Awesome Real Time Communications ¶
Protocols and methodology for near simultaneous exchange of media and data.
- 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.
- 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.
- 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 - 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.
- Kazoo - Carrier-grade VoIP API platform using FreeSWITCH and Kamailio.
- FusionPBX - Multitenant system built on top of FreeSWITCH.
- FreePBX - Web Manager for Asterisk.
- 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.
- simple-peer - WebRTC video, voice, and data channels abstraction for Node.js and the browser.
- PeerJS - 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.
- 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.
- 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.
- 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.
- 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.
Contributions welcome! Read the contribution guidelines first.