搜索资源列表
bayonne-1.1.3.tar
- 国际上著名的开源项目Bayonne IVR,完全基于电信的语音交互式系统应用.内含源码.-internationally renowned open-source projects Bayonne IVR entirely interactive voice telecommunications systems applications. Intron source.
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.
speex-1.0.5.tar
- 一个非常好的开源音频编解码项目，支持多种音频采用频率，支持多码流，支持可变速率 Speex a free codec for free speech Speex is an Open Source/Free Software patent-free audio compression format designed for speech. The Speex Project aims to lower the barrier of entry for voice applications
MyNetMeeting
- 我的论文项目，利用netmeeting实现视频会议功能
sipserver
- 在项目中应用的exosip开发的一个使用例子，易于初学者
RTP_Presenter
- 采用MS在ConferenceXP项目中使用的RTP组件实现.NET平台下的RTP音视频传输的功能。
linphone-2.1.1.tar
- linphone源代码，基于osip、ortp协议栈的开源项目
openh323proxy-0[1].9.13.tar
- 是Opengatekeeper项目的扩展。它支持通过RTP（音频和视频）网守的路由和T.120通讯，因此没有通讯直接在endpoints间进行。-Opengatekeeper project is the expansion. Through its support for RTP (audio and video) Gatekeeper routing and T.120 communications, there is no direct communication between the
vp2p.rar
- 一个P2P语音通信的项目,经过测试,效果不错,而且占用带宽也小,A P2P voice communication projects, tested the effect of good, and the bandwidth occupied by a small
Skype4Java.rar
- 两个文件夹均是Eclipse下的Java项目，运行Eclipse后分别导入这两个工程就可以看运行结果。如何运行AP2APTest请查看开发指南.,Are two folders of Java under the Eclipse project, Eclipse after running into these two projects, respectively, can be seen on the results. Please check AP2APTest how to run the
H264CODEC.rar
- 公司以前的一个项目，里面包含了H264的编码与解码，还附带一个RTP传输例子，极力推存下载,Company' s former project, which includes the H264 encoding and decoding, but also comes with an example of RTP transport, strongly pushed deposit download
Analysis_of_OpenH323_source_code
- OpenH323是广泛应用的H323开源协议栈，该文档通过对一个典型的EndPoint与Gatekeeper之间进行通信的过程进行分析，来向初学者阐述OpenH323的运作机理，对于基于OpenH323进行开发的项目有重要参考意义。-OpenH323 is widely used open source protocol stack, this documents describe a typical process that how a connection established and d
jrtplib-3.9.1
- 2011年最新rtp通信开源项目源码，适合任何平台，目前此项工作历经近十年的开发，已经基本成熟，近一步的研究推进也比较困难，对于程序员是最后的资源。-The library offers support for the Real-time Transport Protocol (RTP), defined in RFC 3550. It makes it very easy to send and receive RTP packets and the RTCP (RTP Control Pro
webrtc
- google 开源项目 webrtc源码-google open source webrtc source
pa_snapshot_v19.tar
- pro audio 驱动，在很多跨平台项目使用到。特别是VOIP类。-pro audio drivers, in many projects cross-platform use. VOIP category in particular.
void_communication
- 这也是一个语音通信的项目,语音那一块是封装过的,不过有源码,可以修改-This is a voice communications, voice, and that is one-off package, but there is source code, you can modify
xml
- 不错的基于IP网络语音程序，可单独使用也可做为子程序加入已有项目中-Good voice procedures based on IP network can be used alone as a subroutine can also join the existing project
standard_voip_register_flow
- 作为某个成熟voip产品的注册模块的符合rfc的全部流程，如果想自己搞项目应该有参考价值-as some mature voip product s register module base on sip protocol,no implement for copyright reason
BasicCall
- 针对Sip Proxy的多路压力测试程序.本压力测试基于ReSIProcate 1.5版本的basicCall项目修改，并且在ReSIProcate 1.5的Proxy测试通过，理论上应该也能针对其它Proxy进行压力测试。测试的时候请下载ReSIProcate，将basicCall.cxx覆盖到\resiprocate-1.5\resip\dum\test-Sip Proxy for Multi-channel pressure test procedure. The stress tests
VESampler
- 基于voiceelements开发的网络电话测试项目-voiceelements