Open source multi-protocol, cross-platform and software switch.

PBX framework supporting multiple protocols and platforms.

Open source SIP server widely deployed by carriers and providers. Formerly known as OpenSER.

Open source SIP server, tracing its roots in OpenSER (presently Kamailio).

Lightweight SIP proxy, location server, and registrar written in Node.js.

Back-to-back user agent server written in Python.

SIP server suite comprising proxy, presence and group chat functions.

Lightweight open source, general purpose, WebRTC gateway.

Open-source WebRTC infrastructure for building scalable real-time audio and video applications.

General purpose high performance RTP proxy.

RTP and UDP based media traffic proxy, usable as a kernel module.

Specialized WebRTC conferencing system.

Open source media and application server for SIP based VoIP services.

A collection of RTC open source projects, with a focus on conferencing software.

Fully featured TURN/STUN server supporting multiple platforms.

Modern and scalable STUN/TURN server written in Erlang.

NAT type diagnosis CLI. Probes STUN servers, classifies mapping behaviour per RFC 5780, and reports a WebRTC…

RFC compliant open source STUN implementation.

Terminal based SIP flow viewer.

Console tool for sniffing, capturing and exploring SIP traffic.

Detect, reconstruct and analyze RTP sessions.

Multi-protocol capturing and monitoring framework for RTC.

Self-hosted one stop client-side WebRTC troubleshooter.

Exposes client-side NAT traversal debug data.

VoIP & RTC traffic monitoring and analysis platform.

Traffic generator for the SIP protocol.

Suite of security tools that can be used to audit SIP based VoIP systems.

SIP stress and diagnostics utility.

Modern and flexible SIP command line tool.

Tooling for creating lean secure FreeSWITCH Docker images.

Open source programmable-voice/telephony API platform.

Carrier-grade VoIP API platform using FreeSWITCH and Kamailio.

Multitenant system built on top of FreeSWITCH.

Web Manager for Asterisk.

Telecommunication stack built with Node.js.

VoIP API platform built on top of Asterisk, Kamailio and RTPEngine.

Open source CPaaS built for communications service providers.

Multitenant solution for VoIP telephony providers.

Real-time speech infrastructure for voice AI with WebSocket streaming, SIP telephony and pluggable STT/TTS providers.

Carrier grade open source billing/rating server.

Billing system for Asterisk for multiple applications.

Wholesale billing platform for Kamailio and FreeSWITCH.

Entry level WebRTC resources.

WebRTC tutorial by HTML5 Rocks.

Collection of samples demonstrating various parts of the WebRTC APIs.

Comprehensive list of samples by Muaz Khan.

30 minutes step by step live tutorial by Google.

Node.js SIP server framework.

JavaScript shim for abstracting WebRTC spec changes and inconsistencies.

Lightweight open source JavaScript SIP library.

Open source JavaScript SIP client with WebRTC media stack.

WebRTC video, voice, and data channels abstraction for Node.js and the browser.

Isomorphic JavaScript peer to peer transport API for client and server.

Data and media peer-to-peer connection API implemented over WebRTC.

A WebSocket Real-Time Communication (RTC) API framework. Realtime Front-end, Back-end reactivity.

Portable SIP Stack along with companion libraries for media handling, STUN/TURN and a modular user agent.

Multi-protocol RTC library written in C.

eXtended osip is a mature C library for abstracting the SIP protocol.

Standalone WebRTC DataChannels C++ implementation.

C++20 WebRTC media runtime with FFmpeg pipeline, Symple signalling, and RFC 5766 TURN.

Secure Real-time Transport Protocol (SRTP) library for C.

Portable Stream Control Transmission Protocol (SCTP) user-land stack.

WebRTC and ORTC library with a small footprint.

General purpose C++ library for Real Time Communications.

WebRTC development toolkit with bindings for multiple platforms.

Open source SIP library used by FreeSWITCH.

Extensive software stack for WebRTC written in Go.

SIP stack for stateful user agents written in Go.

Fast SIP and SDP packet parser.

RFC compliant Diameter protocol library.

RFC 3261 compliant SIP parsing and rendering library for PHP 7.4+.

WebRTC and ORTC implementation for Python using asyncio.

SIP stack application framework.

Python port of the PeerJS peer-to-peer connection library.

Extendable SIP server framework.

Library comprising building blocks for SIP applications.

SIP implementation, with a focus towards softphone clients.

Rust framework for creating SIP applications.

WebRTC stack, supporting SDP, RTP, RTCP and SRTP.

Dart-lang port of JsSIP, capable of SIP over WebSocket.

Blog by Tsahi Levent-Levi with a strong focus on WebRTC.

Unified communications blog by Andrew Prokop.

WebRTC blog by independent technologists.

Join #freeswitch and #freeswitch-dev for user and developer support.

Developer oriented Google Group for WebRTC discussions.

Annual conference held in Chicago for telecommunications developers. Birthplace of FreeSWITCH.

Berlin hosted annual event focused on Kamailio as well as VoIP, WebRTC, IMS, VoLTE and more.

Asterisk focus event held every year across the US.

Annual conference held in the UK focused on telecommunications in general and WebRTC in particular.

Meeting place for the OpenSIPS community.

AI and RTC event in San Francisco.

Free event for software developers, with a RTC component, held every year in Europe.

JanusCon is a live event for Janus and RTC implementers.

Global hackathon focused on programmable communications.