webrtc tutorial by dean bubley of disruptive analysis & tim panton of westhawk ltd

Tutorial: WebRTC

Tim Panton, Westhawk (ex-Tropo)

Dean Bubley, Disruptive Analysis

Originally Delivered at ICIN, Venice, October 14th 2013

[email protected] @disruptivedean

[email protected] @steely_glint

mailto:[email protected]




Agenda for today

15.30 Introduction & background for WebRTC (DB)

15:50 WebRTC Technical Aspects & Standards (TP)

16:30 Q&A

16:45 Comfort Break

17:00 WebRTC Business issues & industry structure (DB)

17:20 WebRTC Integration for Telcos (TP)

18:00 Recommendations & action points (DB)

18:15 Q&A

18:30 Close & drinks

Copyright Disruptive Analysis Ltd 2013 October 2013

OVERVIEW OF WEBRTC:

CONCEPT & KEY BENEFITS (& SOME PROVOCATIONS)


About Disruptive Analysis

London-based analyst house & strategic consulting firm

Cross-silo, contrarian, visionary, independent

Advisor to telcos, vendors, regulators & investors

Covering VoIP since 1997 & 3G/4G mVoIP since 2007

Covering WebRTC since mid-2011

Published report on “Telco-OTT Strategies”, Feb 2012

Report & updates on WebRTC, from Feb 2013

Workshops on Future of Voice & TelcoOTT

Twitter @disruptivedean Blog: disruptivewireless.blogspot.com



For WebRTC report & quarterly

update details email

[email protected]

Remember these?


Design & software simpler via the Web


Benefits of WebRTC

Democratises voice & video in websites & apps

Add context to communications & vice-versa

Cheap / easy / open-source components

Advocacy from Google, major vendors, telcos, IETF,

W3C etc

Enterprises & telcos can extend comms over the Internet

Real momentum & enthusiasm

No predefined signalling

Growing ecosystem even pre-standardisation

Realtime data even more disruptive


WebRTC = disruptive service innovation


Source: Disruptive Analysis WebRTC Strategy Report, Feb 2013 & Q2 Update August 2013

Definitions & methodology in report - See disruptivewireless.blogspot.com for details

Device base supporting WebRTC growing Zero4bn in 4 years

Mil

lio

n

Voice ≠ Telephony


• Now: 2G & 3G • Future: Smartphones & LTE

Voice

Telephony

Voice

Telephony

Video, context, sense Video

Gaming, CEBP,

surveillance, social

voice, TV voice etc

Voicemail

Conferencing

PTT

Comms moving “in-context”

Service

e.g. SMS, Telephony Product

e.g. Viber, Uberconference

Feature

e.g. In-game chat

Function

Voice/video moving from service to function

Fragmenting voice/video comms models

Standalone calls

Non-call comms

Embedded app/web

calls

Copyright Disruptive Analysis Ltd 2013 Oct 2013

Circuit IP

“Hegemony

of the

caller”

Better than

“being there”

Telephony aspires to be

“like being there”

“There”–

same place

and time

The role of video in communications

Video really needs to have a clear “purpose”

We will not default to “video everywhere”

Will be even more contextualised than voice & messaging

Different issues of ergonomics, social norms, behaviour

Very little desire for interruptive straight-to-video calls

Numerous niches for B2B, B2C, C2C

Skype calls between distant relatives / expats / diaspora

Customer service but need for extensive retraining of staff

Personal consulting eg doctors, therapists, trainers, interviewers

Need for interoperability unclear as often in-

context/in=app


Comforting myths


QoS is critical

Interoperability is essential

Minutes / messages = value

Uncomfortable reality


QoS is sometimes critical

Interoperability is essential

for lowest-common

denominator services only

Intention & outcomes = value

Intent & purpose....


Why do people make

phone calls, anyway?

WEBRTC STANDARDS

AND UNDERPINNINGS

Tim Panton, Westhawk Ltd - @steely_glint

What is webRTC?

Haven’t used webRTC

Your laptop is on wifi

Have Chrome/firefox

Browse to:

http://phono.com – click ‘call’

http://phono.com/

What did we just do?

Placed a video call with no plugins

zero config

just by browsing to a site

encrypted

free

over local wifi

WebRTC Definition

Realtime

voice / video /data

browser based

no plugins

secure

interoperable

Standards

IETF

W3C

Loosely based on pre-existing standards

$100s Millions of IPR - donated

Google, Cisco, Mozilla, Skype, Tropo, ATT,

E///, Lucent etc.

HTTP(s) signaling via webserver

Peer to Peer media between Browsers

Big Picture

The IETF has responsibility for the wire

protocols in RTCWeb

Protocol standards (IETF)

Signaling Standards

None! It is up to the javascript in the

browser to do what ever is needed.

Media Standards (RIA 2.0)

Many! STUN

ICE

TURN

DTLS

SRTP

RTCP

OPUS

ULAW

Why so many?

The network environment of a web browser is not the same as a desk phone.

Security – hostile lans (coffee shops/hotels)

Variablity – home networks, wifi/3g

Programability – Javascript is dynamically loadable

This stack of media standards addresses the differences.

ICE sends multiple STUN packets down all possible interfaces to try and

find a path. First Bi-directional route found is used.

STUN and ICE address NAT

NAT NAT

A TURN server in the cloud acts as a packet reflector

Many Telco 3g networks isolate users from each other.

TURN is for when ICE fails to find a viable path

NAT NAT

TURN

If additional services are required – recording, conferences, PSTN interop

etc. then bridging via a media server may be required.

Bridging via a media server.

NAT NAT

MS

DTLS

DTLS is the UDP version of TLS (as used in

https etc)

It serves 2 purposes in the WebRTC

Exchange keys used by the SRTP media

encryption

Carry the data channel streams

Demo – sharefest.me

Data channel provides Peer 2 Peer data between browsers.

It can be used for file transfer, game moves etc.

It may also become important in M2M or IOT as a secure NAT friendly P2P protocol.

Demo show file transfer between 2 browsers by sharing a URL, but not through that server.

SRTP + RTCP

Encrypted version of the classic RTP protocol,

with the RTCP reporting mechanism.

Multiple media streams may be multiplexed over

the same pair of ports – still under discussion.

Standard Codecs

Audio OPUS

Wideband

Flexible, efficient, loss correcting

Expensive to transcode

ULAW

Narrowband PSTN codec High bandwidth

Poor in lossy/variable networks

Codecs - No video standard yet.

Video VP8

H264

Differences are largely commercial

and legal rather than technical, either

is plenty good enough.

These are the API’s offered to the javascript

programmer – aka ‘JSEP’

Standard APIs (W3c)

Javascript : getUserMedia()

navigator.webkitGetUserMedia({

'audio':true,

'video':true

},

function(stream) {

var url =webkitURL.createObjectURL(stream);

createPeer(stream);

},

function(error) {

});

Javascript : RTCPeerConnection

pc = new RTCPeerConnection(configuration,constraints);

pc.onicecandidate = function(evt) {

sendCandyToAlice(evt.candidate);

};

pc.onaddstream = function (event) {

var remotePlay = document.getElementById(”videoTag");

remotePlay.src = webkitURL.createObjectURL(event.stream);

};

pc.addStream(localStream);

pc.createOffer(

function(offer) {

pc.setLocalDescription(offer);

sendOfferToAlice(pc.localDescription.sdp);

},

null, constraints);

HTML : <Video> tag

Extended to accept a webRTC stream as a

source

Either a local or remote

See previous slides

Additional mute etc.

Javascript:RTCSessionDescription

function onMessageFromAlice(message){

var sd = new RTCSessionDescription(

{'sdp':message,'type':"answer”} );

pc.setRemoteDescription(sd,sessionOk,sessionFail);

}

Dissention with SDP as an API

SDP is emitted by pc.createOffer()

Complex SDP

May be manipulated to select

codecs/candidates…

Source of much complaint

Currently ill defined

May be replaced in 2.0 std

Other co-operating W3C APIs

WebGL

Can be used to apply effects to video

WebAudio

Can be used to apply effects to audio

Both will be able to be applied to local or remote

streams

Javascript libraries

Most web coders will use a library

JS Sip

ATT.js

Phono

OpenRTC

Demo – WebGL effects

Demo of WebGL doing realtime effects on a

webRTC video stream

WEBRTC BUSINESS ISSUES &

INDUSTRY STRUCTURE


WebRTC key use cases


Browser-to-browser

(or web-app)

comms

Browser-to-Telco

VoIP / IMS

IMS

Browser-to-Telco

CS / PSTN

Browser-to-

webserver

Web

Browser-to-contact

centre

Browser/app

conferencing

Plus: M2M, gaming, TV-based, data-centric & various others

Browser-to-UC

or IP-PBX

Verticals

Healthcare

App/web-embedded RTC not new concept


• Expensive

• Inflexible

• Poor developer support

• Limited use-cases

• Reliance on “call” model

• Poor audio/video

• Hard to integrate

= Patchy adoption & little

developer enthusiasm / buzz

WebRTC has created the buzz & excitement

Democratised the idea of realtime voice/video/data

Easy elevator pitch

“It’s like Skype – but in the browser, with no plug-ins!”

“Three simple Javascript APIs – millions can use it!”

“It’s being evangelised by Google & all these other

guys!”

Mix of (fast) standardisation & “pragmatic proprietary”

Encourages experimentation with a very low bar

Usable at multiple levels of abstraction / effort /

commitment

Carries a sense of inevitability & scope for innovation


... although not quite as “easy as it looks” yet


Signalling

“WebRTC isn’t a standard. It’s a movement”

Irony: problems making WebRTC stronger not weaker

“Hardcore” comms developers going “down to the metal”

Building around core RTCWeb protocols & media engine

Embedding WebRTC elements into desktop applications

Longer-tail developers being addressed by API/cloud

players

“Packaged” WebRTC capabilities like multiparty video

Abstraction to avoid risk from changing standards

APIs for iOS, Android apps

Next tier up of service platforms emerging too

Renewed interest in “realtime everywhere”

No “religion” about WebRTC “purity” – just get on with it!


(Quote from Tsahi Levent-Levi @tsahil)

WebRTC made easier via 3rd-party APIs


CU-RTC-Web?

3-way support for WebRTC now expanding

WebRTC

Enterprise

Telco Consumer Web


Devices Network

tools

TV &

gaming

WebRTC rapidly expanding past “calls”


View Oct’13: lead WebRTC use-cases


Early enterprise adoption

• “Call me” buttons

• Contact centre

• First telehealth apps

Initial consumer web apps/devices

• Remote 1-1 education/training/sales

• Free standalone video-calling

• Chromecast

• Developer SDKs & APIs

Live &

commercial

• Vertical niche solutions (finance, health)

• Corporate conferencing

• Full enterprise UC

Pilots / pre-

commercial

Trials &

demos

• Telco core/IMS extension

• Entertainment & consumer electronics

• M2M, CDN & data-centric apps

What use-cases lead?


Existing

web

services

Adding real-time

comms capabilities

Existing

realtime

comms

services

Extending via the web,

blending web

capabilities

?

Provisional WebRTC timeline (at Q1’2013)

2011 2012 2013 2014 2015 2016


Chrome &

Opera

browsers start

supporting

WebRTC APIs

Chrome

supports

WebRTC in

stable channel

History Prediction

Firefox supports

WebRTC in stable

channel by default

1 billion WebRTC

capable devices 2 billion WebRTC

capable devices

3 billion WebRTC

capable devices

IETF working

group set up

Google, Cisco,

Skype, Mozilla

RTC-Web

workshop

Microsoft IE

supports WebRTC

or CU-RTC-Web

W3C final

draft

submitted

Google open-

sources GIPS IPR

Ericsson

WebRTC

demo at MWC

AT&T

announces

alpha WebRTC

APIs

Telefonica

acquires

TokBox

First operator-

branded WebRTC

/IMS apps emerge Native-WebRTC

smartphones gain

traction 1 billion individual

active WebRTC

users 1st WebRTC-primary

social/calling app goes

viral

Broad adoption of

WebRTC in massmarket

contact centres

User familiarity

with in-app

voice/video

User familiarity

with in-website

voice/video

Source: Disruptive Analysis WebRTC Strategy Report, Feb 2013

Assumptions - See disruptive-analysis.com for details

Various “flavours” of WebRTC gateway


IMS Internet WebRTC

Gateway

PSTN Internet WebRTC

Gateway

IP-PBX Internet WebRTC

Gateway

M2M Internet WebRTC

Gateway

-Signalling, eg SIP-over-WebSocket

- Voice/video/data media over SRTP

- STUN/ICE/TURN setup for firewalls

Varying functions & scale for gateways,

eg WebSockets, ICE, SIP/XMPP etc, API

exposure, transcoding, security etc

WebRTC gateways galore

WEBRTC INTEGRATION

OPTIONS

Tim Panton, Westhawk Ltd – @steely_glint

Demo – call a mobile

Demo calls my mobile from a browser

Not everyone wants to interop

Games

Dating sites

Whiteboards

OTT

Mayday

For these sites a home grown signaling protocol may be simplest/best.

(highest value apps will be in this class)

Problem statement

HTTP(S) transport

RIA 2.0 media

Encrypted

Opus

VP8 ?

SIP

RTP

Cleartext

Ulaw (729, amr?)

H263/4

WebRTC Carrier IMS

Unless and until those incompatibilities go

away.

Gateway needed ?

HTTP to SIP – SIP in the Browser

Use javascript to build SIP messages and

protocol

Wrap in HTTP (or Web-sockets)

Send to webserver

Webserver unwraps and forwards to IMS

SIP in the browser

Browser

JS SIP IMS

WebSocket

Server SIP in HTTP

UDP

SIP

Problems

You still have a gateway – albeit a thin one.

You have javascript injecting SIP messages

into IMS

The SDP isn’t compatible

The media isn’t compatible

What is Early media in a browser ?

You have your SIP credentials out on the

internet.

SBC

SIP in the browser with SBC

Browser

JS SIP IMS

WebSocket

Server SIP in HTTP

UDP

SIP UDP

SIP

DMZ

Remaining Problems

The media isn’t compatible

What is Early media in a browser ?

You have SIP credentials out on the internet.

SBC

SIP in the browser with SBC, Media

Gateway and Registration proxy

Browser

JS SIP IMS

WebSocket

Server SIP in HTTP

UDP

SIP UDP

SIP

DMZ

Media

GW RIA 2.0 RTP ulaw

Proxy

Reg

REST in the browser

Use web ‘RESTful’ commands

Sent from the browser

To a webRTC gateway

Gateway generates the SIP IMS needs

Gateway controls transcode resource

REST in the browser with

Gateway

Browser

app IMS

WebRTC

gateway REST/HTTP UDP

SIP

DMZ

Problems

Need to map from web Identity to SIP

Select a web identity provider

webRTC gateways don’t scale (yet)

No standard for REST messages

Home rolled protocol (may have holes)

XMPP in the browser with SBC

Browser

app IMS

WebRTC

gateway

XMPP/BOSH/

HTTP UDP

SIP

DMZ

Problems

Need to map from web Identity to SIP

Select a web identity provider

webRTC gateways don’t scale (yet)

More complex than necessary

Needless protocol mapping?

However

BOSH is tested

XMPP well defined and federates

WebRTC isn’t mobile first yet.

Did we forget mobile?

WebRTC on Mobile

Browser isn’t a natural interface

WebRTC codecs are heavy on battery

No native App friendly API (yet)

SIP (if used) not an efficient mobile protocol

Audio hardware on android variable

Both Chrome and firefox on Android support webRTC

Expect to see RIA 2.0 with native APIs

DTLS can carry a certificate – but which

one should be used?

Identity.

Multiple identities on the web

When I call from a webpage, which identity do I want to present?

E164 to the shop

Facebook Id to my fb friends

Anon to the game

Pseudo id to dating site

Do I ever want to present facebook ID to G+ users?

Demo

Phono using a Jira identity. (perhaps)

RECOMMENDATIONS & NEXT

STEPS FOR TELCO WEBRTC


Traditional telecom services: ugly outlook


Source: Model for STL Partners, developed

by Disruptive Analysis

Focus on broadband, bundling,

M2M, digital services & lower costs

/ better flexibility

For telcos, it’s all looking pretty grim anyway Downsides

Voice & SMS saturation & cannibalisation

Regulation & competitive impacts

Weak content & VAS propositions

Economic pressures

Ecosystem competition

Upsides

Connecting the last unconnected

Smartphones & data growth

Better segmentation & pricing

Innovative services & enablers

Embracing & exploiting fragmentation


Neuroscience explains reluctance to change

Predictable irrationality

Endowment effect

Optimism bias

Confirmation bias

Defence of belief systems


Network/service coupling: historical accident


• Service = network

• Only 1 service

• Interop essential

What is “OTT”?

Any capability offered “over the top” of Public Internet

Can be service, application, feature or function,

decoupled from the underlying access network

Calling OTT a “threat” misses its inevitability

Overlooks 150+ examples of “Telco OTT” services


Buys time if done well

But perceived value will fall

Accounting questions

Needs regulatory OK

Starts unwinnable arms race

Admission of being “dumb”

Add value to bundles

May be revshare upside

QoS not monetisable

Needs developer skills

Likely conflict internally

Few telcos will succeed

Carrier strategies with/against OTT

Telco-OTT

Bundle/ enhance

Block / degrade / Charge

Partner


Also: Exit & allow customers to BYOVoice

For telcos WebRTC is a magnifier/catalyst


Now

With WebRTC

Bigger opportunities

Worse threats

Faster speed

What % of future value comes from 3GPP?


The Theory The Reality

Core ntwk

& legacy

voice

IMS,

eventuall

y

Corporate

UC

Cisco /

MS Lync

Video

conf

Bluejean

s / Vidyo

etc

Developer

APIs

Twillio /

Voxeo

• Various telco business units now disintermediating their own core network / platform

• Increasingly partnering with 3rd-party players for voice/video apps

• What % of future equipment/server need will reside in IMS/3GPP domain vs. 3rd party

equipment or cloud platforms?

VoLTE &

RCS

IMS as a

platform

Corporate

UC

Video

conf

Developers

Network

APIs

Consumer Business OTTs

Main WebRTC strategies for SPs?

Extend on-net services & IMS / SS7

Turbocharge Telco-OTT

apps

Sell packaged WebRTC

services to subscriber

Enhance developer platform

Strengthen enterprise &

verticals


Also: invest /

incubate

Improve own CRM

& systems

Maybe M2M,

devices etc

Conclusions

WebRTC is emerging very fast

Standards are still settling

Important to be “a part of the community”

Early experimentation is mandatory

Don’t confine WebRTC to IMS & Labs

Probably no more than 30% total WebRTC effort / resources

should go on IMS integration

Every unit in the telco with a website should be using it

Think about second-order problems now

Speed & design & purpose >> quality & evaluation



For WebRTC report & quarterly

update details email

[email protected]

www.disruptive-analysis.com

disruptivewireless.blogspot.com

@disruptivedean

[email protected]

Skype:disruptiveanalysis


www.westhawk.co.uk

@steely_glint

[email protected]

http://www.disruptive-analysis.com/






webrtc tutorial by dean bubley of disruptive analysis & tim panton of westhawk ltd

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