awesome-rtc
RTC toolkit
A curated list of resources and tools for building real-time communication systems
A curated list of awesome Real Time Communications resources
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last commit: over 2 years ago
Linked from 3 awesome lists
awesomeawesome-listreal-time-communicationsrtcsiptelecommunicationstelephonyvoipwebrtc
Awesome Real Time Communications / Server Software / General Purpose | |||
| FreeSWITCH | Open source multi-protocol, cross-platform and software switch | ||
| Asterisk | PBX framework supporting multiple protocols and platforms | ||
Awesome Real Time Communications / Server Software / SIP Servers | |||
| 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 | 172 | about 2 months ago | Back-to-back user agent server written in Python |
| Flexisip | 150 | 6 days ago | SIP server suite comprising proxy, presence and group chat functions |
Awesome Real Time Communications / Server Software / Media Servers | |||
| Janus | Lightweight open source, general purpose, WebRTC gateway | ||
| RTPProxy | General purpose high performance RTP proxy | ||
| RTP:Engine | 796 | 7 days ago | RTP and UDP based media traffic proxy, usable as a kernel module |
| mediasoup | Specialized WebRTC conferencing system | ||
| SEMS | 161 | 6 months ago | 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 | ||
Awesome Real Time Communications / Server Software / STUN/TURN | |||
| coturn | 11,418 | 7 days ago | Fully featured TURN/STUN server supporting multiple platforms |
| STUNTMAN | 1,442 | 5 months ago | RFC compliant open source STUN implementation |
Awesome Real Time Communications / Operations / Monitoring | |||
| sngrep | 1,020 | about 1 month ago | Terminal based SIP flow viewer |
| sipgrep | 165 | 6 months ago | Console tool for sniffing, capturing and exploring SIP traffic |
| rtpbreak | 18 | almost 8 years ago | Detect, reconstruct and analyze RTP sessions |
| HOMER | 1,650 | 10 days ago | Multi-protocol capturing and monitoring framework for RTC |
| WebRTC Troubleshooter | 478 | 8 months ago | 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 | ||
Awesome Real Time Communications / Operations / Testing | |||
| SIPp | Traffic generator for the SIP protocol | ||
| SIPVicious | 893 | about 2 years ago | Suite of security tools that can be used to audit SIP based VoIP systems |
| sipsak | 140 | 11 months ago | SIP stress and diagnostics utility |
| sipexer | 293 | 3 months ago | Modern and flexible SIP command line tool |
Awesome Real Time Communications / Operations / Deployment | |||
| slimswitch | 18 | 12 months ago | Tooling for creating lean secure FreeSWITCH Docker images |
Awesome Real Time Communications / Operations / Web/API Interfaces | |||
| 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 | 6,361 | 6 days ago | 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 | 197 | 6 days ago | Multitenant solution for VoIP telephony providers |
Awesome Real Time Communications / Operations / Billing | |||
| CGRateS | Carrier grade open source billing/rating server | ||
| A2Billing | Billing system for Asterisk for multiple applications | ||
| PyFreeBilling | 99 | 6 days ago | Wholesale billing platform for Kamailio and FreeSWITCH |
Awesome Real Time Communications / Developer Resources / Tutorials | |||
| 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 | ||
Awesome Real Time Communications / Developer Resources / JavaScript Libraries | |||
| drachtio | Node.js SIP server framework | ||
| adapter.js | 3,635 | 7 months ago | 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 | 7,440 | 5 months ago | WebRTC video, voice, and data channels abstraction for Node.js and the browser |
| Netflux | 213 | almost 3 years ago | Isomorphic JavaScript peer to peer transport API for client and server |
| PeerJS | Data and media peer-to-peer connection API implemented over WebRTC | ||
Awesome Real Time Communications / Developer Resources / C/C++ Libraries | |||
| libre | 522 | 10 months ago | 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 | 1,802 | 4 days ago | Standalone WebRTC DataChannels C++ implementation |
| libSRTP | 1,228 | 7 days ago | Secure Real-time Transport Protocol (SRTP) library for C |
| usrsctp | 673 | 18 days ago | Portable Stream Control Transmission Protocol (SCTP) user-land stack |
| rawrtc | 375 | almost 3 years ago | WebRTC and ORTC library with a small footprint |
| OSS Core | 26 | about 3 years ago | General purpose C++ library for Real Time Communications |
| Open WebRTC Toolkit | WebRTC development toolkit with bindings for multiple platforms | ||
| Sofia-SIP | 274 | about 2 months ago | Open source SIP library used by FreeSWITCH |
Awesome Real Time Communications / Developer Resources / Go Libraries | |||
| Pion | Extensive software stack for WebRTC written in Go | ||
| gossip | 339 | almost 5 years ago | SIP stack for stateful user agents written in Go |
| siprocket | 71 | about 2 years ago | Fast SIP and SDP packet parser |
| go-diameter | 254 | 9 months ago | RFC compliant Diameter protocol library |
Awesome Real Time Communications / Developer Resources / PHP Libraries | |||
| RTCKit/SIP | 36 | 10 days ago | RFC 3261 compliant SIP parsing and rendering library for PHP 7.4+ |
Awesome Real Time Communications / Developer Resources / Python Libraries | |||
| aiortc | 4,265 | 5 months ago | WebRTC and ORTC implementation for Python using asyncio |
| Katari | 3 | almost 2 years ago | SIP stack application framework |
| peerjs-python | 89 | 10 months ago | Python port of the PeerJS peer-to-peer connection library |
Awesome Real Time Communications / Developer Resources / Erlang Libraries | |||
| NkSIP | 357 | 6 months ago | Extendable SIP server framework |
| ersip | 124 | about 2 months ago | Library comprising building blocks for SIP applications |
Awesome Real Time Communications / Developer Resources / Rust Libraries | |||
| libsip | SIP implementation, with a focus towards softphone clients | ||
| sipcore | 29 | over 3 years ago | Rust framework for creating SIP applications |
| rtcrs/webrtc | 4,165 | 4 days ago | WebRTC stack, supporting SDP, RTP, RTCP and SRTP |
Awesome Real Time Communications / Developer Resources / Dart Libraries | |||
| dart-sip-ua | 337 | 30 days ago | Dart-lang port of JsSIP, capable of SIP over WebSocket |
Awesome Real Time Communications / Blogs | |||
| 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 | ||
Awesome Real Time Communications / Discussion | |||
| FreeSWITCH Slack | Join #freeswitch and #freeswitch-dev for user and developer support | ||
| discuss-webrtc | Developer oriented Google Group for WebRTC discussions | ||
Awesome Real Time Communications / Events | |||
| 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 Real Time Communications / Related Lists | |||
| Awesome RIPT | 27 | about 4 years ago | Real Time Internet Peering for Telephony |
| Awesome RTC Hacking | 413 | almost 2 years ago | Real Time Communications hacking and penetration testing resources |
| Awesome 5G | 732 | 4 months ago | 5G frameworks, libraries, software and resources |
| Awesome Cellular Hacking | 2,901 | 13 days ago | Research resources in the 3G/4G/5G Cellular security space |
| Awesome Telco | 696 | 28 days ago | Telco resources and projects |
| SIP Resources | 210 | 3 months ago | Useful SIP resources curated by Kamailio's head developer |
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