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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: been there, done that
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.
About QUOBIS
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- WebRTCHacks.com co-founder and blogger- Independent Expert at European Commission
What does Quobis provide? KNOW-HOW
● Consulting services● Products
WebRTC standards
(Media)
(Signaling)
(Signaling)
“Set or RTC APIs for Web Browsers”
“New protocol profile”
Some discussion on the topic: http://webrtchacks.com/a-hitchhikers-guide-to-webrtc-standardization/
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:
● SIP over Websockets
– Standard mechanism (draft-ietf-sipcore-sip-websocket) – soon to be RFC
– 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
– 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-webrtc-applications/
Takeaway (1/3):each deployment/vendor is implementing its own (proprietary) signaling approach
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
Takeaway (2/3):WebRTC signaling and media is incompatible with existing VoIP
deployments – gateways are required to bridge the two worlds
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?
iswebrtcreadyyet.com
Some discussion on the topic: http://webrtchacks.com/why-the-webrtc-api-has-it-wrong-interview-with-webrtc-object-api-ortc-co-author-inaki-baz-3-2/
Takeaway (3/3):the WebRTC API can have different
flavors
WebRTC Client: SIPPO from Quobis
Signaling agnostic.
Browser agnostic.
API to build your own apps.
The BIG picture
3GPP architecture (under discussion)
Third Party WebRTC-SIP gateway
SIPPO Server = WebRTC Portal + more things
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.
Sippo Web Collaborator
Main features:
- Audio/video
- Interactive chat
- Presence
- Contact list
- File transfer
- Screen sharing
- Dialpad
- etc.
Corporate endpoint fully-interoperable with SIP networks and 3rd party WebRTC gateways
VoiceInstant: WebRTC "Happy button"
WebRTC gateway
End user
Contact Center Platform
• Customer visiting the website clicks on "Contact us" button.
• No need to enter any personal number or to install any software
• The call is transferred to contact center application.
• Agent's can use the same client and applications
• Customer can also see the agent's video.
• Agents can use its own softphone or SIPPO (a webRTC endpoint)
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