[IP电话/视频会议] Hardware-Zoom-Controller
这是一个9键的机械键盘，我黑客固件，使它可以与Zoom作为硬件控制器。代码库已构建...，
This is a 9-key mechanical keyboard that I hacked the firmware so it can work with Zoom as a hardware controller. The code base is built in C, but there is an accompanying script written in node that when deployed, tracks the mute state of the conference and updates the color of the keyboard buttons accordingly, if the mic is muted, or if video ... (2022-04-07, JavaScript, 0KB, 下载0次)
[IP电话/视频会议] viewports
“视口”是一个浏览器扩展，可以在一个窗口屏幕中打开多个视口（Web）。这可以帮助用户享受更多的网络（视频...，
"viewports" is a browser extension which could open multi-viewports(webs) in one window screen. This could help user enjoy more webs(videos) at one time, browse the Internet efficiently , especially useful in Conference/Hall Super screen (2018-10-03, JavaScript, 0KB, 下载0次)
[IP电话/视频会议] mozo
Mozo是一款实时视频会议和聊天应用程序，允许您与...中的其他人共享网络摄像头、扬声器和屏幕...，
Mozo is a real-time video conferencing and chat application that allows you to share your webcam, speaker and screen with other people in the chat room. (2021-10-20, JavaScript, 0KB, 下载0次)
[IP电话/视频会议] webrtc_video_conf
视频会议和消息传递应用程序，使用WebRTC技术进行数据传输，Socket.io用于信令服务器，React用于fron...，
A Video conferencing and messaging app made using WebRTC technology for data transfer, Socket.io for signalling server and React for frontend. (2023-01-06, JavaScript, 0KB, 下载0次)
[IP电话/视频会议] Serverless-Chat-WebRTC
WebRTC视频聊天，没有用于两台计算机之间的视频会议的信号服务器。这是为我在罗格斯大学的研究而创建的...，
WebRTC Video Chat without a signaling server for video conferencing between two computers. This was created for my research at Rutgers University for a teleconferencing platform called the VideoWall. For info on the video wall visit http://videowall.cs.rutgers.edu/ (2017-01-16, JavaScript, 0KB, 下载0次)
[IP电话/视频会议] VideoWall-project-WebRTC
在没有信令服务器的情况下使用WebRTC进行两台计算机之间的视频会议的演示。本演示进一步扩展为视频，
A demo of using WebRTC without a signalling server for video conferencing between two computers. This demo is further expanded into a video, (2016-03-04, JavaScript, 0KB, 下载0次)
[IP电话/视频会议] libg7221
ITU 的G722.1编码器，包含Annex C，是当前复杂度最低的编码器，并且压缩质量非常好，适合高并发voip使用，实测单i5cpu可以支持3000路以上的实时并行处理，采样率16000和32000 码率16000-48000，以800为梯度。
The ITU G722.1 encoder, including Annex C, is currently the least complex of the encoder, and the compression quality is very good, suitable for high concurrent voip use, found a single road i5cpu can support more than 3000 real-time parallel processing, sampling rate 16,000 and 32000 16000-48000 rate to 800 gradient. (2015-09-12, Visual C++, 717KB, 下载42次)
[IP电话/视频会议] pjproject-2.1
开源的SIP协议库，它实现了SIP、SDP、RTP、STUN、TURN和ICE。PJSIP作为基于SIP的一个多媒体通信框架提供了非常清晰的API，以及NAT穿越的功能。在官网下了很久才下载下载了..不知道是不是服务器在外国的问题...
Open source SIP protocol library, which implements the SIP, SDP, RTP, STUN, TURN and ICE. PJSIP as the SIP-based multimedia communications framework provides a very clear API, as well as NAT traversal function. Official website for a long time to download Download .. do not know is not a server problem in foreign countries ... (2013-04-10, Visual C++, 6715KB, 下载28次)
[IP电话/视频会议] IIP_phone_sysP
IP电话会议系统包含服务器端和客户端端源程序，支持语音方面的压缩和解呀处理
IP telephone conference system includes a server-side and client side source support voice zipping it handle (2012-08-20, Visual C++, 997KB, 下载11次)
[IP电话/视频会议] crbt
品视视频电话会议网关。发送视频会议有关的消息到品视视频会议服务器，开始一个视频会议。
Products as video conferencing gateway. Send a message related to video conferencing to video conferencing product, as the server, start a video conference. (2011-06-08, Visual C++, 10379KB, 下载9次)
[IP电话/视频会议] opensips-1.6.3-tls_src.tar
OpenSIPS是一个成熟的开源SIP服务器，除了提供基本的SIP代理及SIP路由功能外，还提供了一些应用级的功能。OpenSIPS的结构非常灵活，其核心路由功能完全通过脚本来实现，可灵活定制各种路由策略，可灵活应用于语音、视频通信、IM以及Presence等多种应用。同时OpenSIPS性能上是目前最快的SIP服务器之一，可用于电信级产品构建。
OpenSIPS is a mature open source SIP server, in addition to providing basic SIP proxy and SIP routing capabilities, but also provides some application-level functions. OpenSIPS structure is very flexible routing capabilities of its core achieved entirely through the script can be flexible and customize a variety of routing strategies can be applied in a flexible voice, video, communications, IM and Presence, and other applications. At the same time OpenSIPS performance is one of the fastest growing SIP server can be used to build carrier-class products. (2010-12-22, C/C++, 4126KB, 下载28次)
[IP电话/视频会议] YSvideoconferencing
功能全面，支持多人视频会议，单播，多播，文字聊天，文件传输，有中央服务器监控客户端
Full-featured, multiplayer video conferencing, unicast, multicast, text chat, file transfer, there is a central server monitoring the client (2009-12-01, Visual C++, 2900KB, 下载103次)
[IP电话/视频会议] IP_phone_system_client_server
IP电话会议系统
包含服务器端和客户端源程序，支持语音方面的压缩和解呀处理
IP telephone conference system includes a server-side and client source code, support for voice compression in dealing with the reconciliation (2009-09-02, Visual C++, 992KB, 下载49次)
[IP电话/视频会议] SIP8
目前，会话初始协议(SIP)大部分认证机制只提供了服务器到客户端的认证，H ITI P摘要
认证便是其中的一种。分析了这种机制容易遭受服务器伪装攻击和密码窃取攻击的缺陷，提出了一
种弥补这些缺陷的安全认证机制。试验表明该算法具备较高的效率。
At present, the Session Initiation Protocol (SIP) authentication mechanism most only provide a server to the client authentication, H ITI P summary of certification is one of them. Analysis of such a mechanism are vulnerable to server spoofing attacks and password theft attacks defects, a make up for deficiencies in the safety of these authentication mechanisms. Tests show that the algorithm with higher efficiency. (2009-01-20, Others, 251KB, 下载13次)
[IP电话/视频会议] 2005050214592410040
视频会议源码(客户端和服务器端,包含完整的开发文档)
Video conferencing source code (client and server, including the complete development documentation) (2008-11-12, Visual C++, 27KB, 下载399次)
[IP电话/视频会议] livemedia
RTP RTSP的实现代码，适合在Livemedia的基础上开发自己的流媒体客户端
Realize the RTP RTSP code, suitable for use in Livemedia based on developing its own streaming media client (2008-05-13, Unix_Linux, 444KB, 下载91次)
[IP电话/视频会议] vocal_ua-1.3.0
很好用的基于sip的代理，可以在vc等编译器下编译在windows下进行验证。
Sip on a good use of agents, such as in vc compiler to compile in windows under the circumstances to verify. (2007-07-25, Visual C++, 2387KB, 下载30次)
[IP电话/视频会议] dm642tcp ip原代码
这是在ti的媒体处理器DM642上进行网络视频传输中的tcp ip协议的实现
This is the ti DM642 media processor for network video transmission of tcp ip Protocol Implementation (2005-12-29, C/C++, 18KB, 下载134次)
[IP电话/视频会议] iTalk
一个用java实现的局域网语音聊天系统，包括服务器端和客户端，可以实现文字聊天，语音聊天，文件传输等功能，并实现了密码验证。
used to achieve a LAN voice chat system, including the server and client, can achieve a text chat, voice chat, file transfer and other functions, and to achieve a password verification. (2005-05-16, Java, 175KB, 下载785次)
[IP电话/视频会议] WinStun
windows下的STUN服务器，已经测试通过，可以在windows下运行通过
windows under the STUN server, the test can be run through windows (2005-03-18, Visual C++, 1305KB, 下载1180次)