Search Results for "webrtc"
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Showing 110 open source projects for "webrtc"
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Pion WebRTC
Pure Go implementation of the WebRTC API
...Pion has multiple large production users already. Come learn from them and be the next! Example applications contains code samples of common things people build with Pion WebRTC. Example-webrtc-applications contains more full featured examples that use 3rd party libraries. -
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Flutter-WebRTC
WebRTC plugin for Flutter Mobile/Desktop/Web
WebRTC plugin for Flutter Mobile/Desktop/Web. Additional platform/OS support from the other community. -
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SRS
SRS is a simple, high efficiency and realtime video server
SRS is a simple, high-efficiency and real-time video server that supports RTMP, WebRTC, HLS, HTTP-FLV and SRT. SRS is licenced under MIT or MulanPSL-2.0, and note that MulanPSL-2.0 is compatible with Apache-2.0, but some third-party libraries are distributed using their own licenses. Besides of FFmpeg or OBS, it's also able to publish by H5 if WebRTC is enabled, please remember to set the CANDIDATE for WebRTC. Highly recommend that directly run SRS by docker, Cloud Virtual Machine, or K8s, however it's also easy to build SRS from source code, for detail please see Getting Started. ... -
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Ant Media Server
Adaptive, ultra low latency streaming
Real-time streaming engine delivers content with sub-0.5secs latency. Ant Media Server supports WebRTC, CMAF, HLS, RTMP, RTSP, SRT, Zixi and more for your needs. Ant Media Server is a real-time streaming engine software that provides adaptive, ultra low latency streaming by using WebRTC technology with ~0.5 seconds latency. Ant Media Server is auto-scalable and can run on-premise or on-cloud. Increase interaction and experience real-time video streaming. -
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5
Trystero
Serverless WebRTC matchmaking for painless P2P
Serverless WebRTC matchmaking for painless P2P: make any site multiplayer in a few lines. Trystero manages a clandestine courier network that lets your application's users talk directly with one another, encrypted and without a server middleman. To establish a direct peer-to-peer connection with WebRTC, a signalling channel is needed to exchange peer information (SDP). -
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audio_video_streaming
Compilation of authoritative information on audio and video streaming
...It serves as a learning hub for developers interested in multimedia systems, covering topics such as encoding, decoding, transmission protocols, and real-time communication frameworks. The repository includes example implementations like multi-user video chat systems, WebRTC demos, and cross-platform media players to provide hands-on learning opportunities. It also documents widely used technologies such as RTP, RTMP, HLS, and WebRTC, helping users understand the full lifecycle of streaming pipelines from capture to rendering. In addition to educational materials, it references industry tools, libraries, and frameworks, making it a valuable roadmap for both beginners and advanced engineers. ... -
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WebTorrent
Streaming torrent client for the web
...It's written completely in JavaScript – the language of the web – so the same code works in both runtimes. In node.js, this module is a simple torrent client, using TCP and UDP to talk to other torrent clients. In the browser, WebTorrent uses WebRTC (data channels) for peer-to-peer transport. It can be used without browser plugins, extensions, or installations. It's Just JavaScript™. Note: WebTorrent does not support UDP/TCP peers in browser. To make BitTorrent work over WebRTC (which is the only P2P transport that works on the web) we made some protocol changes. Therefore, a browser-based WebTorrent client or "web peer" can only connect to other clients that support WebTorrent/WebRTC. -
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MediaDevices
Go implementation of the MediaDevices API
mediadevices is a Go library developed by the Pion WebRTC team that enables real-time access to audio and video devices for building native Go applications involving media streaming and conferencing. It provides a cross-platform, unified API for capturing and manipulating media streams and is often used in combination with Pion WebRTC for peer-to-peer communications. Its support for device enumeration, media constraints, and frame processing makes it a powerful building block for custom voice and video solutions in Go. -
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Weron
Overlay networks based on WebRTC.
Weron is a peer-to-peer overlay network built using WebRTC and WebSockets, enabling remote access and tunneling between machines behind NATs without the need for centralized servers. It uses Golang on the backend and supports secure connections using WireGuard-compatible encryption. Weron allows developers to create scalable mesh or point-to-point networks for applications like remote desktop, Kubernetes access, or collaborative tools—all without requiring public IPs or complicated port forwarding. -
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10
LiveKit
End-to-end stack for WebRTC. SFU media server and SDKs
LiveKit is an open-source project that provides a scalable, multi-user conferencing system based on WebRTC, designed to offer real-time video, audio, and data capabilities for developers. -
11
Matchbox
Painless peer-to-peer WebRTC networking for rust wasm
Matchbox is a WebRTC-based networking library for the Bevy game engine, enabling multiplayer games to run in the browser or native environments with peer-to-peer communication. It handles matchmaking, NAT traversal, and synchronization in real-time, making it easier for game developers to add fast and scalable multiplayer features. Designed with simplicity and performance in mind, Matchbox supports both WebAssembly and desktop targets. -
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mediasoup
Cutting Edge WebRTC Video Conferencing
mediasoup is a Node.js library that provides a cutting-edge WebRTC server capable of handling real-time communications with efficient media routing and processing. -
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Selkies-GStreamer
Open-Source Low-Latency Accelerated Linux WebRTC HTML5 Remote Desktop
selkies-gstreamer is a GStreamer-based media streaming component used in the Selkies project, a cloud-native platform designed for interactive desktop and application streaming. This module acts as a high-performance media pipeline that captures video, encodes it with low latency, and streams it via WebRTC to client browsers. It is optimized for GPU-accelerated encoding and integrates with Kubernetes-based deployments to enable scalable, real-time remote desktop sessions. This component plays a critical role in delivering smooth, responsive experiences for cloud-based workstations, gaming, or visualization tools. -
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FastRTC
The python library for real-time communication
FastRTC is a Python library designed to simplify real-time communication (RTC), especially for audio and video streaming applications. It abstracts away much of the complexity that typically comes with implementing WebRTC by providing a simple interface — e.g. a Stream class — that can be mounted within a web backend (for example a FastAPI application). This makes it particularly well suited for building real-time voice (or video) interfaces for applications such as AI assistants, live chat, or collaborative audio/video tools. FastRTC also integrates nicely with UI frameworks (e.g. via a web demo using Gradio), so developers can rapidly prototype and deploy real-time streaming applications without deep knowledge of low-level WebRTC internals. ... -
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libdatachannel
C/C++ WebRTC network library featuring Data Channels, Media Transport
libdatachannel is a C++ library that implements WebRTC DataChannels and ICE transports, enabling peer-to-peer communication without relying on media components. It is ideal for developers who want efficient, low-latency P2P communication for data transmission in IoT, multiplayer games, decentralized apps, or custom signaling architectures. Built for performance and minimalism, it allows fine-grained control over STUN, TURN, DTLS, and SCTP layers, with a clean and modern C++ interface. -
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008
Open-source event-driven AI powered Softphone
...This project is a WebRTC softphone, and communication is achieved via SIP over a socket. Leading PBX systems like Asterisk or Freeswitch support socket connections. If your provider does not offer this feature, consider using a SIP proxy such as Kamailio, Opensip or Routr. The softphone is internally configured using a JSON definition. The configuration file can be loaded from either a server or a local file. 008 reads the file only once. -
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Billd-Live
Live broadcast room built on Vue3 + WebRTC + Nodejs + SRS
Billd-live is a full-stack live streaming platform that enables users to create and participate in real-time video broadcasts through a web-based interface. Built using technologies such as Vue3, WebRTC, Node.js, and FFmpeg, it provides a complete solution for live streaming similar to modern platforms. Users can start live sessions as hosts, while viewers can join and watch streams directly in the browser. The system supports multiple streaming protocols and includes features such as real-time interaction, multi-user connections, and content distribution. ... -
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smart rtmpd
Multi-system-supported and easy-to-maintain streaming media server
...The server is designed to be lightweight and easy to deploy, requiring minimal configuration and no external dependencies. It supports a wide range of protocols including RTMP, HLS, DASH, RTSP, and WebRTC, making it suitable for diverse streaming environments. The system is built with scalability in mind, supporting clustering and cascading setups for large-scale deployments. It also includes features such as authentication, logging, and remote configuration through web interfaces. Overall, it provides a powerful yet simple solution for building streaming infrastructure. -
19
Narrowlink
A self-hosted solution to enable secure connectivity between devices
Narrowlink is a zero-config tunneling and reverse proxy solution that enables secure access to services behind firewalls or NATs without exposing public IPs. Unlike traditional tools like ngrok, Narrowlink is peer-to-peer and privacy-focused, using WireGuard and WebRTC to establish direct encrypted tunnels between peers. It is designed to make exposing local services simple, fast, and secure with no need for port forwarding or cloud relays. Narrowlink is ideal for developers, self-hosters, and remote teams looking for frictionless, encrypted connectivity. -
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P2P Media Loader
An open-source engine for P2P streaming of live and on demand video
p2p-media-loader is a JavaScript library that enables peer-to-peer delivery of video content using WebRTC, reducing server load and improving scalability for streaming platforms. It works seamlessly with HLS and DASH formats by extending existing players like hls.js and dash.js, allowing viewers to share video segments with one another. The library is optimized for browser environments and is ideal for developers looking to build decentralized or bandwidth-efficient media applications. -
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Ovmeet
Video conferencing and collaboration platform
...Supports WebRTC, RTMP, RTSP, SIP protocols, enabling interoperability with streaming devices, hardware endpoints, and different media sources. -
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go2rtc
Ultimate camera streaming application
go2rtc is a lightweight, zero-dependency streaming server designed to unify and convert video streams across a wide range of protocols and devices, particularly in smart home and surveillance environments. Written in Go, it provides real-time streaming capabilities with extremely low latency by supporting protocols such as RTSP, WebRTC, RTMP, HTTP, and HomeKit, while also enabling seamless transcoding using FFmpeg when needed. The application can ingest streams from IP cameras, USB devices, or cloud-based sources and redistribute them to multiple clients or platforms, including browsers and smart home systems like Home Assistant. Its architecture emphasizes flexibility, allowing users to mix multiple input sources, negotiate codecs dynamically, and even enable two-way audio communication with supported devices. go2rtc also includes features for publishing streams to external platforms like YouTube or Telegram, making it useful beyond surveillance scenarios. -
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Yjs
Shared data types for building collaborative software
...It enables multiple users to edit shared data structures—such as text documents, arrays, maps, and XML trees—synchronously and offline. Yjs is network-agnostic and works with WebRTC, WebSocket, or any other transport layer, making it ideal for collaborative editors, whiteboards, and design tools. Its compact updates and powerful reconciliation system make it one of the most efficient and developer-friendly CRDT libraries available. -
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Coturn TURN server
coturn TURN server project
coturn is a free open source implementation of TURN and STUN Server. The TURN Server is a VoIP media traffic NAT traversal server and gateway. -
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Android Emulator Container Scripts
Minimal scripts to run the emulator in a container for various systems
...It includes scripts and container images that configure the emulator for headless operation, wire up networking, and expose endpoints for ADB and web access. A built-in WebRTC bridge lets you stream the emulator screen to a browser with interactive input, which is ideal for CI dashboards, remote debugging, or demo environments. The project focuses on reproducibility and scale: you define which system image to boot, how to persist or reset data, and how many instances to run, then schedule them like any other workload. ...
