Search Results for "java voip" - Page 2

jSipgateConnect library allows Java-based access to all functions specified in the 'Sip Advanced Messaging, Usability, Routing and Accounting Interface' specification.

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.

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.

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.

I'm sad to say that, unfortunately, the serMyAdmin project is being discontinued starting in September 1st, 2010. After three years of development, the project no longer makes sense. The main reason to create the serMyAdmin was the lack of a GUI for OpenSER/OpenSIPS. The OpenSIPS control panel fulfill this need and it is not wise to maintain two GUI projects. Our team will support the opensips-cp initiative to avoid splitting the community between two open source projects for OpenSIPS GUI. I...

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

Eclipse plugin providing form editing and validations on JAIN SLEE 1.1 deployment descriptors. This plugin requires Eclipse 3.5 (Galileo) or later.

Java SIP stack, originally based on mjsip.

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.

This is a repository for QueueMetrics add-ons that are contribuited by the community and are open-source.

Xen-Agi is an open source java library that allow you to build java applications that interact with the Asterisk Server for VOIP and PBX functionality. This project supports the FastAGI protocol exclusively and is inspired by the Asterisk-Java project.

VoIP Toolkit / Call Control with Integrated Media. High-level Java API for creating SIP enabled VoIP applications. Suitable for either the desktop (softphone, phone applet, incoming call gatekeeper) or server-side (auto attendant, ACD, voicemail).

SIP to Skype Gateway/Bridge/Converter/Adapter. Make and receive SIP to Skype Calls and Skype to SIP Calls. Call Skype users using speed dial or use SkypeOut. Make SIP calls from Skype using a SIP provider or SIP PBX. Use as a Skype Trunk with a PBX.

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.

German Stundent Projekt - Java VoIP Softphone

RAMS is a java based server useful for managing inter-domain VoIP routing, called number translation and Call Detail Record collection. RAMS supports the European Telecommunications Standard Institute's OSP Peering protocol (ETSI TS 101 321).

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.

Java CTI application server for the open-source Asterisk (TM) PBX (includes GUI). OrderlyCalls supports both the AGI / FastAGI & Manager protocols, and includes a WebDeployer that allows you to create your own integrated VOIP and HTML applications.

Open Source implementation of SIP stack according to RFC 3261 and related standards.

Cross-platform Java Teamspeak client

A complete contact center solution for the Asterisk PBX, capable of handling inbound and outbound campaigns. Provides an agent toolbar for CTI. For managers an admin panel controls every aspect of the call center. Stunning historical and realtime stats.

This C++ library has been designed as a Chrome SIP stack. Sippet is an open-source SIP User-Agent library, compliant with the IETF RFC 3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and P2P communication services. The main target was to enable Javascript applications to use UDP, TCP and TLS transports along WebSocket. Existing SIP solutions for the browser are forced to use the WebSockets API to...