Prompt
What should I use for low-latency video calls in poor networks?
Latest observation
Nov 2, 2026
OpenAI APIWeb search: off
For low-latency video calls over poor networks, the best choice is usually WebRTC.
Why WebRTC
- Built for real-time communication: optimized for voice/video calls
- Adaptive to bad networks: handles jitter, packet loss, and changing bandwidth
- Low latency: designed to keep delay minimal
- NAT/firewall traversal: works better across real-world networks
- Widely supported: browsers, mobile apps, desktop apps
What to pair with it
To improve performance on poor networks, use:
- Opus for audio
- H.264 or VP8/VP9 for video, depending on device support
- TURN servers as fallback when direct peer-to-peer connection fails
- Adaptive bitrate / simulcast / SVC for dynamic quality adjustment
If your priority is even lower bandwidth
Consider:
- Audio-first mode
- Lower resolution / frame rate
- Server-side relaying or transcoding if you need tighter control
- SRT / RIST only if this is more of a broadcast contribution workflow than a conversational call
Short answer
If you mean interactive two-way video calling, use WebRTC.
If you want, I can also recommend:
- the best codec settings for poor networks, or
- a full stack for building the call system.