WebRTC Development

Media Capture and Stream:

Capturing audio and video from a device’s camera, microphone and streaming it in real-time.

Peer-to-Peer Communication:

Establishing direct communication between two or more peers, reducing latency (or very low delay between an instruction to transfer data), and dependency on intermediary servers.

NAT Traversal:

Using techniques like STUN (Session Traversal Utilities for NAT), and TURN (Traversal Using Relays around NAT) to handle network address translation and ensure peers can connect even behind firewalls and routers.

Encryption:

Ensuring secure communication through mandatory encryption of audio, video, and data streams using protocols like SRTP (Secure Real-Time Protocol) to establish, manage and terminate connections.

Signaling:

Exchanging control messages like SIP (Session Initiation Protocol) or custom signaling server.

SFU/MCU Server:

Selective Forwarding Unit (SFU), and Multipoint Control Unit (MCU) are media servers used to handle large number of users in a video conference and streaming. Depending on the application requirements, each server is decided.

APIs:

Providing a set of JavaScript APIs that developers can use to build real-time communication features into web applications. Key APIs include:

• getUserMedia(): Captures media from the user’s camera, screen and microphone.
• RTCPeerConnection: Manages the peer-to-peer connection, handling the setup, media exchange, and closing of the connection.
• RTCDataChannel: Enables peer-to-peer data exchange, allowing for real-time data transfer.