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

Open source communications soft switches / IP Telephony applications

• FreeSWITCH
• Asterisk
• Kamailio
• OpenSIPS

FreeSWITCH

A Cross-platform, Scalable, Stable Multi-Protocol Soft Switch, The project inspired by renowned 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

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, a 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://goautodial.org/

OSDial

OSDial is a pull-featured predictive dialer solution built over asterisk
http://osdial.com/

Communications Frameworks

ICTCore

ICTCore core is an open-source unified communications framework for developers and integrators to rapidly develop ICT-based applications using their existing development skills. By using ICTCore developers 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
• JsSIP
• SIP.js

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 that leverages the latest developments in SIP and WebRTC to provide a fully-featured SIP endpoint on any website. With JsSIP any website can get Real-Time Communications features using audio, video, and more with just a few lines of code. JsSIP Features are below

• 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 a full-featured SIP stack written in JavaScript. With SIP.js, you can harness the power of WebRTC to build audio, video, and real-time data into your application. SIP.js is fast, lightweight, and easy to use.

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

Broadcasting Platform

ICTDialer

ICTDIALER is an open-source, unified communications autodialer 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.
http://ictdialer.org/

Billing Systems

• Pyfreebilling
• ASTPP
• A2billing
• CDR Stats
• Magnusbilling

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//

ASTPP

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

A2billing

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

CDR Stats

CDR States
CDR-Stats is a web based CDR (Call Data Record) billing mediation platform with call rating and CDR analysis for multiple tenants having the capability to support Asterisk, FreeSWITCH, Kamailio, and almost any other open source and proprietary switch CDR format including Cisco and Alcatel-Lucent.

https://www.cdr-stats.org//

Magnusbilling

Magnusbilling is softswitch and an open-source billing and customer service software solution. MagnusBilling is a fast, secure, efficient, high availability, VOIP Billing, it provides complete WEB panel for the administrator, reseller and customers

/https://www.magnusbilling.org//

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 a complete faxing solution
http://www.ictfax.org

VoIP softphones / Clients

• sipml5
• Webrtc-to-SIP
• Linphone
• Jisti
• Ekiga

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 relies 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 it 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/

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 Video bridge 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
• Lumicall
• Sipdroid
• Siphon

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 a native android client developed over the PJSIP stack.
http://sipdroid.org/

Siphon

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