ModelRefs / Voice Realtime Stack — Architecture Pattern
Voice Realtime Stack — Architecture Pattern
WebRTC-based voice transport with realtime model APIs and interruption-aware turn management. Turn manager handles interruptions and silence.
Overview
WebRTC for low-latency duplex audio paired with realtime LLM APIs that natively accept and emit audio. Turn manager handles interruptions and silence.
When to use it: You need true-realtime voice with sub-300ms perceived latency.
Pattern details
| Pattern class | real-time |
|---|---|
| Difficulty | advanced |
| Topology | event-driven |
| Also known as | webrtc llm, realtime voice api |
| Last reviewed | 2026-06-07 |
Known failure modes
- Packet loss — Audio glitches degrade ASR. Mitigation: FEC + jitter buffer tuning.
When not to use it
- Routing realtime audio through HTTP polling.
Continue your research
Use these connected ModelRefs sections to compare alternatives, inspect implementation paths, and review the evidence and governance boundaries relevant to Voice Realtime Stack — Architecture Pattern.
Frequently asked questions
When should I adopt the Voice Realtime Stack?
You need true-realtime voice with sub-300ms perceived latency.
What are common failure modes of Voice Realtime Stack?
Packet loss
Is Voice Realtime Stack production-ready?
Yes when paired with the safety controls and observability hooks documented on the pattern page.