实现视音频信息服务器传输的IP网络

运用用户数据报协议 (UDP) ,实时传输控制协议 (RTP/RTCP)和H .32 3协议 ,设计实现解决视音频信息的实时传输服务器。设计中对视音频信息的打包格式采用MPEG 1的格式将信息打包后作为RTP报文的应用数据加上一个RTP头 ,给信息包再加上一个UDP头和IP头 ,组成一个完整的信息包后 ,采用非连接的UDP用户数据报的方式 ,将它发送出去 ,实现组播和多组会议 ,采用MPEG 1的数据压缩技术 ,可以较高的传输速率在宽带IP网络中的传输 ,实时性比较好     

基于UDP的可靠传输协议的研究与实现

在高速数据传输网络中,用户数据报协议(UDP)有着其他数据传输协议无法比拟的优势,但同时也存在着传输可靠性差的问题.文章作者在详细分析UDP特点的基础上,对其关键技术进行了改进,设计了一种可靠的传输机制.文章最后给出了在VxWorks操作系统下实现UDP可靠传输的方法和流程.     

SSCOP信令传输时延分析与检测

SSCOP协议是ATM宽带网络信令系统数据链路层的一部分,SSCOP信令链路为ATM网络元素之间提供可靠的信令信息传输。信令链路传输时延是ATM网络传输质量的一个重要参数,SSCOP协议的很多重要参数的取值都依赖于传输时延的大小。介绍一种SSCOP信令链路传输时延的检测方法,通过构造一种特殊的SSCOP报文来测试信令网络的传输质量,并计算信令报文的传输时延。对提高ATM信令系统的网络质量和优化SSCOP协议参数具有一定的现实意义。     

基于移动网络图像监控终端的设计与实现

文章介绍了一种移动图像监控设备,该设备以GPRS网络为移动通讯平台,通过因特网网络与监控中心实现信息交互。该设备采用一种建立在UDP协议基础上的保序重传协议解决移动网络大数据量数据流传输问题,避免了使用TCP协议易造成的移动网络传输效率降低问题和仅采用UDP协议造成的丢包、乱序问题,并在实践中得到验证。     

网络音像传输系统的软件设计-发送端

首先介绍了系统的方案,该方案采用G．729A标准和H．263标准分别对音频和视频数据进行压缩,并通过UDP协议对数据包进行网络传输。其次对系统的组成和工作流程做了简单概述,最后对语音传输系统中常见的问题,提出了相应的解决办法。     

数字视频网络传输层协议的选择

随着视频压缩技术的日益成熟，数字视频监控产品逐渐成为了市场的主流。人们迫切希望网络技术能够成功地应用于数字视频监控领域，以使人们能够通过网络实现异地监控。正是在这种背景下，“基于网络的数字视频传输”课题研究日显重要。本文围绕网络视频传输的实时性和传输质量两大重要指标，从“解决传输层协议问题”入手，通过对TCP和RTP的比较，提出了“基于UDP协议的RTP实时视频传输”的设计思想，较好地保证了数字视频传输的实时性和服务质量。     

一种移动网络数据传输协议的设计与应用

主要介绍了一种移动网数据传输协议(STP协议),该协议是为更好地实现移动网 数据流传输而提出的一种基于UDP协议的扩展协议。STP协议保留了UDP协议全部功能及 其高效的特点,同时结合移动网的实际特征,提供了可靠性机制和流量控制机制,弥补了UDP 协议的不足,并具有间断可重入、自适应速率传输等优势,可以为上层应用提供高效、可靠的数 据传输服务。     

基于流媒体技术的音频实时播放系统的实现

介绍了2种建立音频播放系统的方法：DirectSound技术以及直接调用低级音频设备。该系统基于RTP／RTCP实时协议控制音媒体流的传输，采用UDP作为传输层协议，在PC机＋MS Windows OS系统上，实现了音频流媒体播放器。     

流媒体在广电网络中的架构与实现

流媒体是采用流式传输的方式在Intemet中播放的媒体格式，将音频、视频等多媒体文件经过压缩，进行连续的传输。在流式传输的实现方案中，一般采用HrITP／TCP来传输控制信息，而用RTP／UDP来传输实时声音数据。文中介绍了流媒体技术在有线电视网络中建立宽带流媒体平台的结构和主要应用服务。     

基于GPRS的远程监控系统设计

针对当前输油管道压力监控现状,结合GPRS通信模块MC39i,给出了能够实现远程数据传输并可广泛应用于远程监控系统的设计方案。重点阐述了系统的硬件组成以及基于嵌入式实时操作系统FreeRTOS实现数据传输模块的软件设计;采用基于状态机的结构设计方法实现PPP链路层协议,实现GPRS模块与互连网络的连接;分析和比较了UDP和TCP两种传输层数据传输方式,并给出了基于uIP TCP/IP协议栈实现TCP和UDP传输协议的原理;对UDP协议的传输可靠性进行改进,采用带握手的UDP协议可以较好地满足实时性的可靠性要求,并对数据传输性能进行了测试。     

声纳信号处理中UDP协议数据传输研究与设计

为了在声纳系统中通过以太网口进行大批量、高速率的数据传输处理,在FPGA中硬件实现了嵌入式UDP协议栈,完成了架构设计、软件仿真验证及硬件实现。用FPGA硬件实现UDP协议栈,加速了网络数据处理能力,使信号传输速率达到了80MB/s,实现了千兆级通信,很好地提高了声纳系统中数据传输速率和系统性能。同时,用FPGA硬件实现UDP协议,栈减小了PCB版图面积和布局布线复杂度,提高了开发效率,有效地降低了开发成本。     

Game of protocols: Is QUIC ready for prime time streaming?

Quick User Datagram Protocol (UDP) Internet Connections (QUIC) is an experimental and low‐latency transport protocol proposed by Google, which is still being improved and specified in the Internet Engineering Task Force (IETF). The viewer's quality of experience (QoE) in HTTP adaptive streaming (HAS) applications may be improved with the help of QUIC's low‐latency, improved congestion control, and multiplexing features. We measured the streaming performance of QUIC on wireless and cellular networks in order to understand whether the problems that occur when running HTTP over TCP can be reduced by using HTTP over QUIC. The performance of QUIC was tested in the presence of network interface changes caused by the mobility of the viewer. We observed that QUIC resulted in quicker start of media streams, better streaming, and seeking experience, especially during the higher levels of congestion in the network and had a better performance than TCP when the viewer was mobile and switched between the wireless networks. Furthermore, we measured QUIC's performance in an emulated network that had a various amount of losses and delays to evaluate how QUIC's multiplexing feature would be beneficial for HAS applications. We compared the performance of HAS applications using multiplexing video streams with HTTP/1.1 over multiple TCP connections to HTTP/2 over one TCP connection and to QUIC over one UDP connection. To that effect, we observed that QUIC provided better performance than TCP on a network that had large delays. However, QUIC did not provide a significant improvement when the loss rate was large. Finally, we analyzed the performance of the congestion control mechanisms implemented by QUIC and TCP, and tested their ability to provide fairness among streaming clients. We found that QUIC always provided fairness among QUIC flows, but was not always fair to TCP.     

Capacity of a wireless LAN with voice and data services

The capacity of a carrier sense multiple-access wireless LAN with voice and data services using the TCP/IP protocol is analyzed to obtain a lower bound for the capacity of the wireless networks with voice and data services. The voice traffic is given a higher priority over the data traffic to accommodate the delay requirement for an acceptable quality of service. This is implemented by assigning the TCP protocol for data and the UDP protocol for voice. The relationship between the data throughput and the number of the voice users supported in this environment is analyzed by using a nonpreemptive queuing model. While the analysis in this paper can be applied to any voice encoding system, the improved multiband excitation voice encoding technique is adopted to provide a low transmission rate with an acceptable quality     

Timeliness of real-time IP communication in switched industrial Ethernet networks

Through several giant evolutionary steps, Ethernet has become an almost ubiquitous technology for communication. Being versatile enough to be employed in new and various fields of application, it is now making inroads in factories. However, automated systems are different from many other applications of Ethernet,first and foremost because they require the network technology to deliver real-time performance. In the present study, a number of critical aspects of Ethernet, usually referred to as an Industrial Ethernet, are examined. More specifically, there is a focus on the application-to-application delay and jitter characteristics of such networks, when using Internet protocols such as UDP and TCP. It is demonstrated how important it is to take control of the latency in the station nodes, since the main communication delays occur inside the nodes, and different solutions are presented for controlling these delays. In particular, a priority-based protocol stack is assessed. The results show that real-time, Ethernet-based IP communication is now adequate even for demanding automated applications. In this paper, substation automation (power distribution) is used as an example of a demanding automation system.     

Throughput Analysis and Measurements in IEEE 802.11 WLANs with TCP and UDP Traffic Flows

There is a vast literature on the throughput analysis of the IEEE 802.11 media access control (MAC) protocol. However, very little has been done on investigating the interplay between the collision avoidance mechanisms of the 802.11 MAC protocol and the dynamics of upper layer transport protocols. In this paper, we tackle this issue from an analytical, simulative, and experimental perspective. Specifically, we develop Markov chain models to compute the distribution of the number of active stations in an 802.11 wireless local area network (WLAN) when long-lived transmission control protocol (TCP) connections compete with finite-load user datagram protocol (UDP) flows. By embedding these distributions in the MAC protocol modeling, we derive approximate but accurate expressions of the TCP and UDP throughput. We validate the model accuracy through performance tests carried out in a real WLAN for a wide range of configurations. Our analytical model and the supporting experimental outcomes show that 1) the total TCP throughput is basically independent of the number of open TCP connections and the aggregate TCP traffic can be equivalently modeled as two saturated flows; and 2) in the saturated regime, n UDP flows obtain about n times the aggregate throughput achieved by the TCP flows, which is independent of the overall number of persistent TCP connections.     

基于UDP协议的设备健康管理系统

分析了设备健康管理的重要性以及其的设计与实现,介绍了系统的框架、功能模块设计和UDP协议,以及如何应用UDP协议设计和采用设备健康管理系统,实现系统对实时状态数据的存储与分析、故障预警、寿命预测功能。     

Fast Handoff in SIP-Based Next Generation Mobile Networks

It is commonly held that next generation mobile systems will be developed on the Internet in combination with diverse access technologies, as the future network architecture will be the coming together of various overlapping wireless access networks. Integrating various wireless networks in future heterogeneous networking environments poses many difficulties, the most critical challenge of which is efficient support for seamless mobility. SIP is a promising nominee for managing mobility in heterogeneous networks as it provides mobility within the application layer and the characteristics of the lower layer protocols are invisible to it. However, the performance of SIP-based mobility management is downgraded, resulting from its adoption of TCP/UDP for signaling and its strict separation between the lower layers and the application layer of the protocol stack. In this paper, a SIP-based cross-layer design for fast handoffs is proposed to shorten the service interruption time when a mobile node crosses the overlapped area of a WLAN/3G cellular system. As will be shown by the simulation results, the SIP-based solution proposed in this paper effectively lessens the handoff delays caused by either the horizontal handoff or vertical handoff in future all-IP heterogeneous wireless networks.     

End-to-End Performance Improvement of the Wireless Channel for Video Transmission

Real time video transmission in wireless environment considers various parameters of wireless channel like information rate, error resiliency, security, end-to-end latency, quality of service etc. The available internet protocols are transmission control protocol and user datagram protocol (UDP). But most of the real-time applications uses UDP as their transport protocol. UDP is a fast protocol suitable for delay sensitive applications like video and audio transmission as it does not provide flow control or error recovery and does not require connection management. Due to the tremendous growth in wired and wireless real-time applications, some improvements should be made in the existing systems or protocols. Various techniques to improve end-to-end performance of system for real time video transmission over wireless channel are available in literature. Authors claim that the solution suggested in the paper provide more reliability in wireless video transmission. In the proposed solution, adaptive redundant packets are added in every block (or datagram) transmitted in order to achieve a desired recovery rate at the receiver. The suggested method dose not use any retransmission mechanism. The network simulator NS-2 is used to evaluate the method and the simulation results indicate that the proposed method can guarantee satisfied end-to-end performance by increased packet delivery ratio, reduced end-to-end delay and hence increased network throughput for video transmission in wireless network.     

Optimization of SIP Session Setup Delay for VoIP in 3G Wireless Networks

Wireless networks beyond 2G aim at supporting real-time applications such as VoIP. Before a user can start a VoIP session, the end-user terminal has to establish the session using signaling protocols such as H.323 and session initiation protocol (SIP) in order to negotiate media parameters. The time interval to perform the session setup is called the session setup time. It can be affected by the quality of the wireless link, measured in terms of frame error rate (FER), which can result in retransmissions of packets lost and can lengthen the session setup time. Therefore, such protocols should have a session setup time optimized against loss. One way to do so is by choosing the appropriate retransmission timer and the underlying protocols. In this paper, we focus on SIP session setup delay and propose optimizing it using an adaptive retransmission timer. We also evaluate SIP session setup performances with various underlying protocols (transport control protocol (TCP), user datagram protocol (UDP), radio link protocols (RLPs)) as a function of the FER. For 19.2 Kbps channel, the SIP session setup time can be up to 6.12s with UDP and 7s with TCP when the FER is up to 10 percent. The use of RLP (1, 2, 3) and RLP (1, 1, 1, 1, 1, 1) puts the session setup time down to 3.4s under UDP and 4s under TCP for the same FER and the same channel bandwidth. We also compare SIP and H.323 performances using an adaptive retransmission timer: SIP outperforms H.323, especially for a FER higher than 2 percent.