WebRTC allows direct peer-to-peer communication between browsers without plugins. It uses technologies like DTLS and SRTP for secure media, and ICE and TURN for network traversal through NATs and firewalls. The WebRTC API defines a JavaScript API and SDP standard for applications to establish sessions between peers. While WebRTC does not specify a signaling protocol, it is commonly used with SIP via gateways to connect to SIP networks and devices. WebRTC has many applications beyond just calls, including games, dating sites, and transferring arbitrary data directly in browsers.
2. Introduction to
WebRTC
Browser to Browser communication
V 1.2 oej@edvina.net 2014-11-07
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3. The idea
Browers can play
audio and video. Why not let them use
the webcam and the
microphone?
If so, users can talk to
each other using the
browser.
Using NO PLUGINS
10. Using a turn server
TURN
on Internet IP
TURN
on Internet IP
11. WebRTC API
JavaScript API defined by
the W3C
User needs to allow usage
of camera and
microphone.
Defines the session in a
SDP document
The applications needs to
provide the SDP exchange
between peers
12. The WebRTC protocols
Defined by the IETF
No signalling protocol
specified
Use SRTP for media
Use ICE (TURN, STUN) for
network discovery
13. WebRTC media
Can be bundled and
mixed to one single
RTP port
Always use secure
RTP with DTLS key
exchange
Use ICE to find the
best media path
Use TURN to traverse
NAT and firewalls.
15. P2P call setup
with network issues
Web
Server
Media
SDP SDP
The web application defines the signaling
TURN
Server
TURN
Server
16. P2P call setup
with Changing networks
Wifi
Media
If a device moves from one network
to another during a session,"
a turn server can handle session
handover to the new network.
TURN
Server
TURN
Server
4G
18. Compatibility with SIP
WebRTC use a lot of updates to
SDP and RTP.
Most old SIP clients and servers
does not directly support it.
A gateway will unfortunately
be needed.
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