TBE012 - WebRTC Technology in OpenTouch R2.2.1 - Ed03a

WebRTC Technology in OpenTouch R2.2.
1
Presales Presentation
November 2016
Pre-Sales Presentation | November 2016
WebRTC in OT R2.2.1 ed03a
With slide notes
1
COPYRIGHT 2016 ALE International. ALL RIGHTS RESERVED.
History
Edition 01: Creation
Edition 02: Update for OpenTouch release 2.2
Edition 03: Update for OpenTouch release 2.2.1 OTES Removal
Disclaimer
This documentation is provided for reference purposes only and does not fully describe the capabilities of each Product and related features. Therefore, ALE International
declines any liability for inaccuracies contained herein. For an exhaustive view on features list and product limits for the current product release please see the required
Feature List/Product Limits document available through the ALE eBusiness Portal web site.
In the interest of continued product development, ALE International reserves the right to make improvements or other changes to this document and the products it
describes at any time without prior notice.
Copyright
Copyright ALE International 2016. Distribution of substantively modified versions of this document is prohibited without the explicit permission of the copyright holder.
Distribution of the work or derivative of the work in any standard (paper) book form for commercial purposes is prohibited unless prior permission is obtained from the
copyright holder.

2
Preamble
Objective of this presentation is to describe the usage

the WebRTC technology and how it is implemented into
our OpenTouch Suite for MLE H2-2016 solution with
product release 2.2.1

3
Agenda
1. Technology information
2. WebRTC integration into OpenTouch solution
3. Quotation guidance
4. Conclusion
5. Documents reference

4
Agenda
4. Conclusion

5
WebRTC Generalities
Web Real-Time Communication is an API definition being
drafted by the World Wide Web Consortium (W3C) to enable browser based
applications for voice and video calling, IM/chat and P2P file sharing without
plugins
Other sources of information: http://en.wikipedia.org/wiki/WebRTC and http://www.webrtc.org/
Major web browsers implementing WebRTC:
Supported mobile platforms:
Strong support from Google company

MS Internet Explorer not compatible
Apple Safari not compatible

6
WebRTC Generalities
IDC# US41209416 (April 2016)

Source: Source: IDC U.S. Enterprise Communications Survey, January 2016, N=361

7
WebRTC Definitions and Components

The major technical specificities of WebRTC are the following:
WebRTC is primarily defined to support communications between peers over the Internet. The
procedure to establish a communication requires the support of different protocols for IP
connectivity (ICE / STUN / TURN)
Only encrypted media streams are supported. SRTP is used for this and the cipher key
negotiation is done via DTLS
WebRTC multiplexes audio and video streams in a single SRTP bundle flow but should also be
able to carry them over separate SRTP flows for compatibility only
Audio codecs G.711 and OPUS in WebRTC definition (G.722 also supported by some browsers)
Video codecs still under definition but the current browser implementations support the VP8
and H.264
No call control signaling is defined for WebRTC. Only some low level media streams
negotiation protocol are standardized: used to ensure a bi-directional communication
between the server and the browser. The signaling protocol can be transported based on
different mechanisms (e.g. WebSocket for SIP, JSEP or other proprietary mechanisms)
8
IP Connectivity
ICE (Interactive Connectivity Establishment)

ICE is a standardize method to retrieve usable IP addresses and ports for the signaling and for the
media paths. In other terms it provides a way to discover a path between the web browser connected
on the public Internet and the other peer (Media server or endpoint) located in the corporate network.
Supporting ICE is mandatory to enable a WebRTC communication, even if no NAT and Firewall are in the
path.
STUN (Session Traversal Utilities for NAT)

STUN is a protocol providing the browser with the capability to discover its public IP@. It implies a
STUN server located in the Internet (meaning a server having a public IP@). This server is a mirror
indicating to the browser how it is seen from the Internet.
TURN (Traversal Using Relay NAT)

TURN is a protocol providing the browser with the capacity to receive data over TCP or UDP coming
from the Internet. A TURN server is a server located in the Internet (meaning a server having a public
IP@) that will act as a relay server for the media traffic between the web browser and the
Communication server. The Turn server encapsulates the media flow so it can pass through NAT and
firewall to reach the browser.

9

DTLS (Datagram Transport Layer Security)
SECURITY
DTLS provides communication security for datagram based protocols. It allows the exchange of the
encryption keys computed by the two peers of the communication.
SRTP (Secure Real Time Protocol)

SRTP is used to encrypt the media flows that can be audio or video.

10

Standards and RFCs related to WebRTC:
RFC
RFC
RFC
RFC
RFC
RFC
RFC
3711
5246
6347
5245
5389
7118
7478
The Secure Real-time Transport Protocol (SRTP)

The Transport Layer Security (TLS) Protocol Version 1.2
Datagram Transport Layer Security Version 1.2 (DTLS)
Interactive Connectivity Establishment (ICE lite)
Session Traversal Utilities for NAT (STUN)
WebSocket Protocol as a Transport for the Session Initiation Protocol (SIP)
Web Real-Time Communication Use Cases and Requirements
Draft IETF RFCs for rtcweb:

draft-ietf-rtcweb-data-channel,
draft-ietf-rtcweb-data-protocol,
draft-ietf-rtcweb-rtp-usage,
draft-ietf-rtcweb-stun-consent-freshness,
draft-ietf-rtcweb-video

11
Agenda
4. Conclusion

12
WebRTC in OpenTouch solution

Generalities
WebRTC and its usage within OTC Web:
OTC Web is the OpenTouch conferencing client that gives access to collaboration services
(such as IM, active talker indication, document or desktop sharing) as well as audio media
(based on WebRTC). Support of video media is not yet available (refer to product roadmap for
more information)
OTC Web is a pure Web based client running into any (or quite) recent browser. OTC Web is
primarily targeted at OTs conference guest participants, who are generally people outside of
the enterprise and connect to the OpenTouch conference system via the Internet. OTC Web
can also be used by employees of the company, with Conversation or Connection user profiles
WebRTC for audio media is supported into OTC Web when the user (guest or corporate) is
connected remotely from the public Internet through the OT SBC installed in the DMZ

13

Solution Blueprint
Reverse Proxy (RP):
- acts as an applicative gateway between remote clients and the OpenTouch server
- exposes a web collaboration interface for enabling access to OpenTouch scheduled conferencing
services to remote guests participants
Internet
Corporate
Internet
firewall
RP
HTTPS
Guest participant
office network
RTP
OT-SBC
SIP over WebSocket

secured
SIP over TCP/UDP
OpenTouch SBC (OT SBC):

HTTPS
OTC-Web
SRTP
- ICE-lite support (for IP connectivity)

- SIP signaling and media traffic between
remote WebRTC clients and OT server
- SRTP to RTP
Guest participant
over Internet
OTC Web:
Corporate network
- Terminates the SIP signaling in a Javascript module

- Manages the media handling thru the WebRTC API
OTBE/OTMS
DMZ
SRTP
OTC-Web
OTC-Web
HTTPS
14
OpenTouch server
(OTBE/OTMS):
- Web application server. Provides SIP

configuration to the client (thru a data
channel flowing thru OTES)
- Hosts Data/Audio/Video conferences
No usage of WebRTC within

OTC Web inside the
corporate network with
current product release !

OTC Web with WebRTC user experience
Allow remote users to join an OT conference with browser embedded VOIP:
- For remote users connecting through the OT SBC (enterprise border element)
- As an alternative to dial in/out with a phone set
- G.722 and G.711 audio codecs supported for VoIP
- Supported OS for desktop: Microsoft Windows 7/8+ and Apple MacOSx 10.9+
- Supported web browsers for WebRTC within OTC Web client:
Firefox version 39 minimum
Chrome version 43 minimum
NB: MS Internet Explorer and Apple Safari are not supported for VoIP within OTC Web
15

OTC Web with WebRTC user experience
NEW!
NB: Call me on my computer for VoIP on the (based on WebRTC) is only proposed if the web browser
used is compatible, the PC has microphone and speaker and the user is connecting remotely through the
OT SBC installed at the border of the corporate network on the OT Conference server side
16

Integration into OT SBC
OT SBC is the mandatory WebRTC/SIP Gateway:
-
SIP over Secure Web Socket as a signaling protocol managed by the OT SBC
OT SBC supports the ICE protocol for IP connectivity of WebRTC clients

NB: OT SBC implements an ICE-lite and HNAT (Hosted NAT mechanism) to handle IP addressing for remote
OTC Web with WebRTC. OT SBC is able to reply to STUN requests from the browser with WebRTC even if
it does not implement a complete STUN server. The advantage is that the OT SBC can have a NATted WAN
interface meaning a private IP@ behind a Firewall (instead of public IP@)
Encryption mandatory between WebRTC clients and the OT SBC:

-
Support of DTLS/SRTP for encryption key exchange managed by the OT SBC
OT SBC supports WebRTC feature as of product release 2.1.1 (a specific software

patch is delivered with this release refer to Technical Release Note document for more details -)

17

Integration into OT SBC
Product Limits with OT SBC release 2.1.1:
OT SBC R2.1.1
OTSBC virtualized
sizing:
x1 vCPU*
RAM 4GB
OTSBC virtualized
sizing:
x4 vCPU*
RAM 8GB

SIP calls
SIP calls Remote
Transcodi
SRTP/RTP WebRTC
(audio (audio+vide SIP
ng
Sessions Sessions
only)
o)
users
Sessions
1000
500
1000
1000
750
300**
4000
2000
6000
4000
750
N/A
18
Agenda
4. Conclusion

19
Quotation of WebRTC in OpenTouch solution

Software licenses in the OT SBC product
SW licenses: WebRTC session and redundant WebRTC session

WebRTC session is an option of SIP calls
Item to configure in ACTIS (OT SBC tab): number of WebRTC sessions
Min value: 0
Max value: maximum WebRTC sessions in product limits
NB: the SRTP/RTP session option does not need to be configured in addition to the
WebRTC session option (encryption of signaling&media with WebRTC is by default)

20
Quotation of WebRTC in OpenTouch solution

ACTIS
WebRTC
As an example, if the need is to allow 10 OTC Web
users to access simultaneously an OT audio
conference through WebRTC compatible browser
ACTIS v20.0
Enter: 10 SIP calls
Enter: 10 WebRTC sessions

to enable WebRTC with OTC Web
client
Enter: 10 remote SIP users

to enable remote users with
OTC Web/WebRTC
to register through the OT SBC

21
Agenda
4. Conclusion

22
Conclusion
Customers, partners
OPENNESS
SECURE
Nomads
Remote workers

STATE OF ART
OPENNESS AND FLEXIBILITY

ALTERNATIVE TO PSTN CALL IN/OUT
ENCRYPTION OF SIGNALING AND MEDIA OVER INTERNET
FUTURE PROOF SOLUTION

READY FOR EVOLUTION (EXTEND WEBRTC CAPABILITIES WITH VIDEO)
23
Agenda
4. Conclusion

24
Reference Documentation
Documentation available in eBusiness Portal:
Resources Center > Presales Documentations
Presales Presentation - OpenTouch SBC Design and Quotation Guidance
Presales Presentation Solution for Remote User Access
Customer Support > Technical Support
OpenTouch Client Administrator Manual

25
enterprise.alcatel-lucent.com

26
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27
BACKUP SLIDES

28
Secure Websocket
HTTP
TCP
IP
HTTP
TLS
TCP
IP
HTTP
HTTPS

Websocket
Websocket
TLS
TCP
TCP
IP
IP
WS
29
WSS

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WebRTC Technology in OpenTouch R2.2.

With slide notes

Pre-Sales Presentation | November 2016

Objective of this presentation is to describe the usage

Pre-Sales Presentation | November 2016

Pre-Sales Presentation | November 2016

Pre-Sales Presentation | November 2016

Major web browsers implementing WebRTC:

Supported mobile platforms:

Strong support from Google company

Pre-Sales Presentation | November 2016

IDC# US41209416 (April 2016)

Pre-Sales Presentation | November 2016

WebRTC Definitions and Components

WebRTC Definitions and Components

ICE (Interactive Connectivity Establishment)

STUN (Session Traversal Utilities for NAT)

TURN (Traversal Using Relay NAT)

Pre-Sales Presentation | November 2016

WebRTC Definitions and Components

SRTP (Secure Real Time Protocol)

Pre-Sales Presentation | November 2016

WebRTC Definitions and Components

The Secure Real-time Transport Protocol (SRTP)

Draft IETF RFCs for rtcweb:

Pre-Sales Presentation | November 2016

Pre-Sales Presentation | November 2016

WebRTC in OpenTouch solution

Pre-Sales Presentation | November 2016

WebRTC in OpenTouch solution

SIP over WebSocket

SIP over TCP/UDP

OpenTouch SBC (OT SBC):

- ICE-lite support (for IP connectivity)

- Terminates the SIP signaling in a Javascript module

- Web application server. Provides SIP

No usage of WebRTC within

WebRTC in OpenTouch solution

WebRTC in OpenTouch solution

WebRTC in OpenTouch solution

OT SBC supports the ICE protocol for IP connectivity of WebRTC clients

Encryption mandatory between WebRTC clients and the OT SBC:

Support of DTLS/SRTP for encryption key exchange managed by the OT SBC

OT SBC supports WebRTC feature as of product release 2.1.1 (a specific software

Pre-Sales Presentation | November 2016

WebRTC in OpenTouch solution

Pre-Sales Presentation | November 2016

Pre-Sales Presentation | November 2016

Quotation of WebRTC in OpenTouch solution

SW licenses: WebRTC session and redundant WebRTC session

Max value: maximum WebRTC sessions in product limits

Pre-Sales Presentation | November 2016

Quotation of WebRTC in OpenTouch solution

Enter: 10 SIP calls

Enter: 10 WebRTC sessions

Enter: 10 remote SIP users

Pre-Sales Presentation | November 2016

Pre-Sales Presentation | November 2016

Pre-Sales Presentation | November 2016

OPENNESS AND FLEXIBILITY

ENCRYPTION OF SIGNALING AND MEDIA OVER INTERNET

FUTURE PROOF SOLUTION

Pre-Sales Presentation | November 2016