The document provides an agenda for a WebRTC workshop covering the following key points: - The workshop will provide a deep technical and business overview of WebRTC through presentations and demonstrations. - Attendees will learn about the current status of WebRTC standardization and implementations, and what capabilities may emerge over the next 1-2 years. - The workshop includes sessions on technology details like APIs, media protocols, and interoperability, as well as implications for service providers, enterprises, and use cases. - An afternoon demo session will provide hands-on experiences of WebRTC applications from various companies and allow networking among attendees.

1. WebRTC Workshop
The HTML5 Real-Time Web
April 22nd 2013
Pre-conference Workshop for the IMS World Forum
Alan Quayle
alan@alanquayle.com
www.alanquayle.com/blog
Jose de Castro
jdecastro@voxeolabs.com
www.voxeolabs.com
4/21/2013 © 2013 Alan Quayle Business and Service Development 1

2. Objectives
• Bring together deep technical and deep business thought leadership on
WebRTC with Jose de Castro, Alan Quayle, and many of the audience to
providing attendees with a unique independent workshop.
• Provide a deep-dive quantified analysis of the WebRTC status, enabling
attendees to understand what is likely to emerge over the next 18
months to 2 years, in this complex rapidly emerging ecosystem and
what it will mean to their business.
• Provide attendees with a series of WebRTC demonstrations, to share
their experiences on implementing WebRTC, and provide ample
networking opportunities at the end of the workshop to discuss and
consolidate what has been learned through the day.
4/21/2013 © 2013 Alan Quayle Business and Service Development 2

3. Structure (1 of 6)
• Registration
• 09:30 - Introduction to WebRTC and Initial Market Review
o What is it and what it is not,
o Cutting through the mis-information and hype
o Non-technical introduction
o Web browser implementation status
o Taxonomy of suppliers / service providers
o Codecs and devices - is certification necessary?
o What is Google's aim?
• 10:30 Standardization deep dive
o Standardization process
o Current status
o Battles and likely outcomes
o IETF and RTCWEB documents

4. Structure (2 of 6)
• 11:30Technology deep dive
o Peer connect API
o Setting up local media and media flow
o Protocols
o WebRTC triangle / trapezoid
o SIP, Jingle and the PSTN.
• 13:00-14:00 Lunch
• 14:00 What WebRTC means to Service Providers and IMS:
o Extending enhanced communications services to web browsers
o Impact on OTT (Over The Top) and existing voice, messaging, video and VAS
o Impact of device compliance
o Customer experiences and behaviors
o Revenue, churn and relevance impacts
• 14:30 What WebRTC means to enterprises:
o Impact on Unified Communication and the Contact Center
o Impact on company's website
o Security and operational issues
o Potential cost savings and innovations

5. Structure (3 of 6) DEMO TIME 15:00-17:00+
• Demo Time will be divided into 2 sessions, its aim is to be informal
and provide ample networking opportunities for attendees to
consolidate their learning from the workshop:
• Demo presentation to the group: each demo will be 5 minutes long,
and 5 minutes for questions; and
• Demo one-on-one: attendees can chat one-on-one with the demo
presenters, notionally 30 minutes but can run on into discussions at
the bar through the evening.

6. Structure (4 of 6) DEMO TIME 15:00-17:00
• Zingaya ('Call' button for websites)
o Embed a 'Call' button into the website. Visitors can click that button and the call is
forwarded to the website operator's preferred land-line or mobile phone. All that is
required is a website; all the visitors need is a browser and microphone.
• Voxeo Labs (Ameche (new IMS/Web services), Tropo (leading call control API),
Phono (Web comms innovation)). They will demo Phono’s three types of
identity:
o Anonymous Identity: user lands on web site and is able to call directly into the contact
center
o Web Identity: use your web identity (twitter, foursquare, etc) to call each other.
o Telco Identity: Phono sessions can attach to the telco network and assume the real
identity (phone number) of the subscriber, allowing calls to be routed to both the mobile
and the browser simultaneously.
• Telestax
o Provides a complete stack from the client-side with Javascript JAIN SIP JS and WebRTC
as well as the server side with our SIP Over WebSockets. The demo will be a WebRTC
video conferencing and IM.

7. Structure (5 of 6) DEMO TIME 15:00-17:00
• Solaiemes WebRTC to Rich Communication Suite demo
o Demonstration of RCS messaging and WebRTC to access to media
components of devices to revamp the value of PSTN (and also mobile) lines.
Shows how Unified Communications could be built just a mash-up of
standards and APIs.
• Quobis
o Their approach to WebRTC is based on QoffeeSIP, a complete open source
Javascript SIP stack that can be used in a website to exploit all the multimedia
capabilities of WebRTC technology. Thanks to QoffeeSIP they have developed
a corporate WebRTC webphone that can interop with different network
devices; this webphone is going to be released at IMS World Forum event.
• Huawei leading NEP
o WebRTC / RCS insurance application demo

8. Structure (6 of 6) DEMO TIME 15:00-17:00
• Drum by NetDev (conference calls and online meetings)
o Allows providers of fixed, mobile and next generation VoIP services to deliver audio
conferencing as a direct, branded service. Hosted within your IP network on your
servers, Drum audio conferencing is a standalone software solution with an integrated
media server.
• Bistri (Social Video)
o Video chat with fun video effects, take screenshots of calls, share them with friends or
social networks. Bistri runs in the browser, so there's no need to install additional
software or plugins.
• apidaze.io
o Is a cloud communications API for developers with tools for building web or mobile
communication services, with a special focus on WebRTC. The demo will show how a
web developer can easily use the regular WebRTC API to place calls to external numbers
and audio conference rooms accessible from the PSTN too, using a simple raw
WebSocket connection that carries JSON text.

9. Introduction to WebRTC and
Initial Market Review

10. What The Geeks Say
Open, Nothing Proprietary
No Plugs-Ins
Multi Platform / Device

11. Real-time stuff for your
browser with no plug-ins

14. M2M and Telematics

15. Surveillance
& Monitoring

16. Lots &
Lots &
Lots of
Devices

17. Embedding Communications
Everywhere!

21. Codec Wars
Opus,
VP8
G.711,
AMR-
WB,
EVS,
H.264

24. Browser GetUserMedia PeerConnection DataChannel
Chrome Yes Yes Q2 ‘13
Chrome for mobile Yes (March ‘13) Yes (March ‘13) Q2 ‘13
Firefox (desktop) Yes Yes Yes (first one)
Firefox (mobile) Yes Yes Yes (first one)
Opera Yes H2 ‘13 2014
Opera Mini H2 ‘13 2014 2014
IE (desktop) Chrome Frame /
2014
Chrome Frame /
2014
Chrome Frame /
2014
IE (mobile) 2014/2015 2014/2015 2014/2015
Safari (desktop) 2014/2015 2014/2015 2014/2015
Safari (mobile) 2014/2015 2014/2015 2014/2015
WebRTC is NOT Everywhere

25. Lies, Damned Lies, and Statistics

26. Regardless IE Matters

27. Mobile is Even More Complex
Native browser Natively in OS 2nd browser 3rd party SDK

28. Business
Technology

29. Business
Technology
Latency
Efficiency
Resilience
Performance
Implementation Complexity
Supporting Devices
Ecosystem Support
Customer Needs
Interoperability
Use Cases
IPR

30. 30
“Given the ability to deliver a royalty-free
platform with no compromises on quality, we
see no reason to include mandatory royalty-
bearing codecs.”
“H.264 support is a requirement in some regulatory
frameworks, such as emergency services. AMR narrow-band
is playing a key role in mobile telephony and has a huge
footprint.”
“G.711 is universal, unencumbered, and widely implemented.
A mandate for Opus will limit initial RTCWeb clients to use
software-based codecs”
“We would like to recommend AMR-WB and
EVS, since we expect them to be available in
mobile chipsets.”

31. Optional Codec Diversity will Reign

32. Codec Wars
• G711a/u (RFC 3551): supported by all the devices. Needs to use a lot of
bandwidth.
• DTMF tones (RFC 4733, updates RFC 2833): needed for interactions with
several systems (for instance IVRs).
• Opus (RFC 6716): bitrate variable, low latency and high quality for human voice
and music. Specially designed for real time communications.
• In order to interact with VoIP systems, in several scenarios, it will need
transcoding or interworking of DTMFs (RFC 4733-> INFO, RFC 4733-> in-band,
etc).
• Can’t we just have both G.711 AND Opus? YES!!!!
Regardless Transcoding Will be Needed

33. And With Transcoding Comes
33
Delay
Third Parties
Packet Loss
Quality Loss
Cost

34. H.264
H.265
VP8
VP9
Can’t we just have both H.264/5 AND VP8/9?
Video Battle is Getting Nowhere

35. There’s No Approval Process

36. In The Limit Which Browser Gives you the Best
Experience?

37. Device base supporting WebRTC
Copyright Disruptive Analysis Ltd
2013
Feb 2013
Million
Source: Disruptive Analysis WebRTC Strategy Report, Feb 2013
Definitions & methodology in report - See disruptivewireless.blogspot.com for details
0
500
1000
1500
2000
2500
3000
3500
4000
Tablets
Smartphones
PCs

38. The WebRTC Train has left the station
and it isn’t going to wait for Telecom

39. WebRTC is a car without wheels!

40. WebRTC Triangle
• Both browsers running the same web application from web server
• Peer Connection media session is established between them
• Signaling is not standardized, could be SIP, Jingle, proprietary.
Uses HTTP or WebSockets for transport
Web Server
(Application)
Browser M
(Running HTML5 Application
from Web Server)
Browser L
(Running HTML5 Application
from Web Server)
Peer Connection (Audio, Video, and/or Data)
40Intro to WebRTC February 2013
The wheels!

41. The Beauty and Value of WebRTC is
when we mash it up with other stuff

42. Keep Calm and
Do SOMETHING
(Just NOT THROUGH
THE GSMA or TMF)
HMS Government Advisory

43. Standardization Deep Dive
(c) 2012 Alan Quayle Business and Service Development

44. Technology Deep Dive
(c) 2012 Alan Quayle Business and Service Development

45. What WebRTC Means to Service
Providers
(c) 2012 Alan Quayle Business and Service Development

46. NetHead CustHead
Our Industry’s Multiple Identity Disorder

47. A Telcos Two Unique Assets

48. Impact of WebRTC?
• Voice becomes just like all your other communications: organized into your
preferred social or office tools.
• It will be important for the IMS/RCS world to inter-operate with the
WebRTC world, currently these browsers will be a closed book to IMS. For
RCS to become pervasive, it cannot remain trapped in phones that have
implemented the IMS/RCS client.
• For all the OTT (Over The Top) applications, they can now use their
"directory service" i.e. your list of contacts also using their service to enable
Viber / Skype / Whatsapp everywhere. On your PC, smartphone, tablet, TV;
and they can offer chargeable services without Apple taking 30%.
• As long as you're data connected, communications is in the cloud, people
need only break out to PSTN when the other person is not data connected,
or the call quality is too low due to their internet connection. PSTN
becomes the communications path of last resort!

49. Impact of WebRTC?
• The company's website now becomes its call center front end. A weblog
becomes your personal communications assistant.
o Lots of start-ups in this space
• Communication service aggregators save customers running multiple clients
on their phone, that would run in the cloud and be controlled from the
browser.
• Click to call doesn't require an operator's voice network, just access to the
internet.
• Communications becomes like using any application on a smartphone,
users can add features, capabilities, people throughout a call, e.g. N-way
calling finally becomes simple and obvious with a simple point and swipe.
• Directory services become critical sources of value in connecting all the
different IDs: telephone numbers, SIP IDs (IDentifier), web session IDs,
other OTT IDs, etc.

50. Impact of WebRTC?
• VAS (Value Added Services) leaves telco. Any web developer can create value and
solve problems for customers, it the customer who will decide, and those developers
who fail fastest win the innovation race.
• Advertising finally enters the communications space, opening up business model
innovation.
• New CRM (Customer Relationship Management) methods: click from email, from
webpage, from app, from TV. The ability to communicate becomes embedded in
most transactions.
• QoS (Quality of Service) remains an issue, but for the people using Vonage and Skype
over the years will attest, QoS is rarely an issue.
• Your phone number is no longer relevant anymore. It's a gateway to the past.
Customers will only know the PSTN is involved because of the poor audio quality
(G.711) – BUT it is the customers only unique ID that they own.
• Gaming becomes interesting as all the devices become controllers using gesture
controls as well as the more traditional methods for network-based games.

51. Other Telco Impacts
• Impact of WebRTC on IP Messaging
o No need to for a messaging client to be downloaded
o Unified communications across voice, video and data
o Integrated experience across the web and communications client
• Opportunities
o Integrated charging
o Bundle APIs: WebRTC, RCS API, payment API, Call Control APIs
o QoS for those willing to pay (those running their business on your network)
• QoS API? NO! keep that for YOUR SERVICES
o Extend network services over the web
o Enhance enterprise Unified Communication offers
o Enhance OTT? Why they’re competitors!
o Gateway for the WebRTC codec mess
o Web phone for existing customers

52. Key Points
• Voice traffic is going to be through the web
• Browsers are the new endpoints
• A website of a company can be the call center
• An individual’s website (Facebook Page) their communications
assistant
• Security, identity and privacy are very important
o Telephone number is not important unless operators pull their finger out!
• New business opportunities abound

53. DO Communications
better thaN the
competition
ELSE BE AN ISP
HMS Government Advisory

54. What WebRTC Means to Enterprises
(c) 2012 Alan Quayle Business and Service Development

55. WebRTC and the Enterprise
• Enterprise is both excited and confused
about WebRTC
o Lower communication costs
o Lower IT costs as fewer clients to maintain?
o Better home and mobile comms that are
integrated with corporate systems
o New customer communications options
• Confusion arises from
o How to integrate
o What needs to change
o How does it work with legacy devices like
desktop videophones, mobile clients, desktop
phones, Microsoft Lync, Cisco UC, etc.
o What about our corporate firewall, what about
our SBC?

56. What Enterprises are Saying in Interviews on WebRTC
“WebRTC can lower our communications costs by 30%”
“WebRTC will finally give us a workable video communication solution
across all employees”
“WebRTC can make our contact center integrated into all channels, e.g.
Web and Mobile, improving customer service.”
“WebRTC can improve collaboration both internally and with our
partners as it just works between browsers.”
“WebRTC appears to be not part of the Telco’s offer, it is OTT.”
“WebRTC is confusing, it doesn’t work in most browsers.”
“WebRTC looks like a lot of hype, will it go the same was as VoIP?”
“WebRTC’s security, private, and ability to get hacked remain unclear”
“WebRTC will be killed by Microsoft!”

57. IT’s Technical Concerns
• Open standards makes it easier to hack
• Clientless, plugin-less browser audio and video for real-
time communications means we’re not dependent on
browser software suppliers
• Open-source codec
• ICE/STUN NAT transversal does not work all the time
• SRTP (Secure Real-time Transport Protocol) –
configuration and SSL certification issues
• No rules on signaling protocol – security issues?
• Multiplex RTP and RTCP on single port – management
issues?
• Multiplex audio and video on a single port – management
issues?

58. Understanding Old-IT
• Hardware endpoints with hard to upgrade software
• Few soft clients tied to the PBX
• Limited codec support, generally H.26x for video
• No experience with ICE, STUN, and STRP
• RTCP shall be on the RTP port +1
• Audio and Video are managed separately
• Look how long BYOD is taking
• Skype blocked, Facebook blocked,
• WebRTC blocked!!!!!!

59. Typical Large Enterprise Deployment
SBC
Router
Desktop Network
Telepresence Network

60. IT Generation Gap
Multiplexed
Media
STUN / ICE
VP8
RTP/RTCP
Separated
NO STUN /
ICE!!!!
H.26x

61. The Solution:
A Gateway
• Signaling Gateway
o SIP stack in JavaScript?
o Break to SIP at the Gateway
• De-ICE
o Validate SDP ICE candidates
• De-Multiplex Media
• STUN Binding Handling
o Response to STUN bindings on
RTP channels
o Validate
• Transcoding

62. Use Cases
• Social Media and CRM Integration
• Video Conferencing to any device
• Inbound Click to call a New channel
• Can be deploy by outsourcers
• Calls in internal directories

63. In Summary
• Enterprise is interested in WebRTC BUT…
• Some Enterprises change even slower than Telcos….
• If Telco’s don’t help them, they will go to other service providers
• Gateway as a Service
o Signaling Conversion
o Media Interworking

64. Keep Calm and
Do SOMETHING
(Just NOT THROUGH
THE GSMA or TMF)
HMS Government Advisory

65. http://webrtcbook.com
65Intro to WebRTC February 2013

66. What WebRTC Means to Enterprises
(c) 2012 Alan Quayle Business and Service Development
DEMO
TIME