搜索资源列表
ip电话
- IP电话演示，Delphi7编译通过-IP telephony demonstration compile Delphi7
sflphone.tar
- IP电话源程序，在Unix环境下编译通过，基于h.323协议。-source IP telephony, in the Unix environment compiler, based on the H.323 agreement.
网络视频电话
- 高性能的可视电话软件 NetTalk效率很高，比我所见的一些可视通话软件要好（如AVPHONE，注：在此并无破坏AVPHONE所在公司形象之意图，仅将软件作客观上的比较） 全图形界面 全图形的界面使得操作方便快捷-high-performance video telephony software NetTalk high efficiency, I have seen some better visualization software calls (such
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.
可视电话
- 可视电话，一个小的可视电话源码。voip方面内容-video phone, a small video telephony source. Voip aspects.
vis_h323lib_dll_2.0r
- H.323通信库的源程序，用于IPVoice，做IP电话的可以参考-H.323 library source for IPVoice do IP telephony can reference
IP-yd1046
- 中国国家信息产业部的IP电话网关设备互通技术规范yd1046（中文）-Chinese Ministry of IP telephony gateway equipment interoperability specifications yd1046 (Chinese)
intercom-0.4.1
- 流行IP电话源程序，内含G.711、G.726、GSM、iLBC、回声消除等源代码，Linux平台C语言-popular source IP telephony, containing G.711, G.726, GSM, VoIP, echo cancellation, etc. source code, C language Linux
IPphonemeetting
- 一个非常好的ＩＰ电话电视会议系统，对于初学者非常有用.-a very good video conferencing, IP telephony systems, very useful for beginners.
OReilly.Asterisk.The.Future.of.Telephony.Sep.2005.
- voip开发资料，详细介绍了开源软件asterisk的发展及里面的脚本设计
IP电话的TCP,IP实现方法
- IP电话的TCP,IP实现方法.pdf-TCP IP telephony, IP method. Pdf
asterisk-chn2
- voip telephony asterisk chinese doc
H323BeaconClient-v1.4_win_src
- 支持h323的视频电话客户端-support h323 video telephony client
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
libosip-0[1].7.9.tar
- oSIP实现了会话初始协议Session Initiation (由IETF发布为RFC 3261)。它可以为多媒体应用程序(IP电话，等等)提供信令功能。它为SIP语法提供了一个完全可用的语法分析器，实现了草案中定义的“事务”层。它也提供了一个SDP分析器和为用户代理提供的额外功能。它可以用于建造代理服务器和IP电话-oSIP achieve Session Initiation Protocol Session Initiation (released by the IETF as RFC
asterisk-1.8.3.2.tar
- 最新的asterisk源码，pbx服务器，sip网络电话服务器，可实现模拟电话和网络电话互通-Asterisk source, pbx server, sip network telephony server, analog phones and VoIP interoperability
Delphi写的IP电话源码
- Delphi写的IP电话源码-Delphi written source IP telephone
H323PHONE
- C++ 的源代码，H323，大家可以作为开发软电话的参考，我以前帮朋友做的 数字电视网络电话软件就参考的此源码-C++ source code, H323, everyone can serve as a reference for the development of the soft phone, I used to help make digital TV Friend Internet telephony software on the reference of this source
Vax_SIP_User_Agent_SDK_7.0.3.4
- To resolve these issues, ITSP (IP-Telephony service provider) support SIP OUTBOUND PROXY. Outbound proxy is the only way to let the NAT/firewall user make and receive phone calls. If the NAT/firewall router does not support SIP pass-through, you ne
tmndec-2.0
- H263的标准解码器，适用于视频电话，移动通信等领域。输入可以是.3gp的格式-H263 standard decoder for video telephony, mobile communication and other fields. Input can be a .3 gp format