转自:http://blog.chinaunix.net/uid-11857489-id-2814490.html
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/.
相关推荐
随着VoIP(Voice over Internet Protocol)技术和下一代网络(NGN)的发展,通信领域正经历从H.323标准向SIP(Session Initiation Protocol)标准的转变。SIP作为一种更简单、更灵活的协议,相比H.323具有诸多优势,...
开源voip计费系统,支持freeswitch、kamailio等sip通信场景
该开源项目是针对大型VoIP和实时通信平台的SIP服务器kamailio的设计源码,由6692个文件构成,主要使用C语言编写,辅以Shell、Python、C++、Java、PHP、HTML、JavaScript、Lua、CSS、Ruby和C#等多种编程语言。...
Android SipDemo是一个官方提供的示例应用,用于演示如何在Android平台上使用SIP(Session Initiation Protocol)进行VoIP(Voice over IP)通信。这个项目对于开发者来说,尤其是那些想要在自己的应用中集成语音...
源地址为https://www.microsip.org/downloads,这是一个轻量级的开源软件,注册后可实现voip电话
1. **WebRTC**: Web Real-Time Communication(WebRTC)是一种开源项目,由Google维护,旨在提供浏览器和移动应用程序之间的实时通信能力。WebRTC提供了API,允许开发者在网页或应用中直接集成音视频通信功能,无需...
标签中包含的"boghe"是该软件的名称,"doubango"可能是与该客户端相关的开源项目或库,它可能被用于实现SIP功能。"voip"代表VoIP技术,"sip"再次强调了SIP协议的重要性,而"freeswitch"是一个流行的开源通信平台,...
【sip开源服务器】是通信领域中的一个重要组成部分,主要用于实现即时通讯功能,如IM(即时消息)、音频和视频通话。SIP(Session Initiation Protocol)是一种应用层控制协议,用于建立、修改和终止多媒体通信会话...
LumiSoft.Net是一个全面的开源库,专为.NET开发者设计,提供了丰富的SIP协议实现。这个库包括了SIP消息的构建、解析、路由以及会话管理等功能,使得开发者无需从底层开始编写SIP协议代码,大大简化了开发过程。...
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。
**miniSipServer** 是一个开源的SIP服务器项目,它支持多种功能,包括用户注册、呼叫处理、媒体协商等。通过配置peers,我们可以定义SIP用户及其属性,比如用户名、密码、联系地址等。在实际部署中,miniSIPServer...
在使用Kphone开源SIP时,用户需要了解基本的SIP概念,如SIP地址(URI)、呼叫信令流程、媒体编码(如G.711、Opus或VP8)以及如何配置SIP服务器和客户端参数。同时,熟悉Linux命令行或编程知识对于深度定制和故障排查...
FreeSWITCH是一个强大的开源通信平台,专为VoIP(Voice over Internet Protocol)和视频会议设计。这个实战指南,"FreeSWITCH VoIP 实战.rar",由merelyjh2提供,旨在帮助研发人员深入理解并应用FreeSWITCH进行sip...
Sipdroid是一款开源的Android应用,用于实现VoIP(Voice over Internet Protocol)服务,它基于SIP(Session Initiation Protocol)协议进行通信。SIP是一种应用层控制协议,主要用于建立、修改和终止多媒体会话,如...
OPAL(Open Protocol Abstraction Library)是一个开源的VoIP(Voice over Internet Protocol)通信系统,它支持多种网络通信协议,包括OPENH323和SIP(Session Initiation Protocol)。这个开源项目允许开发者构建...
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。同时,应该注意的...
本代码包含基本的VoIP 技术实现的代码和技巧
总的来说,jphonelite项目是一个基于Android平台,利用Java编程语言实现的SIP拨号应用,它的开源性质为开发者提供了学习和实践移动通信技术的机会。开发者需要掌握Android应用开发的基本技能,理解SIP协议的工作机制...