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Unified communications framework and REST API for developers
ICTCore is the communications framework the rest of our open source stack is built on. It sits above FreeSWITCH and gives web developers a REST API for voice calls, fax, SMS and email instead of a dialplan to learn.
The idea is that a communications workflow should be something you write in your own application. You describe what should happen, and ICTCore deals with the switch, the media and the state machine underneath.
It is the engine behind ICTFax and the open source ICTDialer, so...
OpenSIPS (former OpenSER) is an GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP server platforms.
IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips
SIP Video Multiconference Media Server with WebRTC support.
REPOSITORY MOVED TO GITHUB!!
https://github.com/medooze/media-server
Video Multiconference Media Server with WebRTC support.
Provide Multiconference and video broadcasting services to any SIP service.
Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface.
JSR309 driver implementation under development.
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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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Small and effective program for SIP traces anonymization
The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
A good way to design optimization techniques for SIP deployment would be to analyze SIP traffic from existing networks. However, publicly available analyses of SIP traffic are rare and thus not a lot of knowledge exists about typical behavior of a SIP server (as opposed to, for...
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
A suite of open-source tools and frameworks for creating SIP apps
Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java.
sipsak is a command line tool which can send simple requests to a SIP server. It can run additional tests on a SIP server which are usefull for admins and developers of SIP enviroments.
Note: the project is maintained on GitHub https://github.com/nils-ohlmeier/sipsak
This is a simple Sip server with UI. GPL Licensed. Available for Windows OS for the time being. Tested with Windows XP,2003,2008,Windows 7,Vista. Support presence,registar,proxy (stateful/stateless),border controller,lawful interception,early media
The G.O.N.E. is a softphone (or soft phone) running over the web, fully multi-plataform, it implements the SIP protocol, and is built to work on any SIP server, like Asterisk, and others. GONE will work on a complex sistem, but this will be showed a bit