Search Results for "java open source" - Page 3

VOIP client/server in python >= 2.6. Audio in/out: ossaudiodev (UNIX like) or SoX. Network: bzip2 compression, speex or ogg audio compression, you can configure all, minimum bytes per second: 350-400 in speex mode: U8, 6 kHz, quality 0, bzip2, buf 4K

Java implementation of MIKEY (Multimedia Internet KEYing, RFC 3830).

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

Fideliphone: High Quality Internet Telephony

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.

THUNDER!Box synchronisiert das Thunderbird-Adressbuch an einen FRITZ!Box DSL-Modem/Router. --- THUNDER!Box synchronizes the Thunderbird addressbook to the FRITZ!Box DSL-Modem/Router.

Fluent is a suite of tools for the management of diverse VoIP networks. Fluent RTE, a reporting and templating engine, is at the core. Add-on modules are available to manage provisioning, CDR, feature sets, routing and more.

This is an initiative to assemble/develop the cheapest, easiest to setup, easiest to manage, scalable, Open Source, standards-based, wireless local do-it-yourself telephone company toolkit in the world.

The web application offers to import call protocols of phone calls and lists of billed calls from different telephone providers. They can be matched to existing call protocol entries to generate cross-provider phone bills.

trixbox CE is an easy to install, VOIP phone system based on the Asterisk PBX. trixbox is designed for home or office use. trixbox CE includes CentOS linux, mysql, and all the tools needed to run a business quality phone system. (formerly asterisk@home)

PJSGUA is an Open Source SIP UAC based on PJSUA reference UAC from PJSIP project

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 is a preconfigured VoIP PBX VM Image based on Asterisk. But provides more advance features, such as LDAP integration, VQ monitoring via RTCP XR reports, LDAP user login, integrated billing system and telephone directory PDF generator.

Plug and play provisioning server for snom VoIP phones written in Python. The server supports the plug and play provisioning feature embedded in snom phones of the snom 3xx, snom8xx and snom MeetingPoint series.

PHP based Viewer for Voice Servers like Mumble.

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.

...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 would like to say thanks to all the people that in some way contributed with the project. The svn code is available to anyone who wants to maintain the project. Most of our development efforts are now focused in a new commercial product based on OpenSIPS called SIP Pulse, completely rewritten in Java and Glassfish www.sippulse.com supporting pre and post paid users with billing and reseller portal.

This is a SIP signaling layer to create a fully operative multipoint (video) conference server using SIP clients and RTP media streams in combination with strManager as a media management layer.

MurMan, the Murmur Manager, is a user friendly, robust, and attractive web management interface for Murmur servers based on PHP, using Murmur's (server component for the Open-Source voice communication Mumble) Ice RCP.

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.