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WebRTC Core Concepts

Below are the core components you see in WebRTC libraries (including Rust implementations like webrtc-rs).

Media

Handles audio and video streams:

  • microphone
  • webcam
  • screen share
  • encoding/decoding (Opus, VP8, H264, etc.)
  • track pipelines
  • jitter buffers

Represents the actual media content.

Interceptor

Middleware for packet flow in WebRTC.

Used for:

  • congestion control
  • retransmissions
  • analytics
  • logging
  • bitrate adaptation
  • simulcast filters

Data

Refers to WebRTC Data Channels.

Supports:

  • text and binary
  • reliable or unreliable delivery
  • ordered or unordered delivery

Used for chat, game sync, file transfer, etc.

RTP (Real-time Transport Protocol)

Packet format for real-time media transmission.

Includes:

  • timestamps
  • sequence numbers
  • SSRC identifiers

Carries media data.

RTCP (RTP Control Protocol)

Feedback channel for RTP.

Reports:

  • packet loss
  • jitter
  • round trip time
  • bandwidth
  • keyframe requests

SRTP (Secure RTP)

Encrypted RTP for media.

Uses keys negotiated via DTLS.

SCTP (Stream Control Transmission Protocol)

Protocol used by WebRTC Data Channels.

Provides:

  • multiple streams
  • configurable reliability
  • no head-of-line blocking

Runs inside DTLS. Can be configured to be TCP-like or UDP-like or anything in between.

DTLS (Datagram TLS)

TLS over UDP.

Used for:

  • verifying peer identity
  • negotiating encryption keys
  • securing SCTP
  • securing SRTP

DTLS does not typically go through CA Root Verification in WebRTC; it is verified in the signalling phase using the key fingerprint.

mDNS

Masks local LAN IP addresses for privacy (e.g. *.local hostnames).

STUN

Discovers:

  • public IP
  • public port
  • NAT mappings

Used for peer-to-peer connectivity.

TURN

Relay server fallback when direct P2P isn't possible.

Used in restrictive NAT environments.

ICE (Interactive Connectivity Establishment)

System that tests multiple network paths and selects the best one.

Uses:

  • host candidates
  • STUN candidates
  • TURN relay candidates

Continuously monitors connectivity.

SDP (Session Description Protocol)

Text-based connection description used for signaling.

Contains:

  • codecs
  • ICE credentials
  • DTLS fingerprints
  • media types
  • transport configs

Used in offers and answers.

Util

Shared helper utilities.

Commonly includes:

  • random generators
  • buffer helpers
  • parsers
  • timers
  • UUID helpers

Interoperability

Check WebRTC interoperability tests for samples in other WebRTC libraries like libwebrtc, pion, SipSorcery and whether they can interoperate with each other.

Extras

Selective Forwarding Unit (SFU)

This is not peer-to-peer WebRTC; a central server manages the media streams. This is how certain "mute" functions or noise detection is implemented in multiuser video conferencing. Used to restrict audio/video flow from users, due to administrative controls or due to being "idle" in a video/audio call.

Multipoint Control Unit (MCU)

This is an alternate architecture to the above, still not peer to peer but the central server itself takes the responsibility to mix/re-encode the streams.