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--- visibility: public --- # Real-Time Communications **repo:** [rtckit/awesome-rtc](https://github.com/rtckit/awesome-rtc) **category:** [[networking|Networking]] **related:** [[pcaptools|Pcaptools]] · [[software-defined-networking|Software Defined Networking]] --- # Awesome Real Time Communications [](https://awesome.re) > Protocols and methodology for near simultaneous exchange of media and data. ## Contents - [Server Software](#server-software) - [General Purpose](#general-purpose) - [SIP Servers](#sip-servers) - [Media Servers](#media-servers) - [STUN/TURN](#stunturn) - [Operations](#operations) - [Monitoring](#monitoring) - [Testing](#testing) - [Deployment](#deployment) - [Web/API Interfaces](#webapi-interfaces) - [Billing](#billing) - [Developer Resources](#developer-resources) - [Tutorials](#tutorials) - [JavaScript Libraries](#[javascript](/@harrisonqian/awesome/wiki/programming-languages/javascript)-libraries) - [C/C++ Libraries](#cc-libraries) - [Go Libraries](#go-libraries) - [PHP Libraries](#php-libraries) - [Python Libraries](#[python](/@harrisonqian/awesome/wiki/programming-languages/python)-libraries) - [Erlang Libraries](#[erlang](/@harrisonqian/awesome/wiki/programming-languages/erlang)-libraries) - [Rust Libraries](#[rust](/@harrisonqian/awesome/wiki/programming-languages/rust)-libraries) - [Dart Libraries](#[dart](/@harrisonqian/awesome/wiki/programming-languages/dart)-libraries) - [Blogs](#blogs) - [Discussion](#discussion) - [Events](#events) - [Related Lists](#related-lists) - [Contribute](#contribute) ## Server Software ### General Purpose - [FreeSWITCH](http://freeswitch.org) - Open source multi-protocol, [cross-platform](/@harrisonqian/awesome/wiki/platforms/cross-platform) and software switch. - [Asterisk](http://asterisk.org) - PBX framework supporting multiple protocols and platforms. ### SIP Servers - [Kamailio](http://www.kamailio.org) - Open source SIP server widely deployed by carriers and providers. Formerly known as OpenSER. - [OpenSIPS](http://www.opensips.org) - Open source SIP server, tracing its roots in OpenSER (presently Kamailio). - [Routr](https://routr.io) - Lightweight SIP proxy, location server, and registrar written in [Node.js](/@harrisonqian/awesome/wiki/platforms/node-js). - [Sippy B2BUA](https://github.com/sippy/b2bua) - Back-to-back user agent server written in [Python](/@harrisonqian/awesome/wiki/programming-languages/python). - [Flexisip](https://github.com/BelledonneCommunications/flexisip) - SIP server suite comprising proxy, presence and group chat functions. ### Media Servers - [Janus](https://janus.conf.meetecho.com) - Lightweight open source, general purpose, WebRTC gateway. - [LiveKit](https://livekit.io) - Open-source WebRTC infrastructure for building scalable real-time audio and video applications. - [RTPProxy](https://www.rtpproxy.org) - General purpose high performance RTP proxy. - [RTP:Engine](https://github.com/sipwise/rtpengine) - RTP and UDP based media traffic proxy, usable as a kernel module. - [mediasoup](https://mediasoup.org) - Specialized WebRTC conferencing system. - [SEMS](https://github.com/sems-server/sems) - Open source media and application server for SIP based VoIP services. - [Jitsi](https://jitsi.org/projects) - A collection of RTC open source projects, with a focus on conferencing software. ### STUN/TURN - [coturn](https://github.com/coturn/coturn) - Fully featured TURN/STUN server supporting multiple platforms. - [eturnal](https://eturnal.net/) - Modern and scalable STUN/TURN server written in [Erlang](/@harrisonqian/awesome/wiki/programming-languages/erlang). - [STUNTMAN](https://github.com/jselbie/stunserver) - RFC compliant open source STUN implementation. ## Operations ### Monitoring - [sngrep](https://github.com/irontec/sngrep) - Terminal based SIP flow viewer. - [sipgrep](https://github.com/sipcapture/sipgrep) - Console tool for sniffing, capturing and exploring SIP traffic. - [rtpbreak](https://github.com/Naishy/rtpsplit) - Detect, reconstruct and analyze RTP sessions. - [HOMER](https://github.com/sipcapture/homer) - Multi-protocol capturing and monitoring framework for RTC. - [WebRTC Troubleshooter](https://github.com/webrtc/testrtc) - Self-hosted one stop client-side WebRTC troubleshooter. - [Trickle ICE](https://webrtc.[github](/@harrisonqian/awesome/wiki/development-environment/github).io/samples/src/content/peerconnection/trickle-ice) - Exposes client-side NAT traversal debug data. - [SIP3](https://sip3.io) - VoIP & RTC traffic monitoring and analysis platform. ### Testing - [SIPp](http://sipp.sourceforge.net) - Traffic generator for the SIP protocol. - [SIPVicious](https://github.com/EnableSecurity/sipvicious) - Suite of [security](/@harrisonqian/awesome/wiki/security/security) tools that can be used to audit SIP based VoIP systems. - [sipsak](https://github.com/nils-ohlmeier/sipsak) - SIP stress and diagnostics utility. - [sipexer](https://github.com/miconda/sipexer) - Modern and flexible SIP command line tool. ### Deployment - [slimswitch](https://github.com/rtckit/slimswitch) - Tooling for creating lean secure FreeSWITCH [Docker](/@harrisonqian/awesome/wiki/back-end-development/docker) images. ### Web/API Interfaces - [Eqivo](https://eqivo.org) - Open source programmable-voice/telephony API platform. - [Kazoo](https://www.2600hz.org) - Carrier-grade VoIP API platform using FreeSWITCH and Kamailio. - [FusionPBX](https://www.fusionpbx.com) - Multitenant system built on top of FreeSWITCH. - [FreePBX](https://www.freepbx.org) - Web Manager for Asterisk. - [Fonoster](https://github.com/fonoster/fonoster) - Telecommunication stack built with [Node.js](/@harrisonqian/awesome/wiki/platforms/node-js). - [Wazo](https://wazo-platform.org) - VoIP API platform built on top of Asterisk, Kamailio and RTPEngine. - [jambonz](https://www.jambonz.org) - Open source CPaaS built for communications service providers. - [IVOZ Provider](https://github.com/irontec/ivozprovider) - Multitenant solution for VoIP telephony providers. - [Sayna](https://github.com/SaynaAI/sayna) - Real-time speech infrastructure for voice AI with WebSocket [streaming](/@harrisonqian/awesome/wiki/big-data/streaming), SIP telephony and pluggable STT/TTS providers. ### Billing - [CGRateS](http://cgrates.org) - Carrier grade open source [billing](/@harrisonqian/awesome/wiki/business/billing)/rating server. - [A2Billing](http://www.asterisk2billing.org) - [Billing](/@harrisonqian/awesome/wiki/business/billing) system for Asterisk for multiple applications. - [PyFreeBilling](https://github.com/mwolff44/pyfreebilling) - Wholesale [billing](/@harrisonqian/awesome/wiki/business/billing) platform for Kamailio and FreeSWITCH. ## Developer Resources ### Tutorials - [Official Website](https://webrtc.org) - Entry level WebRTC resources. - [Getting Started With WebRTC](https://www.html5rocks.com/en/tutorials/webrtc/basics) - WebRTC tutorial by [HTML5](/@harrisonqian/awesome/wiki/front-end-development/html5) Rocks. - [WebRTC Samples](https://webrtc.[github](/@harrisonqian/awesome/wiki/development-environment/github).io/samples) - Collection of samples demonstrating various parts of the WebRTC APIs. - [WebRTC Experiments](https://www.webrtc-experiment.com) - Comprehensive list of samples by Muaz Khan. - [Interactive Codelab](https://codelabs.developers.google.com/codelabs/webrtc-web) - 30 minutes step by step live tutorial by Google. ### JavaScript Libraries - [drachtio](https://drachtio.org) - [Node.js](/@harrisonqian/awesome/wiki/platforms/node-js) SIP server framework. - [adapter.js](https://github.com/webrtcHacks/adapter) - [JavaScript](/@harrisonqian/awesome/wiki/programming-languages/javascript) shim for abstracting WebRTC spec changes and inconsistencies. - [JsSIP](http://jssip.net) - Lightweight open source [JavaScript](/@harrisonqian/awesome/wiki/programming-languages/javascript) SIP library. - [sipML5](https://www.doubango.org/sipml5) - Open source [JavaScript](/@harrisonqian/awesome/wiki/programming-languages/javascript) SIP client with WebRTC media stack. - [simple-peer](https://github.com/feross/simple-peer) - WebRTC video, voice, and data channels abstraction for [Node.js](/@harrisonqian/awesome/wiki/platforms/node-js) and the browser. - [Netflux](https://github.com/coast-team/netflux) - Isomorphic [JavaScript](/@harrisonqian/awesome/wiki/programming-languages/javascript) peer to peer transport API for client and server. - [PeerJS](https://peerjs.com) - Data and media peer-to-peer connection API implemented over WebRTC. - [Socio](https://github.com/Rolands-Laucis/Socio) - A WebSocket Real-Time Communication (RTC) API framework. Realtime Front-end, Back-end reactivity. ### C/C++ Libraries - [libre](https://github.com/creytiv/re) - Portable SIP Stack along with companion libraries for media handling, STUN/TURN and a modular user agent. - [PJSIP](https://www.pjsip.org) - Multi-protocol RTC library written in C. - [eXosip](http://savannah.nongnu.org/projects/exosip) - eXtended osip is a mature C library for abstracting the SIP protocol. - [libdatachannel](https://github.com/paullouisageneau/libdatachannel) - Standalone WebRTC DataChannels C++ implementation. - [libSRTP](https://github.com/cisco/libsrtp) - Secure Real-time Transport Protocol (SRTP) library for C. - [usrsctp](https://github.com/sctplab/usrsctp) - Portable Stream Control Transmission Protocol (SCTP) user-land stack. - [rawrtc](https://github.com/rawrtc/rawrtc) - WebRTC and ORTC library with a small footprint. - [OSS Core](https://github.com/joegen/oss_core) - General purpose C++ library for Real Time Communications. - [Open WebRTC Toolkit](https://01.org/open-webrtc-toolkit) - WebRTC development toolkit with bindings for multiple platforms. - [Sofia-SIP](https://github.com/freeswitch/sofia-sip) - Open source SIP library used by FreeSWITCH. ### Go Libraries - [Pion](https://pion.ly) - Extensive software stack for WebRTC written in Go. - [gossip](https://github.com/StefanKopieczek/gossip) - SIP stack for stateful user agents written in Go. - [siprocket](https://github.com/marv2097/siprocket) - Fast SIP and SDP packet parser. - [go-diameter](https://github.com/fiorix/go-diameter) - RFC compliant Diameter protocol library. ### PHP Libraries - [RTCKit/SIP](https://github.com/rtckit/php-sip) - RFC 3261 compliant SIP parsing and rendering library for PHP 7.4+. ### Python Libraries - [aiortc](https://github.com/aiortc/aiortc) - WebRTC and ORTC implementation for [Python](/@harrisonqian/awesome/wiki/programming-languages/python) using [asyncio](/@harrisonqian/awesome/wiki/programming-languages/asyncio). - [Katari](https://github.com/hyperioxx/Katari) - SIP stack application framework. - [peerjs-python](https://github.com/ambianic/peerjs-python) - [Python](/@harrisonqian/awesome/wiki/programming-languages/python) port of the PeerJS peer-to-peer connection library. ### Erlang Libraries - [NkSIP](https://github.com/NetComposer/nksip) - Extendable SIP server framework. - [ersip](https://github.com/poroh/ersip) - Library comprising building blocks for SIP applications. ### Rust Libraries - [libsip](https://docs.rs/libsip/0.2.4/libsip) - SIP implementation, with a focus towards softphone clients. - [sipcore](https://github.com/armatusmiles/sipcore) - [Rust](/@harrisonqian/awesome/wiki/programming-languages/rust) framework for creating SIP applications. - [rtcrs/webrtc](https://github.com/rtcrs/webrtc) - WebRTC stack, supporting SDP, RTP, RTCP and SRTP. ### Dart Libraries - [dart-sip-ua](https://github.com/cloudwebrtc/dart-sip-ua) - [Dart](/@harrisonqian/awesome/wiki/programming-languages/dart)-lang port of JsSIP, capable of SIP over WebSocket. ## Blogs - [BlogGeekMe](https://bloggeek.me/blog) - Blog by Tsahi Levent-Levi with a strong focus on WebRTC. - [SIP Adventures](https://andrewjprokop.wordpress.com) - Unified communications blog by Andrew Prokop. - [WebRTCHacks](https://webrtchacks.com) - WebRTC blog by independent technologists. ## Discussion - [FreeSWITCH Slack](https://signalwire.community) - Join #freeswitch and #freeswitch-dev for user and developer support. - [discuss-webrtc](https://groups.google.com/forum/?fromgroups#!forum/discuss-webrtc) - Developer oriented Google Group for WebRTC discussions. ## Events - [ClueCon](http://cluecon.com) - Annual conference held in Chicago for telecommunications developers. Birthplace of FreeSWITCH. - [Kamailio World](https://www.kamailioworld.com) - Berlin hosted annual event focused on Kamailio as well as VoIP, WebRTC, IMS, VoLTE and more. - [AstriCon](https://www.asterisk.org/community/astricon-user-conference) - Asterisk focus event held every year across the US. - [CommCon](https://commcon.xyz) - Annual conference held in the UK focused on telecommunications in general and WebRTC in particular. - [OpenSIPS Summit](https://www.opensips.org/events) - Meeting place for the OpenSIPS community. - [Kranky Geek](https://krankygeek.com) - AI and RTC event in San Francisco. - [FOSDEM](https://fosdem.org) - Free event for software developers, with a RTC component, held every year in Europe. - [JanusCon](https://www.januscon.it) - JanusCon is a live event for Janus and RTC implementers. - [TADHack](https://tadhack.com) - Global [hackathon](/@harrisonqian/awesome/wiki/learn/hackathon) focused on programmable communications. ## Related Lists - [Awesome RIPT](https://github.com/rtckit/awesome-ript) - Real Time Internet Peering for Telephony. - [Awesome RTC Hacking](https://github.com/EnableSecurity/awesome-rtc-hacking) - Real Time Communications [hacking](/@harrisonqian/awesome/wiki/security/hacking) and penetration [testing](/@harrisonqian/awesome/wiki/testing/testing) resources. - [Awesome 5G](https://github.com/calee0219/awesome-5g) - 5G [frameworks](/@harrisonqian/awesome/wiki/front-end-development/frameworks), libraries, software and resources. - [Awesome Cellular Hacking](https://github.com/W00t3k/Awesome-Cellular-Hacking) - Research resources in the 3G/4G/5G Cellular [security](/@harrisonqian/awesome/wiki/security/security) space. - [Awesome Telco](https://github.com/ravens/awesome-telco) - Telco resources and projects. - [SIP Resources](https://github.com/miconda/sip-resources) - Useful SIP resources curated by Kamailio's head developer. ## Contribute Contributions welcome! Read the [contribution guidelines](CONTRIBUTING.md) first.