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rtckit/awesome-rtc

Networking  1 year ago  132
📡 A curated list of awesome Real Time Communications resources
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Nov 11th, 2020

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  • Oct 29th, 2020

    Server Software

    Media Servers

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

    C/C++ Libraries

  • librestars461 - Portable SIP Stack along with companion libraries for media handling, STUN/TURN and a modular user agent.
  • Developer Resources

    PHP Libraries

  • RTCKit/SIPstars16 - RFC 3261 compliant SIP parsing and rendering library for PHP 7.4+.
  • Mar 18th, 2020

    Developer Resources

    Dart Libraries

  • dart-sip-uastars182 - Dart-lang port of JsSIP, capable of SIP over WebSocket.
  • Server Software

    SIP Servers

  • Flexisipstars85 - SIP server suite comprising proxy, presence and group chat functions.
  • Developer Resources

    Rust Libraries

  • libsip - SIP implementation, with a focus towards softphone clients.
  • sipcorestars21 - Rust framework for creating SIP applications.
  • rtcrs/webrtcstars1.4k - WebRTC stack, supporting SDP, RTP, RTCP and SRTP.
  • Mar 10th, 2020

    Developer Resources

    JavaScript Libraries

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

    Python Libraries

  • peerjs-pythonstars51 - Python port of the PeerJS peer-to-peer connection library.
  • Mar 6th, 2020

    Operations

    Monitoring

  • WebRTC Troubleshooter - One stop client-side WebRTC troubleshooter.
  • Feb 27th, 2020

    Events

  • CommCon - Annual conference held in the UK focused on telecommunications in general and WebRTC in particular.
  • Feb 12th, 2020

    Operations

    Monitoring

  • SIP3 - VoIP & RTC traffic monitoring and analysis platform.
  • Feb 6th, 2020

    Developer Resources

    Tutorials

  • Official Website - Entry level WebRTC resources.
  • Nov 19th, 2019

    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.
  • Developer Resources

    C/C++ Libraries

  • Open WebRTC Toolkit - WebRTC development toolkit with bindings for multiple platforms.
  • PJSIP - Multi-protocol RTC library written in C.
  • eXosip - eXtended osip is a mature C library for abstracting the SIP protocol.
  • libdatachannelstars554 - Standalone WebRTC DataChannels C++ implementation.
  • libSRTPstars863 - Secure Real-time Transport Protocol (SRTP) library for C.
  • usrsctpstars445 - Portable Stream Control Transmission Protocol (SCTP) user-land stack.
  • rawrtcstars335 - WebRTC and ORTC library with a small footprint.
  • OSS Corestars19 - General purpose C++ library for Real Time Communications.
  • Events

  • OpenSIPS Summit - Meeting place for the OpenSIPS community.
  • 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.
  • Kranky Geek - AI and RTC event in San Francisco.
  • FOSDEM - Free event for software developers, with a RTC component, held every year in Europe.
  • Server Software

    Media Servers

  • SEMSstars118 - Open source media and application server for SIP based VoIP services.
  • Janus - Lightweight open source, general purpose, WebRTC gateway.
  • RTPProxy - General purpose high performance RTP proxy.
  • RTP:Enginestars498 - RTP and UDP based media traffic proxy, usable as a kernel module.
  • mediasoup - Specialized WebRTC conferencing system.
  • 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 B2BUAstars119 - Back-to-back user agent server written in Python.
  • Server Software

    STUN/TURN

  • coturnstars7k - Fully featured TURN/STUN server supporting multiple platforms.
  • STUNTMANstars1.1k - RFC compliant open source STUN implementation.
  • Operations

    Monitoring

  • sngrepstars664 - Terminal based SIP flow viewer.
  • sipgrepstars141 - Console tool for sniffing, capturing and exploring SIP traffic.
  • rtpbreakstars10 - Detect, reconstruct and analyze RTP sessions.
  • HOMERstars1k - 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.
  • SIPViciousstars607 - Suite of security tools that can be used to audit SIP based VoIP systems.
  • sipsakstars83 - SIP stress and diagnostics utility.
  • Operations

    Billing

  • CGRateS - Carrier grade open source billing/rating server.
  • A2Billing - Billing system for Asterisk for multiple applications.
  • PyFreeBillingstars73 - 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.jsstars2.9k - 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-peerstars5.8k - WebRTC video, voice, and data channels abstraction for Node.js and the browser.
  • Netfluxstars192 - Isomorphic JavaScript peer to peer transport API for client and server.
  • Developer Resources

    Go Libraries

  • Pion - Extensive software stack for WebRTC written in Go.
  • gossipstars291 - SIP stack for stateful user agents written in Go.
  • siprocketstars61 - Fast SIP and SDP packet parser.
  • go-diameterstars182 - RFC compliant Diameter protocol library.
  • Developer Resources

    Python Libraries

  • aiortcstars2.6k - WebRTC and ORTC implementation for Python using asyncio.
  • Kataristars23 - SIP stack application framework.
  • Developer Resources

    Erlang Libraries

  • NkSIPstars332 - Extendable SIP server framework.
  • ersipstars98 - Library comprising building blocks for SIP applications.
  • Blogs

  • 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

  • FreeSWITCH Slack - Join #freeswitch and #freeswitch-dev for user and developer support.
  • discuss-webrtc - Developer oriented Google Group for WebRTC discussions.
  • Last Checked At: 2021-10-25T04:31:30.833Z
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