Webrtc
Showing 35 of 71 projects
A single-file TLS 1.0-1.3 and DTLS implementation in C, using libtomcrypt for cryptography.
A React Native wrapper for the Agora RTC SDK, enabling real-time video and audio communication in mobile apps.
A high-performance, AI-native software-defined PBX built in Rust for programmable contact centers.
A curated list of JavaScript libraries and tools for network programming, peer-to-peer communication, and protocol implementation.
A portable C library providing SIP, RTP, STUN/TURN/ICE, HTTP, and WebSocket stacks for building real-time communication applications.
A curated list of security resources for penetration testing and vulnerability assessment of VoIP, WebRTC, and VoLTE systems.
A frontend JavaScript framework for developing DNS rebinding exploits against vulnerable LAN devices and IoT products.
A web-based diagnostic tool that automatically tests WebRTC components like microphone, camera, network, and connectivity.
A high-performance, zero-allocation Go implementation of the RFC 5389 STUN protocol for NAT traversal.
A lightweight desktop app that turns your smartphone into a remote controller for PowerPoint, Google Slides, Canva, and Keynote presentations.
A curated list of awesome resources for Real-Time Communications (RTC), including protocols, servers, libraries, and tools.
A peer-to-peer pipe across the web using WebRTC and GitHub keys for authentication, enabling direct communication between users.
An Electron application for recording screen and audio using WebRTC and desktopCapture API.
A modern CLI tool for crafting and sending SIP requests to test VoIP and WebRTC signaling servers.
A lightweight C library implementing WebRTC and ORTC APIs for real-time communication.
A Dart SIP user agent stack for building cross-platform VoIP and messaging apps with Flutter and WebRTC.
An open-source voice dictation tool that types your speech at the cursor in any application, powered by customizable AI transcription and formatting.
A curated collection of open-source and commercial SIP (Session Initiation Protocol) resources, tools, libraries, and software.
A library for building fast, web-based augmented reality applications using the PlayCanvas Editor and Engine.
A Jupyter widget that brings WebRTC and MediaStream API to notebooks and JupyterLab for real-time media streaming and recording.
An open-source, self-hostable communication tool with messaging, video calls, and screen sharing for company use.
A unified JavaScript library for discovering BitTorrent and WebTorrent peers via trackers, DHT, and LSD.
A lightweight, extensible socket manager for Node.js with pluggable transports and protocol support.
Isomorphic JavaScript peer-to-peer transport API for building secure, fault-tolerant full mesh networks using WebRTC and WebSocket.
Embed realtime voice and video communication into React Native apps using WebRTC and the Voximplant cloud platform.
A browser-based tool for creating and hosting full-featured HTML websites peer-to-peer via WebRTC, WebTorrent, and IPFS.
A high-performance, RFC-compliant SIP stack library written in Rust for building SIP applications like proxies, registrars, and user agents.
A simple, EventEmitter-based WebSocket client and server library for Node.js and browsers.
A high-performance C# ORM for SQL and NoSQL databases with built-in messaging, pub-sub, and client-server architecture.
A high-performance real-time voice processing server in Rust providing unified STT/TTS services via WebSocket and REST APIs.
Share files directly from your browser using IPFS — no cloud needed, with peer-to-peer transfers.
A peer-to-peer chat application built with Rust, WebAssembly, and WebRTC using the Yew framework.
Python client for PeerJS enabling WebRTC peer-to-peer connections between Python apps and web browsers.
A Blazor wrapper for the Media Capture and Streams browser API, enabling access to microphones, cameras, and media streams in .NET web apps.
A low-latency, web-based open-source intercom solution for broadcast and media production using WebRTC.
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