Open Source VOIP Software
From P2P Foundation
Extensive directory of Open Source VOIP applications, both clients and servers at http://www.voip-info.org/wiki-Open%20Source%20VOIP%20Software
Directory
Copied from http://www.voip-info.org/wiki-Open%20Source%20VOIP%20Software
No live links, for hyperlinks go to that original compilation.
SIP Proxies
* Net-SIP A Perl SIP framework that includes a stateless proxy * sipd SIP Proxy * SIP Express Router (SER): the SIP router/proxy/jack-in-all-trades from IPtel.org * partysip * SaRP SIP and RTP Proxy in Perl * Siproxd SIP and RTP Proxy * sipX The SIP PBX for Linux: Complete, native SIP PBX solution from SIPfoundry * Vocal SIP softswitch with H.323 and MGCP translators for non-SIP endpoints * Yxa: Written in the Erlang programming language * JAIN-SIP Proxy * Mini-SIP-Proxy A very tiny perl POE based SIP proxy * OpenSER: GPL SIP Server with TLS support * MjServer: cross-platform SIP proxy/registrar/redirect, written in java, based on MjSip stack * OpenSBC: MPL licensed SIP proxy/registrar/B2BUA with NAT traversal and ENUM * MySIPSwitch: SIP Proxy server which allows using multiple SIP accounts with a single SIP login * SIPVicious tool suite: tools for auditing sip devices
SIP Clients (UA's)
Linux clients:
* Cockatoo
* Ekiga: SIP, H.323 audio and video softphone for various unices
* Kphone
* Linphone audio and video SIP softphone for Linux and Windows XP
* minisip cross-platform SIP softphone, Linux, Windows XP and soon Windows Mobile 2003 SE
* MjUA: simple cross-platform SIP softphone, written in java, based on MjSip stack
* PhoneGaim
* PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
* SFLphone, open-source multiplatform multi-protocol VoIP client
* OpenWengo: a fully SIP compliant multiplatform softphone with many features
* OpenZoep: GPL telephone and IM messaging client engine
* Shtoom: SIP softphone in Python, runs on Windows, Mac, Linux
* sipXphone from SIPfoundry, previously known as the Pingtel phone
* sipXezPhone ("sipX easy phone") from SIPfoundry based on sipXtapi
* Twinkle
* YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.
* YeaPhone: A SIP softphone for the Yealink USB-P1K handset based on the libLinphone backend
* FreeSWITCH
* http://www.opensipstack.org MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.
MacOS X clients:
* FreeSWITCH: Console client for SIP, IAX2, Woomera and Jingle/Google Talk * PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc. * SFLphone, open-source multiplatform multi-protocol VoIP client * Shtoom: SIP softphone in Python, runs on Windows, Mac, Linux
Windows clients:
* 1videoConference alpha: a web2.0 VoIP video conferencing software for Asterisk. * FreeSWITCH: Console client for SIP, IAX2, Woomera and Jingle/Google Talk * minisip cross-platform SIP softphone, Linux, Windows XP and soon Windows Mobile 2003 SE * Linphone audio and video SIP softphone for Linux and Windows XP * MjUA: simple cross-platform SIP softphone, written in java, based on MjSip stack * Eyeball Messenger: Standards based soft client that is SIP and XMPP compliant * OpenWengo: a fully SIP compliant multiplatform softphone with many features * OpenZoep: GPL telephone and IM messaging client engine * PhoneGaim * PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc. * SIP COMMUNICATOR Java based softphone * Shtoom: SIP softphone in Python, runs on Windows, Mac, Linux * sipXphone from SIPfoundry, previously known as the Pingtel phone * http://www.opensipstack.org MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available. * sipXezPhone ("sipX easy phone") from SIPfoundry based on sipXtapi * wxCommunicator Windows softphone based on sipXtapi and wxWidgets 2.8.x, multi-account, conferencing, NAT support * YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.
SIP tools
* Callflow: Generates SIP Call Flow diagrams * SIP-CallerID: SIP Caller ID retrieval and lookup * SIPbomber: SIP proxy testing tool * Sipp: SIP performance tester * SIP Proxy: SIP security testing tool. * pjsip-perf: SIP transaction and call performance measurement tool * Sipsak: SIP testing tool * SMAP: Locating and fingerprinting remote SIP devices * Vovida.org load balancer: SIP Load Balancer * PROTOS Test-Suite: SIP Testing tools * SFTF: SIP Forum Test Framework - a SIP UA test suite primarily targeted at UA software developers hosted by SIPfoundry * SIP Soft client: Software development kit for SIP Softphone * SIPVicious tool suite: tools for auditing SIP devices
SIP Protocol Stacks and Libraries
* YASS - Statefull SIP stack used in Yate written in C++ usable for client, server or proxy in a multithread or single thread model. It's working on both Windows and Linux, it's very small but full featured. * MjSip - complete and powerful java-based SIP library for both J2SE and J2ME platforms. * oSIP Library SIP Library * eXosip - eXtended osip library * Vovida SIP Vovida SIP stack * reSIProcate SIP stack and sample Application from SIPfoundry * NIST SIP Various SIP appications and tools in Java * PJSIP: Small footprint, high performance, and ultra-portable SIP stack written in C, and has language binding for Python. * Twisted Python protocol stacks and applications includes SIP support * OSP client protocol stack and SIPfoundry * libdissipate SIP stack * sipXtackLib an RFC 3261, 3263 complient SIP stack from SIPfoundry * minisip includes a SIP stack * http://sofia-sip.sourceforge.net Sofia-Sip is SIP stack implementation with STUN and presense support * http://www.opensipstack.org MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available. * Verona - GPL licenesed VOIP engine based on oSIP,eXosip,oRTP,ffmepg, works on Linux,Windows Mac-OS/X * PhClickDial - Verona based Active/X plugin for IE allowing ClickToDial functionallity
H.323 Clients
Linux clients:
* Ekiga * GnomeMeeting * YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support. * FreeSWITCH
MacOS X clients:
* ohphoneX * FreeSWITCH
Windows clients:
* OpenPhone * FreeSWITCH
H.323 Gatekeeper
* GNU Gatekeeper - for Linux, Windows, Mac etc.
IAX clients
* IAXComm for Linux, MacOS X and Windows * Kiax - for Linux (QT3) and Windows (QT4), based on iaxclient, GPL * QtIax from http://www.holgerschurig.de/qtiax.html * SFLphone, open-source multiplatform multi-protocol VoIP client (IAX support is planned) * MozIAX * YakaPhoneSimple, Free, Open Source, Skinnable IAX/IAX2 Softphone from YakaSoftware * YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support. * FreeSWITCH
RTP Proxies
* Maxim Sobolev's RTPproxy: Works with SIP express router to traverse NAT, also functions as RTP gateway between IPv4 and IPv6 * AG Projects: SER MediaProxy works with SIP express router, has load-balancing using DNS SRV records and accounting capabilities
RTP Protocol Stacks
* JRTPLIB CUCL Common Multimedia Library includes cross platform RTP stack * oRTP Written in C, running on linux, win32 and arm-linux. * ccRTP C++ library based on GNU Common C++ * LIVE.COM Streaming Media includes C++ RTP stack * Vovida RTP Stack * RTPlib C library * libRTP part of gnome-o-phone * sipXmediaLib RTP + audio bridges, audio splitters, echo suppression, tone from generation (e.g. DTMF), streaming support, RTCP, G711 codecs, etc. from SIPfoundry * Secure RTP - see;"> SRTP * YRTP - Yate RTP stack, that can be used in other projects. * FreeSWITCH * PJMEDIA: Small footprint media stack with a tiny RTP/RTCP stack suitable for DSP or embedded deployment
Other tools
* Vovida.org STUN server: A STUN server * Vomit converts a Cisco IP phone conversation (recorded with TCPdump) into a standard WAV file * Oreka capture and retrieval of SIP, Cisco Skinny (SCCP) and raw RTP sessions with audio compression, rdbms metadata storage and web based user interface. * MORCC - automated online Calling Card store. Paypal integrated. * Voipong - Voice over IP (VoIP) sniffer and call detector.
PBX platforms
Some of these include SIP proxy functionality
* Asterisk: Open Source PBX. Supports IAX, SIP, MGCP, H.323 and other protocols * CallWeaver: a fork of Asterisk with T.38 termination * OpenPBX: Open Source PBX developed using Perl * PBX4Linux: ISDN PBX with H.323 GW * sipX - The SIP PBX for Linux from SIPfoundry, sipX on freshmeat.net * SIPexchange PBX Pingtel's SIP PBX * YATE Yet Another Telephony Engine - supports H.323, SIP, IAX, PSTN * FreeSWITCH
IVR platforms
* Asterisk: Open Source PBX with built-in IVR server * Bayonne: GNU project IVR server * CT Server Perl based Open Source client/server library supporting Voicetronix Telephony hardware. * OpenVXI: Implementation of VoiceXML * sipX PBX The SIP PBX for Linux (open source) with built-in IVR (voice mail & auto-attendant) * YATE Yet Another Telephony Engine * FreeSWITCH * See Also: VoiceXML
Voicemail servers
* Asterisk: Open Source PBX with built-in Voicemail Server * OpenPBX: Open Source PBX with built in voicemail * sipX PBX The SIP PBX for Linux (open source) with built-in IVR (voice mail & auto-attendant) * Lintad: Linux Telephone Answering Device - A Voice and Faxmail Server * OpenUMS: Linux Voicemail and Unified Messaging Server * VOCP: A Voicemail Server for voice modems * YATE Yet Another Telephony Engine with H.323, SIP and IAX support. * FreeSWITCH
Speech
Text-to-speech and speech-to-text (voice recognition)
* Festival: Voice synthesis system (implemented with a trainable neural network) * OpenSALT: Implementation of SALT * OpenVXI: Implementation of VoiceXML * Sphinx: speaker-independent speech recognizer * FreeSWITCH
Fax Servers
* Lintad: Linux Telephone Answering Device - A Voice and Faxmail Server * Hylafax * Asterisk Fax Email Gateway
Development platforms, protocol stacks
* OpenMGCP: Open Source MGCP Protocol Stack Developed with C and POSIX APIs, * OpenSS7: SS7 Protocol Stack * H323plus: Open Source H.323 Protocol Stack following on from the original openH323 * ooh323c: Open Source H.323 Protocol Stack Developed in C * ++Skype C library for skype add-on platform independent software development. It is platform independent, easy to use, and easy to extend because of the flexible library design, inspired by modern C++ design ideas. Performance is one of the goals. * OpenBloX: OpenBloX Open Source Java Diameter framework with all IMS and SIP servers interfaces; maintained by Traffix Systems, * OpenSS7: SS7 Protocol Stack
Radius Servers
* Aradial: Radius server and Billing for VoIP * BSDRadius: Radius server for VoIP * Interlink RADIUS Server RADIUS Server Software * RadBox RADIUS Server + Billing System. (For a work, you nead instal Framework 2.0)
Middleware
* Mobicents: The most popular Open Source Service Logic Execution Environment (JSLEE) and SIP Application Server for the Java platform. * Ernie: Open Source Python based applications platform for VoIP and presence based applications
Suite Solutions
* Zoontelecom: Zoon Suite is a Open Source solution for make VoIP services with billing and more. (Spanish)
