WebRTC for Live Streaming: Achieving Sub-Second Latency

WebRTC (Web Real-Time Communication) enables real-time audio and video communication directly in web browsers with sub-second latency. Originally designed for video conferencing, WebRTC is increasingly used for interactive live streaming applications.

WebRTC Architecture Overview

WebRTC is a collection of protocols and APIs that enable peer-to-peer media communication without plugins.

Core Components

RTCPeerConnection: The main API for establishing and managing connections between peers. Handles ICE negotiation, DTLS encryption, and media transport.

MediaStream: Represents synchronized streams of audio and video tracks captured from cameras, microphones, or screen sharing.

RTCDataChannel: Enables bidirectional data transfer between peers, useful for chat, game state, or control messages.

Signaling Server: Not part of WebRTC specification but required to exchange connection information (SDP offers/answers and ICE candidates) between peers.

Latency Comparison

ProtocolTypical LatencyUse Case HLS10-30 secondsVOD, large-scale live LL-HLS2-5 secondsLive events RTMP2-5 secondsIngest only WebRTCUnder 500msReal-time interactive

Signaling Process

WebRTC requires an external signaling mechanism to exchange connection information before media can flow.

Connection Establishment Steps

1. Create RTCPeerConnection with ICE server configuration

2. Generate Offer: Caller creates SDP offer describing desired session

3. Exchange via Signaling: Send offer through your signaling server (WebSocket, HTTP, etc.)

4. Generate Answer: Callee creates SDP answer

5. Exchange ICE Candidates: Both peers share network candidates as they are discovered

6. Media Flows: Once ICE completes, encrypted media streams begin

Signaling Server Requirements

WebSocket or Server-Sent Events for real-time delivery
Authentication and room management
Message routing between specific peers
Graceful handling of disconnections

NAT Traversal: STUN and TURN

Most devices sit behind NAT (Network Address Translation), making direct peer connections challenging.

STUN (Session Traversal Utilities for NAT)

STUN servers help peers discover their public-facing IP address and port mapping.

How STUN Works:

1. Client sends request to STUN server

2. Server responds with client's public IP and port

3. Client shares this information via signaling

4. Peers attempt direct connection

Limitations: STUN fails with symmetric NATs where each destination gets a different port mapping.

TURN (Traversal Using Relays around NAT)

TURN servers act as media relays when direct connection is impossible.

When TURN is Needed:

Symmetric NAT configurations
Strict corporate firewalls
VPN connections
Mobile networks with carrier-grade NAT

Cost Consideration: TURN relays all media traffic, consuming bandwidth. Plan for 10-20% of connections needing TURN.

ICE Server Configuration

Production deployments should include both STUN and TURN servers:

Primary STUN server (can use free Google STUN)
Backup STUN server
TURN server with UDP transport
TURN server with TCP transport (firewall fallback)
TURN server with TLS (port 443 for restrictive networks)

Scaling WebRTC: Media Server Architectures

Pure peer-to-peer WebRTC does not scale beyond small groups due to the N-squared problem.

Mesh Topology (P2P)

Each participant connects to every other participant directly.

Connections Required: N × (N-1) / 2

2 participants: 1 connection
5 participants: 10 connections
10 participants: 45 connections

Best For: 2-4 participants maximum

SFU (Selective Forwarding Unit)

Central server receives one stream from each participant and forwards to others.

Advantages:

Linear scaling (N connections per participant)
No transcoding required (low CPU)
Supports simulcast for adaptive quality
Maintains low latency

Popular SFU Solutions:

mediasoup (Node.js)
Janus Gateway (C)
LiveKit (Go)
Pion (Go library)

Best For: 5-1000+ participants

MCU (Multipoint Control Unit)

Server receives all streams, mixes into single composite stream.

Advantages:

Single download stream per participant
Lower client bandwidth
Consistent layout for all viewers

Disadvantages:

High server CPU (transcoding)
Added latency from mixing
Less flexible layouts

Best For: Limited bandwidth scenarios, recording

Optimization Techniques

Simulcast

Send multiple quality levels simultaneously, let the SFU select best for each viewer.

Typical Configuration:

Low: 320x180 at 150kbps
Medium: 640x360 at 500kbps
High: 1280x720 at 1500kbps

The SFU forwards appropriate quality based on receiver bandwidth and preferences.

Scalable Video Coding (SVC)

Encode video in layers that can be selectively dropped.

Layer Types:

Temporal: Frame rate layers (15/30/60 fps)
Spatial: Resolution layers
Quality: Fidelity layers

SVC reduces encoding overhead compared to simulcast but has less codec support.

Bandwidth Estimation

WebRTC includes built-in bandwidth estimation using:

REMB (Receiver Estimated Maximum Bitrate)
Transport-CC (Transport-Wide Congestion Control)

Monitor these metrics to adapt quality proactively.

Jitter Buffer Tuning

The jitter buffer compensates for network timing variations.

Trade-offs:

Larger buffer: Smoother playback, higher latency
Smaller buffer: Lower latency, potential stuttering

For interactive applications, use aggressive (small) jitter buffer settings.

Use Cases for WebRTC

ApplicationWhy WebRTCKey Requirements Video ConferencingBidirectional, real-timeEcho cancellation, bandwidth adaptation Live AuctionsInstant bid feedbackUltra-low latency, reliability Virtual ClassroomsInteractive Q&AScreen sharing, chat integration Live Sports BettingReal-time odds updatesSub-second latency critical TelemedicineReal-time diagnosisPrivacy, recording, HIPAA compliance GamingPlayer reactionsMinimal input lag Remote ProductionLive event coordinationHigh quality, synchronization

WebRTC vs Other Protocols

When to Use WebRTC

Latency under 1 second is required
Bidirectional communication needed
Interactive features (chat, reactions)
Small to medium audience size

When to Use HLS/DASH

Large audience (thousands+)
Latency tolerance of 5+ seconds
CDN distribution required
Device compatibility priority

Hybrid Approach

Many platforms combine both:

WebRTC for interactive features and hosts
HLS for large-scale viewer distribution
Gateway servers convert between protocols

Conclusion

WebRTC provides unmatched latency for interactive applications but requires careful architecture for scale. Key decisions include:

STUN/TURN infrastructure for NAT traversal
SFU media server for scaling beyond peer-to-peer
Simulcast or SVC for adaptive quality
Proper jitter buffer configuration for your latency requirements

For platforms requiring both real-time interactivity and large-scale delivery, consider hybrid architectures combining WebRTC with HLS.

Platforms like dcast.tv can integrate WebRTC for interactive features while using HLS for broad delivery, offering the best of both worlds without the infrastructure complexity.