WebRTCThe End Of The World (As We Know It)

WelcomeTo The Beginning Of The Post-Telephony Era

I’m SteveSteve Sokol, Entrepreneur In Residence /

Director of Strategic Programsat Digium

What is WebRTC?

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

How does it work?

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

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WebRTC Call In A Nutshell

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

Web Server

Web Browser

SRTP Media Streams

Signa

ling

Signaling

Media ServerGateway

PBX

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

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What About Signaling?

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?

Yes and no...

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

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...

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:

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)

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

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!

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 Sokolssokol@digium.com+1 (256) 428-6101

Thanks