Search Results for "java telephony"
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Showing 38 open source projects for "java telephony"
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JavaComm
Desktop communication tool: chat, telephony, multimedia
JavaComm is a client–server communication platform offering chat, telephony, screen sharing, conferencing, a media player, and configurable media streams. Users can define and manage their own streams for flexible playback. All server-based communication runs over HTTPS (SSL/TLS), while file transfers are handled directly between users without passing through the server. The JavaComm Server runs on any Jakarta EE Web Profile 10 platform with the required extensions, using Tomcat 10.1.54 as... -
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oreka
Enterprise telephony recording and retrieval system
Enterprise telephony recording and retrieval system with web based user interface. The project currently supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP and audio sound device and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc... It is amongst others being used in Call... -
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MCU Media Server
SIP Video Multiconference Media Server with WebRTC support.
REPOSITORY MOVED TO GITHUB!! https://github.com/medooze/media-server Video Multiconference Media Server with WebRTC support. Provide Multiconference and video broadcasting services to any SIP service. Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface. JSR309 driver implementation under development. . -
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JAVIER
JAvascript Voicexml InterpretER
JAVIER is a JAvascript Voicexml InterpretER, designed (but not restricted) to run inside a web browser, its main engine has less than 1000 lines of code. It's maybe, the tiniest but (almost) FULL VoiceXML implementation. -
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NoiseGator (Noise Gate)
A simple noise gate app intended for use with VOIPs like Skype.
Ever wanted to cut out background noise when talking with others on Skype? Now it's possible! NoiseGator is a light-weight noise gate application that routes audio through an audio input to an audio output. In real-time the audio level is analysed and if the average level is higher than the threshold the audio bypasses as normal. However, if the average level goes below the threshold, the gate closes and the audio is cut. When used with a virtual audio cable it can act as a noise gate... -
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mod_MumbleLink
Positional Audio Communication for Minecraft with Mumble
A Mod so that Minecraft now natively supports Mumble's positional audio feature. This means: Directional and positionally attenuated VOIP in relation to the game world. Please visit the Forum for information about the newest Version! Main Forum-Thread: http://www.minecraftforum.net/forums/mapping-and-modding/minecraft-mods/1272675 TheSkorm's Fork on GitHub: https://github.com/TheSkorm/mod_mumblelink Mumble: http://mumble.sourceforge.net Donations are greatly appreciated:... -
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Mobicents is the leading Open Source VoIP Platform. It is the First and Only Open Source Certified implementation of JSLEE 1.1 (JSR 240), and SIP Servlets 1.1 (JSR 289). Mobicents also includes a powerful and extensible Media Server.
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SMPTE 2022-1 library for VLC & co
Optimized and cross platform SMPTE 2022 FEC library in C, Python, Java
Project moved to GitHub. https://github.com/davidfischer-ch/smpte2022lib -
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IMS-Communicator is an IMS softphone based on the old version of the sip-communicator java project, implemented on top of the JAIN-SIP stack and the Java Media Framework API. It can be used with Open IMS Core (http://www.openimscore.org).
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Zanzibar is a complete, standards based IVR. It includes an MRCPv2 Server with ASR and TTS engines as well as an voiceXML interpreter so that you can deploy and run voiceXML applications. It integrates with VOIP PBX’s (like Asterisk) using SIP and RTP.
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A JNI wrapper for pjsip. You can use this wrapper to develop Java applications using the pjsip library. At the moment only the pjsua API is implemented. If you would like to obtain a commercial license, or need customisations, please contact us.
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Cairo sets out to provide an enterprise grade, MRCPv2 compliant speech solution utilizing existing open source speech resources such as FreeTTS and Sphinx-4.
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SIP Inspector is a tool written in JAVA to simulate different SIP messages and scenarios. You can create your own SIP signaling scenarios, customize SIP messages and monitor incoming and outgoing messages. The tool can play RTP streams from a pcap fi
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Java GUI for editing the Fritz!Box phone book.
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DTMF detector library and/or application that reads in the specified audio file and returns/outputs whether DTMF detected and/or list of detected digits. May have GUI and command line interfaces.
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Prototype testbed implementation of the IETF Media Server Control (MEDIACTRL) SIP Control Framework, comprehensive of both control and processing functionality (as in IMS MRF, Media Resource Function).
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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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Free Open Source VoiceXML editor programmed in Java (Swing). The VoiceXML document is regularly parsed, a tree view is built and syntax errors are reported in a specific table.
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The SpeakRight Framework is a speech application framework written in Java. SpeakRight applications are fast to create and work on any (VoiceXML) speech platform. Applications are written in Java with full debug and unit testing available.
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VxmlSurfer is a modular, multi-platform, extensible VoiceXML 2.0 interpreter, geared to Web browsing as opposed to telephony. VxmlSurfer is currently used as a user interface for the HearSay non-visual Web browser.
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It is an application for Bluetooth-enabled mobile phones that allows you to connect to other BluetunA users in range and share music recommendations. Mobile music, metadata sharing, Bluetooth applications, proximity-based interactions, social awareness
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vox-attendant is a telephony voice enabled virtual receptionist designed to work with VoiceXML and includes a distribution for the Voxeo Prophecy platform. It provides a web based interface for managing the directory.
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CapisuiteConv is an application for the popular ISDN telecommunication suite capisuite to automate the conversion from the voicebox files to mp3 files in a directory of your choice (maybe a smb share...).
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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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VoxForge collects user-submitted speech audio files for the creation of Acoustic Models for Free and Open Source Speech Recognition Engines such as HTK, Julius, ISIP and Sphinx.