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Description: VoIP(voice over IP) 就是通过IP 网络承载语音业务,也称IP 网络电话。当网络出现拥塞或传输差错时,语音包就会产生时延、抖动甚至丢失,导致语音不连续或中断,-VoIP (voice over IP) through the IP network to carry voice traffic. also known as IP telephone network. When the network is expected to congestion or transmission errors, the packet will have a voice delay, jitter or loss, voice not lead to continuous or interrupted,
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Size: 4808 |
Author: 张路宜 |
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Description: rtp的c++库。rtp是VoIP等IP多媒体传输协议，是处理网络延时、抖动、丢包的关键模块。-rtp the c library. Rtp of VoIP and other IP multimedia transmission protocol, network delay, jitter, packet loss of key modules.
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Size: 560441 |
Author: raosiyong |
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Description: iLBC 产生背景
在VoIP的应用中，大部分厂商采用CELP (Code Excited Linear Prediction) 算法的低速率语音编解码，如ITU G.729和G.723.1等。而VoIP应用主要在包交换的IP网络上进行传输，无法避免IP网络的丢包、延时、抖动等实时传输问题，而传统的这几个CELP算法对高丢包的处理不是很好，因而很大程度上会影响语音通话效果。
-iLBC background in the application of VoIP, Most manufacturers to use CELP (Code Excited Linear Prediction) Algorithm low-rate voice codecs, such as G.729 and ITU G.723.1, etc.. And the main application of VoIP in the IP packet-switched network for transmission, could not avoid IP network packet loss, delay, jitter and other real-time transmission, and a few traditional CELP algorithm to handle the high packet loss is not very good, which will largely affect voice calls effect.
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Size: 68232 |
Author: 王刚 |
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Description: jitterbug是基于matlab的工具箱，允许对在不同的时域条件下的线性系统的二次性能指标计算。用这个工具箱，可以很容易看出系统对时延、jitter和数据丢失等的响应。-jitterbug is based on Matlab Toolbox, allowing for the different time domain under the conditions of the linear quadratic system performance computing. Using this toolkit, you can easily see that system to delay, jitter and data loss, such as the response.
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Size: 57082 |
Author: 张婷 |
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Description: jitterbug是基于matlab的工具箱，允许对在不同的时域条件下的线性系统的二次性能指标计算。用这个工具箱，可以很容易看出系统对时延、jitter和数据丢失等的响应。-jitterbug is based on Matlab Toolbox, allowing for the different time domain under the conditions of the linear quadratic system performance computing. Using this toolkit, you can easily see that system to delay, jitter and data loss, such as the response.
Platform: |
Size: 56320 |
Author: 张婷 |
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Description: VoIP(voice over IP) 就是通过IP 网络承载语音业务,也称IP 网络电话。当网络出现拥塞或传输差错时,语音包就会产生时延、抖动甚至丢失,导致语音不连续或中断,-VoIP (voice over IP) through the IP network to carry voice traffic. also known as IP telephone network. When the network is expected to congestion or transmission errors, the packet will have a voice delay, jitter or loss, voice not lead to continuous or interrupted,
Platform: |
Size: 4096 |
Author: 张路宜 |
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Description: VoIP 业务有着严格的实时性要求,时延、抖动和丢包这3 个影响VoIP 服务质量的主要因素与承载网的性能密切相关-VoIP has strict real-time requirements, delay, This jitter and packet loss three impact the quality of VoIP services and the main factors bearing net closely related to the properties
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Size: 63488 |
Author: 张路宜 |
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Description: rtp的c++库。rtp是VoIP等IP多媒体传输协议，是处理网络延时、抖动、丢包的关键模块。-rtp the c library. Rtp of VoIP and other IP multimedia transmission protocol, network delay, jitter, packet loss of key modules.
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Size: 560128 |
Author: raosiyong |
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Description: ADS7846/7843芯片 适合用在4线制触摸屏，它通过标准SPI协议和CPU通信，操作简单，精度高，当触摸屏被按下时（即有触摸事件发生）则ADS7846向CPU发中断请求，CPU接到请求后，应延时一下再响应其请求，目的是为了消除抖动使得采样更准确-ADS7846/7843 chip suitable for use in the 4-wire Touchscreen, SPI through standard protocols and CPU communication, simple operation, high accuracy, When the touch screen is pressed (that is, touching incident) ADS7846 to the CPU interrupt request, CPU upon request, should be about responding to delay its request, the purpose of eliminating jitter make more accurate sampling
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Size: 4096 |
Author: YMCHUNG |
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Description: iLBC 产生背景
在VoIP的应用中，大部分厂商采用CELP (Code Excited Linear Prediction) 算法的低速率语音编解码，如ITU G.729和G.723.1等。而VoIP应用主要在包交换的IP网络上进行传输，无法避免IP网络的丢包、延时、抖动等实时传输问题，而传统的这几个CELP算法对高丢包的处理不是很好，因而很大程度上会影响语音通话效果。
-iLBC background in the application of VoIP, Most manufacturers to use CELP (Code Excited Linear Prediction) Algorithm low-rate voice codecs, such as G.729 and ITU G.723.1, etc.. And the main application of VoIP in the IP packet-switched network for transmission, could not avoid IP network packet loss, delay, jitter and other real-time transmission, and a few traditional CELP algorithm to handle the high packet loss is not very good, which will largely affect voice calls effect.
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Size: 67584 |
Author: 王刚 |
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Description: 使用遗传算法实现交换机的分组调度，从而可以提高交换效率，减小时延抖动以及时延-The use of genetic algorithm for packet scheduling switch, which can improve the exchange efficiency, reducing the delay jitter and delay
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Size: 5120 |
Author: 陈艳玲 |
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Description: ns2下仿真有线网络的延时、抖动、吞吐量的实验源码-ns2 simulation under the cable network delay, jitter, throughput experimental source
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Size: 359424 |
Author: 蔡凌 |
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Description: 有线网络NS2模拟时,线路的抖动计算,awk源码-NS2 simulation of a cable network, the line jitter calculation, awk source
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Size: 1024 |
Author: shan |
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Description: 在网络仿真NS中对延时分析，丢包率分析，延时抖动分析，网络吞吐量分析等给出了可行性算法，直接就能用，对学NS2的人很有用的-Network simulation in NS for latency analysis, analysis of packet loss rate, delay jitter analysis, network throughput analysis give a feasibility algorithm, can be used directly on people who learn NS2 very useful
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Size: 3072 |
Author: 林华 |
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Description: 使用ns2网络仿真软件时，统计无线场景吞吐量、延时、抖动和丢包率的通用脚本文件。-Using the ns2 network simulation software, statistical radio scene throughput, delay, jitter and packet loss rate of the general-purpose script file.
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Size: 3072 |
Author: cn |
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Description: collection of awk script use to measure and calculate end-to-end delay, average delay, jitter, throughput-collection of awk script use to measure and calculate end-to-end delay, average delay, jitter, throughput..
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Size: 6144 |
Author: ismail |
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Description: 用TCP 和UDPprotocol来传输数据，具体请见英文描述-Introduction
In this assignment, you will build a client for a simple streaming transport protocol.
Media streams such as compressed video or audio are typically delay and jitter sensitive-
real-time conversations require 100 ms or less round-trip delay and human ear is very
sensitive to irregular sampling in audio. The long delay imposed by retransmission makes
Transmission Control Protocol (TCP) an unlikely candidate to carry media streams.
Fortunately, with proper error concealment, human perception is not very sensitive to
data loss in video and audio. Thus, User Datagram Protocol (UDP) is commonly
employed to transport media streams. However, there are many problems with UDP-
delay jitter, out-of-order arrival and packet loss. A commonly used technique is to buffer
up some packets to obtain a smoother play-back in the expense of some small delay. An
example is given by the following diagram:
As packet arrives from the network, the stream transport layer will delay the
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Size: 3072 |
Author: Mengmei Liu |
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Description: 在状态转移规则的基础上，局部更新规则和全局更新的蚁群算法的规则，我们提出一个IM证明蚂蚁的最低成本的服务质量蚁群算法（QoS）的路径用于解决网络路由问题与延迟以及背包约束中进行模拟。约52.33％蚂蚁找到成功的QoS路由。并收敛到最佳。-based on the state transition rule，the local updating rule and the global updating rule of ant colony algorithm ，we propose an im proved ant colony algorithm of the least cost quality of service(QoS)unicast routing．The algorithm is used for solving the routing problem with delay．delay jitter．bandwidth，and packet lOSS constrained． In the simulation，about 52．33 ants find the Successful QoS routing ．and converge to the best． It is proved that the algorithm is efficient and effective．
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Size: 176128 |
Author: lifei |
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Description: 这是一个关于NS AWK的分析文件，包括一些参数的分析，例如丢包、延迟、抖动。-This is an analysis of documents on the NS AWK, including the analysis of some parameters, such as packet loss, delay, jitter.
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Size: 2048 |
Author: 肖文泽 |
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Description: 关于抖动缓冲区 jitterbufer的一篇论文-Characteristics of network delay and delay jitter and its effect on voice over IP
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Size: 420864 |
Author: 冯月白 |
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