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:satellite: A curated list of awesome Real Time Communications resources

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[ Daily / Weekly / Overview ]

Mar 28 - Apr 03, 2022

Server Software / Media Servers

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

Operations / Monitoring

• WebRTC Troubleshooter (⭐440) - Self-hosted one stop client-side WebRTC troubleshooter.

Operations / Testing

• sipexer (⭐142) - Modern and flexible SIP command line tool.

Operations / Deployment

• slimswitch (⭐6) - Tooling for creating lean secure FreeSWITCH Docker images.

Operations / Web/API Interfaces

• Eqivo - Open source programmable-voice/telephony API platform.

• Fonoster (⭐5.4k) - 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 (⭐156) - Multitenant solution for VoIP telephony providers.

Developer Resources / C/C++ Libraries

• Sofia-SIP (⭐141) - Open source SIP library used by FreeSWITCH.

Events / Dart Libraries

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

• 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.

Related Lists / Dart Libraries

• SIP Resources (⭐121) - Useful SIP resources curated by Kamailio's head developer.

Nov 16 - Nov 22, 2020

Related Lists / Dart Libraries

• Awesome RIPT (⭐17) - Real Time Internet Peering for Telephony.

• Awesome RTC Hacking (⭐215) - Real Time Communications hacking and penetration testing resources.

• Awesome 5G (⭐411) - 5G frameworks, libraries, software and resources.

• Awesome Cellular Hacking (⭐2.1k) - Research resources in the 3G/4G/5G Cellular security space.

• Awesome Telco (⭐361) - Telco resources and projects.

Nov 02 - Nov 08, 2020

Developer Resources / C/C++ Libraries

• libre (⭐490) - Portable SIP Stack along with companion libraries for media handling, STUN/TURN and a modular user agent.

Developer Resources / PHP Libraries

• RTCKit/SIP (⭐23) - RFC 3261 compliant SIP parsing and rendering library for PHP 7.4+.

Mar 23 - Mar 29, 2020

Server Software / SIP Servers

• Flexisip (⭐104) - SIP server suite comprising proxy, presence and group chat functions.

Developer Resources / Rust Libraries

• libsip - SIP implementation, with a focus towards softphone clients.

• sipcore (⭐24) - Rust framework for creating SIP applications.

• rtcrs/webrtc (⭐2.3k) - WebRTC stack, supporting SDP, RTP, RTCP and SRTP.

Developer Resources / Dart Libraries

• dart-sip-ua (⭐247) - Dart-lang port of JsSIP, capable of SIP over WebSocket.

Mar 16 - Mar 22, 2020

Developer Resources / JavaScript Libraries

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

Developer Resources / Python Libraries

• peerjs-python (⭐66) - Python port of the PeerJS peer-to-peer connection library.

Feb 17 - Feb 23, 2020

Operations / Monitoring

• SIP3 - VoIP & RTC traffic monitoring and analysis platform.

Feb 10 - Feb 16, 2020

Developer Resources / Tutorials

• Official Website - Entry level WebRTC resources.

Nov 25 - Dec 01, 2019

Server Software / General Purpose

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

• Asterisk - PBX framework supporting multiple protocols and platforms.

Server Software / SIP Servers

• 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 (⭐140) - Back-to-back user agent server written in Python.

Server Software / Media Servers

• Janus - Lightweight open source, general purpose, WebRTC gateway.

• RTPProxy - General purpose high performance RTP proxy.

• RTP:Engine (⭐601) - RTP and UDP based media traffic proxy, usable as a kernel module.

• mediasoup - Specialized WebRTC conferencing system.

• SEMS (⭐132) - Open source media and application server for SIP based VoIP services.

Server Software / STUN/TURN

• coturn (⭐8.3k) - Fully featured TURN/STUN server supporting multiple platforms.

• STUNTMAN (⭐1.2k) - RFC compliant open source STUN implementation.

Operations / Monitoring

• sngrep (⭐779) - Terminal based SIP flow viewer.

• sipgrep (⭐146) - Console tool for sniffing, capturing and exploring SIP traffic.

• rtpbreak (⭐11) - Detect, reconstruct and analyze RTP sessions.

• HOMER (⭐1.2k) - Multi-protocol capturing and monitoring framework for RTC.

• Trickle ICE - Exposes client-side NAT traversal debug data.

Operations / Testing

• SIPp - Traffic generator for the SIP protocol.

• SIPVicious (⭐683) - Suite of security tools that can be used to audit SIP based VoIP systems.

• sipsak (⭐98) - SIP stress and diagnostics utility.

Operations / Web/API Interfaces

• Kazoo - Carrier-grade VoIP API platform using FreeSWITCH and Kamailio.

• FusionPBX - Multitenant system built on top of FreeSWITCH.

• FreePBX - Web Manager for Asterisk.

Operations / Billing

• CGRateS - Carrier grade open source billing/rating server.

• A2Billing - Billing system for Asterisk for multiple applications.

• PyFreeBilling (⭐80) - Wholesale billing platform for Kamailio and FreeSWITCH.

Developer Resources / Tutorials

• 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.

Developer Resources / JavaScript Libraries

• drachtio - Node.js SIP server framework.

• adapter.js (⭐3.3k) - 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 (⭐6.5k) - WebRTC video, voice, and data channels abstraction for Node.js and the browser.

• Netflux (⭐198) - Isomorphic JavaScript peer to peer transport API for client and server.

Developer Resources / C/C++ Libraries

• PJSIP - Multi-protocol RTC library written in C.

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

• libdatachannel (⭐897) - Standalone WebRTC DataChannels C++ implementation.

• libSRTP (⭐992) - Secure Real-time Transport Protocol (SRTP) library for C.

• usrsctp (⭐536) - Portable Stream Control Transmission Protocol (SCTP) user-land stack.

• rawrtc (⭐348) - WebRTC and ORTC library with a small footprint.

• OSS Core (⭐21) - General purpose C++ library for Real Time Communications.

• Open WebRTC Toolkit - WebRTC development toolkit with bindings for multiple platforms.

Developer Resources / Go Libraries

• Pion - Extensive software stack for WebRTC written in Go.

• gossip (⭐313) - SIP stack for stateful user agents written in Go.

• siprocket (⭐66) - Fast SIP and SDP packet parser.

• go-diameter (⭐212) - RFC compliant Diameter protocol library.

Developer Resources / Python Libraries

• aiortc (⭐3.1k) - WebRTC and ORTC implementation for Python using asyncio.

• Katari (⭐29) - SIP stack application framework.

Developer Resources / Erlang Libraries

• NkSIP (⭐337) - Extendable SIP server framework.

• ersip (⭐112) - Library comprising building blocks for SIP applications.

Blogs / Dart Libraries

• 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.

Discussion / Dart Libraries

• FreeSWITCH Slack - Join #freeswitch and #freeswitch-dev for user and developer support.

• discuss-webrtc - Developer oriented Google Group for WebRTC discussions.

Events / Dart Libraries

• 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.

• OpenSIPS Summit - Meeting place for the OpenSIPS community.

• Kranky Geek - AI and RTC event in San Francisco.