openai-realtime-voice
Build production OpenAI Realtime voice agents and low-latency spoken interactions with live audio sessions, WebRTC or WebSocket transport, voice activity detection, tool/function calling, prompt design, logging, consent, privacy, safety, latency, and cost controls. Use for speech-to-speech agents and live voice UX; do not use for separate request-based transcription, text-to-speech, or offline audio generation work.
npx skills add calesthio/generative-media-skills --skill openai-realtime-voice --agent claude-code
Same command for any agent — swap --agent for codex, cursor, copilot.
Weekly change comes from our own snapshots, not the repository page — it measures attention, not adoption.
What it does
Guides a live spoken-interaction agent (browser voice assistant, phone-like agent, or live voice UI) to listen, reason, speak, interrupt, and potentially call tools within a Realtime session. It is not intended for batch transcription, offline TTS, or file-based audio tasks. It emphasizes low-latency, turn-taking, tool use, logging, consent, privacy, safety, latency, and cost controls.
How it works
- The skill targets production OpenAI Realtime voice sessions (live audio with low latency) and distinguishes them from batch or transcription workflows.
- It instructs choosing between architecture options: use Realtime voice-agent session for latency-sensitive, interruptible conversations with tool calls; or a chained STT -> text -> TTS pipeline when deterministic transcripts are priority.
- It prescribes using WebRTC for browser/mobile media transport and WebSocket for server-to-server integration, with session management via a Session object, a Conversation, and Responses. It notes that the voice cannot be changed after the first audio response in a session and that the maximum session duration is 60 minutes.
- It details input-audio handling (base64 via input-audio-buffer for WebSocket; WebRTC handling media), VAD configuration (server_vad or semantic_vad) and tuning (silence_duration_ms, threshold, prefix_padding_ms).
- It allows attaching tools at session or per-response level, including MCP tools, and outlines cost accounting (audio tokens) and data handling expectations (transcripts as observability artifacts, not exact reasoning inputs).
- It provides extensive prompts design guidance for spoken agents, including role, voice persona, turn length, safety, privacy, and escalation rules. It covers turn-taking state machine, interruption rules, and explicit UI controls for stopping/muting.
- It defines when to use tools, how to validate and return results, and safety/approval policies for high-impact actions.
When to use it
Use this skill when the deliverable is a live spoken interaction with real-time turn-taking, tool usage, and user-centric audio UX (e.g., browser voice assistant, low-latency tutoring conversation, live kiosk). Do not use for batch speech-to-text, offline TTS, or audio file editing tasks.
What it can touch
- Tools: function tools, MCP tools, and built-in connectors may be attached at session level or for a single response.
- Media and transport: WebRTC for browser/mobile, WebSocket for server-to-server transport.
- Session/state: model, voice, and configuration managed within a Realtime Session object; conversation items and responses tracked.
Caveats
- OpenAI model names, session limits, and data handling policies are subject to change; producers should verify current references before release.
- Data may be logged (sessions, tool calls, IDs) per the guidance; transcripts are treated as observability artifacts, not exact reasoning inputs.
- VAD behavior is configurable and may require careful tuning for different environments and languages; invalid or disabled VAD places more responsibility on application-side turn boundaries.
- Some capabilities depend on external tool servers or remote connectors and require proper authorization, logging, and privacy measures.
# OpenAI Realtime voice Use this skill when the deliverable is a live spoken interaction: a browser voice assistant, phone-like agent, low-latency speech-to-speech support flow, guided interview, tutoring conversation, live kiosk, or realtime voice UI that listens, reasons, speaks, interrupts, and may call tools. Do not use this skill for batch speech-to-text, offline TTS, voiceover generation, au
What does the openai-realtime-voice skill do?
Build production OpenAI Realtime voice agents and low-latency spoken interactions with live audio sessions, WebRTC or WebSocket transport, voice activity detection, tool/function calling, prompt design, logging, consent, privacy, safety, latency, and cost controls. Use for speech-to-speech agents and live voice UX; do not use for separate request-based transcription, text-to-speech, or offline audio generation work.
How do I install it?
Run `npx skills add calesthio/generative-media-skills --skill openai-realtime-voice --agent claude-code` — it drops the skill into your project so the agent can pick it up. Swap the --agent value for codex, cursor or copilot if you use one of those.
Where does this skill come from?
From calesthio/generative-media-skills, a repository with 112 stars. We read it straight from the repository tree rather than a submitted listing, so what you see here is what is actually published.
Is a popular skill a good skill?
Not necessarily. Stars measure attention, not adoption — a repository can trend for a week and be abandoned. That is why we show the weekly change from our own snapshots next to the total, instead of a single flattering number.