Achieving sub-50ms streaming latency requires understanding the entire media pipeline—from encoder to viewer. This guide breaks down every component and provides actionable optimizations for real-time streaming.
In traditional broadcasting, a 30-second delay between capture and display is acceptable—viewers at home watch the same feed, just slightly behind the live event. But modern interactive applications demand more. Sports bettors need odds that match what's happening on the field. Auction participants need to see bids in real-time. Gamers expect their audience to react to plays as they happen.
Ultra-low latency streaming—typically defined as sub-500ms end-to-end—enables these interactive experiences. But achieving it requires rethinking the entire streaming pipeline, from capture to delivery.
This guide provides a comprehensive technical overview of ultra-low latency streaming: where latency comes from, how different protocols handle it, and specific optimizations you can implement today.
Different streaming protocols offer vastly different latency characteristics. Choosing the right protocol is the first step to achieving your target latency.
| Protocol | Typical Latency | Real-Time | Interactive |
|---|---|---|---|
| Traditional HLS | 15-30 seconds | — | — |
| Low Latency HLS (LL-HLS) | 2-5 seconds | — | — |
| RTMP to HLS | 3-10 seconds | — | — |
| SRT | 100-500ms | ||
| WebRTC (WHIP/WHEP) | 50-200ms | ||
| WAVE OMT | <16ms |
Understanding latency requires examining every step in the media pipeline. Each component contributes to total end-to-end delay.
Use hardware encoding (NVENC/QuickSync) with ultrafast/zerolatency preset
Minimize jitter buffer size, use adaptive playback
Edge locations close to both broadcaster and viewers
Use UDP-based protocols (WebRTC, SRT) instead of TCP
Skip transcoding or use hardware-accelerated ABR
Hardware decoding, optimized video player
Each low-latency protocol makes different tradeoffs between latency, reliability, and ease of deployment.
Web Real-Time Communication with standardized ingest (WHIP) and egress (WHEP) protocols.
Browser-based interactive streaming, video calls, auctions
Open-source protocol designed for reliable, low-latency video transport over unpredictable networks.
Professional contribution, remote production, broadcast ingest
Proprietary ultra-low latency protocol optimized for one-way media transport.
Sports betting, financial trading, competitive gaming, live auctions
Encoder settings have a significant impact on latency. Here are optimized configurations for different latency targets.
Different industries have different latency requirements based on their interactive needs and regulatory considerations.
| Industry | Requirement | Reason | Recommended Solution |
|---|---|---|---|
| Sports Betting | <100ms | Prevent delayed viewers from exploiting odds | WebRTC with edge transcoding, OMT for premium |
| Live Auctions | <500ms | Real-time bidding requires immediate feedback | WebRTC with fallback to LL-HLS |
| Esports | <200ms | Match viewer experience to gameplay | SRT ingest with WebRTC delivery |
| Financial Trading | <50ms | Market-moving information delivery | OMT protocol with dedicated infrastructure |
| Interactive Education | <1s | Natural conversation flow | WebRTC for interactive, HLS for scaled viewing |
| Remote Production | <100ms | Director feedback and coordination | SRT with bonded connections |
You can't optimize what you can't measure. WAVE provides comprehensive tools for latency monitoring and troubleshooting.
Real-time visualization of end-to-end latency across all protocols
Deep packet inspection for troubleshooting delivery issues
Aggregate latency data from actual viewer sessions
WAVE provides the infrastructure and protocols you need to deliver sub-50ms streaming at scale. Get started with a free trial today.