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RTP text/t140 Library is a reference implementation for RTP Payload Type for Text Conversation (RFC 4103). The library has source code for encoding and decoding RFC 4103 data, and may be used either as a plug-in to JMF or in a separate RTP sender/receive
Le stelib sono librerie php per costruire dinamicamente pagine web per PC, palmari, cellulari o telefoni IP; queste librerie sono un insieme di classi che ci aiutano a gestire database, PABX VoIP (Asterisk), mootools , phpPlot etc.
trakkcor tracks your position via gps and submits the position data to your server. All you need is a j2me-compliant mobilephone which supports JSR-82 and an external gps receiver.
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).
JSR180 is the standard API spec by jcp.org which is been implemented over Resiprocate Stack. Our purpose is to enable "Open Source Code with Open APIs". By implementing JSR180 we could easily access the SIP Stack by the API documented in the JSR180 Spe
Murmur WebAdministrator is an web interface for end user to control a murmur server ( stand alone mumble server). With Murmur WebAdministrator you can : - start and stop server - edit config - users manager - see logs. READ README FILE !!!
Astercon is an open-source software switching system running over linux. it is built with many best of best technology in linux family, such as asterisk, perl and your favorate web server. it's function include register/sign in, fee counting, calling ca
This project has been superseded by OpalVoip (https://sourceforge.net/projects/opalvoip/) and H323Plus (https://sourceforge.net/projects/h323plus/)
The OpenH323 project provides full featured, interoperable, Open Source implementation of the ITU H.323 teleconferencing protocol that can be used by personal developers and commercial users without charge.
Sydewynder is an SMS receiver and sender application written in Python for S60 phones. It automates the responses of messages and can be used as a mobile application server. It very useful for prototyping mobile applications and educational projects.
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This project contains a SIP stack and server applications built on top of the stack, examples of which are: Stateless Proxy, Registrar,NAT Keepalives,MWI Notifier server. The server applications have been designed to work in association with Asterisk.
AdminsParadise VoIP and Fax server offers a FREE Enterprise grade clustered phone and fax solution for your network for ultimate reliability. Features Web based faxing, scheduled conferencing, voice-mail, roaming users, and much more. Runs Asterisk 1
This is a simple software, comprised of a server and a client, that can be used together with any IP-phone using the SIP protocol (TELE2 and Bredbandsbolget in sweden). It also retrives information, sush as name, adress, etc about the number from eniro.s
SameDesk is a common workspace for web applications. Supporting applications written in AJAX, Java, Flash, & Server-generated HTML, it lets applications across domains work together for the users via a familiar window GUI delivered via web browsers.
SIPStir is a bulk call generator for SIP devices/servers. The server piece is capable of generating an infinite variety of call scenarios based on a simple scripting language. A wxPython client can connect to the server to monitor and modify progress.
VGPS is a PHP/MySQL based provisioning system intended to generate vendor-specific configuration files for Voice-over-IP (VoIP) devices via a generic HTTP API.
H.323 Beacon is a tool that can be used to measure/monitor/qualify/troubleshoot performance of H.323 Videoconference sessions both at the network and at the host by using a end-node(client) and core-node(server) architecture.
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.
Asterisk Manage API and FAGI C++ Framework. Implemented with high efficient OS independent Sockets wrapper (ACE). Provide C++ API wrappers for Asterisk Manager API and FAGI. A Manage API relay(proxy) server and a FAGI socket server will be implemented.
SIPatHome was based on the SIP Express Router (SER) which is a well-tested, configurable, free and RFC3261-compliant SIP proxy. The project was providing VoIP server software for the embedded Linux distribution OpenWRT on affordable Linksys WRT hardware.
Teamspeak Display queries your Teamspeak server and display's the channels and players in it on your website with an option to join the server by clicking a link
TeamSwitch is a perl script connecting an America's Army Operations (AAO) game server to a TeamSpeak (TS) server. TeamSwitch moves players on the right TS channel, according to the AAO team they belong and their status (live/dead).