Understanding WebRTC: What It Is and What It Isn’t

Introduction

WebRTC is the native real-time audio, and video plus low-latency data exchange stack of the web that offers peer-to-peer enabled communication with built-in NAT traversal, encryption support as well as adaptive transport selection. It opens up browser APIs for media capture, secure connection establishment, and streaming audio/video/data between peers by use of RTCPeerConnection, MediaStream tracks, plus DataChannels. WebRTC does not constitute any signaling protocol; hence, a separate signaling channel (like WebSocket, HTTP, or MQTT) has to be provided by the application for exchanging session descriptions (SDP) plus ICE candidates between peers.

In this blog, we will explore:

• How WebRTC works under the hood͏
• Core Components: getUserMedia, RTCPeerConnection, RTCDataChannel​
• Signaling/ICE negotiation​
• STUN/TURN traversal​
• Performance & scaling in production
• Working implementation

Core Notions: ICE, STUN, TURN

ICE (Interactive Connectivity Establishment): gathers candidate endpoints for connectivity checks exchanges them with the remote peer runs connectivity checks and selects the best working route between the peers.
STUN (Session Traversal Utilities for NAT): The use of STUN by a client most often allows it to determine its public-facing address as well as the NAT behavior; in most cases, this permits direct peer-to-peer connections if allowed by the particular network.
TURN (Traversal Using Relays around NAT): In TURN, all media/data is relayed when direct traversal fails due to the presence of strict NATs/firewalls; It is the most reliable but introduces latency as well as server bandwidth cost.
Candidate lifecycle: Gather candidates, exchange via signaling and perform connectivity checks, select the successful pair and monitor ICE state changes in connection.

Minimal Architecture required for a Simple WebRTC Application

Client:

• Capture media using getUserMedia.
• Create an RTCPeerConnection with ICE servers (STUN/TURN) configuration.
• Add local tracks to it.
• Create SDP offer/answer, then exchange them over a signaling channel.
• Exchange ICE candidates over signaling also.
• Play remote media when ontrack events fire.

Signaling Server:

Any two-way channel (often WebSocket) that just shuttles offers/answers/candidates between clients.
STUN/TURN:

Public STUN server for discovery; provide TURN server for robustness under restrictive NATs.

Step-by-Step Guide to Building a Simple 1:1 Video Chat App

Here’s a simple working demo of a video chat application. This example focuses mainly on the client-side logic and assumes you’re using a basic WebSocket signaling server to exchange messages between the two peers (the “caller” and the “callee”).

Step 1: Capture local media and set up the peer connection

• Configure your ICE servers (at minimum, a STUN server; for production, you’ll also need a TURN server).
• Add your local audio and video tracks to the connection.
• Set up event handlers for ICE candidates and remote tracks.
  

  Create Connection And Load Tracks

It’s common practice to add your local tracks before establishing the connection. Once the remote peer starts sending media, the ontrack event will fire, and you’ll receive their audio and video streams. The icecandidate event handler is responsible for sending each discovered candidate through your signaling server to the other peer.

Step 2: Implement Offer/Answer Signaling

• The caller creates an offer, sets it as their local description, and sends it to the callee.
• The callee receives the offer, sets it as their remote description, creates an answer, sets it as their local description, and sends that back.

Both peers then exchange and add ICE candidates received through the signaling server to complete the connection.

Read Message Events

• The flow mirrors common code-lab examples: create RTCPeerConnection, add local tracks, exchange ICE candidates and SDP, and handle track events to play remote media.
• The “signaling” transport is app-defined; WebRTC does not standardize it.
  

  3) Observability and ICE state

• Log iceconnectionstatechange to diagnose failures.
• Use getStats to introspect the selected candidate pair (useful to confirm if you’re relaying via TURN).

State Change Event Listener

• Monitoring and candidate-type logging helps distinguish direct vs relayed paths and troubleshoot NAT traversal issues.
  

  4) DataChannel (optional)

• For low-latency data (game state, cursors), create a DataChannel.
• DataChannel uses SCTP over the established ICE path.

Data Channel and Peer State

• WebRTC data is useful for real-time collaboration or metadata synchronized with media.
  

  Production Readiness Checklist

• Provision TURN for reliability in restrictive networks; TURN relays all traffic and adds latency/bandwidth cost but guarantees connectivity.
• Harden the signaling layer and authenticate peers; signaling is essential even though traffic is peer-to-peer.
• Consider CPU and bandwidth ceilings: group calls multiply outbound streams per participant without SFU; use an SFU for multiparty.
• Track ICE failures and fallbacks, collect stats, and expose telemetry to operations teams.
• Carefully select codecs and constraints to strike a balance between quality, CPU usage, and bitrate.

Drawbacks and Limitations of WebRTC

• Requires servers: A signaling server is a must to run this application; TURN is often needed in production, adding operational cost and complexity for startups.
• Scaling challenges in mesh: CPU and uplink increase with each additional participant; SFUs are often necessary for group calls.
• Complexity and learning curve: Developers manage ICE/STUN/TURN, SDP, codecs, and transport behavior.
• Latency and cost under TURN: Relayed traffic increases latency and server egress costs.

Alternatives to WebRTC

WebSockets:

• Best for reliable, ordered, server-mediated messaging (chat, dashboards, collaborative text) where data integrity is crucial.
• Not ideal for high-bitrate audio/video; traffic goes through a central server.

HLS/DASH (HTTP streaming):

• Ideal for large-scale one-to-many broadcasts via CDNs; latency typically in seconds, not real-time.

WebTransport (HTTP/3 QUIC-based, emerging):

• Offers reliable/unreliable streams and lower overhead for some real-time use cases; good for custom real-time data transports without the full media stack.
• The ecosystem is still maturing; not a drop-in replacement for WebRTC’s A/V pipeline.

Hosted SFU/MCU SDKs and Platforms:

• Managed APIs that hide ICE/STUN/TURN details, and offer features like recording, moderation, simulcast/SVC, and scalability. Trades some control for faster delivery.

Practical Tips

• Always specify ICE servers (STUN/TURN) when constructing RTCPeerConnection; include TURN in production for reliability across NATs.
• Add tracks before negotiation and wire ontrack to render remote media quickly.
• Keep signaling schema simple (offer, answer, candidate), and secure it end-to-end (auth, input validation).
• Instrument ICE states and log candidate types to debug NAT traversal issues quickly.
• Start with 1:1 mesh; switch to an SFU when moving to group calls.

Conclusion

WebRTC is a powerful technology that brings real-time communication directly to the web. On the surface, its APIs look simple, but building a reliable connection means digging deeper—understanding ICE negotiation, signaling, and how to deal with NAT traversal.

Whether you’re creating a video conferencing tool, a multiplayer game, or a secure chat app, WebRTC gives you the flexibility and performance needed to make it all work smoothly.