搜索资源列表
siproxd-0.5.7.tar
- Siproxd is a proxy/masquerading daemon for the SIP protocol. It allows SIP clients (like kphone, linphone) to work behind an IP masquerading firewall or router.
astguiclient_1.0.3
- This program was designed as a GUI client for the Asterisk PBX with Digium Zaptel cards and SIP VOIP hard or softphones as extensions, it could be adapted to other functions, but It was designed for Zap/SIP users. The program will run on X and Win32.
t38modem_0.6.2.tar
- modem for voip
ip电话实例
- 一个ip电话实例的实现，以vc为开发环境。-an example of the realization of telephone, vc for the development environment.
librtp-0.1.tar
- 一个RTP库，用于音视频的打包实时传输，c语言编写-a library, audio and video for the real-time transmission package, c language
回声消除算法
- 回声消除算法的C＋＋程序，运行在unix/linux上， 也可以在vc中被编译， 测试， 效果还行-The program file of the algorithm for echo cancellation written by C++, can be runed on unix/linux, and also can be complied in VC It efficient good by testing.
iphone2
- 本程序自带反向回传语音代理服务，可以使两个不同局域网内的客户机通过国际互连网进行语音通讯，本程序使用点对点方式，不通过任何中介服务器速度快，语音延迟极小，声音清晰逼真，其效果和真实IP电话不相上下，程序界面简洁，操作简单，是在互连网上打IP电话的极好工具，唯一的要求是通话双方都要有电脑，并连入互连网。 本程序能实现以下连网操作功能： 1.通话双方是同一局域网内的用户 2.通话双方是不同局域网内的用户，并通过各自的网关连入互连网 3.通话双方一个直接连入互连网，另一个处在
desh2Phone_src_rate
- VOIP行业的H323协议终端软电话的源代码，对VOIP行业的开发、技术维护人员有较大帮助。-H323 VOIP industry agreement soft phone terminal source code for the development of the VoIP industry, technical maintenance staff have more help.
VideoNet
- 该程序可以用于两个人在LAN/Intranet(或者Internet)上进行视频会议。现在有许多视频会议程序，每个都有各自的性能提升技术。主要的问题是视频会议视频帧的尺寸对于传输来说太大。因此，性能依赖于对帧的编解码。 -the two procedures can be used in LAN / Intranet (or the Internet) for video conferencing. Now there are many video conference proceedings,
h225v301
- H.225 video-telephone protocol,specification document,for reference only
SIP消息实例讲解ppt
- SIP协议message交互实例剖析，相当浅显易懂的资料，适合初学者。-SIP message interactive examples analysis, and very easy-to-digest information for beginners.
SIP to Freshman
- 非常适合初学者了解SIP协议整体结构框架和功能的ppt资料-very suitable for beginners to understand SIP framework of the overall structure and function of information ppt
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
asterisk-1.2.0.tar
- 这个是Asterisk的源码。是一个IP PBX开源的项目，适合做IP电话等业务。支持sip，h323，mgcp等协议。-this is the source Asterisk. IP PBX is an open-source project, suitable for IP telephony services. Support sip, h323, Controller and other agreements.
SIP多方会话消息之实例讲解
- SIP多方会话消息之实例讲解，适合音频会议。-SIP multi-session news on the case, suitable for audio conferencing.
jrtplib-3.3.0
- Jrtplib库，附件版本3.3.0。实现RTP/RTCP协议，用于音频、视频网频传输。开发VOIP的朋友可以参考。-Jrtplib library annex version 3.3.0. Achieving RTP / RTCP for audio, video frequency transmission network. Development of VoIP friends reference.
An H.264-Based Solution on the DM642 for Video Bro
- An H.264-Based Solution on the DM642 for Video Broadcast Applications
CiscoQoSVoIPSolutionsGuide
- This preface introduces the QoS for Voice Over IP Solutions Guide, which explains quality of service for Voice over IP (QoS for VoIP) and addresses how to deploy end-to-end QoS for voice traffic throughout the components of an internetwork. It al
a-system-for-facetoface-meeting
- 大型多媒体视频会议服务器端和客户端.很好用的，希望对大家有帮助。-a system for facetoface meeting.
zlchat-for-oa-2.2-Crack-patch
- zlchat for oa 2.2 Crack patch