Open Source Windows Softphones

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.

Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.

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.

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).

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.

jfPhone is a VoIP/SIP SoftPhone for Desktops.

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 .

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.

OfficeSIP Softphone and Messenger are two enterprise VoIP SIP clients written in C# in .NET Framework. The SIP clients make use of Microsoft UCC API SDK, ensuring the highest quality of audio and video. Compatible with Office Communications Server. See also open source, cross-platform: 1) simple messenger Brief Msg at http://briefmsg.org 2) MUVConf is a multi-user video conferencing, see demo video http://youtu.be/YrBU-Aqtvrk, download https://code.google.com/p/muvconf/downloads/list

ICTAgent is a advance webphone offered as chrome extesion and when installed it will convert all available contacts as click to dial. This extension will embed WebRTC based VoIP Phone in browser which will remain connected to the main server, and will be responsible for inbound and outbound calls, transfer calls, sending DTMF as well sending Fax. It can harvest contacts from the random pages. Agent can access contents, contacts from a single application. Browser can automatically fetch required URL depending on the course of call. Automatic URL will eliminate any delay which agent take while searching and finding desired data. An Embedded Web Phone. It is Operating System Independent, we just need a browser. Click 2 Call. Agent can access contents, contacts and communications from a single application(browser). It will make agent performance much better. It can harvest contacts from random pages / websites/ web applications. CRM Integration. Inbound and Outbound Calls.

TIPcon1 is a standards-compliant soft phone using text, voice and video with accessibility features.

The Bria Desktop API has moved to github at https://github.com/CounterPathAPI. Additional samples for Java and Javascript can be found there. The Bria Desktop allows third-party applications to control Bria for Windows softphone clients. By leveraging the Application API, third-party applications can perform commands such as starting an audio or video call, answering a call or placing a call on hold. CounterPath is now encouraging third-party-application developers to integrate their applications with Bria. Suggested applications range from simple functions such as enabling click-to-call from web pages, to integration with sophisticated Call Center, Customer Relationship Management (CRM) and Health Care applications. This project provides sample and convenient wrapper code to utilize this API. *Note* you need a Bria desktop application to utilize the API.

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.

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

Teleweb is a simple but very efficient call center software. It is the only solution out there that is completely wireless and hosted online. No need for landline, softphone or Voip. All you need is your computer, a headset and you can call the world online.

Softphone SIP com voz, vídeo e Mensagem Instantânea. DTMF, Nat-Traversal, Suporte a Skin(Macromedia Flash), SMS, Historico de Chamadas, Presença, Hold, Auto-Answer e etc. Sip Softphone with Voice, Video and IM.

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.

Cisco IP Telephony plugin for creating CiscoXML structure and Push/Pull to/from Cisco IP phone 7XXX and SoftPhone.

CrossTalk is a VoIP softphone based on sipXtapi.

SIP SoftPhone desenvolvido em Java baseado na JAIN-SIP, JMF e Sip-Communicator 1.0. Possui também o envio de SMS, Agenda, NAT Traversal, DTMF Send/Receive, CallTo e etc. A Java SIP SoftPhone based in JAIN-SIP, JMF and Sip-Comminicator 1.0.

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

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.

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).

The G.O.N.E. is a softphone (or soft phone) running over the web, fully multi-plataform, it implements the SIP protocol, and is built to work on any SIP server, like Asterisk, and others. GONE will work on a complex sistem, but this will be showed a bit