Open Source Mobile Operating Systems VoIP Software

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 Video conferencing with jPBXLite/0.12+ NOTE : THIS PROJECT HAS BEEN SUPERSEDED BY jfPhone.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.

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.

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

Astmontray is a systray applet written in PyQt so users can monitor whether their PABX servers running Asterisk.org, the open source telephony platform, are up or not. It supports AMI authentication and runs fine on Linux, OSX and Maemo tablets.

COMONet is a novel concept in which a voice call is routed via an ad hoc mobile network avoiding the conventional Mobile Service Provider. COMONet is consisting of community of mobile users who stay between two mobile users who are making the call.

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.

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.

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.