Open Source Voip Software , Top IP Telephony application of 2017

Open source software has leading role in emerging of present Information and communication technologies (ICT), Following is list of top rated , mature and reliable open source applications that revolutionized the communication concepts. We have tried our best to include best applications in each category however if you have any news please share with us to keep this post updated .

Open source communications softswitches / IP Telephony applications

  • FreeSWITCH

    A Cross-platform, Scalable, Stable Multi-Protocol Soft Switch ,The project inspired from renown Open Source asterisk PBX system but very well designed and has a bright future
    http://www.freeswitch.org

  • Asterisk

    A software-based, Open Source Converged PBX system that has revolutionized the traditional PBX industry
    http://www.asterisk.org

  • Kamailio

    SIP proxy server, call router, and user agent registration server used in Voice over Internet Protocol and instant messaging applications.
    http://www.kamailio.org/w/

  • OpenSIPS

    SIP proxy server, call router, registration server and redirect server suitable for internet telephony service providers to offer SIP based telephony services to their customers.
    http://www.opensips.org

  • Gnugk

    H.323 gatekeeper based on the OpenH323 stack. A gatekeeper provides address translation, admissions control, call routing, authorization and accounting services
    http://www.gnugk.org

Asterisk & Freeswitch GUI interfaces

  • FusionPBX

    FusionPBX can be used as a highly available single or domain based multi-tenant PBX, carrier grade switch, call center server, fax server, voip server, voicemail server, conference server, voice application server, appliance framework and more. FreeSWITCH™ is a highly scalable, multi-threaded, multi-platform communication platform.
    http://www.fusionpbx.com

  • FreePBX

    A full-featured PBX GUI supporting both asterisk and freeswitch
    http://freepbx.org

  • RasPBX

    RasPBX is a project dedicated to Asterisk and FreePBX running on the Raspberry Pi. For any information related to Raspberry Pi, check the original website at raspberrypi.org.
    http://www.raspberry-asterisk.org

Call Center Solutions

  • GoAutodial

    Goautodial (vicidial) is a professional web based inbound / outbound call center solution built over asterisk
    http://www.goautodial.org/

Communications Frameworks

    ICTCore

  • ICTCore core is open source unified communications framework for developers and integrators to rapidly develop ICT based applications using their existing development skills. By using ICTCore, developer can create communication based applications such as Auto attendant, Fax to Email, Click to Call etc.. they can program custom business logic that can control incoming and outgoing communication instances.

    http://ictcore.org/

SIP protocol stack / SIP development Libraries

  • PJSIP

    Pjsip is sip based open source high performance communication library with multimedia capabilities written in C language for building embedded/non-embedded VoIP applications.
    http://www.pjsip.org/

  • JsSIP

    JsSIP is a simple to use JavaScript library which leverages latest developments in SIP and WebRTC to provide a fully featured SIP endpoint in any website.

    With JsSIP any website can get Real Time Communications features using audio, video and more with just a few lines of code.
    Features

    SIP over WebSocket transport.
    Audio/video calls, instant messaging and presence.
    Lightweight!.
    100% pure JavaScript built from the ground up.
    Easy to use and powerful user API.
    Works with OverSIP, Kamailio and Asterisk servers.

    https://github.com/versatica/JsSIP

  • SIP.js

    SIP.js is a simple, intuitive, and powerful JavaScript signaling library . It is full-featured SIP stack written in JavaScript. With SIP.js, you can harness the power of WebRTC to build audio, video, and realtime data into your application. SIP.js is fast, lightweight, and easy to use.

    https://github.com/onsip/SIP.js

Broadcasting Platform

Billing Systems

  • ASTPP

    ASTPP is a Most Powerful Advanced Open Source VoIP Billing Solution based on open source softswitch freeswitch and it provide Class 4 & Class 5 SoftSwitch Billing Solution & freeswitch Billing
    http://www.astppbilling.org/

  • Pyfreebilling

    pyfreebilling is an open source wholesale billing platform for FreeSWITCH designed for high performance and scalability. It is suitable for residential, business and wholesale VoIP providers https://www.pyfreebilling.com//

  • A2billing

    A2Billing combined with Asterisk is a physical Telecom Platform and Soft-Switch providing a wide range of telecoms services using both traditional telephone technology or VoIP. It contains a real time billing engine
    http://www.asterisk2billing.org

  • Freeside

    Freeside is the premier open-source billing, CRM, trouble ticketing and provisioning automation software for wired and wireless ISPs, VoIP, hosting, service and content providers and other online businesses.
    http://www.sisd.com/freeside

  • ICTFAX

    An email to fax gateway, supports G.711 faxing , PSTN faxing and T.38 origination and termination .ICTFAX is complete faxing solution
    http://www.ictfax.org

  • CitrusDB

    A Multi-user Billing solution for internet service, subscriptions, consulting, and telecommunications , It Provides CRM, Ticketing, Invoicing, and Credit Card Batches
    http://www.citrusdb.org/

Voip softphones / Clients

  • sipml5

    This is the world's first open source HTML5 SIP client entirely written in javascript for integration in social networks (FaceBook, Twitter, Google+), online games, e-commerce websites, email signatures... No extension, plugin or gateway is needed. The media stack rely on WebRTC.
    The client can be used to connect to any SIP or IMS network from your preferred browser to make and receive audio/video calls and instant messages.
    https://www.doubango.org/sipml5/

  • Webrtc-to-SIP

    Setup for a WEBRTC client and Kamailio server to call SIP clients
    https://github.com/havfo/WEBRTC-to-SIP/

  • Linphone

    Linphone is an open source Voice Over IP phone (or SIP phone) that makes possible to communicate freely with people over the internet, with voice, video, and text instant messaging.

    Linphone makes use of the SIP protocol (an open standard for internet telephony) and can be used with any SIP VoIP operator, including our free SIP audio/video service.

    https://linphone.org/

  • Twinkle

    Twinkle is sip based softphone and support both voice over IP and and instant messaging communications , It is sip standard compliant with advance features including secure voice communications using ZRTP / SRTP.
    http://www.twinklephone.com/

  • Jisti

    Jitsi is a set of open-source projects that allows you to easily build and deploy secure videoconferencing solutions. At the heart of Jitsi are Jitsi Videobridge and Jitsi Meet, which let you have conferences on the internet, while other projects in the community enable other features such as audio, dial-in, recording, and simulcasting.
    It supports HD sound quality and video up to DVD size and quality. https://jitsi.org/

  • Ekiga

    Ekiga 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. http://ekiga.org/

Mobile Clients

  • Csipsimple

    A native android client based open source PJSIP library stack , secure version of Csipsimple also support ZRTP, SIP TLS and SRTP
    http://code.google.com/p/csipsimple/

  • Lumicall

    Open source and convenient app for encrypted phone calls from Android. Uses the SIP protocol to interoperate with other apps and corporate telephone systems.
    https://lumicall.org/

  • Sipdroid

    Sipdroid is native android client developed over PJSIP stack .
    http://sipdroid.org/

  • Siphon

    Siphone is sip compliant Iphone client based on PJSIP SIP stack
    http://code.google.com/p/siphon/