Open Source VOIP Software
Open Source VOIP applications, both clients and servers.
Open source means all source code is available!! Do not post any "free but not open" software here!
SIP Proxies
- NethidPro3.0.6 Opensource Sip Encryption Bridge: www.vonets.com
- 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 - renamed to Kamailio
- OpenSIPS forked from OpenSER.
- 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
- JPhone Rich software SDK support softphone development, Windows, Linux, ThreadX, Vxworks etc.
- 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
- SipTheeSkype 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
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.
- 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
- SipTheeSkype from mhspot.com Skype SIP UA - Multiplatform - Open Source
Windows clients
- 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 Java based softphone
- SipTheeSkype 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
- http://www.cytco.net شرکت سیتکو - ارتباط تلفن کوروش ارائه دهنده نسل جدید مراکز تلفن و سیستم های ارتباطی، مرکز تماس، تلفن گویا
- YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.
- 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
- 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
- Sipp: SIP performance tester
- 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
- Sipper: SIPr (called Sipper) is an open source and a comprehensive SIP application testing framework. Generate any call flow in minutes.
SIP Protocol Stacks and Libraries
- Aloha Spring based J2SE SIP A/S which leverages optimistic concurrent model and supports multiple persistence models
- 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
- Zocol Rich software SDK support SIP, SDP, XML, RTP/RTCP, HTTP, STUN, ABNF etc. Support Windows, Linux, ThreadX, Vxworks etc.
H.323 Clients
Linux clients:
MacOS X clients:
Windows clients:
- OpenPhone
- YATE: YateClient is multiprotocol and multiplatform phone with H.323, SIP and IAX support.
H.323 Gatekeeper
IAX clients
RTP Proxies
RTP Protocol Stacks
- JRTPLIB C++ object oriented RTP library
- UCL 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
- Zocol Rich software SDK include RTP/RTCP stack. Support Windows, Linux, ThreadX, Vxworks etc.
MSRP Relays
XCAP servers
- 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.
Some of these include SIP proxy functionality
- 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)
Fax Servers
- 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
Billing
Codecs
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
- Zivios Enterprise Management Suite: Zivios is an open source web based enterprise management system. It offers identity management, single sign on, certificate authority service, user, group and computer provisioning as well as auditing and work flow. Zivios has an extensible plug-in based architecture. Zivios is released with 4 user plug-ins; Mail. Samba, Squid and Asterisk. For more details, please visit Zivios Screen shots and Zivios Architecture. Zivios is OSI certified and is released under GNU GPL.
- Zoontelecom: Zoon Suite is a Open Source solution for make VoIP services with billing and more. (Spanish)
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