Optimize your audio and video sync by managing latency effectively.
Latency is the time delay between when something is said and when it's received as captions or interpreted audio by your audience.
Minimizing latency is crucial for:
⚠️ Some delay is unavoidable — especially with AI voice synthesis or broadcast streaming protocols — but it can be optimized.
Factor | Impact on Latency |
---|---|
Streaming Protocol | RTMP adds 3–10s; SRT is faster; WHIP is low-latency; Web Agent/Desktop Agent are lowest |
AI Voice Interpretation | Adds 5–15s for voice processing and synthesis |
Network Conditions | Poor bandwidth, high jitter, or Wi-Fi instability cause inconsistent delivery |
Device Performance | Underpowered computers or overworked CPUs can delay audio encoding and push |
Video Embeds (YouTube/Vimeo) | Built-in platform buffering adds 5–30s depending on player and CDN behavior |
For in-person or real-time interaction, use:
Avoid RTMP/SRT/WHIP for local audiences — they're best suited for online-only events where latency is less critical
Keep your signal chain clean Minimize unnecessary devices (e.g., mixers > converters > interfaces) — each adds processing time
Monitor latency live Ask a teammate to join as a test viewer and report the delay between live speech and captions/audio
Keep software updated Agents, streaming tools, and firmware often receive latency fixes in new releases
Can I eliminate latency completely? No — some delay is inevitable. But with proper setup, you can keep latency low enough for real-time use.
Do more languages increase latency? No — translations are processed in parallel.
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