Description: 这是一个sip软终端pjsip的开发文档,里面详细介绍了pjsip。是英文的PDF文档。-This is a sip softphone pjsip the development of the document, which detailed the pjsip. PDF documents are in English. Platform: |
Size: 826368 |
Author:刘焱磊 |
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Description: 基于SIP的客户端,可以实现与服务器的语音视频通信 ,很有用-SIP-based client can be achieved with the server s voice and video communications, very useful Platform: |
Size: 10390528 |
Author:ccxxzcv |
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Description: yate是一个软交换的sip电话。也是一个voip服务器或客户端。
主要支持功能:
VoIP 服务器 VoIP 客户端
VoIP to PSTN 网关 PC2Phone and Phone2PC 网关
H.323 网守
H.323 多端点服务器
H.323<->SIP 转换代理
SIP session border controller
SIP 路由 S
IP 注册服务
Jingle 即时聊天 I
SDN passive and active recorder
IAX2服务器客户端
电话服务器和客户端
呼叫中心服务器 (会议,队列)
IVR语音交互应答
预付费,后付费电话卡系统
兼容Asteirsk的zaptel中继卡
支持linux /windows-Yate is a next-generation telephony engine while currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. Voice, video, data and instant messaging can all be unified under Yate s flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses. Platform: |
Size: 2009088 |
Author:Banlyst Yeh |
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Description: 为解决视频监控系统客户端的视频播放问题,提出了SIP(Session Initiation
Protocol)协议、RTP(Real-time Transport Protocol 协议以及Directshow 技术接收服务
器端发来的视频流数据并实时播放且成功存储的方法。-To solve the problem of video broadcast in the client of video surveillance syetem,a
methord of receiving the video stream,real-time play and video stream storage based on SIP
protocol,RTP protocol and Directshow technology is presented Platform: |
Size: 351232 |
Author:赵一方 |
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Description: 基于SIP协议栈的IP电话客户端 支持音视频传输-IP phone client support audio and video transmission based on SIP protocol stack Platform: |
Size: 9456640 |
Author:YOU |
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Description: SIP视频电话客户端 支持linux平台 是通过QT库开发的-SIP video telephony client support for linux platform is developed through the QT library Platform: |
Size: 1992704 |
Author:jia wenquan |
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Description: SIP视频电话客户端 支持linux平台 是通过QT库开发的-SIP video telephony client support for linux platform is developed through the QT library Platform: |
Size: 1167360 |
Author:jia wenquan |
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Description: SIP视频电话客户端 支持linux平台 是通过QT库开发的-SIP video telephony client support for linux platform is developed through the QT library Platform: |
Size: 3701760 |
Author:jia wenquan |
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Description: SIP视频电话客户端 支持linux平台 是通过QT库开发的-SIP video telephony client support for linux platform is developed through the QT library Platform: |
Size: 278528 |
Author:jia wenquan |
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Description: vc实现基于sip协议的客户端,Linux C实现基于sip协议服务器。客户端类似于QQ聊天工具,可实现文本聊天,语音对讲以及文件传输和视频聊天。-vc the sip protocol-based client, Linux C achieve the sip protocol-based server. The client is similar to the QQ chat, text chat, the voice intercom as well as the file transfer and video chat. Platform: |
Size: 11876352 |
Author:sjz |
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Description: jVoIP is a simple SIP Phone based on NIST Sip Communicator.
The phone won t be a fully functional SIP Client:
it won t support REGISTER request
it will be design for LAN voice communication
This phone won t function behind NAT or FireWall
This project isn t actively developed, the interface language is only in italian
Features
nice and simple to use GUI
simple configuration
high-level assumptions
We will not, consider certain functional areas like internationalization, high security, istant messaging and video phone
Screenshots
jVoip main window appears like
jVoIP settings window appears like
Notes
This project is built using Eclipse. To checkout source code refer to the CVS instructions. Then to build it you need install and configure JMF, the Java Media Framework.
To Do list
Localize in english and other languages
Remove uneeded class
Fix and add comments
Related resources
Sip Communicator: http://sip-communicator.dev.java.net/
Platform: |
Size: 6132736 |
Author:lifawen |
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Description: jssip 一个基于webrtc的web sip电话功能。可以连接websocket-JsSIP: The JavaScript SIP Library
Runs in the browser and Node.js
SIP over WebSocket (use real SIP in your web apps)
Audio/video calls (WebRTC) and instant messaging
Lightweight!
100 pure JavaScript built the ground up
Easy to use and powerful user API
Works with OverSIP, Kamailio, Asterisk, OfficeSIP and more (more info)
Written by the authors of RFC 7118 and OverSIP
Platform: |
Size: 6700032 |
Author:ljt |
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