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Description: internet的tcp/ip协议客户端的实现源码，包括rtp,rtcp,rtsp-the internet tcp / ip client agreements to achieve the source, including rtp. rtcp, rtsp
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Size: 3043 |
Author: apple_spring_hu |
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Description: internet的tcp/ip协议的服务器端实现源码，包括rtp,rtcp,rtsp协议的数据封装实现代码-the internet tcp / ip agreement server achieving source, including rtp. rtcp, rtsp agreement data encapsulation code
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Size: 829213 |
Author: apple_spring_hu |
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Description: G726局域网语音通话程序和源代码
这是使用G726语音压缩（16kbps）和RTP进行传输的程序，因为我没有带WIFI的PPC，所以每个程序都是单独测试的，PC端和PPC端分别都工作正常。
G726编解码算法来自OpenH323.传输使用的RTP可以在RTP程序中找到讲解，这个程序主要是G726的函数。将整个 G726封装为g726_Encode和g726_Decode两个函数，参数为压缩和解压数据存储的地址指针，可以将960字节压缩到120字节和将 120字节解压为960字节。这里G726使用的时候，音频设置为8kHz，16位量化，单声道。
使用方法很简单，只用两端各自输入对方的IP，然后按下“开始对话”，就可以进行语音通信了。
PPC端的运行比较稳定，已经进行了自收自发近一个小时连续工作的测试，非常稳定，话音清晰。-G726 LAN voice calls procedures and the source code is the use of voice compression G726 (16kbps ) and RTP for the transmission process, because I did not bring WIFI the PPC, there are separate procedures for each test, PC - and PPC-were all normal work. G726 codec algorithm from the OpenH323. RTP transmission can be used in the process to find RTP stresses Xie, this procedure is mainly a function of G726. Packaging whole G726 for g726_Encode and g726_Decode two functions, Parameters for the compression and decompression of data storage address pointer, 960 bytes can be compressed to 120 to 120 bytes and bytes to 960 bytes decompression. G726 used here, Audio set to 8kHz, 16 quantify mono. It is very simple to use, with only two ends of the respective input each other's IP, then press
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Size: 708753 |
Author: Xia Tao |
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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: G726局域网语音通话源代码
这是使用G726语音压缩（16kbps）和RTP进行传输的程序，使用方法很简单，因为没多少时间，并且RTP不面向连接，所以我也没做连接确认的，只用两端各自输入对方的IP，然后按下“开始对话”，就可以进行语音通信了。-G726 LAN voice calls source code is the use of voice compression G726 (16kbps) and R TP for transmission, the use of approach is very simple, because no amount of time, and no connection-oriented RTP, Therefore, I do not recognize the link, only two ends of the respective input each other's IP, then press the "start dialogue" it can implement a voice communication.
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Size: 535552 |
Author: tutu11911 |
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Description: IP to Serial port Pass-Through Server.zip
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Size: 76800 |
Author: jact |
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Description: 这是VOIP的rtp协议源文件，写代码时需要参考-This is the source document rtp agreement, the need to write code reference
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Size: 199680 |
Author: wanghg |
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Description: 一个不错的实时通讯开发开源代码，可用于ip电话/视频会议，希望对大家有用，-A good real-time communication to develop open-source code, can be used for ip phone/video conferencing, in the hope that useful to everybody,
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Size: 12288 |
Author: 付鹰 |
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Description: voip语音技术，本书描述了因特网和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。本书的覆盖范围包括：VoIP的全面解决方案；VoIP网关和网闸的作用；7号信令(SS7)和IP、H.323的网间互通；支持VoIP组播的协议(IGMP和MBONE)，带宽预留协议(RSVP、RTP、RTCP)及安全业务。本书是一本中、高级教科书，无论你是在对VoIP技术进行评估还是正在使用VoIP技术，本书都可以将你所需要深入理解的信息传送给你，就如一位世界级的专家在你的身边。
-err
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Size: 11891712 |
Author: 俄俄 |
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Description: 本书描述了因特网和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。本书的覆盖范围包括：VoIP的全面解决方案；VoIP网关和网闸的作用；7号信令(SS7)和IP、H.323的网间互通；支持VoIP组播的协议(IGMP和MBONE)，带宽预留协议(RSVP、RTP、RTCP)及安全业务。本书是一本中、高级教科书，无论你是在对VoIP技术进行评估还是正在使用VoIP技术，本书都可以将你所需要深入理解的信息传送给你，就如一位世界级的专家在你的身边。
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Size: 15643648 |
Author: 李步遴 |
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Description: 可用于做IP电话使用.我在VC下使用可行-Can be used to make IP phone. I use VC feasible
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Size: 489472 |
Author: 成龙 |
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Description: 本文采用了递进的方式，先介绍了IP网络视频监控系统的组成及其关键技术，接着阐述了MPEG-4视频流的RTP分组净荷格式。最后，在视频流的RTP传输中，着重分析了MPEG-4视频流的封装格式，并给出相应的实现方法。
-In this paper, a progressive manner, first introduce the IP network video surveillance system and its key technologies, and then expounded on MPEG-4 video streaming of RTP packet payload format. Finally, in the RTP video stream transmission, focusing on analysis of the MPEG-4 video streaming package format, and gives the corresponding method.
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Size: 257024 |
Author: zengzhen |
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Description: 基于IP的H.264关键技术：介绍了 IP网络应用环境，H.264的错误恢复工具；介绍如何在IP网络使用RTP分组传输H.264的视频技术和RTP载荷格式草案。-IP-based H.264 key technologies: introduction of the IP network application environments, H.264 error recovery tools introduce how to use IP network RTP packet transmission H.264 video technology and the draft RTP payload format.
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Size: 11264 |
Author: menmen |
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Description: The board is opened with FRAMERS (E1), even though this example
* does not have PSTN Users, only IP Users.
* * The board is opened and controlled via the PCI interface.
* * Channels 0, 1 are opened (Trunk 0, BChannels 1, 2) and
* activated through RTP. RTP packets are sent by the second
* channel and received by the first channel (using the internal
* IP-loopback option, or using a loopback connector on the IPM-260
* board s NI connector). -The board is opened with FRAMERS (E1), even though this example
* does not have PSTN Users, only IP Users.
* * The board is opened and controlled via the PCI interface.
* * Channels 0, 1 are opened (Trunk 0, BChannels 1, 2) and
* activated through RTP. RTP packets are sent by the second
* channel and received by the first channel (using the internal
* IP-loopback option, or using a loopback connector on the IPM-260
* board s NI connector).
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Size: 8192 |
Author: jefferychang |
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Description: Real-Time Protocol: RTP 是一个打包协议，和UDP（User Datagram Protocol）联合使用,可以用来在使用IP（Internet Protocol）协议的网络上传输实时多媒体数据。-Real-Time Protocol: RTP is a package agreement, and UDP (User Datagram Protocol) joint use, can be used in the use of IP (Internet Protocol) network protocol in real-time multimedia data transmission.
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Size: 61440 |
Author: LL |
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Description: Voice over IP Intercom
Linux 环境下，带有回声消除模块，VC++开发。-The application can:
dial an intercom partner via short-dial buttons
transport your voice over IPv4 with RTP UDP unicast data packets
make a telephone conference
support wideband (16kHz sample frequency) Speex codec
do call diversion
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Size: 405504 |
Author: Liu Yi |
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Description: RTP协议栈的开源源码,主要用于VoIP流媒体相关的开发 eg：音视频终端/服务器端开发-This is the code of the RTP protocol stack(Open Source ),can be used for the IP telecommunication developing
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Size: 334848 |
Author: alphajay |
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Description: Mediastreamer开发手册
Mediastreamer is a library written in C that allows you to create and run audio and video streams. It is designed for any kind of voice over IP applications. It features RTP connectivity, audio codecs (Speex, iLBC, G711, GSM), video codecs (MPEG4, H263, Theora), I/O from soundcards, wav files, webcams, echo-cancelation, conferencing, and various other utilities. It has a modular design that makes it extensible through plugins. This is the media-streaming component of linphone, a GPL SIP video phone.-Mediastreamer is a library written in C that allows you to create and run audio and video streams. It is designed for any kind of voice over IP applications. It features RTP connectivity, audio codecs (Speex, iLBC, G711, GSM), video codecs (MPEG4, H263, Theora), I/O from soundcards, wav files, webcams, echo-cancelation, conferencing, and various other utilities. It has a modular design that makes it extensible through plugins. This is the media-streaming component of linphone, a GPL SIP video phone.
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Size: 209920 |
Author: 於佳健 |
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Description: 获得接收端的IP地址和端口号
创建RTP会话
指定RTP数据接收端
设置RTP会话默认参数
发送流媒体数据
-Obtain the receiver' s IP address and port number specified to create RTP session to set the RTP data receiver parameters RTP session to send streaming media data by default
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Size: 2048 |
Author: 周伟 |
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Description: 采用jrtplib与jthread开发的Rtp网络程序，里面有RtpClient与RtpServer工程，可直接运行,设定IP与端口即可收发数据。
开发背景：主要用于开发视频监控服务器的测试程序。
2012-8-19 upload jackshen-Rtp network program of using jrtplib jthread developed, there are the RtpClient and RtpServer project development background can be run directly: the main test program for the development of video surveillance server. 2012-8-19 upload jackshen
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Size: 14516224 |
Author: jackshen |
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