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https://github.com/rtckit/awesome-rtc

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

List: awesome-rtc

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:satellite: A curated list of awesome Real Time Communications resources

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README 

        
# Awesome Real Time Communications [![Awesome](https://awesome.re/badge.svg)](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-libraries)

  - [C/C++ Libraries](#cc-libraries)

  - [Go Libraries](#go-libraries)

  - [PHP Libraries](#php-libraries)

  - [Python Libraries](#python-libraries)

  - [Erlang Libraries](#erlang-libraries)

  - [Rust Libraries](#rust-libraries)

  - [Dart Libraries](#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 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.

- [Sippy B2BUA](https://github.com/sippy/b2bua) - Back-to-back user agent server written in 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.

- [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.

- [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.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 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 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.

- [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.



### Billing



- [CGRateS](http://cgrates.org) - Carrier grade open source billing/rating server.

- [A2Billing](http://www.asterisk2billing.org) - Billing system for Asterisk for multiple applications.

- [PyFreeBilling](https://github.com/mwolff44/pyfreebilling) - Wholesale 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 Rocks.

- [WebRTC Samples](https://webrtc.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 SIP server framework.

- [adapter.js](https://github.com/webrtcHacks/adapter) - JavaScript shim for abstracting WebRTC spec changes and inconsistencies.

- [JsSIP](http://jssip.net) - Lightweight open source JavaScript SIP library.

- [sipML5](https://www.doubango.org/sipml5) - Open source 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 and the browser.

- [Netflux](https://github.com/coast-team/netflux) - Isomorphic 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.



### 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 using asyncio.

- [Katari](https://github.com/hyperioxx/Katari) - SIP stack application framework.

- [peerjs-python](https://github.com/ambianic/peerjs-python) - 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 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-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 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 and penetration testing resources.

- [Awesome 5G](https://github.com/calee0219/awesome-5g) - 5G frameworks, libraries, software and resources.

- [Awesome Cellular Hacking](https://github.com/W00t3k/Awesome-Cellular-Hacking) - Research resources in the 3G/4G/5G Cellular 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.