The VoIP Users Conference may not be around, but the spirit and people live on.

Open Source Telephony

Asterisk is an open source framework for building communications applications.
The Node.js framework for SIP server applications.
FreePBX is a web-based open source GUI (graphical user interface) that controls and manages Asterisk (PBX), an open source communication server.
FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware.
FusionPBX can be used as a highly available single or domain based multi-tenant PBX, carrier grade switch, call center server, fax server, voip server, voicemail server, conference server, voice application server, appliance framework and more.
HOMER is part of the SIPCAPTURE stack: A robust, carrier-grade and modular VoIP and RTC Capture Framework for Analysis and Monitoring with native support for all major OSS Voice platforms and vendor-agnostic Capture agents.
Janus is a WebRTC Server developed by Meetecho conceived to be a general purpose one.
Multi-platform open-source video conferencing
Kamailio® (successor of former OpenSER and SER) is an Open Source SIP Server released under GPL, able to handle thousands of call setups per second.
OpenSIPS is an Open Source SIP proxy/server for voice, video, IM, presence and any other SIP extensions.
The Modern Stack for Web Real-Time Communication.
PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE.
Sofia SIP
Sofia-SIP - a RFC3261 compliant SIP User-Agent library.

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© 2020 by Fred Posner. Contact through Always be kind.