end of the world presentation

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WebRTC The End Of The World (As We Know It)

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Page 1: End of the world presentation

WebRTCThe End Of The World (As We Know It)

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WelcomeTo The Beginning Of The Post-Telephony Era

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I’m SteveSteve Sokol, Entrepreneur In Residence /

Director of Strategic Programsat Digium

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What is WebRTC?

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Photo Credits: Tom Keating - TMC.net, Eric Hernaez - Netsapiens

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How does it work?

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WebRTC leverages existing VoIPtechnologies

WebRTC exposes communicationsdevelopment to the 20M web developers inthe world

WebRTC sets rules for media, leavessignaling up to the application developer

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Get media streams from camera, mic

Create an “offer” session description

Send the offer to the far-end party

Receive an “answer” session descriptionfrom the far-end party

Discover a path that works by testing allpaths

Send media to the far-end party

WebRTC Call In A Nutshell

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Web Server

Web Browser Web Browser

SRTP Media Streams

Offe

r Sign

aling

(SD

P)O

ffer Signaling (SDP)

Answ

er S

ignali

ng (S

DP)

Answer Signaling (SDP)

HTTP

or W

ebSo

cket HTTP or W

ebSocket

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Web Server

Web Browser

SRTP Media Streams

Signa

ling

Signaling

Media ServerGateway

PBX

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New JavaScript APIs

Media Capture

Peer-To-Peer Networking

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Creating A Connection

Built-In NAT Traversal using ICE

STUN - Discover network details

TURN - Relay as last resort

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Encoding Media

Audio Codecs

Mandatory: Opus, G.711

Optional Codecs

Video Codecs

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Encrypting Media

Mandatory

Secure Realtime Protocol (SRTP)

SDES vs. DTLS-SRTP Key Brokering

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No mandatory protocol or mechanism

Can be done using SIP or Jingle usingJavaScript libraries

Can be done better using other methods:

WebSockets or XMLHttpRequest transport

Simple JSON signaling

Use a protocol that suits your use caseperfectly, not a protocol built to handle alluse cases adequately

What About Signaling?

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No mandatory signaling protocol is aGOOD THING™

Gives developers absolute control overthe user experience

Avoids the tendency to rebuild the PSTN

Avoids the “federation” issue

Allows for identity to be more than anumber

It’s The Web (Stupid?)

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URL-Based Calling

http://www.digium.com/contact/sales

http://www.digium.com/contact/ssokol

Directory-Based Calling

Facebook

Twitter

Linked-In

Corporate LDAP

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“Inside” users will use a web-based ormobile client

“Outside” users will use portal pages torequest access to various resources

People

Departments

Expert Support

Identity can be from email, Facebook,

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You will still need a communicationssystem or a communications service

You (eventually) may not need a “phonecompany”

Prediction: wired and wireless carrierswill become glorified ISPs within the decade

WebRTC will make rich communicationsa 100% “OTT” business

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So, Is It Ready To Use?

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Yes and no...

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Implementations in Chrome, Mozilla

Not currently interoperable

Great for “controlled environments”

Not yet ready for use by “normal” users

Will be ready by the end of 2013

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Challenges

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Mobile Deployments

Large-Scale Multi-Party

Legacy Integration

Codec Selection

Fragmentation (Microsoft’s CU-RTC-Web)

Encryption Keys

Audio Quality / Echo Cancellation

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Future features andenhancements...

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Peer-To-Peer Data

Real-Time Text (Captions)

Media Recording

Screen / Desktop / Tab Sharing

Statistics / Monitoring

Possibly low-level APIs

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A few use cases:

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Social Media

Call Center Agent Interface

Conferencing & Collaboration

Enhanced Customer Care

Distance Learning

In-Game Communications

Broadcasting

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Big Changes(Welcome To The Post-Telephony Era)

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Telephony has been holding backcommunications for the past decade.

SIP was hijacked: what started out as apeer-to-peer system was twisted into“PSTN-Over-IP”

Improvements and price reductions inbandwidth, mobile, web make a real changepossible

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Fully Unified Communications

Integration of communications directlyinto business and social applications

Communications as a feature orfunction rather than as a service

Customized User Experience

Excellent Privacy / Security

Significant Cost Reduction

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Asterisk And WebRTC

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Asterisk 11 added ICE, STUN, TURNsupport, WebSocket transport for SIPchannel and other tweaks

You can now create web endpoints usingAsterisk and a JavaScript SIP library

SIPML5

JS-SIP

Asterisk can bridge between WebRTCand legacy communications technologies

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Demo Time!

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Future versions of Asterisk will do more:

Recording and playback of audio andvideo

Interfaces for additional / customsignaling protocols

Interactive voice and video applications

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Steve [email protected]+1 (256) 428-6101

Thanks