Open Source Mac Softphones

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 Video conferencing with jPBXLite/0.12+ NOTE : THIS PROJECT HAS BEEN SUPERSEDED BY jfPhone.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

Peers is a very simple softphone. It's a SIP User-Agent, written in java, it works on windows, linux and mac. It can be used with SIP servers like opensips or asterisk IPBX. It supports G711 codec (PCMU and PCMA) and telephone-events (DTMF).

wxCommunicator is a cross platform SIP softphone written in C++ utilizing customized sipXtapi user agent library and wxWidgets 2.8.9 GUI library. For a list of supported features see http://wxcommunicator.sourceforge.net/features.html .

jfPhone is a VoIP/SIP SoftPhone for Desktops.

Vodia PBX has included a complete suite of business telephony features for on-premise and cloud-based telephony deployments. Sign up at https://vodia.com/en/register and receive your free Vodia license.

MagnusBilling is a VoIP server system that brings together the best IP Billing Telephony software in the market, creating a comprehensive, flexible and superior tool. Enterprises that need the freedom to build their own VoIP server. You will be able to chose your termination, giving your business larger flexibility and autonomy. Many VoIP distributors use the best of each VoIP server, termination to be able to be more competitive and to have better price and quality. Magnus Sever is softphone integration service installed in LINUX featuring program like: asterisk, freepbx, a2billing, dahdi among others. It allows integration of how many terminations as you wish, the best second track and the best price for each prefix.

The OpenSIPStack Library is an implementation of the Session Initiation Protocol as described in RFC 3261. Applications: * OpenSBC (B2BUA) * OSSPhone (Softphone) Source code is available at http://www.opensipstack.org.

camerubuntu-phone-12.04 est un système ubuntu intégrant la téléphonie IP. Avec ce système Linux, vous disposerez d'un serveur de téléphonie IP Asterisk orienté web qui vous permettra d'effectuer sur votre réseau des appels téléphoniques par ordinateurs, par téléphones IP et mếme en utilisant vos smartphones. Une fois installé, il suffit juste de connecter le PC au réseau, de créer les numéros en utilisant un navigateur, installer un client softphone comme 3CX ou autre dans les PC clients et vous pourrez passer des appels dans votre réseau de façon illimitée.

008 is an open-source event-driven AI powered WebRTC Softphone compatible with macOS, Windows, and Linux. It is also accessible on the web (though official support for browser-related issues is not provided). The name '008' or 'agent 008' reflects our ambition: beyond crafting the premier Open Source Softphone, we aim to introduce a programmable, event-driven AI agent. This agent utilizes embedded artificial intelligence models operating directly on the softphone, ensuring efficiency and reduced operational costs. This project is a WebRTC softphone, and communication is achieved via SIP over a socket. Leading PBX systems like Asterisk or Freeswitch support socket connections. If your provider does not offer this feature, consider using a SIP proxy such as Kamailio, Opensip or Routr. The softphone is internally configured using a JSON definition. The configuration file can be loaded from either a server or a local file. 008 reads the file only once.

connect-rtc.js provides softphone support to AmazonConnect customers when they choose to directly integrate with AmazonConnect API and not use the AmazonConnect web application. It implements Amazon Connect WebRTC signaling protocol and integrates with browser WebRTC APIs to provide a simple contact session interface that can seamlessly integrate with Amazon Connect StreamJS. In a typical amazon-connect-streams integration, connect-rtc-js is not required on parent page. Softphone call handling is done by embedded CCP. Load connect-rtc-js along with amazon-connect-streams on parent page. In the gh-pages branch prebuilt ready-to-use files can be downloaded/linked directly.

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German Stundent Projekt - Java VoIP Softphone

PartiSIPation is a sip user agent which has an easy replaceable gui. So it can be used as a softphone as well as it can be integrated in any kind of application, e.g. online games.

Rabboni VOIP is an Open Source Cross Platform Softphone developed with trolltech QT LIBRARY it uses PJSIP - Open Source SIP Stack

Generic Softphone connector for USB Skype Hardware. Would you like to use your skype usb phone with other softphones? this software is for you.

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.

VoIP Toolkit / Call Control with Integrated Media. High-level Java API for creating SIP enabled VoIP applications. Suitable for either the desktop (softphone, phone applet, incoming call gatekeeper) or server-side (auto attendant, ACD, voicemail).