Open Source Java Internet Phone Software for BSD
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Java Internet Phone Software for BSD
Browse free open source Java Internet Phone Software for BSD and projects below. Use the toggles on the left to filter open source Java Internet Phone Software for BSD by OS, license, language, programming language, and project status.
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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).
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Kablink open team collaboration software uses social networking to unify team workspaces w/ real-time web conferencing. Collaboration for knowledge networking, program management, communities-of-practice, telework, other business process/functional areas
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The XTapi project is aimed at someday growing up to be a reference implementation for JTAPI. At this time it is not the intention to cover every JTAPI package, just the Core and Media packages are covered.
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This is a Java wrapper for Ffmpeg. The JMF (Java Media Framework) plugins system lets one use JMStudio or other application to play H.263/rfc2190 streams. Support for H.263+ and MPEG-4 is currently under test, please contact us for further information.
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Real-Time Collaboration (RTC) and conferencing project. This product has been deprecated. Please see the replacement product, CoolCollaborator, at http://coolcollaboratr.sourceforge.net/
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PNX system develops the middle-ware source code to glue Asterisk with a number of powerful telephony products such as: 1) OpenH323 H.323 stack 2) Vovida SIP stack 3) Bayonne Voice Automation Platform The advantages? An advanced IP PBX supporting the wide
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A C++ framework utilizing Design Patterns for creating Linux and Windows communications applications that contain Dialogic® products. Includes media and network classes (analog, digital, SIP, H323), multithreaded event handling, distributed app support.
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CafeSip - Look what Java and SIP can do
A suite of open-source tools and frameworks for creating SIP apps
Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java. -
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VoIP's project with Speex codec, speech detecting and crypto coding with traffic counting. All code is a pure Java, so completely cross-platforms. Traffic is about 9 MB per hour in both ends, sources: http://www.open-source-soft.narod.ru/arrow.7z
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FreePhone is an easy to use IP phone application. FreePhone supports major platforms and protocols. Features include Conference calling, Voice mail and Chat rooms. A SDK is available for you to make additional features.
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HATEFNA هاتفنا .....http://www.hatefna.org ..... (PC&Mobile)IP SoftPhone over SIP and ASTERISK using JAVA for PC and Symbian C++ for Symbian Mobiles Phones ..... برمجيات للمهاتفة ببروتوكول الإنترنت
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JAGIServer is an Open Source, Multi-Threaded 100% Pure Java AGI Server for use with the Asterisk PBX (TM), providing full AGI call handling functionality. THIS PROJECT HAS BEEN REPLACED BY ORDERLYCALLS
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A set of JNI classes providing direct access to Asterisk PBX functionality from Java. This is not a socket-level interface into the Asterisk manager app but a true Java<->Asterisk integration at the Thread level.
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Jabast connects XMPP Clients to a asterisk PBX to allow extension listening, calls and presence controle for Agents.
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JabbahVoice is an open-source framework which allows developers to easily create and extend voice portals using Asterisk (www.asterisk.org), the famous open-source IP PBX.
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LibDreamSpeak is an example implementation of the TeamSpeak2 protocol (client-side) in Java. Some day it may be used as base for alternative TeamSpeak2 (mobile?) clients.
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This project aim to provide a web based application to plan Conference Call using Asterisk application MeetMe(). The engine is based on Vaadin language to improve the portability of the application, Hibernate and some AGI script based on PHP.
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OpenIM is a instant messenger system utilising the XMPP (Internet-standard communication protocol) and is built on the Java platform. It supports many popular systems (MSN, ICQ, AIM, Jabber, Google Talk etc.) and supports Voice (and Video) Over IP.
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The purpose of this project is to write a plugin for the Openexchange (OX) groupware server (http://www.openexchange.com) that supports various VoIP (SIP) functionality, like direct dialing a SIP telephone, broadcasting, messaging, etc.
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rtusd is a foundation set of real time enabled User Space Devices/Drivers designed to link to HRT high resolution Timer controlled POSIX threads and link re-factored Linux hdw drivers for 2.6.18+. rtusd propogates Linux RT to user space generically.
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With SIP Proxy you will have the opportunity to eavesdrop and manipulate SIP traffic. Furthermore, predefined security test cases can be executed to find weak spots in VoIP devices. Security analysts can add and execute custom test cases.
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A Sip live audio feeding agent. The agent captures audio from sound card and sends live audio stream(uLaw) to caller(sip phone) using RTP. It is based on Peers 0.3(http://peers.sourceforge.net/). Can be used in IP telephony to broadcast live audio.
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SipSpy is a distributed monitoring tool for SIP networks. SpyAgents run on each of the nodes to be monitored, and a SipSpy connects to each of these nodes, receiving information and displaying it in real-time for all the SIP packets monitored.
