Realtime Voice & Live Translation
Last updated 15 Jun 2026
Overview
ToRun supports realtime, bidirectional voice — you speak, the assistant listens and responds in audio and text over a live session — and live translation, where speech is translated as it happens. These run on the realtime capability (bidirectional audio + text) and are billed per second of session time.
Capabilities involved
- Realtime (audio + text) — a live WebSocket-style session with a capable provider model for natural back-and-forth voice conversation.
- Text-to-speech (TTS) — synthesized speech output across 29 languages.
- Speech-to-text (STT) — transcription, including speaker diarization (who said what).
- Live translation — speech translated in real time between languages.
Starting a realtime session
- Open the voice/realtime surface.
- Grant microphone access.
- Speak naturally — the assistant transcribes, responds, and (in voice mode) speaks back.
- End the session when done. The session is tracked as a voice session entity.
For live translation, choose the source and target languages; each party hears the other's speech translated as it streams.
Billing
- Realtime and live-translation usage is metered per second of audio.
- Each metered unit of work produces a
BillingRecordwith a pricing snapshot, so the cost of a voice session is transparent and auditable. - Voice/realtime sits behind the Voice/Audio entitlement, which is included from the Pro tier upward.
STT and diarization
Speech-to-text supports speaker diarization, labeling segments by speaker. This is useful for transcribing meetings, interviews, or multi-party calls where you need to attribute lines to specific speakers.
Tips
- Use a headset or a quiet room — clean input audio improves both transcription and translation accuracy.
- Because realtime is metered per second, end sessions you're not actively using.
- For one-off transcription of an existing recording, STT (not a live realtime session) is the cheaper path.