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按分类查找All IP电话/视频会议(280) 

[IP电话/视频会议] Video-Conference-and-Chat-App

使用webrtc、简单对等和socket.io.React typescript客户端、node.js服务器和MongoDB数据库进行组视频会议和消息传递,
Group video conference and messaging using webrtc, simple-peer and socket.io. React-typescript client, node.js server and MongoDB database, (2020-09-20, TypeScript, 0KB, 下载0次)

http://www.pudn.com/Download/item/id/1696774225368175.html

[IP电话/视频会议] linphone-3.5.2

google到了windows上面的sip server 2008 + x-lite的挺简单的(带图文教程),所以照着做了一下,也可以连上自己的服务器。
Google came to windows SIP Server 2008 + X-Lite's simple (with graphics and tutorials), so do it, or connect to your server. (2018-01-16, Unix_Linux, 8363KB, 下载3次)

http://www.pudn.com/Download/item/id/1516086174272225.html

[IP电话/视频会议] sipp_register

sipp测试工具所使用的测试脚本代码,包含sip用户注册服务器端、客户端、以及正常、异常、随机、混合多种业务模型下的脚本demo。
Test script code sipp testing tools used, including the sip user registration server, client, and normal, abnormal, random, mixed model demo script. (2015-11-25, Others, 21KB, 下载24次)

http://www.pudn.com/Download/item/id/1448441702786798.html

[IP电话/视频会议] gist.tar

东西一般大家多包涵@@自己写的一个rtsp简易调试服务器,用于模拟rtsp信令给客户端,调试客户端处理流程,并发送1个udp数据用于客户端调试rtp处理流程。作者本人就是自己开发这套工具
just soso it very smipler (2013-05-22, Python, 1KB, 下载5次)

http://www.pudn.com/Download/item/id/2253306.html

[IP电话/视频会议] PESQcvibuild.sk

PESQ客观评估语音质量,为评定宽带电话网络和语音编解码器对P.862建议书 的宽带扩展
Perceptual evaluation of speech quality (PESQ): An objective method for end-to-end speech quality assessment of narrow-band telephone networks and speech codec (2012-12-22, matlab, 1332KB, 下载28次)

http://www.pudn.com/Download/item/id/2092598.html

[IP电话/视频会议] pjsip_mediastream_test

基于pjsip 1.10的开源代码实现的测试mediastream 功能的测试代码,支持流媒体的收发。此文件是模拟代理服务器端代码。
The open source code based on pjsip 1.10 the test code test mediastream functions to support streaming transceiver. The project is proxy server-side code. (2012-12-08, Visual C++, 388KB, 下载31次)

http://www.pudn.com/Download/item/id/2076113.html

[IP电话/视频会议] Smr_sip_8826i

实现sip电话穿越防火墙的服务器端代码,利用媒体中转的的方式使防火墙外的sip电话可以呼叫防火墙内的sip电话
Sip call through the firewall to achieve the server-side code, using the media of the way transit outside the firewall can call sip phone sip phone within a firewall (2012-05-20, Perl, 12KB, 下载2次)

http://www.pudn.com/Download/item/id/1878804.html

[IP电话/视频会议] soundcode

这是一段简洁而有效的基于c/s模式语音通信源码,在vc或vs下通过编译,输入ip地址,客户端即可听到服务器端的声音
This is a simple and effective based on c/s mode of voice communication source, vs in vc or through the compiler, enter the ip address, the client can hear the voice server (2011-11-17, Visual C++, 4KB, 下载12次)

http://www.pudn.com/Download/item/id/1701930.html

[IP电话/视频会议] ServerPClient

一个完整的视频会议系统, 包括客户端,服务器, 数据库, 无残缺。
自动检测中英文中译英英译中百度翻译 翻译结果(中 > 英)复制结果 A complete video conferencing systems, including client, server, database. No missing parts. (2011-09-25, Delphi, 9885KB, 下载115次)

http://www.pudn.com/Download/item/id/1654443.html

[IP电话/视频会议] ns2VoIPPP150310

ns-2 VoIP仿真器补丁。 需要先下载ns-2.34,再用一下命令加补丁: cd /usr/local/ns-allinone-2.34/ns-2.34 cat /tmp/ns2VoIP++150310.patch | patch -Np1
ns-2 VoIP emulator patch. Need to download the ns-2.34, and then click Add patch command: cd/usr/local/ns-allinone-2.34/ns-2.34 cat/tmp/ns2VoIP++150310. Patch | patch-Np1 (2011-05-28, Unix_Linux, 45KB, 下载11次)

http://www.pudn.com/Download/item/id/1550821.html

[IP电话/视频会议] WEB

:概述   讯通视频语音程序(以下简称BD)是基于网络视频语音文字通讯的一套服务系统。适用于语音聊天室,网络会议,网上在线服务,网上教育,企业客服等 应用。   BD采用B/S形式实现视频聊天,服务端运行在win2000或2003服务器上,客户端使用IE浏览器即可,无需单独安装客户端软件,方便快捷
: almost  Assyria XunTong video voice program (hereinafter referred to as BD) is based on the network video speech text communications set of service system. Applicable to chat room, network conference, online services, online education, enterprise customer service, etc Applications. BD adopts B/S form video chat, the server realize running on win2000 or 2003 server, the client can use IE browser without requiring a separate installation client software, convenient and quick (2011-05-19, Java, 31KB, 下载22次)

http://www.pudn.com/Download/item/id/1538441.html

[IP电话/视频会议] YouToonew9

基于sip协议,用asterisk作为服务器,实现voip软电话功能,实现了接听,呼叫,挂机,呼叫转移,以及通信录等功能。
Based on sip protocol, with the asterisk as a server, voip soft phone capabilities to achieve and realize the answer, call, hang up, call forwarding, and address book functions. (2010-09-16, Visual C++, 24543KB, 下载75次)

http://www.pudn.com/Download/item/id/1297814.html

[IP电话/视频会议] VOIP

VOIP编码,实现PC上的语音通话功能,类似QQ语音。本代码需要至少两个电脑测试功能,其中一个作为服务器
VOIP coding, to achieve PC voice calls over the function, similar to the QQ voice. The code requires at least two computer test functions, one as a server (2010-05-25, Visual C++, 1631KB, 下载125次)

http://www.pudn.com/Download/item/id/1188395.html

[IP电话/视频会议] china-code.net.zhy.2009661649413662343

MeChat 实现多语音视频和视频会议功能 ,同时可以有十几个人同时语音视频交流.数据的传输采用P2P一样的原理,非常有效的智能的选择使用网络带宽,既保证语音和视频的高质量传输,又对服务的网络带宽没有高要求.客户容易自己美化界面,用户更多的控制功能。 MeChat安装、维护极其简单。MeChat采用C++编制,速度奇快,支持超过5000人同时在线! MeChat灵活的模板结构,提供您一个充分展示自己个性的平台。支持WindowNT/2000/XP/2003/linux/solaris/unix。本系统无需任何WEB SERVER、数据库系统支持,完全独立运行。MeChat Server可以把数据存储在文件中,也可以放在数据库中,使用数据库,更容易和其他程序一起使用。支持的数据库有Access,Ms Sql Server,MySql,Oralce等. 不同与一般的CGI和ASP的聊天程序,MeChat占用服务器的资源非常少。服务器硬件配置为:DELL PIII-550 256M内存,同一台服务器上还运行着许多应用, 500人同时在线时,MeChat仅消耗2 -6 的CPU,5M左右的内存。而且连续运行了三个多月从未出现过崩溃的情况。 如果用ASP或其他基于Web server的聊天程序,恐怕CPU至少也到99 了!
MeChat 实现多语音视频和视频会议功能 ,同时可以有十几个人同时语音视频交流.数据的传输采用P2P一样的原理,非常有效的智能的选择使用网络带宽,既保证语音和视频的高质量传输,又对服务的网络带宽没有高要求.客户容易自己美化界面,用户更多的控制功能。 MeChat安装、维护极其简单。MeChat采用C++编制,速度奇快,支持超过5000人同时在线! MeChat灵活的模板结构,提供您一个充分展示自己个性的平台。支持WindowNT/2000/XP/2003/linux/solaris/unix。本系统无需任何WEB SERVER、数据库系统支持,完全独立运行。MeChat Server可以把数据存储在文件中,也可以放在数据库中,使用数据库,更容易和其他程序一起使用。支持的数据库有Access,Ms Sql Server,MySql,Oralce等. 不同与一般的CGI和ASP的聊天程序,MeChat占用服务器的资源非常少。服务器硬件配置为:DELL PIII-550 256M内存,同一台服务器上还运行着许多应用, 500人同时在线时,MeChat仅消耗2-6 的CPU,5M左右的内存。而且连续运行了三个多月从未出现过崩溃的情况。 如果用ASP或其他基于Web server的聊天程序,恐怕CPU至少也到99 了! (2009-07-30, VBScript, 3808KB, 下载59次)

http://www.pudn.com/Download/item/id/861088.html

[IP电话/视频会议] mr_sip_826

实现sip电话穿越防火墙的服务器端代码,利用媒体中转的方式使防火墙外的sip电话可以呼叫防火墙内的sip电话
Sip call through the firewall to achieve the server-side code, using the media of the way transit outside the firewall can call sip phone sip phone within a firewall (2009-04-30, Perl, 12KB, 下载10次)

http://www.pudn.com/Download/item/id/738726.html

[IP电话/视频会议] JRTPUnicast

JRTP可用于组播也可用于单播,JRTP主页上有一个关于组播的客户端和服务器的例子,还有一个单播的客户端和服务器在同一个程序的例子,本人一直想做一个单播的客户端和服务器分开的例子,在想了一两天后终于想通,做完后发上来,秉承本人一贯风格,注释详尽、附上测试结果,倒不是什么难的东西,只是牛人都不愿做这样的东西,但还是对初学者非常有用,希望得此帮助的给个好评,只此一家,别无分店。
JRTP can be used for multicast can also be used for unicast, JRTP have a home page on the multicast client and server example, there is a unicast client and server program in the same example, I have always wanted to do a single broadcast of the separate client and server example, thinking about it for a finally realized after 12 days, after finishing up, and to inherit has been my style, notes in detail, together with test results, it is not a difficult thing, but cattle were are unwilling to do this kind of thing, but still very useful for beginners, I hope to have the help of praise, only this one, there is no branch. (2009-04-16, Visual C++, 3KB, 下载84次)

http://www.pudn.com/Download/item/id/718818.html

[IP电话/视频会议] ONS

一套商用的视频会议源代码~多的废话不说!有心人自己下来看看,对你以后写代码有很大帮助。
A commercial video-conferencing source code ~ number of do not talk nonsense! People look down their own, after you write the code are very helpful. (2008-12-11, Visual C++, 2894KB, 下载314次)

http://www.pudn.com/Download/item/id/601959.html

[IP电话/视频会议] voip_voice_design

VoIP语音卡的应用及硬件设计:介绍一种已经应用于路由器产品中的VoIP语音卡的硬件设计和工作原理。
VoIP voice card applications and hardware design: A router products have been used in the VoIP voice card hardware design and working principle. (2008-06-27, Unix_Linux, 176KB, 下载55次)

http://www.pudn.com/Download/item/id/499598.html

[IP电话/视频会议] sipXezPhone-0.35a

采用SIP协议编写的IP电话程序,运行于Windows平台,VC++编写
prepared using SIP IP telephony program, which runs on the Windows platform, VC prepared (2006-04-01, Visual C++, 991KB, 下载529次)

http://www.pudn.com/Download/item/id/164175.html

[IP电话/视频会议] ChatRoom+G

这是关于语音聊天软件的一个程序代码 当运行该软的时候,至少需要有一方按下“建立服务器”按钮,表示该台机子用于做服务器,成功时文本显示区会显示“Server Has Been Set OK!”,才有可能接受来自其他机子的连接,当然客户端要输入服务器的IP地址和指定端口后,然后按“连接服务器”按钮,若成功则会在本地文本显示区显示“Client Connection Succeed”,双方建立了连接后,就可以开始通信了。你可以在文本输入区输入你想说的话,按回车,就会在本地文本显示区显示你刚才说的话,差不多同时会在对方的文本显示区显示你说的话,当然为了区别双方说的话,会在自己说的话前加“host:”,在对方说的话前加"guest:";在建立连接后,若一方想传文件到另一方,则要按下“文件传输”按钮,此时会跳出一个对话框,要求你选择你要传的文件的目录,按确定以后会在对方屏幕上跳出一个对话框,要求你选择要保存的路径,若按了确定,则会显示文件传输进度条,当文件传完,还会有结束提示。若双方要进行语音通信,则需要双方约定,只有双方后按下了“语音聊天”才可以进行,当然想结束语音聊天只要再按一下那个按钮就可以了。退出软件按“The End”就可以了。
this is about voice chat software code as a soft run, at least one of the parties? "Establishment server" button and the station for the machine to do server success text display area will show "Server Has Been Set OK!" Is it possible to accept from the other machine connected, of course, to bring the client server and the IP address of the designated ports, and click on the "Connect server" button, if successfully text will be shown in the local area shows that the "Client Connection Succeed" and the two sides established a connection, the communication can begin. You can input text input area you wish to say, according to enter, the text will be shown in the local area that you just say so, almost the same time in each other's text display area (2006-03-18, Visual C++, 665KB, 下载112次)

http://www.pudn.com/Download/item/id/155581.html
总计:280