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.

1. ß
#sthlmwebrtc
http://www.meetup.com/WebRTC-Stockholm/

2. Introduction to
WebRTC
Browser to Browser communication
V 1.2 oej@edvina.net 2014-11-07
twitter @oej
Ⓒ Copyright Edvina AB, Sollentuna, Sweden 2014.All rights reserved.

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

4. Much more than ”calls”
GAMES
DATING SITES
ART

5. Much more than ”media”
DATACHANNELS
p2p (or
browser2browser)
Anything!

6. Secure peer-to-peer
Using DTLS + SRTP for media
Web Server
“Rendevouz”

7. Much more than browsers
Browser
Thing with
WebRTC stack
Gateway
with WebRTC stack
Something else.
SIP, XMPP, DecNet

8. P2P SCTP over UDP
P2P SRTP
P2P SRTP
Multiple channels
AUDIO
VIDEO
DATA

9. NAT"
Firewall
Network problems

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.

14. P2P call setup
Web
Server
Media
SDP SDP
The web application defines the signaling

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

17. No signalling?
Means ANY
signaling
XMPP
SIP
?
The users need to
exchange SDPs. That’s
all.

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.

19. SIP & WebRTC
Web
Server
SIP server with
web socket
support
SIP phone
Media
SIP Javascript
client

20. SIP: Using gateway
Web
Server
SIP gateway with
web socket
support
SIP phone
Media
SIP Javascript
client

21. SIP: Using gateways
Web
Server
SIP gateway with
web socket
support
SIP phone
SIP Javascript
client
Needs to be able to transcode
Turn
Server

22. Using Open Source
Apache
Web
Server
Asterisk + Opus
Patch
SIP gateway with
web socket
support
Jitsi
SIP phone
JSSIP Javascript
client
ReTurn
Server

23. That’s the intro.
Let’s talk!

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