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
- 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
- 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
- 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
- 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
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
- 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 Java based softphone
- 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.
- 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
- 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.
- 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
- 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.
- 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
- 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.
- 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:
- 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:
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
IAX clients
RTP Proxies
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.
- Zocol 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
XCAP servers
- Howler Technologies - optimised G.729 codec for softswitch market.
- MORCC - automated online Calling Card store. Paypal integrated.
- 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
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.
- FreeSWITCH
- 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
- 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
- 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
- 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
Billing
Codecs
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.
Suite Solutions
- Zoontelecom: Zoon Suite is a Open Source solution for make VoIP services with billing and more. (Spanish)
Comments
333multiple calls softphone
I am doing research on VoIP now. I need a softphone to simulate multiple calls to evaluate the capacity of the VoIP network.
Can anybody suggest me a opensource softphone which can simulate multiple calls? It would be highly appreciated. My email address i s viruschidai@gmail.com.
333voice quality
Does anyone know free software, to measure voice quality in MOS scale (P.800, PSQM, or whatever)? I spent a lot of time on google but didn't find anything free :(
333we pay
333Need SIP Softphone and we pay
333setup Asterisk Prepadi Calling card
and some body can teach me how to create asterisk with mysql as prepaid calling card
current iam already install asterisk with mysql cdr and Asterisk Management Portal.
other questions how to setup like SIP.conf in AMP ?, if iam using ExpressTalk Softphones any sample conf in mysq DB ?
thanks
333Re: SIP-communicator.org
333Easy SIP PROXY?
I Have a VoIP Gateway that handles all the VoIP Packets for translating them for PSTN or to go over IP. But the gateway doesnt have a way to restrict specific users from accessing it. I need to put a SIP Proxy on top of the Gateway so that only paying subscribers can use our services. Are there any EASY to use open sorce SIP PROXYs? I tried installing VOCAL but I keep getting errors, and there isnt much documentation on the internet that i can find on the installation of VOCAL. Thanks for your help.
333MGCP
333sflphone doesn't do IAX yet
333"but not open" !