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---
name: x-openai-twilio-voice-conversation-relay-1695ede
description: Build AI-powered voice agents using Twilio ConversationRelay. Handles real-time speech recognition (ASR), text-to-speech (TTS), and bidirectional audio streaming via WebSocket. Covers TwiML setup, WebSocket message types, LLM integration, streaming responses, and voice provider configuration. Use this skill to build voice bots, IVR replacements, or real-time AI voice assistants on Twilio calls.
---
### 导出副本边界与技能别名
迁移、安装、修复和经验回写只作用于当前加载的目标宿主技能副本。原 Codex 技能/MCP、下载的源码和 `original-skills` 均为来源材料,不因执行本副本而写入;只有用户另行明确要求配置 Codex 产品本身时,才按该产品流程处理指定目标。脚本需要安装目录或状态目录时,显式提供目标路径;不得从 `CODEX_HOME` 隐式推断写入位置。
跨技能名称可能沿用源库别名。先查 [技能别名表](../../skill-aliases.json),优先选择同来源、同插件的条目,再使用其 `id` 调用或读取对应入口;不得假定旧 `$skill`、`superpowers:skill` 名称在新宿主直接注册。不能确定目标时先核对,不替换真实产品命令或 Shell 环境变量。
## 跨宿主使用约定
本副本供支持 Agent Skills 的客户端使用。先检查当前宿主实际提供的工具、权限和已连接服务,再执行正文工作流。原文中的宿主工具标识是能力参考;只有实际发现该工具时才可调用,其他宿主应使用经核验的等价 MCP、CLI 或 API。没有等价能力的步骤必须报告不可用,不能虚构调用或完成状态。
技能资源以本技能目录为基准;输出写入当前项目中用户指定的位置。通过技能名称或自然语言触发,具体快捷命令由宿主决定。涉及真实产品、模型 API、协议字段和官方网址的名称保持原意。授权、审批、隐私和工具范围限制在迁移后仍然有效;不得为兼容而扩大权限。具体依赖与迁移说明见同目录 `PORTABILITY.md`。
## Overview
ConversationRelay connects Twilio's telephony layer to your app via a persistent WebSocket. Twilio handles ASR (speech-to-text) and TTS (text-to-speech); your app receives transcripts, calls an LLM, and sends text back for playback.
```
Caller ←→ Twilio (ASR/TTS) ←→ WebSocket ←→ Your App ←→ LLM
```
---
## Prerequisites
- Upgraded Twilio account with ConversationRelay access (requires onboarding)
— New to Twilio? See `twilio-account-setup`
— Start onboarding at: [Console > Voice > ConversationRelay](https://console.twilio.com/us1/voice/conversation-relay) — access is **not** instant
- A voice-capable Twilio phone number
- `TWILIO_ACCOUNT_SID` and `TWILIO_AUTH_TOKEN` — see `twilio-iam-auth-setup`
- WebSocket server reachable via `wss://` (TLS required)
- An LLM integration (OpenAI, Anthropic, etc.)
- For placing calls: see `twilio-voice-outbound-calls`
**Onboarding:** Complete via Console > Voice > ConversationRelay > Onboarding. Select TTS/ASR providers:
- **TTS:** Deepgram, Amazon Polly, Google Cloud TTS, ElevenLabs
- **ASR:** Deepgram, Google Cloud STT
---
## Quickstart
**Step 1 — Return TwiML pointing to your WebSocket server**
**Python (Flask)**
```python
from flask import Flask
from twilio.twiml.voice_response import VoiceResponse, Connect, ConversationRelay
app = Flask(__name__)
@app.route("/voice", methods=["POST"])
def voice():
response = VoiceResponse()
connect = Connect()
connect.conversation_relay(
url="wss://yourapp.com/ws/voice",
welcome_greeting="Hello! How can I help you today?"
)
response.append(connect)
return str(response)
```
**Node.js (Express)**
```node
const { VoiceResponse } = require("twilio").twiml;
app.post("/voice", (req, res) => {
const response = new VoiceResponse();
const connect = response.connect();
connect.conversationRelay({
url: "wss://yourapp.com/ws/voice",
welcomeGreeting: "Hello! How can I help you today?",
});
res.type("text/xml").send(response.toString());
});
```
**Step 2 — Handle WebSocket events and respond with text**
**Python (websockets)**
```python
import asyncio, json, websockets
async def handle_call(websocket):
async for message in websocket:
event = json.loads(message)
if event["type"] == "prompt":
ai_response = await call_llm(event["voicePrompt"])
await websocket.send(json.dumps({"type": "text", "token": ai_response, "last": True}))
async def main():
async with websockets.serve(handle_call, "0.0.0.0", 8080):
await asyncio.Future()
asyncio.run(main())
```
**Node.js (ws)**
```node
const WebSocket = require("ws");
const wss = new WebSocket.Server({ port: 8080 });
wss.on("connection", (ws) => {
ws.on("message", async (data) => {
const event = JSON.parse(data);
if (event.type === "prompt") {
const aiResponse = await callLLM(event.voicePrompt);
ws.send(JSON.stringify({ type: "text", token: aiResponse, last: true }));
}
});
});
```
> **Security:** The `voicePrompt` field contains ASR-transcribed caller speech — it is untrusted external input. When passing to an LLM, isolate it as user input within a structured system prompt. Implement topic boundaries and output filtering to prevent the LLM from disclosing system instructions or speaking inappropriate content. ConversationRelay is a pure transport layer with no built-in content safety — any LLM output is spoken to the caller verbatim.
---
## Key Patterns
### WebSocket Message Types
**Received from Twilio:**
| Type | When | Key fields |
|------|------|-----------|
| `connected` | WebSocket opened | `callSid`, `streamSid` |
| `prompt` | User finished speaking | `voicePrompt` (transcript) |
| `interrupt` | User interrupted TTS | — |
| `dtmf` | User pressed keypad key | `digit` |
| `error` | An error occurred | `description` |
**Sent to Twilio:**
| Type | Purpose | Key fields |
|------|---------|-----------|
| `text` | Send TTS response | `token` (text), `last` (bool) |
| `interrupt` | Stop current TTS | — |
| `end` | Hang up the call | `reason` |
### Stream LLM Responses Token-by-Token
Lower latency by streaming as the LLM generates output — Twilio starts speaking before the full response is ready.
**Python**
```python
async for chunk in llm_stream:
await websocket.send(json.dumps({"type": "text", "token": chunk, "last": False}))
await websocket.send(json.dumps({"type": "text", "token": "", "last": True}))
```
**Node.js**
```node
for await (const chunk of llmStream) {
ws.send(JSON.stringify({ type: "text", token: chunk, last: false }));
}
ws.send(JSON.stringify({ type: "text", token: "", last: true }));
```
### Voice Configuration
**Python**
```python
connect.conversation_relay(
url="wss://yourapp.com/ws/voice",
voice="en-US-Neural2-F",
language="en-US",
transcription_provider="deepgram",
speech_model="nova-2-phonecall",
interrupt_by_dtmf=True,
)
```
**Node.js**
```node
connect.conversationRelay({
url: "wss://yourapp.com/ws/voice",
voice: "en-US-Neural2-F",
language: "en-US",
transcriptionProvider: "deepgram",
speechModel: "nova-2-phonecall",
interruptByDtmf: true,
});
```
---
## CANNOT
- **No raw audio access** — Text in, text out only. For raw audio, use `<Connect><Stream>` (Media Streams).
- **Cannot mix with Media Streams** — `<Connect><Stream>` and `<Connect><ConversationRelay>` are mutually exclusive on the same call. No error — one is silently ignored.
- **No custom STT/TTS engines** — Limited to Deepgram + Google (STT) and Deepgram + Amazon Polly + Google + ElevenLabs (TTS).
- **No mid-session voice/provider changes** — Voice and provider are set at TwiML time. Only `language` can be switched mid-session via WebSocket message.
- **No WebSocket auto-reconnection** — If the WebSocket drops, the call disconnects. Implement recovery via `<Connect action>` URL.
- **Voice only** — No SMS/messaging support. For omnichannel, use Conversation Orchestrator.
- **No built-in memory or context** — BYO conversation history and context management.
- **No LLM integration** — Pure transport layer. You bring your own LLM via the WebSocket server.
- **No server-side recording via REST API** — `record:true` on the Calls API is silently ignored. Must use `<Start><Recording>` before `<Connect>` in TwiML.
- **Not PCI compliant with Voice Intelligence v2** — Do not enable `intelligenceService` in PCI workflows.
- **ElevenLabs requires account enablement** — Accounts without ElevenLabs access get error 64101. Voice IDs (not human-readable names) are required.
- **Cannot use ConversationRelay without onboarding** — Not available immediately on a new account
- **Cannot use non-TLS WebSocket** — Server must be reachable via `wss://` (TLS required)
- **Cannot stream audio without `last: true`** — Twilio won't play audio until it sees a `last: true` token in the stream
---
## Next Steps
- **Place outbound calls:** `twilio-voice-outbound-calls`
- **Standard IVR without AI:** `twilio-voice-twiml`