搜索资源列表
rat-4.2.25.tar
- Linux下的voip通信終端軟件,common目綠為多媒體開發包,主要提供內存mbus,md5,hmac,網絡接口方面的函數.rat為主要的函數集.-under Linux software voip communication terminals, common green head for multimedia development kits, main memory to provide legal, md5, hmac, network interface functions. ra
ser-0.9.4
- 用来作为linux中SIP SERVER,完成VOIP网络电话中服务器的功能-used as a SIP SERVER complete VoIP telephone network server functions
intercom-0.4.1
- 流行IP电话源程序,内含G.711、G.726、GSM、iLBC、回声消除等源代码,Linux平台C语言-popular source IP telephony, containing G.711, G.726, GSM, VoIP, echo cancellation, etc. source code, C language Linux
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
sipp
- voip压力测试工具源代码,sipp功能非常强大,可以对任何类型的sip服务器进行压力测试。该程序支持linux和win32。-1/1/2006 pressure testing tools source code, sipp is very powerful, can sip any type of server stress testing. The procedure support Linux and win32.
ser-0.9.6_src.tar
- SIP Express Router, Linux下的SIP代理服务器,小巧实用,开发测试VoIP设备和应用的必备.-SIP Express Router, Linux under the SIP proxy server, Small practical, the development of VoIP test equipment and the necessary application.
kphoneSI_1.0.2.tar
- KphoneSI (kpsi) is a SIP (Session Initiation Protocol) user agent for Linux, with which you can initiate VoIP (Voice over IP) connections over the Internet, send Instant Messages, subscribe your friends presence information and start othe
sip_voip_sample
- linux下的sip voip程序,这个是日本人开发的,对开发sip voip程序十分有参考意义-under linux sip 1/1/2006 procedures, the development of the Japanese, development of procedures sip 1/1/2006 very reference
ortp-0.13.1.tar
- 由基于SIP协议的国际VOIP组织linphone推出的标准RTP/RTCP实现。并提供了多个例子程序,可以在linux或者windows平台下实现对流媒体的传输与控制。
Freeiris 终端运营系统 voip soft
- 简单的画面,简单的架设方式,稍懂linux的人都可架设成功的 voip终端运营系统
基于exosip的UA结合ORTP实现的VOIP通信
- 基于eXosip实现UA之间通信; 基于ORTP实现RTP语音通信; 基于LINUX声卡采集语音; 基于播放器播放WAV声音文件
pjproject-1.0-rc2.rar 基于sip协议的VoIP、视频会议源码
- 基于sip协议的VoIP、视频会议源码,对防火墙穿透协议支持较好,跨平台,支持Linux/Unix,Wingdows,CE,Symbian等,Sip agreement based on VoIP, video conferencing source of support to better penetrate the firewall, cross-platform support for Linux/Unix, Wingdows, CE, Symbian, etc.
arm-voip.tar
- 嵌入式linux voip 语音电话 2440ARM qt图形界面 可以拨号-Embedded linux voip voice dialing 2440ARM qt graphical interface
speex
- speex1.2.beta3最新源码,支持网络语音压缩和回声抵消,应用与VOIP,内附源码和中文使用说明。-speex1.2.beta3 the latest source code, support for voice compression and echo cancellation network, application and VOIP, containing source code and instructions in Chinese.
ARMDSPVoI
- 利用ARM与DSP相联系,实现VoIP电话系统的安全可靠链接-Use ARM and DSP linked to VoIP phone systems to achieve safe and reliable link
658jrtplibmediaplayer
- 利用rtp库实现实时语音传送,是做voip的必备协议-Rtp library using real-time voice transmission, is so essential agreement voip
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
opal-3.8.0
- opal3.8.0,VoIP协议库,支持sip,h.323等,2010-02-01最新发布,支持多平台,如windows、linux-opal3.8.0, VoIP protocol library, support sip, h.323, etc. From 2010-02-01-date releases, support multiple platforms, such as windows, linux
1652s_2
- The AT76C901 is highly integrated ASIC that can be used as a part of a wireless phone that utilizes an 802.11 LAN-based wireless medium and carries Voice over IP (VoIP) packets. Specified in this datasheet, an ARM® processor-based subsystem (
voip
- 在linux底下编写voip网络电话,输入对方的ip号可实现对话。还没有界面化。可能需要调试一些电脑硬件。-Prepared under the linux voip Internet phone, enter the other' s ip number can be achieved dialogue. No interface of. May need to debug some computer hardware.