awesome-rtc

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

Sippy B2BUA

Back-to-back user agent server written in Python

Flexisip

SIP server suite comprising proxy, presence and group chat functions

Lightweight open source, general purpose, WebRTC gateway

General purpose high performance RTP proxy

RTP:Engine

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

Specialized WebRTC conferencing system

SEMS

Open source media and application server for SIP based VoIP services

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

coturn

Fully featured TURN/STUN server supporting multiple platforms

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

Exposes client-side NAT traversal debug data

VoIP & RTC traffic monitoring and analysis platform

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

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

Fonoster

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

IVOZ Provider

Multitenant solution for VoIP telephony providers

Carrier grade open source billing/rating server

Billing system for Asterisk for multiple applications

PyFreeBilling

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

adapter.js

JavaScript shim for abstracting WebRTC spec changes and inconsistencies

Lightweight open source JavaScript SIP library

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

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

libre

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

libdatachannel

Standalone WebRTC DataChannels C++ implementation

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

WebRTC development toolkit with bindings for multiple platforms

Sofia-SIP

Open source SIP library used by FreeSWITCH

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

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

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

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