Open Source Voip Software , Top IP Telephony Projects in 2021

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

  • Asterisk

    A software-based, Open Source Converged PBX system that has revolutionized the traditional PBX industry

  • Kamailio

    SIP proxy server, call router, and user agent registration server used in Voice over Internet Protocol and instant messaging applications.

  • 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.

  • Gnugk

    H.323 gatekeeper based on the OpenH323 stack. A gatekeeper provides address translation, admissions control, call routing, authorization and accounting services

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.

  • FreePBX

    A full-featured PBX GUI supporting both asterisk and freeswitch

  • 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

Call Center Solutions

  • GoAutodial

    Goautodial (vicidial) is a professional web based inbound / outbound call center solution built over asterisk

  • OSDial

    OSDial is a pull-featured predictive dialer solution built over asterisk

Communications Frameworks


  • 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.

SIP protocol stack / SIP development Libraries


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

  • 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.

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

  • 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.

Broadcasting Platform

  • ICTDialer

    CTDIALER is an open source, unified communications auto dialer software . ICTDialer is multi-tenant with Voice, SMS and Fax broadcasting capabilities developed over re-known open source Content Management System Drupal and Freeswitch based powerful ICTCore Communication Framework.

  • Billing Systems

    • 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

    • 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

    • 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

    • 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.

    • CitrusDB

      A Multi-user Billing solution for internet service, subscriptions, consulting, and telecommunications , It Provides CRM, Ticketing, Invoicing, and Credit Card Batches

    Fax over IP application Clients

    • ICTFAX

      An email to fax gateway, supports G.711 faxing , PSTN faxing and T.38 origination and termination .ICTFAX is complete faxing solution

    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.

    • Webrtc-to-SIP

      Setup for a WEBRTC client and Kamailio server to call SIP clients

    • 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.

    • 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.

    • 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.

    Mobile Clients