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{"slug": "rtckit--awesome-rtc", "title": "Awesome Rtc", "description": ":satellite: A curated list of awesome Real Time Communications resources", "github_url": "https://github.com/rtckit/awesome-rtc", "stars": "466", "tag": "Networking", "entry_count": 106, "subcategory_count": 19, "subcategories": [{"name": "General", "parent": "", "entries": [{"name": "Server Software", "url": "#server-software", "description": ""}, {"name": "Operations", "url": "#operations", "description": ""}, {"name": "Developer Resources", "url": "#developer-resources", "description": ""}, {"name": "Blogs", "url": "#blogs", "description": ""}, {"name": "Discussion", "url": "#discussion", "description": ""}, {"name": "Events", "url": "#events", "description": ""}, {"name": "Related Lists", "url": "#related-lists", "description": ""}, {"name": "Contribute", "url": "#contribute", "description": ""}]}, {"name": "General Purpose", "parent": "Server Software", "entries": [{"name": "FreeSWITCH", "url": "http://freeswitch.org", "description": "Open source multi-protocol, cross-platform and software switch."}, {"name": "Asterisk", "url": "http://asterisk.org", "description": "PBX framework supporting multiple protocols and platforms."}]}, {"name": "SIP Servers", "parent": "Server Software", "entries": [{"name": "Kamailio", "url": "http://www.kamailio.org", "description": "Open source SIP server widely deployed by carriers and providers. Formerly known as OpenSER."}, {"name": "OpenSIPS", "url": "http://www.opensips.org", "description": "Open source SIP server, tracing its roots in OpenSER (presently Kamailio)."}, {"name": "Routr", "url": "https://routr.io", "description": "Lightweight SIP proxy, location server, and registrar written in Node.js."}, {"name": "Sippy B2BUA", "url": "https://github.com/sippy/b2bua", "description": "Back-to-back user agent server written in Python.", "stars": "195"}, {"name": "Flexisip", "url": "https://github.com/BelledonneCommunications/flexisip", "description": "SIP server suite comprising proxy, presence and group chat functions.", "stars": "176"}]}, {"name": "Media Servers", "parent": "Server Software", "entries": [{"name": "Janus", "url": "https://janus.conf.meetecho.com", "description": "Lightweight open source, general purpose, WebRTC gateway."}, {"name": "LiveKit", "url": "https://livekit.io", "description": "Open-source WebRTC infrastructure for building scalable real-time audio and video applications."}, {"name": "RTPProxy", "url": "https://www.rtpproxy.org", "description": "General purpose high performance RTP proxy."}, {"name": "RTP:Engine", "url": "https://github.com/sipwise/rtpengine", "description": "RTP and UDP based media traffic proxy, usable as a kernel module.", "stars": "912"}, {"name": "mediasoup", "url": "https://mediasoup.org", "description": "Specialized WebRTC conferencing system."}, {"name": "SEMS", "url": "https://github.com/sems-server/sems", "description": "Open source media and application server for SIP based VoIP services.", "stars": "181"}, {"name": "Jitsi", "url": "https://jitsi.org/projects", "description": "A collection of RTC open source projects, with a focus on conferencing software."}]}, {"name": "STUN/TURN", "parent": "Server Software", "entries": [{"name": "coturn", "url": "https://github.com/coturn/coturn", "description": "Fully featured TURN/STUN server supporting multiple platforms.", "stars": "14k"}, {"name": "eturnal", "url": "https://eturnal.net/", "description": "Modern and scalable STUN/TURN server written in Erlang."}, {"name": "STUNTMAN", "url": "https://github.com/jselbie/stunserver", "description": "RFC compliant open source STUN implementation.", "stars": "1.6k"}]}, {"name": "Monitoring", "parent": "Server Software", "entries": [{"name": "sngrep", "url": "https://github.com/irontec/sngrep", "description": "Terminal based SIP flow viewer.", "stars": "1.2k"}, {"name": "sipgrep", "url": "https://github.com/sipcapture/sipgrep", "description": "Console tool for sniffing, capturing and exploring SIP traffic.", "stars": "172"}, {"name": "rtpbreak", "url": "https://github.com/Naishy/rtpsplit", "description": "Detect, reconstruct and analyze RTP sessions.", "stars": "22"}, {"name": "HOMER", "url": "https://github.com/sipcapture/homer", "description": "Multi-protocol capturing and monitoring framework for RTC.", "stars": "1.9k"}, {"name": "WebRTC Troubleshooter", "url": "https://github.com/webrtc/testrtc", "description": "Self-hosted one stop client-side WebRTC troubleshooter.", "stars": "492"}, {"name": "Trickle ICE", "url": "https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice", "description": "Exposes client-side NAT traversal debug data."}, {"name": "SIP3", "url": "https://sip3.io", "description": "VoIP & RTC traffic monitoring and analysis platform."}]}, {"name": "Testing", "parent": "Server Software", "entries": [{"name": "SIPp", "url": "http://sipp.sourceforge.net", "description": "Traffic generator for the SIP protocol."}, {"name": "SIPVicious", "url": "https://github.com/EnableSecurity/sipvicious", "description": "Suite of security tools that can be used to audit SIP based VoIP systems.", "stars": "1k"}, {"name": "sipsak", "url": "https://github.com/nils-ohlmeier/sipsak", "description": "SIP stress and diagnostics utility.", "stars": "170"}, {"name": "sipexer", "url": "https://github.com/miconda/sipexer", "description": "Modern and flexible SIP command line tool.", "stars": "378"}]}, {"name": "Deployment", "parent": "Server Software", "entries": [{"name": "slimswitch", "url": "https://github.com/rtckit/slimswitch", "description": "Tooling for creating lean secure FreeSWITCH Docker images.", "stars": "19"}]}, {"name": "Web/API Interfaces", "parent": "Server Software", "entries": [{"name": "Eqivo", "url": "https://eqivo.org", "description": "Open source programmable-voice/telephony API platform."}, {"name": "Kazoo", "url": "https://www.2600hz.org", "description": "Carrier-grade VoIP API platform using FreeSWITCH and Kamailio."}, {"name": "FusionPBX", "url": "https://www.fusionpbx.com", "description": "Multitenant system built on top of FreeSWITCH."}, {"name": "FreePBX", "url": "https://www.freepbx.org", "description": "Web Manager for Asterisk."}, {"name": "Fonoster", "url": "https://github.com/fonoster/fonoster", "description": "Telecommunication stack built with Node.js.", "stars": "7.7k"}, {"name": "Wazo", "url": "https://wazo-platform.org", "description": "VoIP API platform built on top of Asterisk, Kamailio and RTPEngine."}, {"name": "jambonz", "url": "https://www.jambonz.org", "description": "Open source CPaaS built for communications service providers."}, {"name": "IVOZ Provider", "url": "https://github.com/irontec/ivozprovider", "description": "Multitenant solution for VoIP telephony providers.", "stars": "221"}, {"name": "Sayna", "url": "https://github.com/SaynaAI/sayna", "description": "Real-time speech infrastructure for voice AI with WebSocket streaming, SIP telephony and pluggable STT/TTS providers.", "stars": "131"}]}, {"name": "Billing", "parent": "Server Software", "entries": [{"name": "CGRateS", "url": "http://cgrates.org", "description": "Carrier grade open source billing/rating server."}, {"name": "A2Billing", "url": "http://www.asterisk2billing.org", "description": "Billing system for Asterisk for multiple applications."}, {"name": "PyFreeBilling", "url": "https://github.com/mwolff44/pyfreebilling", "description": "Wholesale billing platform for Kamailio and FreeSWITCH.", "stars": "109"}]}, {"name": "Tutorials", "parent": "Developer Resources", "entries": [{"name": "Official Website", "url": "https://webrtc.org", "description": "Entry level WebRTC resources."}, {"name": "Getting Started With WebRTC", "url": "https://www.html5rocks.com/en/tutorials/webrtc/basics", "description": "WebRTC tutorial by HTML5 Rocks."}, {"name": "WebRTC Samples", "url": "https://webrtc.github.io/samples", "description": "Collection of samples demonstrating various parts of the WebRTC APIs."}, {"name": "WebRTC Experiments", "url": "https://www.webrtc-experiment.com", "description": "Comprehensive list of samples by Muaz Khan."}, {"name": "Interactive Codelab", "url": "https://codelabs.developers.google.com/codelabs/webrtc-web", "description": "30 minutes step by step live tutorial by Google."}]}, {"name": "JavaScript Libraries", "parent": "Developer Resources", "entries": [{"name": "drachtio", "url": "https://drachtio.org", "description": "Node.js SIP server framework."}, {"name": "adapter.js", "url": "https://github.com/webrtcHacks/adapter", "description": "JavaScript shim for abstracting WebRTC spec changes and inconsistencies.", "stars": "3.7k"}, {"name": "JsSIP", "url": "http://jssip.net", "description": "Lightweight open source JavaScript SIP library."}, {"name": "sipML5", "url": "https://www.doubango.org/sipml5", "description": "Open source JavaScript SIP client with WebRTC media stack."}, {"name": "simple-peer", "url": "https://github.com/feross/simple-peer", "description": "WebRTC video, voice, and data channels abstraction for Node.js and the browser.", "stars": "7.8k"}, {"name": "Netflux", "url": "https://github.com/coast-team/netflux", "description": "Isomorphic JavaScript peer to peer transport API for client and server.", "stars": "216"}, {"name": "PeerJS", "url": "https://peerjs.com", "description": "Data and media peer-to-peer connection API implemented over WebRTC."}, {"name": "Socio", "url": "https://github.com/Rolands-Laucis/Socio", "description": "A WebSocket Real-Time Communication (RTC) API framework. Realtime Front-end, Back-end reactivity.", "stars": "126"}]}, {"name": "C/C++ Libraries", "parent": "Developer Resources", "entries": [{"name": "libre", "url": "https://github.com/creytiv/re", "description": "Portable SIP Stack along with companion libraries for media handling, STUN/TURN and a modular user agent.", "stars": "544"}, {"name": "PJSIP", "url": "https://www.pjsip.org", "description": "Multi-protocol RTC library written in C."}, {"name": "eXosip", "url": "http://savannah.nongnu.org/projects/exosip", "description": "eXtended osip is a mature C library for abstracting the SIP protocol."}, {"name": "libdatachannel", "url": "https://github.com/paullouisageneau/libdatachannel", "description": "Standalone WebRTC DataChannels C++ implementation.", "stars": "2.5k"}, {"name": "libSRTP", "url": "https://github.com/cisco/libsrtp", "description": "Secure Real-time Transport Protocol (SRTP) library for C.", "stars": "1.4k"}, {"name": "usrsctp", "url": "https://github.com/sctplab/usrsctp", "description": "Portable Stream Control Transmission Protocol (SCTP) user-land stack.", "stars": "744"}, {"name": "rawrtc", "url": "https://github.com/rawrtc/rawrtc", "description": "WebRTC and ORTC library with a small footprint.", "stars": "390"}, {"name": "OSS Core", "url": "https://github.com/joegen/oss_core", "description": "General purpose C++ library for Real Time Communications.", "stars": "26"}, {"name": "Open WebRTC Toolkit", "url": "https://01.org/open-webrtc-toolkit", "description": "WebRTC development toolkit with bindings for multiple platforms."}, {"name": "Sofia-SIP", "url": "https://github.com/freeswitch/sofia-sip", "description": "Open source SIP library used by FreeSWITCH.", "stars": "322"}]}, {"name": "Go Libraries", "parent": "Developer Resources", "entries": [{"name": "Pion", "url": "https://pion.ly", "description": "Extensive software stack for WebRTC written in Go."}, {"name": "gossip", "url": "https://github.com/StefanKopieczek/gossip", "description": "SIP stack for stateful user agents written in Go.", "stars": "346"}, {"name": "siprocket", "url": "https://github.com/marv2097/siprocket", "description": "Fast SIP and SDP packet parser.", "stars": "74"}, {"name": "go-diameter", "url": "https://github.com/fiorix/go-diameter", "description": "RFC compliant Diameter protocol library.", "stars": "282"}]}, {"name": "PHP Libraries", "parent": "Developer Resources", "entries": [{"name": "RTCKit/SIP", "url": "https://github.com/rtckit/php-sip", "description": "RFC 3261 compliant SIP parsing and rendering library for PHP 7.4+.", "stars": "43"}]}, {"name": "Python Libraries", "parent": "Developer Resources", "entries": [{"name": "aiortc", "url": "https://github.com/aiortc/aiortc", "description": "WebRTC and ORTC implementation for Python using asyncio.", "stars": "5k"}, {"name": "Katari", "url": "https://github.com/hyperioxx/Katari", "description": "SIP stack application framework.", "stars": "4"}, {"name": "peerjs-python", "url": "https://github.com/ambianic/peerjs-python", "description": "Python port of the PeerJS peer-to-peer connection library.", "stars": "97"}]}, {"name": "Erlang Libraries", "parent": "Developer Resources", "entries": [{"name": "NkSIP", "url": "https://github.com/NetComposer/nksip", "description": "Extendable SIP server framework.", "stars": "362"}, {"name": "ersip", "url": "https://github.com/poroh/ersip", "description": "Library comprising building blocks for SIP applications.", "stars": "128"}]}, {"name": "Rust Libraries", "parent": "Developer Resources", "entries": [{"name": "libsip", "url": "https://docs.rs/libsip/0.2.4/libsip", "description": "SIP implementation, with a focus towards softphone clients."}, {"name": "sipcore", "url": "https://github.com/armatusmiles/sipcore", "description": "Rust framework for creating SIP applications.", "stars": "31"}, {"name": "rtcrs/webrtc", "url": "https://github.com/rtcrs/webrtc", "description": "WebRTC stack, supporting SDP, RTP, RTCP and SRTP.", "stars": "5k"}]}, {"name": "Dart Libraries", "parent": "Developer Resources", "entries": [{"name": "dart-sip-ua", "url": "https://github.com/cloudwebrtc/dart-sip-ua", "description": "Dart-lang port of JsSIP, capable of SIP over WebSocket.", "stars": "370"}, {"name": "BlogGeekMe", "url": "https://bloggeek.me/blog", "description": "Blog by Tsahi Levent-Levi with a strong focus on WebRTC."}, {"name": "SIP Adventures", "url": "https://andrewjprokop.wordpress.com", "description": "Unified communications blog by Andrew Prokop."}, {"name": "WebRTCHacks", "url": "https://webrtchacks.com", "description": "WebRTC blog by independent technologists."}, {"name": "FreeSWITCH Slack", "url": "https://signalwire.community", "description": "Join #freeswitch and #freeswitch-dev for user and developer support."}, {"name": "discuss-webrtc", "url": "https://groups.google.com/forum/?fromgroups#!forum/discuss-webrtc", "description": "Developer oriented Google Group for WebRTC discussions."}, {"name": "ClueCon", "url": "http://cluecon.com", "description": "Annual conference held in Chicago for telecommunications developers. Birthplace of FreeSWITCH."}, {"name": "Kamailio World", "url": "https://www.kamailioworld.com", "description": "Berlin hosted annual event focused on Kamailio as well as VoIP, WebRTC, IMS, VoLTE and more."}, {"name": "AstriCon", "url": "https://www.asterisk.org/community/astricon-user-conference", "description": "Asterisk focus event held every year across the US."}, {"name": "CommCon", "url": "https://commcon.xyz", "description": "Annual conference held in the UK focused on telecommunications in general and WebRTC in particular."}, {"name": "OpenSIPS Summit", "url": "https://www.opensips.org/events", "description": "Meeting place for the OpenSIPS community."}, {"name": "Kranky Geek", "url": "https://krankygeek.com", "description": "AI and RTC event in San Francisco."}, {"name": "FOSDEM", "url": "https://fosdem.org", "description": "Free event for software developers, with a RTC component, held every year in Europe."}, {"name": "JanusCon", "url": "https://www.januscon.it", "description": "JanusCon is a live event for Janus and RTC implementers."}, {"name": "TADHack", "url": "https://tadhack.com", "description": "Global hackathon focused on programmable communications."}, {"name": "Awesome RIPT", "url": "https://github.com/rtckit/awesome-ript", "description": "Real Time Internet Peering for Telephony.", "stars": "29"}, {"name": "Awesome RTC Hacking", "url": "https://github.com/EnableSecurity/awesome-rtc-hacking", "description": "Real Time Communications hacking and penetration testing resources.", "stars": "516"}, {"name": "Awesome 5G", "url": "https://github.com/calee0219/awesome-5g", "description": "5G frameworks, libraries, software and resources.", "stars": "872"}, {"name": "Awesome Cellular Hacking", "url": "https://github.com/W00t3k/Awesome-Cellular-Hacking", "description": "Research resources in the 3G/4G/5G Cellular security space.", "stars": "3.6k"}, {"name": "Awesome Telco", "url": "https://github.com/ravens/awesome-telco", "description": "Telco resources and projects.", "stars": "904"}, {"name": "SIP Resources", "url": "https://github.com/miconda/sip-resources", "description": "Useful SIP resources curated by Kamailio's head developer.", "stars": "253"}]}]} |