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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MORE INFORMATION

VICTOR PASCUALChief Strategy Officer

victor.pascual@quobis.comTwitter: @victorpascual