Open Source VOIP applications, both clients and servers.

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
• 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
• Open IP Phone Business IP Phone sdk support, ims compliant, good interoperability.
• 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
• 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.
• 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
• 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
• OpenSoftphone: A simple Java based SIP softphone using the PjSip-jni wrapper.

MacOS X clients:

• 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.
• 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 mhspot.com Skype SIP UA - Multiplatform - Open Source

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
• 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.
• 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
• YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.
• OpenSoftphone: A simple Java based SIP softphone using the PjSip-jni wrapper

SIP tools

• Callflow: Generates SIP Call Flow diagrams
• Open Source Asterisk AMI: Open Source Asterisk AMI interface application
• pjsip-perf: SIP transaction and call performance measurement tool
• miTester for SIP: SIP testing tool; Automates test execution.
• 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
• 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
• 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 Symbian and support ZRTP encryption.
• reSIProcate SIP stack and sample Application from SIPfoundry
• SailFin Adds SIP support the the Java GlassFish Application Server
• Twisted Python protocol stacks and applications includes SIP support
• Open Sip Stack MPL licensed SIP stack with ENUM, Presence (XMPP/SIMPLE) and NAT traversal. Reference implementation of Session Border Controller (OpenSBC) available.
• SIP SIMPLE client SDK - High level middleware on top of SIP, RTP, MSRP and XCAP protocols
• 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
• 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.
• Juphoon SIP Stack Rich software SDK support SIP, SDP, XML, RTP/RTCP, HTTP, STUN, ABNF etc. Support Windows, Linux, ThreadX, Vxworks etc.

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
• YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.

MacOS X clients:

• FreeSWITCH: Console client using OPAL
• ohphoneX

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
• YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.

H.323 Gatekeeper

• GNU Gatekeeper - for Linux, Windows, Mac etc.

IAX clients

• IAXComm for Linux, MacOS X and Windows
• FreeSWITCH
• 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
• YATE: YateClient is multiprotocol and multiplatform phone 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 RTP gateway between IPv4 and IPv6
• MediaProxy 2 is more scalable using kernel space switching and works with OpenSIPs

RTP Protocol Stacks

• Secure RTP - see: SRTP
• ccRTP C++ library based on GNU Common C++
• 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
• UCL Common Multimedia Library includes cross platform RTP stack
• Vovida RTP Stack
• YRTP - Yate RTP stack, that can be used in other projects.
• Juphoon RTP Stack Rich software SDK include RTP/RTCP stack. Support Windows, Linux, ThreadX, Vxworks etc.
• 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

• 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
• Encours Teleconferencing in your web browser with an integrated VOIP layer (Java) and an optional Asterisk connectivity on the server side.

PBX platforms

• 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

• 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

• See Open Source Billing Systems

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.

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