WebRTC standards update - November 2014

1. WebRTC standards update (November 2014)
Victor
Pascual
Avila
Victor.Pascual@quobis.com
@victorpascual
Antón
Román
Portabales
Anton.Roman@quobis.com
@antonroman

2. 2

3. 3
So here we are today…

4. 4
@victorpascual
@antonroman
#sthlmwebrtc

5. About Victor
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
- Associate Professor at Universitat Pompeu Fabra

6. About Anton
• CTO
@
Quobis:
involved
in
first
industry
deployments
(on-­‐going
RFX’s,
PoCs
and
field
trials).
• Background
in
SIP
and
IMS.
• GeVng
involved
in
IETF
acXviXes.
• Involved
in
more
than
80
WebRTC
PoC
with
operators,
call
centers,
gateway
vendors
and
open
source
soluXons.
• Reviewer
of
some
WebRTC
books
and
author
of
some
posts
at
webrtchacks.com
6

7. What is WebRTC?
A browser-embedded media engine

8. WebRTC standards
(Signaling)
(Media)
(Signaling)
“Set or RTC APIs
for Web Browsers”
“New protocol
profile”

9. RTCWeb WG (and other)
- Audio codecs – G.711, Opus
- Video codecs – H.264, 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

10. RTCWeb WG

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

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18. 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 f00b4r
Some discussion on the topic: http://webrtchacks.com/signalling-options-for-webrtc-applications/

19. Interworking Towards Legacy VoIP?
• 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

20. WebRTC signaling and media is NOT
compatible with existing VoIP/IMS
deployments – gateways are required to
bridge the two worlds

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

23. 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
No decision
Some discussion on the topic: http://webrtchacks.com/ietf-finally-made-decision-mandatory-
implement-mti-video-codec-webrtc/

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26. 26
hap://blogs.cisco.com/collaboraXon/ciscos-­‐openh264-­‐
now-­‐part-­‐of-­‐firefox/

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35. Photo
taken
by
Dan
York
35

36. Fresh News! (from last night -­‐-­‐ yup, we
woke up at 3am)
36

37. But…
37

38. VP9
vs.
H.265
38

39. 39

40. 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).”
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 or even 1.1”, Backward
interoperability?
Obtain
local
media
Setup Peer
Connection
Attach
media
or Data
Close
Connection
← getUserMedia(),
etc.
← RTCPeerConnection(),
etc.
← addStream(),
createOffer(),
etc.

41. iswebrtcreadyyet.com
Browser Support

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44. Browser API
1.1
2.0
?
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/

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

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

47. Browser API

48. Plug-­‐in free or free plug-­‐in?
48

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55. 55
hap://webrtchacks.com/ortc-­‐qa-­‐robin-­‐raymond/

56. 56
hap://blog.webrtc.is/2014/07/01/google-­‐
chrome-­‐38-­‐39-­‐to-­‐ship-­‐with-­‐ortc-­‐webrtc-­‐1-­‐1-­‐apis/
Well…
not
really

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58. 58
ORTC editor

59. 59
WebRTC
editor

60. 11/17/14
60

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

62. Adding New Wheels to IMS
with WebRTC

63. 3GPP TS 23.228 V12.5.0
(2014-06)

64. Reference Architecture
P
C
E
F
N
A
T
I
P
-
C
A
N
WWSF
W1
W2
WIC
UE
I / S - CSCF
W4 WAF
W5
eP - CSCF Mw
Iq
eIMS - AGW
H / V - PCRF
Gx
Rx
W3
IMS - ALG

65. Interworking Towards Legacy
IMS
codec 1
SRTP
BFCP
SCTP
DTLS
UDP UDP
IP
codec 1 codec 2
SRTP RTP
TCP
IP
codec 2
RTP
SCTP
DTLS
IP
UDP UDP UDP
IP IP
UDP
IP
IP
UE eIMS - AGW peer
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

66. SIP Forum WebRTC Task
Group
“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”

67. Alliance for Telecom Solutions

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69. 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”

70. OMA
RESTful Network API for WebRTC Signaling v.1.0 (Feb 2014)

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

72. This
slide
might
be
outdated
72

73. This
slide
might
be
outdated
73

74. 74

75. hap://www.gsma.com/newsroom/wp-­‐content/uploads//WebRTC-­‐Whitepaper-­‐v13.pdf
75

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77. Thank You!
Victor
Pascual
Avila
Victor.Pascual@quobis.com
@victorpascual
Antón
Román
Portabales
Anton.Roman@quobis.com
@antonroman