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. Platform: |
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. Platform: |
Size: 68232 |
Author:王刚 |
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Description: NIST Net – A Linux-based Network Emulation Tool, It is a raw IP packet filter with many controllable channel parameters such as packet loss ratio, jitter, bandwidth variation, delay, and network buffer size. To simulate different network environments Platform: |
Size: 2309412 |
Author:北科 |
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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: 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 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 Platform: |
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. Platform: |
Size: 560128 |
Author:raosiyong |
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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 Platform: |
Size: 5120 |
Author:陈艳玲 |
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Description: NIST Net – A Linux-based Network Emulation Tool, It is a raw IP packet filter with many controllable channel parameters such as packet loss ratio, jitter, bandwidth variation, delay, and network buffer size. To simulate different network environments-NIST Net- A Linux-based Network Emulation Tool, It is a raw IP packet filter with many controllable channel parameters such as packet loss ratio, jitter, bandwidth variation, delay, and network buffer size. To simulate different network environments Platform: |
Size: 2309120 |
Author:北科 |
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Description: 当IP语音包的网络时延抖动较小时，一般的语音缓冲算法可以得到较好的语音质量。当网络中存在突发大时延时，就会出现极大丢包率或极大端到端时延，从而难以获得好的语音质量。为此，提出针对突发大时延下的自适应语音缓冲算法。通过估算网络平均时延和学习语音包经过的网络路径上的状态，来确定需要控制端到端时延大小和语音包的丢包率，动态调整Jitter Buffer队列的最小深度和最大深度，从而可以尽量减小语音裂缝（gap）的出现。通过基于听觉模型的客观音质评价（PESQ）仿真计算以及在实际语音网关设备中的应用表明算法对语音通信质量有一定的改善作用。-The continuous playout of voice packets in the presence of variable network delays is often achieved by buffering the received voice packets for sufficient time. Basic jitter buffering algorithms can work well only when the delay does not spike status of the networks, is presented to promote the quality of voice communication. It timely adjusts the minimal and maximal depth of buffer queue according to the control target of end-to-end delay and packet loss rate. The algorithm can much more easily achieve the continuous playout because it plays voice packet at a fixed inter-play time in the most time of a talk-spurt. The control target of packet loss rate can be extended to 20 . However, the basic algorithms can only bear 5-10 of the packet loss rate. Perceptual evaluation of speech quality(PESQ) is applied to assess the speech quality in the simulation. It is shown that the algorithm can obviously promote the quality of voice communication in IP networks with spike delay. The practic Platform: |
Size: 329728 |
Author:瞿志超 |
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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. Platform: |
Size: 3072 |
Author:cn |
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Description: 描述了因特网和IP的主要特征，包括包丢失和时延抖动，并让读者了解数字信号处理器(DSP)和语音编码器在VoIP中所扮演的角色。本书还为读者讲述了如何通过ISDN、xDSL、HFC本地环路或其他途径建立与业务提供商之间的通路，以及目前主要的IP电话协议-Describes the main features of the Internet and IP, including packet loss and delay jitter, and to allow readers to understand the digital signal processor (DSP) and voice coder in the role of VoIP. The book also tells the reader how to ISDN, xDSL, HFC, or other means to establish a local loop between the suppliers and business access, and the current major IP telephony protocol Platform: |
Size: 19512320 |
Author:han |
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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.. Platform: |
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 Platform: |
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． Platform: |
Size: 176128 |
Author:lifei |
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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 Platform: |
Size: 420864 |
Author:冯月白 |
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Description: The new emerging QoS architectures are motivated by the desire to improve the
overall performance of an IP network. Differentiated Services (Diffserv) define a model for
implementing scalable differentiation of QoS in the Internet. Multiprotocol Label
Switching (MPLS) is a fast label-based switching technique that offers new QoS
capabilities for large scale IP networks. When an MPLS network supports DiffServ, traffic
flows can receive class-based network treatment that provides bases for QoS guarantees.
The objective of this work is to study the influence of the QoS mechanism via
DiffServ-MPLS on network parameters such as jitter, delay and throughput. The
comprehensive study showed general improvement in the throughput, jitter and delay
particularly of voice and video transmission when using DiffServ-aware MPLS network as
compared to pure IP only or MPLS only. Platform: |
Size: 444416 |
Author:muhand |
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Search - delay and delay jitter