Protocols and methodology for near simultaneous exchange of media and data.

• Server Software
  • General Purpose
  • SIP Servers
  • Media Servers
  • STUN/TURN
• Operations
  • Monitoring
  • Testing
  • Deployment
  • Web/API Interfaces
  • Billing
• Developer Resources
  • Tutorials
  • JavaScript Libraries
  • C/C++ Libraries
  • Go Libraries
  • PHP Libraries
  • Python Libraries
  • Erlang Libraries
  • Rust Libraries
  • Dart Libraries
• Blogs
• Discussion
• Events
• Related Lists
• Contribute

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

• 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 - Self-hosted 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.
• sipexer - Modern and flexible SIP command line tool.

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

• 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.
• Fonoster - 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 - Multitenant solution for VoIP telephony providers.

• 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.
• adapter.js - 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 - 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.
• 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.
• Sofia-SIP - Open source SIP library used by FreeSWITCH.

• 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.

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

• 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.
• JanusCon - JanusCon is a live event for Janus and RTC implementers.
• TADHack - Global hackathon focused on programmable communications.

• Awesome RIPT - Real Time Internet Peering for Telephony.
• Awesome RTC Hacking - Real Time Communications hacking and penetration testing resources.
• Awesome 5G - 5G frameworks, libraries, software and resources.
• Awesome Cellular Hacking - Research resources in the 3G/4G/5G Cellular security space.
• Awesome Telco - Telco resources and projects.
• SIP Resources - Useful SIP resources curated by Kamailio's head developer.

Contributions welcome! Read the contribution guidelines first.