{"id":483,"url":"https://github.com/rtckit/awesome-rtc","name":"awesome-rtc","description":":satellite: A curated list of awesome Real Time Communications resources","projects_count":100,"last_synced_at":"2026-06-09T08:00:25.325Z","repository":{"id":42677075,"uuid":"222767354","full_name":"rtckit/awesome-rtc","owner":"rtckit","description":":satellite: A curated list of awesome Real Time Communications resources","archived":false,"fork":false,"pushed_at":"2026-05-18T19:39:18.000Z","size":69,"stargazers_count":481,"open_issues_count":2,"forks_count":39,"subscribers_count":18,"default_branch":"master","last_synced_at":"2026-05-18T21:46:30.069Z","etag":null,"topics":["awesome","awesome-list","real-time-communications","rtc","sip","telecommunications","telephony","voip","webrtc"],"latest_commit_sha":null,"homepage":"","language":null,"has_issues":true,"has_wiki":null,"has_pages":null,"mirror_url":null,"source_name":null,"license":"cc0-1.0","status":null,"scm":"git","pull_requests_enabled":true,"icon_url":"https://github.com/rtckit.png","metadata":{"files":{"readme":"README.md","changelog":null,"contributing":"CONTRIBUTING.md","funding":null,"license":"LICENSE","code_of_conduct":null,"threat_model":null,"audit":null,"citation":null,"codeowners":null,"security":null,"support":null,"governance":null,"roadmap":null,"authors":null,"dei":null,"publiccode":null,"codemeta":null,"zenodo":null,"notice":null,"maintainers":null,"copyright":null,"agents":null,"dco":null,"cla":null}},"created_at":"2019-11-19T18:56:07.000Z","updated_at":"2026-05-18T19:39:23.000Z","dependencies_parsed_at":"2026-03-08T03:00:43.573Z","dependency_job_id":null,"html_url":"https://github.com/rtckit/awesome-rtc","commit_stats":null,"previous_names":[],"tags_count":0,"template":false,"template_full_name":null,"purl":"pkg:github/rtckit/awesome-rtc","repository_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories/rtckit%2Fawesome-rtc","tags_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories/rtckit%2Fawesome-rtc/tags","releases_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories/rtckit%2Fawesome-rtc/releases","manifests_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories/rtckit%2Fawesome-rtc/manifests","owner_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/owners/rtckit","download_url":"https://codeload.github.com/rtckit/awesome-rtc/tar.gz/refs/heads/master","sbom_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories/rtckit%2Fawesome-rtc/sbom","scorecard":null,"host":{"name":"GitHub","url":"https://github.com","kind":"github","repositories_count":286080680,"owners_count":34096955,"icon_url":"https://github.com/github.png","version":null,"created_at":"2022-05-30T11:31:42.601Z","updated_at":"2026-05-26T15:22:16.424Z","status":"online","status_checked_at":"2026-06-09T02:00:06.510Z","response_time":63,"last_error":null,"robots_txt_status":"success","robots_txt_updated_at":"2025-07-24T06:49:26.215Z","robots_txt_url":"https://github.com/robots.txt","online":true,"can_crawl_api":true,"host_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub","repositories_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories","repository_names_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repository_names","owners_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/owners"}},"created_at":"2024-01-04T17:24:20.666Z","updated_at":"2026-06-09T08:00:25.326Z","primary_language":null,"list_of_lists":false,"displayable":true,"categories":["Discussion","Operations","Developer Resources","Events","Server Software","Related Lists"],"sub_categories":["Dart Libraries","Web/API Interfaces","Tutorials","C/C++ Libraries","Python Libraries","Monitoring","Media Servers","SIP Servers","Testing","JavaScript Libraries","STUN/TURN","Deployment","Billing","PHP Libraries","Go Libraries","Erlang Libraries","General Purpose"],"readme":"# Awesome Real Time Communications [![Awesome](https://awesome.re/badge.svg)](https://awesome.re)\n\n\u003e Protocols and methodology for near simultaneous exchange of media and data.\n\n\n## Contents\n\n- [Server Software](#server-software)\n  - [General Purpose](#general-purpose)\n  - [SIP Servers](#sip-servers)\n  - [Media Servers](#media-servers)\n  - [STUN/TURN](#stunturn)\n- [Operations](#operations)\n  - [Monitoring](#monitoring)\n  - [Testing](#testing)\n  - [Deployment](#deployment)\n  - [Web/API Interfaces](#webapi-interfaces)\n  - [Billing](#billing)\n- [Developer Resources](#developer-resources)\n  - [Tutorials](#tutorials)\n  - [JavaScript Libraries](#javascript-libraries)\n  - [C/C++ Libraries](#cc-libraries)\n  - [Go Libraries](#go-libraries)\n  - [PHP Libraries](#php-libraries)\n  - [Python Libraries](#python-libraries)\n  - [Erlang Libraries](#erlang-libraries)\n  - [Rust Libraries](#rust-libraries)\n  - [Dart Libraries](#dart-libraries)\n- [Blogs](#blogs)\n- [Discussion](#discussion)\n- [Events](#events)\n- [Related Lists](#related-lists)\n- [Contribute](#contribute)\n\n\n## Server Software\n\n### General Purpose\n\n- [FreeSWITCH](http://freeswitch.org) - Open source multi-protocol, cross-platform and software switch.\n- [Asterisk](http://asterisk.org) - PBX framework supporting multiple protocols and platforms.\n\n### SIP Servers\n\n- [Kamailio](http://www.kamailio.org) - Open source SIP server widely deployed by carriers and providers. Formerly known as OpenSER.\n- [OpenSIPS](http://www.opensips.org) - Open source SIP server, tracing its roots in OpenSER (presently Kamailio).\n- [Routr](https://routr.io) - Lightweight SIP proxy, location server, and registrar written in Node.js.\n- [Sippy B2BUA](https://github.com/sippy/b2bua) - Back-to-back user agent server written in Python.\n- [Flexisip](https://github.com/BelledonneCommunications/flexisip) - SIP server suite comprising proxy, presence and group chat functions.\n\n### Media Servers\n\n- [Janus](https://janus.conf.meetecho.com) - Lightweight open source, general purpose, WebRTC gateway.\n- [LiveKit](https://livekit.io) - Open-source WebRTC infrastructure for building scalable real-time audio and video applications.\n- [RTPProxy](https://www.rtpproxy.org) - General purpose high performance RTP proxy.\n- [RTP:Engine](https://github.com/sipwise/rtpengine) - RTP and UDP based media traffic proxy, usable as a kernel module.\n- [mediasoup](https://mediasoup.org) - Specialized WebRTC conferencing system.\n- [SEMS](https://github.com/sems-server/sems) - Open source media and application server for SIP based VoIP services.\n- [Jitsi](https://jitsi.org/projects) - A collection of RTC open source projects, with a focus on conferencing software.\n\n### STUN/TURN\n\n- [coturn](https://github.com/coturn/coturn) - Fully featured TURN/STUN server supporting multiple platforms.\n- [eturnal](https://eturnal.net/) - Modern and scalable STUN/TURN server written in Erlang.\n- [natcheck](https://github.com/1mb-dev/natcheck) - NAT type diagnosis CLI. Probes STUN servers, classifies mapping behaviour per RFC 5780, and reports a WebRTC direct-P2P forecast.\n- [STUNTMAN](https://github.com/jselbie/stunserver) - RFC compliant open source STUN implementation.\n\n\n\t## Operations\n\n### Monitoring\n\n- [sngrep](https://github.com/irontec/sngrep) - Terminal based SIP flow viewer.\n- [sipgrep](https://github.com/sipcapture/sipgrep) - Console tool for sniffing, capturing and exploring SIP traffic.\n- [rtpbreak](https://github.com/Naishy/rtpsplit) - Detect, reconstruct and analyze RTP sessions.\n- [HOMER](https://github.com/sipcapture/homer) - Multi-protocol capturing and monitoring framework for RTC.\n- [WebRTC Troubleshooter](https://github.com/webrtc/testrtc) - Self-hosted one stop client-side WebRTC troubleshooter.\n- [Trickle ICE](https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice) - Exposes client-side NAT traversal debug data.\n- [SIP3](https://sip3.io) - VoIP \u0026 RTC traffic monitoring and analysis platform.\n\n### Testing\n\n- [SIPp](http://sipp.sourceforge.net) - Traffic generator for the SIP protocol.\n- [SIPVicious](https://github.com/EnableSecurity/sipvicious) - Suite of security tools that can be used to audit SIP based VoIP systems.\n- [sipsak](https://github.com/nils-ohlmeier/sipsak) - SIP stress and diagnostics utility.\n- [sipexer](https://github.com/miconda/sipexer) - Modern and flexible SIP command line tool.\n\n### Deployment\n\n- [slimswitch](https://github.com/rtckit/slimswitch) - Tooling for creating lean secure FreeSWITCH Docker images.\n\n### Web/API Interfaces\n\n- [Eqivo](https://eqivo.org) - Open source programmable-voice/telephony API platform.\n- [Kazoo](https://www.2600hz.org) - Carrier-grade VoIP API platform using FreeSWITCH and Kamailio.\n- [FusionPBX](https://www.fusionpbx.com) - Multitenant system built on top of FreeSWITCH.\n- [FreePBX](https://www.freepbx.org) - Web Manager for Asterisk.\n- [Fonoster](https://github.com/fonoster/fonoster) - Telecommunication stack built with Node.js.\n- [Wazo](https://wazo-platform.org) - VoIP API platform built on top of Asterisk, Kamailio and RTPEngine.\n- [jambonz](https://www.jambonz.org) - Open source CPaaS built for communications service providers.\n- [IVOZ Provider](https://github.com/irontec/ivozprovider) - Multitenant solution for VoIP telephony providers.\n- [Sayna](https://github.com/SaynaAI/sayna) - Real-time speech infrastructure for voice AI with WebSocket streaming, SIP telephony and pluggable STT/TTS providers.\n\n### Billing\n\n- [CGRateS](http://cgrates.org) - Carrier grade open source billing/rating server.\n- [A2Billing](http://www.asterisk2billing.org) - Billing system for Asterisk for multiple applications.\n- [PyFreeBilling](https://github.com/mwolff44/pyfreebilling) - Wholesale billing platform for Kamailio and FreeSWITCH.\n\n\n## Developer Resources\n\n### Tutorials\n\n- [Official Website](https://webrtc.org) - Entry level WebRTC resources.\n- [Getting Started With WebRTC](https://www.html5rocks.com/en/tutorials/webrtc/basics) - WebRTC tutorial by HTML5 Rocks.\n- [WebRTC Samples](https://webrtc.github.io/samples) - Collection of samples demonstrating various parts of the WebRTC APIs.\n- [WebRTC Experiments](https://www.webrtc-experiment.com) - Comprehensive list of samples by Muaz Khan.\n- [Interactive Codelab](https://codelabs.developers.google.com/codelabs/webrtc-web) - 30 minutes step by step live tutorial by Google.\n\n### JavaScript Libraries\n\n- [drachtio](https://drachtio.org) - Node.js SIP server framework.\n- [adapter.js](https://github.com/webrtcHacks/adapter) - JavaScript shim for abstracting WebRTC spec changes and inconsistencies.\n- [JsSIP](http://jssip.net) - Lightweight open source JavaScript SIP library.\n- [sipML5](https://www.doubango.org/sipml5) - Open source JavaScript SIP client with WebRTC media stack.\n- [simple-peer](https://github.com/feross/simple-peer) - WebRTC video, voice, and data channels abstraction for Node.js and the browser.\n- [Netflux](https://github.com/coast-team/netflux) - Isomorphic JavaScript peer to peer transport API for client and server.\n- [PeerJS](https://peerjs.com) - Data and media peer-to-peer connection API implemented over WebRTC.\n- [Socio](https://github.com/Rolands-Laucis/Socio) - A WebSocket Real-Time Communication (RTC) API framework. Realtime Front-end, Back-end reactivity.\n\n### C/C++ Libraries\n\n- [libre](https://github.com/creytiv/re) - Portable SIP Stack along with companion libraries for media handling, STUN/TURN and a modular user agent.\n- [PJSIP](https://www.pjsip.org) - Multi-protocol RTC library written in C.\n- [eXosip](http://savannah.nongnu.org/projects/exosip) - eXtended osip is a mature C library for abstracting the SIP protocol.\n- [libdatachannel](https://github.com/paullouisageneau/libdatachannel) - Standalone WebRTC DataChannels C++ implementation.\n- [icey](https://github.com/nilstate/icey) - C++20 WebRTC media runtime with FFmpeg pipeline, Symple signalling, and RFC 5766 TURN.\n- [libSRTP](https://github.com/cisco/libsrtp) - Secure Real-time Transport Protocol (SRTP) library for C.\n- [usrsctp](https://github.com/sctplab/usrsctp) - Portable Stream Control Transmission Protocol (SCTP) user-land stack.\n- [rawrtc](https://github.com/rawrtc/rawrtc) - WebRTC and ORTC library with a small footprint.\n- [OSS Core](https://github.com/joegen/oss_core) - General purpose C++ library for Real Time Communications.\n- [Open WebRTC Toolkit](https://01.org/open-webrtc-toolkit) - WebRTC development toolkit with bindings for multiple platforms.\n- [Sofia-SIP](https://github.com/freeswitch/sofia-sip) - Open source SIP library used by FreeSWITCH.\n\n### Go Libraries\n\n- [Pion](https://pion.ly) - Extensive software stack for WebRTC written in Go.\n- [gossip](https://github.com/StefanKopieczek/gossip) - SIP stack for stateful user agents written in Go.\n- [siprocket](https://github.com/marv2097/siprocket) - Fast SIP and SDP packet parser.\n- [go-diameter](https://github.com/fiorix/go-diameter) - RFC compliant Diameter protocol library.\n\n### PHP Libraries\n\n- [RTCKit/SIP](https://github.com/rtckit/php-sip) - RFC 3261 compliant SIP parsing and rendering library for PHP 7.4+.\n\n### Python Libraries\n\n- [aiortc](https://github.com/aiortc/aiortc) - WebRTC and ORTC implementation for Python using asyncio.\n- [Katari](https://github.com/hyperioxx/Katari) - SIP stack application framework.\n- [peerjs-python](https://github.com/ambianic/peerjs-python) - Python port of the PeerJS peer-to-peer connection library.\n\n### Erlang Libraries\n\n- [NkSIP](https://github.com/NetComposer/nksip) - Extendable SIP server framework.\n- [ersip](https://github.com/poroh/ersip) - Library comprising building blocks for SIP applications.\n\n### Rust Libraries\n\n- [libsip](https://docs.rs/libsip/0.2.4/libsip) - SIP implementation, with a focus towards softphone clients.\n- [sipcore](https://github.com/armatusmiles/sipcore) - Rust framework for creating SIP applications.\n- [rtcrs/webrtc](https://github.com/rtcrs/webrtc) - WebRTC stack, supporting SDP, RTP, RTCP and SRTP.\n\n### Dart Libraries\n\n- [dart-sip-ua](https://github.com/cloudwebrtc/dart-sip-ua) - Dart-lang port of JsSIP, capable of SIP over WebSocket.\n\n\n## Blogs\n\n- [BlogGeekMe](https://bloggeek.me/blog) - Blog by Tsahi Levent-Levi with a strong focus on WebRTC.\n- [SIP Adventures](https://andrewjprokop.wordpress.com) - Unified communications blog by Andrew Prokop.\n- [WebRTCHacks](https://webrtchacks.com) - WebRTC blog by independent technologists.\n\n\n## Discussion\n\n- [FreeSWITCH Slack](https://signalwire.community) - Join #freeswitch and #freeswitch-dev for user and developer support.\n- [discuss-webrtc](https://groups.google.com/forum/?fromgroups#!forum/discuss-webrtc) - Developer oriented Google Group for WebRTC discussions.\n\n\n## Events\n\n- [ClueCon](http://cluecon.com) - Annual conference held in Chicago for telecommunications developers. Birthplace of FreeSWITCH.\n- [Kamailio World](https://www.kamailioworld.com) - Berlin hosted annual event focused on Kamailio as well as VoIP, WebRTC, IMS, VoLTE and more.\n- [AstriCon](https://www.asterisk.org/community/astricon-user-conference) - Asterisk focus event held every year across the US.\n- [CommCon](https://commcon.xyz) - Annual conference held in the UK focused on telecommunications in general and WebRTC in particular.\n- [OpenSIPS Summit](https://www.opensips.org/events) - Meeting place for the OpenSIPS community.\n- [Kranky Geek](https://krankygeek.com) - AI and RTC event in San Francisco.\n- [FOSDEM](https://fosdem.org) - Free event for software developers, with a RTC component, held every year in Europe.\n- [JanusCon](https://www.januscon.it) - JanusCon is a live event for Janus and RTC implementers.\n- [TADHack](https://tadhack.com) - Global hackathon focused on programmable communications.\n\n\n## Related Lists\n\n- [Awesome RIPT](https://github.com/rtckit/awesome-ript) - Real Time Internet Peering for Telephony.\n- [Awesome RTC Hacking](https://github.com/EnableSecurity/awesome-rtc-hacking) - Real Time Communications hacking and penetration testing resources.\n- [Awesome 5G](https://github.com/calee0219/awesome-5g) - 5G frameworks, libraries, software and resources.\n- [Awesome Cellular Hacking](https://github.com/W00t3k/Awesome-Cellular-Hacking) - Research resources in the 3G/4G/5G Cellular security space.\n- [Awesome Telco](https://github.com/ravens/awesome-telco) - Telco resources and projects.\n- [SIP Resources](https://github.com/miconda/sip-resources) - Useful SIP resources curated by Kamailio's head developer.\n\n\n## Contribute\n\nContributions welcome! Read the [contribution guidelines](CONTRIBUTING.md) first.\n","projects_url":"https://awesome.ecosyste.ms/api/v1/lists/rtckit%2Fawesome-rtc/projects"}