Search Results for "webrtc"
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Showing 18 open source projects for "webrtc"
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Elixir WebRTC
An Elixir implementation of the W3C WebRTC API
Elixir WebRTC is an Elixir implementation of the W3C WebRTC API for building real-time communication features in Elixir applications. It gives developers a way to create peer connections, exchange media, and work with browser-compatible WebRTC behavior from the BEAM ecosystem. The project is especially useful for applications that need live audio, video, or data communication without leaving Elixir. -
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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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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. -
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ZLMediaKit
WebRTC/RTSP/RTMP/HTTP/HLS/HTTP-FLV/WebSocket
ZLMediaKit is a high-performance C++11 media server and streaming SDK for real-time audio and video applications. It supports a wide range of protocols, including RTSP, RTMP, HLS, HTTP-FLV, WebSocket-FLV, GB28181, HTTP-TS, HTTP-fMP4, MP4, and WebRTC. The project is designed for protocol conversion, live streaming, low-latency playback, recording, forwarding, and large-scale client access. Its asynchronous, multithreaded network model makes it suitable for commercial streaming deployments with many concurrent connections. ZLMediaKit can be deployed directly as a complete media server or embedded through its standard C API as an SDK for other languages and systems. ... -
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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. -
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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. -
9
AnalysisAVP
Encode decode, rgb yuv h264 aac flv mp4 rtmp
...It covers foundational topics such as encoding, decoding, color formats like RGB and YUV, and widely used codecs including H.264 and AAC. The project also explores media container formats like MP4 and FLV, along with streaming protocols such as RTMP and WebRTC, offering a broad understanding of media transmission. It includes practical examples and experimental code that demonstrate real-world implementations such as camera capture, rendering, and encoding workflows. The repository also references various multimedia frameworks and libraries like FFmpeg, SDL2, and x264, helping users understand their integration. ... -
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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. ... -
11
Metastream
Watch streaming media with friends
...New features are added on top of streaming websites such as real-time chat and timestamp markers. Synchronized playback of streaming media across various websites. Public, private, and offline sessions. Support for WebRTC peer-to-peer connections. Easily add watch party support to your website by redirecting the user to Metastream. Metastream used to be an Electron desktop application until development was stopped due to roadblocks in acquiring a Widevine license. The latest version can be found on the GitHub releases page. -
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Membrane Core
The core of Membrane Framework, multimedia processing framework
membrane_core is the foundation of the Membrane multimedia framework for Elixir, providing the abstractions and runtime needed to build real-time audio and video pipelines. It models media processing as a graph of lightweight, supervised OTP processes—elements connected by links—so work is isolated, fault-tolerant, and easy to scale or reconfigure at runtime. The core defines a clear lifecycle and callback API for elements, plus concepts like buffers, events, and capabilities/format... -
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Peer Calls
Group peer to peer video calls for everyone written in Go
Peer Calls is a self-hosted, open-source WebRTC-based video and audio calling platform for group communication. Designed for simplicity and privacy, it allows anyone to run their own video conferencing service without relying on third-party providers. Peer Calls supports multi-user rooms, screen sharing, and chat, all delivered via a clean web interface. It’s great for small teams, communities, and educational groups seeking secure and customizable alternatives to mainstream conferencing tools. -
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simple-peer
Simple WebRTC video, voice, and data channels
simple-peer is a lightweight JavaScript library that simplifies peer-to-peer WebRTC connections for real-time communication directly between browsers and devices. The project abstracts away much of the complexity of the native WebRTC API, allowing developers to establish audio, video, and data channels with minimal configuration. It supports direct peer communication without requiring heavy signaling infrastructure beyond the initial connection exchange. simple-peer is commonly used in browser-based chat applications, multiplayer games, collaborative tools, decentralized systems, and streaming platforms. ... -
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Telegram WebRTC (VoIP)
Voice chats, private incoming and outgoing calls in Telegram
Telegram WebRTC (VoIP) is a Python and C++ library that enables real-time voice and video communication features for Telegram bots and clients. It provides an interface for joining, managing, and streaming audio or video in Telegram group calls and voice chats. The library is built on top of low-level communication protocols, ensuring efficient handling of real-time media streams. -
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anyLive
RTMP streamer, RTMP (HLS) second-on player, live streaming on demand
anyLive is a cross-platform live streaming framework that provides a complete end-to-end solution for audio and video capture, encoding, transmission, decoding, and rendering. Built on a C++ core and leveraging WebRTC and RTMP technologies, it supports multiple operating systems including Android, iOS, Windows, and Linux using a unified codebase. The project is designed for real-time communication and live broadcasting applications, offering low-latency streaming and compatibility with standard RTMP servers and CDNs. It integrates support for multiple protocols such as RTMP, HLS, HTTP, and RTSP, enabling flexible streaming scenarios. ... -
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PHP HTML5 Videochat
Simple PHP setup for HTML5 Videochat web application by VideoWhisper.
Using the HTML5 videochat interface is highly recommended as main browsers have plans to discontinue Flash support in 2020. HTML5 Videochat uses WebRTC technology to allow broadcasting webcam directly from website, without need for Flash. This is a simple embedding preview edition, with simple scripts to embed app and showcase few features. Live demo for this edition is available at: https://videowhisper.com/demos/html5-videochat/ For a full implementation of advanced capabilities, see turnkey site edition available as WordPress plugin with full php source: https://paidvideochat.com/html5-videochat/ Compatible turnkey complete hosting plans with HTML5 relay streaming and installation included: https://webrtchost.com/hosting-plans/#Complete-Hosting Contact VideoWhisper anytime for clarifications, suggestions, custom development evaluation: https://videowhisper.com/tickets_submit.php -
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MCU Media Server
SIP Video Multiconference Media Server with WebRTC support.
REPOSITORY MOVED TO GITHUB!! https://github.com/medooze/media-server Video Multiconference Media Server with WebRTC support. Provide Multiconference and video broadcasting services to any SIP service. Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface. JSR309 driver implementation under development. .
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