FreeSWITCH

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

Asterisk

PBX framework supporting multiple protocols and platforms.

Kamailio

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

OpenSIPS

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

Routr

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

Sippy B2BUA

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

Flexisip

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

Janus

Lightweight open source, general purpose, WebRTC gateway.

LiveKit

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

RTPProxy

General purpose high performance RTP proxy.

RTP:Engine

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

mediasoup

Specialized WebRTC conferencing system.

SEMS

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

Jitsi

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

coturn

Fully featured TURN/STUN server supporting multiple platforms.

eturnal

Modern and scalable STUN/TURN server written in Erlang.

natcheck

NAT type diagnosis CLI. Probes STUN servers, classifies mapping behaviour per RFC 5780, and reports a WebRTC direct-P2P forecast.

STUNTMAN

RFC compliant open source STUN implementation.

sngrep

Terminal based SIP flow viewer.

sipgrep

Console tool for sniffing, capturing and exploring SIP traffic.

rtpbreak

Detect, reconstruct and analyze RTP sessions.

HOMER

Multi-protocol capturing and monitoring framework for RTC.

WebRTC Troubleshooter

Self-hosted one stop client-side WebRTC troubleshooter.

Trickle ICE

Exposes client-side NAT traversal debug data.

SIP3

VoIP & RTC traffic monitoring and analysis platform.

SIPp

Traffic generator for the SIP protocol.

SIPVicious

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

sipsak

SIP stress and diagnostics utility.

sipexer

Modern and flexible SIP command line tool.

slimswitch

Tooling for creating lean secure FreeSWITCH Docker images.

Eqivo

Open source programmable-voice/telephony API platform.

Kazoo

Carrier-grade VoIP API platform using FreeSWITCH and Kamailio.

FusionPBX

Multitenant system built on top of FreeSWITCH.

FreePBX

Web Manager for Asterisk.

Fonoster

Telecommunication stack built with Node.js.

Wazo

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

jambonz

Open source CPaaS built for communications service providers.

IVOZ Provider

Multitenant solution for VoIP telephony providers.

Sayna

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

CGRateS

Carrier grade open source billing/rating server.

A2Billing

Billing system for Asterisk for multiple applications.

PyFreeBilling

Wholesale billing platform for Kamailio and FreeSWITCH.

Official Website

Entry level WebRTC resources.

Getting Started With WebRTC

WebRTC tutorial by HTML5 Rocks.

WebRTC Samples

Collection of samples demonstrating various parts of the WebRTC APIs.

WebRTC Experiments

Comprehensive list of samples by Muaz Khan.

Interactive Codelab

30 minutes step by step live tutorial by Google.

drachtio

Node.js SIP server framework.

adapter.js

JavaScript shim for abstracting WebRTC spec changes and inconsistencies.

JsSIP

Lightweight open source JavaScript SIP library.

sipML5

Open source JavaScript SIP client with WebRTC media stack.

simple-peer

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

Netflux

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

PeerJS

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

Socio

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

libre

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

PJSIP

Multi-protocol RTC library written in C.

eXosip

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

libdatachannel

Standalone WebRTC DataChannels C++ implementation.

icey

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

libSRTP

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

usrsctp

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

rawrtc

WebRTC and ORTC library with a small footprint.

OSS Core

General purpose C++ library for Real Time Communications.

Open WebRTC Toolkit

WebRTC development toolkit with bindings for multiple platforms.

Sofia-SIP

Open source SIP library used by FreeSWITCH.

Pion

Extensive software stack for WebRTC written in Go.

gossip

SIP stack for stateful user agents written in Go.

siprocket

Fast SIP and SDP packet parser.

go-diameter

RFC compliant Diameter protocol library.

RTCKit/SIP

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

aiortc

WebRTC and ORTC implementation for Python using asyncio.

Katari

SIP stack application framework.

peerjs-python

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

NkSIP

Extendable SIP server framework.

ersip

Library comprising building blocks for SIP applications.

libsip

SIP implementation, with a focus towards softphone clients.

sipcore

Rust framework for creating SIP applications.

rtcrs/webrtc

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

dart-sip-ua

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

BlogGeekMe

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

SIP Adventures

Unified communications blog by Andrew Prokop.

WebRTCHacks

WebRTC blog by independent technologists.

FreeSWITCH Slack

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

discuss-webrtc

Developer oriented Google Group for WebRTC discussions.

ClueCon

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

Kamailio World

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

AstriCon

Asterisk focus event held every year across the US.

CommCon

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

OpenSIPS Summit

Meeting place for the OpenSIPS community.

Kranky Geek

AI and RTC event in San Francisco.

FOSDEM

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

JanusCon

JanusCon is a live event for Janus and RTC implementers.

TADHack

Global hackathon focused on programmable communications.

Awesome RIPT

Real Time Internet Peering for Telephony.

Awesome RTC Hacking

Real Time Communications hacking and penetration testing resources.

Awesome 5G

5G frameworks, libraries, software and resources.

Awesome Cellular Hacking

Research resources in the 3G/4G/5G Cellular security space.

Awesome Telco

Telco resources and projects.

SIP Resources

Useful SIP resources curated by Kamailio's head developer.