Merck Moving Beyond Passwords: FIDO Paris Seminar.pptx
WebRTC introduction
1.
2. WHAT IS WEBRTC?
Web Browsers with Real-Time-Communication
A new API for embedding real-time communications into websites and
browser-based applications.
Become standard capabilities of the modern web browsers.
3. WEBRTC
• Accessed through
JavaScript API
• No
plugins, downloads or
installs
• Cross browsers and
platforms (Chrome
and Firefox)
• Audio/Video
conference
• File/Screen Sharing
• Speech Recognition
• Voice/Text translation
• Audio/Video recording
• Broadcast
• Integrate with PSTN or
mobile networks to
receive calls on any
device
10. IF YOU WANT TO KNOW MORE
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WebRTC client requirements
Codecs
Enterprise level architecture
Online resources
11. WEBRTC CLIENT REQUIREMENTS
Port Requirements
To use WebRTC please ensure that the following ports
are not blocked.
TCP: port 80 and 443
UDP: all ports between 10,000 and 60,000
Minimum Bandwidth Requirements
Upload: 8 kilobytes/second
Download: 8 kilobytes/second
12. WHAT CODECS ARE SUPPORTED?
The currently supported voice codecs are
Opus, G.711, G.722, iLBC, and iSAC, and VP8 is the supported
video codec. The list of supported codecs may change in the
future.
Opus is both preferred audio codec in Chrome and Firefox now.
14. ENTERPRISE LEVEL ARCHITECTURE
ICE servers (mandatory)
WebRTC is bound to use
ICE servers because we
need to make sure users'
Firewalls MUST NOT
block UDP or TCP ports.
Also we need to fallback
to relaying ICE server to
traverse NATs.
15. ENTERPRISE LEVEL ARCHITECTURE
Service capable to
capture and process
RTP packets e.g.
transcode, record or
merge them on server
end.
Capable to handle huge
bandwidth/relay stream
over peers.
Media servers (optional)