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libcu30-1[1].0-1.src
- 高质量，底带宽，实时的视频会议软件，使用了一个Cornell大学的新算法。-high-quality, end-bandwidth, real-time video conferencing software, the use of a Cornell University's new algorithm.
gaim-1.5.0.tar
- linux平台下的一款Instant messaging软件，包含全部的源代码，configure/make之后即可运行。-under the new Instant messaging software, contains all the source code, configure / make can run after.
yate-1.1.0-1.tar
- 新一代电话引擎，有很强的扩展性，目前主要用于VoIP和PSTN-new generation of telephone engine is a highly scalable, mainly for VoIP and PSTN
RTCIncoming
- RTC下接收呼叫的例子（如果是VC用户，请新建一个APPLICATION 32空工程，然后，用ADD FILE的方法把所有的文件包含在工程中编译）-RTC receive calls example (If yes VC users, a new air APPLICATION 32 works, and then use the ADD FILE all the documents included in the project compiler)
SoftphoneA.zip
- VaxVoIP offers affordable software customization for our products. Whether you need simple changes or need new functionality, we offer services to meet your needs. If you are interested in custom software development, please contact customdev@vax
EchoCancellationinIPTelephoneSystem
- 分析了IP 电话系统中回声的特点, 并针对其回声具有路径延迟、延迟抖动大等特点, 提出 了一种基于延迟分析的自适应回声消除器设计的新思想和相关算法。介绍了自适应回声消除器的 基本结构及各个功能模块, 例如: 变阶数自适应滤波器, 语音状态检测器, 延迟分析器和NLM S 算 法控制器. 本文还详细论述了以TM S320C6201DSP 为硬件平台来实现能运用在IP 电话网关中的 自适应回声消除器及其在IP 电话系统中的连接关系。在文章的最后, 列举了对自适应回声消除器
sflphone-0.9.6.tar
- sip 软电话终端，新版本是基于pjsip(老版本是exosip)-sip soft phone terminal, the new version is based on pjsip (the old version is exosip)
shipingchuanshu
- 这篇文章是关于在VC中实现视频传输的一个创新思维，提出了一些新思想-This article is about the realization of video transmission VC in a creative thinking, put forward some new ideas
VOIP7
- 详细描述了基于SIP的VoIP复杂的安全问题，并且阐述了身份认证、加载MIME消息体 来扩展SIP消息和加密包等解决方法。提出一套新的安全框架，在链路层和网络层(交换机)、传输和会话层(防火墙)及应用层(基f-jl~．务器的软件等)多个层实施安全措施来根本解决诸多安全隐患。-A detailed descr iption of SIP-based VoIP complex security issues and expounded on authentication, loading MIM
kamailio-1.5.0-tls_src.tar
- Kamailio is a GPL licensed SIP server implementation. It started in 2005 as a fork of Fokus Fraunhofer SIP Express Router (SER) project. Kamailio wants to be a more open project, not only from license point of view, but more open as project managemen
CiscoVoIPKonfiguration
- Cisco的网络产品已经使世界发生了革命性的变化。准确地讲，Cisco网关、路由器和交换机构筑了Internet的主干网。每天都会有数亿美元的商业利润通过它们滚滚涌来。而近几年来发生的这些惊人变化还只是这场革命的开端。在网络短暂的历史进程中，最具活力的事件也许即将发生：把语音和数据技术结合成一种统一的传输模式。随着Cisco新的语音解决方案系列的发布，从传统的PBX语音解决方案到基于IP的语音解决方案的转移将会以指数级加速进行。 -Cisco' s networking product
opensips-1.5.2-tls_src.tar
- sip服务器原代码,要求在linux或者unix下安装,可以用于建立自己的voip系统,基于sip协议-OpenSIPS brings: robust and performant SIP (RFC3261) Registrar server, Location server, Proxy server and Redirect server small footprint- the binary file is small size, functionality can be stri
ns2voip-2.31-070906.patch
- voip for ns2.. you can add new traffic/VoIP
kamailio-3.0.2_linux_i386.tar
- Kamailio是一个开源的SIP服务器，原名OpenSER 该版本主要修复了代码中的一些小问题，完善了文档，建议使用 3.0 和 3.0.1 版本的用户升级。 -SIP Router (sip-router) is an industrial-strength, free VoIP server based on the Session Initiation Protocol (SIP RFC3261). It is engineered to power IP te
BroadVoice16OpenSource.v1.1
- BV16 codec Broadcam company generated. This file is new for 16 bit.
openmeeting
- 非常遗憾的是，OpenMeetings没有提供基于第三方SSO（单点登录，例如CAS）系统的集成方法。但它提供了一个Web Service，官网将其命名为SOAP-Gateway，亦即一个基于SOAP的网关。这个Gateway集成了一系列的方法，可以实现从OpenMeetings的Flash客户端（前台）之外完成登录、进入房间、以及部分后台管理（如添加房间、注册新用户等）功能。-Very unfortunately, OpenMeetings none provided based on thi
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.
openmcu-4.0
- OpenMCU-ru is an open-source Mutli Conference Unit using the H.323, SIP and RTSP protocols (software MCU). It is fork of the original OpenMCU with many new features. Server runs on Linux, Windows and FreeBSD.
bjrjy
- Several new methods sar image denoising, ML estimation method can be a good signal to noise ratio, DC-DC power single-part set-loop control.