搜索资源列表
sndrec
- 实现Linux环境下,录音与放音的程序,便于VOIP开发
sofia-sip-1.12.7.tar
- Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and person-to-person communicat
sofia-sip-1.12.6.tar
- Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and person-to-person communicat
openser-1.3.1-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
asterisk
- linux下面,sip,h.323代理服务器c语言原码,要在linux环境下安装,可以构建voip系统服务器
sofia-sip-1.12.9.tar
- Sofia SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and person-to-person communication
source
- 具有浮点指令的g.729语音压缩编码,符合ITU-T G.729 Annex C 规范,通过修改makefile可以支持windows/linux/solaris等操作系统,仅需要很少的更改就可以应用到voip领域
g729AnnexA
- g.729a语音压缩编码最新版本,符合ITU-T G.729 Annex A 2006年的规范,通过修改makefile可以支持windows/linux/solaris等操作系统,g.729a是g.729的简化复杂度后的结果,适合应用在嵌入式领域的voip实现。
KOMtraffgen100
- voip检测程序,能够在windows or linux运行-This is a voip detecting program,ti can run in windows or linux.
jrtplib-2.9
- 国外牛人用C++编写的voip的库文件,多媒体实时流传输协议(RTP/RTCP)lib包,平台要求(linux,unix,windows)用于网络音频视频传输-foreign cattle with a C voip prepared by the Treasury, the documents, multimedia real-time streaming protocol (RTP / RTCP) lib packages, platform requirements (Linux, Uni
sip协议的实现
- 详细说明: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。-By the SIP-based VOIP international standards organization launched Linphone RTP/RTCP realize. And provide a number of examples of the procedure, you can under
linphone-3.4.3.tar
- linux下的VoIP电话linphone的源码,最新版,编译安装测试过,可实现语音视频通话及文本聊天-VoIP phone linphone under linux source code, the latest version, compiled and tested and is available for voice calls and text chat with video
thesis-jori-original
- 国外牛人的毕业论文,英文版本,实时语音传输相关,其中的两个库文件,jrtplib与jvoiplib已经上传。enjoy -foreign elite s graduate paper in english.related to real time voice transimission.it has two library,jrtplib,jvoiplib.enjoy
tstone[1]
- VC++写的支持H.323和SIP的点对点网络电话VOIP系统源代码-TStone is p2p voip system.It can run windows or linux. It support h.323 and sip signaling.It is writed by c/c++ code.
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
demo_uas
- linux下的voip服务端uas的代码,有助于理解sip,希望对研究人员有用-voip under linux server uas the code helps to understand the sip, the researchers hope to be useful
asterisk-1.2.35.tar
- asterisk-linux下的安装文件-asterisk-sip voip
openam-v1_4_0_LINUX.tar
- LINUX系统下的基于H323的VOIP网络电话,需要预先安装H323标准库和PTLIB库函数才可以运行,运行在LINUX下 -LINUX systems of VOIP H323 based VoIP, requires pre-installed libraries and PTLIB H323 standard library functions can run, running under LINUX
t38modem-v1_0_2_LINUX.tar
- LINUX下的基于H323的VOIP网络电话,需要预先安装H323标准库和PTLIB库函数才可以运行,运行在LINUX下 -LINUX H323 under the VOIP network telephone based on the need to pre-install the H323 standard library and PTLIB library functions can run, running under LINUX
first_voip
- voip电话 实现局域的通话和无线情况下的通话,开发环境在linux下-voip phone calls to achieve local and wireless phone case, the development environment under linux