Search Results for "debian gnu linux" - Page 6
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Showing 189 open source projects for "debian gnu linux"
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Open Unified Recording (OUR) is a full featured Linux based Open Source VoIP/SIP Call Recording engine, indexing and retrieval system. The system resides on the network and passively captures SIP sessions. Project is sponsored by UnifiedRecording.com
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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.
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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.
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A modularized Linux platform optimized for embedded hardware hosting various network appliance packages (i.e. Router,Voip PBX,NAS,etc..) with an easy to use web interface for customization.
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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
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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
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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.
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Fast & Lightweight PHP Based GUI For Asterisk opensource IP-PBX and doesnot requires any database. Designed to be used with Embedded Appliance or PC Based Solution. Its is different than Asterisk GUI & FreePBX ,no AJAX or MYSQL.
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This project provides a meter of QoS on VoIP using the protocol SIP Media Loopback
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QoS Number is a project to control de quality of service ( QoS ) in a IP-Centrex Plattform .
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An XCAP server is used by XCAP clients to store data like buddy lists and presence policy in combination with a SIP Presence server that supports PUBLISH, SUBSCRIBE and NOTIFY methods to provide a complete SIP SIMPLE server solution.
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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.
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The goal of this project is to provide a liveCD allowing the user to analyze networks for VoIP installations. This project gives you a global network state.
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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.
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8ix Zenith CE spells Asterisk derived IP Telephony solution.When you want robust communication, without the huge cost and recurring bill that comes with the way you do business, 8ix Zenith CE addresses your every communication need.
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Astmontray is a systray applet written in PyQt so users can monitor whether their PABX servers running Asterisk.org, the open source telephony platform, are up or not. It supports AMI authentication and runs fine on Linux, OSX and Maemo tablets.
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A command line SIP/H323 softphone capable of sending and receiving audio files as well as sending out of band DTMF digits. Supports a BNF format configuration language for scripting call scenarios. Useful for example system testing.
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CHAROZT is an Asterisk MOH (Music on Hold) Uploader. It is a Linux-based application installed in Linux desktops that converts MP3 to WAV for Asterisk supported MOH. It can also upload converted files to the Asterisk Server.
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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.
