Upperside WebRTC conference - WebRTC intro

1. WebRTC Intro and Standards update (December 2014)
Victor
Pascual
Avila
Victor.Pascual@quobis.com
@victorpascual

2. About Me
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
- Adjunct Professor at Universitat Pompeu Fabra

3. Disclaimer:
All
opinions
and
nonsense
expressed
are
my
own
3

4. 4

5. 12/15/14
5

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

7. 7

8. Any good defini,on
for ‘media engine’?
8

9. 9

10. Once upon a ,me…
10

11. 11

12. 12

13. 13

14. 14

15. 15

16. 16

17. 17

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

19. RTCWeb WG (and other)
- Audio codecs – G.711, Opus
- Video codecs – H.264, VP8 (oh, well…)
- 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

20. RTCWeb WG

21. 21

22. 22

23. 23

24. 24

25. 25

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

27. 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/

28. 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

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

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

32. 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/
November
2013

33. 33

34. 34

35. 35
hJp://blogs.cisco.com/collaboraLon/ciscos-­‐openh264-­‐
now-­‐part-­‐of-­‐firefox/

36. 36

37. 37

38. 38

39. 39

40. 40

41. 41

42. 42

43. 43

44. 44
etc.

45. 45

46. Photo
taken
by
Dan
York
46

47. Consensus!
47
November
2014

48. But…
48

49. 49
WebRTC-­‐compaLble

50. Consensus!
50

51. VP9
vs.
H.265
51

52. 19
Nov
2014
52

53. 53

54. 54

55. 55

56. 56

57. 57
“I
also
do
not
support
the
dra-
proposal
for
the
following
reasons:
1.
The
proposal
imposes
a
dual
MTI
obligaLon
on
devices
and
applicaLons.
We
should
expect
this
recommendaLon
to
be
widely
ignored
in
pracLce
because
it
offers
no
interoperability
benefits,
while
increasing
footprint
and
adding
unnecessary
liability/fees.
The
RFC
6919
"MUST
(BUT
WE
KNOW
YOU
WON'T)"
is
appropriate
here.
DefiniLon:
The
phrase
"MUST
(BUT
WE
KNOW
YOU
WON'T)"
is
used
to
indicate
requirements
that
are
needed
to
meet
formal
review
criteria
(e.g.,
mandatory-­‐to-­‐implement
security
mechanisms),
when
these
mechanisms
are
too
inconvenient
for
implementers
to
actually
implement.
2.
The
codec
standardizaLon
process
is
important
from
a
legal
and
IPR
disclosure
point
of
view.
VP8
has
not
yet
completed
this
process.”
“I
don’t
support
the
proposed
text
either”
“we
can
not
support
this
dra-,
due
to
the
existence
of
“unwilling
to
license"
declaraLon
on
a
technology
required
for
compliance”

58. Consensus?
58

59. 59

60. 60

61. iswebrtcreadyyet.com
Browser Support

62. Wait
aier
the
coffee
break!
62

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

64. Adding New Wheels to IMS
with WebRTC

65. 3GPP TS 23.228 V12.5.0
(2014-06)

66. 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

67. 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

68. 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”

69. Alliance for Telecom Solutions

70. 70

71. 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”

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

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

74. This
slide
might
be
outdated
74

75. 75

76. hJp://www.gsma.com/newsroom/wp-­‐content/uploads//WebRTC-­‐Whitepaper-­‐v13.pdf
76

77. Thank You!
Victor
Pascual
Avila
Victor.Pascual@quobis.com
@victorpascual