Twingate: It's time to ditch your VPNTwingate makes Zero Trust Network Access easy to deploy, even easier to use, and always secure. Twingate makes Zero Trust Network Access easy to deploy, even easier to use, and always secure.
Distribution @ Twitch - Media over QUIC
Source This document was originally shared on MoQ IETF mailing list. This is a verbatim copy of the original, preserved without Google Docs. Introduction This is an attempt to document the issues that Twitch/Amazon IVS has encountered with various distribution protocols over the last 8 years. HLS We initially used RTMP for distribution but switched to HLS something like 8 years ago. We use MSE to
GitHub - httptoolkit/mockrtc: Powerful friendly WebRTC mock peer & proxy
Let's write an automated test with MockRTC. To test WebRTC-based code, you will typically need to: Start a MockRTC mock session Define rules that match and mock the traffic you're interested in Create a WebRTC connection to a mock peer, by either: Using MockRTC's offer or answer directly. Applying the provided MockRTC.hookWebRTCConnection hook function to your RTC connection, and then connecting t
SDP の Unified Plan と Plan B | blog.jxck.io
Intro 新年早々、 Blink Dev で Unified Plan の Intent to Implement という嬉しい知らせが届いた。 Intent to Implement: WebRTC Unified Plan SDP SDP の互換性についてインパクトの大きいこの変更について簡単に解説する。 Update 実装が進み SdpFormat は sdpSemantics に変わったため、記事を修正。 PSA: Unified Plan SDP testing flag is now available on Canary PSA: RTCRtpTransceiver shipping in M69 behind sdpSemantics:'unified-plan' なお、以下のフラグを付けて起動するとデフォルトで有効にできる。 --enable-blink-feature
Stadia’s loss is Clubhouse’s gain: the social audio company has poached a longtime Google engineer
Tech/Google/AppsStadia’s loss is Clubhouse’s gain: the social audio company has poached a longtime Google engineer Stadia’s loss is Clubhouse’s gain: the social audio company has poached a longtime Google engineer / Justin Uberti is joining Clubhouse By Jay Peters, a news editor who writes about technology, video games, and virtual worlds. He’s submitted several accepted emoji proposals to the Uni
はじめに
WebRTC 1.0 is a W3C Recommendation
The Web Real-Time Communications Working Group has published WebRTC 1.0: Real-Time Communication Between Browsers as a W3C Recommendation. This document defines a set of JavaScript APIs to allow media and generic application data to be exchanged with another browser or device implementing the appropriate set of real-time protocols defined in IETF. WebRTC already serves as a cornerstone of online c
ReactとWebRTCでZoomのようなビデオチャットアプリを作ってデータフローを図解してみた
フロントエンド x RTC界隈の最近とこれから - console.lealog();
chrome://webrtc-internals のしくみ - console.lealog();
輪講資料 ICE（Interactive Connectivity Establishment）
輪講資料 ICE(Interactive Connectivity Establishment) 名城大学 理工学部 情報工学科 渡邊研究室 B4 140441043 鴨下友馬 1/61 ICE 基本事項 RFC5245で標準化されているUDPベースのNAT越えプロトコル オファー／アンサーモデルを利用する任意のプロトコルで 使用可能である． オファー／アンサーモデル：セッションを開始するために，一方が自らの視点 から情報を提供し（オファー），もう一方が相手に対応する情報を返答する （アンサー）モデル．RFC3264で標準化． オファー／アンサーモデルを利用するプロトコルの例： SIP(Session Initiation Protocol)：音声や映像，テキストメッセージの交換などを 行うために必要なセッションの生成・変更・切断を行うプロトコル STUN，TURNという2つ
On the Road to WebRTC 1.0, Including VP8
Safari 11 was the first Safari version to support WebRTC. Since then, we have worked to continue improving WebKit’s implementation and compliance with the spec. I am excited to announce major improvements to WebRTC in Safari 12.1 on iOS 12.2 and macOS 10.14.4 betas, including VP8 video codec support, video simulcast support and Unified Plan SDP (Session Description Protocol) experimental support.
RTCQuicTransport Coming to an Origin Trial Near You (Chrome 73) | Blog | Chrome for Developers
What? The RTCQuicTransport is a new web platform API that allows exchanging arbitrary data with remote peers using the QUIC protocol. It’s intended for peer to peer use cases, and therefore is used with a standalone RTCIceTransport API to establish a peer-to-peer connection through ICE. The data is transported reliably and in order (see section below for details on unordered & unreliable delivery)
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GitHub - livekit/livekit: End-to-end stack for WebRTC. SFU media server and SDKs.
GitHub - shinyoshiaki/werift-webrtc: WebRTC Implementation for TypeScript (Node.js), includes ICE/DTLS/SCTP/RTP/SRTP/WEBM/MP4
Deskreen
モンスターストライクのリアルタイム通信を支える技術
RTCQuicTransport Explainer/Design Doc
RTCQuicTransport & RTCIceTransport Chrome Design Doc
WebRTC の未来
