Search Results for "sofia-sip"
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55 programs for "sofia-sip" with 1 filter applied:
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Android Contacts, Reborn
Android Contacts API Library written in Kotlin
This library provides a complete set of APIs to do everything you need with Contacts in Android. You no longer have to deal with the Contacts Provider, database operations, and cursors. Whether you just need to get all or some Contacts for a small part of your app (written in Kotlin or Java), or you are looking to create your own full-fledged Contacts app with the same capabilities as the AOSP Android Contacts app and Google Contacts app, this library is for you. -
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wxPython Project Phoenix
wxPython's Project Phoenix. A new implementation of wxPython
wxPython, the cross-platform GUI toolkit for the Python language. With wxPython software developers can create truly native user interfaces for their Python applications, that run with little or no modifications on Windows, Macs and Linux or other Unix-like systems. Welcome to wxPython's Project Phoenix! Phoenix is the improved next-generation wxPython, "better, stronger, faster than he was before." This new implementation is focused on improving speed, maintainability and extensibility.... -
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A C++ framework utilizing Design Patterns for creating Linux and Windows communications applications that contain Dialogic® products. Includes media and network classes (analog, digital, SIP, H323), multithreaded event handling, distributed app support.
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Elastix
Unified Communications Server
Elastix is a software-based PBX powered by 3CX and based on Debian. An open-standards solution, Elastix is an easy to install and manage UC system compatible with popular IP phones, gateways and SIP trunks. Elastix is complete with unified communications features such as integrated WebRTC video conferencing, chat, presence and softphones and smartphone clients for Windows, Mac, iOS and Android. -
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jfBroadcast
VoIP/SIP Autodialer
VoIP/SIP AutoDialer. Broadcasts a message with the option to transfer person to another number. Now includes new survey options. -
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oreka
Enterprise telephony recording and retrieval system
Enterprise telephony recording and retrieval system with web based user interface. The project currently supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP and audio sound device and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc... It is amongst others being used in Call... -
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CRUSH
Data reduction and imaging for select astronomical cameras
CRUSH is a pioneering reduction and imaging suite for many astronomical cameras operating at far-infrared, submillimeter and millimeter wavelengths, such as SHARC-2, SCUBA-2, LABOCA, GISMO, or SOFIA/HAWC+ (to name a few). -
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wxCommunicator is a cross platform SIP softphone written in C++ utilizing customized sipXtapi user agent library and wxWidgets 2.8.9 GUI library. For a list of supported features see http://wxcommunicator.sourceforge.net/features.html .
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jpbxlite
Java VoIP/SIP PBX system (replaced by jfPBX)
jPBXLite is a VoIP/SIP PBX. Supports SIP extensions, voicemail, trunks, conferences, queues (ACD) and an IVR system. Support video conferencing with jPhoneLite/1.4.0. NOTE:THIS PROJECT WAS RENAMED AND IS NOW jfPBX. Please go to jfpbx.sourceforge.net -
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Mobicents is the leading Open Source VoIP Platform. It is the First and Only Open Source Certified implementation of JSLEE 1.1 (JSR 240), and SIP Servlets 1.1 (JSR 289). Mobicents also includes a powerful and extensible Media Server.
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Git repo: https://github.com/asipto/siremis Web management interface for Kamailio (OpenSER) - handle subscriber profiles, access control lists, accounting, least cost routing and load balancing, monitoring charts, xmlrpc communication with SIP server
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Peers is a very simple softphone. It's a SIP User-Agent, written in java, it works on windows, linux and mac. It can be used with SIP servers like opensips or asterisk IPBX. It supports G711 codec (PCMU and PCMA) and telephone-events (DTMF).
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baresip
Baresip is a modular SIP User-Agent with audio and video support
Baresip is a portable and modular SIP User-Agent with audio and video support. the latest source code can be found here: https://github.com/alfredh/baresip -
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KAMAILIO (OpenSER) - robust, secure and scalable Open Source (GPL) SIP (RFC3261) server implementation with large features set (over 90 extension modules). As of May 2009, source code is hosted by GIT repository at http://sip-router.org
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LuaSofia is a Lua binding of Sofia-Sip library. Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification. We decided to use git, source code can be get at: git://github.com/ppizarro/luasofia.git
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dtmfbox is a small softswitch application (SIP/CAPI), that can be used to control different tasks over telephone keyboard via DTMF. Mostly, it was made to run on the AVM FRITZ!Box 7170 (mipsel) but works under Unix/Linux and Windows, too.
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PartiSIPation is a sip user agent which has an easy replaceable gui. So it can be used as a softphone as well as it can be integrated in any kind of application, e.g. online games.
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This is an university project for the Software Technologies course in the Faculty of Mathematics and Informatics department of the Sofia University. It will present a repository for agile projects that will allow creation and search for projects.
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SerMyAdmin
Web interface for OpenSIPS
... 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. -
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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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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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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.