WebRTC allows for real-time communication directly in web browsers without plugins. It uses existing VoIP technologies and exposes communications APIs to web developers. WebRTC handles media streams and leaves signaling up to applications. While implementations exist, challenges remain around mobile, encryption, and fragmentation. WebRTC has the potential to disrupt telephony and enable rich, customized unified communications applications on the web.

1. WebRTC
The End Of The World (As We Know It)

2. Welcome
To The Beginning Of The Post-Telephony Era

3. I’m Steve
Steve Sokol, Entrepreneur In Residence /
Director of Strategic Programs
at Digium

4. What is WebRTC?

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

6. How does it work?

7. ❖ WebRTC leverages existing VoIP
technologies
❖ WebRTC exposes communications
development to the 20M web developers in
the world
❖ WebRTC sets rules for media, leaves
signaling up to the application developer

8. WebRTC Call In A Nutshell
❖ Get media streams from camera, mic
❖ Create an “offer” session description
❖ Send the offer to the far-end party
❖ Receive an “answer” session description
from the far-end party
❖ Discover a path that works by testing all
paths

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10. Web Server
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11. New JavaScript APIs
❖ Media Capture
❖ Peer-To-Peer Networking

12. Creating A Connection
❖ Built-In NAT Traversal using ICE
❖ STUN - Discover network details
❖ TURN - Relay as last resort

13. Encoding Media
❖ Audio Codecs
❖ Mandatory: Opus, G.711
❖ Optional Codecs
❖ Video Codecs

14. Encrypting Media
❖ Mandatory
❖ Secure Realtime Protocol (SRTP)
❖ SDES vs. DTLS-SRTP Key Brokering

15. What About Signaling?
❖ No mandatory protocol or mechanism
❖ Can be done using SIP or Jingle using
JavaScript libraries
❖ Can be done better using other methods:
❖ WebSockets or XMLHttpRequest transport
❖ Simple JSON signaling
❖ Use a protocol that suits your use case
perfectly, not a protocol built to handle all

16. ❖ No mandatory signaling protocol is a
GOOD THING™
❖ Gives developers absolute control over
the user experience
❖ Avoids the tendency to rebuild the PSTN
❖ Avoids the “federation” issue
❖ Allows for identity to be more than a
number

17. ❖ URL-Based Calling
❖ http://www.digium.com/contact/sales
❖ http://www.digium.com/contact/ssokol
❖ Directory-Based Calling
❖ Facebook
❖ Twitter
❖ Linked-In
❖ Corporate LDAP

18. • “Inside” users will use a web-based or
mobile client
• “Outside” users will use portal pages to
request access to various resources
• People
• Departments
• Expert Support

19. ❖ You will still need a communications
system or a communications service
❖ You (eventually) may not need a “phone
company”
❖ Prediction: wired and wireless carriers
will become glorified ISPs within the decade
❖ WebRTC will make rich communications
a 100% “OTT” business

20. So, Is It Ready To Use?

21. Yes and no...

22. ❖ 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

23. Challenges

24. ❖ Mobile Deployments
❖ Large-Scale Multi-Party
❖ Legacy Integration
❖ Codec Selection
❖ Fragmentation (Microsoft’s CU-RTC-
Web)
❖ Encryption Keys

25. Future features and
enhancements...

26. ❖ Peer-To-Peer Data
❖ Real-Time Text (Captions)
❖ Media Recording
❖ Screen / Desktop / Tab Sharing
❖ Statistics / Monitoring
❖ Possibly low-level APIs

27. A few use cases:

28. ❖ Social Media
❖ Call Center Agent Interface
❖ Conferencing & Collaboration
❖ Enhanced Customer Care
❖ Distance Learning
❖ In-Game Communications
❖ Broadcasting

29. Big Changes
(Welcome To The Post-Telephony Era)

30. • Telephony has been holding back
communications for the past decade.
• SIP was hijacked: what started out as a
peer-to-peer system was twisted into
“PSTN-Over-IP”
• Improvements and price reductions in
bandwidth, mobile, web make a real change
possible

31. ❖ Fully Unified Communications
❖ Integration of communications directly
into business and social applications
❖ Communications as a feature or
function rather than as a service
❖ Customized User Experience
❖ Excellent Privacy / Security
❖ Significant Cost Reduction

32. Asterisk And WebRTC

33. ❖ Asterisk 11 added ICE, STUN, TURN
support, WebSocket transport for SIP
channel and other tweaks
❖ You can now create web endpoints using
Asterisk and a JavaScript SIP library
❖ SIPML5
❖ JS-SIP
❖ Asterisk can bridge between WebRTC
and legacy communications technologies

34. Demo Time!

35. Future versions of Asterisk will do more:
❖ Recording and playback of audio and
video
❖ Interfaces for additional / custom
signaling protocols
❖ Interactive voice and video applications

36. Thanks
Steve Sokol
ssokol@digium.com
+1 (256) 428-6101