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Webrtc - rich communication - quobis - victor pascual


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Presentation on Quobis from Victor Pascual given at the WebRTC pre-workshop at Rich Communications in Berlin on 28th Oct 2013

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Webrtc - rich communication - quobis - victor pascual

  1. 1. WebRTC: been there, done that
  2. 2. About QUOBIS Quobis is a leading european company in the delivery of carrier-class unified communication solutions with a special focus on security, interconnection and identity management for service providers and enterprises. Seven years working on VoIP projects. Three years developing own products.
  3. 3. About Me Victor Pascual – Chief Strategy Officer (CSO) at Quobis Main focus: help make WebRTC happen – involved in WebRTC standardization, development and first industry deployments (on-going RFX's, PoC's and field trials) Side activities: - IETF contributor (SIP, Diameter and WebRTC areas) - IETF STRAW WG co-chair - SIP Forum WebRTC Task Group co-chair - co-founder and blogger - Independent Expert at European Commission
  4. 4. What does Quobis provide? KNOW-HOW ● Consulting services ● Products
  5. 5. WebRTC standards (Signaling) (Signaling) “Set or RTC APIs for Web Browsers” (Media) “New protocol profile” Some discussion on the topic:
  6. 6. WebRTC does not define signaling Don’t panic, it’s not a bad thing!
  7. 7. Signaling plane WebRTC has no defined signaling method. JavaScript app downloaded from web server. Popular choices are: ● SIP over Websockets – – Extend SIP directly into the browser by embedding a SIP stack directly into the webpage – typically based on JavaScript – WebSocket create a full-duplex channel right from the web browser – ● Standard mechanism (draft-ietf-sipcore-sip-websocket) – soon to be RFC Popular examples are jsSIP, sip-js, QoffeeSIP, or sipML5 Call Control API – – • proprietary signaling scheme based on more traditional web tools and techniques GSMA/OMA extending RCS “standard” API to include WebRTC support Other alternatives based on XMPP, JSON or foobar Some discussion on the topic:
  8. 8. Takeaway (1/3): each deployment/vendor is implementing its own (proprietary) signaling approach
  9. 9. Media plane (1/2) ● ● A browser-embedded media engine – Best-of-breed echo canceler – Video jitter buffer, image enhancer – Audio codecs – G.711, Opus are MTI – Video codecs – H.264 vs. VP8 (MTI TBD - IPR discussion) – Media codecs are negotiated with SDP (for now at least) – Requires Secure RTP (SRTP) – DTLS – Requires Peer-2-peer NAT traversal tools (STUN, TURN, ICE) – trickle ICE – Multiplexing: RTPs & RTP+RTCP Yes, your favorite SIP client implementation is compatible with most of this. But, the vast majority of deployments – – – – Use plain RTP (and SDES if encrypted) Do not support STUN/TURN/ICE Do not support multiplexing (ok, not really an issue) Use different codecs that might not be supported on the WebRTC side
  10. 10. Takeaway (2/3): WebRTC signaling and media is incompatible with existing VoIP deployments – gateways are required to bridge the two worlds
  11. 11. Media plane (2/2) How do applications access the WebRTC media engine in the web browser? ● W3C API – Currently working on 1.0 2.0: Backward compatibility? Competing API: CU-RTC-Web (Microsoft) Competing API: ORTC (Microsoft and others) Apple? – ● ● ● Some discussion on the topic:
  12. 12. Takeaway (3/3): the WebRTC API can have different flavors
  13. 13. WebRTC Client: SIPPO from Quobis Signaling agnostic. Browser agnostic. API to build your own apps.
  14. 14. The BIG picture
  15. 15. 3GPP architecture (under discussion) SIPPO Server = WebRTC Portal + more things Third Party WebRTC-SIP gateway
  16. 16. SIPPO Server: Control, provision, configure and customize your WebRTC Clients ● RESTful APIs for management of users and web clients ● Seven modules: Authentication, Authorization, Accounting, Contact mgmt, Branding, File sharing, Statistics. ● Connection to LDAP/AD for Authentication, Authorization and Contact Management. ● Integration with Facebook, Gmail, etc. ● Support for identity federation ● Diameter for integration with backend. ● Etc.
  17. 17. Sippo Web Collaborator Corporate endpoint fully-interoperable with SIP networks and 3rd party WebRTC gateways Main features: - Audio/video - Interactive chat - Presence - Contact list - File transfer - Screen sharing - Dialpad - etc.
  18. 18. VoiceInstant: WebRTC "Happy button" Contact Center Platform WebRTC gateway 2 1 The call is transferred to contact center application. • End user • Agent's can use the same client and applications 3 • Customer visiting the website clicks on "Contact us" button. • No need to enter any personal number or to install any software • Customer can also see the agent's video. • Agents can use its own softphone or SIPPO (a webRTC endpoint)
  19. 19. MORE INFORMATION VICTOR PASCUAL Chief Strategy Officer Twitter: @victorpascual