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awesome-rtc

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

Last synced: about 11 hours ago
JSON representation

  • Operations

    • Web/API Interfaces

      • Eqivo - Open source programmable-voice/telephony API platform.
      • FreePBX - Web Manager for Asterisk.
      • Fonoster - Telecommunication stack built with Node.js.
      • Kazoo - Carrier-grade VoIP API platform using FreeSWITCH and Kamailio.
      • IVOZ Provider - Multitenant solution for VoIP telephony providers.
    • Testing

      • 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.
    • Deployment

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

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

      • 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.
  • Developer Resources

    • JavaScript Libraries

      • drachtio - Node.js SIP server framework.
      • 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

      • libre - Portable SIP Stack along with companion libraries for media handling, STUN/TURN and a modular user agent.
      • eXosip - eXtended osip is a mature C library for abstracting the SIP protocol.
      • libdatachannel - Standalone WebRTC DataChannels C++ implementation.
      • eXosip - eXtended osip is a mature C library for abstracting the SIP protocol.
      • 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.
      • Sofia-SIP - Open source SIP library used by FreeSWITCH.
    • Go Libraries

      • gossip - SIP stack for stateful user agents written in Go.
      • siprocket - Fast SIP and SDP packet parser.
      • go-diameter - RFC compliant Diameter protocol library.
    • PHP Libraries

      • RTCKit/SIP - RFC 3261 compliant SIP parsing and rendering library for PHP 7.4+.
    • Python Libraries

      • peerjs-python - Python port of the PeerJS peer-to-peer connection library.
      • Katari - SIP stack application framework.
      • aiortc - WebRTC and ORTC implementation for Python using asyncio.
    • Erlang Libraries

      • NkSIP - Extendable SIP server framework.
      • ersip - Library comprising building blocks for SIP applications.
    • Tutorials

  • Server Software

    • Media Servers

      • SEMS - Open source media and application server for SIP based VoIP services.
      • RTP:Engine - RTP and UDP based media traffic proxy, usable as a kernel module.
    • STUN/TURN

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

      • Asterisk - PBX framework supporting multiple protocols and platforms.
    • SIP Servers

      • Sippy B2BUA - Back-to-back user agent server written in Python.
      • Flexisip - SIP server suite comprising proxy, presence and group chat functions.
      • Routr - Lightweight SIP proxy, location server, and registrar written in Node.js.
  • 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

      • 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.
      • 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.
      • Kamailio World - Berlin hosted annual event focused on Kamailio as well as VoIP, WebRTC, IMS, VoLTE and more.
      • JanusCon - JanusCon is a live event for Janus and RTC implementers.