the only solution currently available to enable high frame rate and low-bandwidth streaming to a web browser, a low-latency streaming model for audio, video and binary data delivery for Web & Mobile apps.
WebRTC is a protocol designed for secure low-latency streaming of video, audio, and arbitrary data.
Since Modern web browsers and platforms can decode H.264 video streams without any extra plugins. Once the client has a local GPU with hardware video decoding, WebRTC automatically uses the GPU to accelerate the decoding of the stream, which offloads work from the CPU and improves frame rates. The HTML5 video element is the only markup element required and enables streaming of content inline with existing websites, single app streaming, or full-screen remote-desktop use cases.
Actutally, the FB messenger & Google meet & Grome Remote Desktop Apps use WebRTC instead of VNC / RDP.
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Social Meida
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Gaming
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Science
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Retail
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VR/AR
git clone https://github.com/QueenieCplusplus/ChatRoom_Nodejs_App.git
cd ChatRoom_Nodejs_App
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Firebase Hosting 託管 Web App 的 靜態資產
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Cloud Firestore 可將結構化數據保存在雲端，並在數據更新時獲得即時通知 類似 Firebase Messaging
注意：用於發信號的另一個選項可以是Firebase Cloud Messaging。但是，Chrome目前僅支持該功能， 在此代碼實驗室中，我們將重點介紹一種可在所有支持 WebRTC 的瀏覽器上使用的解決方案。 -
FirebaseRTC 聊天室 API (SDP)
npm -g install firebase-tools
check the version is v6.7.1
firebase login
firebase use --add
firebase serve --only hosting
[output]
hosting: Local server: http://localhost:5000
您應該看到已連接到 Firebase 項目的 FirebaseRTC republicate。
該應用程序已自動連接到您的 Firebase 項目。
在此應用程序中，每個視頻聊天會話稱為一個房間。 用戶可以通過單擊應用程序中的按鈕來創建新房間。這將生成一個 ID，供遠程方加入同一房間。 該ID用作Cloud Firestore中每個房間的密鑰。
每個房間都將包含要約 Invitation 和答案 Answer 的 RTCSessionDescriptions 會議描述， 以及來自各方的帶有 ICE 候選人的兩個單獨的集合。
該 ID 用作 Cloud Firestore 中每個房間的密鑰。
https://cloud.google.com/solutions/gpu-accelerated-streaming-using-webrtc
https://developer.mozilla.org/zh-TW/docs/Web/API/WebRTC_API
The H.264 profile for WebRTC does not support lossless encoding. If you require lossless formats, you should consider other solutions such as PCoIP from Teradici.