Ant Media Server - RTMP, RTSP, WebRTC and Adaptive Bitrate. It can also record videos in MP4, HLS and FLV

•    Java

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Ant Media Server is an open source media server that supports RTMP, RTSP, WebRTC and Adaptive Bitrate. It can also record videos in MP4, HLS and FLV. It also supports WebRTC to RTMP Adapter, IP camera.

media-server webrtc broadcast rtmp rtsp hls mp4 ip-camera mp4-hls flv streaming livestream video-streaming audio-streaming streaming-server video-platform

SmarterStreaming - 国内外为数不多致力于极致体验的超强全自研跨平台(windows/android/iOS)流媒体内核，通过模块化自由组合，支持实时RTMP推流、RTMP播放器、RTSP播放器、录像、多路流媒体转发、音视频导播、动态视频合成、音频混音、直播互动、内置轻量级RTSP服务等，比快更快，业界真正靠谱的超低延迟直播SDK(1秒内，低延迟模式下200~400ms)。

•    Java

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国内外为数不多致力于极致体验的超强全自研跨平台(windows/android/iOS)流媒体内核，通过模块化自由组合，支持实时RTMP推流、RTMP播放器、RTSP播放器、录像、多路流媒体转发、音视频导播、动态视频合成、音频混音、直播互动、内置轻量级RTSP服务等，比快更快，业界真正靠谱的超低延迟直播SDK(1秒内，低延迟模式下200~400ms)。

rtmp-pusher rtmp-player android-rtmp ios-rtmp windows-rtmp daniulive rtsp-player rtmp-client android-publisher rtsp2rtmp player rtmp rtsp rtsp-relay rtmp-h265 hevc rtmpclient rtspclient rtmp-broadcaster rtsp-to-rtmp

webrtc-sdk - WebRTC Simple Calling API + Mobile SDK - A simplified approach to RTCPeerConnection for mobile and web video calling apps

•    Javascript

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WebRTC SDK Upgraded! ES6, new camera control and 100x less code than v1. At PubNub we believe simplicity is essential for our SDK usability. We've taken a simplified approach to WebRTC Peer Connections by creating and easy-to-use SDK for developers. The ideas of simplicity should span all platforms and devices too and that's why we also support Android WebRTC mobile calling with compatibility for iOS native Objective-C based WebRTC SDK. This simple developer WebRTC SDK is powered by PubNub Data Stream Network.

webrtc webrtc-call webrtc-sdk webrtc-video webrtc-receiver webrtc-javascript-library webrtc-phone webrtc-demo webrtc-dialing

webtorrent-hybrid - WebTorrent (with WebRTC support in Node.js)

•    Javascript

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In node.js, the webtorrent package only connects to normal TCP/UDP peers, not WebRTC peers. If you want to connect to all types of peers, including WebRTC peers, from node.js, you need to use this package (webtorrent-hybrid). Previous versions (0.x) of this package used wrtc, a WebRTC implementation via native extensions, the current one is based on electron-webrtc (which in turn uses electron-prebuilt) for better compatibility. It creates a hidden Electron process (which is based on Chromium, so WebRTC support is great!) and communicates with that process to enable WebRTC in Node.js. This adds a lot of overhead, so we are looking forward to using a pure JavaScript implementation, like perhaps Node-RTCPeerConnection when it's ready.

webtorrent nodejs browser torrent bittorrent hybrid webrtc bittorrent-client download mad-science streaming webrtc-data

gortmp - Implement RTMP protocol by golang

•    Go

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RTMP protocol implementation.

rtmp rtmp-server rtmp-player rtmpdump rtmp-protocol

webrtc - A reference gradle project that let you explore WebRTC Android in Android Studio.

•    C++

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WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose. Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

webrtc webrtc-demos webrtc-android

nginx-http-flv-module - Media streaming server based on nginx-rtmp-module, HTTP-FLV/RTMP/HLS/DASH supported

•    C

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Media streaming server based on nginx-rtmp-module. 中文说明.

rtmp http-flv gop-cache media-server nginx-rtmp virtual-hosts nginx-http-flv chunked-transmission live-streaming h264 aac flvjs dynamic-module

yasea - RTMP live streaming client for Android

•    C

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Yasea is an Android streaming client. It encodes YUV and PCM data from camera and microphone to H.264/AAC, encapsulates in FLV and transmits over RTMP. non-gpuimage for demo without GPUImage module.

rtmp android rtmp-stream camera media live-streaming android-sdk android-library gles shader opengles glsl android-development

spreed-webrtc - WebRTC audio/video call and conferencing server.

•    Javascript

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Spreed WebRTC implements a WebRTC audio/video call and conferencing server and web client. The latest source of Spreed WebRTC can be found on GitHub. If you are a user, just wanting a secure and private alternative for online communication make sure to check out the Spreedbox, providing a ready to use hardware with Spreed WebRTC included.

nginx-rtmp-win32 - Nginx-rtmp-module Windows builds.

•    XSLT

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Nginx-rtmp-module Windows builds.

nginx-rtmp win32

html5-dash-hls-rtmp - :sunflower: HTML5播放器、M3U8直播/点播、RTMP直播、低延迟、推流/播流地址鉴权、优化浏览器兼容性，HLS+扩展

•    Javascript

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:sunflower: HTML5播放器、M3U8直播/点播、RTMP直播、低延迟、推流/播流地址鉴权、优化浏览器兼容性，HLS+扩展

rtmp videojs hls m3u8 mediaelement html5 dash player

KSYMediaPlayer_Android - 金山云Android播放SDK（KSYUN Live Streaming player SDK），支持RTMP HTTP-FLV HLS 协议（supporting RTMP HTTP-FLV HLS protocol），直播延时2-3秒（Living delay 2 or 3 seconds），supporting jitter control，software&hardware decode

•    Java

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金山云Android播放SDK（KSYUN Live Streaming player SDK），支持RTMP HTTP-FLV HLS 协议（supporting RTMP HTTP-FLV HLS protocol），直播延时2-3秒（Living delay 2 or 3 seconds），supporting jitter control，software&hardware decode

player player-recording player-api

KSYMediaPlayer_iOS - 金山云iOS播放SDK（KSYUN Live Streaming player SDK），支持RTMP HTTP-FLV HLS 协议（supporting RTMP HTTP-FLV HLS protocol），直播延时2-3秒（Living delay 2 or 3 seconds）

•    Objective-C

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金山云iOS播放SDK（KSYUN Live Streaming player SDK），支持RTMP HTTP-FLV HLS 协议（supporting RTMP HTTP-FLV HLS protocol），直播延时2-3秒（Living delay 2 or 3 seconds）

player player-recording

webrtc - A pure Golang implementation of the WebRTC Native API

•    Go

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See DESIGN.md for an overview of features and future goals. Check out the example applications to help you along your Pion WebRTC journey.

webrtc pion-webrtc ortc rtp srtp webrtc-api

wave-share - Serverless, peer-to-peer, local file sharing through sound

•    C++

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A proof-of-concept for WebRTC signaling using sound. Works with all devices that have microphone + speakers. Runs in the browser. Nearby devices negotiate the WebRTC connection by exchanging the necessary Session Description Protocol (SDP) data via a sequence of audio tones. Upon successful negotiation, a local WebRTC connection is established between the browsers allowing data to be exchanged via LAN.

data-over-sound webrtc fsk p2p webrtc-signaling file-sharing data-transfer modem ultrasonic

webrtc-streamer - WebRTC streamer for V4L2 capture devices and RTSP sources

•    C++

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This is a try to stream video sources through WebRTC using simple mechanism. It embeds a HTTP server that implements API and serve a simple HTML page that use them through AJAX.

webrtc streamer rtsp c-plus-plus v4l2 webrtc-streamer

testrtc - WebRTC Troubeshooter: Collection of tests and tools to help diagnose WebRTC systems.

•    Javascript

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WebRTC troubleshooter provides a set of tests that can be easily run by a user to help diagnose WebRTC related issues. The user can then download a report containing all the gathered information or upload the log and create a temporary link with the report result. Due to their time duration they are not part of the normal test suite and need to be run explicitly.

rtmp - Golang rtmp server

•    Go

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Golang rtmp server

webtty - Share a terminal session over WebRTC

•    Go

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There are a handful of bugs to fix, but everything works pretty well at the moment. Please open an issue if you find a bug. Or, install directly with Go. WebTTY requires go version 1.9 or higher.

webrtc webrtc-experiments tty pion-webrtc

react-native-webrtc - A WebRTC module for React Native.

•    Objective-C

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A WebRTC module for React Native. the order of commit revision is nothing to do with the order of cherry-picks, for example, the earlier committed cherry-pick-#2 may have higher revision than cherry-pick-#3 and vice versa.

react-component react-native ios android webrtc