搜索资源列表
VideoChat
- 用VC实现的一个非常完整的网络会议系统。视频用Mpeg4实现）、语音用G729a实现、传输用RTP协议实现.-With the VC to achieve a very complete web conferencing system. Achieved with the Mpeg4 video), voice use G729a implemented RTP protocol for transmission.
VideoChat
- 基于JMF的即时视频语音模拟实现，通信采用rtp协议，eclipse开发环境-jmf rtp eclipse
rtp g729 g711 协议栈
- rtp g729 g711 协议栈
AXIS-rtsp
- 通过rtp协议实现了从安讯士axis摄像头读取完整的rtsp视频流rtp数据包-Rtp protocol by Axis axis from the camera to read the complete rtp rtsp video stream packets
ortp-0.16.2.tar
- rtp协议栈源码，实现实时传输，传输文本或多媒体流-RTP
rtsp_server
- rtsp server源代码，里面有完整的rtsp rtp rtcp sdp协议源码-rtsp server source code, which has the complete source code protocol rtsp rtp rtcp sdp
live
- 流媒体服务器和客户端的源代码，可以直接构建点播直播系统，基于RTP/RTSP协议栈，实现MP3等多种视频格式-Streaming media server and client source code, can be directly built on-demand broadcast system, based on RTP/RTSP protocol stack to achieve a variety of video formats such as MP3
MorDirectShowFilter
- 这是RTP DirectShow Filters服务器与客户端可执行程序源码与服务器端代码，支持 RTP， RTCP， RTSP and SDP协议．-This is a RTP DirectShow Filters server and the client-side executable program source code with the server-side code, support for RTP, RTCP, RTSP and SDP protocols.
jrtplib-3.7.1
- jrtplib3.7.1版，是RTP（实时传输协议）的开发类库，对多媒体编程有帮助
rtp
- rtp C源码 实时流协议（RTSP）是应用层协议，控制实时数据的传送 -rtp C source Real Time Streaming Protocol (RTSP) is an application layer protocol, to control the transmission of real-time data
sflphone_0.9.1.tar
- 最新版的sflphone，支持SIP协议，exosip2，osip2,RTP。界面用QT写的，有跨平台特性，编译通过。-The latest version of sflphone, support the SIP protocol, exosip2, osip2, RTP. Interface written using QT, there are cross-platform features, the compiler through.
WiresharkOpreatRTPStream
- 使用WireShark分析RTP流，Wireshark 是一个强大的抓包及网络分析软件，可以用来嗅探和分析多种网络协议的数据包和流，RTP 和 RTCP 也是其中的两种。-WireShark analysis using the RTP stream, Wireshark is a powerful Ethereal and network analysis software, can be used to sniff and analyze a variety of network proto
rtpdocument
- rtp实时传输协议，后面有很多源代码，我就是按照上面的算法实现了一个视频采集传输设备。-rtp real-time transport protocol, followed by a lot of source code, I was in accordance with the above algorithm, a video capture transmission equipment.
The_Speex_Codec_Manual_Version_1.2_Beta_3
- Speex是一套开源的专门压缩声音的库，压缩的性能非常高，常用在VoIP或者其它网络程序中。Speex声称自己是不受任何专利限制，并授权根据修订后的BSD许可证发布。它可以用来与Ogg容器格式或直接在UDP / RTP协议下传输。 这份是Speex的编码手册英文版，下面的地址是维基百科中关于Speex的介绍: http://en.wikipedia.org/wiki/Speex-Speex is a free software speech codec that may be used on
e-comm
- 代码来自sourceforge，在linux下的实时语音聊天程序，使用了adpcm编码，同时还使用了RTP实时传输协议，是一个很好的学习实时传输协议的程序。- The code comes from sourceforge, chats the procedure under the linux real-time pronunciation, has used the adpcm code, meanwhile has used the RTP real-time transmissi
JRTPConsole
- 使用RTP协议写的一个传输程序，包括发送端和接收端-RTP written agreement to use a transmission process, including the sending end and receiving end
41695084rtp
- RTP实现的库 有美国一大学生开发 适合学习RTP协议的用-RTP library to achieve the development of the United States suitable for a university to learn to use RTP protocol
rfc3550
- rtp的详细RFC文档，适合英语强的程序员阅读，内容包含协议分析和使用例子-err
live.2008.11.13.tar
- 用C++写的流媒体程序库，实现了标准协议。例如RTP/RTCP,RTSP以及SIP-C write streaming media libraries, which achieved the standard protocols. For example, RTP / RTCP, RTSP and SIP, etc. -With C++ Write streaming media library and achieve a standard protocol. For example, RTP/
Rtsp
- rtsp协议的主要实现代码.对开发流媒体，rtp/rtsp传输，mpeg4等值得参考。-The main achievement of the agreement rtsp code. on the development of streaming media, rtp/rtsp transmission, mpeg4, such as worth considering.