WebRTC allows for real-time communication through a web browser or mobile app using audio, video, and data without requiring a centralized server. It works by using STUN and TURN servers to handle signaling and traverse NATs to establish a direct peer-to-peer connection when possible. WebRTC implements encryption and permissions to help secure access to devices and media streams, though vulnerabilities still exist around stolen credentials and unverified certificates. It supports advanced features like simulcast for adaptive streaming and SFUs for flexible multi-party video routing.

2. Table of Content
• Introduction
– Why WebRTC is Important?
– What We Can Do With WebRTC?
– Browser Supports
• Architecture of WebRTC
– Architecture of WebRTC app
– Signaling
• Security measures in WebRTC
• WebRTC Vulnerabilities
• Advanced Features

3. WebRTC stands for …?
• Web Real Time Communication
 WebRTC let you to Cooperate with Video, Audio and Data Communication.
 Can Communicate peer to peer without centralized server.
 This standard enables users to communicate in real-time through a web-browser
or a mobile app

4. Why WebRTC is Important…?
• No centralized server required,
• Adapted by wide range of browsers,
• The WebRTC client makes use of a microphone, speakers, and/or
camera attached to the computer to send audio and/or video.
• This means that a user can communicate with others with basically
nothing but a browser and internet connection.

5. Browser Supports…

6. What We Can Do With WebRTC…?
• Potential Use Cases and Inspiration
– File Sharing
– Screen Sharing
– Video Conferencing/Tele presence
– Integrated Text Chat
– Virtual Conference
– Remote Desktop Applications
– Multi Player

7. Architecture of WebRTC

8. Architecture of WebRTC app

9. Signaling…
• WebRTC signaling refers to the process of setting up, controlling, and
terminating a communication session. In order for two endpoints to
begin talking to one another, three types of information must be
exchanged:
– Session control information
– Network Data
– Media Data

10. Signaling Servers…
• STUN – Session Traversal Utilities for NAT
– NAT – Network Address Translation
• TURN – Traversal Using Relay NAT
• ICE – Interactive Connectivity Establishment

11. STUN server…
• STUN server allows clients to discover their public IP address and the
type of NAT they are behind.
• In most cases, a STUN server is only used during the connection setup
and once that session has been established, media will flow directly
between clients.

12. TURN Server
• If a STUN server cannot establish the connection, ICE can turn to TURN
(pardon the pun).
• TURN is an extension to STUN that allows media traversal over a NAT
that does not do the “consistent hole punch” required by STUN traffic.
• TURN servers are often used in the case of a symmetric NAT.

13. ICE
• ICE is a framework that allows WebRTC to overcome the complexities
of real-world networking.
• It’s ICE’s job to find the best path to connect peers. It may be able to
do that with a direct connection between the clients, but it also works
for clients where a direct connection is not possible (i.e. behind NATs).

14. Security measures in WebRTC…
● Encrypt the data as it travels between users.So interception could only
be done in the end points or media gateway.
● Browsers will ask users to verify access to their cameras and
microphones.

15. WebRTC Vulnerabilities
• Access to devices
HTML and JavaScript are executed by the browser as a sandbox
designed to be isolated from the rest of the computer. How ever bug may
be exist.
-Threats
• Web page’s access to user’s camera and microphone without
permission.
• Users can be potentially being recorded with JavaScript code
downloaded from the malicious web server.

16. Security of the turn server
• Turn is necessary to establish bi-
directional flows.
• Media relaying is an expensive
resource. So it is protected with
credentials.
• Those credentials can be long-term.
If these credentials are stolen the
turn server can be abused.

17. Protection
• Signaling over TLS
TLS provide privacy ,integrity and authentication. It is also provides
server authentication and client authentication if a client certificate is
provided.
If the client certificate is signed by a trusted certification authority the
real time communication can have legal value. But attacker can use
dummy certificate for the attack. Due to that HTTPS must used when
working with WebRTC.

18. Advanced features
• Simulcast
• SFU

19. Simulcast
ability to adopt the quality of video for each participants.
Desktop
Desktop
Mobile
Cr. Book
1080 P
720 P
360 P
1080P
720 P
360 P
SFU
SFU=Selective Forwarding Unit

20. Selective forwarding unit(SFU)
• Early days MCU(Multipoint Control Unit) used for that case .But this
approach is very expensive computationally.
• Thus, Selective Forwarding Units (SFUs) have become popular as a
more economical solution for multi-party video routing.
• It is selects the which sender stream to forward to a receiver.
• It is making video encoding more flexible.

21. Thank You