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kphone-4.0.2.tar
- KPhone is a SIP (Session Initiation Protocol) user agent for Linux, with which you can initiate VoIP (Voice over IP) connections over the Internet. KPhone requires KDE (K Desktop Environment) or at least various libraries from the KDE package. KPhone
kphoneSI.tar
- 一个比较好的基于LINUX平台的网络电话，代码写很有特点，很值得一读-a better platform for Linux-based Internet phones, code written with unique features and is well worth a read
myicq-win_src
- MyICQ是一套完全公开源代码的即时通讯软件，包括服务器端和客户端，可以用于互联网或局域网中。可以运行在Windows或Linux(KDE/Qt)操作系统上，这是Windows的客户端。MyICQ完全支持服务器之间的松散连接，这意味着任何人都可以在Internet上架设自己的服务器。群组功能使您能够享受到聊天室和网络游戏所带来的快乐。 --------------- 基本功能特点 --------------- 采用MySQL数据库 日志文件 配置文
libosip2-2.0.6
- SIP Stacks libmsip: A C++ SIP stack for Linux developed for the miniSIP project. Homepage: http://www.minisip.org/libmsip/.
kphone
- SIP Stacks dissipate2: C++ Linux, requries the qt-library, License: GPL Homepage: http://www.wirlab.net/kphone/ A enhanced dissipate, is part of the kphone distribution.
eXosip-0.7.6
- SIP Stacks GNU eXosip: C Linux+Windows+... License: GPL Homepage: http://savannah.nongnu.org/projects/exosip/ The extensible osip: \"...It aims to implement a simple high layer API to control the SIP for sessions establishements and common extensio
fobbit-0.61.tar
- Linux下开发的一款网络电话-Linux built a network phone
ser-0.8.14_src.tar
- 一个linux下的高速sip router-a sip of high-speed router
rtpproxy
- rtp relay,SIP中穿越NAT的一种方式，这是源代码，已经在red hat linux 9中调通过-rtp relay, cross SIP NAT a way, this is the source code, in red hat linux 9 emphasized through
JRTP
- 使用JRTPLIB库实现RTP的传输RTP是目前解决流媒体实时传输问题的最好办法，如果需要在Linux平台上进行实时流媒体编程，可以考虑使用一些开放源代码的RTP库，如LIBRTP、JRTPLIB等。JRTPLIB是一个面向对象的RTP库，它完全遵循RFC 1889设计，在很多场合下是一个非常不错的选择，下面就以JRTPLIB为例，讲述如何在Linux平台上运用RTP协议进行实时流媒体编程。-JRTPLIB use of RTP for achieving the transmission RT
jrtplib-2.8.rar 用C++写的实时传输协议RTP
- 一个外国牛人用C++写的实时传输协议RTP,源代码的编写风格很好，可读性很强。可以用于网络视频和音频的传输，enjoy it, A foreign Niu Ren real-time transmission agreement RTP which writes with C, the source code compilation style very good, the readability is very strong. May use in the network video fr
e-comm-0.7.tar
- 简单的局域网中用的IP电话程序-A simple IP phone program which used in LAN network
e-comm
- 代码来自sourceforge，在linux下的实时语音聊天程序，使用了adpcm编码，同时还使用了RTP实时传输协议，是一个很好的学习实时传输协议的程序。- The code comes from sourceforge, chats the procedure under the linux real-time pronunciation, has used the adpcm code, meanwhile has used the RTP real-time transmissi
uctimsclient1[1].0.12.tar
- ims sip客户端的实现，在linux以图形界面操作-ims sip client to achieve, in a graphical interface linux operation
yate2.tar
- 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
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
komssys-snap-19feb2007
- 出版《多媒体技术: 计算、通讯及应用(英文影印版)》牛人使用的RTP通信服务程序，可以用来做多媒体通信快速实验，只支持MPEG-1编码。服务器功能相当于IBM s VideoCharger. -The KOM(S) Streaming System provides a server, a client, and a proxy cache for audio/video streaming. The first and foremost encoding format that we use
Linux-audioavideo-SDK
- 一套不错的Linux音视频SDK，可用于二次开发，无回声问题，支持1080P高清视频通讯，采用了P2P技术，支持多平台互通，公开了代码跟开发文档，很适合做项目。-A good set of Linux audio and video SDK for secondary development, no echo problems, support 1080P HD video communications using P2P technology to support multi-platform
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.
nghbi
- Arrival process is a Poisson process, Optimization class contains several simple sample programs, PSS primary synchronization signal in the time domain simulation related.