Search Results for "live"

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...

http://www.astchannelslive.com/sourceforge.php This is a dashboard work with Asterisk(FreePBX) any other installation: 1. Showing the peer status 2. queue,agent at real time 3. Hangup, Spy,....

Source code content for Open Source packages used by the ARRIS SecureMedia Encryptonite Server, which provides a streamlined process for service providers to ingest, transcode and securely deliver live and on-demand content.

Project moved to GitHub. https://github.com/davidfischer-ch/smpte2022lib

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.

Cairo sets out to provide an enterprise grade, MRCPv2 compliant speech solution utilizing existing open source speech resources such as FreeTTS and Sphinx-4.

Process flow designer that creates applications (Saflets) for us in the SafiServer engine. SafiWorkshop is used for the design, real time debugging and deployment of advanced flow based applications. Currently tailored for use with Asterisk PBX.

A Client-Server application to make votings with life evaluation via Bluetooth. Server is written in Java for use an any platform. Client is a Java-midlet for use on most mobile phones. It is result of a TU-Ilmenau software project. (See the Credits)

No port forwarding, DMZ, and others are required, Trate allows other applications - almost all applications(VNC,WinSCP,BitTorrent,Filezilla,ssh,Web/Ftp etc) - to auto-create dynamic connections between two peers behind two separate NATs and firewalls.

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.

This project aims to build a software reference implementation of the EBU standard for the transmission of high quality, low latency, audio streams over IP networks. (EBU-tech 3326)

Development of software and alternative firmware for Asterisk-appliance D-Link HorstBox (DVA-G3342SB), free kubuntu live-dvd for firmware development, sourcecode (GPL) for original D-Link firmware, modifications/patches.

ST-PBX Live is a GNU/Debian Linux distribution integrating Asterisk, freePBX and other tools to try from CD or install in HD a complete VoIP system.

The MRCPv2 protocol is designed to allow client devices to control media processing resources, such as speech recognition engines. MRCP4J provides a Java API that encapsulates the MRCPv2 protocol and can be used to implement MRCP clients and/or servers.

Utilities enhance AMP, A@H, Asterisk and Cisco XML. Speed Dialing revised using web interface. Paging using custom meetme routines. Intercom by dialing 0 + Ext #. Browsing AMP's extensions on your phones LCD. Backup patch for Asterisk@Home maint screen.

Kasterx is a live-cd knoppix/debian based distro with asterisk preconfigured with AMP (asterisk management portal). Designed with new users to asterisk in mind to jumpstart you into the world of using linux as a pbx/voip system.

RUMPI: Real Time Multi Peer Interaction. RUMPI want to be as ubiquitous as Flash-browser-plugin, but enabling browsers for real time communication between peers: VoIP (IAX2) and Real-time video Streaming. RUMPI is not a plattform, is a Browser plugin,