- 浏览: 231333 次
- 性别:
- 来自: 南京
最新评论
-
baby8117628:
vc下mp3 IDv1和IDV2的读取 -
gezexu:
你好,我按照你的步骤一步步进行但是安装libvorbis的时候 ...
linux如何搭建强大的FFMPEG环境 -
ini_always:
帅哥,转载也把格式做好点,另外出处也要注明一下吧。。。
MP3文件格式解析
VoIP bookmarks from Klaus Darilion
Below you will find descriptions and links to SIP and RTP stacks, applications, test utilities, SIP proxies, SIP PBXs and STUN server and clients. Most of them are open source :-), but not all of them :-(
If you have any comments please feel free to contact me: --> klaus.darilion at pernau.at <--
There are also other VoIP related portals and link collections .
Note: I mainly searched for C/C++ stacks and applications. There also exist a lot of stacks and applications for other programming languages, especially for java. If you are looking for Java stacks/applications, please ask Google (search for: NIST java jain).
RTP Stacks (mainly open source C/C++ stacks)
* jrtplib : A very nice, simple C++ RTP stack. Works on Windows, Linux.... ; License: Free; Homepage: http://lumumba.luc.ac.be/jori/jrtplib/jrtplib.html . This stack is not symmetrical, but you can use my version of rtpconnection.cpp (for jrtp version 2.8) to make it symmetrical. (send RTP and receive RTP on the same port, send RTCP and receive RTCP on the same port).
* Common Multimedia Library : from UCL London, includes RTP stack; C; License: Free; Homepage: http://www-mice.cs.ucl.ac.uk/multimedia/software/common/
* ortp : C; License: LGPL ; Homepage: http://www.linphone.org/ortp/ ; without RTCP, from linphone
* GNU ccRTP : C++; License: GPL (with linking exception ); Homepage: http://www.gnu.org/software/ccrtp/
* LIVE.COM Streaming Media : C++; License: LGPL ; Homepage: http://live.com/liveMedia/
* Morgan RTP DirectShow Filters : C++; License: ?; Homepage: http://www.morgan-multimedia.com/RTP/ ; based on liveMedia library
* RTP from vovida.org : C++; License: VOCAL ; Homepage: http://www.vovida.org/protocols/downloads/rtp/
* RTPlib : RTP library from Lucent Technologies/Cloumbia University; C; License: Non-exklusive source code license ; Homepage: http://www-out.bell-labs.com/project/RTPlib/
* librtp : C; License: GPL ; Homepage: http://gphone.sourceforge.net/template.php3?page=librtp ; from Gnome-o-phone
* Microsoft RTC API : The Mircosoft RTC API is a high level SIP and RTP Stack. It's included in Windows XP and also comes with the several Windows Messenger. Version 1.2 introduced a lot of new features is behaves strange when used with other SIP clients. Developer Homepage: http://www.microsoft.com/downloads/details.aspx?FamilyID=ae0bdc75-9f2f-4217-b97f-dfa0adf264aa&displaylang=en .
* sipXmediaLib : Part of pingtel's open source releases for VoIP. License: LGPL ; Homepage: sipfoundry.org .
SIP Stacks
external SIP stack comparison
* dissipate : C++; Linux, requries the qt-library, License: GPL ; Homepage: http://www.div8.net/dissipate/ ; The original dissipate by Billy Biggs.
* dissipate2 : C++; Linux, requries the qt-library, License: GPL ; Homepage: http://www.wirlab.net/kphone/ ; A enhanced dissipate, is part of the kphone distribution.
* GNU osip : C; Linux+Windows+...; License: LGPL ; Homepage: http://www.gnu.org/software/osip/ ; Also known as libosip. Note: The interface of osip has been changed and from now on it will be called osip2! Download the tar file from http://osip.atosc.org/download/osip/ .
* 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 extensions. Once completed, this eXtended library should provide an API for call management, messaging and presence features.... Download the tar file from http://osip.atosc.org/download/exosip/ .
* SIP from vovida.org : C++; Linux+Windows+...; License: Vovida Software License ; Homepage: http://www.vovida.org/protocols/downloads/sip/
* resiprocate : C++; Linux+Windows+...; Includes now a high level API (DialogUsageManager) which supports refers, ... License: VOCAL ; Homepage: http://www.sipfoundry.org/reSIProcate/ .
* Microsoft RTC API : The Mircosoft RTC API is a high level SIP and RTP Stack. It's included in Windows XP and also comes with the several Windows Messenger. Version 1.2 introduced a lot of new features is behaves strange when used with other SIP clients. Developer Homepage: http://www.microsoft.com/downloads/details.aspx?FamilyID=ae0bdc75-9f2f-4217-b97f-dfa0adf264aa&displaylang=en .
* sipXtackLib : Part of pingtel's open source releases for VoIP. License: LGPL ; Homepage: sipfoundry.org . There is also a high level call library (sipXcallLib ), which implements JTAPI in C++.
* libmsip : A C++ SIP stack for Linux developed for the miniSIP project. Homepage: http://www.minisip.org/libmsip/ .
RTP Applications
* RAT - Robust Audio Tool ; Supports a large number of codecs, ... License: Free; Homepage: http://www-mice.cs.ucl.ac.uk/multimedia/software/rat/
* JMF - Java Media Framework : Can receive and send RTP streams; Homepage: http://java.sun.com/products/java-media/jmf/
* MP3/RTP Plugin for Winamp : Homepage: http://www.live.com/multikit/winamp-plugin.html
* Vomit - Voice over Missconfigured Internet Telephones: Plays back captured voice conversation; Homepage: http://vomit.xtdnet.nl
* RTP Tools : Several RTP utilities from the Columbia University; Homepage: http://www.cs.columbia.edu/IRT/software/rtptools/
* UDP Packet Reflector/Forwarder : A tiny tool which forwards or reflects UDP packets. You can also add delay and packet loss. Very useful if you want to test RTP applications. Homepage: http://www.cs.ucl.ac.uk/staff/s.bhatti/teaching/z02/reflector.html . As I was not able to compile this tool I searched and found a binary somewhere in the web. You can download it local
SIP Phones (SIP User Agents)
* x-lite, x-pro : A SIP client for Windows; Mac OS and Windows CE, http://www.xten.com/ . A really nice SIP UA with a lot of features. The light version is free and really rocks , the pro version not. Supports multiple proxies.
* eyeP Phone Lite : A SIP client for Windows, a FWD version is available for free http://www.eyepmedia.com/eyePPhoneFWD.htm .
* SIPPS : SIP softphone with answering machine and a lot of features. They have also integrated support for nikotel.com for SIP-PSTN termination.http://www.sippstar.com/ . A Demo for testing is available. The configuration is a bit weird (what's the difference between a proxy and a redirect server?).
* MSN Messenger : Microsofts Messenger, Version 4.6 allows also connections to other SIP servers than microsofts one. Nice design, works very well. Can be used with the SIP service of iptel.org. Homepage: http://messenger.microsoft.com ; local download of Version 4.6 for Windows NT (2000).
* MSN Messenger : Microsofts Messenger, Version 4.7 allows also connections to other SIP servers than microsofts one. Nice design, works very well. Can be used with the SIP service of iptel.org. Homepage: http://messenger.microsoft.com ; local download of Version 4.7 for Windows XP.
* Microsoft portrait : Windows SIP client that supports Audio, Video and IM. Uses RTC API 1.2 and therefore has poor compatibility with other SIP clients.http://research.microsoft.com/~jiangli/portrait/ .
* Ubiquity User Agent : Java based SIP Client for Windows, very useful, you have to register (free) to get an license; Homepage: http://www.ubiquity.net/useragent.php
* EZ-Phone (Evaluation Version) : SIP Phone for Windows; Homepage: http://www.hssworld.com/voip/download.htm
* MySIP : SIP User Agent from Siemens; Homepage: http://www.mysip.ch/
* SJPhone : SIP and H.323 Softphone for Windows, Linux and PocketPC from: http://www.sjlabs.com/ . The configuration for SIP is a little bit tweaky. And there must not be another SIP client running on port 5060 or the SJPhone won't work.
* Linphone : A SIP Softphone for Linux (GNOME), needs libosip ans oRTP; Homepage: http://www.linphone.org/
* KPhone : A SIP Softphone for Linux (KDE); Homepage: http://www.wirlab.net/kphone/index.html
* Vovida : Complete SIP Suite for Linux (Uaser Agent, Proxy, ...), very, very big software contruct; Homepage: Vovida.org
* Siphon : Linux SIP Softphone; Homepage: http://siphon.sourceforge.net/index.html
* ActXPhone : An ActiveX-Control SIP Softphone based on the Microsoft Real Time Communications (RTC) API.http://www.pernau.at/kd/voip/ActXPhone/ .
* sipXphone : Part of pingtel's open source releases for VoIP. License: LGPL ; Homepage: sipfoundry.org . This softphone also requires lots of other libraries from the sipX... software at sipfoundry.org.
* Shtoom : An open source, cross plattform SIP client written in Python. License: LGPL ; Homepage: http://www.divmod.org/Home/Projects/Shtoom/index.html .
* Cornfed SIP-UA : A SIP user agent for Linux. License: Free for non-commercial use (binary distribution); Homepage: http://www.cornfed.com/products/ .
* MiniSIP : An open source SIP user agent for Linux which runs on PDAs. It is based on several libraries, including libmsip, a C++ SIP stack. Homepage: http://www.minisip.org/index.html .
SIP Test Utility
* sipsak : SIP Swiss Army Knife, very useful test utility (Linux); Homepage: http://sipsak.berlios.de/
* SIPNess : Ortena Networks SIP Messenger, very useful test utility for windows; Homepage: http://www.ortena.com/download.htm
* SIP request generator : A web based generator of SIP requests: send SIP requests to SIP UAS and waits for final response: Download at http://obelix.ict.tuwien.ac.at/sip-gen/sip-gen.zip or test it online at Download at http://obelix.ict.tuwien.ac.at/sip-gen/sip-request-gen.php
* Nastysip A simple Linux-program from SX-Design that generates bogus SIP-messages and sends them to any peer. Download at http://www.sxdesign.com/index.php?page=developer&submnu=nastysip .
* sipXtest : Part of pingtel's open source releases for VoIP. License: LGPL ; Homepage: sipfoundry.org .
* SIP Forum Test Framework (SFTF) : A Framework to test SIP devices for common errors. License: GPL ; Homepage: sipfoundry.org .
* callflow : a powerful SIP call flow visualizer; Homepage: http://callflow.sourceforge.net/ .
* SIP Scenario Generator : a powerful SIP call flow visualizer; Homepage: http://www.iptel.org/~sipsc/ .
* SIPp : a powerful SIP performance testing tool sponsered by HP; Homepage: http://sipp.sourceforge.net/ .
SIP Applications (Proxy, Location Server)
*
Sip Express Router (ser)
: Highspeed GNU SIP proxy with a lot of features and a lot of ongoing development. Homepage: http://www.iptel.org/ser/ . A really cool SIP proxy - I like it! You can also take a look at the development homepage with web CVS. At the beginning you should read the admin guide and the mailing lists archive .
*
Ser Media Server (sems)
: Media Server add-on for ser SIP proxy. Homepage: http://sems.berlios.de/ . Supports voicemail, IVR, SIP/PSTN gateway ...
* Asterisk : Linux Software PBX with Gateway, SIP Proxy, Gateway (SIP, H.323, PSTN, ...); Homepage: http://www.asteriskpbx.com/
* sipd : A Linux SIP proxy from SX-Design written in C (GPL): http://www.sxdesign.com/index.php?page=developer&submnu=sipd
* partysip : A Linux SIP proxy based on osip2 (LGPL). Developer homepage is at: http://savannah.nongnu.org/projects/partysip/ , you can download tar packages from: http://osip.atosc.org/download/partysip/ .
* mysip : A SIP proxy server from Siemens for Windows platforms. Homepage: http://www.mysip.ch/
* Fomine RTC server : A SIP proxy server for Windows which uses its own SIP stack (does NOT need the RTC API) Homepage: http://www.fomine.com/rtc-server.html . The unregistered version can be used up to 5 users.
* sipXpbx : Part of pingtel's open source releases for VoIP. License: LGPL ; Homepage: sipfoundry.org . This PBX combines various sipX applications like a SIP proxy (sipXregistry, sipXproxy), a media server (sipXvxml) and lots more.
* yate : Yet Another Telephony Engine - a PSTN gateway. License: GPL ; Homepage: yate.null.ro . This gateway supports H.323, SIP and zaptel (->asterisk) based PSTN cards.
STUN server and clients
* mystun : STUN server and client library from the iptel.org guys. License: GPL, Homepage: http://developer.berlios.de/projects/mystun/ . You have to download the file via CVS .
* Vovida STUN server : STUN server and client library/application for Linux and Windows from the Vovida guys. License: Vovida Software License 1.0, Homepage: http://www.vovida.org/applications/downloads/stun/ . The files are hosted at sourceforge .
NAT traversal ALG (application level gateway)
This applications can be installed on a linux NAT-box. They will rewrite your SIP messages and have some kind of UDP/RTP proxy for the media stream.
* SaRP - SIP and RTP proxy : Perl implementation, License: GPL, Homepage: http://sourceforge.net/projects/sarp/ .
* siproxd : Siproxd is a proxy/masquerading daemon for the SIP protocol based on osip. License: GPL; Homepage: http://sourceforge.net/projects/siproxd/ .
发表评论
-
vc下mp3 IDv1和IDV2的读取
2010-01-25 10:52 2378/*这是修改后的代码,VC下读ID3v2 & ID3v ... -
使用ffmpeg为库编写的小型多媒体播放器源代码
2010-01-21 16:52 4332今天突发奇想,就在以前音频播放器(详细情况请看这里——http ... -
ffmpeg提取音频播放器总结
2010-01-21 16:31 5962ffmpeg提取音频播放器总 ... -
ffmpeg开发指南
2010-01-20 17:26 3370ffmpeg 中的Libavformat 和 li ... -
linux下安装ffmpeg过程
2010-01-18 15:48 1882最近互联网视频共享的 ... -
【PNG overview】PNG专题!
2010-01-18 13:39 3345【PNG overview】PNG专题! 作者 鼯鼠 ... -
Big Endian 和 Little Endian
2010-01-18 13:29 1542Peter Lee 2008-04-20 一、字节序 ... -
MediaInfo开源工程
2010-01-18 13:22 2353一、简介 MediaInfo 用来 ... -
MP3文件格式解析
2010-01-18 10:58 3540MP3文件格式解析 Peter Lee 2008-06-0 ... -
LAME-mp3
2010-01-18 10:40 2009LAME - 压缩 MP3 的最佳利 ... -
FLV文件格式分析(图示讲解的清楚)
2010-01-14 15:56 5090FLV是一个二进制文件, ... -
我对FLV 文件格式的理解
2010-01-14 15:52 3345我对FLV 文件格式的理解 ----------------- ... -
常用的音频文件介绍
2010-01-13 10:56 1364MP3全称是动态影像专家压缩标准音频层面3(Moving Pi ... -
RTSP客户端的JAVA实现
2010-01-12 16:12 8233参考资料 1. 《RTSP简单命 ... -
国外嵌入式、音视频处理等重要网站
2010-01-08 10:07 2017嵌入式方面: 1.关于嵌入式开发的站点,提供非常多关于嵌入 ... -
RTSP点播——消息流程实例
2010-01-08 09:44 5095RTSP点播消息流程实例(客户端:VLC, RTSP服务器:L ... -
live555代码解读之三:SETUP和PLAY请求消息处理过程
2010-01-08 09:43 3440SETUP请求消息处理过程 ... -
live555代码解读之二:DESCRIBE请求消息处理过程
2010-01-08 09:42 3775ve555代码解读之二:DESCRIBE请求消息处理过程 ... -
live555代码解读之一:RTSP连接的建立过程
2010-01-08 09:42 4430TSPServer类用于构建一个RTSP服务器,该类同时在其内 ... -
live555源代码概述
2010-01-08 09:41 3861述 liveMedia项目(http://www ...
相关推荐
kamailio (OpenSIPS)是一个成熟的电信级SIP Server平台,可广泛应用于SIP应用的路由分发、负载均衡,可用于搭建SIP代理,提供SIP注册服务等。而且目前OpenSIPS自身也提供SIP Presence以及IM功能。...
开源voip计费系统,支持freeswitch、kamailio等sip通信场景
一款基于C#、WPF、Ozeki.Voip 的软电话,支持在一个软件中管理多个分机,单软件支持自动呼叫或者自动接听功能。相关源码参考博客:http://mp.blog.csdn.net/postedit/79281818
本代码包含基本的VoIP 技术实现的代码和技巧
OPAL工程是一个集合了OPENH323和SIP协议的开源VoIP协议,上面给出的是OPAL工程的源码
Mumble 是一个用于游戏中语音沟通VoIP的开源项目,包含服务端和客户端程序
开源VoIP软件MjSip 用来拨打接听互联网电话 本文档对其进行基本的架构分析,解决了一些软件自身的常见问题 初次接触VoIP的朋友可参考 使用MjSip做开发的朋友也可使用
VoIP资料VoIP资料VoIP资料VoIP资料VoIP资料VoIP资料VoIP资料VoIP资料VoIP资料VoIP资料VoIP资料VoIP资料VoIP资料VoIP资料VoIP资料VoIP资料VoIP资料VoIP资料
很多人希望了解是否可能构建企业级开源VoIP解决方案以及这样做是否有好处。答案是这不仅可能,而且Asterisk的独到之处在于它是一个开源IP电话平台。Asterisk不仅作为一个IP呼叫信号服务器(有时称作IP PBX),而且它...
VoIP服务器----Asterisk,从实现电话免费大餐、免费的VOIP服务、Asterisk的安装和配置等深入讲解
SipDroid是一款开源的SIP/Voip客户端软件,针对Android手机开发平台。它是通过SIP provider来提供电话通信服务的,在它的最新的版本1.5.5beta中也提供了视频通话的服务。 它运行在标准的SIP协议之上,由于...
对于企业来说,通过ADSL线路和VoIP设备实现语音通信是最为廉价的接入方式,其实现质量和安全性方面也没有政府、军队等系统的要求高,所以企业更愿意选择ADSL+VoIP的语音解决方案。
很好用的开源网络电话拨打软件,开源的,免费的,好用的
详述VOIP基础详述VOIP基础详述VOIP基础详述VOIP基础详述VOIP基础
jPBXLite是VoIP / SIP PBX。 支持SIP扩展,语音邮件,中继,会议,队列(ACD)和IVR系统。 通过jPhoneLite / 1.4.0支持视频会议。 注意:此项目已重命名,现在为jfPBX。 请转到jfpbx.sourceforge.net
voip的课件及实验材料,内有开源软件介绍,自己手动配置voip哦亲。
Cisco+VoIP配置技术Cisco+VoIP配置技术Cisco+VoIP配置技术Cisco+VoIP配置技术Cisco+VoIP配置技术
Ozeki VoIP服务检查器是可用于检查VoIP服务可用性的工具。... 之所以开发此工具,是因为许多移动Internet... 该工具是开源的。 为不同的平台提供了源代码。 您可以自由使用,修改,开发或使用此工具做任何您想做的事情。
VoIP性能的优化要求几方面的能力,如对话音业务进行非常灵活的分类能力,保证最小带宽设置的能力,向网络上的其他业务类别提供出色的QoS保证的能力,以及提供真正的多业务支持的能力。在企业网中应用以IP技术为基础...
开源图鸭VoIP技术的服务器源代码及客户端网络模块代码。VoIP库、WIN/MAC/iOS/Android端demo以及文档均在图鸭科技官网的开发者板块中。服务端提供编译后可直接运行的代码,客户端部分仅提供视频通讯用SDK