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awesome-rtc

:satellite: A curated list of awesome Real Time Communications resources
https://github.com/rtckit/awesome-rtc

Last synced: 5 days ago
JSON representation

  • Server Software

    • General Purpose

      • FreeSWITCH - Open source multi-protocol, cross-platform and software switch.
      • Asterisk - PBX framework supporting multiple protocols and platforms.

    • 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 - Back-to-back user agent server written in Python.
      • Flexisip - SIP server suite comprising proxy, presence and group chat functions.

    • Media Servers

      • Janus - Lightweight open source, general purpose, WebRTC gateway.
      • RTPProxy - General purpose high performance RTP proxy.
      • mediasoup - Specialized WebRTC conferencing system.
      • Jitsi - A collection of RTC open source projects, with a focus on conferencing software.
      • RTP:Engine - RTP and UDP based media traffic proxy, usable as a kernel module.
      • SEMS - Open source media and application server for SIP based VoIP services.

    • STUN/TURN

      • coturn - Fully featured TURN/STUN server supporting multiple platforms.
      • STUNTMAN - RFC compliant open source STUN implementation.

  • Operations

    • Monitoring

      • Trickle ICE - Exposes client-side NAT traversal debug data.
      • SIP3 - VoIP & RTC traffic monitoring and analysis platform.
      • 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.

    • Testing

      • 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.
      • SIPp - Traffic generator for the SIP protocol.

    • Web/API Interfaces

      • 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.
      • Wazo - VoIP API platform built on top of Asterisk, Kamailio and RTPEngine.
      • jambonz - Open source CPaaS built for communications service providers.
      • Fonoster - Telecommunication stack built with Node.js.
      • IVOZ Provider - Multitenant solution for VoIP telephony providers.

    • Billing

      • CGRateS - Carrier grade open source billing/rating server.
      • A2Billing - Billing system for Asterisk for multiple applications.
      • CGRateS - Carrier grade open source billing/rating server.
      • PyFreeBilling - Wholesale billing platform for Kamailio and FreeSWITCH.

    • Deployment

      • slimswitch - Tooling for creating lean secure FreeSWITCH Docker images.

  • Developer Resources

    • Tutorials

    • JavaScript Libraries

      • drachtio - Node.js SIP server framework.
      • JsSIP - Lightweight open source JavaScript SIP library.
      • sipML5 - Open source JavaScript SIP client with WebRTC media stack.
      • PeerJS - Data and media peer-to-peer connection API implemented over WebRTC.
      • adapter.js - JavaScript shim for abstracting WebRTC spec changes and inconsistencies.
      • 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.

    • C/C++ Libraries

      • PJSIP - Multi-protocol RTC library written in C.
      • eXosip - eXtended osip is a mature C library for abstracting the SIP protocol.
      • libre - Portable SIP Stack along with companion libraries for media handling, STUN/TURN and a modular user agent.

    • Go Libraries

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

    • Python Libraries

      • Katari - SIP stack application framework.

    • Rust Libraries

      • 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 Libraries

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

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

  • 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.
      • 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.
      • TADHack - Global hackathon focused on programmable communications.
      • Kranky Geek - AI and RTC event in San Francisco.
      • CommCon - Annual conference held in the UK focused on telecommunications in general and WebRTC in particular.
Programming Languages
C 7 JavaScript 4 C++ 3 Python 3 Shell 2 TypeScript 1 HTML 1 PHP 1 Go 1
Categories
Operations 25 Developer Resources 21 Server Software 15 Events 12 Discussion 4 Blogs 3
Sub Categories
Dart Libraries 20 Web/API Interfaces 8 Monitoring 7 JavaScript Libraries 7 Media Servers 6 Tutorials 5 SIP Servers 5 Testing 5 Billing 4 C/C++ Libraries 3 Rust Libraries 3 STUN/TURN 2 General Purpose 2 Go Libraries 1 Deployment 1 Python Libraries 1
Keywords
sip 11 voip 10 webrtc 8 telephony 5 c 2 javascript 2 rtp 2 nodejs 2 pcap 2 hep 2 flow 2 turn 2 server 2 kamailio 2 security 2 p2p 2 password-cracker 1 hacking-tools 1 audit-sip 1 analytics 1 troubleshooting 1 callflow 1 statistics 1 capture-agent 1 rtc 1 cdr 1 packet-sniffer 1 correlation 1 encapsulation 1 packet-capture 1 opensips 1 monitoring 1 asterisk 1 avaya 1 broadsoft 1 broadworks 1 cisco-webex 1 phone 1 python 1 sip-library 1 sip-server 1 sip-stack 1 sip-uri 1 telecommunications 1 voip-communications 1 voip-server 1 linphone 1 networking 1 debug 1 ncurses 1