VOIP开源项目源码地址

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
NastysipA 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/.


(c) Klaus Darilion 2004
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Android版VoIP客户端App代码,用以实现经典的语音及视频通话功能,开发环境的配置下载下来SipDroid代码后,不能直接导入到eclipse中,否则是无法使用和编译的。   首先需要通过command line参照工程目录下的BUILT.txt文件对工程进行update。   如:android update project -p . -t 2(这里 -p指的是 --path -t指的是 --target)   由于是开源软件,需要版本控制的服务器,可以安装Subclipse(可以参见官网)   这样就可以将工程import到eclipse中了。   整理一些Android视频通话技术开发的资料:   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Android内核本身没有提供SIP协议栈,所以SipDroid软件中通过JNI直接实现了SIP协议栈。它能够在如下一些网络中提供VoiP服务:WLAN、3G、EDGE网络。甚至能够和Skype、Google Talk, MSN, Yahoo, AIM, and ICQ等用户进行通话。但前提必须到一些SIP provider注册账号(有免费的,也有收费的),SipDroid强烈建议使用PBXes这个sip provider来得到服务。   对于SipDroid实现视频通话的原理,目前的理解是:发送方是将是将视频数据传输到视频流媒体服务器,然后再传输给接收方的。具体细节目前不是很清楚,因为涉及到许多通信协议。   本项目共250 个目标文件,还有jni文件下c 文件需要编译。 感兴趣的同学必须得好好研究研究。

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