MobileTea Boston presentation on getting started with WebRTC. Includes: *References on major WebRTC deployments *WebRTC use cases *What WebRTC is *Intro to the WebRTC API's *How to start developing with WebRTC *WebRTC scaling challenges *Chad's favorite WebRTC resources

1. Getting started with WebRTC
Chad Hart
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http://www.meetup.com/WebRTC-Boston/
Meetup #1 Links & Resources: goo.gl/UvbI27

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IP Telephony + WebRTC Media Server
PowerMedia™ XMS
Multi-party conferencing InterworkingTranscoding
Stream processing Person-to-machineRecording

5. Who uses WebRTC today?
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6. Who uses WebRTC today?
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7. Who uses WebRTC today?

8. Who uses WebRTC today?
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9. Who uses WebRTC today?
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10. Use case: Augmented Reality
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11. Use case: facial biometric verification
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12. Use case: controlling robots
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http://www.tate.org.uk/whats-on/tate-britain/special-event/after-dark
12

13. Use case: dog training
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https://webrtchacks.com/javascript-dog-trainer/

14. Use case: peer-to-peer data
CDNs
File Transfer
Private Networks
bitcoin
Peer web-servers
Internet of Things
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grimwire
wearables, security, industrial machine vision, ???

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WebRTC is on a lot of devices

16. WebRTC will have a lot of users
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WebRTC is world-class VoIP engine for free
OPUS
On2
Global IP
Solutions
.org
Video Codec
Google acquires On2 for $133
million
VoIP Media Engine
Google acquires Global IP
Solutions for $68.2M
Audio codec
Opus - open, royalty-free, highly
versatile standardized audio codec
Open Source Communities

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WebRTC is a Web and Internet Standard
webrtc rtcweb
http://www.w3.org/TR/webrtc/ https://tools.ietf.org/wg/rtcweb/

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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

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WebRTC is 3 JavaScript APIs (& Java)
getUserMedia
CreatePeer
Connection
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

21. 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/
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22. Working with WebRTC
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23. Browser Support
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iswebrtcreadyyet.com

24. Browser Support: WTF - No IE & Safari??
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iswebrtcreadyyet.com
Fallback
to a
plug-in
or
(ugh)
Flash

25. http://upliketoast.blogspot.com/2011/01/search-results-vintage-women-boxing.html
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Many ways to work with WebRTC
Level of effort
Low-level
source
Frameworks,
Tools, and
SDKs
Cloud API’s
Amount of control
Website Plug-
ins & widgets

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Building WebRTC from scratch – 2 options
Official Build by Google Ericsson Lab’s OpenWebRTC

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Build with tools
Open Source
Commercial
Server-side libraries
JavaScript libraries
Mobile SDK’s
Cross-platform frameworks
Infrastructure
elements

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Consuming WebRTC API’s from the cloud
Signaling
Authentication
Push Notifications
SIP Gateway
Transcoding
STUN
TURN Servers
Low Latency
Elasticity
Recording
Geographic
redundancy
Bandwidth
Service
monitoring
Multi-party
Identity

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Plug-ins for your web admin

31. 130+ tool vendors & major projects
https://webrtchacks.com/vendor-directory/
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32. Back-end Considerations
• Servers
• Signaling server
• STUN/TURN
• Media servers/processing
• Gateways
• Network
• Bandwidth
• Latency
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33. 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
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TURN will relay your media
media
client
Signaling
Server
TURN
Server
STUN
Server
client

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Mesh does not Scale for Many Video Conferees
35
Full Mesh
Clients get overloaded
Encode costs more than decode
Limited uplink bandwidth
Inconsistent performance
across participants

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Media servers for scalable multi-party conferencing
MCU SFU
Multipoint Control Unit Selective Forwarding Unit

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Core WebRTC Gateway & Peripheral Functions
HTTP-to-SIP
(H2S)
Media GW
Transcoder
Web SIP
SBC
STUN 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

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Example: Large, Web-Centric Network Architecture
Web
server
SIP/IMS
Network
SIP
Media
ServerFirewall
APIManager
API Calls
H2S
Server
Identity
Server
App
Server
App
Server
WSS
server
STUN/
TURN
Internal
REST APIs
WebRTC
Browser

39. Chad’s Favorite WebRTC Resources
• webrtcHacks
• github.com/webrtc
• WebRTC Experiment
• html5rocks.com
• webrtcweekly.com
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40. My contact info
linkedin.com/in/chadwallacehart
chad.hart@dialogic.com
@chadwallacehart
webrtcHacks.com
chadwhart@gmail.com
@webrtcHacks
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