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PII Redaction

For the complete documentation index, see llms.txt

Overview

Streaming PII Redaction lets you automatically detect and remove personally identifiable information from your streaming transcripts in real time. When enabled, the API redacts PII in final turns only before sending them to the client.

PII redaction supports all streaming models: u3-rt-pro, universal-streaming-english, and universal-streaming-multilingual.

PII redaction only applies to final turns. When redact_pii is true, include_partial_turns defaults to false automatically so no unredacted text reaches the client. Only set include_partial_turns to true if you explicitly want partial (non-final) turns, which will contain unredacted PII alongside the redacted final turns.

When you enable PII redaction, your final turns will look like this:

With hash substitution: Hi, my name is ####!
With entity_name substitution: Hi, my name is [PERSON_NAME]!

For PII redaction on pre-recorded audio, including generating redacted audio files, see Redact PII from transcripts.

Connection parameters

Parameter	Type	Required	Default	Description
`redact_pii`	boolean	Yes	`false`	Enable PII text redaction. Only applies to final turns.
`redact_pii_policies`	array	No	All	PII entity types to redact. Over the raw WebSocket, pass a JSON-encoded array of policy names (e.g. `["person_name","phone_number"]`). The SDKs accept a native list/array. If omitted and `redact_pii` is `true`, all detected PII is redacted. See PII policies for the full list.
`redact_pii_sub`	string	No	`hash`	Replacement scheme. `hash` replaces PII with `#` characters, `entity_name` replaces with `[ENTITY_TYPE]`.
`include_partial_turns`	boolean	No	`false` when `redact_pii` is `true`, otherwise `true`	Whether to include partial (non-final) turns. Defaults to `false` automatically when PII redaction is enabled, so no unredacted text reaches the client. Set to `true` only if you explicitly want to receive partial turns, which will contain unredacted PII.

Quickstart

Get started with streaming PII redaction using the code below. This example streams audio from your microphone and prints each turn with PII redacted.

Install the required libraries

bash

pip install websocket-client pyaudio

Create a new file main.py and paste the code below. Replace `` with your API key.

Run with python main.py and speak into your microphone.

python

import pyaudio
import websocket
import json
import threading
import time
from urllib.parse import urlencode

YOUR_API_KEY = "<YOUR_API_KEY>"
CONNECTION_PARAMS = {
    "sample_rate": 16000,
    "speech_model": "u3-rt-pro",
    "format_turns": "true",
    "redact_pii": "true",
    "redact_pii_policies": json.dumps(["person_name", "phone_number", "email_address"]),
    "redact_pii_sub": "entity_name",
}
API_ENDPOINT_BASE_URL = "wss://streaming.assemblyai.com/v3/ws"
API_ENDPOINT = f"{API_ENDPOINT_BASE_URL}?{urlencode(CONNECTION_PARAMS)}"

FRAMES_PER_BUFFER = 800
SAMPLE_RATE = CONNECTION_PARAMS["sample_rate"]
CHANNELS = 1
FORMAT = pyaudio.paInt16

audio = None
stream = None
ws_app = None
audio_thread = None
stop_event = threading.Event()

def on_open(ws):
    print("WebSocket connection opened.")

    def stream_audio():
        global stream
        while not stop_event.is_set():
            try:
                audio_data = stream.read(FRAMES_PER_BUFFER, exception_on_overflow=False)
                ws.send(audio_data, websocket.ABNF.OPCODE_BINARY)
            except Exception as e:
                print(f"Error streaming audio: {e}")
                break

    global audio_thread
    audio_thread = threading.Thread(target=stream_audio)
    audio_thread.daemon = True
    audio_thread.start()

def on_message(ws, message):
    try:
        data = json.loads(message)
        msg_type = data.get("type")
        if msg_type == "Begin":
            print(f"Session began: ID={data.get('id')}")
        elif msg_type == "Turn":
            transcript = data.get("transcript", "")
            end_of_turn = data.get("end_of_turn", False)
            if end_of_turn:
                print(f"\r{' ' * 80}\r{transcript}")
        elif msg_type == "Termination":
            print(f"\nSession terminated: {data.get('audio_duration_seconds', 0)}s of audio")
    except Exception as e:
        print(f"Error handling message: {e}")

def on_error(ws, error):
    print(f"\nWebSocket Error: {error}")
    stop_event.set()

def on_close(ws, close_status_code, close_msg):
    print(f"\nWebSocket Disconnected: Status={close_status_code}")
    global stream, audio
    stop_event.set()
    if stream:
        if stream.is_active():
            stream.stop_stream()
        stream.close()
    if audio:
        audio.terminate()

def run():
    global audio, stream, ws_app
    audio = pyaudio.PyAudio()
    stream = audio.open(
        input=True,
        frames_per_buffer=FRAMES_PER_BUFFER,
        channels=CHANNELS,
        format=FORMAT,
        rate=SAMPLE_RATE,
    )
    print("Speak into your microphone. Press Ctrl+C to stop.")
    ws_app = websocket.WebSocketApp(
        API_ENDPOINT,
        header={"Authorization": YOUR_API_KEY},
        on_open=on_open,
        on_message=on_message,
        on_error=on_error,
        on_close=on_close,
    )
    ws_thread = threading.Thread(target=ws_app.run_forever)
    ws_thread.daemon = True
    ws_thread.start()
    try:
        while ws_thread.is_alive():
            time.sleep(0.1)
    except KeyboardInterrupt:
        print("\nStopping...")
        stop_event.set()
        if ws_app and ws_app.sock and ws_app.sock.connected:
            ws_app.send(json.dumps({"type": "Terminate"}))
            time.sleep(2)
        if ws_app:
            ws_app.close()
        ws_thread.join(timeout=2.0)

if __name__ == "__main__":
    run()

Install the required libraries

bash

pip install "assemblyai>=0.64.0" pyaudio

Create a new file main.py and paste the code below. Replace `` with your API key.

Run with python main.py and speak into your microphone.

python

import logging
from typing import Type
import assemblyai as aai
from assemblyai.streaming.v3 import (
    BeginEvent,
    StreamingClient,
    StreamingClientOptions,
    StreamingError,
    StreamingEvents,
    StreamingParameters,
    TurnEvent,
    TerminationEvent,
)

api_key = "<YOUR_API_KEY>"
logging.basicConfig(level=logging.INFO)
logger = logging.getLogger(__name__)

def on_begin(self: Type[StreamingClient], event: BeginEvent):
    print(f"Session started: {event.id}")

def on_turn(self: Type[StreamingClient], event: TurnEvent):
    print(f"{event.transcript} (end_of_turn={event.end_of_turn})")

def on_terminated(self: Type[StreamingClient], event: TerminationEvent):
    print(
        f"Session terminated: {event.audio_duration_seconds} seconds of audio processed"
    )

def on_error(self: Type[StreamingClient], error: StreamingError):
    print(f"Error occurred: {error}")

def main():
    client = StreamingClient(
        StreamingClientOptions(
            api_key=api_key,
            api_host="streaming.assemblyai.com",
        )
    )
    client.on(StreamingEvents.Begin, on_begin)
    client.on(StreamingEvents.Turn, on_turn)
    client.on(StreamingEvents.Termination, on_terminated)
    client.on(StreamingEvents.Error, on_error)

    client.connect(
        StreamingParameters(
            sample_rate=16000,
            speech_model="u3-rt-pro",
            format_turns=True,
            redact_pii=True,
            redact_pii_policies=["person_name", "phone_number", "email_address"],
            redact_pii_sub="entity_name",
        )
    )
    try:
        client.stream(
            aai.extras.MicrophoneStream(sample_rate=16000)
        )
    finally:
        client.disconnect(terminate=True)

if __name__ == "__main__":
    main()

Install the required libraries

bash

npm install ws

Create a new file index.mjs and paste the code below. Replace `` with your API key.

Run with node index.mjs and speak into your microphone.

javascript

import WebSocket from "ws";

const YOUR_API_KEY = "<YOUR_API_KEY>";
const params = new URLSearchParams({
  sample_rate: "16000",
  speech_model: "u3-rt-pro",
  format_turns: "true",
  redact_pii: "true",
  redact_pii_policies: JSON.stringify(["person_name", "phone_number", "email_address"]),
  redact_pii_sub: "entity_name",
});

const url = `wss://streaming.assemblyai.com/v3/ws?${params}`;
const ws = new WebSocket(url, {
  headers: { Authorization: YOUR_API_KEY },
});

ws.on("open", () => {
  console.log("Connected to AssemblyAI Streaming API");
  // Stream audio data by sending binary frames:
  // ws.send(audioBuffer);
});

ws.on("message", (data) => {
  const msg = JSON.parse(data);
  if (msg.type === "Turn" && msg.end_of_turn) {
    console.log(msg.transcript);
  }
});

ws.on("error", (err) => console.error("WebSocket error:", err));
ws.on("close", () => console.log("Disconnected"));

Install the required libraries

bash

npm install assemblyai node-record-lpcm16

Create a new file index.mjs and paste the code below. Replace `` with your API key.

Run with node index.mjs and speak into your microphone.

javascript

import { AssemblyAI } from "assemblyai";
import recorder from "node-record-lpcm16";

const apiKey = "<YOUR_API_KEY>";
const SAMPLE_RATE = 16000;

const client = new AssemblyAI({ apiKey });

const transcriber = client.streaming.transcriber({
  sampleRate: SAMPLE_RATE,
  speechModel: "u3-rt-pro",
  formatTurns: true,
  redactPii: true,
  redactPiiPolicies: ["person_name", "phone_number", "email_address"],
  redactPiiSub: "entity_name",
});

transcriber.on("open", ({ id }) => {
  console.log(`Session started: ${id}`);
});

transcriber.on("turn", (turn) => {
  if (turn.end_of_turn) {
    console.log(turn.transcript);
  }
});

transcriber.on("close", (code, reason) => {
  console.log(`Session terminated: ${code} ${reason}`);
});

transcriber.on("error", (error) => {
  console.error(`Error occurred: ${error}`);
});

async function main() {
  await transcriber.connect();
  console.log("Speak into your microphone. Press Ctrl+C to stop.");

  const recording = recorder.record({
    channels: 1,
    sampleRate: SAMPLE_RATE,
    audioType: "raw",
  });

  recording.stream().on("data", (chunk) => transcriber.sendAudio(chunk));

  process.on("SIGINT", async () => {
    recording.stop();
    await transcriber.close(true);
    process.exit(0);
  });
}

main();

Example output

With entity_name substitution:

plain

Hi, my name is [PERSON_NAME] and you can reach me at [PHONE_NUMBER] or [EMAIL_ADDRESS].

With hash substitution:

plain

Hi, my name is #### and you can reach me at ###-###-#### or ####@#####.###.

Supported PII policies

Streaming PII redaction supports the same policies as pre-recorded PII redaction, including person_name, phone_number, email_address, credit_card_number, us_social_security_number, date_of_birth, and more.

For the full list of available policies, see PII policies.

Troubleshooting

PII redaction only applies to final turns. If you're seeing PII, you likely set include_partial_turns to true, which returns unredacted partial turns alongside redacted finals. Remove that override (or set it to false) to only receive redacted final turns — this is the default when redact_pii is enabled.

Audio redaction is not available for streaming. To generate a redacted audio file, use pre-recorded PII redaction with the redact_pii_audio parameter.

PII Redaction ​

Overview ​

Connection parameters ​

Quickstart ​

Example output ​

Supported PII policies ​

Troubleshooting ​

PII Redaction

Overview

Connection parameters

Quickstart

Example output

Supported PII policies

Troubleshooting