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Universal Queue and Unified Communications core implementation upon the Platform as a Service paradigm. An abstraction layer for the hardware communications platform among a plugin based model brings independence from technology and media to the project
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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Use this when attachments aren't offered with your email service. (i.e. Proxy Mail). Simply Supply a link to download the 11KB Audio Player and Copy/Paste your Message into the Email. See Website For Demo. http://audiotext.sourceforge.net
[es ] Aplicación web central para particulares. [en ] Core web application for individuals. [fr] Application Web centrale pour les personnes. [pt] Central de aplicação web para indivíduos. [ar] المركزي تطبيق ويب للأفراد
jSipgateConnect library allows Java-based access to all functions specified in the 'Sip Advanced Messaging, Usability, Routing and Accounting Interface' specification.
Castadiva is a test bed based on low-cost off-the-shelf devices, which is used to test protocols developed for MANETs. Castadiva is completely compatible with the file format used by the ns-2 simulator.
TALK provides an efficient and scalable architecture to make CTI more easy and web oriented. TALK includes a generic JTAPI connector, a full AJAX client, a Gadget, and a library to allows your web applications to be connected to your PBX.
Realistic Workplace Simulations that Show Applicant Skills in Action
Skillfully transforms hiring through AI-powered skill simulations that show you how candidates actually perform before you hire them. Our platform helps companies cut through AI-generated resumes and rehearsed interviews by validating real capabilities in action. Through dynamic job specific simulations and skill-based assessments, companies like Bloomberg and McKinsey have cut screening time by 50% while dramatically improving hire quality.
The Java Teamspeak 3 Server Query Software Development Kit is a Java Library that provides Classes to access the Teamspeak Server Query for easy integreating TSSQ into Java Desktop- or Web-Applications.
The project "Sippie" is an additional tool for automatically converting Wireshark-Traces to XML - scenarios, which can be easily used as infile for the OpenSource SIPp test tool and traffic generator.
A simple lightweight instant messaging and VoIP client wirtten in Java. It implements the SIP protocol as described in RFC 3621 and other related RFCs and implementation requierements.
Experimental open source project, which aims to test possibility of VoIP calls "from browser to browser". Main part is Java application implementing UDP hole punching, which is sent to both clients using JWS. See it online in action at project homepage
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
The system is used to control elements of a smart home remotely, with a web or a telephony interface.
Automation can also be created to automatically execute actions based on events.
A secure VoIP client for Google Android based on SipDroid (http://sipdroid.org). CounterTap uses SRTP to secure the media stream. Please read the FAQs in the help forum before installing. NB: I will restart dev work later in 2010, pending resources.
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.
miTester for SIP is an automated SIP testing tool designed and developed to take care of the complex pre-deployment testing of SIP applications easily. This SIP testing tool can be used to simulate SIP call-flows & automate functional, regression tests.
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.