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

1. WebRTC: been there, done that

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. 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

4. What does Quobis provide? KNOW-HOW
●
Consulting services
● Products

5. WebRTC standards
(Signaling)
(Signaling)
“Set or RTC APIs
for Web Browsers”
(Media)
“New protocol
profile”
Some discussion on the topic: http://webrtchacks.com/a-hitchhikersguide-to-webrtc-standardization/

6. WebRTC does not define signaling
Don’t panic, it’s not a bad thing!

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: http://webrtchacks.com/signallingoptions-for-webrtc-applications/

8. Takeaway (1/3):
each deployment/vendor is implementing
its own (proprietary) signaling approach

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. Takeaway (2/3):
WebRTC signaling and media is
incompatible with existing VoIP
deployments – gateways are required to
bridge the two worlds

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:
http://webrtchacks.com/why-thewebrtc-api-has-it-wrong-interview-withwebrtc-object-api-ortc-co-author-inakibaz-3-2/
iswebrtcreadyyet.com

12. Takeaway (3/3):
the WebRTC API can have different
flavors

13. WebRTC Client: SIPPO from Quobis
Signaling
agnostic.
Browser
agnostic.
API to build
your own
apps.

14. The BIG picture

15. 3GPP architecture (under discussion)
SIPPO Server = WebRTC Portal + more things
Third Party WebRTC-SIP gateway

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. 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. 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. MORE
INFORMATION
VICTOR PASCUAL
Chief Strategy Officer
victor.pascual@quobis.com
Twitter: @victorpascual