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Description: 如何成功的运用OPENH323 来开发商业的H.323 协议栈
Openh323 有以下的特征：
1>动态的静音检测算法, 以减小语音包的传输量。
2>支持Windows，Linux 和FreeBSD 的多种客户端命令。
3>包含有MCU，PSTN 网关，和自动语音应答机等多种应用平台。
4>软件支持GSM，LPC-10，G.711 uLaw/Alaw 软件编解码方式。
5>特定硬件（Quicknet/LineXJ）支持G.723.1, G.728 和 G.729 。
6>采用Jitter Buffer 技术对语音信号做接受的缓冲处理。
7>软件产生舒适噪音。
8>采用H.261 视频压缩协议。
9>支持广播的方式查找网守（GateKeeper）。
10>支持H.235 附件D 中和网守的身份认证。(部分地支持H.235)
11>支持部分H.450 补充协议。
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Size: 394463 |
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Description: 本文详细讨论了添加到 RTC 的媒体改进特性，这些改进使得最终用户和开发者都能有更愉快的体验。当应用程序被构建在 RTC 客户端 API 之上，最终用户能获得丰富的音视频体验，而开发者可以使程序得到一系列免费的改进。使用这些 API 构建的应用程序还能够访问 RTC 提供的即时消息和出席功能。有关这些API的信息，可在 Windows Platform SDK中获得。
本文讨论了以下的特性和改进之处：
音频视频编解码器的可获得性
回波抵消（AEC）
冗余音频编码
动态抖动缓冲和调整
-discussed in detail in this paper added to the RTC to improve the characteristics of the media, these improvements allow end users and developers will have a more enjoyable experience. When the application has been built on the RTC client API on top, end-users can access the audio-video experience, and the developer can make a series of procedures to improve free. Construction of the use of these API applications can also visit the RTC to provide instant messaging and attend functions. The API information available in the Windows Platform SDK gain. This paper discusses the following features and improvements : the audio and video codec availability echo canceller (AEC) Redundant audio coding dynamic jitter buffer and adjustment
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Size: 5503130 |
Author: 2 |
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Description: 实时数据采集的屏幕抖动显示问题一直困扰着许多人，本程序采用双缓冲的方法解决了上述问题。-screen jitter that the problem has been troubling many people and the adoption of double-buffer solution to the above problems.
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Size: 91506 |
Author: 杨国锋 |
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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
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Size: 2309412 |
Author: 北科 |
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Description: jitter buffer算法的源代码-jitter buffer algorithm source code
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Size: 2194 |
Author: 张浦 |
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Description: * Network Coding Network Simulator
*
* 06/10/05 - Aaron Drew
*
* This simulator is designed to test out the performance of a variety of network coding approaches.
*
* It simulates networks with dynamic topologies, reading from NS2 compatible node mobility files.
* I assume that:
* - wireless nodes are effected by a simple exponentially distributed noise floor.
* - latency = distance x speed of light + fixed processing time + random jitter.
*
* These assumptions can be easily changed in the code below.
*
* The simulator is a simple discrete event-based simulator like NS2, GloMoSim and others but designed specifically to support
* my network coding experiments. It has callbacks for 'transmit buffer empty' and may eventually support 'in air' collisions
* of packets and multiple radio channels. I didn't want to have to build upon the unnecessary MAC layers that exist in the
* other simulators.
*
* The network coding module in rlnc.h implements basic block encoding and decoding as well as block generation by intermediate
* nodes that have received partial data.
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Size: 12023 |
Author: yeyueyushen@163.com |
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Description: 如何成功的运用OPENH323 来开发商业的H.323 协议栈 Openh323 有以下的特征： 1>动态的静音检测算法, 以减小语音包的传输量。 2>支持Windows，Linux 和FreeBSD 的多种客户端命令。 3>包含有MCU，PSTN 网关，和自动语音应答机等多种应用平台。 4>软件支持GSM，LPC-10，G.711 uLaw/Alaw 软件编解码方式。 5>特定硬件（Quicknet/LineXJ）支持G.723.1, G.728 和 G.729 。 6>采用Jitter Buffer 技术对语音信号做接受的缓冲处理。 7>软件产生舒适噪音。 8>采用H.261 视频压缩协议。 9>支持广播的方式查找网守（GateKeeper）。 10>支持H.235 附件D 中和网守的身份认证。(部分地支持H.235) 11>支持部分H.450 补充协议。-How successful the use of OpenH323 to develop commercial H.323
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Size: 394240 |
Author: 陈学军 |
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Description: 实时数据采集的屏幕抖动显示问题一直困扰着许多人，本程序采用双缓冲的方法解决了上述问题。-screen jitter that the problem has been troubling many people and the adoption of double-buffer solution to the above problems.
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Size: 91136 |
Author: |
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Description: jitter buffer算法的源代码-jitter buffer algorithm source code
Platform: |
Size: 2048 |
Author: 张浦 |
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Description: 本文详细讨论了添加到 RTC 的媒体改进特性，这些改进使得最终用户和开发者都能有更愉快的体验。当应用程序被构建在 RTC 客户端 API 之上，最终用户能获得丰富的音视频体验，而开发者可以使程序得到一系列免费的改进。使用这些 API 构建的应用程序还能够访问 RTC 提供的即时消息和出席功能。有关这些API的信息，可在 Windows Platform SDK中获得。
本文讨论了以下的特性和改进之处：
音频视频编解码器的可获得性
回波抵消（AEC）
冗余音频编码
动态抖动缓冲和调整
-discussed in detail in this paper added to the RTC to improve the characteristics of the media, these improvements allow end users and developers will have a more enjoyable experience. When the application has been built on the RTC client API on top, end-users can access the audio-video experience, and the developer can make a series of procedures to improve free. Construction of the use of these API applications can also visit the RTC to provide instant messaging and attend functions. The API information available in the Windows Platform SDK gain. This paper discusses the following features and improvements : the audio and video codec availability echo canceller (AEC) Redundant audio coding dynamic jitter buffer and adjustment
Platform: |
Size: 5502976 |
Author: |
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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
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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
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Size: 329728 |
Author: 瞿志超 |
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Description: VoIP系统的关键技术语音抖动缓冲（Jitter Buffer）的详细资料。-Detailed information of VoIP system s key technology jitter buffer.
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Size: 4625408 |
Author: Dayspring |
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Description: jitterbuffer实现网络数据流的抖动，可以有效解决网络带宽波动的时候音视频数据的丢包现象-jitterbuffer network data stream to achieve jitter, can be an effective solution to network bandwidth fluctuations, when the phenomenon of audio and video data packet loss
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Size: 9250816 |
Author: KDM |
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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: 关于VOIP接受端中消除抖动的一些文档，其中也包含ITU-T的协议 -jitter buffer
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Size: 3499008 |
Author: 高维 |
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Description: Adaptive jitter buffer for Speex.
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Size: 7168 |
Author: joucanyun |
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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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Description: 因特网下，流媒体视频连续播放算法论文,动态抖动缓冲-stream meida jitter buffer
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Size: 247808 |
Author: 冯月白 |
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