en source means all source code is available!! Do not post any "free but not open" software here!
SIP Proxies
- Mini-SIP-Proxy A very tiny perl POE based SIP proxy
- MjServer cross-platform SIP proxy/registrar/redirect, written in java, based on MjSip stack
- MySIPSwitch SIP Proxy server which allows using multiple SIP accounts with a single SIP login
- NethidPro3.0.6 Opensource Sip Encryption Bridge: www.vonets.com
- Net-SIP A Perl SIP framework that includes a stateless proxy
- JAIN-SIP Proxy
- OpenJSIP Opensource distributed standalone SIP proxy, SIP registrar, SIP location service run by Java VM. Based on NIST SIP and derived from JAIN-SIP Proxy.
- OpenSBC: MPL licensed SIP proxy/registrar/B2BUA with NAT traversal and ENUM
- OpenSER: GPL SIP Server with TLS support - renamed to Kamailio
- OpenSIPS forked from OpenSER.
- partysip SIP proxy server
- SaRP SIP and RTP Proxy in Perl
- sipd SIP Proxy
- SIP Express Router (SER): the SIP router/proxy/jack-in-all-trades from IPtel.org
- Siproxd SIP and RTP Proxy
- SIPVicious tool suite: tools for auditing sip devices
- 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
SIP Clients (UA's)
Linux clients:
- Cockatoo
- Ekiga || SIP, H.323 audio and video softphone for various linux, solaris, windows, and various unix systems. Formerly GnomeMeeting
- FreeSWITCH: Console client for SIP, IAX2, Woomera and Jingle/Google Talk
- 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
- Open IP Phone Business IP Phone sdk support, ims compliant, good interoperability.
- OpenSIPStack MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.
- OpenSoftphone: A simple Java based SIP softphone using the PjSip-jni wrapper.
- OpenZoep: GPL telephone and IM messaging client engine
- Peers Minimalist SIP softphone written in java (tested on linux and windows)
- PhoneGaim
- PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
- QuteCom ex-OpenWengo: a fully SIP compliant multiplatform softphone with many features
- SFLphone, open-source multiplatform multi-protocol VoIP client
- Shtoom: SIP softphone in Python, runs on Windows, Mac, Linux
- SIP Communicator Audio/Video phone and messenger - Multiplatform - Open Source (also supports XMPP, MSN, AIM, Yahoo! and others).
- SipToSis from mhspot.com Skype SIP UA - Multiplatform - Open Source
- sipXezPhone ("sipX easy phone") from SIPfoundry based on sipXtapi
- sipXphone from SIPfoundry, previously known as the Pingtel phone
- Twinkle
- YateClient is multiprotocol and multiplatform softphone with H.323, SIP, Jingle and IAX support.
- YeaPhone: A SIP softphone for the Yealink USB-P1K handset based on the libLinphone backend
MacOS X clients:
- Blink: It supports wideband VoIP, Instant Messaging, File Transfer and Desktop Sharing based on MSRP
- FreeSWITCH: Console client for SIP, IAX2, Woomera and Jingle/Google Talk
- PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
- QuteCom ex-OpenWengo: a fully SIP compliant multiplatform softphone with many features
- SFLphone, open-source multiplatform multi-protocol VoIP client
- Shtoom: SIP softphone in Python, runs on Windows, Mac, Linux
- SIP Communicator Audio/Video phone and messenger - Multiplatform - Open Source (also supports XMPP, MSN, AIM, Yahoo! and others).
- SipToSis from http://www.mhspot.com Skype SIP UA - Multiplatform - Open Source
- Telephone: A SIP softphone designed for the Mac (written in Objective-C/Cocoa). Very good integration with Mac OSX : Dial from Addressbook, dial tel: URIs from Safari, notifications with Growl.
- YateClient skinnable VoIP client based on QT library which supports H.323, SIP, Jingle and IAX protocols
Windows clients
- Ekiga || SIP, H.323 audio and video softphone for various linux, solaris, windows, and various unix systems. Formerly GnomeMeeting
- FreeSWITCH: Console client for SIP, IAX2, Woomera and Jingle/Google Talk
- JPhone Rich software SDK support softphone development, Windows, Linux, ThreadX, Vxworks etc.
- 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
- OfficeSIP Messenger is audio-video softphone and instant messenger, open source alternative to MS Office Communicator.
- OfficeSIP Softphone GPL audio-video softphone.
- OpenSIPStack MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.
- OpenZoep: GPL telephone and IM messaging client engine
- Peers Minimalist SIP softphone written in java (tested on linux and windows)
- PhoneGaim
- PJSUA: Command line SIP UA with SIMPLE, IM, call transfer, RTCP/RTCP, etc.
- OpenSoftphone: A simple Java based SIP softphone using the PjSip-jni wrapper
- QuteCom ex-OpenWengo: a fully SIP compliant multiplatform softphone with many features
- Shtoom: SIP softphone in Python, runs on Windows, Mac, Linux
- SIP Communicator Audio/Video phone and messenger - Multiplatform - Open Source (also supports XMPP, MSN, AIM, Yahoo! and others).
- SipToSis from mhspot.com Skype SIP UA - Multiplatform - Open Source
- sipXezPhone ("sipX easy phone") from SIPfoundry based on sipXtapi
- sipXphone from SIPfoundry, previously known as the Pingtel phone
- VMukti (formerly 1videoConference) alpha: a web2.0 VoIP video conferencing software for Asterisk.
- wxCommunicator Windows softphone based on sipXtapi and wxWidgets 2.8.x, multi-account, conferencing, NAT support
- YateClient is multiprotocol and multiplatform softphone with H.323, SIP,Jingle and IAX support.
SIP tools
- Callflow: Generates SIP Call Flow diagrams
- miTester for SIP: SIP testing tool; Automates test execution.
- Open Source Asterisk AMI: Open Source Asterisk AMI interface application
- pjsip-perf: SIP transaction and call performance measurement tool
- 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-CallerID: SIP Caller ID retrieval and lookup
- SIPbomber: SIP proxy testing tool
- SIP SIMPLE Command Line Tools for SIP sessions (complete console based SIP UA) and SIMPLE Presence (Publish, Subscribe, Notify) and XCAP document manipulation
- Sipp: SIP performance tester
- Sipper: SIPr (called Sipper) is an open source and a comprehensive SIP application testing framework. Generate any call flow in minutes.
- SIP Proxy: SIP security testing tool.
- Sipsak: SIP testing tool
- SIP Soft client: Software development kit for SIP Softphone
- SIPVicious tool suite: tools for auditing SIP devices
- SMAP: Locating and fingerprinting remote SIP devices
- Vovida.org load balancer: SIP Load Balancer
SIP Protocol Stacks and Libraries
- Aloha Spring based J2SE SIP A/S which leverages optimistic concurrent model and supports multiple persistence models
- eXosip - eXtended osip library
- Juphoon SIP Stack Rich software SDK support SIP, SDP, XML, RTP/RTCP, HTTP, STUN, ABNF etc. Support Windows, Linux, ThreadX, Vxworks etc.
- libdissipate SIP stack
- minisip includes a SIP stack
- MjSip - complete and powerful java-based SIP library for both J2SE and J2ME platforms.
- MSRP Library - MSRP protocol (RFC4975) and its relay extension (RFC4976) written in Python
- NIST SIP Various SIP appications and tools in Java
- Open Sip Stack MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.
- oSIP Library SIP Library
- OSP client protocol stack and SIPfoundry
- PhClickDial - Verona based Active/X plugin for IE allowing ClickToDial functionallity
- PJSIP: Small footprint, high performance, and ultra-portable SIP stack written in C, and has language binding for Python. Works on smartphones (Symbian, Windows, iPhone/iOS, Android) as well as desktops and support ZRTP encryption.
- reSIProcate SIP stack and sample Application from SIPfoundry
- SailFin Adds SIP support the the Java GlassFish Application Server
- sipXtackLib an RFC 3261, 3263 complient SIP stack from SIPfoundry
- http://sofia-sip.sourceforge.net Sofia-Sip is SIP stack implementation with STUN and presense support
- SIP SIMPLE client SDK - High level middleware on top of SIP, RTP, MSRP and XCAP protocols
- Twisted Python protocol stacks and applications includes SIP support
- Verona - GPL licenesed VOIP engine based on oSIP,eXosip,oRTP,ffmepg, works on Linux,Windows Mac-OS/X
- Vovida SIP Vovida SIP stack
- XCAP Library - XCAP client library written in Python
- 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.
H.323 Clients
Linux clients:
- Ekiga || SIP, H.323 audio and video softphone for various linux, solaris, windows, and various unix systems. Formerly GnomeMeeting
- FreeSWITCH: Console client using OPAL
- GnomeMeeting
- YateClient is multiprotocol and multiplatform softphone with H.323, SIP and IAX support.
MacOS X clients:
- FreeSWITCH: Console client using OPAL
- ohphoneX
- YateClient skinnable VoIP client based on the QT library which supports H.323, SIP, Jingle and IAX protocols
Windows clients:
- Ekiga || SIP, H.323 audio and video softphone for various linux, solaris, windows, and various unix systems. Formerly GnomeMeeting
- FreeSWITCH: Console client using OPAL
- OpenPhone
- YateClient is multiprotocol and multiplatform softphone with H.323, SIP and IAX support.
H.323 Gatekeeper
- GNU Gatekeeper - for Linux, Windows, Mac etc.
IAX clients
- FreeSWITCH
- IAXComm for Linux, MacOS X and Windows
- Kiax - for Linux, Windows and MacOS, based on iaxclient, GPL
- MozIAX
- QtIax from http://www.holgerschurig.de/qtiax.html
- SFLphone, open-source multiplatform multi-protocol VoIP client (IAX support is planned)
- YakaPhoneSimple, Free, Open Source, Skinnable IAX/IAX2 Softphone from YakaSoftware
- YateClient is multiprotocol and multiplatform softphone with H.323, SIP and IAX support.
RTP Proxies
- AG Projects: MediaProxy 1 works with SIP express router and OpenSER, has load-balancing using DNS SRV records and accounting capabilities
- Maxim Sobolev RTPproxy: Works with SIP express router to traverse NAT, also functions as RTPgateway between IPv4 and IPv6
- MediaProxy 2 is more scalable using kernel space switching and works with OpenSIPs
RTP Protocol Stacks
- ccRTP C++ library based on GNU Common C++
- Juphoon RTP Stack Rich software SDK include RTP/RTCP stack. Support Windows, Linux, ThreadX, Vxworks etc.
- JRTPLIB C++ object oriented RTP library
- libRTP part of gnome-o-phone
- libzrtpcpp - ZRTP extension library for ccRTP stack
- LIVE.COM Streaming Media includes C++ RTP stack
- oRTP Written in C, running on linux, win32 and arm-linux.
- PJMEDIA: Small footprint media stack with a tiny RTP/RTCP stack suitable for DSP or embedded deployment
- RTPlib C library
- 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
- UCL Common Multimedia Library includes cross platform RTP stack
- Vovida RTP Stack
- YRTP - Yate RTP stack, that can be used in other projects.
- zrtp4j - ZRTP stack for Java, based on GNU ZRTP, used in SIP Communicator
MSRP Relays
- MSRPRelay from AG Projects
XCAP servers
- OpenXCAP from AG Projects
Other tools
- Encours Teleconferencing in your web browser with an integrated VOIP layer (Java) and an optional Asterisk connectivity on the server side.
- Howler Technologies - optimised G.729 codec for softswitch market.
- MORCC - automated online Calling Card store. Paypal integrated.
- OgonPhonesXML .NET Library for Aastra SIP Phones and Cisco SIP/IP phones for fast and easy XML Interfacement.
- Oreka capture and retrieval of SIP, Cisco Skinny (SCCP) and raw RTP sessions with audio compression, rdbms metadata storage and web based user interface.
- Vovida.org STUN server: A STUN server
- Voipong - Voice over IP (VoIP) sniffer and call detector.
- Vomit converts a Cisco IP phone conversation (recorded with TCPdump) into a standard WAV file
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
- FreeSWITCH Open Source PBX and Soft Switch
- OpenPBX: Open Source PBX developed using Perl
- PBX4Linux: ISDN PBX with H.323 GW
- SIPexchange PBX Pingtel's SIP PBX
- sipwitch: GNU project's Pure SIP call server, sipwitch on freshmeat.net
- sipX - The SIP PBX for Linux from SIPfoundry, sipX on freshmeat.net
- SIP - It's the Rage! - Rage! Business Office Xchange based on SipFoundry
- YATE Yet Another Telephony Engine - supports H.323, SIP, IAX, PSTN
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.
- FreeSWITCH
- OpenVXI: Implementation of VoiceXML
- SEMS: Free/Open Source SIP media server with IVR capabilities
- sipX PBX The SIP PBX for Linux (open source) with built-in IVR (voice mail & auto-attendant)
- YATE Yet Another Telephony Engine
- See Also: VoiceXML
Voicemail servers
- Asterisk: Open Source PBX with built-in Voicemail Server
- FreeSWITCH
- Lintad: Linux Telephone Answering Device - A Voice and Faxmail Server
- OpenPBX: Open Source PBX with built in voicemail
- OpenUMS: Linux Voicemail and Unified Messaging Server
- SEMS: Free/Open Source SIP media server with built-in Voicemail and Voicebox Server
- sipX PBX The SIP PBX for Linux (open source) with built-in IVR (voice mail & auto-attendant)
- VOCP: A Voicemail Server for voice modems
- YATE Yet Another Telephony Engine with H.323, SIP and IAX support.
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
- UniMRCP: cross-platform MRCP client and server
Fax Servers
- Asterisk Fax Email Gateway
- Lintad: Linux Telephone Answering Device - A Voice and Faxmail Server
- Hylafax
Development platforms, protocol stacks
- H323plus: Open Source H.323 Protocol Stack following on from the original openH323
- OpenBloX: OpenBloX Open Source Java Diameter framework with all IMS and SIP servers interfaces; maintained by Traffix Systems,
- OpenMGCP: Open Source MGCP Protocol Stack Developed with C and POSIX APIs,
- OpenSS7: SS7 Protocol Stack
- 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.
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)
Billing
Codecs
- See Codec Software
Middleware
- Ernie: Open Source Python based applications platform for VoIP and presence based applications
- Mobicents: The most popular Open Source Service Logic Execution Environment (JSLEE) and SIP Application Server for the Java platform.
- TALK: Web based CTI Solution (AJAX client) which provides call control, presence and directorty features.
Suite Solutions
- Zoontelecom: Zoon Suite is a Open Source solution for make VoIP services with billing and more. (Spanish)
CTI Dialer utilities
- Asterisk phonebook A common shared phone book directory for Asterisk PBX
- TALK Powerful directory management and scalable architecture to create Click to call or Select and Dial applications + AJAX libraries to implement these features in your web site.