WebRTC for Beginners Webinar Slides
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You have undoubtedly heard about WebRTC and understand that it has to do with real-time communications to and from a web browser, but do you really understand what it entails? This session will help demystify WebRTC in terms of what it is, what it requires, how it works, and who will use it.
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WebRTC for Beginners Webinar Slides
- 1. WebRTC For Beginners Andrew Prokop Director of Vertical Industries
- 2. 2 Agenda > What is WebRTC > Uses > Under the Covers > Tools > Issues > What’s Next
- 3. 3 What is WebRTC > Web Real-Time Communications > Plug-In Free > Application Free > Browser to Browser Media – Voice – Video – Data > A standard from W3C and IETF
- 4. 4 Markets > Healthcare > Higher Education > Customer Service > Conferencing > Social Media > Customer Portals > Etc.
- 5. 5 Give Me a Few Reasons > Viruses / Malware / Industrial Spying – Growing reluctance to run plugins or new software > Need for Rapid Development > Embedded Communications – Add communications to the tools and systems that require it > The Internet of Things
- 6. 6 A Few Real World Examples > Unify Circuit > MindMe > LiveNinja Katana > Cisco Spark > WhatsApp (somewhat) > Facebook Messenger (mobile version) > OnSIP (hosted VoIP) > Vidyo VidyoWorks > Interactive Intelligence PureCloud
- 8. 8 Click to Call Amir Zmora VP Alliances & Partnerships AudioCodes Mobile: +972-54-7769603 Call my Lync extension using WebRTC
- 9. 9 The WebRTC Stack WebRTC API Identity SDP ICE / STUN / TURN Security Codecs
- 11. 11 The WebRTC API > An Extension to HTML-5 – RTCRtpSender, RTCRtpReceiver, etc. – Standalone and Embedded Web Browsers > Standard Tools – HTML for display – JavaScript for WebRTC Objects
- 12. 12 <!DOCTYPE html> <html> <head> <title>My WebRTC application</title> <style type="text/css"> div#lVideo { width: 45%; height: 200px; background: black; float: left; } </style> <script type="text/javascript"src="myrtclib.js"></script> <script type="text/javascript"src="myrtcadapter.js"></script> </head> <body onLoad="onPageLoad();"> HTML
- 13. 13 WebRTC Signaling Whatever you want it to be! > WebSocket > XHR / Comet > XMPP / Jingle > SIP via JsSIP > Smoke Signals Flexibility by design
- 14. 14 Media > Defined by Session Description Protocol (SDP) – Just like SIP! > Audio Codecs – G.711 Narrowband Audio > And possibly iLBC and iSAC – Opus Wideband Audio > Video Codecs – H.264 and VP8 (headed towards VP9) > Separate Point-to-Point Data Channel
- 15. 15 Opus > Bit-rates from 6 kb/s to 510 kb/s > Sampling rates from 8 kHz (narrowband) to 48 kHz (fullband) > Frame sizes from 2.5 ms to 60 ms > Support for both constant bit-rate and variable bit-rate Audio > Support for speech and music, mono and stereo > Dynamically adjustable bitrate, audio bandwidth, and frame size > Good loss robustness and packet loss concealment (PLC)
- 16. Ah, but what about firewalls and private IP addresses?
- 17. 17 STUN
- 18. 18 TURN
- 19. 19 Security > Signaling – DTLS (Datagram Transport Layer Security) – Think HTTPS for UDP > Media – SRTP
- 20. 20 Identity Browser is configured with identity provider(s) for the user > User “logs on” using protocol downloaded from identity provider > Browser gets an assertion from identity provider that binds with DTLS fingerprint to the identity such as aprokop@arrowsi.com > The calling JavaScript passes the assertion to the far side > The far side’s browser verifies the assertion with identity provider and checks that DTLS fingerprint matches the assertion > Browser displays “secure to aprokop@arrowsi.com”
- 21. 21 Challenges > WebRTC is still in flux > No defined signaling protocol > Not all parties are on board – Apple and Microsoft still not fully onboard – Microsoft has proposed competing/complimentary technology -- ORTC
- 22. 22 Taming the Beast > 99 (and counting) WebRTC SDKs and Toolkits – 39 focused on mobile devices – 34 focused on messaging > 18 Open Source Projects
- 23. 23 I n t e r n e t HTML 5 Consumer Facing Web Application Make Call SM SMSM SM SM SM Breeze Breeze Breeze Web RTC Media Services (AMS) SIP/H.323 Agent STUN/ TURN Avaya SBC-E WebRTC Agent WebRTC Media Reverse Proxy SIP SIP WebRTC Snap-In STUN/ TURN Avaya SBC-E Customer WebRTC Media
- 24. 24 AudioCodes WebRTC Gateway Access SBC Transcoding GW (Optional) SIP/WebSockets ICE/DTLS Opus/UDP/SRTP
- 25. 25 What’s Next > Specification will solidify > Multiparty media > Widespread IoT adoption > Middleware (e.g. media stream recording) > Better statistics and reporting > ORTC and WebRTC will merge > Apple will finally make up its mind
- 26. Questions