[IP电话/视频会议] asv_client
绝对安全的视频聊天是一种纯粹的基于Web RTC的视频会议和消息传递应用程序。它的主要特点是完全匿名、免费，并且不在服务器端存储任何数据。
Absolutely secure video chat is an pure Web-RTC-based video conferencing and messaging application. The main feature is that it is completely anonymous, free and does t store any data on the server side. (2024-02-12, JavaScript, 0KB, 下载0次)
[IP电话/视频会议] GGMeeting_a5
语音视频数据都是实时采集、实时播放的数据，所以测试时，服务器的带宽要求最好是独享带宽，共享带宽一般无法满足实时语音视频的要求。 实时语音视频而言，100M的共享带宽，还不如5M的独享。
Voice and video data are real-time data acquisition and playback, so when testing, the best bandwidth requirement of the server is exclusive bandwidth. Shared bandwidth generally can not meet the requirements of real-time voice and video. For real-time voice and video, the shared bandwidth of 100M is not as good as that of 5M. (2019-08-01, C#, 14189KB, 下载5次)
[IP电话/视频会议] SIPPlayer
SIP是一个应用层的信令控制协议。用于创建、修改和释放一个或多个参与者的会话。这些会话可以好似Internet多媒体会议、IP电话或多媒体分发。这是一个基于SIP协议的媒体播放器
SIP is an application-layer signaling protocol. Used to create, modify and release one or more of the participants of the session. These sessions can be like Internet multimedia conferencing, IP telephony and multimedia distribution.it s a media player based on SIP (2016-09-09, Visual C++, 75668KB, 下载13次)
[IP电话/视频会议] apache-openmeetings-2.1.1-src
OpenMeetings是一个多语言可定制的视频会议和协作系统。它支持音频、视频，能让你查看每个与会者的桌面。OpenMeetings还包含一个白板，通过白板可以导入各种格式的图片和涂鸦。
它是基于OpenLaszlo’s的新流媒体格式和开源的Flash服务器---Red5! [1]
OpenMeetings is a multi-language customizable video conferencing and collaboration system. It supports audio, video, allowing you to view each participant' s desktop. OpenMeetings also includes a whiteboard, whiteboard can import a variety of image formats and graffiti. It is based on OpenLaszlo' s new streaming media formats and open source Flash server--- Red5! [1] (2013-07-13, Java, 16093KB, 下载9次)
[IP电话/视频会议] sdk
大华网络sdk开发手册
欢迎使用网络SDK编程手册，网络SDK是软件开发者在开发网络硬盘录像机、网络视频服务器、网络摄像机、网络球机、智能设备等产品监控联网应用时的开发套件。本文档详细描述了开发包中各个函数实现的功能、接口及其函数之间的调用关系和示例实现.
本开发套件主要包括业务操作和设备管理两大部分：
业务操作
状态侦听、实时监视、实时预览、字符叠加、音频控制、录像回放和下载、数据保存、云台控制、语音对讲、透明串口、码流统计、智能分析等功能。
设备管理
远程升级、远程重启/关闭、设备参数配置（系统通用配置、报警布/撤防设置、录像配置、串口配置、图像配置、日志管理、用户管理、设备校时、动态检测配置、网络配置）等功能。
dahua sdk (2013-07-11, Visual C++, 2681KB, 下载86次)
[IP电话/视频会议] SipPhone
该项目是一个基于SIP协议的IP电话，是项目的1.0版本，实现了基本的语音功能，项目基于VOIP开发完成，该项目可以与IP电话模拟器进行正常的通话
The project is a SIP-based IP telephony, version 1.0 of the project to achieve a basic voice functions, project-based VOIP development is complete, the project can be normal calls with IP telephony simulator (2013-04-10, Visual C++, 7999KB, 下载13次)
[IP电话/视频会议] nettalker
套软件视频会议包括客户端+服务器端全套完整源代码，还有开发文档。也可进行二次开发。可与贵单位原有的网站和业务管理软件进行很好的融合（如OA 系统、MIS系统、ERP系统等），为您提供广泛的应用服务
Sets of software video conferencing+ server side of the client complete source code, as well as development of the document. Secondary development. A good integration with your organization' s existing website and business management software (such as OA systems, MIS systems, ERP systems, etc.), and to provide you with a wide range of application services (2012-09-06, Visual C++, 1253KB, 下载65次)
[IP电话/视频会议] oosiip_eXosiiS
osip实现RFC3261，全部用C语言实现，跨平台编译，小巧，简单易懂，但是没有实现UA的能力；eXosip实现了UAA的基本功能，是一种基于osip实现的。两个库唯一的缺点就是性能差了一些，不太适合服务器产产品的开发，但是适合终端的开发。
osip achieve RFC3261, all using C language, cross-platform compilation, compact, easy to understand, but not the ability of UA eXosip achieve the basic functions of the UAA is a based on osip achieve. Two libraries only drawback is the poor performance of some, not suitable for the development of server products production, but for the development of the terminal. (2012-08-30, C/C++, 1110KB, 下载7次)
[IP电话/视频会议] trtcc_for_VCCh
通过RTP协议进行通讯的完整源码实例，本人已经在Visual C6上编译通过过。能向支持SIP协议的服务器发起VOIP呼叫，并能进行文字、视频、文件的传输。
Through the RTP protocol to communicate with complete source instance, I have compiled through Visual C6 too. VOIP call can be initiated to support SIP server, and the text, video, and file transfers. (2012-07-27, Visual C++, 124KB, 下载9次)
[IP电话/视频会议] RFC3550
本文描述RTP（real-time transport protocol），实时传输协议。RTP在多点传送（多播）或单点传送（单播）的网络服务上，提供端对端的网络传输功能，适合应用程序传输实时数据，如：音频，视频或者仿真数据。RTP没有为实时服务提供资源预留的功能，也不能保证QoS（服务质量）。数据传输功能由一个控制协议（RTCP）来扩展，通过扩展，可以用一种方式对数据传输进行监测控制，该协议（RTCP）可以升级到大型的多点传送（多播）网络，并提供最小限度的控制和鉴别功能。RTP和RTCP被设计成和下面的传输层和网络层无关。协议支持RTP标准的转换器和混合器的使用。
本文的大多数内容和旧版的RFC1889相同。在线路里传输的数据包格式没有改变，唯一的改变是使用协议的规则和控制算法。为了最小化传输，发送RTCP数据包时超过了设定的速率，而在这时，很多的参与者同时加入了一个会话，在这样的情况下，一个新加入到（用于计算的可升级的）计时器算法中的元素是最大的改变。
This paper describes the RTP (real-time transport the protocol), real-time transport protocol. RTP in a multicast (multicast) or unicast (unicast) network services, providing end-to-end network transport functions suitable for applications transmit real-time data, such as: audio, video or simulation data. RTP does not provide for real-time service resource reservation function, we can not guarantee QoS (Quality of Service). Data transfer function is extended by a control protocol (RTCP), and by extension, a way of data transmission monitoring and control, the Protocol (RTCP) can upgrade to large multicast (Multicast) network, and provide a minimum limit control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP standards converter and mixer.
Most of the content of this article and previous versions of RFC1889. The line transmission packet format has not changed, only change (2012-03-28, Visual C++, 51KB, 下载10次)
[IP电话/视频会议] shiyan6
用RMI构建一个分布式共享会议议程服务，不同的客户应该能够使用共享会议议程服务，该服务提供会议的查询，增加和删除功能，会议议程服务器有允许用户注册和撤销会议的功能。
RMI to build a distributed shared with the meeting agendas, different customers should be able to use the shared services agenda, the service provider meeting the query, add and delete functions allow the user to the agenda server has the function of registration and revocation of the meeting. (2011-06-08, Java, 28KB, 下载11次)
[IP电话/视频会议] mumble-1.1.8
Mumble是一个自由、开源 和跨平台的VOIP软件。它的主要用户是游戏玩家，它类似于软件 TeamSpeek and Ventrilo。和相同游戏服务器通话的玩家组成了C/S架构。
Mumble is a free, open source and cross-platform VOIP software. Its main users are gamers, which is similar to the software TeamSpeek and Ventrilo. And the same game server calls the players and the C/S architecture. (2009-11-02, C++, 1531KB, 下载14次)
[IP电话/视频会议] freeswitch.1.0.rc1
一个开源的sip软交换交换机的源代码.FreeSWITCH 是一个电话的软交换解决方案，包括一个软电话和软交换机用以提供语音和聊天的产品驱动。FreeSWITCH 可以用作交换机引擎、PBX、多媒体网关以及多媒体服务器等。
An open-source softswitch sip switch the source code. FreeSWITCH is a telephony Softswitch solution, including a soft-phones and soft-switches to provide voice and chat-driven products. FreeSWITCH can be used as a switch engine, PBX, media gateways, and multimedia servers. (2009-08-11, Windows_Unix, 24367KB, 下载54次)
[IP电话/视频会议] iNetTalkNew
修正了NetTalk的原作者指出的问题，另外增加了与远程服务器通讯并能进行P2P连接内网用户的功能。并在通讯模式方面简化了而直接采用“电话号码”的方式进行拨叫远程用户。
NetTalk amended the original author pointed out that the issue of additional communications with the remote server and can connect to P2P users within the network function. And simplify the communication model and direct the use of " phone number" to the remote user to dial. (2009-04-27, Visual C++, 9380KB, 下载61次)
[IP电话/视频会议] SipGen20051013
免费开放的SIP消息产生器，用于SIP并发测试，能以多种方式循环发送和接收SIP消息，并能根据事先配置的脚本自动统计、显示消息发送或是接收的情况。
Free and open source SIP generator for SIP concurrent testing cycle to a variety of ways to send and receive SIP messages and can be configured in accordance with the prior script automatically statistics, displays the message sent or received by the situation. (2008-07-14, Visual C++, 43KB, 下载28次)
[IP电话/视频会议] 200851313207271
本视频会议系统使用户可以利用普通的PC机、标准的视频采集设备（USB摄像头或者视频头+视频采集卡）、耳机和麦克风，实现基于Internet、广域网、局域网的虚拟会议。与传统的基于硬件的H.323解决方案相比，本系统是纯软件解决方案。用户无需投入高昂的成本，就能够实现高质量、高可靠性的音视频通讯的会议功能，有效地节约时间和经费，提高企业的工作效率。
本视频会议系统由服务器软件和客户端软件组成。
The video conferencing system allows users to use an ordinary PC, the standard video capture device (USB cameras or video head+ Video capture card), headphones and microphone, based on Internet, WAN, LAN virtual meetings. With traditional hardware-based solutions compared to H.323, the system is a pure software solution. Users without the high cost of inputs, we can realize high-quality, high reliability, communications, audio and video conferencing capabilities, effectively saving time and funds to enhance enterprise work efficiency. The video conferencing system from server software and client software. (2008-05-25, Visual C++, 6766KB, 下载1414次)
[IP电话/视频会议] 为VoIP铺平道路ExpandVoIP-CH
本白皮书分析了VoIP的性能和带宽要求，并介绍了扩展网络公司的加速器（Expand Networks
ACCELERATORs）如何有助于提供所需的性能，而同时在集成网络中降低WAN的成本。
VoIP analysis of the performance and bandwidth requirements, and expansion of the company's network accelerator (Expand Networks, ACCELERATOR s) can help provide the necessary performance, while the integrated WAN network to reduce costs. (2005-12-27, Windows_Unix, 224KB, 下载28次)
[IP电话/视频会议] IP电话源码
开机：打开监听服务，以便别人可以和你通话。如果你有多个IP地址，会提示让你选择用哪个地址开机。
关机：关闭监听服务，已经接受的连接不受影响，你可以继续通话。
拨号：连接已经开机的机器，弹出对话框后必须输入对方的IP和端口，如果需要使用代理，请正确设置代理服 务器。
挂机：断开正在进行的通话。
开机后会显示你监听的IP和端口，别人可以通过这个IP和端口和你连接。
连接成功（你拨号连接别人或别人来连接你）后会显示对方的IP和端口。
关于代理服务器：
开机：支持Socks 4/5代理或者同时使用HTTP代理，如果不使用Socks代理，则HTTP代理无效。使用代理开机成功后显示的是代理服务器分配的IP和端口，别人必须使用该IP和端口才能和你连接。开机代理在主窗口上设置。
拨号：支持Socks 4/5和HTTP代理，或者二者的组合，组合使用时的连接顺序是HTTP->Socks。拨号代理在点击“拨号”后弹出的对话框里设置。
本版本修复了以前版本的一些小问题，增加了动态调整延时，打开和关闭话筒，声音，保存和自动加载通话地址列表，代理服务器列表，使用全部或部分本机地址开机。从本版本起，地址都为IP:PORT格式。
boot : Open monitoring services, and so that other people can talk to you. If you have multiple IP addresses, so you are prompted to choose which addresses boot. Shutdown : Close monitoring services, has accepted the connection will not be affected, you can continue to talk. Dial-up : Connecting switched on the machine, pop-up dialog box appears to be imported each other's IP and port, if the need to use the agent, the right to set up the proxy server. Coolness : disconnect the ongoing conversations. After boot monitor will show you the IP and port, the others through IP and port and connect you. Connecting success (your dial-up connection or other people to connect you) will show the other side of the IP and port. Acting on the server : boot : support for Socks 4/5 proxy or at the s (2005-11-28, Others, 262KB, 下载79次)
[IP电话/视频会议] 网络电话
使用说明： 开机：打开监听服务，以便别人可以和你通话。如果你有多个IP地址，会提示让你选择用哪个地址开机。 关机：关闭监听服务，已经接受的连接不受影响，你可以继续通话。 拨号：连接已经开机的机器，弹出对话框后必须输入对方的IP和端口，如果需要使用代理，请正确设置代理服 务器。 挂机：断开正在进行的通话。
use : boot : Open monitoring services, and so that other people can talk to you. If you have multiple IP addresses, so you are prompted to choose which addresses boot. Shutdown : Close monitoring services, has accepted the connection will not be affected, you can continue to talk. Dial-up : Connecting switched on the machine, pop-up dialog box appears to be imported each other's IP and port, if the need to use the agent, the right to set up the proxy server. Coolness : disconnect the ongoing conversations. (2005-04-06, Visual C++, 263KB, 下载109次)
[IP电话/视频会议] vocal-1.5.0.tar
Vovida出品的包括RTP、RTCP、SIP等各种基础协议栈和SIP UAC、UAS等功能的IP电话服务器端程序。编译环境为linux
Vovida produced in RTP, RTCP, and other SIP based protocol stack and SIP UAC and UAS capabilities of IP telephony server procedures. Linux compiler environment (2005-02-22, Unix_Linux, 16441KB, 下载370次)