搜索资源列表
p2psip
- sip p2p的很好实现，学习p2p voip 好工具-sip p2p
YouToo
- VOIP方面的,SIP协议源码,编缉后生成DLL,可进行二次开发-VOIP, SIP protocol source code editing generated DLL secondary development
freeswitch
- FreeSWITCH是以C撰写而成的开放源码电话应用软件，可以连接SIP H.323、IAX2、LDAP、Zeroconf、XMPP / Jingle，Google Talk等现有技术，架构出开放源码PBX系统或开放源码的VoIP交换平台。-FreeSWITCH based on open source telephony application software written in C by, can to connect SIP H.323, IAX2, LDAP, Zeroconf, X
exosip2-3.6.0
- exosip3.6.0源码，开源SIP协议栈，具有WINDOWS和LINUX多平台的支持。对研究VOIP很有参考价值。-the source of exosip3.6.0, open source SIP protocol stack, multi-platform support for WINDOWS and LINUX. Research VOIP great reference value.
SipPhone
- 该项目是一个基于SIP协议的IP电话，是项目的1.0版本，实现了基本的语音功能，项目基于VOIP开发完成，该项目可以与IP电话模拟器进行正常的通话-The project is a SIP-based IP telephony, version 1.0 of the project to achieve a basic voice functions, project-based VOIP development is complete, the project can be normal ca
3CXPhone5
- 3cxphone,voip客户端 sip服务-3cxphone, voip sip client service
openbts-P2.8.0Opelousas
- OpenBTS 是从基带收发站（BTS）向上，完全替代传统的 GSM 运营商的网络交换构架。替代传统的传递呼叫到运营商的移动交换中心（MSC)的方法，它是通过 SIP 和 VOIP 传递数据到 Asterisk PBX 的。项目是由 Harving Samra 和 David A. Burgess 启动的。其目的是把边远和发展中地区地区的GSM服务费用减少到每个用户每月在1美元以下。项目的最初开发者涉及执照事宜，在初期的相关工作中官司连连（现已解决），也就是说一些底层的 GSM 代码被重新改写过
sipchat-0.3
- 实现sip的聊天客户端，它为架设voip提供参考和学习-Achieve sip chat client, it is set up voip reference and learning
T_long_to_float_2
- SIP Communicator, the OpenSource Java VoIP and Instant Messaging client but has been significantly changed for Andriod.
SipUA
- 他是网络电话（Voip）、SIP协议、android 网络电话源码 可直接导入eclipse 使用。-He is a VoIP (Voip), SIP protocol. Internet phone android source code can be used directly into eclipse.
baresip-0.4.7
- 一个小巧，性能不错的sip协议栈，c语言编写： * Minimalistic and modular VoIP client * SIP, SDP, RTP/RTCP, STUN/TURN/ICE * IPv4 and IPv6 support * RFC-compliancy * Robust, fast, low footprint * Portable C89 and C99 source code-* Minimalistic and modular VoIP c
Voip-SIP-Telephone
- 网络电话。哥伦比亚大学CSEE 4840课程设计-Internet telephony. Columbia University CSEE 4840 Course Design
sphone.tar
- Linux下一个voip软电话，C++语言编写，支持打电话和接电话。使用了sip和rtp协议，使用QT开发了简单的界面。-A voip soft phone on linux,calling and answering is supported.And the phone uses sip protocol and rtp protocol.Also,there is a simple gui made by QT.
voipmonitor-9.3-src.tar
- 广域网质量监控软件，VoIP Monitor 模块帮助您测定与跟踪最需要关注的广域网语音质量性能。-VoIPmonitor is open source network packet sniffer with commercial frontend for SIP SKINNY RTP and RTCP VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on
sipdroid
- android上开源voip代码，对了解sip很有帮助-voip android open source code is helpful for understanding sip
yate-5.3.0-1.tar
- yate软电话，在linux和unix下的VOIP SIP 协议软件电话工具-Yate soft phone, VOIP SIP protocol under Linux and UNIX software phone tools
vmeet
- sip 服务端 voip 电话视频 调试用的-sip server
libeXosip2-4.1.0.tar
- 基于osip库实现的exosip源码，进行了api接口的封装，方便使用完成sip相关的开发工作，不用再复杂的使用osip底层接口。-exosip src code,you can do something like ims\voip using it，
voipmonitor-11.7.7-src.tar
- VoIPmonitor 是一个开源的实时网络抓包并分析SIP和RTP protocol的程序. -VoIPmonitor is open source live network packet sniffer which analyze SIP and RTP protocol. It can run as daemon or analyzes already captured pcap files. For each detected VoIP call voipmonitor ca
sipeksdk
- Sipek Softphone is an open source project that is intended to share common VoIP software design concepts and practices. Beside that it s a simple and easy-to-use SIP softphone with many useful features. New version released - Sipek Softphone v0.