End of the world presentation

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What is WebRTC and How Does it Work?

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End of the world presentation

  1. 1. WebRTCThe End Of The World (As We Know It)
  2. 2. WelcomeTo The Beginning Of The Post-Telephony Era
  3. 3. I’m SteveSteve Sokol, Entrepreneur In Residence / Director of Strategic Programs at Digium
  4. 4. What is WebRTC?
  5. 5. Photo Credits: Tom Keating - TMC.net, Eric Hernaez - Netsapiens
  6. 6. How does it work?
  7. 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. 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
  9. 9. Web Browser et ck So eb P) W or g (SD P) TP lin SD HT igna g( rS lin ffe na O Sig er sw An SRTP Media StreamsWeb Server P) SD g( lin P) na Sig ( SD er ng et ali ck Web Browser sw ign So An rS eb ffe W O or TP HT
  10. 10. Web Server Sig g lin na na lin Sig g SRTP Media Streams Media Server Gateway PBXWeb Browser
  11. 11. New JavaScript APIs❖ Media Capture❖ Peer-To-Peer Networking
  12. 12. Creating A Connection❖ Built-In NAT Traversal using ICE❖ STUN - Discover network details❖ TURN - Relay as last resort
  13. 13. Encoding Media❖ Audio Codecs ❖ Mandatory: Opus, G.711 ❖ Optional Codecs❖ Video Codecs
  14. 14. Encrypting Media❖ Mandatory❖ Secure Realtime Protocol (SRTP)❖ SDES vs. DTLS-SRTP Key Brokering
  15. 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. 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. 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. 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. 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. 20. So, Is It Ready To Use?
  21. 21. Yes and no...
  22. 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. 23. Challenges
  24. 24. ❖ Mobile Deployments❖ Large-Scale Multi-Party❖ Legacy Integration❖ Codec Selection❖ Fragmentation (Microsoft’s CU-RTC- Web)❖ Encryption Keys
  25. 25. Future features and enhancements...
  26. 26. ❖ Peer-To-Peer Data❖ Real-Time Text (Captions)❖ Media Recording❖ Screen / Desktop / Tab Sharing❖ Statistics / Monitoring❖ Possibly low-level APIs
  27. 27. A few use cases:
  28. 28. ❖ Social Media❖ Call Center Agent Interface❖ Conferencing & Collaboration❖ Enhanced Customer Care❖ Distance Learning❖ In-Game Communications❖ Broadcasting
  29. 29. Big Changes(Welcome To The Post-Telephony Era)
  30. 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. 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. 32. Asterisk And WebRTC
  33. 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. 34. Demo Time!
  35. 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. 36. Thanks Steve Sokolssokol@digium.com +1 (256) 428-6101

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