Getting Started with WebRTC

Getting started with WebRTCChad Hart

12/10/2014 Scotch Streaming and Scaling 1


http://www.meetup.com/WebRTC-Boston/

Meetup #1 Links & Resources: goo.gl/UvbI27

http://www.meetup.com/WebRTC-Boston/

http://goo.gl/UvbI27


COMPANY CONFIDENTIAL © COPYRIGHT 2013 DIALOGIC INC. ALL RIGHTS RESERVED. 4

IP Telephony + WebRTC Media Server

PowerMedia™ XMS

Multi-party conferencing InterworkingTranscoding

Stream processing Person-to-machineRecording

Who uses WebRTC today?







http://youtu.be/Z9h30NcVy4E

http://youtu.be/Z9h30NcVy4E



Use case: Augmented Reality


Use case: facial biometric verification


Use case: controlling robots

12/10/2014 Scotch Streaming and Scaling

http://www.tate.org.uk/whats-on/tate-britain/special-event/after-dark

12

Use case: dog training


https://webrtchacks.com/javascript-dog-trainer/

Use case: peer-to-peer data

CDNs

File Transfer

Private Networks

bitcoin

Peer web-servers

Internet of Things


grimwire

wearables, security, industrial machine vision, ???


WebRTC is on a lot of devices

WebRTC will have a lot of users



WebRTC is world-class VoIP engine for free

OPUS

On2

Global IP Solutions

.org

Video CodecGoogle acquires On2 for $133

million

VoIP Media EngineGoogle acquires Global IP

Solutions for $68.2M

Audio codecOpus - open, royalty-free, highly

versatile standardized audio codec

Open Source Communities


WebRTC is a Web and Internet Standard

webrtc rtcweb

http://www.w3.org/TR/webrtc/ https://tools.ietf.org/wg/rtcweb/

http://www.w3.org/TR/webrtc/

https://tools.ietf.org/wg/rtcweb/


WebRTC is State-of-the-Art Technology

Firewall/NAT Traversal• ICE/STUN/TURN• RTCP Multiplex• RTP Bundle

Audio Codecs• Opus• G.711

Video Codecs• VP8, VP9• H.264

Mandatory Encryption• DTLS

Media Negotiation Signaling• Session Description Protocol


WebRTC is 3 JavaScript APIs (& Java)

getUserMedia

CreatePeerConnection

Data Channel

Creates a media connection (RTP) between 2 peers

Grants access to local microphone, camera, or screen

Allows the user to send arbitrary data between peers

WebRTC API Examples

• webrtc.github.io/samples/src/content/getusermedia/resolution/

• webrtc.github.io/samples/src/content/peerconnection/pc1/

• webrtc.github.io/samples/src/content/datachannel/


http://webrtc.github.io/samples/src/content/getusermedia/resolution/

http://webrtc.github.io/samples/src/content/peerconnection/pc1/

http://webrtc.github.io/samples/src/content/datachannel/

Working with WebRTC


Browser Support


iswebrtcreadyyet.com

http://iswebrtcreadyyet.com/

Browser Support: WTF - No IE & Safari??


iswebrtcreadyyet.com

Fallback to a

plug-in or

(ugh) Flash

http://iswebrtcreadyyet.com/

http://upliketoast.blogspot.com/2011/01/search-results-vintage-women-boxing.html



Many ways to work with WebRTC

Level of effort

Low-level source

Frameworks, Tools, and

SDKsCloud API’s

Amount of control

Website Plug-ins & widgets


Building WebRTC from scratch – 2 options

Official Build by Google Ericsson Lab’s OpenWebRTC


Build with tools

Open Source

Commercial

Server-side libraries

JavaScript libraries

Mobile SDK’s

Cross-platform frameworks

Infrastructure elements


Consuming WebRTC API’s from the cloud

SignalingAuthentication

Push Notifications

SIP Gateway

Transcoding

STUN

TURN Servers

Low Latency

ElasticityRecording

Geographic redundancy

Bandwidth

Service monitoring

Multi-party

Identity


Plug-ins for your web admin

130+ tool vendors & major projects

https://webrtchacks.com/vendor-directory/12/10/2014 Scotch Streaming and Scaling 31

https://webrtchacks.com/vendor-directory/

Back-end Considerations

• Servers• Signaling server

• STUN/TURN

• Media servers/processing

• Gateways

• Network• Bandwidth

• Latency


The hard parts of WebRTC as you grow

Server-side• Defining signaling for your

signaling server

• TURN servers for NAT traversal

• Multi-party video conferencing

• Real-time transcoding

• Interworking with existing telephony networks (gateways)

Client-side• Mobile optimizations

• Non-WebRTC browser support

• Keeping up with browser updates


COMPANY CONFIDENTIAL: © COPYRIGHT 2015 DIALOGIC INC. ALL RIGHTS RESERVED. 34

TURN will relay your media

media

client

Signaling Server

TURN Server

STUN Server

client


Mesh does not Scale for Many Video Conferees

35

Full Mesh

Clients get overloadedEncode costs more than decode

Limited uplink bandwidth

Inconsistent performance across participants


Media servers for scalable multi-party conferencing

MCU SFU

Multipoint Control Unit Selective Forwarding Unit


Core WebRTC Gateway & Peripheral Functions

HTTP-to-SIP(H2S)

Media GW

Transcoder

Web SIP

SBCSTUN Server

TURN Server

API Gateway

Regulates API calls

Provides media anchoring for NAT Traversal

Tells a client what its IP address is

Converts non-standard web

signaling to SIP

Web SDK/ Widget

Mobile SDK/ Widget

Kit for adding Calls to

webpage

Kit for adding Calls to native applications

Converts one codec to another, trans-rates/sizing

Encryption & transport interworking

SIP interworking, media binding, and

security


Example: Large, Web-Centric Network Architecture

Web server

SIP/IMS Network

SIP

Media ServerFirewall

AP

I Man

ager

API Calls

H2S Server

Identity Server

App Server

App Server

WSS server

STUN/ TURN

Internal REST APIs

WebRTCBrowser

Chad’s Favorite WebRTC Resources

• webrtcHacks

• github.com/webrtc

• WebRTC Experiment

• html5rocks.com

• webrtcweekly.com


https://webrtchacks.com/

https://github.com/webrtc

https://www.webrtc-experiment.com/

http://www.html5rocks.com/

https://webrtcweekly.com/

My contact info

linkedin.com/in/chadwallacehart

[email protected]

@chadwallacehart

[email protected]

@webrtcHacks


http://linkedin.com/in/chadwallacehart

mailto:[email protected]

http://webrtchacks.com/

mailto:[email protected]

Date post:	16-Jul-2015
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