WebRTC: Standards Update – Where are we?
About QUOBIS
Quobis is a leading european company in the delivery of
carrier-class unified communication solutions with a
...
About Me (victor.pascual@quobis.com)
Victor Pascual – Chief Strategy Officer (CSO) at Quobis
Main focus: help make WebRTC ...
Technology Angle
A browser-embedded media engine

“No need to install upgrade/configure software”
Business Angle

Is WebRTC something disruptive or simply yet
another access framework? BOTH!
RTC → Web
Web → RTC

- global...
WebRTC standards

(Signaling)

(Signaling)

“Set or RTC APIs
for Web Browsers”

(Media)
“New protocol
profile”

Some discu...
RTCWeb WG (and others)

- Audio codecs – G.711, Opus
- Video codecs – H.264 vs. VP8
- Media codecs are negotiated with SDP...
WebRTC does not define signaling

Don’t panic, it’s not a bad thing!
Signaling plane
WebRTC has no defined signaling method. JavaScript app downloaded
from web server. Popular choices are:
●
...
(1/3)
each deployment/vendor is implementing
its own proprietary signaling mechanism
Interworking?
●

●

A browser-embedded media engine
– Best-of-breed echo canceler
– Video jitter buffer, image enhancer
– ...
(2/3)
WebRTC signaling and media is NOT
compatible with existing VoIP deployments
– gateways are required to bridge the tw...
The video codec battle

Some discussion on the topic: http://webrtchacks.com/cisco-openh264/
Result of the
discussion?
Room participants: 30/50 in favor of H.264
Remote participants (minority): 75/25 in favor of VP8...
No decision

Some discussion on the topic: http://webrtchacks.com/ietf-finally-made-decisionmandatory-implement-mti-video-...
WebRTC WG
“The mission of the W3C WebRTC WG is to define client-side APIs to enable Real-Time Communications in
Web-browse...
How do applications access the media engine?
●

W3C API
– Currently working on 1.0
2.0: Backward compatibility?
Competing ...
(3/3)
the WebRTC API can have different
flavors
WebRTC Access to IMS (r12)
SA1 (requirements): reusing IMS client security credentials and/or
public identities/credentials; how IMS clients communic...
SIP Forum WebRTC Task Group
“the initial focus of the Task Group is to determine what
the needs are for successful interop...
GSMA
Alliance for Telecom Solutions
”Device Solutions Initiative (DSI), an initiative that will host a range of projects to
dev...
WebRTC interop Activity Group

“focuses on interoperability issues relating to the use of WebRTC”
“the group is focused on...
Summary

●

●

●

each deployment/vendor is implementing its own
proprietary signaling mechanism
WebRTC signaling and medi...
MORE
INFORMATION
VICTOR PASCUAL
Chief Strategy Officer
victor.pascual@quobis.com
Twitter: @victorpascual
MORE
INFORMATION
BACKUP SLIDES
WebRTC client app: SIPPO from Quobis

Corporate endpoint fully-interoperable with
SIP networks and 3rd party WebRTC gatewa...
How to make things work?
Reference Architecture
SIPPO: Client + Server component
3GPP architecture (under discussion)

SIPPO Server = WebRTC Portal + more things

Third Party WebRTC-SIP gateway
SIPPO Server: Control, provision, configure and
customize your WebRTC Clients

● RESTful APIs for management of users and ...
WebRTC standards update (13 Nov 2013)
WebRTC standards update (13 Nov 2013)
WebRTC standards update (13 Nov 2013)
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WebRTC standards update (13 Nov 2013)

  1. 1. WebRTC: Standards Update – Where are we?
  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@quobis.com) 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 - WebRTCHacks.com co-founder and blogger - Independent Expert at European Commission @victorpascual
  4. 4. Technology Angle A browser-embedded media engine “No need to install upgrade/configure software”
  5. 5. Business Angle Is WebRTC something disruptive or simply yet another access framework? BOTH! RTC → Web Web → RTC - global business – browsers are connected to the Internet → it's time to go OTT Pure Web vs Interworked Not only Web browsers but also native support via apps or OS (e.g. set-top boxes, FirefoxOS) - expand footprint – extend existing services to new subscribers - decrease churn – enhance current services to existing subscribers - new service revenues – create new services and subscribers
  6. 6. WebRTC standards (Signaling) (Signaling) “Set or RTC APIs for Web Browsers” (Media) “New protocol profile” Some discussion on the topic: http://webrtchacks.com/a-hitchhikers-guide-to-webrtc-standardization/
  7. 7. RTCWeb WG (and others) - Audio codecs – G.711, Opus - Video codecs – H.264 vs. VP8 - Media codecs are negotiated with SDP (for now at least) - Requires Secure RTP (SRTP) – DTLS-SRTP (SDES is prohibited) - Requires Peer-2-peer NAT traversal tools (STUN, TURN, ICE) – trickle ICE - Multiplexing: RTPs & RTP+RTCP - Tools for firewall traversal - DataChannel - Etc. NEW PROTOCOL PROFILE FOR MEDIA
  8. 8. WebRTC does not define signaling Don’t panic, it’s not a bad thing!
  9. 9. 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: http://webrtchacks.com/signalling-options-for-webrtcapplications/
  10. 10. (1/3) each deployment/vendor is implementing its own proprietary signaling mechanism
  11. 11. Interworking? ● ● 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 at all) 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
  12. 12. (2/3) WebRTC signaling and media is NOT compatible with existing VoIP deployments – gateways are required to bridge the two worlds
  13. 13. The video codec battle Some discussion on the topic: http://webrtchacks.com/cisco-openh264/
  14. 14. Result of the discussion? Room participants: 30/50 in favor of H.264 Remote participants (minority): 75/25 in favor of VP8 → No clear consensus
  15. 15. No decision Some discussion on the topic: http://webrtchacks.com/ietf-finally-made-decisionmandatory-implement-mti-video-codec-webrtc/
  16. 16. WebRTC WG “The mission of the W3C WebRTC WG is to define client-side APIs to enable Real-Time Communications in Web-browsers. These APIs should enable building applications that can be run inside a browser, requiring no extra downloads or plugins, that allow communication between parties using audio, video and supplementary real-time communication, without having to use intervening servers (unless needed for firewall traversal).” Obtain local media Setup Peer Connection Attach media or Data Close Connection ← getUserMedia(), etc. ← RTCPeerConnection(), etc. ← addStream(), createOffer(), etc. Discussion: provides the current API in its form (e.g. based on SDP O/A) the flexibility Web developers need? Answer: well, not really but it's good enough for most of the use cases we have today Competing proposals: Microsoft's CURTC-WEB (Aug'12), WebRTC Object API (ORTC) (Aug'13) Next step: “Done is better than perfect”, Let's finish WebRTC 1.0, Let the industry adopt it Future work: “fix/improve things in WebRTC 2.0”, Backward interoperability?
  17. 17. How do applications access the media engine? ● W3C API – Currently working on 1.0 2.0: Backward compatibility? Competing API: CU-RTC-Web (Microsoft) Competing API: ORTC (Microsoft and others) Apple? Since last week Opera includes some support – ● ● ● ● Some discussion on the topic: http://webrtchacks.com/why-the-webrtc-apihas-it-wrong-interview-with-webrtc-object-apiortc-co-author-inaki-baz-3-2/ iswebrtcreadyyet.com
  18. 18. (3/3) the WebRTC API can have different flavors
  19. 19. WebRTC Access to IMS (r12)
  20. 20. SA1 (requirements): reusing IMS client security credentials and/or public identities/credentials; how IMS clients communicate with WebRTC clients connected to IMS; IMS services to the WebRTC client; regulatory functions and charging; offer IMS services to users interacting with a 3rd party website, etc. SA2 (architecture): expand the IMS architecture and stage 2 procedures as required by the support of WebRTC clients access to IMS; media plane aspects; PBX emulation; signalling; only UNI covered, NNI out of scope. SA3 (security): WebRTC client authentication mechanisms, media plane security
  21. 21. SIP Forum WebRTC Task Group “the initial focus of the Task Group is to determine what the needs are for successful interoperability of WebRTCto-SIP deployments” covering both Enterprises and Service Providers “recommendations, Reference Architecture Documents, Certifications, and/or White Papers”
  22. 22. GSMA
  23. 23. Alliance for Telecom Solutions ”Device Solutions Initiative (DSI), an initiative that will host a range of projects to develop network- and protocol-agnostic, client-side bindings that will make real-time communications more accessible to web developers” “The tools being created by the Initiative will offer developers a code-once, run anywhere approach, replacing the need for carrier-specific coding to add functions such as basic call signaling and network control to applications. DSI will focus initially on call signaling but is expected to advance from there to address other network-specific functions” “The DSI’s formation and launch was led by Alcatel-Lucent, AT&T, CenturyLink, Ericsson, Sprint and Verizon” “The first project under the DSI has already started. In July, ATIS launched ORCA, which stands for Open Real-Time Communications APIs, an open source project that will mask the complexity of end-to-end signaling for real-time communication, allowing developers to focus on embedding innovative new functionality to their applications, such as high-quality voice and/or video calls. ORCA has successfully developed initial client-side software called orca.js, with the “.js” denoting creation of a JavaScript library. The client-side binding is network- and protocol- independent, linking to supporting implementation libraries to allow developers consistent access to the robust services provided by IMS networks”
  24. 24. WebRTC interop Activity Group “focuses on interoperability issues relating to the use of WebRTC” “the group is focused on enterprise WebRTC , interworking of WebRTC and other carrier technologies, and other existing videoconferencing systems” “develop an interoperability test framework and prepare for IOT events”
  25. 25. Summary ● ● ● each deployment/vendor is implementing its own proprietary signaling mechanism WebRTC signaling and media is incompatible with existing VoIP deployments – gateways are required to bridge the two worlds the WebRTC API can have different flavors
  26. 26. MORE INFORMATION VICTOR PASCUAL Chief Strategy Officer victor.pascual@quobis.com Twitter: @victorpascual
  27. 27. MORE INFORMATION BACKUP SLIDES
  28. 28. WebRTC client app: SIPPO from Quobis 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. Signaling agnostic - Browser agnostic - API to build your own apps.
  29. 29. How to make things work?
  30. 30. Reference Architecture SIPPO: Client + Server component
  31. 31. 3GPP architecture (under discussion) SIPPO Server = WebRTC Portal + more things Third Party WebRTC-SIP gateway
  32. 32. 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.

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