World's Fair 2025
Your realtime AI is ngmi — Sean DuBois (OpenAI), Kwindla Kramer (Daily)
Overview
This talk emphasizes the critical role of low latency in building effective voice AI experiences, arguing that current real-time AI is often inadequate. The core thesis is that achieving natural, human-like voice interactions requires a fundamental shift in how audio and video are handled over networks, moving beyond traditional methods like WebSockets to leverage technologies like WebRTC. The speakers suggest that voice is poised to become a primary interface for the next generation of AI applications.
Who should watch
- AI Engineers building voice-enabled applications
- Product Managers evaluating voice AI features
- Developers working with real-time audio and video
- Anyone interested in the future of human-computer interaction via voice
Key takeaways
- Voice AI experiences demand extremely low latency, with user expectations mirroring natural human conversation (around 500ms). Latencies above one second significantly harm completion rates and user satisfaction.
- WebRTC is essential for real-time edge-to-cloud audio and video, offering superior performance over WebSockets for these use cases due to its handling of packet loss, jitter, and bandwidth fluctuations.
- WebSockets are suitable for prototyping and short, structured data transfers but are ill-equipped for the demands of real-time audio streams, often leading to glitches or high latency.
- WebRTC provides built-in mechanisms for resampling, packetization, and bandwidth estimation, simplifying development compared to building these features from scratch with WebSockets.
- The adoption of voice as a primary interface for AI is compared to the desktop-to-mobile platform shift, suggesting a significant upcoming transformation in user interfaces.
- Tools and frameworks like Pipcat and Pion are enabling developers to build sophisticated voice AI applications more easily, lowering the barrier to entry for innovative projects.
- Real-world applications of WebRTC are widespread, powering services like Zoom, WhatsApp, and Discord, and enabling advanced use cases such as remote surgery and teleoperated vehicles.
- The flexibility of WebRTC allows for various connection topologies, including peer-to-peer local connections, cloud-based processing, and multi-party integrations.
Notable quotes
*If you want to build a conversational voice experience that people really love, you you're going to stress a lot about latency. Nothing else matters if your AI responds too slowly.*
*Websockets are great if you're trying to deliver audio and you want something really easy that can target all platforms. If you're trying to build a prototype lots of different platforms, websockets the way to go. On the other hand, WebRTC solves a bunch of things around handling giving you high quality audio, high bandwidth, low latency.*
*I'm convinced voice is going to be the core building block of the next generation of UIs, of the UIs for the generative AI era.*
Unofficial community note. Prefer the recording for nuance.