Open Source Android VoIP Apps

The best non-root FOSS call recorder app for Android 11+! ShizuCallRecorder empowers ADB through Shizuku to use an advanced list of permissions given to the shell application. It is free, respect your privacy, and try to apply the best security practices. This app main focus are Cellular/VoLTE calls, however, we do try support third-party applications like WhatsApp, Telegram, Line and carriers VoIP call starting Android 14+

ARK-phone is a complete replacement for the Phone and Messages apps on Android, with free, end-to-end encrypted internet calls between friends. Share your ARK code, link a friend's code and call as usual: when the other phone is reachable the call goes over the internet free of charge, otherwise the same call goes through your carrier. No account, phone number or e-mail is needed. The audio is encrypted with WebRTC (DTLS-SRTP) and travels phone to phone; a relay is used only when no direct path exists and cannot decrypt anything. The dialer has a full in-call screen, a T9 keypad, recent calls with filters, contacts with favourites, dual SIM and flexible call blocking. As the default SMS app it handles SMS and MMS, including groups, with quick reply. No ads, no analytics and no tracking. Available in 11 languages. Source code, issues and discussions: https://github.com/jrs8205/ARK-phone

Baresip is a portable and modular SIP User-Agent with audio and video support. the latest source code can be found here: https://github.com/alfredh/baresip

A secure VoIP client for Google Android based on SipDroid (http://sipdroid.org). CounterTap uses SRTP to secure the media stream. Please read the FAQs in the help forum before installing. NB: I will restart dev work later in 2010, pending resources.

Jabbin is an Open Source social application that combines VoIP, Instant Messaging and Social Networking, allowing you to focus on what you really care about: your friends.

The Ozeki VoIP Service Checker is a tool you can use for checking the availability of VoIP services. It can be used to verify if a give service exists, whether it can be reached from a particular network and it checks if the user details, such as the username and password to access that service is correct. This tool was developed because many mobile Internet providers block VoIP traffic and no other tool exists that makes it possible to quickly verify if a certain network support VoIP. The tool is as made available to the public free of charge with the GPL license. The tool is open source. The source code is provided for different platforms. You are free to use, modify, develop or do whatever you wish with this tool.

Project moved to GitHub. https://github.com/davidfischer-ch/smpte2022lib

VitalPBX is a free telephone and communications PBX system for companies. It is a complete platform that can be installed on the physical hardware on the site or as a hosted application. VitalPBX acts as the upper layer interface for the Linux base and then Asterisk (one of the most popular communication toolkits in the world). For this reason, VitalPBX is the graphic user interface between you and the complex world of modern communications. VitalPBX will help you implement a secure telephone system for your company, save, take advantage of recent innovations and provide opportunities to integrate your business processes if you wish to do so.

Client frontend built by Omnitor AB, Sweden. See README for instructions on how to build and run.

jphonelite is a Java SIP VoIP SoftPhone for Desktops (Windows, Linux, Mac) and Android. Features 6 lines with transfer, hold, conference (up to all 6 lines), g711 u/a, g722, g729a, and video (video support in Linux or Windows only and includes H263/H264/VP8). Applet includes full JavaScript support. STUN/TURN/ICE supported. Encrypt media with SRTP. DTLS Key Exchange. Video conferencing with jPBXLite/0.12+ NOTE : THIS PROJECT HAS BEEN SUPERSEDED BY jfPhone. Please go to jfphone.sourceforge.net to get latest version.

This C++ library has been designed as a Chrome SIP stack. Sippet is an open-source SIP User-Agent library, compliant with the IETF RFC 3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and P2P communication services. The main target was to enable Javascript applications to use UDP, TCP and TLS transports along WebSocket. Existing SIP solutions for the browser are forced to use the WebSockets API to send/receive SIP messages. But for many service providers, the WebSocket protocol could not be a feasible solution due to scalability, support and other business constraints. By the way, the stack can be used as a full client-side C++ Stack (full-featured) without Javascript integration.