线链路下一时刻的误码率进行了估计，根据这个估计对TCP进行了修改以自适应选取合适的TCP报文段尺寸来传输数据。在网络仿真器（NS2）中仿真一个误码率时变的无线信道，把修改后的TCP置于NS仿真环境中进行仿真，实验结果表明了该改进方法有效。     

基于RTT自适应的无线TCP改进算法

针对TCP Reno在无线环境下的性能恶化问题,在研究分析TCP Reno拥塞控制算法问题的基础上,提出一种基于RTT自适应的改进算法.该算法实现了丢包区分的拥塞窗口与慢启动门限调整,减轻了传统TCP由于无法区分拥塞丢包与误码丢包、盲目将拥塞窗口减半带来的性能下降.分析了该算法的可行性,并通过NS仿真对其吞吐量、带宽利用率、公平性等指标进行评估.仿真结果表明,相对TCP Reno,改进算法实现了无线环境下的TCP性能改善,同时具有一定的友好性与公平性.     

Investigating the performance of TCP in mobile ad hoc networks

Mobile ad hoc networks have several inherent characteristics (e.g. dynamic topology, time-varying and bandwidth constrained wireless channels, multi-hop routing, and distributed control and management). The goal of this work is to investigate the impact of these characteristics on the performance of TCP. First, we investigate throughput performance of TCP as a function of path length (i.e. multiple wireless hops), node mobility, and traffic intensity. Next, we examine the ‘fairness’ of the ad hoc network with regard to equal sharing of network bandwidth among multiple TCP flows. Third, we evaluate the impact of two on-demand routing protocols (i.e. AODV and DSR) on the throughput of TCP. Finally, a factorial design experiment is conducted to quantify the effects and interactions of three factors, which influence the throughput of TCP. These factors include routing, node speed, and node pause time. Two key results were observed. Results show that traffic intensity (e.g. number of concurrent flows) is significantly affects TCP throughput, suggesting the need for congestion control, scheduling and traffic management schemes. Second, source routing achieves higher throughputs while also generating significantly less routing overhead than AODV. Results also show that in some instances, the fairness of the network is very uneven among concurrent TCP flows, resulting in several sending stations achieving very little or no throughput.     

A high performance backoff protocol for fast execution of composite web services

Using web services to expose applications over the Internet is now a widely accepted practice. Currently, there are several ongoing efforts that provide ways to effectively compose web services distributed across different organizations. One of the problems underlying the deployment of such composite services on the web, however, is service co-allocation that arises when a composite service needs to ensure all the required component services to be available for execution at the same time. Motivated by this, this paper presents a new decentralized protocol, named web service co-allocation protocol (WSCP), which can facilitate fast execution of composite web services. The proposed framework is an enhancement of the famous two phase commit protocol through the incorporation of tentative hold phase as well as the employment of a new high performance backoff protocol developed to better address the dynamics of the service co-allocation problem. The simulation results show that the proposed approach yields significant improvements over existing protocols.     

AppTCP: The design and evaluation of application-based TCP for e-VLBI in fast long distance networks

Electric Very Long Baseline Interferometry (e-VLBI) is a typical astronomical interferometry used in radio astronomy. It allows observations of an object that are made simultaneously by many radio telescopes to be combined, emulating a telescope with the size equal to the maximum separation between the telescopes. The main requirements of transporting e-VLBI data are the high and constant transmission rate. However, the traditional TCP and its variants cannot meet these requirements. In an effort to solve the problem of transporting e-VLBI data in fast long distance networks, we propose an application-based TCP (AppTCP) congestion control algorithm, using Closed-Loop Control theory to keep the stable and constant transmission rate. AppTCP can swiftly reach the required transmission rate by increasing the congestion control window, and keep the transmission rate and allows the other TCP flows to share the remaining bandwidth. We further conduct extensive experiments in both fast long distance network test-bed and actual national networks (i.e., from Beijing to Shanghai in China) and international networks (i.e., from Hongkong in China to Chicago in USA) to evaluate and verify the performance and effectiveness of AppTCP. The results show that the AppTCP can effectively utilize the link capacity and maintain the constant rate during the data transmission, and its performance significantly outperforms that of the existing TCP variants.     

MRTP: a multiflow real-time transport protocol for ad hoc networks

Real-time multimedia transport has stringent quality of service requirements, which are generally not supported by current network architectures. In emerging mobile ad hoc networks, frequent topology changes and link failures cause severe packet losses, which degrade the quality of received media. However, in such mesh networks, there usually exist multiple paths between any source and destination nodes. Such path diversity has been demonstrated to be effective in combating congestion and link failures for improved media quality. In this paper, we present a new protocol to facilitate multipath transport of real-time multimedia data. The proposed protocol, the multiflow real-time transport protocol (MRTP), provides a convenient vehicle for real-time applications to partition and transmit data using multiple flows. We demonstrate through analysis that data partitioning, which is an essential function of MRTP, can effectively reduce the short-range dependence of multimedia data, thus improving its queueing performance in underlying networks. Furthermore, we show that a few flows are sufficient for MRTP to exploit most of the benefits of multipath transport. Finally, we present a comprehensive simulation study on the performance of MRTP under a mobile ad hoc network. We show that with one additional path, MRTP outperformed single-flow RTP by a significant margin.     

Unreliable transport protocol using congestion control for high-speed networks

Currently there is no control for the real-time traffic of multimedia applications using UDP (User Datagram Protocol) in high-speed networks. Therefore, although a number of high-speed TCP (Transmission Control Protocol) protocols have been developed for gigabit-speed (or faster) links, the real-time traffic could also congest the network and result in unfairness and throughput degradation of TCP traffic. In this paper, a new unreliable transport protocol, FAST DCCP, is presented for the real-time traffic in high-speed networks. FAST DCCP is based on the DCCP protocol and adopts the FAST scheme to realize congestion control. Some modifications have been made to the mechanisms inherited from DCCP so as to let the proposed protocol can efficiently operate under a large size window. In addition, an enhanced protocol, EEFAST DCCP, using the measurements of one-way delay to dynamically adjust the window size is also proposed to improve the throughput of FAST DCCP with the effect of reverse traffic. Simulation results show that FAST DCCP not only can satisfy the requirements of real-time data delivery, but also perform well in bandwidth utilization and fairness in high-speed networks. Meanwhile, EEFAST DCCP is able to effectively conquer the throughput degradation caused by the reverse traffic.     

An improved transport layer protocol for wireless sensor networks

Wireless sensor networks (WSNs) are a kind of communication networks having independent sensor nodes that form multi-hop ad hoc network to transfer data. In the past few years, various transport control protocols in wireless sensor networks (WSNs) have been developed and proposed in the literature. In this paper, we have analyzed pump slowly, fetch quickly (PSFQ) protocol and presented an improved transport layer protocol for wireless sensor networks. The improved protocol has been analyzed based on various factors such as average latency and average error tolerance and it is found that the proposed protocol is better than PSFQ in terms of these factors.     

On-line retrainable neural networks: improving the performance ofneural networks in image analysis problems

A novel approach is presented in this paper for improving the performance of neural-network classifiers in image recognition, segmentation, or coding applications, based on a retraining procedure at the user level. The procedure includes: 1) a training algorithm for adapting the network weights to the current condition; 2) a maximum a posteriori (MAP) estimation procedure for optimally selecting the most representative data of the current environment as retraining data; and 3) a decision mechanism for determining when network retraining should be activated. The training algorithm takes into consideration both the former and the current network knowledge in order to achieve good generalization. The MAP estimation procedure models the network output as a Markov random field (MRF) and optimally selects the set of training inputs and corresponding desired outputs. Results are presented which illustrate the theoretical developments as well as the performance of the proposed approach in real-life experiments.     

TRAP: a high performance protocol for wireless local area networks

The fixed number of available random addresses for randomly addressed polling (RAP) poses a significant problem in terms of scalability in case of many active stations. In such cases, the protocol's performance is significantly degraded. In this paper, we propose a TDMA-based randomly addressed polling (TRAP) protocol. The protocol employs a variable-length TDMA-based contention stage with the length based on the number of active stations. At the beginning of each polling cycle, the base station invites all active mobile stations to register their intention to transmit via transmission of a short pulse. The base station uses the aggregate received pulse in order to obtain an estimate of the number of contending stations and schedules the contention stage to comprise an adequate number of time slots for these stations to successfully register their intention to transmit. Then it transmits a READY message carrying the number of time slots P. Each mobile station calculates a random address in the interval [0…P−1], transmits its registration request in the respective time slot and then the base station polls according to the received random addresses. Simulation results are presented that reveal the superiority of TRAP against the RAP protocol in case of medium and high offered loads. Furthermore, the implementation of the proposed protocol is much simpler than that of CDMA-based versions of RAP, since no extra hardware is needed for the orthogonal reception of the random addresses.     

基于快速传输协议实现卫星TCP性能的改善

在经由卫星通信系统的网络应用中, TCP的性能较低。分段连接机制作为应用最广的一种性能优化方案,可以改善TCP端对端的性能,而在此机制中,卫星链路部分采用何种传输协议直接决定性能改善的效果。XTP协议是一种应用于有线网络的高速传输协议,但经过深入研究发现：XTP协议同样也非常适用于卫星通信系统。所以建议：在分段连接机制中,采用XTP协议作为卫星链路的传输协议。性能测试结果表明采用该建议非常显著地提高了端对端TCP的性能。     

Simulation methodology and performance analysis of network coding based transport protocol in wireless big data networks

The Multi-Path, Multi-Hop (MPMH) communications have been extensively used in wireless network. It is especially suitable to big data transmissions due to its high throughput. To provide congestion and end-to-end reliability control, two types of transport layer protocols have been proposed in the literature: the TCP-based protocols and the rateless coding based protocols. However, the former is too conservative to explore the capacity of the MPMH networks, and the latter is too aggressive in filling up the communication capacity and performs poorly when dealing with congestions. To overcome their drawbacks, a novel network coding scheme, namely, Adjustable Batching Coding (ABC), was proposed by us, which uses redundancy coding to overcome random loss and uses retransmissions and window size shrink to relieve congestion. The stratified congestion control strategy makes the ABC scheme especially suitable for big data transmissions. However, there is no simulation platform built so far that can accurately test the performance of the network coding based transport protocols. We have built a modular, easy-to-customize simulation system based on an event-based programming method, which can simulate the ABC-based MPMH transport layer behaviors. Using the proposed simulator, the optimal parameters of the protocol can be fine-tuned, and the performance is superior to other transport layer protocols under the same settings. Furthermore, the proposed simulation methodology can be easily extended to other variants of MPMH communication systems by adjusting the ABC parameters.