Projects tagged ‘java’ and ‘voice’

Centreon is a network, system, applicative supervision and monitoring tool, it is based upon the most effective Open Source monitoring engine : Nagios. Centreon provides a new frontend and new functionnalities to Nagios. It allows you to be more ... [More] efficient in your network monitoring, but also allows you to make your supervision information readable by a largest range of users. Indeed, a non technical user can now use the Centreon/Nagios couple to easily understand your network infrastructure thanks to charts and graphical representations of the gathered information. Skilled users still have access to specific and technical information collected by Nagios though. [Less]

JMeeting is a next generation type of business or personal meeting. JMeeting will have audio, video and chat conferencing easily set up and viewable. JMeeting will be able to share and create presentations. During the meeting you can click present ... [More] and it will present the presentation or document full screen to all users with a red laser to show important points. [Less]

This is web-based application that takes syndication feeds and turns them into PodCasting-style spoken sources via text-to-speech.

MARF is an open-source research platform and a collection of voice/sound/speech/text and natural language processing (NLP) algorithms written in Java and arranged into a modular and extensible framework facilitating addition of new algorithms. MARF ... [More] can run distributedly over the network and may act as a library in applications or be used as a source for learning and extension. [Less]

free VoiceXML interpreter for JAVA supporting JAVA APIs such as JSAPI and JTAPI. JVoiceXML is an implementation of VoiceXML 2.1, the Voice Extensible Markup Language, specified at http://www.w3.org/TR/2005/CR-voicexml21-20050613/. This is an ... [More] extension to VoiceXML 2.0, specified at http://www.w3.org/TR/voicexml20/ VoiceXML is designed for creating audio dialogs that feature synthesized speech, digitized audio, recognition of spoken and DTMF key input, recording of spoken input, telephony, and mixed initiative conversations. Major goal is to have a platform independent implementation that can be used for free. [Less]

SpeakRight is an open-source Java framework for speech recognition applications in VoiceXML. SpeakRight's "flow objects" are easy to use Java classes that hide the complexity of VoiceXML issues such as prompt escalation, error handling, and platform ... [More] quirks. SpeakRight apps run in Java servlets and produce VoiceXML dynamically. SpeakRight provides a suite of reusable VUI objects and many extension points. Application Development is faster than many proprietary VXML tools. Java IDEs like Eclipse provide code completion, refactoring tools, a great debugger, and unit testing with JUnit. [Less]

High-level Java API for easily creating SIP enabled VoIP applications. Suitable for either a desktop (softphone, phone applet, incoming call screener) or server-side (auto attendant, ACD, voicemail) application. Defines a generic media provider ... [More] interface, and includes an implementation of that interface which uses the Java Media Framework. Flibble-VoIP is the underlying engine of the VoxCenter softphone NEWS 6/11/2008 - Flibble-VoIP Beta 2 has been released. In addition to the addition of the roadmap items below, many fixes were made to STUN, SIP Signaling, media, DTMF, and authentication. Tested with Vonage, VoIP.com, Broad Voice, and IdeaSIP. News - 5/26/2007: I've modified the way flibble-voip is being packaged. This should be less confusing now, and all of the source and binary packages are available (on sourceforge). Fixed an issue with authentication challenges on outgoing calls. This means the PlaceCall example, with two way audio, is functional with VoIP providers, such as Vonage. News - 5/5/2007: Tested with a Vonage softphone account. Can receive calls, verified two way audio! Roadmap Phase 0: (completed rev-50) SIP proxy registration (tested with the SipExchange proxy) (completed rev-50) Place a Call (completed rev-50) End a Call (completed rev-50) Blind Transfer Roadmap BETA 1: (completed rev-81) Media Integration (JMF) (completed rev-81) G711 / PCMU codec (completed rev-81) BYE handling (completed rev-83) Answer Call (completed rev-87) STUN probe for Public IP on init (completed rev-100) Media File Streaming (.WAV / .MP3) (completed rev-100) Allow for Media Source change during a call (completed rev-106) Responding to INVITE authentication challenges (completed rev-118) DTMF receiving (completed rev-122) DTMF Sending Roadmap BETA 2: (completed rev-148) Cancel Call (completed rev-148) Reject Call (completed rev-148) Hold / Unhold (completed rev-148) Echo Suppression (completed rev-148) SIP Keepalives Roadmap RELEASE 1.1: Call Recording Voice Recognition Text-To-Speech Roadmap RELEASE 1.2: Mute local audio playout Mute microphone Transferee Support SIP over TCP GSM Audio Codec Video (JPEG, H263) Consultatvie Transfer Presence Instant Messaging (IM / SIMPLE) [Less]