1. Introduction to
WebRTC
November 20th, 2012

2. In a few words
WebRTC (Real Time Communications) will be the next big thing in terms of
unified communications during the next years. Web browser will be able to
manage voice and video in a native way.
During 60 minutes, Elías Pérez, Antón Román and Iago Soto will talk about
their vision about this technology and how is the best way to implement
WebRTC in the legacy and VOIP networks of corporations and telcos.
Elías Pérez Carrera Iago Soto Mata Antón Román
CEO CMO CTO
@epcarrera @iagosoto @antonroman
elias.perez@quobis.com iago.soto@quobis.com anton.roman@quobis.com

3. Agenda
- Introduction to HTML5 and WebRTC
- Network architecture
- Tech challenges
- Application cases
- Demos
- Identity Management
- Questions and answers

4. HTML5 and WebRTC. Short intro
... is the fifth version of the HTML language, that
offers new capabilities to create web pages,
specially in terms of dynamic elements.
Defines WebSockets as the new transport
protocol between browser and servers
It is promoted by W3C
... is an opensource project that makes possible
to manage multimedia communications in the
web browsers, using simples API's in Javascript,
in a native way.
It is promoted by the team of Google Chrome

5. HTML5 and WebRTC. Standards
Two main players:
● IETF RTCWeb WG ( Internet world)
● W3C WebRTC WG (web world)
Drafts:
WebRTC 1.0 Real-time Communication Between Browsers
WebRTC MediaCapture
The WebSocket API
draft-ietf-rtcweb-data-channel
draft-ietf-rtcweb-jsep
draft-ietf-rtcweb-rtp-usage
... etc ...
Just a few days ago !!!

6. HTML5 and WebRTC. Increasing interest

7. HTML5 and WebRTC. Advantages
Opensystems, with no propietary implementations
¡No plugins!
Multi-platform... and multi-device!

8. HTML5 and WebRTC. Application cases
Software is going to migrate
to WEbRTC like collab apps

9. HTML5 and WebRTC. Application cases
And Click2call will be a trend in
CRM or marketing websites

10. Architectures. Browser to browser
Any browser will have
capabilities to call any
other with RTC.

11. Architectures. Multiconference

12. Architectures. SIP interconnection
SIP to WebRTC gateway will be implemented for interconnection.

13. Architectures. PABX interconnection

14. Tech challeges. Codecs
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 latence 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 be needed transcoding or interworking of DTMFs
(RFC 4733-> INFO, RFC 4733-> in-band, etc).
And this only for audio...
... battle in vídeo VP8 vs H264

15. Tech challeges. SRTP vs DTLS-SRTP
VoIP devices implement
normally RTP encryption
using SRTP. They share
the key in the SDP
protocol.
DTLS-SRTP implements a new method to manage the
SRTP key. This is done using DTLS, a version of TLS
based on datagrams.

16. Tech challeges. NAT. STUN and ICE
Browsers are allways behind NAT and Firewalls, so it's
difficult to exchange packets in real time.
The user want a seamless communication, with no problems
(similar to Skype).
It is needed to found a "way" for RTP/RTCP traffic,
independent to the architecture..
ICE, makes possible (using STUN and TURN, protocols) to
discover y to choose the addresses that are going to be used
to exchange packets.
ICE was a difficult protocol to implemnt and had to be defined
during some years.

17. Tech challeges. Media negotiation
Uses SDP for negotiating media.
Classic negotiation offer-answer
There is a draft IETF:
SDP for the WebRTC / draft-nandakumar-rtcweb-sdp-00
Example SDP:
| v=0
| o=bob 16833 0 IN IP4 0.0.0.0
| s=
| t=0 0
| a=ice-ufrag:c300d85b
| a=ice-pwd:de4e99bd291c325921d5d47efbabd9
| a2
| a=fingerprint:sha-1
| 99:41:49:83:4a:97:0e:1f:ef:6d:f7:c9:c7:7
| 0:9d:1f:66:79:a8:07 |
| m=audio 49203 RTP/AVP 109

18. Tech challeges. Signalling. Options
WebRTC does not define how to manage signalling
There are some possibilities
● XMPP / Jingle
● SIP
● Protocol ad-hoc that manages SDP's
Quobis chooses SIP:
● "Standard": NGN networks are supported by SIP
● "Interoperable": it's possible to connect to everything!!!
● "Powerful": allows to build apps over it
Exisiting SIP implementations:
● sipML5: first implementation available
● JsSIP: authors of the draft-ibc-sipcore-sip-websocket

19. Introduction to QoffeeSIP
Complete implementation of SIP
protocol using Javascript
It runs directly in the browser
Focused on developers,
written using CoffeeScript
Easy to extend.
Light application (5 KB)
Will be opensource in the nexts
weeks.

20. Introduction to QoffeeSIP
+
SIP
=
Interconnection of browsers
with SIP legacy devices.

21. Tech challeges. Adoption. Browsers
Really involved in WebRTC
First stable versions (for desktop)
Implements PeerConnection, GetUserMedia, etc..
Involved, working more slowly than Chrome
Implements part of the API
Iniciative Firefox OS for mobiles
Involved in WebRTC
First stable versions (for desktop)
Implements PeerConnection, GetUserMedia, etc..

22. Tech challeges. Adoption. Browsers
Unknown roadmap, patents involved
Important for smartphones and tablets
Involved in WebRTC
Suggestion and proposal that do not fit with the standard
There is a plugin developed by Google (Chrome Frame)
Developed by Ericsson labs
First browser with WebRTC in mobiles (without
Websockets)
Beta version for developers.

23. Tech challeges. Security.
Media access: must be allowed by the user
Models of allowance:
● Access to cam and mic in a unique session
● continuous access to cam and mic
● Accesses based on users:
○ Allow calls just to an user
○ Allow calls to known users
LED to show that camera o mic are been used
Automatic off in case of not attendance (change of window)
Defined in draft-ietf-rtcweb-security

24. Tech challeges. Monitoring
It is possible to monitor all the
traffic, similar to standard SIP.
Similar to SIP over TLS, if WSS is
used (secure Websockets)
monitorization should be done at
the edges (most usually in the
server).
We have modified the monitoring
tools in order to see websocket
traffic.

25. Case studies. Applications
Telcos Corporaciones
SP / Social media Fabricantes

26. Case studies. Applications
Telcos
● Webphone for existing customers
● Integration in NGN and IMS
● Inbound channel for corporate
customers
● Alternative to calls from mobiles

27. Case studies. Applications
Corporations
● Inbound Click to call. New channel.
● Can be deploy by ourtsourcers
● Calls in internal directories
● Multiconferences.
● Integration with legacy systems

28. Case studies. Applications
SP / Social media
● Social networks -> next MVNOs
● New services for webs managing IDs
● ISPs will offer UC modules/widgets
● Entertainment: chats, etc...

29. Case studies. Applications
Manufacturers
● Online gaming
● Set-top-boxes and web TV
● Gateways WebRTC to IPBX
● Mobile device manufacturers

30. Demo. TalkSetup
Join a demo of WebRTC at
http://TalkSetup.quobis.com

31. Demo. TalkSetup backend
Easy to manage WebRTC calls and users.

32. Tech challenges. Identity management
Makes possible to be sure of the
identity using a third
party
Adds a second factor of authentications because we
validate the device (smartphone or PC) and the
credentials are introduced ciphered in a SIP
signalling packet.

33. Tech challenges. Identity management

34. Tech challenges. Identity management

35. Tech challenges. Identity management
Agents can be sure
of the identity of
the person who is
calling.

36. Demo. IdentityCall
Demo using national electronic ID card.

37. What we have learned
● Voice traffic is going to be thru web
● Browsers are the new endpoints
● A website of a company can be the call
center
● Telephone number is not important
● Security, identity a privacy is very
important
● New business opportunities

38. Any questions?
Elías Pérez Carrera Iago Soto Mata Antón Román
CEO CMO CTO
@epcarrera @iagosoto @antonroman
elias.perez@quobis.com iago.soto@quobis.com anton.roman@quobis.com