搜索资源列表
rtc_for_VC6
- 通过RTP协议进行通讯的完整例子，本人已经在VC6上编译通过。 可以向支持SIP协议的服务器发起VOIP呼叫,并可以进行文字、视频、文件的传输。-through RTP for a complete communications example, I have compiled the VC6 through. To support SIP Server launched VoIP call, and can be text, video, document transmission.
ser-0.8.11_src.tar
- 一个SIP协议SIP server应用程序,较全面的实现了一个SIP服务器的各项功能.
SIP-NAT
- sip协议中的Nat穿越技术。SIP发送一个INVITE包到FWD SERVER，其中包含有主叫方的RTP的IP地址和端口，FWD将这个包转到对应的被叫方，被叫方接受了呼叫并将它自己的RTP的IP地址和端口返回来
视频会议完整的开发文档
- 视频会议(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)
linphone-3.1.0.tar.gz
- sip软电话，采用SIP协议，可以连接到sip服务器上，实现局域内部拨号连接。,sip soft phone, using SIP protocol, you can connect to the sip server to achieve the internal LAN dial-up connection.
opensips-1.6.2-tls_src.tar
- OpenSIPS开源SIP协议原码，实现SIP服务器。-OpenSIPS open source SIP protocol, SIP server implementation.
sipPhone
- 基于sip的voip网络电话的客户端程序，服务器可用开源平台asterisk，实现了单呼、组呼、强插、强拆等功能。-The voip sip based VoIP client, the server is available open-source platform asterisk, to achieve a single call, group call, Override, demolitions and other functions.
obelisk-1.0.0
- This project contains a SIP stack and server applications built on top of the stack, examples of which are: Stateless Proxy, Registrar,NAT Keepalives,MWI Notifier server. The server applications have been designed to work in association with Asterisk
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 authentic
dtmfbox-0.5.0
- can be used as a good example of implementation of SIP register server
dtmfbox-0.5.0_win32
- can be used as a good example of implementation of SIP register server (win32 platform)
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
kamailio-1.5.0-tls_src.tar
- Kamailio is a GPL licensed SIP server implementation. It started in 2005 as a fork of Fokus Fraunhofer SIP Express Router (SER) project. Kamailio wants to be a more open project, not only from license point of view, but more open as project managemen
live.tar
- 包含RTP/RTCP/RTSP以及SIP的协议栈源码， 内含Mpeg1/2/4以及H.263/H.264的Streaming RTP代码，适用于流媒体相关开发的Client和Server-Contains the RTP/RTCP/RTSP and SIP protocol stack source code, including the Mpeg1/2/4 and Streaming RTP code H.263/H.264 for streaming media related to t
servertest
- This source code has a useful information for all beginners using SIP server and developing an application like softphone. Also can Transmit and Receive voice by replacing the IP address.
SIP-Asterisk
- 如何将sip网关注册到asterisk服务器简要介绍。-How to register to the asterisk server sip gateway briefly.
SIP
- SIP protocol in C++ the client and server are implemented the server can save all port of clients connected to server
sip
- SIP Server的基本原理，包括sip信令的说明-SIP Server' s basic principles, including a descr iption of sip signaling
sip-client
- sip client library to connect with asterisk or another sip server.
sip
- SIP UDP redirect and registrar server