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WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
WebRTC eduCONF
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WebRTC eduCONF

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  • 1. WebRTC What’s going on and is it of use to NRENs Mihály Mészáros, NIIF Institute eduCONF Workshop 13/03/14
  • 2. 2Connect | Communicate | Collaborate Agenda ● Overview, WebRTC and RTCWEB History, API ● WebRTC and NRENs: Is it a good idea to jointly develop WebRTC based RTC service pilot for the GÉANT community? ● Roll Call, status of NREN Web / Desktop Conference services ● adapt the technology level of the training to audience preference ● RTCWEB architecture, a technology deep dive, (nuts & bolts) ● NAT Firewall traversal, codecs, security, identity, troubleshooting ● Experience WebRTC (demonstrations, games) ● Building real world service Frameworks, tools ● Components to build a real world WebRTC service ● SWOT Analysis. Is WebRTC Ready? What would it take? ● Predictions & Summary, WebRTC related Open Discussion
  • 3. 3Connect | Communicate | Collaborate History ● Global IP Solutions ● In May 2010, Google bought GIPS for $68.2 million. ● May 31, 2011 Google released Open Source WebRTC. ● mainly based on GIPS technology ● Dual Standardization Bodies ● RTCWEB IETF 2011-05-01 ● WebRTC W3C 2012-09-12 ● Aug 1, 2012 getUserMedia in Chrome 21 ● Oct 2, 2012 PeerConnection in Chrome 23 ● Nov, 2012 PeerConection in stable Chrome ● Feb 4, 2013 Firefox and Chrome interoperability achieved ● 2013 Hangouts VP8, 2014 Hangouts + WebRTC (H2O Vidyo)
  • 4. 4Connect | Communicate | Collaborate What is WebRTC ? (RTCWEB) ● WebRTC: “A framework, protocols and application programming interface that provide real time interactive voice, video and data in web browsers and other applications” ● Standardization ● WEBRTC (W3C) part of HTML5 ● RTCWEB (IETF) ● / IMS_WebRTC(3GPP) / ● Implementation ● Chrome, FireFox, Opera, Browser (Ericsson Research), etc. ● WebRTC native JAVA / C++ API support ● for Browsers and Apps ● Android, iOS(?)
  • 5. 5Connect | Communicate | Collaborate WebRTC ● WebRTC Peer to Peer Direct media ● Abstract signaling ● Hides complexity from the web developer ● Browser do the heavy lifting ● Signal processing ● Codec handling ● Audio Video synchronization ● Echo cancellation ● Peer to peer communication ● Firewall/NAT traversal ● Security ● Bandwidth management
  • 6. 6Connect | Communicate | Collaborate WebRTC API ● Major API Components ● GetUserMedia ● Acquiring audio and video ● which allows a web browser to access the camera and microphone ● DataChannels ● which allow browsers to share data via peer-to-peer ● PeerConnection ● P2P Communication ● Codec negotiation, Security ● Media handling, Bandwidth Management ● etc. ● Peer-to-peer DTMF ● RTCStatsReport ● Identity
  • 7. 7Connect | Communicate | Collaborate WebRTC API vs Alternative APIs ● Current nearly 1.0 WebRTC API couldn't be perfect. ● World Wide consensus is big challenge. ● First make API stable. ● Redesign takes time. So redesign only after stable API 1.0 ● http://dev.w3.org/2011/webrtc/editor/webrtc.html ● http://dev.w3.org/2011/webrtc/editor/getusermedia.html ● API Alternatives ● WebRTC Object API (ORTC) https://rawgithub.com/openpeer/ortc/master/ortc.html http://www.w3.org/community/orca/ ● Microsoft (CU-RTC-Web) http://lists.w3.org/Archives/Public/public-webrtc/2012Aug/0014.html
  • 8. 8Connect | Communicate | Collaborate WebRTC and NREN's ● TNC2013 TERENA Technical Advisory Council ● Jan Meier: WebRTC Why you should care? ● Big Blue Button WebRTC Support ● Donated by UNINET, NorduNet ● 2013 Aug 26 WebRTC meeting ● Big Blue Button WebRTC support (NORDUNET) ● Videoconference Gateway/MCU (NIIFI, JANET) ● Lecture Recording (REDIRIS) ● GN4 New Idea From ● Open Mailing lists ● discussion@nrenum.net ● webconf@terena.org
  • 9. 9Connect | Communicate | Collaborate Look under the hood technology vs. High-level overview
  • 10. 10Connect | Communicate | Collaborate Introduction / WAYF / Roll Call ● What do your prefer / expect from this WebRTC training? ● High level overview, status, possible directions, implementations ● Deep dive in technical details (nuts and bolts) ● What do you know already about WebRTC technology? ● What functions are mandatory to implement in RTC collaboration solution beyond video conference today? ● Secondary video/Presentation sharing, Buddy list,Presence, Calendar integration, Directory / Phonebook, File sharing, IM/Chat, Whiteboard, integration API MOOC/eLearning etc. ● What solutions does your NREN use today for Desktop/Web Videoconference? (What are the limitations of such product?) ● Does your NREN provides STUN/TURN service? ● Is the exotic platform support is important for your NREN? e.g. Linux distributions, mobile platforms
  • 11. 11Connect | Communicate | Collaborate Technically ● W3C WebRTC JavaScript API ● WebRTC use abstract signaling protocol ● Designed in mind SIP, XMPP/JINGLE compatibility ● WebRTC signaling is fully application specific ● Security Architecture ● IETF RTCWEB WG (wire protocols) ● NAT / Firewall traversal ● IPv4/IPv6 ● Multiplexing data/media ● Security ● Identity,Encryption, Privacy ● DTLS-SRTP, SDES-SRTP (Audio, Video) ● SCTP over DTLS (Data) ● Fresh / Current / leading edge IETF standards ● backward compatibility issues ● SDP capability description ● media bundling ● ICE (STUN/TURN) ● Trickle ICE ● Congestion Control ● RTP SAVPF ● RTCP feedback ● multiplexing ● RTP RTCP ● RTP multiplexing (audio video) ● codecs (e.g. VP8, Opus, etc.)
  • 12. 12Connect | Communicate | Collaborate JavaScript Session Establishment Protocol (JSEP) IETF RTCWEB Workgroup
  • 13. 13Connect | Communicate | Collaborate Offer – Answer model ● Session Description Protocol capability exchange ● Peer State transitions: http://dev.w3.org/2011/webrtc/editor/images/peerstates.svg ● createOffer, createAnswer Image source: http://chimera.labs.oreilly.com/books/1230000000545/ch18.html
  • 14. 14Connect | Communicate | Collaborate SDP Anatomy - Nuts & Bolts ● SDP is complex, WebRTC SDP Anatomy ● http://webrtchacks.com/anatomy-webrtc-sdp/ ● Example: v=0 o=- 13051590608781842 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio video data a=msid-semantic: WMS 0uUCRBhlvHjTrdnKqTaVj6VJCRuSutXJsCET m=audio 52429 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 195.111.192.2 a=rtcp:52429 IN IP4 195.111.192.2 a=candidate:2576070158 1 udp 2122260223 10.10.10.6 33748 typ host generation 0 a=candidate:2576070158 2 udp 2122260223 10.10.10.6 33748 typ host generation 0 a=candidate:2057973986 1 udp 1686052607 178.48.31.2 33748 typ srflx raddr 10.10.10.6 rport 33748 generation 0 a=candidate:2057973986 2 udp 1686052607 178.48.31.2 33748 typ srflx raddr 10.10.10.6 rport 33748 generation 0 a=candidate:3607644926 1 tcp 1518280447 10.10.10.6 0 typ host generation 0 a=candidate:3607644926 2 tcp 1518280447 10.10.10.6 0 typ host generation 0 a=candidate:4040299261 1 udp 41885439 195.11.92.2 52429 typ relay raddr 178.48.31.2 rport 35976 generation 0 a=candidate:4040299261 2 udp 41885439 195.11.92.2 52429 typ relay raddr 178.48.31.2 rport 35976 generation 0 a=ice-ufrag:NmMQwBJpi4qLGvnd a=ice-pwd:HJUMzaN0+ExHkNtLmZjYvpEM a=ice-options:google-ice a=fingerprint:sha-256 41:68:6B:2C:A5:80:AF:9D:5B:FA:3A:3F:D4:51:19:2C:E6:FC:08:2C:DD:D3:E5:ED:C9:84:D2:85:B8:A5:AC:48 …..
  • 15. 15Connect | Communicate | Collaborate Interactive Connectivity Establishment
  • 16. 16Connect | Communicate | Collaborate ICE - Nuts & Bolts ● ICE 1. Candidate gathering ● STUN ● TURN (allocation) 2. Prioritisation 3. Exchange 4. Connectivity checks 5. Coordination 6. Communication ● http://sdstrowes.co.uk/talks/20081031-ice-turn-stun-tutorial.pdf ● http://www.ietf.org/proceedings/67/slides/mmusic-11.pdf ● Trickle ICE ● https://github.com/emcho/trickle-ice/tree/master/slides
  • 17. 17Connect | Communicate | Collaborate Standard Based Firewall/NAT Traversal ● ICE RFC5245 (STUN/TURN) ● Tries to find the best path ● Firewall traversal ● IPv4, IPv6 Inter-working ● Multiple IP addresses ● Beyond ICE ● RFC5245 drawback ● lengthy ● Trickle ICE draft ● Reducing session establishment time ● Reducing ICE processing times ● Incremental Provisioning of Candidates for the Interactive Connectivity Establishment (ICE) Protocol ● XMPP XEP-0176 ● Implemented
  • 18. 18Connect | Communicate | Collaborate ICE vs. Trickle ICE Slide from: trickle-ice-iet86-orlando.pptx STUN Server STUN Server BobAlice disco disco offer and candidates … connectivity checks … answer and candidates Vanilla ICE as per RFC 5245 STUN Server STUN Server BobAlice disco disco O/A with host or no cands … more cands & conn checks …
  • 19. 19Connect | Communicate | Collaborate Architecture Overview Image source: http://www.webrtc.org/_/rsrc/1317202919504/reference/WebRTCpublicdiagramforwebsite%20%282%29.png
  • 20. 20Connect | Communicate | Collaborate Protocol Stack ● Peer-to-Peer media communication ● RTCP Multiplex ● Media Multiplex (audio, video) Image source: http://www.sloreto.com/slides/Aalto022013WebRTC/images/protocolStack.jpg
  • 21. 21Connect | Communicate | Collaborate Security ● Trust in your browser only (TCB) ● Secure End to End Communication ● getUserMedia ● Secure User Interface opt-in (e.g. Camera, audio access) ● User can allow/deny audio video source usage ● Media/Data Encryption is mandatory! ● DTLS-SRTP / DTLS ● SDES-SRTP - “MUST NOT implement” according IETF 87 http://tools.ietf.org/agenda/87/slides/slides-87-rtcweb-5.pdf ● AAI identity provision ● WebRTC Security framework ● SDP attached Identity Assertion (a=identity: base64) ● Signaling protocol is not defined by WebRTC ● Use secure signalling e.g. SIP over WSS(TLS+WebSocket)
  • 22. 22Connect | Communicate | Collaborate RTCWEB Security architecture Overview +----------------+ Unspecified +----------------+ | | protocol | | | Signaling |<----------------->| Signaling | | Server | (SIP, XMPP, ...) | Server | | | | | +----------------+ +----------------+ ^ ^ | | HTTPS | | HTTPS | | | | v v JS API JS API +-----------+ +-----------+ | | Media | | Alice | Browser |<--------------------------->| Browser | Bob | | DTLS+SRTP | | +-----------+ +-----------+ ^ ^--+ +--^ ^ | | | | v | | v +-----------+ | | +-----------+ | |<-------------------------+ | | | IdP1 | | | IdP2 | | | +------------------------>| | +-----------+ +-----------+ A federated call with IdP-based identity
  • 23. 23Connect | Communicate | Collaborate Offer (Fingerprint + Assertion) Origin: http://www.ietf.org/proceedings/82/slides/rtcweb-13.pdf
  • 24. 24Connect | Communicate | Collaborate Answer (Fingerprint+Assertion) Origin: http://www.ietf.org/proceedings/82/slides/rtcweb-13.pdf
  • 25. 25Connect | Communicate | Collaborate Codecs ● Audio ● Opus (royalty free, RFC 6176) , Opus 1.1 mobile ● iSAC (internet Speech Audio Codec) ● iLIBC (internet Low Bitrate Codec RFC 3951) ● G.711 (alaw/ulaw) ● Automatic Gain Control (AGC) ● Acoustic Echo Cancellation (AEC) ● Video ● VP8 Chrome, Firefox ● H.264 Browser(Ericsson Lab), (Firefox planed) ● Future HEVC/H.265 (SVC), VP9 (Vidyo&Google VP9 SVC) ● VoiceEngine, VideoEngine, NetEQ, AEC, etc. all stem from the GIPS acquisition
  • 26. 26Connect | Communicate | Collaborate Battle for Mandatory To Implement(MTI) Video Codec ● Battle for WebRTC mandatory to implement (MTI) codec ● Audio MTI codecs ● G.711 (alaw/ulaw) ● Opus ● Video (?!) ● Google ● Hangout H.264=>VP8 ● Chrome only VP8/VP9 support ● Cisco ● Cisco will open H.264 codec ● Cisco will pay MPEG LA ● Mozilla will support Cisco binary H.264 codec ● http://www.openh264.org/ ● video codec proposals, and backers ● VP8 (VP9) ● Google ● H.264 (H.265) ● Ericsson ● Nokia ● BlackBerry ● Qualcomm ● Orange ● Cisco ● Microsoft ● Apple ● Both has Pros & Cons
  • 27. 27Connect | Communicate | Collaborate Diagnostic / Interoperability ● Browser interoperability: http://www.webrtc.org/interop ● https://code.google.com/p/webrtc/source/browse/trunk/samples/js/base/adapter.js ● Check Network Connectivity: http://www.check-connectivity.com/ ● Developer / Diagnostic tool ● chrome://webrtc-internals ● Firefox planed
  • 28. 28Connect | Communicate | Collaborate Demonstrations: http://goo.gl/d3uftB
  • 29. 29Connect | Communicate | Collaborate Experience it. Demonstrations ● Cube Slam Chrome experiment Game https://www.cubeslam.com ● LifeSize demo http://www.lifesize.com/en/webrtc SIP URI call ● Magic Xylophone (motion detection) http://www.soundstep.com/blog/experiments/jsdetection/ ● getUserMedia Face Gestures http://shinydemos.com/facekat/ ● getUserMedia Filters http://webcamtoy.com/hu/app/ ● getUserMedia ASCII http://idevelop.ro/ascii-camera/ ● Pitch detect http://webaudiodemos.appspot.com/pitchdetect/index.html
  • 30. 30Connect | Communicate | Collaborate AppRTC (WebRTC reference application) ● https://apprtc.appspot.com/ ● Options: ● stereo=true ● hd=true ● debug=loopback ● video= ● audio= ● ss= (stun) ● st=(turn) ● For more parameters see: https://code.google.com/p/webrtc/source/browse/trunk/samples/js/apprtc/apprtc.py ● Loopback test https://apprtc.appspot.com/?r=52215035&hd=true&debug=loopback
  • 31. 31Connect | Communicate | Collaborate JsSIP ● http://tryit.jssip.net/ ● Use generated account, or use your own sip account ● You can follow SIP messages in JavaScript console
  • 32. 32Connect | Communicate | Collaborate Interesting Demonstrations ● Soundtrap https://www.soundtrap.com ● getUserMedia ● GetUserMedia Webcam controlled slides http://lli.web.fh-koeln.de/mocowe/ ● getUserMedia face tracking http://www.simpl.info/headtrackr/ ● getUserMedia constraints https://simpl.info/getusermedia/constraints/ ● getUserMedia + Web Audio http://www.webaudiodemos.appspot.com/AudioRecorder/index.html ● Screen Sharing/Capture https://html5-demos.appspot.com/static/getusermedia/screenshare.html https://simpl.info/screencapture/ ● Tab capture: chrome.tabCapture http://updates.html5rocks.com/2012/12/Screensharing-with-WebRTC
  • 33. 33Connect | Communicate | Collaborate More Demonstrations.. ● Face substitution http://auduno.github.io/clmtrackr/examples/facesubstitution.html (WebGL) ● PeerConnection ● PeerConnection simple vidconf demo http://www.simpl.info/rtcpeerconnection/ ● DataChannel ● P2P file share http://www.sharefest.me/ ● Simple data channel demo http://www.simpl.info/rtcdatachannel/ ● Muaz Khan Demos https://www.webrtc-experiment.com/ ● webrtc.org demos http://www.webrtc.org/demo ● Mozilla http://mozilla.github.io/webrtc-landing/
  • 34. 34Connect | Communicate | Collaborate Architecture Overview Image source: http://www.sloreto.com/slides/Aalto022013WebRTC/images/WebRTC_Architecture0.jpg
  • 35. 35Connect | Communicate | Collaborate Building Real World Service ● Component required to build a Service ● Web server ● Signaling server (WebSocket) ● ICE (NAT, Firewall Traversal) ● STUN server, TURN server ● Session Border Controller / Gateway (signaling/media) ● SIP proxy, XMPP server, H.323 gatekeeper ● Multipoint conference handling ● MCU media mixing CP(Continuous Presence) ● Conference Archiving ● AAI (IdP) ● Vendor Directory http://webrtchacks.com/vendor-directory/
  • 36. 36Connect | Communicate | Collaborate Multipoint (MCU)
  • 37. 37Connect | Communicate | Collaborate Multipoint ● Peer 2 Peer ● One to One ● Mesh ● Small N-way ● Focus Point / Star ● Medium N-way ● MCU / Mixer ● Large N-way ● Video Router ● Large N-way ● Simulcast, layered, scalable video coding support Image source: http://webrtchacks.com/webrtc-beyond-one-one/
  • 38. 38Connect | Communicate | Collaborate MCU, Gateway, SBC ● MCU ● WebRTC is about Peer2Peer ● So limited Multipoint capabilities ● WebRTC endpoint need an MCU for large N-way calls ● Gateway/SBC ● Interoperability ● RTP – SDES-SRTP – DTLS-SRTP – RTP ● Demultiplex – RTCP – Media channel ● SAVPF<=>AVP – RTCP feedback ● ICE(STUN/TURN) ● Security, SPIT ● Transcoding Video, Audio ● e.g. VP8 <=> H.264
  • 39. 39Connect | Communicate | Collaborate WebRTC MCU vendors ● Open Source ● http://www.medooze.com/products/mcu/functionality.aspx Argentinian universities VoIP workgroup has been using for about a year. http://www.youtube.com/watch?v=pocgfJXmwV4 (in Spanish) ● http://lynckia.com/ ● http://code.google.com/p/telepresence/ NIIFI tested ● Commercial ● http://www.requestec.com/site/platform/architecture.jsp ● http://acano.com/tour/ ● PEXIP http://www.pexip.com/requirements NIIFI tested Version 2 SRTP-DTLS (Version 3)
  • 40. 40Connect | Communicate | Collaborate WebRTC Frameworks ● SimpleRTC ● https://github.com/henrikjoreteg/SimpleWebRTC ● EasyRTC ● https://github.com/priologic/easyrtc ● webRTC.io ● https://github.com/webRTC/webRTC.io ● ShareFest ● https://github.com/peer5/sharefest ● PEERJS ● http://peerjs.com/
  • 41. 41Connect | Communicate | Collaborate Open Source components implementations ● IP PBX ● FreeSwitch ● SIP over WebSocket ● SRTP-DTLS (git version) ● video transcoding fs-video branch ● Asterisk ● SIP over WebSocket ● SIP Proxy ● Kamailio ● SIP over WebSocket ● OverSIP ● SIP over WebSocket ● RTP PROXY ● mediaproxy-ng ● STUN, TURN ● stunserver ● http://www.stunprotocol.org/ ● rfc5766-turn-server ● Gateway ● Doubango webrtc2sip (GW) ● JS client library ● SIPML5 ● JsSIP ● QoffeeSIP
  • 42. 42Connect | Communicate | Collaborate Big Blue Button (BBB) ● Current UI Flash Based streaming using ● Red5, FreeSwitch ● Lecture / videoconference ● Desktop Sharing ● Audio, Video ● Slides, blackboard, draw/highlight ● Chat, ● Participant list ● Recording ● HTML5 integration started ● Big Project, Community support ● 1.5K members of development mailing list ● Localized 35 languages ● HTML5 client ● implemented using coffeescript, require.js, backbone.js ● HTML5/WebRTC documentation https://code.google.com/p/bigblu ebutton/wiki/HTML5 ● Demo sever http://webrtc.bigbluebutton.org
  • 43. 43Connect | Communicate | Collaborate BBB & WebRTC Architecture
  • 44. 44Connect | Communicate | Collaborate WebRTC SWOT analysis ● SWOT analysis: ● Strengths ● Weaknesses ● Opportunities ● Threats
  • 45. 45Connect | Communicate | Collaborate SWOT: Strengths ● No plugins ● No Flash, Java, Silverlight etc. ● Client deployed everywhere ● No sw client install needed: ● 1000000000+ WebRTC endpoints ● Client is always up2date. (Browser auto updates) ● Multi Platform ● PC ● Phone, Tablet ● Security is mandatory ● peer-to-peer ● HD video ● Wideband audio ● E2E Security, Opt-in Privacy ● Open ● Open Source, Standards based ● Royalty Free (?) Nothing proprietary(?) ● Multimedia for Web ● Voice,Video (webcam, screencapture), Data ● Standard based Firewall/NAT traversal ● ICE (STUN/TURN) ● IPv6 and IPv4 negotiation, interoperability ● Media multiplexing ● WebRTC is part of HTML5 ● Web JS API is simple and hides complexity ● Implementations ● Browser, and native Java/C API
  • 46. 46Connect | Communicate | Collaborate SWOT: Weaknesses ● Early adopters phase not mature final standard (draft), ● Browser implementation compatibility issues ● Depends other sw infrastructure operations ● STUN/TURN server, MCU, Gateway ● AV Codec HW support (HW VP8 Android KitKat 720p) ● No MTI video Codec (H.264 vs. VP8) future (H.265 vs. VP9), Daala(?), Scalable video coding (SVC) ● RTCWEB Security Architecture not yet implemented. ● For a SAML based WebRTC security architecture implementation more research and development needed. ● Desktop sharing, statistics, DTMF, security architecture is not yet implemented in every browser ● Acoustic echo cancellation and noise suppression ● Backward compatibility issues, handling of low-bandwidth situations
  • 47. 47Connect | Communicate | Collaborate SWOT: Opportunities ● WebRTC Buzzword / Hype ● HTML5 (WebRTC) as an universal application platform. ● Disrupting communication market / Transforming Communication ● Transparent Standard based secure platform for RTC ● Alpha channels, blue box/ green screen ● New possibilities / New applications ● Games, Video support, Call centre, Lecture Recording, streaming ● Apps Mobile, Tablet /Android/ ● Collaborative music composing, etc. ● RTC (Videoconference and beyond) to anyone who has a browser ● Bridge between Telco and Web world ● Trusted, Open Source peer to peer communication ● AAI integration ● Next gen video codecs: e.g. VP9 (SVC) same quality cut bitrate in half.
  • 48. 48Connect | Communicate | Collaborate SWOT: Threats ● Backward Compatibility, WEBRTC implements leading edge IETF standards (current installed videoconference / telepresence room don't.) ● Browser implementation in every browser ● Internet Explorer, Safari ● Mobile adaptation (iOS, Android native Apps) ● Abstract signaling ● Endpoint / User Identification (URI, E.164, etc.) ● Communication Regulation, Legal Issues ● Lawful interception, Emergency calling, E.164 numbering etc. ● No mandatory signaling protocol. It could lead to Walled Gardens compatibility issues. ● Alternative APIs (ORTC, CU-RTC-WEB)
  • 49. 49Connect | Communicate | Collaborate Is WebRTC Ready? ● Yes! Ready to start experience it, and build a leading/bleeding edge pilot service. ● Simple video call in optimal case works between different implementations ● We have many demonstrations, and market players also adopting to it ● Early adopters build frameworks, new services. ● No.. It is not yet ready to build superior, reliable, real world service. ● Backward compatibility. Almost compatible, but only almost. ● IPv6 support implementation, Call setup delay (ICE / Trickle ICE) ● No MTI video, HW support, (SWOT Weakness..) ● What is missing for building real services (Justin Uberti) https://docs.google.com/document/d/1EBOnUXjIlEmYO0fRAtbW-woEcPKRuwmIIxVDhyPvaic/edit
  • 50. 50Connect | Communicate | Collaborate Summary & Prediction ● Multi platform, Standard based, Royalty free technology. designed security and identity management in mind, IPv6 support, and standard based Firewall/NAT traversal, etc. ● An emerging, young technology, a leading edge technology ● Still a lot of growing up to do ● Considerable impact to all RTC market players and Service Providers (Google, Cisco, Vidyo, LifeSize, Oracle, AT&T etc..) ● WebRTC is here! Act Now! Experience it, use it, improve it! ● It is stable enough to start build pilot services. ● If you like the idea to pilot an open source standard based WebRTC based RTC collaboration solution what use federated authentication and serve GÉANT community, then please support my GN4 NIF. ● WebRTC has arrived, choose the right open leading/bleeding edge way, and don't buy or support proprietary walled gardens any more! (Incompatible RTC vendor solutions, with vendor lock-in, etc.)
  • 51. 51Connect | Communicate | Collaborate Let's start to experience WebRTC Let's start making mistakes on WebRTC field “An expert is a person who has found out by his own painful experience all the mistakes that one can make in a very narrow field.” Niels Bohr
  • 52. 52Connect | Communicate | Collaborate Open Discussion ● Has Your NREN implemented WebRTC (or planning to implement a service based on it? How do you plan to use it? Videoconference / Streaming / other ? ● What is your opinion about Video Codec War? ● What do you prefer and why? (VP8 vs. H.264) ● Is it important to choose Mandatory To Implement (MTI) codec? ● WebRTC GN4 New Idea Form ● Please express your support if you like the idea, or comment it ● MTI functionality = ? ● AAI integration ● Questions, AoB ● Open Discussion
  • 53. www.geant.net www.twitter.com/GEANTnews | www.facebook.com/GEANTnetwork | www.youtube.com/GEANTtv Connect | Communicate | Collaborate Thank you!

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