webrtc: bring real-time to the web - barcamp saigon 2012
TRANSCRIPT
WebRTC Bring real-time to the web
NEW TRENDS of WEB TECHNOLOGY ON MOBILE
VIDEO COMMUNICATION with
HCMC University of Technology
09/2012
I. What is WebRTC ?
II. Key Features
1. Media Stream
2. Peer Connection
3. Data Channels
III. Applications
IV. Demos
TỔNG QUAN WebRTC Bring real-time to the web
TỔNG QUAN
Story of Google • Justin Uberti • Google Hangout, Google Video Chat
Gmail Call Phone • Plugins
- Really Complicated - Security - Codec, Licensing - Other browsers, manufacturers
Build one platform, not just for web, but for the entire communications industry.
TỔNG QUAN
What is WebRTC ?
• Real Time Communications meets the web • A state-of-the-art audio/video communication stack in your web
browser • A cross-industry effort to create a new communications platform
“WebRTC and HTML5 could enable the same transformation for real time that the original browser did for information.”
Phil Edholm
TỔNG QUAN
WebRTC Support
• Desktop browsers - Chrome 21 - Opera 12 - Firefox 17 - IE ?
• Mobile browsers • Native C++
• Desktop and mobile
2013
04/2012
01/2012
05/2011
04/2011
Release
Mozilla Firefox nightly build
Google Chrome dev
W3C WebRTC WG
IETF RTCWeb WG
ỨNG DỤNG
TRONG ĐÀO TẠO TỪ XA Key Features
I. What is WebRTC ?
II. Key Features
1. Media Stream
2. Peer Connection
3. Data Channels
III. Applications
IV. Demos
TỔNG QUAN
Web Server
Browser Browser
Signaling
path
Web Server
Media path
Application defined over
HTTP / Websockets
Application defined over
HTTP / Websockets
Key Features
1. Media Stream Access audio and video
Media Stream
• Represent a MediaSource • getUserMedia API to access camera/microphone • Use with <video> as an URL • Send to remote peer Combine with other HTML5 for funny effects • <canvas> • CSS • WebGL
getUserMedia
<script> navigator.webkitGetUserMedia({video:true}, onGotStream, onFailedStream); onGotStream = function(stream) { var url = webkitURL.createObjectURL(stream); video.src = url; } </script> <video id="video" autoplay="autoplay" />
Key Features
2. Peer Connection Audio and video session
PeerConnection
API for establishing audio/video calls Built-in • Peer-to-peer • Codec control • Encryption • Bandwidth management
Setup a session To start a session, a client needs • Local Session Description • Remote Session Description • Remote Session Candidates
Setup a session 1. Create Local Session Description 2. Send it to remote peer B (OFFER) 3. Receive Session Description from peer A 4. Create Session Description send back to peer A (ANSWER) 5,6. Send ICECadidate to other peer 7. Setup media path
2 3
1 4
5 6
7
PeerConnection API
Caller side Create a new PeerConnection PeerConnection(config, iceCallback) addStream(stream) Create local SessionDescription createOffer(hints) setLocalDescription(type, desc) startIce() <wait for response from callee> Receive remote SessionDescription setRemoteDescription(type, desc)
Callee side <receive call from caller> Create PeerConnection PeerConnection(config, iceCallback) setRemoteDescription(type, desc) Create local SessionDescription createAnswer(offer, hints) setLocalDescription(type, desc) startIce()
Sample Code
<script> pc1 = new webkitPeerConnection00 (null, onIceCandidate1); // create PC pc2 = new webkitPeerConnection00 (null, onIceCandidate2); // create PC pc2.onaddstream = onRemoteStream; pc1.addStream (localStream); // add local stream var offer = pc1.createOffer(null); // create an offer pc1.setLocalDescription(pc1.SDP_OFFER, offer); // set it on both PC pc2.setRemoteDescription(pc2.SDP_OFFER, offer); var answer = pc2.createAnswer(offer.toSdp(), null); // create an answer pc2.setLocalDescription(pc2.SDP_ANSWER, answer); // set it on both PC pc1.setRemoteDescription(pc1.SDP_ANSWER, answer); pc1.startIce(); // start the connection process pc2.startIce(); </script>
WebRTC Signaling Channel • XMLHttpRequest (AJAX) • WebSocket • Google App Engine
Key Features
3. Data Channels Peer-to-peer data exchange in browsers
Data Channel Peer-to-peer exchange of arbitrary application data
• Low latency • High message rate/thoughput • Reliable and unreliable semantics
Use cases • Multiplayer game • Remote desktop • Real-time interactive (chat, drawing…) • File transfer • Decentralized networks
Sample Code
<script> dc1 = pc1.createDataChannel ("a label"); // reliable mode dc2 = pc2.createDataChannel ("a label"); dc2.onmessage = function(e) { textarea.value += e.data; } function send() { dc1.send(input.value); } </script>
Web Server
Web Server
CƠ SỞ LÝ THUYẾT Applications
APPLICATIONS Video Communication
Gaming
E-Commerce
Live Video
Record + Replay
Phone Call
File Transfer
Remote Desktop
VIDEO COMMUNICATION
Web Server
Web Server Media Server
Web
Server
Media
Server
Media
Server
Live Video
Providers 1 Providers 2
SIP
Start ups
Zingaya (Call' button for websites) enables voice calls through any computer from a webpage. No download or phone is required.
Voxeo Labs (Open source enabler for WebRTC services) Phono is a jQuery plug-in that turns any Web browser into a multichannel communications platform
Utribo (SaaS Service) 'Connect' by Utribo is a Software as a Service that enables subscribers to receive calls made in a web browser to their computer, phone, ….
Tenhands (Enterprise HD Video Collaboration) Desktop HD video collaboration service, it's free and built for business needs.
Bistri (Social Video) Video chat with fun video effects, take screenshots of calls, share them with friends or social networks. Bistri runs in the browser, no need to install additional software or plugins.
WebRTC Bring real-time to the web
Nguyễn Mậu Quang Vũ [email protected]
WebRTC Bring real-time to the web
Phạm Nguyên Trình [email protected]
HCMC University of Technology