A Webinar by Victor Pascual Avila and Amir Zmora about WebRTC standards. IETF and W3C work on WebRTC as well as interworking with other networks such as IMS. The Webinar also talks about WebRTC signaling options and video codecs.

1. WebRTC standards update (July 2014)
Victor Pascual Avila
Victor.pascual@quobis.com
@victorpascual
Amir Zmora
amzmora@gmail.com
@AmirZmora

2. Upperside WebRTC
Conference, Dec 16-18
7/15/2014 2

3. New WebRTC Blog
7/15/2014 3

4. New WebRTC Blog
7/15/2014 4
blog.uppersideconferences.com

5. Technology, Innovation & Strategy Consultant
Main focus: help make WebRTC happen – involved in WebRTC
standardization, development and first industry deployments (on-going RFX's,
PoC's and field trials)
Other activities:
- Chief Strategy Officer (CSO)
- 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
About Me

6. WebRTC is…..?

7. Oetting
WebRTC is…..?

8. Oetting
WebRTC is…..?
• A media engine in the browser with JavaScript APIs
• Open Source, no royalties, license fees
• Real-time voice, video & data
• No plug-in
• Secured
• Standard interfaces

9. 周小逸 Ian

11. - 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
RTCWeb WG (and other)

12. RTCWeb WG

13. Don’t panic, it’s not a bad thing!
WebRTC Doesn’t Define
Signaling

14. Some discussion on the topic: http://webrtchacks.com/signalling-options-for-webrtc-
applications/
Signaling Plane
• WebRTC has no defined signaling method.
JavaScript app downloaded from web server.
Popular choices are:
• SIP over Websockets
 Standard mechanism (RFC7118)
 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
 “standard” APIs enhanced to include WebRTC support
• Other alternatives based on XMPP, JSON or foobar

15. each deployment/vendor is implementing
its own proprietary signaling mechanism
(1/3)

16. Interworking Towards Legacy?
• 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

17. WebRTC signaling and media is NOT
compatible with existing VoIP/IMS
deployments – gateways are required to
bridge the two worlds
(2/3)

18. Some discussion on the topic: http://webrtchacks.com/cisco-openh264/
The Video Codec Battle

20. Poll Question

21. Room participants: 30/50 in favor of H.264
Remote participants (minority): 75/25 in favor of VP8
→ No clear consensus
No decisionSome discussion on the topic: http://webrtchacks.com/ietf-finally-made-decision-
mandatory-implement-mti-video-codec-webrtc/
Result of The Discussion?

22. “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).”
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
Alternative proposals: Microsoft's CU-
RTC-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?
Obtain
local
media
Setup Peer
Connection
Attach
media
or Data
Close
Connection
← getUserMedia(),
etc.
← RTCPeerConnection(),
etc.
← addStream(),
createOffer(),
etc.
WebRTC WG

23. iswebrtcreadyyet.com
Browser Support

24. Poll Question

25. 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/
Browser API

26. http://status.modern.ie/
Browser API

27. http://status.modern.ie/
Browser API

28. Browser API

29. Plug-in free or free plug-in?
7/15/2014 29

30. the WebRTC API can have different flavors
(3/3)

31. http://webrtchacks.com/ims-approach-webrtc/
WebRTC Access to IMS (r12)

32. Adding New Wheels to IMS
with WebRTC

33. 3GPP TS 23.228 V12.5.0
(2014-06)

34. P
C
E
F
N
A
T
I
P
-
C
A
N
WWSF
W1
W2
UE
WIC
I/S-CSCF
eIMS-AGW
Iq
Mw
eP-CSCF
H/V-PCRF
Gx
Rx
W3
IMS-ALG
WAFW4
W5
Reference Architecture

35. codec1
SRTP
IP IP
UDP
IP
UDP UDP UDP
IP
UE eIMS-AGW peer
SRTP RTP
codec1 codec2
RTP
codec2
BFCP
SCTP
DTLS
IP
SCTP
DTLS
IP
TCP
IP
UDP UDP
BFCP
TCP
IP
UE eIMS-AGW peer
MSRP
SCTP
DTLS
IP
MSRP
SCTP
DTLS
IP
MSRP
TCP
IP
UDP UDP
MSRP
TCP
IP
UE eIMS-AGW peer
Interworking Towards Legacy
IMS

36. “the initial focus of the Task Group is to determine what
the needs are for successful interoperability of
WebRTC-to-SIP deployments” covering both
Enterprises and Service Providers
“recommendations, Reference Architecture Documents,
Certifications, and/or White Papers”
SIP Forum WebRTC Task
Group

37. Alliance for Telecom Solutions

38. “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”
WebRTC Interop Activity
Group

39. How does WebRTC relate to VoLTE
and RCS?
GSMA

40.  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
Summary

41. Thank You!
Victor Pascual Avila
Victor.pascual@quobis.com
@victorpascual
Amir Zmora
amzmora@gmail.com
@AmirZmora