【VOIP】Open source

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
• 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

• 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.

10 Open Source VoIP softphones

Let's continue with our Open Source VoIP series: today topic will be softphones ... There are a lot of softphones, some of them are free, maybe you have to pay for some other ones, but let's focus on what really matters, quality open source softphones. Because open source, was, is and will be always better and it respects your freedom. Freedom to do whatever you want with it, thus includes making phone calls.

Ekiga

Ekiga (formely known as GnomeMeeting) is an open source SoftPhone, Video Conferencing and Instant Messenger application over the Internet.
It supports HD sound quality and video up to DVD size and quality.
It is interoperable with many other standard compliant softwares, hardwares and service providers as it uses both the major telephony standards (SIP and H.323).
Main Features of the Ekiga Softphone Version 3.2 in a nutshell

• Ease of use with a modern Graphical User Interface.
• Audio and Video free calls through the internet.
• Free Instant Messaging through the internet with Presence support.
• Audio (and video) calls to landlines and cell phones with support to the cheapest service providers.
• High Definition Sound (wideband) and Video Quality up to DVD quality (high framerate, state of the art quality codec and frame size).
• Free of choice of the service provider.
• SMS to cell phones if the service provider supports it (like the default provider).
• Standard Telephony features support like Call Hold, Call Transfer, Call Forwarding, DTMF.
• Remote and Local Address Book support: Remote Address Book support with authentification using the standard LDAP technology, Local Address support in Gnome (Evolution).
• Multi platform: Windows and GNU/Linux
• Wide interoperability: Ekiga use the main deployed stantards for telephony protocols (SIP and H.323) and has been tested with a wide range of softphones, hardphones, PBX and service providers.

http://www.ekiga.org/
License: GPL

Twinkle

Twinkle is a softphone for your voice over IP and instant messaging communcations using the SIP protocol. You can use it for direct IP phone to IP phone communication or in a network using a SIP proxy to route your calls and messages.
In addition to making basic voice calls Twinkle provides you the following features:

• 2 call appearances (lines)
• Multiple active call identities
• Custom ring tones
• Call Waiting
• Call Hold
• 3-way conference calling
• Mute
• Call redirection on demand
• Call redirection unconditional
• Call redirection when busy
• Call redirection no answer
• Reject call redirection request
• Blind call transfer
• Call transfer with consultation (attended call transfer)
• Reject call transfer request
• Call reject
• Repeat last call
• Do not disturb
• Auto answer
• Message Waiting Indication
• Voice mail speed dial
• User defineable scripts triggered on call events
•        E.g. to implement selective call reject or distinctive ringing
• RFC 2833 DTMF events
• Inband DTMF
• Out-of-band DTMF (SIP INFO)
• STUN support for NAT traversal
• Send NAT keep alive packets when using STUN
• NAT traversal through static provisioning
• Persistent TCP connections for NAT traversal
• Missed call indication
• History of call detail records for incoming, outgoing, successful and missed calls
• DNS SRV support
• Automatic failover to an alternate server if a server is unavailable
• Other programs can originate a SIP call via Twinkle, e.g. call from address book
• System tray icon
• System tray menu to quickly originate and answer calls while Twinkle stays hidden
• User defineable number conversion rules
• Simple address book
• Support for UDP and TCP as transport for SIP
• Presence
• Instant messaging
• Simple file transfer with instant message
• Instant message composition indication
• Command line interface (CLI)

Twinkle is available for Linux only

http://www.xs4all.nl/~mfnboer/twinkle/
License: GPL

Kiax

Kiax started in early 2004 as a small program mainly aimed to provide a simple user interface for making VoIP calls with Asterisk PBX (an open source VoIP PBX). Its first versions showed the existing need for a user-friendly, free and open softphone. Currently Kiax has been downloaded by more than 60 000 users (stats from SourceForge.net) and is available for direct installation from the repositories of the major Linux distros (Ubuntu, SuSE). While it is functionally rich and considerably stable its development has reached a state where modification and customization became difficult. With the help of Forschung-Direkt and MIXvoip Kiax development has been restarted. Kiax ver.2 is a complete re-write of the softphone which aims to clean up the design issues and to provide a more flexible architecture for extention and customization.
Key features and characteristics of Kiax ver.2:

• Decoupled Signaling, Storage and Visualization aspects
• Modularized, lightweight core layer
• GCC4 ready code
• Single codebase for Linux, Windows and MacOS
• SQLite as default storage backend
• QT4.4 as GUI frontend
• Webkit integration
• Even simpler (than old Kiax) to use UI
• Completely brandable
• Remote configuration
• Simplified integration with service providers (via JSON)
• Support for multiple service providers
• Support for simultaneous calls
• Registry fail-over support
• Live CDR and Contacts search
• Codecs: G711, iLBC, GSM, Speex
• Noise reduction filter
• I18n support

http://kiax.sourceforge.net
License: LGPL-licensed core, GPL GUI

QuteCom

QuteCom is the new name for the open source softphone previously known as WengoPhone. QuteCom began life as OpenWengo developed by French VoIP provider Wengo as a free softphone for its telephony service.
QuteCom is cross-platform (Windows, Linux, Mac OS X) and integrates voice and video calls and instant messaging. The number of protocols supported is on the same level as other multi-protocol IM clients. The application is developed with the Qt cross-platform toolkit.

http://www.qutecom.org/
License: GPL

SIPhone

Home of the World's first free SIP/VoIP application for iPhone and iPod Touch 1 and 2.
Siphon SIP/VoIP project is the first in his category that works on iPhone and iPod Touch 2 with headset for all SIP providers. It is a native application approved running on 2.X using internal micro/speaker and headset.
The Application supports the SIP standard, preserving compatibility with hundreds of SIP providers and offers a GUI which preserves the apple design of native iPhone applications.
Be careful, this version didn't test on iPod Touch 1. One thing is sure, Touchmod's micro doesn't work with iPhone 2.X OS. You need a microphone with a HW key inside “approved” by Apple. Such a microphone are: iVoice III from Macally.
Currently, Siphon is localized in 15 languages.

http://code.google.com/p/siphon/
License: GPLv2

SIP Communicator

SIP Communicator is an audio/video Internet phone and instant messenger that supports some of the most popular VoIP and instant messaging protocols such as SIP, Jabber, AIM/ICQ, MSN, Yahoo! Messenger, Bonjour, IRC and a whole lot of other useful features.
SIP Communicator is completely Open Source / Free Software, and is freely available under the terms of the GNU Lesser General Public License.
These are SIP Communicator main features:

• Audio calls
• Video calls
• Desktop streaming
• Desktop sharing
• Audio conference calls
• Audio level display
• Call recording
• Attended transfer
• Blind transfer
• Call encryption with SRTP and ZRTP
• Mute
• Hold
• Support for ICE
• Wideband audio
• Noise suppression
• Echo cancellation

SIP Communicator is cross-platform and is developed in Java.

http://sip-communicator.org
License: GPL

SFLphone

SFLphone is a SIP and IAX2 (Asterisk) compatible softphone for Linux developed by Canadian Linux consulting company Savoir-Faire Linux.
The SFLphone project's goal is to create a “robust enterprise-class desktop phone” and is designed to cater for home users as well as the “hundred-calls-a-day receptionist”.
Its main features include support of unlimited number of calls, multi-accounts, call transfer and hold. Call recording is another useful feature.
SFLphone has clients for GNOME (integrated options), KDE and Python and it now supports the PulseAudio sound server, so users can experience additional functionality like sound mixing and per-application volume control.
The softphone is designed to connect to the Asterisk open source PABX.

URL: http://www.sflphone.org/
License: GPL3

Empathy

Empathy is a messaging program which supports text, voice, and video chat and file transfers over many different protocols. You can tell it about your accounts on all those services and do all your chatting within one application.
Empathy uses Telepathy for protocol support and has a user interface based on Gossip. Empathy is the default chat client in current versions of GNOME, making it easier for other GNOME applications to integrate collaboration functionality using Telepathy.
Current features:

• Multi-protocol: Google Talk (Jabber/XMPP), MSN, IRC, Salut, AIM, Facebook, Yahoo!, Gadu Gadu, Groupwise, ICQ and QQ. (Supported protocols depend on installed Telepathy Connection Manager components.) Supports all protocols supported by Pidgin.
• File transfer for XMPP, and local networks.
• Voice and video call using SIP, XMPP and Google Talk.
• Some IRC support.
• Conversation theming (see list of supported Adium themes).
• Sharing and viewing location information.
• Private and group chat (with smileys and spell checking).
• Empathy Conversation window
• Conversation logging.
• Automatic away and extended away presence.
• Automatic reconnection using Network Manager.
• Python bindings for libempathy and libempathy-gtk
• Support for collaborative applications (“tubes”).

URL: http://live.gnome.org/Empathy
License: GPL

miniSIP

Minisip is a SIP User Agent ("Internet telephone").
It can be used to make phone calls, instant message and videocalls to your buddies connected to the same SIP network.
Features:

• SIP compliant (RFC 3261 and more)
• Multiple lines (users) on the same phone
• Multiple incoming/outgoing calls simultaneously
• Runs on multiple Operating Systems (Linux PC, Linux familiar IPAQ PDA, Windows XP and soon Windows Mobile 2003 SE)
• Focus on security: TLS, end-to-end security, SRTP, MIKEY (DH, PSK, PKE)
• Instant Messaging
• Video conferencing
• Spatial audio
• Push-to-Talk (P2T)
• Full Mesh audio conferencing
• STUN support
• Call Logging

URL: http://www.minisip.org/
License: GPL

Linphone

Linphone is an internet phone or Voice Over IP phone (VoIP).
Features:

• With linphone you can communicate freely with people over the internet, with voice, video, and text instant messaging.
• Linphone makes use of the SIP protocol, an open standart for internet telephony. You can use Linphone with your favorite SIP VoIP operator.
• Linphone is free-software (or open-source), you can download and redistribute it freely.
• Linphone is available for PCs (linux, windows), MacOSX and for mobile phones: Android, iPhone.

URL: http://www.linphone.org/
License: GPL

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itech 2 years ago from New Delhi, iNdia

Useful article but Help me with one thing: Which software will allow me to call a cellphone directly using my internet connection such as skype without paying a single penny and I does not belong to U.S .

i_see_retards 2 years ago

@itech, you dont understand how VOIP work....

azizzii 2 years ago from Right from Haven

Why don't you try google's soft phone. As far as the free voip calls are concerned, I don't think these calls are reliable. However, you can select the unlimited plans of some voip service provider. These plans are unlimited only in 8$ per month.

BerylS 12 months ago

hello,

At the first phase of my project I also use a free a softphone, because it was only for testing how effective it is. As it turned to be very effective, I decided to implement a more complext softphone for complex text, voice and video phoning to my customers. There are many softphones on the market. Since I increase the capacity, finally, I bought a softphone license, as free softphones were not so reliable as they should be in production. My deciding factors were the support and reliability with high capacity.

Finally, I chose the Ozeki VoIP SIP SDK that is quite flexible enough to develop a softphone (and I also created an autodialer with it). The support team is very responsive and helped me a lot, the right support was critical for me. That is why I purchased this product instead of the others after testing.

Sample source: http://www.voip-sip-sdk.com/p_355-implement-do-not

I hope, my experiences help others...

BR

Private ethernet Burnaby 8 months ago

How do I get a free SIP or VOIP Account to use on my laptop? I have a laptop that I want to basically make a cell phone. I always have Wi-Fi so VoIP/SIP seems like the best option. I want to be able to have free calls to landlines and other softphones.

http://www.itel.ca

VoipGuy 3 weeks ago

Jitsi and Mumble are missing.