LwIP在LPC23/24XX以太网MAC控制器上的移植与应用

针对嵌入式设备接入以太网的需求,搭建了基于LPC23/24XX微控制器和DM9161A器件的硬件平台,在对LPC23/24XX以太网MAC(介质访问控制层)控制器的特性进行分析的基础上,开发和移植Lw IP协议栈的网络接口层和操作系统模拟层,实现了一个低成本嵌入式网络系统。通过对该系统进行TCP性能测试并在上面实现一个简单的WEB服务器,表明了该系统效率较高,具有实用价值。     

一种基于PIC单片机的嵌入式以太网接入系统

本文构建的系统是以PIC16F877单片机作为嵌入式系统的中央处理单元,并通过硬件集成电路S-7600A协议栈芯片来实现TCP/IP协议栈.因为以太网已经成为一种广泛的接入Internet的方式,可以通过RTL8019A以太网控制芯片把该系统接入以太网,然后通过以太网接入Internet,从而可实现对此嵌入式系统的远程访问和控制.     

基于Modbus/TCP协议的以太网监控软件的设计与实现

在对Modbus/TCP协议深入研究的基础上,设计了一种了基于Modbus/TCP协议的三层体系结构的工业以太网监控软件,重点阐述了三层结构中监控及网络管理级的数据通讯、数据管理及控制组态等备功能模块的设计,并对监控软件的实时性能和开环控制性能进行了测试,为将工业以太网直接应用于工业现场设备监控的研究提供了一种设计方案.     

基于以太网硬件协议栈的DSP嵌入式系统设计

应用以太网硬件协议栈芯片技术,设计并实现了具有网络数据高速传输功能的DSP嵌入式系统.以太网选用专用芯片WS100,该芯片适用于嵌入式系统,并具有Intemet协议.通过配置系统网络信息来实现整套系统的TCP和UDP连接,同时以Server或Client模式进行数据传输.以基于WS100的DSP嵌入式数据采集系统为例,通过合理编写程序和配置各寄存器,验证表明,系统可以通过以太网成功传输数据.     

基于LWIP的串口/以太网协议转换器的设计与实现

本文提出了一种串口／以太网协议转换器的设计方案,选择价格低廉的Coldfire系列MCF52235处理器作为控制单元,同时将μC/OS—II移植到该硬件平台上。本设计选择轻量级的LWIP作为以太网协议栈,在此基础上设计了串口与以太网之间的协议转换应用程序。     

网络混成传输话音接口板的设计

本文介绍话音接口板的工作原理与设计。该接口板的主要特点是将网络站点从模拟话音信号的处理中释放出来,从而实现话音与数据在网上的混成传输。本文最后介绍了提高话音质量的几个技术要点。     

网络化智能传感器中以太网接口设计

当前传感器向着微型化、数字化、智能化、网络化的方向发展,论文从网络化智能传感器的结构和特点入手,结合以太网和TCP/IP协议的功能特点,以W78E58单片机为核心控制器,以8029芯片为网络接入模块,提出了一种基于以太网的智能传感器设计思想,重点论述了该网络传感器的网络接口的软硬件设计方法.实验表明,采用该方法设计的以网络传感器具有良好的检测性能和网络通信性能,并且具有成本低廉、结构简单、组网灵活等优点.该网络传感器已经成功应用在广东省农业现代化的环境因子网络测控平台上.     

Profibus DP协议在以太网上的实现研究

讨论了Profibus DP协议功能在以太网上实现的可行性,对基于以太网的实时协议进行了改进.改进的实时协议基于以太网数据链路层的LLC子层,使以太网可以适应实时通讯的要求能够应用于设备级,并且与已经存在的Profibus DP现场总线上的应用保持了兼容性.还介绍了协议的实现和性能分析.     

嵌入式网络连通检测器的实现

介绍了以太网连通错误管理(CFM)的技术背景,指出纯软件开发的以太网的CFM模块具有局限性,并无法满足协议多变的特点.该模块是在SOPC系统上实现IEEE 802.1ag协议中连通性检测功能.系统采用Altera公司提供的DE2开发板作为实现平台,实现了能以毫秒级速度发送及分级告警的嵌入式网络连通检测器.用两个开发板模拟一个小型的网络,通过PC机来对开发板进行操作和控制,利用对开发板上的LCD显示屏及LED显示灯进行监测,验证连通检测器的功能及性能.     

UDP/IP协议下Modbus协议的研究与实现

工业以太网中Modbus/TCP已经成为以太网通信行业事实上的标准.本文对该通信标准进行深入的研究并进行改进,提出在UDP/IP协议上实施Modbus协议,随后从实时性和可靠性两个方面分析了Modbus/UDP协议栈的优点,然后结合实验室开发的一个测控终端组建测试系统对该协议栈进行实验检验.理论和实际表明:Modbus/UDP协议可操作性好容易实现、实时性和可靠性高,组网方便灵活.     

无线语音通信及视频监控系统设计

针对目前监控系统只有视频监控而缺少语音通信的现状,提出了一种基于ARM的无线语音通信及视频监控系统的设计方案。该系统采用自适应差分脉冲编码调制算法进行语音编码压缩,采用离散余弦变换算法对视频数据进行编码压缩,采用TCP/IP协议,通过无线局域网进行数据传输。测试结果表明,该系统不仅实现了视频监控及语音实时通信,且性能可靠,成本低。     

A real-time flood-routed voice protocol for fibre optic local area networks

A simple real-time flood routing voice (RTFRV) protocol which is suitable for point-to-point arbitrarily connected fibre optic local area networks is described. Simulation on a 3 Mbit/s nine-node mesh network which employs the protocol has shown that it is capable of supporting about 50 simultaneous two-way voice conversations (or 100 users) with packet delay of less than 10 ms and without any loss of information. The overall results suggest that voice transmission on such a network is feasible and that it performs better than voice transmission over Ethernet. Furthermore, the network is also very robust and allows easy network reconfiguration.     

An X.25-compatible protocol for packet voice communications

Developments in packet-switched networks have generated interest in how to integrate voice and data traffic in such networks. The paper deals with a proposed communications protocol for voice transmission within X.25 packet-switched networks. Basic protocol procedures for handling voice communications are examined, and possible formats for the identified types of voice frame and packet fitting for current CCITT standards governing packet data communications are presented.     

Data/voice integration for fibre optic LANs with arbitrary topology

A high-performance protocol for arbitrary point-to-point fibre optic local area networks with combined voice and data traffic is described. Simulation on a 3 Mbit/s nine-node mesh network has shown that it is capable of supporting a substantial number of data users and more than 100 active voice calls (200 active telephone users, coding rate 64 kbit/s without silence detection) with a packet delay of less than 10 ms and without any loss of information. The network performance is superior to the previous reported voice protocol which used the modified anarchy flood routing (MAFR) technique. Performance limitation due to growing network size for flood-routed networks is also discussed. A method of extending the flood-routing protocols over subnetworks which are connected via bridges is also presented. This method enables a large network to be divided into smaller subnetworks which reduces the number of links required.     

Performance of a distributed control LAN access protocol supporting voice and data traffic

This paper describes a local area network access protocol for integrated voice and data communications in a token passing ring. The transmission scheme is decentralized and suitable for both voice and data services. In this system access delay for voice packets can be bounded to a desired upper limit, even at heavy network load. The quality of speech remains unaffected at any level of offered data load and all voice packets are served within the specified delay limit. The protocol provides equal priority to pure data at normal network load while voice is given priority over pure data at heavy network load. The delay-throughput characteristic of both voice and data traffic has been evaluated at a transmission speed of 10 Mb s^-1 and it is shown that the mean delay for voice packets, at heavy data load, can be limited to a value equal to half the threshold token rotation time.     

Voice/data integration on Ethernet — backoff and priority considerations

Astudy of the CSMA/CD protocol as given in the Ethernet specifications, for real time (voice in particular) and mixed voice/data environments is described. The objectives of the study were to predict the Ethernet's behaviour under the current protocols, to point out potential improvements within the realm of the current specifications, and to improve understanding of the network behaviour in these environments. Results are given in terms of relative performance of the various system measures given as a function of system and traffic models. It is shown that the support provided to real time voice on Ethernet strongly depends on the assumptions made with regard to the coexisting data user's traffic and that with certain data traffic patterns the observed voice delay is much worse than previously reported. Finally, it is shown that without violating the Ethernet network specification, voice service can be improved by simple adjustments of the Ethernet retransmission ‘backoff’ algorithm.     

Voice transmission over ad hoc network adapting optimum approaches to maximize the performance

Although a lot of research and implementations have been performed for voice transmission over regular network and internet, a little is done for mobile ad hoc network (MANET). Like other computer networks, in MANET voice transmission is also very much demanding and necessary. Since the wireless links in an ad hoc network are highly error prone and may go down frequently because of node mobility, interference, channel fading, and the lack of infrastructure, it is very difficult and challenging to implement voice transmission over MANET. In this research, to maximize the performance of MANET during voice transmission we choose some parameters and methodological approaches e.g. Method of media access, selection of audio codec, selection of routing protocol etc in efficient and optimum way. Since voice applications consume more energy than typical applications, we use an energy aware routing protocol known as WEAC for the study. During the selection process of routing protocol we have a performance comparison study between Improved DSR and WEAC, where WEAC performs better consuming less energy in case of larger network. We also have a comparative study among several audio codecs (G.711, G.729 and G.723.1) where by simulation we show that the G.729 codec is more suitable to use for voice transmission over MANET in terms of latency. Finally, we show that it is possible to launch voice transmission with acceptable quality and throughput over MANET using G.729 and WEAC protocol.     

KNX/EIB通信协议的实时性分析与改进

KNX/EIB通信协议在其链路层使用CSMA/CA机制解决通信冲突问题,虽然提高了KNX/EIB网络防冲突能力,却造成同等级命令帧数据传输时延抖动非常大,传输实时性大打折扣。提出对KNX/EIB实时性的改进方法KNX/EIB-A,在不改变原有的通信协议栈的基础上,将一个调度程序应用于KNX/EIB通信协议的应用层与用户应用程序之间;将原有的分布式平等结构划分为分级主从结构,对数据命令帧的发送进行调度。最后通过对KNX/EIB-A进行分析及原型实现,证明了该方法通过有效地减轻传输时延的抖动从而改进了协议的实时性。     

一种基于星─多环拓扑的综合语音／数据网的设计

本文提出一种新型计算机局域网结构，它具有实电路存取和令牌存取双重功能，因而特别适用于综合语音／数据传输。本文介绍了这种新型网络体系结构和网络接口单元，阐述了实连接访问协议处理机。     

基于TCP/IP协议的语音传输的研究

讨论基于TCP／IP协议，用Winsock完成语音的端－端传输的方法。并给出了相应的实例程序。