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Description: 关于JAVA多点对多点的语音传输问题,采用rtp与rtcp实现语音的传输-on JAVA more point-to-multipoint transmission of voice, the problem with RTCP rtp voice transmission
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Size: 6816 |
Author: 田天 |
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Description: 国外牛人的毕业论文，英文版本，实时语音传输相关，其中的两个库文件，jrtplib与jvoiplib已经上传。enjoy -foreign elite s graduate paper in english.related to real time voice transimission.it has two library,jrtplib,jvoiplib.enjoy
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Size: 649216 |
Author: 杨静文 |
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Description: 从麦克风采集语音数据，直接发送出去，没有RTP部分，做为测试用例不错。-collect data from microphone , and send out directly , have no RTP section . it is useful to be test case
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Size: 1479680 |
Author: 魏伟 |
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Description: 这是一个将语音打包程序, 将数据打包成RTP/RTCP-This is a package of voice procedures, data packing into RTP/RTCP
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Size: 349184 |
Author: 邓来 |
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Description: JMF implement the multimedia communiacation, it will endecoode the voice input to rtp package and send through the network.-JMF implement the multimedia communiacat ion. it will endecoode the voice input to rtp package and send through the network.
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Size: 4803584 |
Author: 杨华 |
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Description: 关于JAVA多点对多点的语音传输问题,采用rtp与rtcp实现语音的传输-on JAVA more point-to-multipoint transmission of voice, the problem with RTCP rtp voice transmission
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Size: 6144 |
Author: 田天 |
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Description: 这是一个外国人写的语音聊天代码,其中使用了RTP协议-This is a foreigner's voice chat write code, which used the RTP
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Size: 759808 |
Author: 欧建林 |
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Description: 视频会议系统，使用RTP协议。可以实现多人视频语音会议-video conferencing systems, the use of RTP. People can achieve video voice conferencing
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Size: 280576 |
Author: weibaby168 |
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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: 1002496 |
Author: Xia Tao |
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Description: G726局域网语音通话源代码(vc++ 6)
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 (vc 6) This the voice compression using G726 (- ns 16-kb ps) and RTP for transmission, the use of approach is very simple, because no amount of time, RTP and not connection-oriented, so I do not recognize the link, only two ends of the respective input each other's IP, then press the "start a dialogue", it can implement a voice communication.
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Size: 577536 |
Author: Arnold |
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Description: 一个可以在局域网进行视频聊天的源代码，语音压缩采用G729，视频压缩采用H.263，网络传输采用RTP。本程序介绍了视频聊天的基本技术，稍加改动就可以直接运用于Internet网络。本人主要进行流媒体传输方面的工作，有兴趣者可以和本人联系。email:cxingpeng@gmail.com-a LAN can video chat source code, voice compression using G729. H.263 video compression and network transmission using RTP. The procedures of the basic video chat technology, tamper on the Internet can be directly applied to the network. I mainly carried out with regard to the transmission of streaming media work, are interested in and I can link. Email : cxingpeng@gmail.com
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Size: 564224 |
Author: cxingpeng |
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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: 完整的点对点语音视频程序
采用RTP协议,支持声音、摄像头、简单文字
全部C++源码，编译即可运行-Complete procedures for the use of peer-to-peer voice and video RTP protocol to support voice, camera, simple text all the C++ Source, the compiler can be run
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Size: 222208 |
Author: aeolus |
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Description: 基于RTP的实时语音传输的实现与研究，用来实现网络语音传输-RTP-based real-time voice transmission with the research, used to realize network voice transmission
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Size: 688128 |
Author: 阿强 |
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Description: 声音进行rtp传输的源码。
该源码我也没看明明白白，因为偶不懂。-Voice rtp transmission source. The source code I did not see clearly, because even understand.
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Size: 150528 |
Author: MyWisest |
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Description: Peers is a very minimalistic SIP User-Agent. It s a softphone, written in java, and it has been tested on linux and windows. It allows a user to call from one PC to another on a local area network, using SIP/SDP/RTP with Ulaw encoding for voice. -Peers is a very minimalistic SIP User-Agent. It sa softphone, written in java, and it has been tested on linux and windows. It allows a user to call from one PC to another on a local area network, using SIP/SDP/RTP with Ulaw encoding for voice.
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Size: 2051072 |
Author: spark |
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Description: 基于java语言的rtp协议源码，基于java语言的rtp协议源码-Based on the java language source rtp agreement, based on the java language rtp protocol source
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Size: 1067008 |
Author: zr |
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Description: 可以將聲音由麥克風取得壓縮後用RTP送出。可以使用PCM GSM。使用簡單方便，也可以在裡面加上加密解密。-Voice can be compressed by a microphone made out with RTP. Can use the PCM GSM. The use of simple and convenient, can also be combined with encryption and decryption on the inside.
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Size: 1189888 |
Author: dearw |
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Description: RTP voice transmission with real time protocol of jmf
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Size: 7168 |
Author: abbes |
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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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