搜索资源列表
Ulive20081113s
- 用C++写的流媒体程序库,实现了标准协议议。例如RTP/RTCP,RTSP和SIP -Written in C++ streaming media process library that implements the standard protocol proposed. Such as RTP/RTCP, RTSP and SIP
linphone-test
- Windows下C语言调用libliphone DLL 进行SIP呼叫的注册,主叫和被叫测试示例程序。自带linphone的配置文件。-A test program of SIP register/outgoing/incoming call demo using liblinphone DLL under windows platform. Attached with linphone configuration file.
uPJSSIP_test_s
- 使用开源SIP协议栈开发的一个小测试程序源码,包含UA与UAS两部分,完成最基本的publish会话。注:简单易懂起见,publish没有设置消息体。PJSIP,一个用C开发,功能强大大的开源SIP协议栈,使用用优秀的内存分配机制开发,运行速度快,支持IM、Presence、PIDF、rPIDF等最新RFC,能用来实现各种SIP应用。PJSIP下载见本站。 -The open source SIP stack developed a small test program source co
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
OSIP
- SIP的一种协议栈开源代码,纯C写的,linux下轻松编译通过,生成可用的库-A SIP protocol stack open source code, written in pure C, linux under easily compile, generate usable libraries
sip_UAC_UAS
- 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 transfe
live.2012.02.29.tar
- Live555 是一个为流媒体提供解决方案的跨平台的C++开源项目,它实现了对标准流媒体传输协议如RTP/RTCP、RTSP、SIP等的支持。Live555实现了对多种音视频编码格式的音视频数据的流化、接收和处理等支持,包括MPEG、H.263+、DV、JPEG视频和多种音频编码。同时由于良好的设计,Live555非常容易扩展对其他格式的支持。-Live555 is a streaming cross-platform solutions C++ open source project, whi
live.2011.06.12.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-Live555 is a written in c++, based on open standards protocol RTP/RTCP, RTSP, SIP multimedia streaming source code library. In the library can build a variety of protocols of
Net
- SIP协议栈,非常少的c#版的,类似c++的pjsip是开源的,同时兼容了底层的stun,upnp,mail,ftp等协议栈-sip stack in c# version ,it is like pjsip ,it is open source in c#,including sip mail stun upnp nat and so on
live555
- Live555是一个为流媒体提供解决方案的跨平台的 C++开源项目,它实现了对标准流媒体传输协议如RTP/RTCP、RTSP、SIP等的支持。-(See also the "LIVE555 Proxy Server".) LIVE555 Streaming Media Source-code libraries for standards-based RTP/RTCP/RTSP/SIP multimedia streaming,
resiprocate1.8.10
- 用C++语言写的最新开源SIP协议栈源码 ,版本号为resiprocate1.8.10,编译环境为vs.net 2005, 2008, 2010-Using C++ language to write the latest open source SIP protocol stack source code, the version number is resiprocate1.8.10, build environment for vs.net 2005, 2008, 2010
live555-latest.tar
- Live555 是一套以 C++ 實作的Multimedia Streaming library,這套函數庫實做了RTP/RTCP/RTSP/SIP 等標準協議,支援MPEG, H.264, H.263+, DV, JPEG 等格式。- Live555 is a set of C++ implemented Multimedia Streaming library, this library implements something RTP/RTCP/RTSP/SIP and othe
live.2013.10.25.tar
- ive555 是一个为流媒体提供解决方案的跨平台的C++开源项目,它实现了对标准流媒体传输协议如RTP/RTCP、RTSP、SIP等的支持-ive555 is a streaming media solutions for the cross-platform C++ open source project, which implements a standard streaming protocols such as RTP/RTCP, RTSP, SIP and other support
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
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.
baresip-0.4.10.tar
- 用c语言实现网络语音通话,用到了sip协议-Implementation of network voice communication with the C language, uses the SIP protocol
sipp-3.3.tar
- sipp最新源代码,可供C++开发 适合linux下IMS、软交换通信开发-for sip test and develope
SIP_CallOut_cSharp
- SIP通讯协议DEMO C#开发 想学SIP的快下载看看吧-Independentsoft.Sip Source. SIP .NET is Session Initiation Protocol API for .NET Framework and Mono. The API is written in 100 managed C# code.
pjproject-2.4
- PJSIP是用C语言编写实施基于标准协议,如SIP , SDP , RTP , STUN , TURN , ICE和一个自由和开放源码的多媒体通信库。它结合了信令协议( SIP)具有丰富的多媒体框架和NAT功能集成到高级别API,几乎适用于任何类型的系统,从台式机,嵌入式系统,移动手机。-PJSIP is a free and open source multimedia communication library written in C language implementing stand
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.