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Add markdown files for basic explanation on WebRTC (#756)
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‎webrtc-explained.md‎

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# WebRTC Core Concepts
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Below are the core components you see in WebRTC libraries (including Rust implementations like webrtc-rs).
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## Media
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Handles audio and video streams:
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- microphone
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- webcam
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- screen share
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- encoding/decoding (Opus, VP8, H264, etc.)
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- track pipelines
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- jitter buffers
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Represents the actual media content.
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## Interceptor
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Middleware for packet flow in WebRTC.
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Used for:
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- congestion control
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- retransmissions
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- analytics
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- logging
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- bitrate adaptation
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- simulcast filters
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## Data
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Refers to WebRTC Data Channels.
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Supports:
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- text and binary
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- reliable or unreliable delivery
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- ordered or unordered delivery
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Used for chat, game sync, file transfer, etc.
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## RTP (Real-time Transport Protocol)
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Packet format for real-time media transmission.
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Includes:
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- timestamps
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- sequence numbers
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- SSRC identifiers
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Carries media data.
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## RTCP (RTP Control Protocol)
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Feedback channel for RTP.
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Reports:
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- packet loss
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- jitter
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- round trip time
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- bandwidth
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- keyframe requests
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## SRTP (Secure RTP)
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Encrypted RTP for media.
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Uses keys negotiated via DTLS.
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## SCTP (Stream Control Transmission Protocol)
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Protocol used by WebRTC Data Channels.
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Provides:
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- multiple streams
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- partial reliability
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- no head-of-line blocking
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Runs inside DTLS.
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## DTLS (Datagram TLS)
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TLS over UDP.
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Used for:
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- verifying peer identity
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- negotiating encryption keys
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- securing SCTP
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- securing SRTP
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## mDNS
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Masks local LAN IP addresses for privacy (e.g. *.local hostnames).
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## STUN
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Discovers:
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- public IP
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- public port
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- NAT mappings
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Used for peer-to-peer connectivity.
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## TURN
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Relay server fallback when direct P2P isn't possible.
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Used in restrictive NAT environments.
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## ICE (Interactive Connectivity Establishment)
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System that tests multiple network paths and selects the best one.
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Uses:
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- host candidates
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- STUN candidates
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- TURN relay candidates
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Continuously monitors connectivity.
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## SDP (Session Description Protocol)
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Text-based connection description used for signaling.
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Contains:
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- codecs
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- ICE credentials
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- DTLS fingerprints
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- media types
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- transport configs
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Used in offers and answers.
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## Util
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Shared helper utilities.
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Commonly includes:
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- random generators
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- buffer helpers
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- parsers
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- timers
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- UUID helpers
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‎webrtc-signaling-explained.md‎

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# WebRTC Signaling Explained
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## What is Signaling?
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Signaling is the process where two peers exchange the information required to establish a WebRTC connection.
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It happens *before* any direct peer connection is possible.
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This includes exchanging:
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- SDP Offer
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- SDP Answer
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- ICE Candidates
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## Why Signaling Is Needed
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WebRTC peers cannot discover each other on their own.
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They need to:
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- agree on codecs and media setup
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- share network information
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- exchange security fingerprints
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Signaling provides this exchange.
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## What Signaling Is Not
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Signaling is **not**:
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- part of WebRTC itself
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- standardized
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- a fixed protocol
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- tied to any transport
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## Signaling Transport Options
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Signaling can use any transport, including:
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- WebSockets
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- HTTP
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- REST
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- SSE
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- MQTT
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- TCP
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- UDP
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- Redis pub/sub
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- anything else
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## What Information Gets Exchanged
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### SDP (Session Description Protocol)
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Contains:
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- codecs
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- media directions
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- ICE usernames/passwords
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- DTLS fingerprints
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- transport description
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- media streams
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### ICE Candidates
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Contain:
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- IP addresses
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- ports
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- transport protocol
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- candidate priority
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## Typical Signaling Flow
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1. Peer A creates PeerConnection
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2. Peer A generates Offer (SDP)
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3. Peer A sends Offer to Peer B using the signaling channel
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4. Peer B sets remote Offer
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5. Peer B generates Answer (SDP)
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6. Peer B sends Answer back via signaling channel
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7. Both sides exchange ICE Candidates
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8. ICE finds a working route
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9. Peers connect directly
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## Signaling Server Responsibilities
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A signaling server:
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- relays Offer
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- relays Answer
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- relays ICE candidates
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It does *not*:
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- relay audio/video (RTP)
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- relay data channel traffic
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- stay involved after connection succeeds
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## Server After Connection
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Once the WebRTC connection is established:
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- the signaling server is no longer required
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- peers communicate directly (unless TURN is used)
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## Simplified Diagram
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Peer A ---- Offer/Answer/ICE ----> Signaling Server ---- Offer/Answer/ICE ----> Peer B
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Peer A <------------------------------------- P2P ------------------------------------> Peer B
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## Summary
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- Signaling exists only to bootstrap WebRTC.
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- It exchanges Offer, Answer, and ICE candidates.
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- It is not part of WebRTC protocol.
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- You can use any transport you want.
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- Once done, peers can communicate directly.
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