WebRTC allows for real-time communication through peer-to-peer connections for voice, video, and data directly in web browsers. It uses open standards and does not require any plugins. WebRTC uses protocols like STUN, TURN, and ICE for NAT traversal and uses SRTP for secure media transmission. Signaling is required to coordinate between peers, which can use protocols like SIP, XMPP, or WebSockets. Popular codecs used in WebRTC include VP8 for video and Opus for voice. WebRTC is supported on over 4 billion devices by 2016 and enables many applications including video calling, remote assistance, and game/desktop streaming directly in web browsers.

1. WebRTC Overview
RouYun Pan
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2. •What is
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3. WebRTC is …?
• WebRTC offers users the ability to conduct a real-time
peer-to-peer communication for vioice, video and data.
• Today, WebRTC is still a work in progress.
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4. History
- Feb, 2010
Google acquire ON2 Technologies for $124 million, and then release the video
engine(VP8).
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- May 2010
Google acquire Global IP Solutions(GIPS) for $68 million, and then release the
source code about audio engine and network.
- Oct 2011
First Public Working Draft - W3C
- Feb 2012
WebRTC Native APIs 2.0
- June 2012
WebRTC Session at Google I/O
- Feb 2013
Firefox and Chrome interoperation achieved

5. What does Webrtc provide?
• Open Source, no royalties, license fees
• Real-time flexible voice, video & data
framework in cross platform
• Standard Web APIs Interoperable between
browsers
• No proprietary plug-in
• Security
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6. Low entry barriers
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P2P VoIP
WebRTC
PSTN
Entry barrier: complexity
Time
VoIP
 Circuit-switched
 Electric gear
 Dedicated lines
 SIP, IP-based
 Somewhat interoperable
 IMS core (for carriers)
 Complex systems
 Pure IP
 Peer-to-peer (P2P)
 Need client software
 „Walled garden“
 HTML5
 No plugin needed
 No client software
 Fully interoperable

7. Standardization
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IETF
*RTCWEB WG formed after BOF at IETF 80, April 2011
*Focus on protocols and interoperability.
W3C
*W3C WEBRTC WG created May 2011
*High level APIs and device control (mic, camera, network)
*PeerConnection API proposal originally proposed in WHATWG
currently being discussed:
http://dev.w3.org/2011/webrtc/editor/webrtc.html

8. WebRTC supported on >4bn devices by 2016
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9. What’s inside WebRTC
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10. For developer
• It is built into browsers and Using SDKs and APIs
of WebRTC can be integrated into Android and
iOS apps
– Session management
– Codec handling
– Peer to peer communication
– Security
– Bandwidth estimation
– Signaling and backend are not part of WebRTC
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11. Peer to peer, Server still be required?
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Client A Client B

12. Webrtc need these severs
• Signaling Server
• ICE Servers
• Media Servers (optional)
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13. Signaling plane
• Signaling is the process of coordinating
communication. In order for a “WebRTC Call”, its
clients may need to exchange information:
– Session control messages used to open or close
communication.
– Error messages.
– Media metadata such as codec settings, bandwidth
and media types.
– Key data, used to establish secure connections.
– Network data, such as a host's IP address and port as
seen by the outside world.
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14. • SIP/SDP
• XMPP/Jingle
• Websockets
• XHR/Comet
Signaling option in WebRTC
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15. For Example: SDP(conti.)
Session Origin Information
Network Information
Audio Information
ICE Candidate for audio
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16. For Example: SDP(conti.)
• http://tools.ietf.org/id/draft-nandakumar-rtcweb-sdp-01.html#rfc.section.5
Indicates NACK RTCP feedback support
Video information
ICE Candidate for video
RTCP setting
data channel information
ICE Candidate for data
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17. WebRTC Signaling triangle
PeerConnection(audio, video and/or data)
Web/Signaling server
Client A Client B
Signaling Signaling
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18. Webrtc Signaling trapezoid
Peerconnection(audio,and video and/or data)
Server A
Client A
Client B
Server B
Jingle or Sip
Signaling
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Signaling

19. WebRTC & SIP
PeerConnection(audio and/or video)
Server A
Client A SIP Phone
SIP Server
sip
sip
Signaling
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20. WebRTC & Jingle
PeerConnection(audio and/or video)
Server A
Client A
Jingle client
XMPP Server
Jingle
jingle
Signaling
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21. WebRTC & PSTN
PeerConnection(audio)
Server
Client A Phone BGateway
Signaling sip
analog
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22. WebRTC protocol
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23. RFC Documents
• ICE: Interactive Connectivity Establishment (RFC 5245)
– STUN: Session Traversal Utilities for NAT (RFC 5389)
– TURN: Traversal Using Relays around NAT (RFC 5766)
• SDP: Session Description Protocol (RFC 4566)
• XMPP: Extensible Messaging and Presence Protocol
(RFC 3921)
• DTLS: Datagram Transport Layer Security (RFC 6347)
• SCTP: Stream Control Transport Protocol (RFC 4960)
• SRTP: Secure Real-Time Transport Protocol (RFC 3711)
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24. For example: Secure pathways
Data(SCTP)
Web/Signaling server
Client A Client B
Audio/video(SRTP)
Signaling(https) Signaling(https)
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25. NAT traversal
Client A NAT NAT
Signaling Signaling
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26. Interactive Connectivity Establishment (ICE)
• A framework for connecting peers, it tries to find
the best path for each call.
– Direct
– STUN (Session Traversal Utilities for NAT)
– TURN (Traversal Using Relays around NAT)
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27. STUN Server
Client A NAT NAT
Signaling Signaling
Stun server
Media
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28. TURN Server
Client A NAT NAT
Signaling Signaling
Stun server
Media
Turn server
Media
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29. Media engine
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30. VoiceEngine
• OPUS (RFC6716)
• G.711(RFC3551)
• NetEQ for Voice
• Acoustic Echo Canceler
• Noise Reduction
* 8 kHz to 48 kHz
* Bitrate is about 6- 510 Kbps
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31. VideoEngine
• VP8(RFC6386)
• Video Jitter Buffer & Packet Loss
• Image enhancements
*1080P at 30 FPS: 2.5+ Mbps
*720p at 30 FPS: 1.0~2.0 Mbps
*360p at 30 FPS: 0.5~1.0 Mbps
*180p at 30 FPS: 0.1~0.5 Mbps
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32. Set up a call
Applicaption PeerConnectionfactory
PeerConnectionfactory()
CreatLocolMediaStream()
CreatLocolVideoTrack()
CreatLocolAudioTrack()
(add the tracks to stream)
AddSream()
PeerConnection
CommitStreamChanges()
OnSingalingMessage() - Offer
Get Answer from the remote peer
Remote Peer
Send Offer to the remote peer
Media
OnAddSream()
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33. Receive a call
Applicaption PeerConnectionfactory
CreatLocolMediaStream()
CreatLocolVideoTrack()
CreatLocolAudioTrack()
(add the tracks to stream)
AddSream()
PeerConnection
CommitStreamChanges()
Send Answer to the remote peer
Remote Peer
Reciever Offer from the remote peer
ProcessingSingalingMessage() - Offer
Media
OnSinglingMessages() - answer
PeerConnectionfactory()
OnAddStream()
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34. Close a call
Applicaption
Close
PeerConnection
OnStateChanges()
Get OK from the remote peer
Remote Peer
Send Shutdown to the remote peer
RemoveStream()
OnSignalingMessgae() - Shutdown
ProcessingMessage() - OK
OnStateChanges()
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35. Comparison with VoIP
Classic VoIP WebRTC
Signaling SIP or H.323 Undefined
Media transport RTP/RTCP RTP/RTCP
Security SRTP in SIP
H.235 in H.323
SRTP
NAT traversal STUN/TURN/ICE in SIP
H.450.x in H.323
STUN/TURN/ICE
Video codec H.263, H.264 VP8
Voice codec G.7xx series G.711, Opus, iLAB, iSAC
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36. What can we do with WebRTC?
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37. Technical support
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38. Home health care
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39. Game streaming
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40. Spy Camera? Wearable device
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41. Q&A
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