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jrtplib-3.7.1
- 推荐做流媒体传输或是视频会议等系统开发人员下载。 实时传输协议RTP（Realtime Transport Protocol）：是针对Internet上多媒体数据流的一个传输协议，由IETF(Internet工程任务组)作为RFC1889发布。RTP被定义为在一对一或一对多的传输情况下工作，其目的是提供时间信息和实现流同步。-JRTPLIB is an object-oriented RTP library written in C++. It was first developed
linuxmobileRTP
- linux 手持嵌入式视频研发 硬件平台 linux平台 编解码资料来源 移植过程 应用过程 都很详细 难的技术文章 pdf格式 但文字复制无加密-linux mobile RTP
2002-06-18-Video-Services-Forum
- rtp:multimedia streaming over ip-rtp: multimedia streaming over ip
voipong-2.0
- 基于流量模型的检测，支持SIP, H323, Cisco s Skinny Client Protocol, RTP and RTCP.- VoIPong is a utility which detects all Voice Over IP calls on a pipeline, and for those which are G711 encoded, dumps actual conversation to seperate wave files. It sup
MSS_v1_0
- RTP开发包3，用于DIRETCTSHOW的开发，很好用的，希望下载-RTP Development Kit 3, for the development of DIRETCTSHOW good use, and want to download
1
- 2008年硕士论文，基于H.264熵编码的视频加密技术研究-Master' s thesis in 2008, based on entropy coding H.264 video encryption technology research
405797jrtplib-2.9
- jrtplib RTP协议栈的实现 RTP库 网络视频会议实时传输协议 实时传输-jrtplib RTP protocol stack RTP library to achieve real-time video conferencing network transport protocol real-time transmission
gsm
- 这个网络电话程序是linux下，用C语言实现的。它既不是实现的H.323 或 SIP协议, 也没有使用RTP协议，更没有使用到任何其它第三方软件，不过，它确实工作的很好。通话话音质量非常不错。-The network telephone program is under linux, using the C language. It is not the realization of the H.323 or SIP protocol, but also did not use the RTP
RFC3550
- RFC3550是关于rtp的详细介绍，是网络电话开发的工具。-RFC3550 is a detailed descr iption on the rtp is a tool for the development of Internet telephony.
SS7GWPD
- The function of an SS7 GW is to allow interconnection between the signalling of the PSTN, based on SS7, and the signalling systems of the VoIP networks – either SIP or H.323. Additionally, the SS7 GW must provide connection control for interworki
111
- 基于RTP协议的音频传输技术的研究与实现-RTP-based audio transmission technology agreements Research and Implementation of
rtpfiletransfer
- rtpfiletransfer 使用udp封装rtp的实现文件传输协议，效率不高 但大家可以-rtpfiletransfer
RFC3550
- 中文版RTP协议的具体内容，根据原英文翻译，整理成了word-RTP Chinese version of the specific content of the agreement, in accordance with the original English translation, consolidation has become a word
live
- live media rtsp/rtp/-rtsp
live.2009.04.20.tar
- live555是一个用C++编写的、基于开放标准协议RTP/RTCP, RTSP, SIP的多媒体流化源码库。用该库可以构建多种协议的多媒体应用程序，比如VLC，MPlayer, LIVE555 Media Server,vobStreamer等，它也可以接收，发送MPEG, H.263+ or JPEG格式的多媒体视频和多种音频文件，包中针对各种应用都有测试源码，为开发者提供了了解和应用该库的便捷途径。--live555 C is a preparation that is based on
jm50h-suehring
- h.264 New features * Sequence and Picture Parameter Sets * writes Annex B bytestream and RTP packets
rtpflood
- 对SIP SERVER进行rtp flood攻击的小程序-rtp flood to SIP SERVER
SoftPhone
- 点对点语音聊天 联系人管理 通话记录管理 使用SIP协议，媒体协议使用RTP协议-softphone