Distribution @ Twitch - Media over QUIC
published 2/15/2022 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
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 㮠Unified Plan 㨠Plan B | blog.jxck.io
Intro 新年早々ã€Blink Dev 㧠Unified Plan ã® Intent to Implement ã¨ã„ã†å¬‰ã—ã„知らã›ãŒå±Šã„ãŸã€‚ Intent to Implement: WebRTC Unified Plan SDP SDP ã®äº’æ›æ€§ã«ã¤ã„ã¦ã‚¤ãƒ³ãƒ‘クトã®å¤§ãã„ã“ã®å¤‰æ›´ã«ã¤ã„ã¦ç°¡å˜ã«è§£èª¬ã™ã‚‹ã€‚ Update 実装ãŒé€²ã¿ SdpFormat 㯠sdpSemantics ã«å¤‰ã‚ã£ãŸãŸã‚ã€è¨˜äº‹ã‚’ä¿®æ£ã€‚ PSA: Unified Plan SDP testing flag is now available on Canary PSA: RTCRtpTransceiver shipping in M69 behind sdpSemantics:'unified-plan' ãªãŠã€ä»¥ä¸‹ã®ãƒ•ãƒ©ã‚°ã‚’付ã‘ã¦èµ·å‹•ã™ã‚‹ã¨ãƒ‡ãƒ•ã‚©ãƒ«ãƒˆã§æœ‰åŠ¹ã«ã§ãる。 --enable-blink-features
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
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好奇心旺盛ãªäººã®ãŸã‚ã®WebRTC #ã“ã®æœ¬ã¯ã€WebRTCã®å®Ÿè£…者ãŒè‹¦åŠ´ã—ã¦å¾—ãŸçŸ¥è˜ã‚’世界ã«å‘ã‘ã¦ç™ºä¿¡ã™ã‚‹ãŸã‚ã«ä½œæˆã•ã‚Œã¾ã—ãŸã€‚ 好奇心旺盛ãªäººã®ãŸã‚ã®WebRTC ã¯ã€å¸¸ã«ã‚ˆã‚Šå¤šãã®ã“ã¨ã‚’求ã‚ã¦ã„る人ã®ãŸã‚ã«æ›¸ã‹ã‚ŒãŸã‚ªãƒ¼ãƒ—ンソースã®æ›¸ç±ã§ã™ã€‚ ã“ã®æœ¬ã¯æŠ½è±¡åŒ–ã•ã‚ŒãŸã‚‚ã®ã§ã¯ã‚ã‚Šã¾ã›ã‚“。 ã“ã®æœ¬ã¯ãƒ—ãƒãƒˆã‚³ãƒ«ã¨APIã«é–¢ã™ã‚‹ã‚‚ã®ã§ã€ç‰¹å®šã®ã‚½ãƒ•ãƒˆã‚¦ã‚§ã‚¢ã«ã¤ã„ã¦èªžã‚‹ã‚‚ã®ã§ã¯ã‚ã‚Šã¾ã›ã‚“。 ç§ãŸã¡ã¯RFCã‚’è¦ç´„ã—ã€æ–‡æ›¸åŒ–ã•ã‚Œã¦ã„ãªã„ã™ã¹ã¦ã®çŸ¥è˜ã‚’一箇所ã«é›†ã‚ã‚‹ã“ã¨ã‚’試ã¿ã¾ã™ã€‚本書ã¯ãƒãƒ¥ãƒ¼ãƒˆãƒªã‚¢ãƒ«ã§ã¯ãªã„ã®ã§ã€ã‚³ãƒ¼ãƒ‰ã¯ã‚ã¾ã‚Šå«ã¾ã‚Œã¾ã›ã‚“。 WebRTCã¯ç´ 晴らã—ã„技術ã§ã™ãŒã€ä½¿ã„ã“ãªã™ã®ã¯é›£ã—ã„ã‚‚ã®ã§ã™ã€‚ã“ã®æœ¬ã¯ãƒ™ãƒ³ãƒ€ãƒ¼ã«ä¾å˜ã›ãšã€åˆ©ç›Šç›¸åを排除ã™ã‚‹ã‚ˆã†ã«ã—ã¦ã„ã¾ã™ã€‚ ã“ã®æœ¬ã¯èª°ã®ãŸã‚ã®ã‚‚ã®ã‹ã€‚ #WebRTC ãŒä½•ã‚’解決ã™ã‚‹ã®ã‹ã•ãˆçŸ¥ã‚‰ãšã€ã‚‚ã£ã¨å¦ã³ãŸã„ã¨æ€ã£ã¦ã„る開発者。既㫠WebRTC を使ã£
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
Reactã¨WebRTCã§Zoomã®ã‚ˆã†ãªãƒ“デオãƒãƒ£ãƒƒãƒˆã‚¢ãƒ—リを作ã£ã¦ãƒ‡ãƒ¼ã‚¿ãƒ•ãƒãƒ¼ã‚’図解ã—ã¦ã¿ãŸ
ã¯ã˜ã‚ã« ã“ã‚“ã«ã¡ã¯ã€‚ 都内在ä½ã®ãƒ•ãƒãƒ³ãƒˆãƒ³ãƒ‰ã‚¨ãƒ³ã‚¸ãƒ‹ã‚¢ã§ã™ã€‚ 僕ã¯ã¨ã‚る会社ã«ã¦ç´„ 1 å¹´åŠã»ã© React 㨠WebRTC を用ã„ã¦æ˜ åƒé…ä¿¡ã®ã‚¢ãƒ—リケーション開発を行ã£ã¦ãã¾ã—ãŸã€‚ ãã“ã§ã¯é–‹ç™ºã‚’スムーズã«é€²ã‚る為㫠WebRTC ã® SDK を利用ã—ã¦ã„ã¦ã€ 本æ¥å¦ç¿’コストãŒé«˜ã„ã¨ã•ã‚Œã¦ã„ã‚‹ WebRTC をカジュアルã«åˆ©ç”¨ã™ã‚‹ã“ã¨ãŒã§ãã¦ã„ã¾ã™ã€‚ ã—ã‹ã—ã€ã‚ˆã‚Šå…¥ã‚Šçµ„ã‚“ã 実装をã—ãŸã‚Šæ˜ åƒé…信特有ã®å•é¡Œï¼ˆå¾Œè¿°ï¼‰ を解決ã™ã‚‹ã¨ãªã‚‹ã¨ä»¥ä¸‹ 3 ã¤ã® WebAPI ã®ç†è§£ã¯é¿ã‘ã¦é€šã‚Œã¾ã›ã‚“。 MediaStream RTCPeerConnection WebSocket 詳ã—ãã¯æ–‡ä¸ã«è¨˜è¼‰ã—ã¾ã™ãŒã“れらã®ç†è§£ã‚’æ·±ã‚ãªã„ã¨é–‹ç™ºã®é€²è¡Œã«å¤§ããªå½±éŸ¿ãŒã‚ã‚‹ã¨æ€ã£ãŸã®ã§ã€WebRTC 関連ã®ãƒ©ã‚¤ãƒ–ラリç‰ã‚’利用ã›ãšã«æ˜ åƒé…ä¿¡ã®ã‚¢ãƒ—リケーションを作ã£ã¦å¦ç¿’ã—よã†ã¨ã„ã†è€ƒãˆã«ãªã‚Šã€å®Ÿéš›ã«ä½œã£ã¦ã¿ã¾ã—ãŸã€‚
フãƒãƒ³ãƒˆã‚¨ãƒ³ãƒ‰ x RTC界隈ã®æœ€è¿‘ã¨ã“ã‚Œã‹ã‚‰ - console.lealog();
フãƒãƒ³ãƒˆã‚¨ãƒ³ãƒ‰ã‚¨ãƒ³ã‚¸ãƒ‹ã‚¢ã‹ã‚‰ã¿ã‚‹ã€ã“ã®ç•Œéšˆã§ä»Šã©ã‚“ãªIssueãŒè©±é¡Œã«ãªã£ã¦ã‚‹ã®ã‹ã¨ã€ã“ã®å…ˆã©ã†ã„ã†å‹•ããŒã‚ã‚Šãã†ã‹ã«ã¤ã„ã¦ã€‚ ãã“ã¾ã§è‡ªåˆ†ã«å…ˆè¦‹ã®æ˜ŽãŒã‚ã‚‹ã¨ã‚‚æ€ã£ã¦ãªã„ã‘ã©ã€ã‚¢ã‚¦ãƒˆãƒ—ットã—ã¦ãŠã‹ãªã„ã¨å¿˜ã‚Œã¦ã—ã¾ã„ãã†ãªã®ã§ãƒ»ãƒ»ã€‚ ã¡ãªã¿ã«ã“ã“ã§ã„ã†ãƒ•ãƒãƒ³ãƒˆã‚¨ãƒ³ãƒ‰ã¯ã€ã„ã‚ゆるブラウザã¨ã‹JavaScriptã®APIã®ã“ã¨ã§ã™ã€‚ プãƒãƒˆã‚³ãƒ«çš„ãªå´é¢ã¯ãã“ã¾ã§è©³ã—ããªã„ã®ã§ã‚ã¾ã‚Šè§¦ã‚Œã¾ã›ã‚“。 WebRTC 1.0 GitHub - w3c/webrtc-pc: WebRTC 1.0 API ã¾ãšã€RTCã¨ã„ãˆã°ã‚ºãƒãƒªã®WebRTCã‹ã‚‰ã€‚ 昨年末ã«WDã‹ã‚‰CRã¸æ ¼ä¸Šã’ã¨ã„ã†ã“ã¨ã§ã€ã‚‚ã†APIãŒæ¿€å¤‰ã—ãŸã‚Šã¯ã—ãªã„・・ã¯ãšã€‚ 実際ã®ã¨ã“ã‚ã€ã“ã“åŠå¹´ãらã„大ããªå¯¾å¿œã—ãŸè¦šãˆãŒãªã„ã§ã™ã€‚（WebRTCãã®ã‚‚ã®ã‚’実装ã—ã¦ã‚‹äººã¯ã€åœ°å‘³ã«ã„ã‚ã„ã‚対応ã—ã¦ã‚‹ã¨æ€ã†ã‘ã©ï¼‰ ガワã®APIã¨ã„ã†è¦³ç‚¹ã§ã„ã†ã¨ã€æœ€è¿‘ã¯ã‚‚
chrome://webrtc-internals ã®ã—ãã¿ - console.lealog();
ãªã‚“ã¨ãªãアテã¯ã¤ã„ã¦ãŸã‘ã©ã€ä¸èº«ãŒæ°—ã«ãªã‚‹ã¨ã„ã†è©±ã‚’èžã„ãŸã®ã§èª¿ã¹ã¦ã¿ãŸã€‚ WebRTCã‚„ã£ã¦ã‚‹ã¿ãªã•ã‚“ãªã‚‰ãŠé¦´æŸ“ã¿ã®ã‚ã®ãƒšãƒ¼ã‚¸ã§ã™ã€‚ chrome://webrtc-internals WebRTCã®ãƒ‡ãƒãƒƒã‚°ã¨ã„ãˆã°ã“ã®ãƒšãƒ¼ã‚¸ã€‚ 特ã«ä½•ã‚‚仕込んã§ãªã„ã®ã«ã€è¦‹ã¦ã‚‹ãƒšãƒ¼ã‚¸ã§`getUserMedia()`ã—ãŸã‚Š`RTCPeerConnection`ãŒä½œã‚‰ã‚Œã‚Œã°ãã®æ§˜åãŒè¦‹ãˆã‚‹ã—ã€å®Ÿéš›ã«æµã‚Œã¦ã‚‹ãƒ¡ãƒ‡ã‚£ã‚¢ã‚„データã®ã“ã¨ã¾ã§ã‚ã‹ã‚‹ã€‚ ã‚ã‚Œã£ã¦ã©ã†ã„ã†ã—ãã¿ï¼Ÿã£ã¦ã„ã†ã®ã‚’調ã¹ã¦ã„ãã¾ã™ã€‚ ãŸã ã®Webページ URLãŒ`chrome://`ã«ãªã£ã¦ã‚‹ã‘ã©ã€ã‚Œã£ãã¨ã—ãŸWebページã§ã™ã€‚ ãªã®ã§DevToolsã§Networkタブ見れã°ã ã„ãŸã„ã‚ã‹ã‚‹ï¼ã¨ã„ã†ã‚ã‘ã§ã€‚ 構æˆè¦ç´ ã¯ã“ã‚“ãªæ„Ÿã˜ã€‚ webrtc-internals.html ãŸã ã®HTML/CSS 以下ã®2ã¤ã®JSã‚’èªã¿è¾¼ã‚“ã§ã‚‹ util.j
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Stackless и stackful? Корутины и аÑинхронноÑÑ‚ÑŒ в Go
輪講資料 ICE（Interactive Connectivity Establishment）
輪講資料 ICE(Interactive Connectivity Establishment) ååŸŽå¤§å¦ ç†å·¥å¦éƒ¨ æƒ…å ±å·¥å¦ç§‘ æ¸¡é‚Šç ”ç©¶å®¤ B4 140441043 鴨下å‹é¦¬ 1/61 ICE åŸºæœ¬äº‹é … ï¬RFC5245ã§æ¨™æº–化ã•ã‚Œã¦ã„ã‚‹UDPベースã®NAT越ãˆãƒ—ãƒãƒˆã‚³ãƒ« ï¬ã‚ªãƒ•ã‚¡ãƒ¼ï¼ã‚¢ãƒ³ã‚µãƒ¼ãƒ¢ãƒ‡ãƒ«ã‚’利用ã™ã‚‹ä»»æ„ã®ãƒ—ãƒãƒˆã‚³ãƒ«ã§ 使用å¯èƒ½ã§ã‚る． ï¬ã‚ªãƒ•ã‚¡ãƒ¼ï¼ã‚¢ãƒ³ã‚µãƒ¼ãƒ¢ãƒ‡ãƒ«ï¼šã‚»ãƒƒã‚·ãƒ§ãƒ³ã‚’開始ã™ã‚‹ãŸã‚ã«ï¼Œä¸€æ–¹ãŒè‡ªã‚‰ã®è¦–点 ã‹ã‚‰æƒ…å ±ã‚’æä¾›ã—（オファー），もã†ä¸€æ–¹ãŒç›¸æ‰‹ã«å¯¾å¿œã™ã‚‹æƒ…å ±ã‚’è¿”ç”ã™ã‚‹ （アンサー）モデル．RFC3264ã§æ¨™æº–化． ï¬ã‚ªãƒ•ã‚¡ãƒ¼ï¼ã‚¢ãƒ³ã‚µãƒ¼ãƒ¢ãƒ‡ãƒ«ã‚’利用ã™ã‚‹ãƒ—ãƒãƒˆã‚³ãƒ«ã®ä¾‹ï¼š ï¬ SIP(Session Initiation Protocol)ï¼šéŸ³å£°ã‚„æ˜ åƒï¼Œãƒ†ã‚ストメッセージã®äº¤æ›ãªã©ã‚’ è¡Œã†ãŸã‚ã«å¿…è¦ãªã‚»ãƒƒã‚·ãƒ§ãƒ³ã®ç”Ÿæˆãƒ»å¤‰æ›´ãƒ»åˆ‡æ–ã‚’è¡Œã†ãƒ—ãƒãƒˆã‚³ãƒ« ï¬STUN，TURNã¨ã„ã†2ã¤
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)
RTCQuicTransport Explainer/Design Doc
ã“ã®ãƒ–ラウザ ãƒãƒ¼ã‚¸ãƒ§ãƒ³ã®ã‚µãƒãƒ¼ãƒˆã¯çµ‚了ã—ã¾ã—ãŸã€‚サãƒãƒ¼ãƒˆã•ã‚Œã¦ã„るブラウザã«ã‚¢ãƒƒãƒ—グレードã—ã¦ãã ã•ã„。
RTCQuicTransport & RTCIceTransport Chrome Design Doc
ã“ã®ãƒ–ラウザ ãƒãƒ¼ã‚¸ãƒ§ãƒ³ã®ã‚µãƒãƒ¼ãƒˆã¯çµ‚了ã—ã¾ã—ãŸã€‚サãƒãƒ¼ãƒˆã•ã‚Œã¦ã„るブラウザã«ã‚¢ãƒƒãƒ—グレードã—ã¦ãã ã•ã„。
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Open Sourcing Dyte’s Device Emulator
Twingate:Â It's time to ditch your VPN
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
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