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Video latency: optimize real-time WebRTC

Short answer

Video latency in WebRTC is the delay between emitting a frame and receiving it on the other side. Under normal conditions, expect 200 ms to 1 second for interaction — well below deferred streaming (HLS, 5–30 s). Main factors: distance to SFU, network quality, ICE path choice, and server sizing.

WebRTC latency vs deferred streaming

Protocol Typical latency Use case
WebRTC 200 ms – 3 s Dialogue, real-time video, support
WebSocket (data) 50 – 500 ms Signaling, chat
HLS / DASH 5 – 30 s Mass webinar, replay

For real-time communication, WebRTC is the right protocol.

Six latency factors in WebRTC

1. Network distance to SFU

Every millisecond of RTT adds up. An SFU in France minimizes latency for European participants.

2. ICE path (direct vs TURN)

  • Direct: minimal latency;
  • TURN relay: added latency (1 server hop);
  • TURN TCP fallback: higher than UDP.

STUN and TURN

3. Codecs and encoding

Hardware-accelerated VP8/VP9/H.264 reduces latency. 720p/30fps is a good latency/quality trade-off.

4. SFU sizing

A saturated SFU adds jitter and buffering.

5. Client network quality

Poor Wi-Fi or 4G triggers bitrate adaptation and variable latency.

6. Architecture (SFU vs MCU)

SFU forwarding without transcoding has lower latency than MCU mixing.

Acceptable latency by use case

Use case Target latency Protocol
Conversation / real-time video < 300 ms ideal WebRTC
Support / diagnosis < 1 s WebRTC
10+ participant meeting < 1–2 s WebRTC + SFU
Interactive live (Q&A) < 3 s WebRTC or hybrid

IT optimization checklist

  1. SFU hosted close to participants (France);
  2. TURN documented with UDP priority, TCP fallback;
  3. SFU sized for participant count;
  4. Hardware-accelerated codecs enabled;
  5. Simulcast / SVC for adaptation without retranscoding;
  6. Continuous monitoring (jitter alerts > 30 ms).

How does Leagora optimize latency?

WebRTC infrastructure with SFU in France, documented TURN, simulcast, monitoring. Contact for a network workshop.

Products: meeting.leagora.io, assistance-video.fr.

FAQ

What minimum latency with WebRTC?

~150–200 ms in optimal direct conditions. In B2B production: 300 ms – 1 s is realistic.

Does TURN increase latency?

Yes, by one server hop (~20–100 ms). Less than connection failure without TURN.

Does latency depend on browser?

Marginally: recent Chrome, Firefox, Safari are comparable. Network dominates.

Does leagora.io guarantee latency figures?

leagora.io documents factors. SLA guarantees are contractual.

Key takeaways

  • WebRTC production latency = 200 ms – 1 s (sub-second in optimal conditions).
  • France SFU + documented TURN + simulcast = optimization triad.