RTP/UDP协议在无线视频传输中的应用研究

设计了RTP/UDP/IP框架下的传输协议,并应用于CDMA IX的无线视频传输系统中.结合无线信道特点和UDP协议的固有缺陷,提出了利用RTP协议的序列号域进行差错控制的方法,在克服网络丢包和传输乱序方面获得了良好的效果.     

XicP协议和SPC拥塞控制算法

本文描述了基于UDP的抗高随机丢包率传输协议——XicP协议,及其设计与实现。讨论了xicP协议的拥塞控制策略,重点研究了两方面内容：（1）基于SPC的拥塞控制算法;（2）基于丢包序列的重传算法。XicP协议是面向链接的协议,结合了窗口流量控制算法和改善的AIMD速率控制算法,支持高随机丢包率广域网的快速数据传榆,极大地提高了数据吞吐量。     

基于多协议的人工影响天气中转服务器

为了满足人工影响天气不同作业决策时段需求,设计了一种基于混合TCP - UDP协议的人工影响天气中转服务器.连接双方(指挥端和客户端)的指令通讯协议采用混合TCPUDP算法的传输协议进行数据连接.该机制使用UDP作为短对话时的传输层协议,而对于大量数据需要传输时则使用TCP作为传输层的协议.这样,对于短对话可以避免TCP的额外开销,而对于长对话又可以得到由TCP提供的可靠传输和拥塞控制.     

基于无线局域网的实时视频传输与控制

针对无线局域网,研究实现了一种基于RTP/UDP/IP协议的端到端H.263实时视频传输系统.针对无线QoS机制系统应用了一种RTCP反馈友好码率控制算法TFRC,根据丢包率来动态调整发送码率实现拥塞控制.实验表明该传输方案能获得较好的视频传输质量.     

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.     

变电站实时监控系统中的千兆以太网传输设计

针对变电站实时监控系统需要传输的数据量大和传统的百兆网络已无法满足传输总带宽需求的问题,文中设计了嵌入式千兆以太网视频传输系统。该系统选用Xilinx公司Artix-7系列的FPGA芯片为主控制器,完成数据的采集、处理和传输控制。使用UDP/IP协议帧通过千兆以太网向远程监控系统传输图像数据时,丢包率仅为0.001 25%。在对变电站高压设备进行监控传输测试时,稳定传输视频图像的最高帧频可达150 frame·s-1。     

UDP协议改进及其在嵌入式核辐射监测系统中的应用

文章通过单播与组播相结合、禁用UDP数据校验、发送端函数设置、消息应答以及超时和重传等机制,对传统UDP协议进行了实时性改进和可靠性控制.改进的UDP协议在核电站嵌入式辐射监测系统中的成功应用证明了文章提出方案的可行性和有效性.     

基于无线网络数据广播算法的多媒体信息传输技术与应用

随着无线网络的迅猛发展,传统声音、文字通讯模式已发生巨大的改变,本文以无线网络数据广播算法为基础,对多媒体信息传输技术进行了研究.通过分析TFB算法、节点选择、转发节点选择、相近转发节点的消除,建立了基于无线网络数据TFB三点转发广播算法的多媒体交互系统构架,采用UDP+ CDN传输协议中转模式,在Windows XP上,采用TFB三点转发广播算法,进行无线移动网络多媒体发包与丢包的测试,通过CDN服务器,连续发5 000个包,实验结果表明,两次发包间不延时,无线网络未出现丢包现象,丢包现象主要发生在发送节点侧与接受节点侧,丢包范围在0.3％ ～2.3％之间,这对用户体验不会造成影响.     

基于前向纠错的自适应低延时传输机制

针对无线网络中视频传输的延时以及用户数据报协议（User Datagram Protocol,UDP）因丢包造成的视频质量大幅下降问题,提出一种基于前向纠错的自适应低延时（Adaptive Low-Latency Forward Error Correction,AL-FEC）传输机制。该机制通过前向纠错方案避免在使用UDP协议传输过程中因丢包而造成的视频失真卡顿现象,通过发包策略及自适应冗余度达到降低端到端时延并提高视频传输质量的目的。根据当前的网络状况以及视频帧特性确定需要发送的源包个数,利用接收包接收到的数据包信息实时估计网络状况,从而根据网络波动情况动态更新冗余包比例。大量实验结果表明,相较于静态的前向纠错机制,AL-FEC能够使得平均端到端时延降低40～50 ms,在高丢包率情况下,视频的峰值信噪比（Peak Signal to Noise Ratio,PSNR）提高5～8 dB。     

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。     

基于TCP、UDP的多路多数据流融合网络系统设计

为节省网络资源,充分利用TCP与UDP两种网络传输协议各自的优势,设计了TCP、UDP的多路多数据流融合网络系统。该系统在数据采集端的多端复杂网路中采用TCP网络协议传输数据,保证了网络传输的可靠性,同时在数据转发端的点对点网络中采用UDP网络协议传输数据,节省了网络资源,解决了网络拥塞问题。该系统在多路TCP数据流融合过程中应用了队列(Queue)融合算法,解决了数据流不同步导致的数据重复转发及丢失问题。经过长时间运行测试,整个系统在网络传输速率、稳定性及可靠性方面都达到了设计要求。     

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。     

Considerations for application‐layer multipath transport control

Multipath transport faces a lot of challenges caused by path diversity, network dynamics, and service diversity. An effective end‐to‐end multipath transport control mechanism becomes essential to efficiently utilize multiple paths. On the base of the general framework of multipath transport system based on application‐level relay proposed in our previous work, this paper presents a multipath transport control mechanism supporting various applications with different transmission requirements. We propose a multipath transport protocol suite, which is extensible and suitable for various applications, and a multipath transport control model in which an application‐dependent splitting granularity named flow block is introduced. Two load distribution models are explored: the earliest idle path first load distribution for reliable data transmission to maximize the data throughput and the packet reordering‐controlled load distribution for real‐time data transmission to minimize the packet reordering thereby reducing end‐to‐end delay and packet loss rate of multipath transport. Simulation results show that the proposed models can effectively improve data throughput for applications with reliable transmission requirements and reduce the total packet loss rate of the destination for applications with real‐time transmission requirements.     

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.     

FJADA: Friendship Based JellyFish Attack Detection Algorithm for Mobile Ad Hoc Networks

TCP and UDP are considered the most popular and well known transport layer protocols to facilitate the end to end communication between two nodes in the network. TCP is used as the transport layer protocol in packet delivery and error sensitive applications, where packet loss cannot be compromised. However, low-rate TCP targeted Denial of Service (DoS) attacks exploit the retransmission timeout and congestion control features of TCP protocol. These low-rate TCP targeted Denial of Service (DoS) attacks are also called JellyFish (JF) attacks. These attacks perform the malicious activities either by delaying, or periodically dropping or mis-ordering the data packets on the route from source to destination node in the network, and cause severe degradation in end-to-end throughput in the network. JellyFish attack is further classified as JF-Delay Variance Attack, JF-Periodic Drop Attack and JF-Reorder Attack based on the type of the malicious activities being performed. JellyFish attack conforms to all existing routing and packet forwarding protocol specifications, and therefore it becomes very difficult to detect its presence in the network. In this paper, a Friendship Based JellyFish Attack Detection Algorithm (FJADA) is presented for Mobile Ad Hoc Networks, where the basic concept of friendship mechanism is added to the existing Direct Trust-based Detection (DTD) algorithm to save the valuable resources of a node in monitoring the activities of its one hop neighbours, through promiscuous mode. FJADA also minimizes the possibility of overestimating the malicious behaviour of innocent nodes due to radio transmission errors, network congestion or packet collisions. The results obtained throughout the simulation experiments clearly show the feasibility and effectiveness of the proposed detection algorithm.     

基于MPC850的GPRS三峡库区水质监测系统

通用分组无线业务是在GSM网络上开通的一种新型的分组数据传输业务,该业务主要具有＂高速＂和＂永远在线＂的优点,正是由于该种优点,用于实时监测将会取得很好的效果。主要介绍了基于MPC850的通用分组无线业务三峡库区水质监测系统,结合三峡库区的实际情况,给出了系统的总体设计,分析了单片机MPC850对数据的采集、处理和发送的基本原理,并介绍了软件实现方法。由于设计的通用性,该系统模型还可用于水文、气象、农情等方面的监测。     

Cross-layer-based adaptive TCP algorithm for cloud computing services in 4G LTE-A relaying communication

Cloud computing provides various diverse services for users accessing big data through high data rate cellular networks, e.g., LTE-A, IEEE 802.11ac, etc. Although LTE-A supports very high data rate, multi-hop relaying, and cooperative transmission, LTE-A suffers from high interference, path loss, high mobility, etc. Additionally, the accesses of cloud computing services need the transport layer protocols (e.g., TCP, UDP, and streaming) for achieving end-to-end transmissions. Clearly, the transmission QoS is significantly degraded when the big data transmissions are done through the TCP protocol over a high interference LTE-A environment. The issue of providing high data rate and high reliability transmissions in cloud computing needs to be addressed completely. Thus, this paper proposes a cross-layer-based adaptive TCP algorithm to gather the LTE-A network states (e.g., AMC, CQI, relay link state, available bandwidth, etc.), and then feeds the state information back to the TCP sender for accurately executing the network congestion control of TCP. As a result, by using the accurate TCP congestion window (cwnd) under a high interference LTE-A, the number of timeouts and packet losses are significantly decreased. Numerical results demonstrate that the proposed approach outperforms the compared approaches in goodput and fairness, especially in high interference environment. Especially, the goodput of the proposed approach is 139.42 % higher than that of NewReno when the wireless loss increases up to 4 %. Furthermore, the throughput and the response functions are mathematically analyzed. The analysis results can justify the claims of the proposed approach.     

Efficient resource allocation for China's 3G/4G wireless networks

The all-IP DiffServ model is expected to be the most promising architecture for QoS provisioning in China's next-generation wireless networks, due to its scalability, convenience for mobility support, and capability of interworking heterogeneous radio access networks. This article focuses on efficient resource allocation in a wireless DiffServ architecture. Resource utilization efficiency is particularly important for China's wireless networks as the mobile user density in China is and will continue to be much higher than that in other countries. More specifically, we propose a novel buffer sharing scheme to provide assured service for real-time layer-coded multimedia traffic, which can guarantee the specific packet loss requirement of each layer with UDP as the transport layer protocol. An adaptive optimal buffer configuration can be applied to achieve maximum resource utilization over the time-varying channel. Assured service is also provided to TCP data traffic for guaranteed throughput, where the cross-layer coupling between the TCP layer and link layer is exploited to efficiently utilize the wireless resources.     

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.