Hidden assistant demo

.pre-commit-config.yaml

@@ -4,5 +4,5 @@ repos:
     hooks:
       - id: ruff
         language_version: python3
-        args: [--fix]
+        args: [ --select,  I, ]
       - id: ruff-format

CHANGELOG.md

@@ -5,104 +5,21 @@ All notable changes to **Pipecat** will be documented in this file.
 The format is based on [Keep a Changelog](https://keepachangelog.com/en/1.0.0/),
 and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0.html).
 
-## [0.0.76] - 2025-07-11
+## [Unreleased]
 
 ### Added
 
-- Added `SpeechControlParamsFrame`, a new `SystemFrame` that notifies
-  downstream processors of the VAD and Turn analyzer params. This frame is
-  pushed by the `BaseInputTransport` at Start and any time a
-  `VADParamsUpdateFrame` is received.
-
-### Changed
-
-- Two package dependencies have been updated:
-  - `numpy` now supports 1.26.0 and newer
-  - `transformers` now supports 4.48.0 and newer
-
-### Fixed
-
-- Fixed an issue with RTVI's handling of `append-to-context`.
-
-- Fixed an issue where using audio input with a sample rate requiring resampling
-  could result in empty audio being passed to STT services, causing errors.
-
-- Fixed the VAD analyzer to process the full audio buffer as long as it contains
-  more than the minimum required bytes per iteration, instead of only analyzing
-  the first chunk.
-
-- Fixed an issue in ParallelPipeline that caused errors when attempting to drain
-  the queues.
-
-- Fixed an issue with emulated VAD timeout inconsistency in
-  `LLMUserContextAggregator`. Previously, emulated VAD scenarios (where
-  transcription is received without VAD detection) used a hardcoded
-  `aggregation_timeout` (default 0.5s) instead of matching the VAD's
-  `stop_secs` parameter (default 0.8s). This created different user experiences
-  between real VAD and emulated VAD scenarios. Now, emulated VAD timeouts
-  automatically synchronize with the VAD's `stop_secs` parameter.
-
-- Fix a pipeline freeze when using AWS Nova Sonic, which would occur if the
-  user started early, while the bot was still working through
-  `trigger_assistant_response()`.
-
-## [0.0.75] - 2025-07-08
-
-### Added
-
-- Added an `aggregate_sentences` arg in `CartesiaTTSService`,
-  `ElevenLabsTTSService`, `NeuphonicTTSService` and `RimeTTSService`, where the
-  default value is True. When `aggregate_sentences` is True, the `TTSService`
-  aggregates the LLM streamed tokens into sentences by default. Note: setting
-  the value to False requires a custom processor before the `TTSService` to
-  aggregate LLM tokens.
-
-- Added `kwargs` to the `OLLamaLLMService` to allow for configuration args to
-  be passed to Ollama.
-
 - Added call hang-up error handling in `TwilioFrameSerializer`, which handles
   the case where the user has hung up before the `TwilioFrameSerializer` hangs
   up the call.
 
 ### Changed
 
-- Updated `RTVIObserver` and `RTVIProcessor` to match the new RTVI 1.0.0 protocol.
-  This includes:
-
-  - Deprecating support for all messages related to service configuaration and
-    actions.
-  - Adding support for obtaining and logging data about client, including its
-    RTVI version and optionally included system information (OS/browser/etc.)
-  - Adding support for handling the new `client-message` RTVI message through
-    either a `on_client_message` event handler or listening for a new
-    `RTVIClientMessageFrame`
-  - Adding support for responding to a `client-message` with a `server-response`
-    via either a direct call on the `RTVIProcessor` or via pushing a new
-    `RTVIServerResponseFrame`
-  - Adding built-in support for handling the new `append-to-context` RTVI message
-    which allows a client to add to the user or assistant llm context. No extra
-    code is required for supporting this behavior.
-  - Updating all JavaScript and React client RTVI examples to use versions 1.0.0
-    of the clients.
-
-  Get started migrating to RTVI protocol 1.0.0 by following the migration guide:
-  https://docs.pipecat.ai/client/migration-guide
-
-- Refactored `AWSBedrockLLMService` and `AWSPollyTTSService` to work
-  asynchronously using `aioboto3` instead of the `boto3` library.
-
 - The `UserIdleProcessor` now handles the scenario where function calls take
   longer than the idle timeout duration. This allows you to use the
   `UserIdleProcessor` in conjunction with function calls that take a while to
   return a result.
 
-### Fixed
-
-- Updated the `NeuphonicTTSService` to work with the updated websocket API.
-
-- Fixed an issue with `RivaSTTService` where the watchdog feature was causing
-  an error on initialization.
-
 ### Performance
 
 - Remove unncessary push task in each `FrameProcessor`.

examples/deployment/modal-example/client/javascript/package.json

@@ -15,7 +15,7 @@
     "vite": "^6.3.5"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/daily-transport": "^1.0.0"
+    "@pipecat-ai/client-js": "^0.3.5",
+    "@pipecat-ai/daily-transport": "^0.3.10"
   }
 }

examples/deployment/modal-example/client/javascript/src/app.js

@@ -5,7 +5,7 @@
  */
 
 /**
- * Pipecat Client Implementation
+ * RTVI Client Implementation
  *
  * This client connects to an RTVI-compatible bot server using WebRTC (via Daily).
  * It handles audio/video streaming and manages the connection lifecycle.
@@ -16,7 +16,7 @@
  * - Browser with WebRTC support
  */
 
-import { PipecatClient, RTVIEvent } from '@pipecat-ai/client-js';
+import { RTVIClient, RTVIEvent } from '@pipecat-ai/client-js';
 import { DailyTransport } from '@pipecat-ai/daily-transport';
 
 /**
@@ -26,7 +26,7 @@ import { DailyTransport } from '@pipecat-ai/daily-transport';
 class ChatbotClient {
   constructor() {
     // Initialize client state
-    this.pcClient = null;
+    this.rtviClient = null;
     this.setupDOMElements();
     this.initializeClientAndTransport();
     this.setupEventListeners();
@@ -59,7 +59,7 @@ class ChatbotClient {
     this.disconnectBtn.addEventListener('click', () => this.disconnect());
 
     // Populate device selector
-    this.pcClient.getAllMics().then((mics) => {
+    this.rtviClient.getAllMics().then((mics) => {
       console.log('Available mics:', mics);
       mics.forEach((device) => {
         const option = document.createElement('option');
@@ -71,16 +71,16 @@ class ChatbotClient {
     this.deviceSelector.addEventListener('change', (event) => {
       const selectedDeviceId = event.target.value;
       console.log('Selected device ID:', selectedDeviceId);
-      this.pcClient.updateMic(selectedDeviceId);
+      this.rtviClient.updateMic(selectedDeviceId);
     });
 
     // Handle mic mute/unmute toggle
     const micToggleBtn = document.getElementById('mic-toggle-btn');
 
     micToggleBtn.addEventListener('click', () => {
-      let micEnabled = this.pcClient.isMicEnabled;
+      let micEnabled = this.rtviClient.isMicEnabled;
       micToggleBtn.textContent = micEnabled ? 'Unmute Mic' : 'Mute Mic';
-      this.pcClient.enableMic(!micEnabled);
+      this.rtviClient.enableMic(!micEnabled);
       // Add logic to mute/unmute the mic
       if (micEnabled) {
         console.log('Mic muted');
@@ -93,12 +93,23 @@ class ChatbotClient {
   }
 
   /**
-   * Set up the Pipecat client and Daily transport
+   * Set up the RTVI client and Daily transport
    */
   async initializeClientAndTransport() {
-    // Initialize the Pipecat client with a DailyTransport and our configuration
-    this.pcClient = new PipecatClient({
+    // Initialize the RTVI client with a DailyTransport and our configuration
+    this.rtviClient = new RTVIClient({
       transport: new DailyTransport(),
+      params: {
+        // REPLACE WITH YOUR MODAL URL ENDPOINT
+        baseUrl:
+          'https://<Modal workspace>--pipecat-modal-bot-launcher.modal.run',
+        endpoints: {
+          connect: '/connect',
+        },
+        requestData: {
+          bot_name: 'openai',
+        },
+      },
       enableMic: true, // Enable microphone for user input
       enableCam: false,
       callbacks: {
@@ -165,8 +176,8 @@ class ChatbotClient {
     // Set up listeners for media track events
     this.setupTrackListeners();
 
-    await this.pcClient.initDevices();
-    window.client = this.pcClient;
+    await this.rtviClient.initDevices();
+    window.client = this.rtviClient;
   }
 
   /**
@@ -201,10 +212,10 @@ class ChatbotClient {
    * This is called when the bot is ready or when the transport state changes to ready
    */
   setupMediaTracks() {
-    if (!this.pcClient) return;
+    if (!this.rtviClient) return;
 
     // Get current tracks from the client
-    const tracks = this.pcClient.tracks();
+    const tracks = this.rtviClient.tracks();
 
     // Set up any available bot tracks
     if (tracks.bot?.audio) {
@@ -220,10 +231,10 @@ class ChatbotClient {
    * This handles new tracks being added during the session
    */
   setupTrackListeners() {
-    if (!this.pcClient) return;
+    if (!this.rtviClient) return;
 
     // Listen for new tracks starting
-    this.pcClient.on(RTVIEvent.TrackStarted, (track, participant) => {
+    this.rtviClient.on(RTVIEvent.TrackStarted, (track, participant) => {
       // Only handle non-local (bot) tracks
       if (!participant?.local) {
         if (track.kind === 'audio') {
@@ -242,7 +253,7 @@ class ChatbotClient {
     });
 
     // Listen for tracks stopping
-    this.pcClient.on(RTVIEvent.TrackStopped, (track, participant) => {
+    this.rtviClient.on(RTVIEvent.TrackStopped, (track, participant) => {
       if (participant.local) {
         this.log('Local mic muted');
         return;
@@ -300,27 +311,21 @@ class ChatbotClient {
 
   /**
    * Initialize and connect to the bot
-   * This sets up the Pipecat client, initializes devices, and establishes the connection
+   * This sets up the RTVI client, initializes devices, and establishes the connection
    */
   async connect() {
     try {
       const botSelector = document.getElementById('bot-selector');
       const selectedBot = botSelector.value;
+      this.rtviClient.params.requestData.bot_name = selectedBot;
 
       // Initialize audio/video devices
       this.log('Initializing devices...');
-      await this.pcClient.initDevices();
+      await this.rtviClient.initDevices();
 
       // Connect to the bot
       this.log(`Connecting to bot: ${selectedBot}`);
-      await this.pcClient.connect({
-        // REPLACE WITH YOUR MODAL URL ENDPOINT
-        endpoint:
-          'https://<your-workspace>--pipecat-modal-fastapi-app.modal.run/connect',
-        requestData: {
-          bot_name: selectedBot,
-        },
-      });
+      await this.rtviClient.connect();
 
       this.log('Connection complete');
     } catch (error) {
@@ -331,9 +336,9 @@ class ChatbotClient {
       this.updateStatus('Error');
 
       // Clean up if there's an error
-      if (this.pcClient) {
+      if (this.rtviClient) {
         try {
-          await this.pcClient.disconnect();
+          await this.rtviClient.disconnect();
         } catch (disconnectError) {
           this.log(`Error during disconnect: ${disconnectError.message}`);
         }
@@ -345,10 +350,10 @@ class ChatbotClient {
    * Disconnect from the bot and clean up media resources
    */
   async disconnect() {
-    if (this.pcClient) {
+    if (this.rtviClient) {
       try {
-        // Disconnect the Pipecat client
-        await this.pcClient.disconnect();
+        // Disconnect the RTVI client
+        await this.rtviClient.disconnect();
 
         // Clean up audio
         if (this.botAudio.srcObject) {

examples/deployment/modal-example/server/requirements.txt

@@ -1,3 +1,2 @@
 python-dotenv==1.0.1
-modal==1.0.5
-fastapi[all]
+modal==0.71.3

examples/deployment/pipecat-cloud-daily-pstn-server/nextjs-webhook-server/pages/api/dial.js

@@ -103,7 +103,7 @@ export default async function handler(req, res) {
       const sip_config = {
         display_name: From,
         sip_mode: 'dial-in',
-        num_endpoints: (call_transfer !== undefined && call_transfer !== null) ? 2 : 1,
+        num_endpoints: call_transfer !== null ? 2 : 1,
         codecs: {"audio": ["OPUS"]},
       };
       daily_room_properties.sip = sip_config;

examples/fal-smart-turn/README.md

@@ -44,7 +44,7 @@ Try the hosted version of the demo here: https://pcc-smart-turn.vercel.app/.
 4. Run the server:
 
    ```bash
-   LOCAL_RUN=1 python server.py
+   LOCAL=1 python server.py
    ```
 
 ### Run the client

examples/fal-smart-turn/client/package.json

@@ -9,9 +9,9 @@
     "lint": "next lint"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/client-react": "^1.0.0",
-    "@pipecat-ai/daily-transport": "^1.0.0",
+    "@pipecat-ai/client-js": "^0.3.5",
+    "@pipecat-ai/client-react": "^0.3.5",
+    "@pipecat-ai/daily-transport": "^0.3.10",
     "next": "15.3.1",
     "react": "^19.0.0",
     "react-dom": "^19.0.0"

examples/fal-smart-turn/client/src/app/layout.tsx

@@ -1,5 +1,5 @@
 import './globals.css';
-import { PipecatProvider } from '@/providers/PipecatProvider';
+import { RTVIProvider } from '@/providers/RTVIProvider';
 
 export const metadata = {
   title: 'Pipecat React Client',
@@ -20,7 +20,7 @@ export default function RootLayout({
         <link rel="icon" href="/favicon.svg" type="image/svg+xml" />
       </head>
       <body>
-        <PipecatProvider>{children}</PipecatProvider>
+        <RTVIProvider>{children}</RTVIProvider>
       </body>
     </html>
   );

examples/fal-smart-turn/client/src/app/page.tsx

@@ -1,22 +1,22 @@
 'use client';
 
 import {
-  PipecatClientAudio,
-  PipecatClientVideo,
-  usePipecatClientTransportState,
+  RTVIClientAudio,
+  RTVIClientVideo,
+  useRTVIClientTransportState,
 } from '@pipecat-ai/client-react';
 import { ConnectButton } from '../components/ConnectButton';
 import { StatusDisplay } from '../components/StatusDisplay';
 import { DebugDisplay } from '../components/DebugDisplay';
 
 function BotVideo() {
-  const transportState = usePipecatClientTransportState();
+  const transportState = useRTVIClientTransportState();
   const isConnected = transportState !== 'disconnected';
 
   return (
     <div className="bot-container">
       <div className="video-container">
-        {isConnected && <PipecatClientVideo participant="bot" fit="cover" />}
+        {isConnected && <RTVIClientVideo participant="bot" fit="cover" />}
       </div>
     </div>
   );
@@ -35,7 +35,7 @@ export default function Home() {
       </div>
 
       <DebugDisplay />
-      <PipecatClientAudio />
+      <RTVIClientAudio />
     </div>
   );
 }

examples/fal-smart-turn/client/src/components/ConnectButton.tsx

@@ -1,17 +1,11 @@
 import {
-  usePipecatClient,
-  usePipecatClientTransportState,
+  useRTVIClient,
+  useRTVIClientTransportState,
 } from '@pipecat-ai/client-react';
 
-// Get the API base URL from environment variables
-// Default to "/api" if not specified
-// "/api" is the default for Next.js API routes and used
-// for the Pipecat Cloud deployed agent
-const API_BASE_URL = process.env.NEXT_PUBLIC_API_BASE_URL || '/api';
-
 export function ConnectButton() {
-  const client = usePipecatClient();
-  const transportState = usePipecatClientTransportState();
+  const client = useRTVIClient();
+  const transportState = useRTVIClientTransportState();
   const isConnected = ['connected', 'ready'].includes(transportState);
 
   const handleClick = async () => {
@@ -24,10 +18,7 @@ export function ConnectButton() {
       if (isConnected) {
         await client.disconnect();
       } else {
-        await client.connect({
-          endpoint: `${API_BASE_URL}/connect`,
-          requestData: { foo: 'bar' },
-        });
+        await client.connect();
       }
     } catch (error) {
       console.error('Connection error:', error);

examples/fal-smart-turn/client/src/components/DebugDisplay.tsx

@@ -6,7 +6,7 @@ import {
   TranscriptData,
   BotLLMTextData,
 } from '@pipecat-ai/client-js';
-import { usePipecatClient, useRTVIClientEvent } from '@pipecat-ai/client-react';
+import { useRTVIClient, useRTVIClientEvent } from '@pipecat-ai/client-react';
 import './DebugDisplay.css';
 
 interface SmartTurnResultData {
@@ -20,7 +20,7 @@ interface SmartTurnResultData {
 
 export function DebugDisplay() {
   const debugLogRef = useRef<HTMLDivElement>(null);
-  const client = usePipecatClient();
+  const client = useRTVIClient();
 
   const log = useCallback((message: string) => {
     if (!debugLogRef.current) return;

examples/fal-smart-turn/client/src/components/StatusDisplay.tsx

@@ -1,7 +1,7 @@
-import { usePipecatClientTransportState } from '@pipecat-ai/client-react';
+import { useRTVIClientTransportState } from '@pipecat-ai/client-react';
 
 export function StatusDisplay() {
-  const transportState = usePipecatClientTransportState();
+  const transportState = useRTVIClientTransportState();
 
   return (
     <div className="status">

examples/fal-smart-turn/client/src/providers/PipecatProvider.tsx

@@ -1,28 +0,0 @@
-'use client';
-
-import { PipecatClient } from '@pipecat-ai/client-js';
-import { DailyTransport } from '@pipecat-ai/daily-transport';
-import { PipecatClientProvider } from '@pipecat-ai/client-react';
-import { PropsWithChildren, useEffect, useState } from 'react';
-
-export function PipecatProvider({ children }: PropsWithChildren) {
-  const [client, setClient] = useState<PipecatClient | null>(null);
-
-  useEffect(() => {
-    const pcClient = new PipecatClient({
-      transport: new DailyTransport(),
-      enableMic: true,
-      enableCam: false,
-    });
-
-    setClient(pcClient);
-  }, []);
-
-  if (!client) {
-    return null;
-  }
-
-  return (
-    <PipecatClientProvider client={client}>{children}</PipecatClientProvider>
-  );
-}

examples/fal-smart-turn/client/src/providers/RTVIProvider.tsx

@@ -0,0 +1,43 @@
+'use client';
+
+import { RTVIClient } from '@pipecat-ai/client-js';
+import { DailyTransport } from '@pipecat-ai/daily-transport';
+import { RTVIClientProvider } from '@pipecat-ai/client-react';
+import { PropsWithChildren, useEffect, useState } from 'react';
+
+// Get the API base URL from environment variables
+// Default to "/api" if not specified
+// "/api" is the default for Next.js API routes and used
+// for the Pipecat Cloud deployed agent
+const API_BASE_URL = process.env.NEXT_PUBLIC_API_BASE_URL || '/api';
+
+console.log('Using API base URL:', API_BASE_URL);
+
+export function RTVIProvider({ children }: PropsWithChildren) {
+  const [client, setClient] = useState<RTVIClient | null>(null);
+
+  useEffect(() => {
+    const transport = new DailyTransport();
+
+    const rtviClient = new RTVIClient({
+      transport,
+      params: {
+        baseUrl: API_BASE_URL,
+        endpoints: {
+          connect: '/connect',
+        },
+        requestData: { foo: 'bar' },
+      },
+      enableMic: true,
+      enableCam: false,
+    });
+
+    setClient(rtviClient);
+  }, []);
+
+  if (!client) {
+    return null;
+  }
+
+  return <RTVIClientProvider client={client}>{children}</RTVIClientProvider>;
+}

examples/fal-smart-turn/server/bot.py

@@ -45,7 +45,7 @@ from pipecat.transports.services.daily import DailyParams, DailyTransport
 load_dotenv(override=True)
 
 # Check if we're in local development mode
-LOCAL = os.getenv("LOCAL_RUN")
+LOCAL = os.getenv("LOCAL")
 
 logger.remove()
 logger.add(sys.stderr, level="DEBUG")

examples/freeze-test/client/src/app.ts

@@ -20,10 +20,11 @@ import {
 } from '@pipecat-ai/client-js';
 import {
   ProtobufFrameSerializer,
-  WebSocketTransport,
-} from '@pipecat-ai/websocket-transport';
+  WebSocketTransport
+} from "@pipecat-ai/websocket-transport";
 
 class RecordingSerializer extends ProtobufFrameSerializer {
+
   private lastTimestamp: number | null = null;
   private recordingAudioToSend: boolean = false;
   private _recordedAudio: { data: ArrayBuffer; delay: number }[] = [];
@@ -39,11 +40,7 @@ class RecordingSerializer extends ProtobufFrameSerializer {
   }
 
   // @ts-ignore
-  serializeAudio(
-    data: ArrayBuffer,
-    sampleRate: number,
-    numChannels: number
-  ): Uint8Array | null {
+  serializeAudio(data: ArrayBuffer, sampleRate: number, numChannels: number): Uint8Array | null {
     if (this.recordingAudioToSend) {
       const now = Date.now();
       // Compute delay since last packet
@@ -58,13 +55,13 @@ class RecordingSerializer extends ProtobufFrameSerializer {
   }
 
   public get recordedAudio() {
-    return this._recordedAudio;
+    return this._recordedAudio
   }
 }
 
 class WebsocketClientApp {
-  private ENABLE_RECORDING_MODE = false;
-  private RECORDING_TIME_MS = 10000;
+  private ENABLE_RECORDING_MODE = false
+  private RECORDING_TIME_MS = 10000
 
   private rtviClient: RTVIClient | null = null;
   private connectBtn: HTMLButtonElement | null = null;
@@ -74,7 +71,7 @@ class WebsocketClientApp {
   private botAudio: HTMLAudioElement;
 
   private declare websocketTransport: WebSocketTransport;
-  private sendRecordedAudio: boolean = false;
+  private sendRecordedAudio: boolean = false
   private declare recordingSerializer: RecordingSerializer;
 
   private playBtn: HTMLButtonElement | null = null;
@@ -94,12 +91,8 @@ class WebsocketClientApp {
    * Set up references to DOM elements and create necessary media elements
    */
   private setupDOMElements(): void {
-    this.connectBtn = document.getElementById(
-      'connect-btn'
-    ) as HTMLButtonElement;
-    this.disconnectBtn = document.getElementById(
-      'disconnect-btn'
-    ) as HTMLButtonElement;
+    this.connectBtn = document.getElementById('connect-btn') as HTMLButtonElement;
+    this.disconnectBtn = document.getElementById('disconnect-btn') as HTMLButtonElement;
     this.statusSpan = document.getElementById('connection-status');
     this.debugLog = document.getElementById('debug-log');
     this.playBtn = document.getElementById('play-btn') as HTMLButtonElement;
@@ -112,12 +105,8 @@ class WebsocketClientApp {
   private setupEventListeners(): void {
     this.connectBtn?.addEventListener('click', () => this.connect());
     this.disconnectBtn?.addEventListener('click', () => this.disconnect());
-    this.playBtn?.addEventListener('click', () =>
-      this.startSendingRecordedAudio()
-    );
-    this.stopBtn?.addEventListener('click', () =>
-      this.stopSendingRecordedAudio()
-    );
+    this.playBtn?.addEventListener('click', () => this.startSendingRecordedAudio());
+    this.stopBtn?.addEventListener('click', () => this.stopSendingRecordedAudio());
   }
 
   /**
@@ -176,9 +165,7 @@ class WebsocketClientApp {
 
     // Listen for tracks stopping
     this.rtviClient.on(RTVIEvent.TrackStopped, (track, participant) => {
-      this.log(
-        `Track stopped: ${track.kind} from ${participant?.name || 'unknown'}`
-      );
+      this.log(`Track stopped: ${track.kind} from ${participant?.name || 'unknown'}`);
     });
   }
 
@@ -188,10 +175,7 @@ class WebsocketClientApp {
    */
   private setupAudioTrack(track: MediaStreamTrack): void {
     this.log('Setting up audio track');
-    if (
-      this.botAudio.srcObject &&
-      'getAudioTracks' in this.botAudio.srcObject
-    ) {
+    if (this.botAudio.srcObject && "getAudioTracks" in this.botAudio.srcObject) {
       const oldTrack = this.botAudio.srcObject.getAudioTracks()[0];
       if (oldTrack?.id === track.id) return;
     }
@@ -206,17 +190,27 @@ class WebsocketClientApp {
     try {
       const startTime = Date.now();
 
-      this.recordingSerializer = new RecordingSerializer();
-      const ws_opts = {
-        serializer: this.ENABLE_RECORDING_MODE
-          ? this.recordingSerializer
-          : new ProtobufFrameSerializer(),
-        recorderSampleRate: 8000,
-        playerSampleRate: 8000,
-      };
+      this.recordingSerializer = new RecordingSerializer()
+      const transport = this.ENABLE_RECORDING_MODE ?
+          new WebSocketTransport({
+            serializer: this.recordingSerializer,
+            recorderSampleRate: 8000,
+            playerSampleRate:8000
+          }) :
+          new WebSocketTransport({
+            serializer: new ProtobufFrameSerializer(),
+            recorderSampleRate: 8000,
+            playerSampleRate:8000
+          });
+      this.websocketTransport = transport
 
       const RTVIConfig: RTVIClientOptions = {
-        transport: new WebSocketTransport(ws_opts),
+        transport,
+        params: {
+          // The baseURL and endpoint of your bot server that the client will connect to
+          baseUrl: 'http://localhost:7860',
+          endpoints: { connect: '/connect' },
+        },
         enableMic: true,
         enableCam: false,
         callbacks: {
@@ -244,34 +238,27 @@ class WebsocketClientApp {
           onMessageError: (error) => console.error('Message error:', error),
           onError: (error) => console.error('Error:', error),
         },
-      };
+      }
       this.rtviClient = new RTVIClient(RTVIConfig);
-      this.websocketTransport = this.rtviClient.transport;
       this.setupTrackListeners();
 
       this.log('Initializing devices...');
       await this.rtviClient.initDevices();
 
       this.log('Connecting to bot...');
-      await this.rtviClient.connect({
-        endpoint: 'http://localhost:7860/connect',
-      });
+      await this.rtviClient.connect();
 
       const timeTaken = Date.now() - startTime;
       this.log(`Connection complete, timeTaken: ${timeTaken}`);
 
       if (this.ENABLE_RECORDING_MODE) {
-        this.log(
-          `Starting to recording the next ${
-            this.RECORDING_TIME_MS / 1000
-          }s of audio`
-        );
-        this.recordingSerializer.startRecording();
-        await this.sleep(this.RECORDING_TIME_MS);
-        this.recordingSerializer.stopRecording();
-        this.log('Recording stopped');
-        this.rtviClient.enableMic(false);
-        this.startSendingRecordedAudio();
+        this.log(`Starting to recording the next ${(this.RECORDING_TIME_MS/1000)}s of audio`);
+        this.recordingSerializer.startRecording()
+        await this.sleep(this.RECORDING_TIME_MS)
+        this.recordingSerializer.stopRecording()
+        this.log("Recording stopped");
+        this.rtviClient.enableMic(false)
+        this.startSendingRecordedAudio()
       }
     } catch (error) {
       this.log(`Error connecting: ${(error as Error).message}`);
@@ -293,16 +280,11 @@ class WebsocketClientApp {
   public async disconnect(): Promise<void> {
     if (this.rtviClient) {
       try {
-        this.stopSendingRecordedAudio();
+        this.stopSendingRecordedAudio()
         await this.rtviClient.disconnect();
         this.rtviClient = null;
-        if (
-          this.botAudio.srcObject &&
-          'getAudioTracks' in this.botAudio.srcObject
-        ) {
-          this.botAudio.srcObject
-            .getAudioTracks()
-            .forEach((track) => track.stop());
+        if (this.botAudio.srcObject && "getAudioTracks" in this.botAudio.srcObject) {
+          this.botAudio.srcObject.getAudioTracks().forEach((track) => track.stop());
           this.botAudio.srcObject = null;
         }
       } catch (error) {
@@ -312,21 +294,21 @@ class WebsocketClientApp {
   }
 
   private startSendingRecordedAudio() {
-    this.sendRecordedAudio = true;
+    this.sendRecordedAudio = true
     if (this.playBtn) this.playBtn.disabled = true;
     if (this.stopBtn) this.stopBtn.disabled = false;
-    void this.replayAudio();
+    void this.replayAudio()
   }
 
   private stopSendingRecordedAudio() {
     if (this.stopBtn) this.stopBtn.disabled = true;
     if (this.playBtn) this.playBtn.disabled = false;
-    this.sendRecordedAudio = false;
+    this.sendRecordedAudio = false
   }
 
   private async replayAudio() {
     if (this.sendRecordedAudio) {
-      this.log('Sending recorded audio');
+      this.log("Sending recorded audio")
       for (const chunk of this.recordingSerializer.recordedAudio) {
         await this.sleep(chunk.delay);
         this.websocketTransport.handleUserAudioStream(chunk.data);
@@ -334,13 +316,14 @@ class WebsocketClientApp {
       const randomDelay = 1000 + Math.random() * (10000 - 500);
       await this.sleep(randomDelay);
 
-      void this.replayAudio();
+      void this.replayAudio()
     }
   }
 
   private sleep(ms: number): Promise<void> {
-    return new Promise((resolve) => setTimeout(resolve, ms));
+    return new Promise(resolve => setTimeout(resolve, ms));
   }
+
 }
 
 declare global {

examples/instant-voice/client/javascript/package.json

@@ -18,7 +18,7 @@
     "vite": "^6.3.5"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/daily-transport": "^1.0.0"
+    "@pipecat-ai/client-js": "^0.3.5",
+    "@pipecat-ai/daily-transport": "^0.3.8"
   }
 }

examples/instant-voice/client/javascript/src/app.ts

@@ -5,7 +5,7 @@
  */
 
 /**
- * Pipecat Client Implementation
+ * RTVI Client Implementation
  *
  * This client connects to an RTVI-compatible bot server using WebRTC (via Daily).
  * It handles audio/video streaming and manages the connection lifecycle.
@@ -18,22 +18,20 @@
 
 import {
   Participant,
-  PipecatClient,
-  PipecatClientOptions,
+  RTVIClient,
+  RTVIClientOptions,
   RTVIEvent,
 } from '@pipecat-ai/client-js';
-import {
-  DailyEventCallbacks,
-  DailyTransport,
-} from '@pipecat-ai/daily-transport';
+import { DailyTransport } from '@pipecat-ai/daily-transport';
 import SoundUtils from './util/soundUtils';
+import { InstantVoiceHelper } from './util/instantVoiceHelper';
 
 /**
  * InstantVoiceClient handles the connection and media management for a real-time
  * voice and video interaction with an AI bot.
  */
 class InstantVoiceClient {
-  private declare pcClient: PipecatClient;
+  private declare rtviClient: RTVIClient;
   private connectBtn: HTMLButtonElement | null = null;
   private disconnectBtn: HTMLButtonElement | null = null;
   private statusSpan: HTMLElement | null = null;
@@ -48,7 +46,7 @@ class InstantVoiceClient {
     document.body.appendChild(this.botAudio);
     this.setupDOMElements();
     this.setupEventListeners();
-    this.initializePipecatClient();
+    this.initializeRTVIClient();
   }
 
   /**
@@ -74,11 +72,16 @@ class InstantVoiceClient {
     this.disconnectBtn?.addEventListener('click', () => this.disconnect());
   }
 
-  private initializePipecatClient(): void {
-    const PipecatConfig: PipecatClientOptions = {
+  private initializeRTVIClient(): void {
+    const RTVIConfig: RTVIClientOptions = {
       transport: new DailyTransport({
         bufferLocalAudioUntilBotReady: true,
       }),
+      params: {
+        // The baseURL and endpoint of your bot server that the client will connect to
+        baseUrl: 'http://localhost:7860',
+        endpoints: { connect: '/connect' },
+      },
       enableMic: true,
       enableCam: false,
       callbacks: {
@@ -110,23 +113,30 @@ class InstantVoiceClient {
         onBotTranscript: (data) => this.log(`Bot: ${data.text}`),
         onMessageError: (error) => console.error('Message error:', error),
         onError: (error) => console.error('Error:', error),
-        onAudioBufferingStarted: () => {
-          SoundUtils.beep();
-          this.updateBufferingStatus('Yes');
-          this.log(
-            `onMicCaptureStarted, timeTaken: ${Date.now() - this.startTime}`
-          );
-        },
-        onAudioBufferingStopped: () => {
-          this.updateBufferingStatus('No');
-          this.log(
-            `onMicCaptureStopped, timeTaken: ${Date.now() - this.startTime}`
-          );
-        },
-      } as DailyEventCallbacks,
+      },
     };
 
-    this.pcClient = new PipecatClient(PipecatConfig);
+    this.rtviClient = new RTVIClient(RTVIConfig);
+    this.rtviClient.registerHelper(
+      'transport',
+      new InstantVoiceHelper({
+        callbacks: {
+          onAudioBufferingStarted: () => {
+            SoundUtils.beep();
+            this.updateBufferingStatus('Yes');
+            this.log(
+              `onMicCaptureStarted, timeTaken: ${Date.now() - this.startTime}`
+            );
+          },
+          onAudioBufferingStopped: () => {
+            this.updateBufferingStatus('No');
+            this.log(
+              `onMicCaptureStopped, timeTaken: ${Date.now() - this.startTime}`
+            );
+          },
+        },
+      })
+    );
     this.setupTrackListeners();
   }
 
@@ -172,8 +182,8 @@ class InstantVoiceClient {
    * This is called when the bot is ready or when the transport state changes to ready
    */
   setupMediaTracks() {
-    if (!this.pcClient) return;
-    const tracks = this.pcClient.tracks();
+    if (!this.rtviClient) return;
+    const tracks = this.rtviClient.tracks();
     if (tracks.bot?.audio) {
       this.setupAudioTrack(tracks.bot.audio);
     }
@@ -184,10 +194,10 @@ class InstantVoiceClient {
    * This handles new tracks being added during the session
    */
   setupTrackListeners() {
-    if (!this.pcClient) return;
+    if (!this.rtviClient) return;
 
     // Listen for new tracks starting
-    this.pcClient.on(RTVIEvent.TrackStarted, (track, participant) => {
+    this.rtviClient.on(RTVIEvent.TrackStarted, (track, participant) => {
       // Only handle non-local (bot) tracks
       if (!participant?.local && track.kind === 'audio') {
         this.setupAudioTrack(track);
@@ -195,7 +205,7 @@ class InstantVoiceClient {
     });
 
     // Listen for tracks stopping
-    this.pcClient.on(RTVIEvent.TrackStopped, (track, participant) => {
+    this.rtviClient.on(RTVIEvent.TrackStopped, (track, participant) => {
       this.log(
         `Track stopped: ${track.kind} from ${participant?.name || 'unknown'}`
       );
@@ -220,25 +230,22 @@ class InstantVoiceClient {
 
   /**
    * Initialize and connect to the bot
-   * This sets up the Pipecat client, initializes devices, and establishes the connection
+   * This sets up the RTVI client, initializes devices, and establishes the connection
    */
   public async connect(): Promise<void> {
     try {
       this.startTime = Date.now();
       this.log('Connecting to bot...');
-      await this.pcClient.connect({
-        // The baseURL and endpoint of your bot server that the client will connect to
-        endpoint: 'http://localhost:7860/connect',
-      });
+      await this.rtviClient.connect();
     } catch (error) {
       this.log(`Error connecting: ${(error as Error).message}`);
       this.updateStatus('Error');
       this.updateBufferingStatus('No');
 
       // Clean up if there's an error
-      if (this.pcClient) {
+      if (this.rtviClient) {
         try {
-          await this.pcClient.disconnect();
+          await this.rtviClient.disconnect();
         } catch (disconnectError) {
           this.log(`Error during disconnect: ${disconnectError}`);
         }
@@ -251,7 +258,7 @@ class InstantVoiceClient {
    */
   public async disconnect(): Promise<void> {
     try {
-      await this.pcClient.disconnect();
+      await this.rtviClient.disconnect();
       if (
         this.botAudio.srcObject &&
         'getAudioTracks' in this.botAudio.srcObject

examples/instant-voice/client/javascript/src/util/instantVoiceHelper.ts

@@ -0,0 +1,39 @@
+import {RTVIClientHelper, RTVIClientHelperOptions, RTVIMessage} from "@pipecat-ai/client-js";
+import {DailyRTVIMessageType} from '@pipecat-ai/daily-transport';
+
+export type InstantVoiceHelperCallbacks = Partial<{
+    onAudioBufferingStarted: () => void;
+    onAudioBufferingStopped: () => void;
+}>;
+
+// --- Interface and class
+export interface InstantVoiceHelperOptions extends RTVIClientHelperOptions {
+    callbacks?: InstantVoiceHelperCallbacks;
+}
+export class InstantVoiceHelper extends RTVIClientHelper {
+
+    protected declare _options: InstantVoiceHelperOptions;
+
+    constructor(options: InstantVoiceHelperOptions) {
+        super(options);
+    }
+
+    handleMessage(rtviMessage: RTVIMessage): void {
+        switch (rtviMessage.type) {
+            case DailyRTVIMessageType.AUDIO_BUFFERING_STARTED:
+                if (this._options.callbacks?.onAudioBufferingStarted) {
+                    this._options.callbacks?.onAudioBufferingStarted()
+                }
+                break;
+            case DailyRTVIMessageType.AUDIO_BUFFERING_STOPPED:
+                if (this._options.callbacks?.onAudioBufferingStopped) {
+                    this._options.callbacks?.onAudioBufferingStopped()
+                }
+                break;
+        }
+    }
+
+    getMessageTypes(): string[] {
+        return [DailyRTVIMessageType.AUDIO_BUFFERING_STARTED, DailyRTVIMessageType.AUDIO_BUFFERING_STOPPED];
+    }
+}

examples/news-chatbot/client/javascript/package.json

@@ -15,7 +15,7 @@
     "vite": "^6.3.5"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/daily-transport": "^1.0.0"
+    "@pipecat-ai/client-js": "^0.3.5",
+    "@pipecat-ai/daily-transport": "^0.3.8"
   }
 }

examples/news-chatbot/client/javascript/src/app.js

@@ -5,7 +5,7 @@
  */
 
 /**
- * Pipecat Client Implementation
+ * RTVI Client Implementation
  *
  * This client connects to an RTVI-compatible bot server using WebRTC (via Daily).
  * It handles audio/video streaming and manages the connection lifecycle.
@@ -16,9 +16,78 @@
  * - Browser with WebRTC support
  */
 
-import { LogLevel, PipecatClient, RTVIEvent } from '@pipecat-ai/client-js';
+import {
+  LogLevel,
+  RTVIClient,
+  RTVIClientHelper,
+  RTVIEvent,
+} from '@pipecat-ai/client-js';
 import { DailyTransport } from '@pipecat-ai/daily-transport';
 
+class SearchResponseHelper extends RTVIClientHelper {
+  constructor(contentPanel) {
+    super();
+    this.contentPanel = contentPanel;
+  }
+
+  handleMessage(rtviMessage) {
+    console.log('SearchResponseHelper, received message:', rtviMessage);
+    if (rtviMessage.data) {
+      // Clear existing content
+      this.contentPanel.innerHTML = '';
+
+      // Create a container for all content
+      const contentContainer = document.createElement('div');
+      contentContainer.className = 'content-container';
+
+      // Add the search_result
+      if (rtviMessage.data.search_result) {
+        const searchResultDiv = document.createElement('div');
+        searchResultDiv.className = 'search-result';
+        searchResultDiv.textContent = rtviMessage.data.search_result;
+        contentContainer.appendChild(searchResultDiv);
+      }
+
+      // Add the sources
+      if (rtviMessage.data.origins) {
+        const sourcesDiv = document.createElement('div');
+        sourcesDiv.className = 'sources';
+
+        const sourcesTitle = document.createElement('h3');
+        sourcesTitle.className = 'sources-title';
+        sourcesTitle.textContent = 'Sources:';
+        sourcesDiv.appendChild(sourcesTitle);
+
+        rtviMessage.data.origins.forEach((origin) => {
+          const sourceLink = document.createElement('a');
+          sourceLink.className = 'source-link';
+          sourceLink.href = origin.site_uri;
+          sourceLink.target = '_blank';
+          sourceLink.textContent = origin.site_title;
+          sourcesDiv.appendChild(sourceLink);
+        });
+
+        contentContainer.appendChild(sourcesDiv);
+      }
+
+      // Add the rendered_content in an iframe
+      if (rtviMessage.data.rendered_content) {
+        const iframe = document.createElement('iframe');
+        iframe.className = 'iframe-container';
+        iframe.srcdoc = rtviMessage.data.rendered_content;
+        contentContainer.appendChild(iframe);
+      }
+
+      // Append the content container to the content panel
+      this.contentPanel.appendChild(contentContainer);
+    }
+  }
+
+  getMessageTypes() {
+    return ['bot-llm-search-response'];
+  }
+}
+
 /**
  * ChatbotClient handles the connection and media management for a real-time
  * voice and video interaction with an AI bot.
@@ -26,7 +95,7 @@ import { DailyTransport } from '@pipecat-ai/daily-transport';
 class ChatbotClient {
   constructor() {
     // Initialize client state
-    this.pcClient = null;
+    this.rtviClient = null;
     this.setupDOMElements();
     this.setupEventListeners();
   }
@@ -91,10 +160,10 @@ class ChatbotClient {
    * This is called when the bot is ready or when the transport state changes to ready
    */
   setupMediaTracks() {
-    if (!this.pcClient) return;
+    if (!this.rtviClient) return;
 
     // Get current tracks from the client
-    const tracks = this.pcClient.tracks();
+    const tracks = this.rtviClient.tracks();
 
     // Set up any available bot tracks
     if (tracks.bot?.audio) {
@@ -107,10 +176,10 @@ class ChatbotClient {
    * This handles new tracks being added during the session
    */
   setupTrackListeners() {
-    if (!this.pcClient) return;
+    if (!this.rtviClient) return;
 
     // Listen for new tracks starting
-    this.pcClient.on(RTVIEvent.TrackStarted, (track, participant) => {
+    this.rtviClient.on(RTVIEvent.TrackStarted, (track, participant) => {
       // Only handle non-local (bot) tracks
       if (!participant?.local && track.kind === 'audio') {
         this.setupAudioTrack(track);
@@ -118,7 +187,7 @@ class ChatbotClient {
     });
 
     // Listen for tracks stopping
-    this.pcClient.on(RTVIEvent.TrackStopped, (track, participant) => {
+    this.rtviClient.on(RTVIEvent.TrackStopped, (track, participant) => {
       this.log(
         `Track stopped event: ${track.kind} from ${
           participant?.name || 'unknown'
@@ -144,13 +213,20 @@ class ChatbotClient {
 
   /**
    * Initialize and connect to the bot
-   * This sets up the Pipecat client, initializes devices, and establishes the connection
+   * This sets up the RTVI client, initializes devices, and establishes the connection
    */
   async connect() {
     try {
-      // Initialize the Pipecat client with a Daily WebRTC transport and our configuration
-      this.pcClient = new PipecatClient({
+      // Initialize the RTVI client with a Daily WebRTC transport and our configuration
+      this.rtviClient = new RTVIClient({
         transport: new DailyTransport(),
+        params: {
+          // The baseURL and endpoint of your bot server that the client will connect to
+          baseUrl: 'http://localhost:7860',
+          endpoints: {
+            connect: '/connect',
+          },
+        },
         enableMic: true, // Enable microphone for user input
         enableCam: false,
         callbacks: {
@@ -175,8 +251,6 @@ class ChatbotClient {
               this.setupMediaTracks();
             }
           },
-          // Handle search response events
-          onBotLlmSearchResponse: this.handleSearchResponse.bind(this),
           // Handle bot connection events
           onBotConnected: (participant) => {
             this.log(`Bot connected: ${JSON.stringify(participant)}`);
@@ -207,22 +281,22 @@ class ChatbotClient {
           },
         },
       });
-
-      //this.pcClient.setLogLevel(LogLevel.DEBUG)
+      //this.rtviClient.setLogLevel(LogLevel.DEBUG)
+      this.rtviClient.registerHelper(
+        'llm',
+        new SearchResponseHelper(this.searchResultContainer)
+      );
 
       // Set up listeners for media track events
       this.setupTrackListeners();
 
       // Initialize audio devices
       this.log('Initializing devices...');
-      await this.pcClient.initDevices();
+      await this.rtviClient.initDevices();
 
       // Connect to the bot
       this.log('Connecting to bot...');
-      await this.pcClient.connect({
-        // The baseURL and endpoint of your bot server that the client will connect to
-        endpoint: 'http://localhost:7860/connect',
-      });
+      await this.rtviClient.connect();
 
       this.log('Connection complete');
     } catch (error) {
@@ -232,9 +306,9 @@ class ChatbotClient {
       this.updateStatus('Error');
 
       // Clean up if there's an error
-      if (this.pcClient) {
+      if (this.rtviClient) {
         try {
-          await this.pcClient.disconnect();
+          await this.rtviClient.disconnect();
         } catch (disconnectError) {
           this.log(`Error during disconnect: ${disconnectError.message}`);
         }
@@ -246,11 +320,11 @@ class ChatbotClient {
    * Disconnect from the bot and clean up media resources
    */
   async disconnect() {
-    if (this.pcClient) {
+    if (this.rtviClient) {
       try {
-        // Disconnect the Pipecat client
-        await this.pcClient.disconnect();
-        this.pcClient = null;
+        // Disconnect the RTVI client
+        await this.rtviClient.disconnect();
+        this.rtviClient = null;
 
         // Clean up audio
         if (this.botAudio.srcObject) {
@@ -265,57 +339,6 @@ class ChatbotClient {
       }
     }
   }
-
-  handleSearchResponse(response) {
-    console.log('SearchResponseHelper, received message:', response);
-    // Clear existing content
-    this.searchResultContainer.innerHTML = '';
-
-    // Create a container for all content
-    const contentContainer = document.createElement('div');
-    contentContainer.className = 'content-container';
-
-    // Add the search_result
-    if (response.search_result) {
-      const searchResultDiv = document.createElement('div');
-      searchResultDiv.className = 'search-result';
-      searchResultDiv.textContent = response.search_result;
-      contentContainer.appendChild(searchResultDiv);
-    }
-
-    // Add the sources
-    if (response.origins) {
-      const sourcesDiv = document.createElement('div');
-      sourcesDiv.className = 'sources';
-
-      const sourcesTitle = document.createElement('h3');
-      sourcesTitle.className = 'sources-title';
-      sourcesTitle.textContent = 'Sources:';
-      sourcesDiv.appendChild(sourcesTitle);
-
-      response.origins.forEach((origin) => {
-        const sourceLink = document.createElement('a');
-        sourceLink.className = 'source-link';
-        sourceLink.href = origin.site_uri;
-        sourceLink.target = '_blank';
-        sourceLink.textContent = origin.site_title;
-        sourcesDiv.appendChild(sourceLink);
-      });
-
-      contentContainer.appendChild(sourcesDiv);
-    }
-
-    // Add the rendered_content in an iframe
-    if (response.rendered_content) {
-      const iframe = document.createElement('iframe');
-      iframe.className = 'iframe-container';
-      iframe.srcdoc = response.rendered_content;
-      contentContainer.appendChild(iframe);
-    }
-
-    // Append the content container to the content panel
-    this.searchResultContainer.appendChild(contentContainer);
-  }
 }
 
 // Initialize the client when the page loads

examples/p2p-webrtc/video-transform/client/typescript/package.json

@@ -18,7 +18,7 @@
     "vite": "^6.3.5"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/small-webrtc-transport": "^1.0.0"
+    "@pipecat-ai/client-js": "^0.3.2",
+    "@pipecat-ai/small-webrtc-transport": "^0.0.2"
   }
 }

examples/p2p-webrtc/video-transform/client/typescript/src/app.ts

@@ -1,236 +1,217 @@
-import { SmallWebRTCTransport } from '@pipecat-ai/small-webrtc-transport';
 import {
-  BotLLMTextData,
-  Participant,
-  PipecatClient,
-  PipecatClientOptions,
-  TranscriptData,
-  TransportState,
-} from '@pipecat-ai/client-js';
+    SmallWebRTCTransport
+} from "@pipecat-ai/small-webrtc-transport";
+import {Participant, RTVIClient, RTVIClientOptions, Transport} from "@pipecat-ai/client-js";
 
 class WebRTCApp {
-  private declare connectBtn: HTMLButtonElement;
-  private declare disconnectBtn: HTMLButtonElement;
-  private declare muteBtn: HTMLButtonElement;
 
-  private declare audioInput: HTMLSelectElement;
-  private declare videoInput: HTMLSelectElement;
-  private declare audioCodec: HTMLSelectElement;
-  private declare videoCodec: HTMLSelectElement;
+    private declare connectBtn: HTMLButtonElement;
+    private declare disconnectBtn: HTMLButtonElement;
+    private declare muteBtn: HTMLButtonElement;
 
-  private declare videoElement: HTMLVideoElement;
-  private declare audioElement: HTMLAudioElement;
+    private declare audioInput: HTMLSelectElement;
+    private declare videoInput: HTMLSelectElement;
+    private declare audioCodec: HTMLSelectElement;
+    private declare videoCodec: HTMLSelectElement;
 
-  private debugLog: HTMLElement | null = null;
-  private statusSpan: HTMLElement | null = null;
+    private declare videoElement: HTMLVideoElement;
+    private declare audioElement: HTMLAudioElement;
 
-  private declare smallWebRTCTransport: SmallWebRTCTransport;
-  private declare pcClient: PipecatClient;
+    private debugLog: HTMLElement | null = null;
+    private statusSpan: HTMLElement | null = null;
 
-  constructor() {
-    this.setupDOMElements();
-    this.setupDOMEventListeners();
-    this.initializePipecatClient();
-    void this.populateDevices();
-  }
+    private declare smallWebRTCTransport: SmallWebRTCTransport;
+    private declare rtviClient: RTVIClient;
 
-  private initializePipecatClient(): void {
-    const opts: PipecatClientOptions = {
-      transport: new SmallWebRTCTransport({ connectionUrl: '/api/offer' }),
-      enableMic: true,
-      enableCam: true,
-      callbacks: {
-        onTransportStateChanged: (state: TransportState) => {
-          this.log(`Transport state: ${state}`);
-        },
-        onConnected: () => {
-          this.onConnectedHandler();
-        },
-        onBotReady: () => {
-          this.log('Bot is ready.');
-        },
-        onDisconnected: () => {
-          this.onDisconnectedHandler();
-        },
-        onUserStartedSpeaking: () => {
-          this.log('User started speaking.');
-        },
-        onUserStoppedSpeaking: () => {
-          this.log('User stopped speaking.');
-        },
-        onBotStartedSpeaking: () => {
-          this.log('Bot started speaking.');
-        },
-        onBotStoppedSpeaking: () => {
-          this.log('Bot stopped speaking.');
-        },
-        onUserTranscript: (transcript: TranscriptData) => {
-          if (transcript.final) {
-            this.log(`User transcript: ${transcript.text}`);
-          }
-        },
-        onBotTranscript: (data: BotLLMTextData) => {
-          this.log(`Bot transcript: ${data.text}`);
-        },
-        onTrackStarted: (
-          track: MediaStreamTrack,
-          participant?: Participant
-        ) => {
-          if (participant?.local) {
-            return;
-          }
-          this.onBotTrackStarted(track);
-        },
-        onServerMessage: (msg: unknown) => {
-          this.log(`Server message: ${msg}`);
-        },
-      },
-    };
-    this.pcClient = new PipecatClient(opts);
-    this.smallWebRTCTransport = this.pcClient.transport as SmallWebRTCTransport;
-  }
-
-  private setupDOMElements(): void {
-    this.connectBtn = document.getElementById(
-      'connect-btn'
-    ) as HTMLButtonElement;
-    this.disconnectBtn = document.getElementById(
-      'disconnect-btn'
-    ) as HTMLButtonElement;
-    this.muteBtn = document.getElementById('mute-btn') as HTMLButtonElement;
-
-    this.audioInput = document.getElementById(
-      'audio-input'
-    ) as HTMLSelectElement;
-    this.videoInput = document.getElementById(
-      'video-input'
-    ) as HTMLSelectElement;
-    this.audioCodec = document.getElementById(
-      'audio-codec'
-    ) as HTMLSelectElement;
-    this.videoCodec = document.getElementById(
-      'video-codec'
-    ) as HTMLSelectElement;
-
-    this.videoElement = document.getElementById(
-      'bot-video'
-    ) as HTMLVideoElement;
-    this.audioElement = document.getElementById(
-      'bot-audio'
-    ) as HTMLAudioElement;
-
-    this.debugLog = document.getElementById('debug-log');
-    this.statusSpan = document.getElementById('connection-status');
-  }
-
-  private setupDOMEventListeners(): void {
-    this.connectBtn.addEventListener('click', () => this.start());
-    this.disconnectBtn.addEventListener('click', () => this.stop());
-    this.audioInput.addEventListener('change', (e) => {
-      // @ts-ignore
-      let audioDevice = e.target?.value;
-      this.pcClient.updateMic(audioDevice);
-    });
-    this.videoInput.addEventListener('change', (e) => {
-      // @ts-ignore
-      let videoDevice = e.target?.value;
-      this.pcClient.updateCam(videoDevice);
-    });
-    this.muteBtn.addEventListener('click', () => {
-      let isCamEnabled = this.pcClient.isCamEnabled;
-      this.pcClient.enableCam(!isCamEnabled);
-      this.muteBtn.textContent = isCamEnabled ? '📵' : '📷';
-    });
-  }
-
-  private log(message: string): void {
-    if (!this.debugLog) return;
-    const entry = document.createElement('div');
-    entry.textContent = `${new Date().toISOString()} - ${message}`;
-    if (message.startsWith('User: ')) {
-      entry.style.color = '#2196F3';
-    } else if (message.startsWith('Bot: ')) {
-      entry.style.color = '#4CAF50';
+    constructor() {
+        this.setupDOMElements();
+        this.setupDOMEventListeners();
+        this.initializeRTVIClient()
+        void this.populateDevices();
     }
-    this.debugLog.appendChild(entry);
-    this.debugLog.scrollTop = this.debugLog.scrollHeight;
-  }
 
-  private clearAllLogs() {
-    this.debugLog!.innerText = '';
-  }
-
-  private updateStatus(status: string): void {
-    if (this.statusSpan) {
-      this.statusSpan.textContent = status;
+    private initializeRTVIClient(): void {
+        const transport = new SmallWebRTCTransport();
+        const RTVIConfig: RTVIClientOptions = {
+            params: {
+                baseUrl: "/api/offer"
+            },
+            transport: transport as Transport,
+            enableMic: true,
+            enableCam: true,
+            callbacks: {
+                onTransportStateChanged: (state) => {
+                    this.log(`Transport state: ${state}`)
+                },
+                onConnected: () => {
+                    this.onConnectedHandler()
+                },
+                onBotReady: () => {
+                    this.log("Bot is ready.")
+                },
+                onDisconnected: () => {
+                    this.onDisconnectedHandler()
+                },
+                onUserStartedSpeaking: () => {
+                    this.log("User started speaking.")
+                },
+                onUserStoppedSpeaking: () => {
+                    this.log("User stopped speaking.")
+                },
+                onBotStartedSpeaking: () => {
+                    this.log("Bot started speaking.")
+                },
+                onBotStoppedSpeaking: () => {
+                    this.log("Bot stopped speaking.")
+                },
+                onUserTranscript: (transcript) => {
+                    if (transcript.final) {
+                        this.log(`User transcript: ${transcript.text}`)
+                    }
+                },
+                onBotTranscript: (transcript) => {
+                    this.log(`Bot transcript: ${transcript.text}`)
+                },
+                onTrackStarted: (track: MediaStreamTrack, participant?: Participant) => {
+                    if (participant?.local) {
+                        return
+                    }
+                    this.onBotTrackStarted(track)
+                },
+                onServerMessage: (msg) => {
+                    this.log(`Server message: ${msg}`)
+                }
+            },
+        }
+        RTVIConfig.customConnectHandler = () => Promise.resolve();
+        this.rtviClient = new RTVIClient(RTVIConfig);
+        this.smallWebRTCTransport = transport
     }
-    this.log(`Status: ${status}`);
-  }
 
-  private onConnectedHandler() {
-    this.updateStatus('Connected');
-    if (this.connectBtn) this.connectBtn.disabled = true;
-    if (this.disconnectBtn) this.disconnectBtn.disabled = false;
-  }
+    private setupDOMElements(): void {
+        this.connectBtn = document.getElementById('connect-btn') as HTMLButtonElement;
+        this.disconnectBtn = document.getElementById('disconnect-btn') as HTMLButtonElement;
+        this.muteBtn = document.getElementById('mute-btn') as HTMLButtonElement;
 
-  private onDisconnectedHandler() {
-    this.updateStatus('Disconnected');
-    if (this.connectBtn) this.connectBtn.disabled = false;
-    if (this.disconnectBtn) this.disconnectBtn.disabled = true;
-  }
+        this.audioInput = document.getElementById('audio-input') as HTMLSelectElement;
+        this.videoInput = document.getElementById('video-input') as HTMLSelectElement;
+        this.audioCodec = document.getElementById('audio-codec') as HTMLSelectElement;
+        this.videoCodec = document.getElementById('video-codec') as HTMLSelectElement;
 
-  private onBotTrackStarted(track: MediaStreamTrack) {
-    if (track.kind === 'video') {
-      this.videoElement.srcObject = new MediaStream([track]);
-    } else {
-      this.audioElement.srcObject = new MediaStream([track]);
+        this.videoElement = document.getElementById('bot-video') as HTMLVideoElement;
+        this.audioElement = document.getElementById('bot-audio') as HTMLAudioElement;
+
+        this.debugLog = document.getElementById('debug-log');
+        this.statusSpan = document.getElementById('connection-status');
     }
-  }
 
-  private async populateDevices(): Promise<void> {
-    const populateSelect = (
-      select: HTMLSelectElement,
-      devices: MediaDeviceInfo[]
-    ): void => {
-      let counter = 1;
-      devices.forEach((device) => {
-        const option = document.createElement('option');
-        option.value = device.deviceId;
-        option.text = device.label || 'Device #' + counter;
-        select.appendChild(option);
-        counter += 1;
-      });
-    };
+    private setupDOMEventListeners(): void {
+        this.connectBtn.addEventListener("click", () => this.start());
+        this.disconnectBtn.addEventListener("click", () => this.stop());
+        this.audioInput.addEventListener("change", (e) => {
+            // @ts-ignore
+            let audioDevice = e.target?.value
+            this.rtviClient.updateMic(audioDevice)
+        })
+        this.videoInput.addEventListener("change", (e) => {
+            // @ts-ignore
+            let videoDevice = e.target?.value
+            this.rtviClient.updateCam(videoDevice)
+        })
+        this.muteBtn.addEventListener('click', () => {
+            let isCamEnabled = this.rtviClient.isCamEnabled
+            this.rtviClient.enableCam(!isCamEnabled)
+            this.muteBtn.textContent = isCamEnabled ? '📵' : '📷';
+        });
 
-    try {
-      const audioDevices = await this.pcClient.getAllMics();
-      populateSelect(this.audioInput, audioDevices);
-      const videoDevices = await this.pcClient.getAllCams();
-      populateSelect(this.videoInput, videoDevices);
-    } catch (e) {
-      alert(e);
     }
-  }
 
-  private async start(): Promise<void> {
-    this.clearAllLogs();
-
-    this.connectBtn.disabled = true;
-    this.updateStatus('Connecting');
-
-    this.smallWebRTCTransport.setAudioCodec(this.audioCodec.value);
-    this.smallWebRTCTransport.setVideoCodec(this.videoCodec.value);
-    try {
-      await this.pcClient.connect();
-    } catch (e) {
-      console.log(`Failed to connect ${e}`);
-      this.stop();
+    private log(message: string): void {
+        if (!this.debugLog) return;
+        const entry = document.createElement('div');
+        entry.textContent = `${new Date().toISOString()} - ${message}`;
+        if (message.startsWith('User: ')) {
+          entry.style.color = '#2196F3';
+        } else if (message.startsWith('Bot: ')) {
+          entry.style.color = '#4CAF50';
+        }
+        this.debugLog.appendChild(entry);
+        this.debugLog.scrollTop = this.debugLog.scrollHeight;
     }
-  }
 
-  private stop(): void {
-    void this.pcClient.disconnect();
-  }
+    private clearAllLogs() {
+        this.debugLog!.innerText = ''
+    }
+
+    private updateStatus(status: string): void {
+        if (this.statusSpan) {
+            this.statusSpan.textContent = status;
+        }
+        this.log(`Status: ${status}`);
+   }
+
+    private onConnectedHandler() {
+        this.updateStatus('Connected');
+        if (this.connectBtn) this.connectBtn.disabled = true;
+        if (this.disconnectBtn) this.disconnectBtn.disabled = false;
+    }
+
+    private onDisconnectedHandler() {
+        this.updateStatus('Disconnected');
+        if (this.connectBtn) this.connectBtn.disabled = false;
+        if (this.disconnectBtn) this.disconnectBtn.disabled = true;
+    }
+
+    private onBotTrackStarted(track: MediaStreamTrack) {
+        if (track.kind === 'video') {
+            this.videoElement.srcObject = new MediaStream([track]);
+        } else {
+            this.audioElement.srcObject = new MediaStream([track]);
+        }
+    }
+
+    private async populateDevices(): Promise<void> {
+        const populateSelect = (select: HTMLSelectElement, devices: MediaDeviceInfo[]): void => {
+            let counter = 1;
+            devices.forEach((device) => {
+                const option = document.createElement('option');
+                option.value = device.deviceId;
+                option.text = device.label || ('Device #' + counter);
+                select.appendChild(option);
+                counter += 1;
+            });
+        };
+
+        try {
+            const audioDevices = await this.rtviClient.getAllMics();
+            populateSelect(this.audioInput, audioDevices);
+            const videoDevices = await this.rtviClient.getAllCams();
+            populateSelect(this.videoInput, videoDevices);
+        } catch (e) {
+            alert(e);
+        }
+    }
+
+    private async start(): Promise<void> {
+        this.clearAllLogs()
+
+        this.connectBtn.disabled = true;
+        this.updateStatus("Connecting")
+
+        this.smallWebRTCTransport.setAudioCodec(this.audioCodec.value)
+        this.smallWebRTCTransport.setVideoCodec(this.videoCodec.value)
+        try {
+            await this.rtviClient.connect()
+        } catch (e) {
+            console.log(`Failed to connect ${e}`)
+            this.stop()
+        }
+
+    }
+
+    private stop(): void {
+        void this.rtviClient.disconnect()
+    }
 }
 
 // Create the WebRTCConnection instance

examples/simple-chatbot/client/javascript/index.html

@@ -1,51 +1,40 @@
 <!DOCTYPE html>
 <html lang="en">
-  <head>
-    <meta charset="UTF-8" />
-    <meta name="viewport" content="width=device-width, initial-scale=1.0" />
-    <title>AI Chatbot</title>
-  </head>
 
-  <body>
-    <div class="container">
-      <div class="status-bar">
-        <div class="status">
-          Status: <span id="connection-status">Disconnected</span>
-        </div>
-        <div class="controls">
-          <button id="connect-btn">Connect</button>
-          <button id="disconnect-btn" disabled>Disconnect</button>
-        </div>
+<head>
+  <meta charset="UTF-8">
+  <meta name="viewport" content="width=device-width, initial-scale=1.0">
+  <title>AI Chatbot</title>
+</head>
+
+<body>
+  <div class="container">
+    <div class="status-bar">
+      <div class="status">
+        Status: <span id="connection-status">Disconnected</span>
       </div>
-
-      <div class="main-content">
-        <div class="bot-container">
-          <div id="bot-video-container"></div>
-          <audio id="bot-audio" autoplay></audio>
-        </div>
-      </div>
-
-      <div class="device-bar">
-        <div class="device-controls">
-          <select id="device-selector"></select>
-          <button id="mic-toggle-btn">Unmute Mic</button>
-        </div>
-        <div class="text-input-container">
-          <input
-            type="text"
-            id="text-input"
-            placeholder="Type your message..." />
-          <button id="send-text-btn" disabled>Send</button>
-        </div>
-      </div>
-
-      <div class="debug-panel">
-        <h3>Debug Info</h3>
-        <div id="debug-log"></div>
+      <div class="controls">
+        <button id="connect-btn">Connect</button>
+        <button id="disconnect-btn" disabled>Disconnect</button>
       </div>
     </div>
 
-    <script type="module" src="/src/app.js"></script>
-    <link rel="stylesheet" href="/src/style.css" />
-  </body>
-</html>
+    <div class="main-content">
+      <div class="bot-container">
+        <div id="bot-video-container">
+        </div>
+        <audio id="bot-audio" autoplay></audio>
+      </div>
+    </div>
+
+    <div class="debug-panel">
+      <h3>Debug Info</h3>
+      <div id="debug-log"></div>
+    </div>
+  </div>
+
+  <script type="module" src="/src/app.js"></script>
+  <link rel="stylesheet" href="/src/style.css">
+</body>
+
+</html>

examples/simple-chatbot/client/javascript/package.json

@@ -15,7 +15,7 @@
     "vite": "^6.3.5"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/daily-transport": "^1.0.0"
+    "@pipecat-ai/client-js": "^0.3.5",
+    "@pipecat-ai/daily-transport": "^0.3.8"
   }
 }

examples/simple-chatbot/client/javascript/src/app.js

@@ -5,7 +5,7 @@
  */
 
 /**
- * Pipecat Client Implementation
+ * RTVI Client Implementation
  *
  * This client connects to an RTVI-compatible bot server using WebRTC (via Daily).
  * It handles audio/video streaming and manages the connection lifecycle.
@@ -16,7 +16,7 @@
  * - Browser with WebRTC support
  */
 
-import { PipecatClient, RTVIEvent } from '@pipecat-ai/client-js';
+import { RTVIClient, RTVIEvent } from '@pipecat-ai/client-js';
 import { DailyTransport } from '@pipecat-ai/daily-transport';
 
 /**
@@ -26,8 +26,9 @@ import { DailyTransport } from '@pipecat-ai/daily-transport';
 class ChatbotClient {
   constructor() {
     // Initialize client state
-    this.pcClient = null;
+    this.rtviClient = null;
     this.setupDOMElements();
+    this.setupEventListeners();
     this.initializeClientAndTransport();
   }
 
@@ -41,9 +42,6 @@ class ChatbotClient {
     this.statusSpan = document.getElementById('connection-status');
     this.debugLog = document.getElementById('debug-log');
     this.botVideoContainer = document.getElementById('bot-video-container');
-    this.deviceSelector = document.getElementById('device-selector');
-    this.micToggleBtn = document.getElementById('mic-toggle-btn');
-    this.sendTextBtn = document.getElementById('send-text-btn');
 
     // Create an audio element for bot's voice output
     this.botAudio = document.createElement('audio');
@@ -56,78 +54,25 @@ class ChatbotClient {
    * Set up event listeners for connect/disconnect buttons
    */
   setupEventListeners() {
-    this.connectBtn.addEventListener('click', () => {
-      console.log('click');
-      this.connect();
-    });
+    this.connectBtn.addEventListener('click', () => this.connect());
     this.disconnectBtn.addEventListener('click', () => this.disconnect());
-
-    // Populate device selector
-    this.pcClient.getAllMics().then((mics) => {
-      console.log('Available mics:', mics);
-      mics.forEach((device) => {
-        const option = document.createElement('option');
-        option.value = device.deviceId;
-        option.textContent = device.label || `Microphone ${device.deviceId}`;
-        this.deviceSelector.appendChild(option);
-      });
-    });
-    this.deviceSelector.addEventListener('change', (event) => {
-      const selectedDeviceId = event.target.value;
-      console.log('Selected device ID:', selectedDeviceId);
-      this.pcClient.updateMic(selectedDeviceId);
-    });
-
-    // Handle mic mute/unmute toggle
-    const micToggleBtn = document.getElementById('mic-toggle-btn');
-
-    micToggleBtn.addEventListener('click', async () => {
-      if (this.pcClient.state === 'disconnected') {
-        await this.pcClient.initDevices();
-      } else {
-        this.pcClient.enableMic(!this.pcClient.isMicEnabled);
-      }
-    });
-
-    const textInput = document.getElementById('text-input');
-
-    const sendTextToLLM = () => {
-      this.sendTextBtn.disabled = true; // Disable button to prevent multiple clicks
-      const text = textInput.value.trim();
-      if (text) {
-        void this.pcClient.appendToContext({
-          role: 'user',
-          content: text,
-          run_immediately: true,
-        });
-      }
-      textInput.value = ''; // Clear the input
-      this.sendTextBtn.disabled = false; // Re-enable button after sending
-    };
-
-    this.sendTextBtn.addEventListener('click', sendTextToLLM);
-
-    // Also handle Enter key in the input
-    textInput.addEventListener('keypress', (e) => {
-      if (e.key === 'Enter') {
-        sendTextToLLM();
-      }
-    });
   }
 
-  updateMicToggleButton(micEnabled) {
-    console.log('Mic enabled:', micEnabled, this.pcClient?.isMicEnabled);
-    this.micToggleBtn.textContent = micEnabled ? 'Mute Mic' : 'Unmute Mic';
-  }
   /**
-   * Set up the Pipecat client and Daily transport
+   * Set up the RTVI client and Daily transport
    */
-  async initializeClientAndTransport() {
-    console.log('Initializing Pipecat client and transport...');
-    // Initialize the Pipecat client with a DailyTransport and our configuration
-    this.pcClient = new PipecatClient({
+  initializeClientAndTransport() {
+    // Initialize the RTVI client with a DailyTransport and our configuration
+    this.rtviClient = new RTVIClient({
       transport: new DailyTransport(),
-      enableMic: true,
+      params: {
+        // The baseURL and endpoint of your bot server that the client will connect to
+        baseUrl: 'http://localhost:7860',
+        endpoints: {
+          connect: '/connect',
+        },
+      },
+      enableMic: true, // Enable microphone for user input
       enableCam: false,
       callbacks: {
         // Handle connection state changes
@@ -141,9 +86,7 @@ class ChatbotClient {
           this.updateStatus('Disconnected');
           this.connectBtn.disabled = false;
           this.disconnectBtn.disabled = true;
-          this.sendTextBtn.disabled = true;
           this.log('Client disconnected');
-          this.updateMicToggleButton(false);
         },
         // Handle transport state changes
         onTransportStateChanged: (state) => {
@@ -163,7 +106,6 @@ class ChatbotClient {
         onBotReady: (data) => {
           this.log(`Bot ready: ${JSON.stringify(data)}`);
           this.setupMediaTracks();
-          this.sendTextBtn.disabled = false;
         },
         // Transcript events
         onUserTranscript: (data) => {
@@ -179,20 +121,14 @@ class ChatbotClient {
         onMessageError: (error) => {
           console.log('Message error:', error);
         },
-        onMicUpdated: (data) => {
-          console.log('Mic updated:', data);
-          this.deviceSelector.value = data.deviceId;
-        },
         onError: (error) => {
           console.log('Error:', JSON.stringify(error));
         },
       },
     });
-    window.client = this; // Expose client globally for debugging
 
     // Set up listeners for media track events
     this.setupTrackListeners();
-    this.setupEventListeners();
   }
 
   /**
@@ -227,10 +163,10 @@ class ChatbotClient {
    * This is called when the bot is ready or when the transport state changes to ready
    */
   setupMediaTracks() {
-    if (!this.pcClient) return;
+    if (!this.rtviClient) return;
 
     // Get current tracks from the client
-    const tracks = this.pcClient.tracks();
+    const tracks = this.rtviClient.tracks();
 
     // Set up any available bot tracks
     if (tracks.bot?.audio) {
@@ -246,34 +182,27 @@ class ChatbotClient {
    * This handles new tracks being added during the session
    */
   setupTrackListeners() {
-    if (!this.pcClient) return;
+    if (!this.rtviClient) return;
 
     // Listen for new tracks starting
-    this.pcClient.on(RTVIEvent.TrackStarted, (track, participant) => {
+    this.rtviClient.on(RTVIEvent.TrackStarted, (track, participant) => {
+      // Only handle non-local (bot) tracks
       if (!participant?.local) {
         if (track.kind === 'audio') {
           this.setupAudioTrack(track);
         } else if (track.kind === 'video') {
           this.setupVideoTrack(track);
         }
-      } else if (track.kind === 'audio') {
-        console.log(`Local audio track started: `, this.pcClient.tracks());
-        // If local audio track starts, update mic
-        this.updateMicToggleButton(true);
       }
     });
 
     // Listen for tracks stopping
-    this.pcClient.on(RTVIEvent.TrackStopped, (track, participant) => {
+    this.rtviClient.on(RTVIEvent.TrackStopped, (track, participant) => {
       this.log(
         `Track stopped event: ${track.kind} from ${
-          participant ? (participant.local ? 'local' : 'bot') : 'unknown'
+          participant?.name || 'unknown'
         }`
       );
-      if (participant?.local && track.kind === 'audio') {
-        // If local audio track stops, update mic toggle button
-        this.updateMicToggleButton(false);
-      }
     });
   }
 
@@ -322,16 +251,17 @@ class ChatbotClient {
 
   /**
    * Initialize and connect to the bot
-   * This sets up the Pipecat client, initializes devices, and establishes the connection
+   * This sets up the RTVI client, initializes devices, and establishes the connection
    */
   async connect() {
     try {
+      // Initialize audio/video devices
+      this.log('Initializing devices...');
+      await this.rtviClient.initDevices();
+
       // Connect to the bot
       this.log('Connecting to bot...');
-      await this.pcClient.connect({
-        endpoint: 'http://localhost:7860/connect',
-        timeout: 25000,
-      });
+      await this.rtviClient.connect();
 
       this.log('Connection complete');
     } catch (error) {
@@ -341,9 +271,9 @@ class ChatbotClient {
       this.updateStatus('Error');
 
       // Clean up if there's an error
-      if (this.pcClient) {
+      if (this.rtviClient) {
         try {
-          await this.pcClient.disconnect();
+          await this.rtviClient.disconnect();
         } catch (disconnectError) {
           this.log(`Error during disconnect: ${disconnectError.message}`);
         }
@@ -355,10 +285,10 @@ class ChatbotClient {
    * Disconnect from the bot and clean up media resources
    */
   async disconnect() {
-    if (this.pcClient) {
+    if (this.rtviClient) {
       try {
-        // Disconnect the Pipecat client
-        await this.pcClient.disconnect();
+        // Disconnect the RTVI client
+        await this.rtviClient.disconnect();
 
         // Clean up audio
         if (this.botAudio.srcObject) {

examples/simple-chatbot/client/javascript/src/style.css

@@ -10,8 +10,7 @@ body {
   margin: 0 auto;
 }
 
-.status-bar,
-.device-bar {
+.status-bar {
   display: flex;
   justify-content: space-between;
   align-items: center;
@@ -21,24 +20,7 @@ body {
   margin-bottom: 20px;
 }
 
-.device-bar {
-  flex-direction: column;
-  gap: 10px;
-}
-
-.controls,
-.device-controls {
-  display: flex;
-  align-items: center;
-  gap: 10px; /* Adds spacing between elements */
-}
-
-.device-controls {
-  margin-left: auto;
-}
-
-.controls button,
-.device-controls button {
+.controls button {
   padding: 8px 16px;
   margin-left: 10px;
   border: none;
@@ -46,56 +28,6 @@ body {
   cursor: pointer;
 }
 
-#bot-selector,
-#device-selector {
-  padding: 8px 16px;
-  padding-right: 40px;
-  border: none;
-  border-radius: 4px;
-  background-color: #6c757d; /* Gray background */
-  color: white; /* White text */
-  cursor: pointer;
-  appearance: none; /* Removes default browser styling for dropdowns */
-  background-image: url("data:image/svg+xml,%3Csvg xmlns='http://www.w3.org/2000/svg' viewBox='0 0 24 24' fill='white'%3E%3Cpath d='M7 10l5 5 5-5z'/%3E%3C/svg%3E"); /* Custom arrow */
-  background-repeat: no-repeat;
-  background-position: right 8px center; /* Position the arrow */
-}
-
-#bot-selector:focus,
-#device-selector:focus {
-  outline: none;
-  box-shadow: 0 0 4px rgba(0, 0, 0, 0.3); /* Add a subtle focus effect */
-}
-
-.text-input-container {
-  display: flex;
-  gap: 8px;
-  margin-left: 10px;
-  width: 100%;
-  flex: 1;
-}
-
-#text-input {
-  flex: 1;
-  padding: 8px 16px;
-  border: 1px solid #e0e0e0;
-  border-radius: 4px;
-  min-width: 200px;
-}
-
-#send-text-btn {
-  padding: 8px 16px;
-  border: none;
-  border-radius: 4px;
-  background-color: #007bff;
-  color: white;
-  flex-shrink: 0;
-}
-
-#send-text-btn:hover {
-  background-color: #0056b3;
-}
-
 #connect-btn {
   background-color: #4caf50;
   color: white;
@@ -106,9 +38,6 @@ body {
   color: white;
 }
 
-#mic-toggle-btn {
-}
-
 button:disabled {
   opacity: 0.5;
   cursor: not-allowed;

examples/simple-chatbot/client/react/package.json

@@ -10,9 +10,9 @@
     "preview": "vite preview"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/client-react": "^1.0.0",
-    "@pipecat-ai/daily-transport": "^1.0.0",
+    "@pipecat-ai/client-js": "^0.3.5",
+    "@pipecat-ai/client-react": "^0.3.5",
+    "@pipecat-ai/daily-transport": "^0.3.8",
     "react": "^18.3.1",
     "react-dom": "^18.3.1"
   },

examples/simple-chatbot/client/react/src/App.tsx

@@ -1,22 +1,22 @@
 import {
-  PipecatClientAudio,
-  PipecatClientVideo,
-  usePipecatClientTransportState,
+  RTVIClientAudio,
+  RTVIClientVideo,
+  useRTVIClientTransportState,
 } from '@pipecat-ai/client-react';
-import { PipecatProvider } from './providers/PipecatProvider';
+import { RTVIProvider } from './providers/RTVIProvider';
 import { ConnectButton } from './components/ConnectButton';
 import { StatusDisplay } from './components/StatusDisplay';
 import { DebugDisplay } from './components/DebugDisplay';
 import './App.css';
 
 function BotVideo() {
-  const transportState = usePipecatClientTransportState();
+  const transportState = useRTVIClientTransportState();
   const isConnected = transportState !== 'disconnected';
 
   return (
     <div className="bot-container">
       <div className="video-container">
-        {isConnected && <PipecatClientVideo participant="bot" fit="cover" />}
+        {isConnected && <RTVIClientVideo participant="bot" fit="cover" />}
       </div>
     </div>
   );
@@ -35,16 +35,16 @@ function AppContent() {
       </div>
 
       <DebugDisplay />
-      <PipecatClientAudio />
+      <RTVIClientAudio />
     </div>
   );
 }
 
 function App() {
   return (
-    <PipecatProvider>
+    <RTVIProvider>
       <AppContent />
-    </PipecatProvider>
+    </RTVIProvider>
   );
 }
 

examples/simple-chatbot/client/react/src/components/ConnectButton.tsx

@@ -1,16 +1,16 @@
 import {
-  usePipecatClient,
-  usePipecatClientTransportState,
+  useRTVIClient,
+  useRTVIClientTransportState,
 } from '@pipecat-ai/client-react';
 
 export function ConnectButton() {
-  const client = usePipecatClient();
-  const transportState = usePipecatClientTransportState();
+  const client = useRTVIClient();
+  const transportState = useRTVIClientTransportState();
   const isConnected = ['connected', 'ready'].includes(transportState);
 
   const handleClick = async () => {
     if (!client) {
-      console.error('Pipecat client is not initialized');
+      console.error('RTVI client is not initialized');
       return;
     }
 
@@ -18,7 +18,7 @@ export function ConnectButton() {
       if (isConnected) {
         await client.disconnect();
       } else {
-        await client.connect({ endpoint: 'http://localhost:7860/connect' });
+        await client.connect();
       }
     } catch (error) {
       console.error('Connection error:', error);

examples/simple-chatbot/client/react/src/components/DebugDisplay.tsx

@@ -6,12 +6,12 @@ import {
   TranscriptData,
   BotLLMTextData,
 } from '@pipecat-ai/client-js';
-import { usePipecatClient, useRTVIClientEvent } from '@pipecat-ai/client-react';
+import { useRTVIClient, useRTVIClientEvent } from '@pipecat-ai/client-react';
 import './DebugDisplay.css';
 
 export function DebugDisplay() {
   const debugLogRef = useRef<HTMLDivElement>(null);
-  const client = usePipecatClient();
+  const client = useRTVIClient();
 
   const log = useCallback((message: string) => {
     if (!debugLogRef.current) return;

examples/simple-chatbot/client/react/src/components/StatusDisplay.tsx

@@ -1,7 +1,7 @@
-import { usePipecatClientTransportState } from '@pipecat-ai/client-react';
+import { useRTVIClientTransportState } from '@pipecat-ai/client-react';
 
 export function StatusDisplay() {
-  const transportState = usePipecatClientTransportState();
+  const transportState = useRTVIClientTransportState();
 
   return (
     <div className="status">

examples/simple-chatbot/client/react/src/providers/PipecatProvider.tsx

@@ -1,16 +0,0 @@
-import { type PropsWithChildren } from 'react';
-import { PipecatClient } from '@pipecat-ai/client-js';
-import { DailyTransport } from '@pipecat-ai/daily-transport';
-import { PipecatClientProvider } from '@pipecat-ai/client-react';
-
-const client = new PipecatClient({
-  transport: new DailyTransport(),
-  enableMic: true,
-  enableCam: false,
-});
-
-export function PipecatProvider({ children }: PropsWithChildren) {
-  return (
-    <PipecatClientProvider client={client}>{children}</PipecatClientProvider>
-  );
-}

examples/simple-chatbot/client/react/src/providers/RTVIProvider.tsx

@@ -0,0 +1,22 @@
+import { type PropsWithChildren } from 'react';
+import { RTVIClient } from '@pipecat-ai/client-js';
+import { DailyTransport } from '@pipecat-ai/daily-transport';
+import { RTVIClientProvider } from '@pipecat-ai/client-react';
+
+const transport = new DailyTransport();
+
+const client = new RTVIClient({
+  transport,
+  params: {
+    baseUrl: 'http://localhost:7860',
+    endpoints: {
+      connect: '/connect',
+    },
+  },
+  enableMic: true,
+  enableCam: false,
+});
+
+export function RTVIProvider({ children }: PropsWithChildren) {
+  return <RTVIClientProvider client={client}>{children}</RTVIClientProvider>;
+}

examples/simple-chatbot/server/bot-openai.py

@@ -4,18 +4,6 @@
 # SPDX-License-Identifier: BSD 2-Clause License
 #
 
-"""OpenAI Bot Implementation.
-
-This module implements a chatbot using OpenAI's GPT-4 model for natural language
-processing. It includes:
-- Real-time audio/video interaction through Daily
-- Animated robot avatar
-- Text-to-speech using ElevenLabs
-- Support for both English and Spanish
-
-The bot runs as part of a pipeline that processes audio/video frames and manages
-the conversation flow.
-"""
 
 import asyncio
 import os
@@ -24,150 +12,72 @@ import sys
 import aiohttp
 from dotenv import load_dotenv
 from loguru import logger
-from PIL import Image
-from runner import configure
 
 from pipecat.audio.vad.silero import SileroVADAnalyzer
-from pipecat.frames.frames import (
-    BotStartedSpeakingFrame,
-    BotStoppedSpeakingFrame,
-    Frame,
-    OutputImageRawFrame,
-    SpriteFrame,
-)
 from pipecat.pipeline.pipeline import Pipeline
 from pipecat.pipeline.runner import PipelineRunner
 from pipecat.pipeline.task import PipelineParams, PipelineTask
 from pipecat.processors.aggregators.openai_llm_context import OpenAILLMContext
-from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
 from pipecat.processors.frameworks.rtvi import RTVIConfig, RTVIObserver, RTVIProcessor
-from pipecat.services.elevenlabs.tts import ElevenLabsTTSService
 from pipecat.services.openai.llm import OpenAILLMService
 from pipecat.transports.services.daily import DailyParams, DailyTransport
+from pipecat.transports.services.helpers.daily_rest import (
+    DailyMeetingTokenParams,
+    DailyMeetingTokenProperties,
+    DailyRESTHelper,
+    DailyRoomParams,
+)
 
 load_dotenv(override=True)
 logger.remove(0)
 logger.add(sys.stderr, level="DEBUG")
 
-sprites = []
-script_dir = os.path.dirname(__file__)
-
-# Load sequential animation frames
-for i in range(1, 26):
-    # Build the full path to the image file
-    full_path = os.path.join(script_dir, f"assets/robot0{i}.png")
-    # Get the filename without the extension to use as the dictionary key
-    # Open the image and convert it to bytes
-    with Image.open(full_path) as img:
-        sprites.append(OutputImageRawFrame(image=img.tobytes(), size=img.size, format=img.format))
-
-# Create a smooth animation by adding reversed frames
-flipped = sprites[::-1]
-sprites.extend(flipped)
-
-# Define static and animated states
-quiet_frame = sprites[0]  # Static frame for when bot is listening
-talking_frame = SpriteFrame(images=sprites)  # Animation sequence for when bot is talking
-
-
-class TalkingAnimation(FrameProcessor):
-    """Manages the bot's visual animation states.
-
-    Switches between static (listening) and animated (talking) states based on
-    the bot's current speaking status.
-    """
-
-    def __init__(self):
-        super().__init__()
-        self._is_talking = False
-
-    async def process_frame(self, frame: Frame, direction: FrameDirection):
-        """Process incoming frames and update animation state.
-
-        Args:
-            frame: The incoming frame to process
-            direction: The direction of frame flow in the pipeline
-        """
-        await super().process_frame(frame, direction)
-
-        # Switch to talking animation when bot starts speaking
-        if isinstance(frame, BotStartedSpeakingFrame):
-            if not self._is_talking:
-                await self.push_frame(talking_frame)
-                self._is_talking = True
-        # Return to static frame when bot stops speaking
-        elif isinstance(frame, BotStoppedSpeakingFrame):
-            await self.push_frame(quiet_frame)
-            self._is_talking = False
-
-        await self.push_frame(frame, direction)
-
 
 async def main():
-    """Main bot execution function.
-
-    Sets up and runs the bot pipeline including:
-    - Daily video transport
-    - Speech-to-text and text-to-speech services
-    - Language model integration
-    - Animation processing
-    - RTVI event handling
-    """
+    """Main bot execution function."""
     async with aiohttp.ClientSession() as session:
-        (room_url, token) = await configure(session)
+        daily_rest_helper = DailyRESTHelper(
+            daily_api_key=os.getenv("DAILY_API_KEY"),
+            daily_api_url=os.getenv("DAILY_API_URL", "https://api.daily.co/v1"),
+            aiohttp_session=session,
+        )
+
+        room = await daily_rest_helper.create_room(
+            DailyRoomParams(properties={"enable_prejoin_ui": False})
+        )
+
+        token_params = DailyMeetingTokenParams(
+            properties=DailyMeetingTokenProperties(
+                is_owner=True,
+                permissions={
+                    "hasPresence": False,  # Example: join as a hidden participant
+                },
+                start_video_off=True,
+                start_audio_off=True,
+            )
+        )
+
+        token = await daily_rest_helper.get_token(room_url=room.url, params=token_params)
 
         # Set up Daily transport with video/audio parameters
         transport = DailyTransport(
-            room_url,
+            room.url,
             token,
             "Chatbot",
             DailyParams(
                 audio_in_enabled=True,
-                audio_out_enabled=True,
-                video_out_enabled=True,
-                video_out_width=1024,
-                video_out_height=576,
                 vad_analyzer=SileroVADAnalyzer(),
                 transcription_enabled=True,
-                #
-                # Spanish
-                #
-                # transcription_settings=DailyTranscriptionSettings(
-                #     language="es",
-                #     tier="nova",
-                #     model="2-general"
-                # )
             ),
         )
 
-        # Initialize text-to-speech service
-        tts = ElevenLabsTTSService(
-            api_key=os.getenv("ELEVENLABS_API_KEY"),
-            #
-            # English
-            #
-            voice_id="pNInz6obpgDQGcFmaJgB",
-            #
-            # Spanish
-            #
-            # model="eleven_multilingual_v2",
-            # voice_id="gD1IexrzCvsXPHUuT0s3",
-        )
-
         # Initialize LLM service
         llm = OpenAILLMService(api_key=os.getenv("OPENAI_API_KEY"))
 
         messages = [
             {
                 "role": "system",
-                #
-                # English
-                #
-                "content": "You are Chatbot, a friendly, helpful robot. Your goal is to demonstrate your capabilities in a succinct way. Your output will be converted to audio so don't include special characters in your answers. Respond to what the user said in a creative and helpful way, but keep your responses brief. Start by introducing yourself.",
-                #
-                # Spanish
-                #
-                # "content": "Eres Chatbot, un amigable y útil robot. Tu objetivo es demostrar tus capacidades de una manera breve. Tus respuestas se convertiran a audio así que nunca no debes incluir caracteres especiales. Contesta a lo que el usuario pregunte de una manera creativa, útil y breve. Empieza por presentarte a ti mismo.",
+                "content": "Summerize the conversation so far in a single sentence.",
             },
         ]
 
@@ -176,8 +86,6 @@ async def main():
         context = OpenAILLMContext(messages)
         context_aggregator = llm.create_context_aggregator(context)
 
-        ta = TalkingAnimation()
-
         #
         # RTVI events for Pipecat client UI
         #
@@ -189,8 +97,6 @@ async def main():
                 rtvi,
                 context_aggregator.user(),
                 llm,
-                tts,
-                ta,
                 transport.output(),
                 context_aggregator.assistant(),
             ]
@@ -204,7 +110,6 @@ async def main():
             ),
             observers=[RTVIObserver(rtvi)],
         )
-        await task.queue_frame(quiet_frame)
 
         @rtvi.event_handler("on_client_ready")
         async def on_client_ready(rtvi):

examples/twilio-chatbot/client/typescript/package.json

@@ -19,7 +19,7 @@
     "vite": "^6.0.2"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/websocket-transport": "^1.0.0"
+    "@pipecat-ai/client-js": "^0.4.0",
+    "@pipecat-ai/websocket-transport": "^0.4.2"
   }
 }

examples/twilio-chatbot/client/typescript/src/app.ts

@@ -5,22 +5,21 @@
  */
 
 import {
-  BotLLMTextData,
-  Participant,
-  PipecatClient,
-  PipecatClientOptions,
-  RTVIEvent, RTVIMessage, TranscriptData,
+  RTVIClient,
+  RTVIClientOptions,
+  RTVIEvent,
 } from '@pipecat-ai/client-js';
 import {
   WebSocketTransport,
   TwilioSerializer,
-} from '@pipecat-ai/websocket-transport';
+} from "@pipecat-ai/websocket-transport";
 
 class WebsocketClientApp {
-  private static STREAM_SID = 'ws_mock_stream_sid';
-  private static CALL_SID = 'ws_mock_call_sid';
 
-  private rtviClient: PipecatClient | null = null;
+  private static STREAM_SID = "ws_mock_stream_sid"
+  private static CALL_SID = "ws_mock_call_sid"
+
+  private rtviClient: RTVIClient | null = null;
   private connectBtn: HTMLButtonElement | null = null;
   private disconnectBtn: HTMLButtonElement | null = null;
   private statusSpan: HTMLElement | null = null;
@@ -39,12 +38,8 @@ class WebsocketClientApp {
    * Set up references to DOM elements and create necessary media elements
    */
   private setupDOMElements(): void {
-    this.connectBtn = document.getElementById(
-      'connect-btn'
-    ) as HTMLButtonElement;
-    this.disconnectBtn = document.getElementById(
-      'disconnect-btn'
-    ) as HTMLButtonElement;
+    this.connectBtn = document.getElementById('connect-btn') as HTMLButtonElement;
+    this.disconnectBtn = document.getElementById('disconnect-btn') as HTMLButtonElement;
     this.statusSpan = document.getElementById('connection-status');
     this.debugLog = document.getElementById('debug-log');
   }
@@ -85,23 +80,13 @@ class WebsocketClientApp {
   }
 
   private async emulateTwilioMessages() {
-    const connectedMessage = {
-      event: 'connected',
-      protocol: 'Call',
-      version: '1.0.0',
-    };
+    const connectedMessage={"event": "connected", "protocol": "Call", "version": "1.0.0"}
 
-    const websocketTransport = this.rtviClient?.transport as WebSocketTransport;
-    void websocketTransport?.sendRawMessage(connectedMessage);
+    const websocketTransport = this.rtviClient?.transport as WebSocketTransport
+    void websocketTransport?.sendRawMessage(connectedMessage)
 
-    const startMessage = {
-      event: 'start',
-      start: {
-        streamSid: WebsocketClientApp.STREAM_SID,
-        callSid: WebsocketClientApp.CALL_SID,
-      },
-    };
-    void websocketTransport?.sendRawMessage(startMessage);
+    const startMessage={"event": "start", "start": {"streamSid": WebsocketClientApp.STREAM_SID, "callSid": WebsocketClientApp.CALL_SID}}
+    void websocketTransport?.sendRawMessage(startMessage)
   }
 
   /**
@@ -124,7 +109,7 @@ class WebsocketClientApp {
     if (!this.rtviClient) return;
 
     // Listen for new tracks starting
-    this.rtviClient.on(RTVIEvent.TrackStarted, (track: MediaStreamTrack, participant?: Participant) => {
+    this.rtviClient.on(RTVIEvent.TrackStarted, (track, participant) => {
       // Only handle non-local (bot) tracks
       if (!participant?.local && track.kind === 'audio') {
         this.setupAudioTrack(track);
@@ -132,10 +117,8 @@ class WebsocketClientApp {
     });
 
     // Listen for tracks stopping
-    this.rtviClient.on(RTVIEvent.TrackStopped, (track: MediaStreamTrack, participant?: Participant) => {
-      this.log(
-        `Track stopped: ${track.kind} from ${participant?.name || 'unknown'}`
-      );
+    this.rtviClient.on(RTVIEvent.TrackStopped, (track, participant) => {
+      this.log(`Track stopped: ${track.kind} from ${participant?.name || 'unknown'}`);
     });
   }
 
@@ -145,10 +128,7 @@ class WebsocketClientApp {
    */
   private setupAudioTrack(track: MediaStreamTrack): void {
     this.log('Setting up audio track');
-    if (
-      this.botAudio.srcObject &&
-      'getAudioTracks' in this.botAudio.srcObject
-    ) {
+    if (this.botAudio.srcObject && "getAudioTracks" in this.botAudio.srcObject) {
       const oldTrack = this.botAudio.srcObject.getAudioTracks()[0];
       if (oldTrack?.id === track.id) return;
     }
@@ -163,19 +143,23 @@ class WebsocketClientApp {
     try {
       const startTime = Date.now();
 
-      const ws_opts = {
+      const transport = new WebSocketTransport({
         serializer: new TwilioSerializer(),
         recorderSampleRate: 8000,
-        playerSampleRate: 8000,
-        ws_url: 'http://localhost:8765/ws',
-      };
-      const RTVIConfig: PipecatClientOptions = {
-        transport: new WebSocketTransport(ws_opts),
+        playerSampleRate: 8000
+      });
+      const RTVIConfig: RTVIClientOptions = {
+        transport,
+        params: {
+          // The baseURL and endpoint of your bot server that the client will connect to
+          baseUrl: 'http://localhost:8765',
+          endpoints: { connect: '/' },
+        },
         enableMic: true,
         enableCam: false,
         callbacks: {
           onConnected: () => {
-            this.emulateTwilioMessages();
+            this.emulateTwilioMessages()
             this.updateStatus('Connected');
             if (this.connectBtn) this.connectBtn.disabled = true;
             if (this.disconnectBtn) this.disconnectBtn.disabled = false;
@@ -186,21 +170,27 @@ class WebsocketClientApp {
             if (this.disconnectBtn) this.disconnectBtn.disabled = true;
             this.log('Client disconnected');
           },
-          onBotReady: (data: any) => {
+          onBotReady: (data) => {
             this.log(`Bot ready: ${JSON.stringify(data)}`);
             this.setupMediaTracks();
           },
-          onUserTranscript: (data: TranscriptData) => {
+          onUserTranscript: (data) => {
             if (data.final) {
               this.log(`User: ${data.text}`);
             }
           },
-          onBotTranscript: (data: BotLLMTextData) => this.log(`Bot: ${data.text}`),
-          onMessageError: (error: RTVIMessage) => console.error('Message error:', error),
-          onError: (error: RTVIMessage) => console.error('Error:', error),
+          onBotTranscript: (data) => this.log(`Bot: ${data.text}`),
+          onMessageError: (error) => console.error('Message error:', error),
+          onError: (error) => console.error('Error:', error),
         },
-      };
-      this.rtviClient = new PipecatClient(RTVIConfig);
+      }
+      // @ts-ignore
+      RTVIConfig.customConnectHandler = () => Promise.resolve(
+          {
+              ws_url: "/ws",
+          }
+      );
+      this.rtviClient = new RTVIClient(RTVIConfig);
       this.setupTrackListeners();
 
       this.log('Initializing devices...');
@@ -233,13 +223,8 @@ class WebsocketClientApp {
       try {
         await this.rtviClient.disconnect();
         this.rtviClient = null;
-        if (
-          this.botAudio.srcObject &&
-          'getAudioTracks' in this.botAudio.srcObject
-        ) {
-          this.botAudio.srcObject
-            .getAudioTracks()
-            .forEach((track) => track.stop());
+        if (this.botAudio.srcObject && "getAudioTracks" in this.botAudio.srcObject) {
+          this.botAudio.srcObject.getAudioTracks().forEach((track) => track.stop());
           this.botAudio.srcObject = null;
         }
       } catch (error) {
@@ -247,6 +232,7 @@ class WebsocketClientApp {
       }
     }
   }
+
 }
 
 declare global {

examples/websocket/client/package.json

@@ -19,8 +19,8 @@
     "vite": "^6.3.5"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/websocket-transport": "^1.0.0",
+    "@pipecat-ai/client-js": "^0.4.0",
+    "@pipecat-ai/websocket-transport": "^0.4.2",
     "protobufjs": "^7.4.0"
   }
 }

examples/websocket/client/src/app.ts

@@ -5,7 +5,7 @@
  */
 
 /**
- * Pipecat Client Implementation
+ * RTVI Client Implementation
  *
  * This client connects to an RTVI-compatible bot server using WebSocket.
  *
@@ -14,14 +14,16 @@
  */
 
 import {
-  PipecatClient,
-  PipecatClientOptions,
+  RTVIClient,
+  RTVIClientOptions,
   RTVIEvent,
 } from '@pipecat-ai/client-js';
-import { WebSocketTransport } from '@pipecat-ai/websocket-transport';
+import {
+  WebSocketTransport
+} from "@pipecat-ai/websocket-transport";
 
 class WebsocketClientApp {
-  private pcClient: PipecatClient | null = null;
+  private rtviClient: RTVIClient | null = null;
   private connectBtn: HTMLButtonElement | null = null;
   private disconnectBtn: HTMLButtonElement | null = null;
   private statusSpan: HTMLElement | null = null;
@@ -29,7 +31,7 @@ class WebsocketClientApp {
   private botAudio: HTMLAudioElement;
 
   constructor() {
-    console.log('WebsocketClientApp');
+    console.log("WebsocketClientApp");
     this.botAudio = document.createElement('audio');
     this.botAudio.autoplay = true;
     //this.botAudio.playsInline = true;
@@ -43,12 +45,8 @@ class WebsocketClientApp {
    * Set up references to DOM elements and create necessary media elements
    */
   private setupDOMElements(): void {
-    this.connectBtn = document.getElementById(
-      'connect-btn'
-    ) as HTMLButtonElement;
-    this.disconnectBtn = document.getElementById(
-      'disconnect-btn'
-    ) as HTMLButtonElement;
+    this.connectBtn = document.getElementById('connect-btn') as HTMLButtonElement;
+    this.disconnectBtn = document.getElementById('disconnect-btn') as HTMLButtonElement;
     this.statusSpan = document.getElementById('connection-status');
     this.debugLog = document.getElementById('debug-log');
   }
@@ -93,8 +91,8 @@ class WebsocketClientApp {
    * This is called when the bot is ready or when the transport state changes to ready
    */
   setupMediaTracks() {
-    if (!this.pcClient) return;
-    const tracks = this.pcClient.tracks();
+    if (!this.rtviClient) return;
+    const tracks = this.rtviClient.tracks();
     if (tracks.bot?.audio) {
       this.setupAudioTrack(tracks.bot.audio);
     }
@@ -105,10 +103,10 @@ class WebsocketClientApp {
    * This handles new tracks being added during the session
    */
   setupTrackListeners() {
-    if (!this.pcClient) return;
+    if (!this.rtviClient) return;
 
     // Listen for new tracks starting
-    this.pcClient.on(RTVIEvent.TrackStarted, (track, participant) => {
+    this.rtviClient.on(RTVIEvent.TrackStarted, (track, participant) => {
       // Only handle non-local (bot) tracks
       if (!participant?.local && track.kind === 'audio') {
         this.setupAudioTrack(track);
@@ -116,10 +114,8 @@ class WebsocketClientApp {
     });
 
     // Listen for tracks stopping
-    this.pcClient.on(RTVIEvent.TrackStopped, (track, participant) => {
-      this.log(
-        `Track stopped: ${track.kind} from ${participant?.name || 'unknown'}`
-      );
+    this.rtviClient.on(RTVIEvent.TrackStopped, (track, participant) => {
+      this.log(`Track stopped: ${track.kind} from ${participant?.name || 'unknown'}`);
     });
   }
 
@@ -129,10 +125,7 @@ class WebsocketClientApp {
    */
   private setupAudioTrack(track: MediaStreamTrack): void {
     this.log('Setting up audio track');
-    if (
-      this.botAudio.srcObject &&
-      'getAudioTracks' in this.botAudio.srcObject
-    ) {
+    if (this.botAudio.srcObject && "getAudioTracks" in this.botAudio.srcObject) {
       const oldTrack = this.botAudio.srcObject.getAudioTracks()[0];
       if (oldTrack?.id === track.id) return;
     }
@@ -141,15 +134,21 @@ class WebsocketClientApp {
 
   /**
    * Initialize and connect to the bot
-   * This sets up the Pipecat client, initializes devices, and establishes the connection
+   * This sets up the RTVI client, initializes devices, and establishes the connection
    */
   public async connect(): Promise<void> {
     try {
       const startTime = Date.now();
 
       //const transport = new DailyTransport();
-      const PipecatConfig: PipecatClientOptions = {
-        transport: new WebSocketTransport(),
+      const transport = new WebSocketTransport();
+      const RTVIConfig: RTVIClientOptions = {
+        transport,
+        params: {
+          // The baseURL and endpoint of your bot server that the client will connect to
+          baseUrl: 'http://localhost:7860',
+          endpoints: { connect: '/connect' },
+        },
         enableMic: true,
         enableCam: false,
         callbacks: {
@@ -177,20 +176,15 @@ class WebsocketClientApp {
           onMessageError: (error) => console.error('Message error:', error),
           onError: (error) => console.error('Error:', error),
         },
-      };
-      this.pcClient = new PipecatClient(PipecatConfig);
-      // @ts-ignore
-      window.pcClient = this.pcClient; // Expose for debugging
+      }
+      this.rtviClient = new RTVIClient(RTVIConfig);
       this.setupTrackListeners();
 
       this.log('Initializing devices...');
-      await this.pcClient.initDevices();
+      await this.rtviClient.initDevices();
 
       this.log('Connecting to bot...');
-      await this.pcClient.connect({
-        // The baseURL and endpoint of your bot server that the client will connect to
-        endpoint: 'http://localhost:7860/connect',
-      });
+      await this.rtviClient.connect();
 
       const timeTaken = Date.now() - startTime;
       this.log(`Connection complete, timeTaken: ${timeTaken}`);
@@ -198,9 +192,9 @@ class WebsocketClientApp {
       this.log(`Error connecting: ${(error as Error).message}`);
       this.updateStatus('Error');
       // Clean up if there's an error
-      if (this.pcClient) {
+      if (this.rtviClient) {
         try {
-          await this.pcClient.disconnect();
+          await this.rtviClient.disconnect();
         } catch (disconnectError) {
           this.log(`Error during disconnect: ${disconnectError}`);
         }
@@ -212,17 +206,12 @@ class WebsocketClientApp {
    * Disconnect from the bot and clean up media resources
    */
   public async disconnect(): Promise<void> {
-    if (this.pcClient) {
+    if (this.rtviClient) {
       try {
-        await this.pcClient.disconnect();
-        this.pcClient = null;
-        if (
-          this.botAudio.srcObject &&
-          'getAudioTracks' in this.botAudio.srcObject
-        ) {
-          this.botAudio.srcObject
-            .getAudioTracks()
-            .forEach((track) => track.stop());
+        await this.rtviClient.disconnect();
+        this.rtviClient = null;
+        if (this.botAudio.srcObject && "getAudioTracks" in this.botAudio.srcObject) {
+          this.botAudio.srcObject.getAudioTracks().forEach((track) => track.stop());
           this.botAudio.srcObject = null;
         }
       } catch (error) {
@@ -230,6 +219,7 @@ class WebsocketClientApp {
       }
     }
   }
+
 }
 
 declare global {

examples/word-wrangler-gemini-live/client/package.json

@@ -9,12 +9,11 @@
     "lint": "next lint"
   },
   "dependencies": {
-    "@pipecat-ai/client-js": "^1.0.0",
-    "@pipecat-ai/client-react": "^1.0.0",
-    "@pipecat-ai/daily-transport": "^1.0.0",
+    "@pipecat-ai/client-js": "^0.3.5",
+    "@pipecat-ai/client-react": "^0.3.5",
+    "@pipecat-ai/daily-transport": "^0.3.10",
     "@tabler/icons-react": "^3.31.0",
     "@tailwindcss/postcss": "^4.1.3",
-    "jotai": "^2.12.5",
     "js-confetti": "^0.12.0",
     "next": "15.2.4",
     "react": "^19.0.0",

examples/word-wrangler-gemini-live/client/src/hooks/useConnectionState.ts

@@ -1,26 +1,16 @@
 import { useEffect, useCallback } from 'react';
 import {
-  usePipecatClient,
-  usePipecatClientTransportState,
+  useRTVIClient,
+  useRTVIClientTransportState,
 } from '@pipecat-ai/client-react';
 import { CONNECTION_STATES } from '@/constants/gameConstants';
-import { useConfigurationSettings } from '@/contexts/Configuration';
-
-// Get the API base URL from environment variables
-// Default to "/api" if not specified
-// "/api" is the default for Next.js API routes and used
-// for the Pipecat Cloud deployed agent
-const API_BASE_URL = process.env.NEXT_PUBLIC_API_BASE_URL || '/api';
-
-console.log('Using API base URL:', API_BASE_URL);
 
 export function useConnectionState(
   onConnected?: () => void,
   onDisconnected?: () => void
 ) {
-  const client = usePipecatClient();
-  const transportState = usePipecatClientTransportState();
-  const config = useConfigurationSettings();
+  const client = useRTVIClient();
+  const transportState = useRTVIClientTransportState();
 
   const isConnected = CONNECTION_STATES.ACTIVE.includes(transportState);
   const isConnecting = CONNECTION_STATES.CONNECTING.includes(transportState);
@@ -45,17 +35,12 @@ export function useConnectionState(
       if (isConnected) {
         await client.disconnect();
       } else {
-        await client.connect({
-          endpoint: `${API_BASE_URL}/connect`,
-          requestData: {
-            personality: config.personality,
-          },
-        });
+        await client.connect();
       }
     } catch (error) {
       console.error('Connection error:', error);
     }
-  }, [client, config, isConnected]);
+  }, [client, isConnected]);
 
   return {
     isConnected,

examples/word-wrangler-gemini-live/client/src/pages/_app.tsx

@@ -1,15 +1,15 @@
-import { ConfigurationProvider } from '@/contexts/Configuration';
-import { PipecatProvider } from '@/providers/PipecatProvider';
-import { PipecatClientAudio } from '@pipecat-ai/client-react';
-import type { AppProps } from 'next/app';
-import { Nunito } from 'next/font/google';
-import Head from 'next/head';
-import '../styles/globals.css';
+import { ConfigurationProvider } from "@/contexts/Configuration";
+import { RTVIProvider } from "@/providers/RTVIProvider";
+import { RTVIClientAudio } from "@pipecat-ai/client-react";
+import type { AppProps } from "next/app";
+import { Nunito } from "next/font/google";
+import Head from "next/head";
+import "../styles/globals.css";
 
 const nunito = Nunito({
-  subsets: ['latin'],
-  display: 'swap',
-  variable: '--font-sans',
+  subsets: ["latin"],
+  display: "swap",
+  variable: "--font-sans",
 });
 
 export default function App({ Component, pageProps }: AppProps) {
@@ -21,10 +21,10 @@ export default function App({ Component, pageProps }: AppProps) {
       </Head>
       <main className={`${nunito.variable}`}>
         <ConfigurationProvider>
-          <PipecatProvider>
-            <PipecatClientAudio />
+          <RTVIProvider>
+            <RTVIClientAudio />
             <Component {...pageProps} />
-          </PipecatProvider>
+          </RTVIProvider>
         </ConfigurationProvider>
       </main>
     </>

examples/word-wrangler-gemini-live/client/src/pages/api/connect.ts

@@ -1,11 +1,11 @@
-import type { NextApiRequest, NextApiResponse } from 'next';
+import type { NextApiRequest, NextApiResponse } from "next";
 
 export default async function handler(
   req: NextApiRequest,
   res: NextApiResponse
 ) {
-  if (req.method !== 'POST') {
-    return res.status(405).json({ error: 'Method not allowed' });
+  if (req.method !== "POST") {
+    return res.status(405).json({ error: "Method not allowed" });
   }
 
   try {
@@ -15,16 +15,16 @@ export default async function handler(
     if (!personality) {
       return res
         .status(400)
-        .json({ error: 'Missing required configuration parameters' });
+        .json({ error: "Missing required configuration parameters" });
     }
 
     const response = await fetch(
       `https://api.pipecat.daily.co/v1/public/${process.env.AGENT_NAME}/start`,
       {
-        method: 'POST',
+        method: "POST",
         headers: {
           Authorization: `Bearer ${process.env.PIPECAT_CLOUD_API_KEY}`,
-          'Content-Type': 'application/json',
+          "Content-Type": "application/json",
         },
         body: JSON.stringify({
           createDailyRoom: true,
@@ -37,15 +37,15 @@ export default async function handler(
 
     const data = await response.json();
 
-    console.log('Response from API:', JSON.stringify(data, null, 2));
+    console.log("Response from API:", JSON.stringify(data, null, 2));
 
-    // Transform the response to match what Pipecat client expects
+    // Transform the response to match what RTVI client expects
     return res.status(200).json({
       room_url: data.dailyRoom,
       token: data.dailyToken,
     });
   } catch (error) {
-    console.error('Error starting agent:', error);
-    return res.status(500).json({ error: 'Failed to start agent' });
+    console.error("Error starting agent:", error);
+    return res.status(500).json({ error: "Failed to start agent" });
   }
 }

examples/word-wrangler-gemini-live/client/src/providers/PipecatProvider.tsx

@@ -1,43 +0,0 @@
-'use client';
-
-import { PipecatClient } from '@pipecat-ai/client-js';
-import { DailyTransport } from '@pipecat-ai/daily-transport';
-import { PipecatClientProvider } from '@pipecat-ai/client-react';
-import { PropsWithChildren, useEffect, useState, useRef } from 'react';
-
-export function PipecatProvider({ children }: PropsWithChildren) {
-  const [client, setClient] = useState<PipecatClient | null>(null);
-  const clientCreated = useRef(false);
-
-  useEffect(() => {
-    // Only create the client once
-    if (clientCreated.current) return;
-
-    const pcClient = new PipecatClient({
-      transport: new DailyTransport(),
-      enableMic: true,
-      enableCam: false,
-    });
-
-    setClient(pcClient);
-    clientCreated.current = true;
-
-    // Cleanup when component unmounts
-    return () => {
-      if (pcClient) {
-        pcClient.disconnect().catch((err) => {
-          console.error('Error disconnecting client:', err);
-        });
-      }
-      clientCreated.current = false;
-    };
-  }, []);
-
-  if (!client) {
-    return null;
-  }
-
-  return (
-    <PipecatClientProvider client={client}>{children}</PipecatClientProvider>
-  );
-}

examples/word-wrangler-gemini-live/client/src/providers/RTVIProvider.tsx

@@ -0,0 +1,72 @@
+"use client";
+
+import { RTVIClient } from "@pipecat-ai/client-js";
+import { DailyTransport } from "@pipecat-ai/daily-transport";
+import { RTVIClientProvider } from "@pipecat-ai/client-react";
+import { PropsWithChildren, useEffect, useState, useRef } from "react";
+import { useConfigurationSettings } from "@/contexts/Configuration";
+
+// Get the API base URL from environment variables
+// Default to "/api" if not specified
+// "/api" is the default for Next.js API routes and used
+// for the Pipecat Cloud deployed agent
+const API_BASE_URL = process.env.NEXT_PUBLIC_API_BASE_URL || "/api";
+
+console.log("Using API base URL:", API_BASE_URL);
+
+export function RTVIProvider({ children }: PropsWithChildren) {
+  const [client, setClient] = useState<RTVIClient | null>(null);
+  const config = useConfigurationSettings();
+  const clientCreated = useRef(false);
+
+  useEffect(() => {
+    // Only create the client once
+    if (clientCreated.current) return;
+
+    const transport = new DailyTransport();
+
+    const rtviClient = new RTVIClient({
+      transport,
+      params: {
+        baseUrl: API_BASE_URL,
+        endpoints: {
+          connect: "/connect",
+        },
+        requestData: {
+          personality: config.personality,
+        },
+      },
+      enableMic: true,
+      enableCam: false,
+    });
+
+    setClient(rtviClient);
+    clientCreated.current = true;
+
+    // Cleanup when component unmounts
+    return () => {
+      if (rtviClient) {
+        rtviClient.disconnect().catch((err) => {
+          console.error("Error disconnecting client:", err);
+        });
+      }
+      clientCreated.current = false;
+    };
+  }, []);
+
+  // Update the connectParams when config changes
+  useEffect(() => {
+    if (!client) return;
+
+    // Update the connect params without recreating the client
+    client.params.requestData = {
+      personality: config.personality,
+    };
+  }, [client, config.personality]);
+
+  if (!client) {
+    return null;
+  }
+
+  return <RTVIClientProvider client={client}>{children}</RTVIClientProvider>;
+}

examples/word-wrangler-gemini-live/server/bot.py

@@ -4,7 +4,6 @@
 # SPDX-License-Identifier: BSD 2-Clause License
 #
 
-import argparse
 import asyncio
 import os
 import sys
@@ -199,15 +198,16 @@ async def bot(args: DailySessionArguments):
 
 
 # Local development
-async def local_daily(args: DailySessionArguments):
+async def local_daily():
     """Daily transport for local development."""
-    # from runner import configure
+    from runner import configure
 
     try:
         async with aiohttp.ClientSession() as session:
+            (room_url, token) = await configure(session)
             transport = DailyTransport(
-                room_url=args.room_url,
-                token=args.token,
+                room_url,
+                token,
                 bot_name="Bot",
                 params=DailyParams(
                     audio_in_enabled=True,
@@ -217,7 +217,7 @@ async def local_daily(args: DailySessionArguments):
             )
 
             test_config = {
-                "personality": args.personality,
+                "personality": "witty",
             }
 
             await main(transport, test_config)
@@ -227,24 +227,7 @@ async def local_daily(args: DailySessionArguments):
 
 # Local development entry point
 if LOCAL_RUN and __name__ == "__main__":
-    parser = argparse.ArgumentParser(
-        description="Run the Word Wrangler bot in local development mode"
-    )
-    parser.add_argument(
-        "-u", "--room-url", type=str, default=os.getenv("DAILY_SAMPLE_ROOM_URL", "")
-    )
-    parser.add_argument(
-        "-t", "--token", type=str, default=os.getenv("DAILY_SAMPLE_ROOM_TOKEN", None)
-    )
-    parser.add_argument(
-        "-p",
-        "--personality",
-        default="witty",
-        choices=["friendly", "professional", "enthusiastic", "thoughtful", "witty"],
-        help="Personality preset for the bot (friendly, professional, enthusiastic, thoughtful, witty)",
-    )
-    args = parser.parse_args()
     try:
-        asyncio.run(local_daily(args))
+        asyncio.run(local_daily())
     except Exception as e:
         logger.exception(f"Failed to run in local mode: {e}")

examples/word-wrangler-gemini-live/server/server.py

@@ -160,15 +160,14 @@ async def rtvi_connect(request: Request) -> Dict[Any, Any]:
     Raises:
         HTTPException: If room creation, token generation, or bot startup fails
     """
-    body = await request.json()
-    print("Creating room for RTVI connection", body)
+    print("Creating room for RTVI connection")
     room_url, token = await create_room_and_token()
     print(f"Room URL: {room_url}")
 
     # Start the bot process
     try:
         proc = subprocess.Popen(
-            [f"python3 -m bot -u {room_url} -t {token} -p {body.get('personality', 'witty')}"],
+            [f"python3 -m bot -u {room_url} -t {token}"],
             shell=True,
             bufsize=1,
             cwd=os.path.dirname(os.path.abspath(__file__)),

pyproject.toml

@@ -25,14 +25,14 @@ dependencies = [
     "docstring_parser~=0.16",
     "loguru~=0.7.3",
     "Markdown~=3.7",
-    "numpy>=1.26.4",
+    "numpy~=1.26.4",
     "Pillow~=11.1.0",
     "protobuf~=5.29.3",
     "pydantic~=2.10.6",
     "pyloudnorm~=0.1.1",
     "resampy~=0.4.3",
     "soxr~=0.5.0",
-    "openai~=1.74.0",
+    "openai~=1.70.0",
 ]
 
 [project.urls]
@@ -42,7 +42,7 @@ Website = "https://pipecat.ai"
 [project.optional-dependencies]
 anthropic = [ "anthropic~=0.49.0" ]
 assemblyai = [ "websockets~=13.1" ]
-aws = [ "aioboto3~=15.0.0", "websockets~=13.1" ]
+aws = [ "boto3~=1.37.16", "websockets~=13.1" ]
 aws-nova-sonic = [ "aws_sdk_bedrock_runtime~=0.0.2" ]
 azure = [ "azure-cognitiveservices-speech~=1.42.0"]
 cartesia = [ "cartesia~=2.0.3", "websockets~=13.1" ]
@@ -68,7 +68,7 @@ local = [ "pyaudio~=0.2.14" ]
 mcp = [ "mcp[cli]~=1.9.4" ]
 mem0 = [ "mem0ai~=0.1.94" ]
 mlx-whisper = [ "mlx-whisper~=0.4.2" ]
-moondream = [ "einops~=0.8.0", "timm~=1.0.13", "transformers>=4.48.0" ]
+moondream = [ "einops~=0.8.0", "timm~=1.0.13", "transformers~=4.48.0" ]
 nim = []
 neuphonic = [ "pyneuphonic~=1.5.13", "websockets~=13.1" ]
 noisereduce = [ "noisereduce~=3.0.3" ]
@@ -79,7 +79,7 @@ perplexity = []
 playht = [ "pyht~=0.1.12", "websockets~=13.1" ]
 qwen = []
 rime = [ "websockets~=13.1" ]
-riva = [ "nvidia-riva-client~=2.21.1" ]
+riva = [ "nvidia-riva-client~=2.19.1" ]
 sambanova = []
 sentry = [ "sentry-sdk~=2.23.1" ]
 local-smart-turn = [ "coremltools>=8.0", "transformers", "torch==2.5.0", "torchaudio==2.5.0" ]

scripts/fix-ruff.sh

@@ -1,10 +1,5 @@
-
-#!/bin/bash
-
-SCRIPT_DIR="$(cd "$(dirname "${BASH_SOURCE[0]}")" && pwd)"
-PROJECT_ROOT="$(dirname "$SCRIPT_DIR")"
-
-echo "Running ruff format..."
-ruff format "$PROJECT_ROOT"
-echo "Running ruff check..."
-ruff check --fix "$PROJECT_ROOT"
+ruff format src
+ruff format examples
+ruff format tests
+ruff format scripts
+ruff check --select I,D --fix

src/pipecat/audio/turn/base_turn_analyzer.py

@@ -76,16 +76,6 @@ class BaseTurnAnalyzer(ABC):
         """
         pass
 
-    @property
-    @abstractmethod
-    def params(self):
-        """Get the current turn analyzer parameters.
-
-        Returns:
-            Current turn analyzer configuration parameters.
-        """
-        pass
-
     @abstractmethod
     def append_audio(self, buffer: bytes, is_speech: bool) -> EndOfTurnState:
         """Appends audio data for analysis.

src/pipecat/audio/turn/smart_turn/base_smart_turn.py

@@ -87,15 +87,6 @@ class BaseSmartTurn(BaseTurnAnalyzer):
         """
         return self._speech_triggered
 
-    @property
-    def params(self) -> SmartTurnParams:
-        """Get the current smart turn parameters.
-
-        Returns:
-            Current smart turn configuration parameters.
-        """
-        return self._params
-
     def append_audio(self, buffer: bytes, is_speech: bool) -> EndOfTurnState:
         """Append audio data for turn analysis.
 

src/pipecat/audio/vad/vad_analyzer.py

@@ -183,37 +183,36 @@ class VADAnalyzer(ABC):
         if len(self._vad_buffer) < num_required_bytes:
             return self._vad_state
 
-        while len(self._vad_buffer) >= num_required_bytes:
-            audio_frames = self._vad_buffer[:num_required_bytes]
-            self._vad_buffer = self._vad_buffer[num_required_bytes:]
+        audio_frames = self._vad_buffer[:num_required_bytes]
+        self._vad_buffer = self._vad_buffer[num_required_bytes:]
 
-            confidence = self.voice_confidence(audio_frames)
+        confidence = self.voice_confidence(audio_frames)
 
-            volume = self._get_smoothed_volume(audio_frames)
-            self._prev_volume = volume
+        volume = self._get_smoothed_volume(audio_frames)
+        self._prev_volume = volume
 
-            speaking = confidence >= self._params.confidence and volume >= self._params.min_volume
+        speaking = confidence >= self._params.confidence and volume >= self._params.min_volume
 
-            if speaking:
-                match self._vad_state:
-                    case VADState.QUIET:
-                        self._vad_state = VADState.STARTING
-                        self._vad_starting_count = 1
-                    case VADState.STARTING:
-                        self._vad_starting_count += 1
-                    case VADState.STOPPING:
-                        self._vad_state = VADState.SPEAKING
-                        self._vad_stopping_count = 0
-            else:
-                match self._vad_state:
-                    case VADState.STARTING:
-                        self._vad_state = VADState.QUIET
-                        self._vad_starting_count = 0
-                    case VADState.SPEAKING:
-                        self._vad_state = VADState.STOPPING
-                        self._vad_stopping_count = 1
-                    case VADState.STOPPING:
-                        self._vad_stopping_count += 1
+        if speaking:
+            match self._vad_state:
+                case VADState.QUIET:
+                    self._vad_state = VADState.STARTING
+                    self._vad_starting_count = 1
+                case VADState.STARTING:
+                    self._vad_starting_count += 1
+                case VADState.STOPPING:
+                    self._vad_state = VADState.SPEAKING
+                    self._vad_stopping_count = 0
+        else:
+            match self._vad_state:
+                case VADState.STARTING:
+                    self._vad_state = VADState.QUIET
+                    self._vad_starting_count = 0
+                case VADState.SPEAKING:
+                    self._vad_state = VADState.STOPPING
+                    self._vad_stopping_count = 1
+                case VADState.STOPPING:
+                    self._vad_stopping_count += 1
 
         if (
             self._vad_state == VADState.STARTING

src/pipecat/examples/run.py

@@ -9,21 +9,6 @@
 This module provides a unified interface for running Pipecat examples across
 different transport types including Daily.co, WebRTC, and Twilio. It handles
 setup, configuration, and lifecycle management for each transport type.
-
-Example usage:
-    SmallWebRTCTransport::
-
-        python bot.py --transport webrtc
-
-    DailyTransport::
-
-        python bot.py --transport daily
-
-    Twilio::
-
-        python bot.py --transport twilio --proxy username.ngrok.io
-        # Note: Concurrently, run an ngrok tunnel to your local server:
-        # ngrok http 7860
 """
 
 import argparse

src/pipecat/frames/frames.py

@@ -28,7 +28,6 @@ from typing import (
 )
 
 from pipecat.audio.interruptions.base_interruption_strategy import BaseInterruptionStrategy
-from pipecat.audio.turn.smart_turn.base_smart_turn import SmartTurnParams
 from pipecat.audio.vad.vad_analyzer import VADParams
 from pipecat.metrics.metrics import MetricsData
 from pipecat.transcriptions.language import Language
@@ -1146,23 +1145,6 @@ class OutputDTMFUrgentFrame(DTMFFrame, SystemFrame):
     pass
 
 
-@dataclass
-class SpeechControlParamsFrame(SystemFrame):
-    """Frame for notifying processors of speech control parameter changes.
-
-    This includes parameters for both VAD (Voice Activity Detection) and
-    turn-taking analysis. It allows downstream processors to adjust their
-    behavior based on updated interaction control settings.
-
-    Parameters:
-        vad_params: Current VAD parameters.
-        turn_params: Current turn-taking analysis parameters.
-    """
-
-    vad_params: Optional[VADParams] = None
-    turn_params: Optional[SmartTurnParams] = None
-
-
 #
 # Control frames
 #

src/pipecat/pipeline/parallel_pipeline.py

@@ -273,17 +273,12 @@ class ParallelPipeline(BasePipeline):
         if not self._down_task:
             self._down_task = self.create_task(self._process_down_queue())
 
-    async def _drain_queue(self, queue: asyncio.Queue):
-        try:
-            while not queue.empty():
-                queue.get_nowait()
-        except asyncio.QueueEmpty:
-            logger.debug(f"Draining {self} queue already empty")
-
     async def _drain_queues(self):
         """Drain all frames from upstream and downstream queues."""
-        await self._drain_queue(self._up_queue)
-        await self._drain_queue(self._down_queue)
+        while not self._up_queue.empty:
+            await self._up_queue.get()
+        while not self._down_queue.empty:
+            await self._down_queue.get()
 
     async def _handle_interruption(self):
         """Handle interruption by cancelling tasks, draining queues, and restarting."""

src/pipecat/processors/aggregators/llm_response.py

@@ -19,8 +19,6 @@ from typing import Dict, List, Literal, Optional, Set
 from loguru import logger
 
 from pipecat.audio.interruptions.base_interruption_strategy import BaseInterruptionStrategy
-from pipecat.audio.turn.smart_turn.base_smart_turn import SmartTurnParams
-from pipecat.audio.vad.vad_analyzer import VADParams
 from pipecat.frames.frames import (
     BotInterruptionFrame,
     BotStartedSpeakingFrame,
@@ -45,7 +43,6 @@ from pipecat.frames.frames import (
     LLMSetToolsFrame,
     LLMTextFrame,
     OpenAILLMContextAssistantTimestampFrame,
-    SpeechControlParamsFrame,
     StartFrame,
     StartInterruptionFrame,
     TextFrame,
@@ -70,13 +67,9 @@ class LLMUserAggregatorParams:
         aggregation_timeout: Maximum time in seconds to wait for additional
             transcription content before pushing aggregated result. This
             timeout is used only when the transcription is slow to arrive.
-        turn_emulated_vad_timeout: Maximum time in seconds to wait for emulated
-            VAD when using turn-based analysis. Applied when transcription is
-            received but VAD didn't detect speech (e.g., whispered utterances).
     """
 
     aggregation_timeout: float = 0.5
-    turn_emulated_vad_timeout: float = 0.8
 
 
 @dataclass
@@ -397,9 +390,6 @@ class LLMUserContextAggregator(LLMContextResponseAggregator):
         """
         super().__init__(context=context, role="user", **kwargs)
         self._params = params or LLMUserAggregatorParams()
-        self._vad_params: Optional[VADParams] = None
-        self._turn_params: Optional[SmartTurnParams] = None
-
         if "aggregation_timeout" in kwargs:
             import warnings
 
@@ -487,10 +477,6 @@ class LLMUserContextAggregator(LLMContextResponseAggregator):
             self.set_tools(frame.tools)
         elif isinstance(frame, LLMSetToolChoiceFrame):
             self.set_tool_choice(frame.tool_choice)
-        elif isinstance(frame, SpeechControlParamsFrame):
-            self._vad_params = frame.vad_params
-            self._turn_params = frame.turn_params
-            await self.push_frame(frame, direction)
         else:
             await self.push_frame(frame, direction)
 
@@ -632,40 +618,9 @@ class LLMUserContextAggregator(LLMContextResponseAggregator):
     async def _aggregation_task_handler(self):
         while True:
             try:
-                # The _aggregation_task_handler handles two distinct timeout scenarios:
-                #
-                # 1. When emulating_vad=True: Wait for emulated VAD timeout before
-                #    pushing aggregation (simulating VAD behavior when no actual VAD
-                #    detection occurred).
-                #
-                # 2. When emulating_vad=False: Use aggregation_timeout as a buffer
-                #    to wait for potential late-arriving transcription frames after
-                #    a real VAD event.
-                #
-                # For emulated VAD scenarios, the timeout strategy depends on whether
-                # a turn analyzer is configured:
-                #
-                # - WITH turn analyzer: Use turn_emulated_vad_timeout parameter because
-                #   the VAD's stop_secs is set very low (e.g. 0.2s) for rapid speech
-                #   chunking to feed the turn analyzer. This low value is too fast
-                #   for emulated VAD scenarios where we need to allow users time to
-                #   finish speaking (e.g. 0.8s).
-                #
-                # - WITHOUT turn analyzer: Use VAD's stop_secs directly to maintain
-                #   consistent user experience between real VAD detection and
-                #   emulated VAD scenarios.
-                if not self._emulating_vad:
-                    timeout = self._params.aggregation_timeout
-                elif self._turn_params:
-                    timeout = self._params.turn_emulated_vad_timeout
-                else:
-                    # Use VAD stop_secs when no turn analyzer is present, fallback if no VAD params
-                    timeout = (
-                        self._vad_params.stop_secs
-                        if self._vad_params
-                        else self._params.turn_emulated_vad_timeout
-                    )
-                await asyncio.wait_for(self._aggregation_event.wait(), timeout)
+                await asyncio.wait_for(
+                    self._aggregation_event.wait(), self._params.aggregation_timeout
+                )
                 await self._maybe_emulate_user_speaking()
             except asyncio.TimeoutError:
                 if not self._user_speaking:

src/pipecat/processors/frameworks/rtvi.py

@@ -44,7 +44,6 @@ from pipecat.frames.frames import (
     InterimTranscriptionFrame,
     LLMFullResponseEndFrame,
     LLMFullResponseStartFrame,
-    LLMMessagesAppendFrame,
     LLMTextFrame,
     MetricsFrame,
     StartFrame,
@@ -72,14 +71,13 @@ from pipecat.processors.frame_processor import FrameDirection, FrameProcessor
 from pipecat.services.llm_service import (
     FunctionCallParams,  # TODO(aleix): we shouldn't import `services` from `processors`
 )
-from pipecat.services.openai.llm import OpenAIContextAggregatorPair
 from pipecat.transports.base_input import BaseInputTransport
 from pipecat.transports.base_output import BaseOutputTransport
 from pipecat.transports.base_transport import BaseTransport
 from pipecat.utils.asyncio.watchdog_queue import WatchdogQueue
 from pipecat.utils.string import match_endofsentence
 
-RTVI_PROTOCOL_VERSION = "1.0.0"
+RTVI_PROTOCOL_VERSION = "0.3.0"
 
 RTVI_MESSAGE_LABEL = "rtvi-ai"
 RTVIMessageLiteral = Literal["rtvi-ai"]
@@ -92,10 +90,6 @@ class RTVIServiceOption(BaseModel):
 
     Defines a configurable option that can be set for an RTVI service,
     including its name, type, and handler function.
-
-    .. deprecated:: 0.0.75
-        Pipeline Configuration has been removed as part of the RTVI protocol 1.0.0.
-        Use custom client and server messages instead.
     """
 
     name: str
@@ -110,10 +104,6 @@ class RTVIService(BaseModel):
 
     Represents a service that can be configured and used within the RTVI protocol,
     containing a name and list of configurable options.
-
-    .. deprecated:: 0.0.75
-       Pipeline Configuration has been removed as part of the RTVI protocol 1.0.0.
-       Use custom client and server messages instead.
     """
 
     name: str
@@ -132,10 +122,6 @@ class RTVIActionArgumentData(BaseModel):
     """Data for an RTVI action argument.
 
     Contains the name and value of an argument passed to an RTVI action.
-
-    .. deprecated:: 0.0.75
-        Actions have been removed as part of the RTVI protocol 1.0.0.
-        Use custom client and server messages instead.
     """
 
     name: str
@@ -146,10 +132,6 @@ class RTVIActionArgument(BaseModel):
     """Definition of an RTVI action argument.
 
     Specifies the name and expected type of an argument for an RTVI action.
-
-    .. deprecated:: 0.0.75
-        Actions have been removed as part of the RTVI protocol 1.0.0.
-        Use custom client and server messages instead.
     """
 
     name: str
@@ -161,10 +143,6 @@ class RTVIAction(BaseModel):
 
     Represents an action that can be executed within the RTVI protocol,
     including its service, name, arguments, and handler function.
-
-    .. deprecated:: 0.0.75
-       Actions have been removed as part of the RTVI protocol 1.0.0.
-       Use custom client and server messages instead.
     """
 
     service: str
@@ -188,10 +166,6 @@ class RTVIServiceOptionConfig(BaseModel):
     """Configuration value for an RTVI service option.
 
     Contains the name and value to set for a specific service option.
-
-    .. deprecated:: 0.0.75
-        Pipeline Configuration has been removed as part of the RTVI protocol 1.0.0.
-        Use custom client and server messages instead.
     """
 
     name: str
@@ -202,10 +176,6 @@ class RTVIServiceConfig(BaseModel):
     """Configuration for an RTVI service.
 
     Contains the service name and list of option configurations to apply.
-
-    .. deprecated:: 0.0.75
-       Pipeline Configuration has been removed as part of the RTVI protocol 1.0.0.
-       Use custom client and server messages instead.
     """
 
     service: str
@@ -216,10 +186,6 @@ class RTVIConfig(BaseModel):
     """Complete RTVI configuration.
 
     Contains the full configuration for all RTVI services.
-
-    .. deprecated:: 0.0.75
-       Pipeline Configuration has been removed as part of the RTVI protocol 1.0.0.
-       Use custom client and server messages instead.
     """
 
     config: List[RTVIServiceConfig]
@@ -230,15 +196,10 @@ class RTVIConfig(BaseModel):
 #
 
 
-# deprecated
 class RTVIUpdateConfig(BaseModel):
     """Request to update RTVI configuration.
 
     Contains new configuration settings and whether to interrupt the bot.
-
-    .. deprecated:: 0.0.75
-        Pipeline Configuration has been removed as part of the RTVI protocol 1.0.0.
-        Use custom client and server messages instead.
     """
 
     config: List[RTVIServiceConfig]
@@ -249,10 +210,6 @@ class RTVIActionRunArgument(BaseModel):
     """Argument for running an RTVI action.
 
     Contains the name and value of an argument to pass to an action.
-
-    .. deprecated:: 0.0.75
-        Actions have been removed as part of the RTVI protocol 1.0.0.
-        Use custom client and server messages instead.
     """
 
     name: str
@@ -263,10 +220,6 @@ class RTVIActionRun(BaseModel):
     """Request to run an RTVI action.
 
     Contains the service, action name, and optional arguments.
-
-    .. deprecated:: 0.0.75
-       Actions have been removed as part of the RTVI protocol 1.0.0.
-       Use custom client and server messages instead.
     """
 
     service: str
@@ -281,80 +234,12 @@ class RTVIActionFrame(DataFrame):
     Parameters:
         rtvi_action_run: The action to execute.
         message_id: Optional message ID for response correlation.
-
-    .. deprecated:: 0.0.75
-        Actions have been removed as part of the RTVI protocol 1.0.0.
-        Use custom client and server messages instead.
     """
 
     rtvi_action_run: RTVIActionRun
     message_id: Optional[str] = None
 
 
-class RTVIRawClientMessageData(BaseModel):
-    """Data structure expected from client messages sent to the RTVI server."""
-
-    t: str
-    d: Optional[Any] = None
-
-
-class RTVIClientMessage(BaseModel):
-    """Cleansed data structure for client messages for handling."""
-
-    msg_id: str
-    type: str
-    data: Optional[Any] = None
-
-
-@dataclass
-class RTVIClientMessageFrame(SystemFrame):
-    """A frame for sending messages from the client to the RTVI server.
-
-    This frame is meant for custom messaging from the client to the server
-    and expects a server-response message.
-    """
-
-    msg_id: str
-    type: str
-    data: Optional[Any] = None
-
-
-@dataclass
-class RTVIServerResponseFrame(SystemFrame):
-    """A frame for responding to a client RTVI message.
-
-    This frame should be sent in response to an RTVIClientMessageFrame
-    and include the original RTVIClientMessageFrame to ensure the response
-    is properly attributed to the original request. To respond with an error,
-    set the `error` field to a string describing the error. This will result
-    in the client receiving a `response-error` message instead of a
-    `server-response` message.
-    """
-
-    client_msg: RTVIClientMessageFrame
-    data: Optional[Any] = None
-    error: Optional[str] = None
-
-
-class RTVIRawServerResponseData(BaseModel):
-    """Data structure for server responses to client messages."""
-
-    t: str
-    d: Optional[Any] = None
-
-
-class RTVIServerResponse(BaseModel):
-    """The RTVI-formatted message response from the server to the client.
-
-    This message is used to respond to custom messages sent by the client.
-    """
-
-    label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
-    type: Literal["server-response"] = "server-response"
-    id: str
-    data: RTVIRawServerResponseData
-
-
 class RTVIMessage(BaseModel):
     """Base RTVI message structure.
 
@@ -384,7 +269,7 @@ class RTVIErrorResponseData(BaseModel):
 class RTVIErrorResponse(BaseModel):
     """RTVI error response message.
 
-    RTVI Formatted error response message for relaying failed client requests.
+    Sent in response to a client request that resulted in an error.
     """
 
     label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
@@ -400,13 +285,13 @@ class RTVIErrorData(BaseModel):
     """
 
     error: str
-    fatal: bool  # Indicates the pipeline has stopped due to this error
+    fatal: bool
 
 
 class RTVIError(BaseModel):
     """RTVI error event message.
 
-    RTVI Formatted error message for relaying errors in the pipeline.
+    Sent when an error occurs that isn't in response to a specific request.
     """
 
     label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
@@ -418,10 +303,6 @@ class RTVIDescribeConfigData(BaseModel):
     """Data for describing available RTVI configuration.
 
     Contains the list of available services and their options.
-
-    .. deprecated:: 0.0.75
-        Pipeline Configuration has been removed as part of the RTVI protocol 1.0.0.
-        Use custom client and server messages instead.
     """
 
     config: List[RTVIService]
@@ -431,10 +312,6 @@ class RTVIDescribeConfig(BaseModel):
     """Message describing available RTVI configuration.
 
     Sent in response to a describe-config request.
-
-    .. deprecated:: 0.0.75
-        Pipeline Configuration has been removed as part of the RTVI protocol 1.0.0.
-        Use custom client and server messages instead.
     """
 
     label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
@@ -447,10 +324,6 @@ class RTVIDescribeActionsData(BaseModel):
     """Data for describing available RTVI actions.
 
     Contains the list of available actions that can be executed.
-
-    .. deprecated:: 0.0.75
-        Actions have been removed as part of the RTVI protocol 1.0.0.
-        Use custom client and server messages instead.
     """
 
     actions: List[RTVIAction]
@@ -460,10 +333,6 @@ class RTVIDescribeActions(BaseModel):
     """Message describing available RTVI actions.
 
     Sent in response to a describe-actions request.
-
-    .. deprecated:: 0.0.75
-        Actions have been removed as part of the RTVI protocol 1.0.0.
-        Use custom client and server messages instead.
     """
 
     label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
@@ -476,10 +345,6 @@ class RTVIConfigResponse(BaseModel):
     """Response containing current RTVI configuration.
 
     Sent in response to a get-config request.
-
-    .. deprecated:: 0.0.75
-        Pipeline Configuration has been removed as part of the RTVI protocol 1.0.0.
-        Use custom client and server messages instead.
     """
 
     label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
@@ -492,10 +357,6 @@ class RTVIActionResponseData(BaseModel):
     """Data for an RTVI action response.
 
     Contains the result of executing an action.
-
-    .. deprecated:: 0.0.75
-        Actions have been removed as part of the RTVI protocol 1.0.0.
-        Use custom client and server messages instead.
     """
 
     result: ActionResult
@@ -505,10 +366,6 @@ class RTVIActionResponse(BaseModel):
     """Response to an RTVI action execution.
 
     Sent after successfully executing an action.
-
-    .. deprecated:: 0.0.75
-        Actions have been removed as part of the RTVI protocol 1.0.0.
-        Use custom client and server messages instead.
     """
 
     label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
@@ -517,30 +374,6 @@ class RTVIActionResponse(BaseModel):
     data: RTVIActionResponseData
 
 
-class AboutClientData(BaseModel):
-    """Data about the RTVI client.
-
-    Contains information about the client, including which RTVI library it
-    is using, what platform it is on and any additional details, if available.
-    """
-
-    library: str
-    library_version: Optional[str] = None
-    platform: Optional[str] = None
-    platform_version: Optional[str] = None
-    platform_details: Optional[Any] = None
-
-
-class RTVIClientReadyData(BaseModel):
-    """Data format of client ready messages.
-
-    Contains the RTVIprotocol version and client information.
-    """
-
-    version: str
-    about: AboutClientData
-
-
 class RTVIBotReadyData(BaseModel):
     """Data for bot ready notification.
 
@@ -548,10 +381,7 @@ class RTVIBotReadyData(BaseModel):
     """
 
     version: str
-    # The config field is deprecated and will not be included if
-    # the client's rtvi version is 1.0.0 or higher.
-    config: Optional[List[RTVIServiceConfig]] = None
-    about: Optional[Mapping[str, Any]] = None
+    config: List[RTVIServiceConfig]
 
 
 class RTVIBotReady(BaseModel):
@@ -588,25 +418,6 @@ class RTVILLMFunctionCallMessage(BaseModel):
     data: RTVILLMFunctionCallMessageData
 
 
-class RTVIAppendToContextData(BaseModel):
-    """Data format for appending messages to the context.
-
-    Contains the role, content, and whether to run the message immediately.
-    """
-
-    role: Literal["user", "assistant"] | str
-    content: Any
-    run_immediately: bool = False
-
-
-class RTVIAppendToContext(BaseModel):
-    """RTVI Message format to append content to the LLM context."""
-
-    label: RTVIMessageLiteral = RTVI_MESSAGE_LABEL
-    type: Literal["append-to-context"] = "append-to-context"
-    data: RTVIAppendToContextData
-
-
 class RTVILLMFunctionCallStartMessageData(BaseModel):
     """Data for LLM function call start notification.
 
@@ -941,11 +752,6 @@ class RTVIObserver(BaseObserver):
         elif isinstance(frame, RTVIServerMessageFrame):
             message = RTVIServerMessage(data=frame.data)
             await self.push_transport_message_urgent(message)
-        elif isinstance(frame, RTVIServerResponseFrame):
-            if frame.error is not None:
-                await self._send_error_response(frame)
-            else:
-                await self._send_server_response(frame)
 
         if mark_as_seen:
             self._frames_seen.add(frame.id)
@@ -1073,22 +879,6 @@ class RTVIObserver(BaseObserver):
         message = RTVIMetricsMessage(data=metrics)
         await self.push_transport_message_urgent(message)
 
-    async def _send_server_response(self, frame: RTVIServerResponseFrame):
-        """Send a response to the client for a specific request."""
-        message = RTVIServerResponse(
-            id=str(frame.client_msg.msg_id),
-            data=RTVIRawServerResponseData(t=frame.client_msg.type, d=frame.data),
-        )
-        await self.push_transport_message_urgent(message)
-
-    async def _send_error_response(self, frame: RTVIServerResponseFrame):
-        """Send a response to the client for a specific request."""
-        if self._params.errors_enabled:
-            message = RTVIErrorResponse(
-                id=str(frame.client_msg.msg_id), data=RTVIErrorResponseData(error=frame.error)
-            )
-            await self.push_transport_message_urgent(message)
-
 
 class RTVIProcessor(FrameProcessor):
     """Main processor for handling RTVI protocol messages and actions.
@@ -1118,7 +908,6 @@ class RTVIProcessor(FrameProcessor):
         self._bot_ready = False
         self._client_ready = False
         self._client_ready_id = ""
-        self._client_version = []
         self._errors_enabled = True
 
         self._registered_actions: Dict[str, RTVIAction] = {}
@@ -1132,7 +921,6 @@ class RTVIProcessor(FrameProcessor):
 
         self._register_event_handler("on_bot_started")
         self._register_event_handler("on_client_ready")
-        self._register_event_handler("on_client_message")
 
         self._input_transport = None
         self._transport = transport
@@ -1148,15 +936,6 @@ class RTVIProcessor(FrameProcessor):
         Args:
             action: The action to register.
         """
-        import warnings
-
-        with warnings.catch_warnings():
-            warnings.simplefilter("always")
-            warnings.warn(
-                "The actions API is deprecated, use server and client messages instead.",
-                DeprecationWarning,
-            )
-
         id = self._action_id(action.service, action.action)
         self._registered_actions[id] = action
 
@@ -1166,15 +945,6 @@ class RTVIProcessor(FrameProcessor):
         Args:
             service: The service to register.
         """
-        import warnings
-
-        with warnings.catch_warnings():
-            warnings.simplefilter("always")
-            warnings.warn(
-                "The actions API is deprecated, use server and client messages instead.",
-                DeprecationWarning,
-            )
-
         self._registered_services[service.name] = service
 
     async def set_client_ready(self):
@@ -1200,22 +970,6 @@ class RTVIProcessor(FrameProcessor):
         """Send a bot interruption frame upstream."""
         await self.push_frame(BotInterruptionFrame(), FrameDirection.UPSTREAM)
 
-    async def send_server_message(self, data: Any):
-        """Send a server message to the client."""
-        message = RTVIServerMessage(data=data)
-        await self._send_server_message(message)
-
-    async def send_server_response(self, client_msg: RTVIClientMessage, data: Any):
-        """Send a server response for a given client message."""
-        message = RTVIServerResponse(
-            id=client_msg.msg_id, data=RTVIRawServerResponseData(t=client_msg.type, d=data)
-        )
-        await self._send_server_message(message)
-
-    async def send_error_response(self, client_msg: RTVIClientMessage, error: str):
-        """Send an error response for a given client message."""
-        await self._send_error_response(id=client_msg.msg_id, error=error)
-
     async def send_error(self, error: str):
         """Send an error message to the client.
 
@@ -1259,6 +1013,9 @@ class RTVIProcessor(FrameProcessor):
             function_name: Name of the function being called.
             llm: The LLM processor making the call.
             context: The LLM context.
+
+        Note:
+            This method is deprecated. Use handle_function_call() instead.
         """
         import warnings
 
@@ -1379,15 +1136,7 @@ class RTVIProcessor(FrameProcessor):
         try:
             match message.type:
                 case "client-ready":
-                    data = None
-                    try:
-                        data = RTVIClientReadyData.model_validate(message.data)
-                    except ValidationError:
-                        # Not all clients have been updated to RTVI 1.0.0.
-                        # For now, that's okay, we just log their info as unknown.
-                        data = None
-                        pass
-                    await self._handle_client_ready(message.id, data)
+                    await self._handle_client_ready(message.id)
                 case "describe-actions":
                     await self._handle_describe_actions(message.id)
                 case "describe-config":
@@ -1399,9 +1148,6 @@ class RTVIProcessor(FrameProcessor):
                     await self._handle_update_config(message.id, update_config)
                 case "disconnect-bot":
                     await self.push_frame(EndTaskFrame(), FrameDirection.UPSTREAM)
-                case "client-message":
-                    data = RTVIRawClientMessageData.model_validate(message.data)
-                    await self._handle_client_message(message.id, data)
                 case "action":
                     action = RTVIActionRun.model_validate(message.data)
                     action_frame = RTVIActionFrame(message_id=message.id, rtvi_action_run=action)
@@ -1409,9 +1155,6 @@ class RTVIProcessor(FrameProcessor):
                 case "llm-function-call-result":
                     data = RTVILLMFunctionCallResultData.model_validate(message.data)
                     await self._handle_function_call_result(data)
-                case "append-to-context":
-                    data = RTVIAppendToContextData.model_validate(message.data)
-                    await self._handle_update_context(data)
                 case "raw-audio" | "raw-audio-batch":
                     await self._handle_audio_buffer(message.data)
 
@@ -1425,20 +1168,9 @@ class RTVIProcessor(FrameProcessor):
             await self._send_error_response(message.id, f"Exception processing message: {e}")
             logger.warning(f"Exception processing message: {e}")
 
-    async def _handle_client_ready(self, request_id: str, data: RTVIClientReadyData | None):
-        """Handle the client-ready message from the client."""
-        version = data.version if data else "unknown"
-        logger.debug(f"Received client-ready: version {version}")
-        if version == "unknown":
-            self._client_version = [0, 3, 0]  # Default to 0.3.0 if unknown
-        else:
-            try:
-                self._client_version = [int(v) for v in version.split(".")]
-            except ValueError:
-                logger.warning(f"Invalid client version format: {version}")
-                self._client_version = [0, 3, 0]
-        about = data.about if data else {"library": "unknown"}
-        logger.debug(f"Client Details: {about}")
+    async def _handle_client_ready(self, request_id: str):
+        """Handle a client-ready message."""
+        logger.debug("Received client-ready")
         if self._input_transport:
             await self._input_transport.start_audio_in_streaming()
 
@@ -1469,45 +1201,18 @@ class RTVIProcessor(FrameProcessor):
 
     async def _handle_describe_config(self, request_id: str):
         """Handle a describe-config request."""
-        import warnings
-
-        with warnings.catch_warnings():
-            warnings.simplefilter("always")
-            warnings.warn(
-                "Configuration helpers are deprecated. If your application needs this behavior, use custom server and client messages.",
-                DeprecationWarning,
-            )
-
         services = list(self._registered_services.values())
         message = RTVIDescribeConfig(id=request_id, data=RTVIDescribeConfigData(config=services))
         await self._push_transport_message(message)
 
     async def _handle_describe_actions(self, request_id: str):
         """Handle a describe-actions request."""
-        import warnings
-
-        with warnings.catch_warnings():
-            warnings.simplefilter("always")
-            warnings.warn(
-                "The Actions API is deprecated, use custom server and client messages instead.",
-                DeprecationWarning,
-            )
-
         actions = list(self._registered_actions.values())
         message = RTVIDescribeActions(id=request_id, data=RTVIDescribeActionsData(actions=actions))
         await self._push_transport_message(message)
 
     async def _handle_get_config(self, request_id: str):
         """Handle a get-config request."""
-        import warnings
-
-        with warnings.catch_warnings():
-            warnings.simplefilter("always")
-            warnings.warn(
-                "Configuration helpers are deprecated. If your application needs this behavior, use custom server and client messages.",
-                DeprecationWarning,
-            )
-
         message = RTVIConfigResponse(id=request_id, data=self._config)
         await self._push_transport_message(message)
 
@@ -1525,15 +1230,6 @@ class RTVIProcessor(FrameProcessor):
 
     async def _update_service_config(self, config: RTVIServiceConfig):
         """Update configuration for a specific service."""
-        import warnings
-
-        with warnings.catch_warnings():
-            warnings.simplefilter("always")
-            warnings.warn(
-                "Configuration helpers are deprecated. If your application needs this behavior, use custom server and client messages.",
-                DeprecationWarning,
-            )
-
         service = self._registered_services[config.service]
         for option in config.options:
             handler = service._options_dict[option.name].handler
@@ -1542,15 +1238,6 @@ class RTVIProcessor(FrameProcessor):
 
     async def _update_config(self, data: RTVIConfig, interrupt: bool):
         """Update the RTVI configuration."""
-        import warnings
-
-        with warnings.catch_warnings():
-            warnings.simplefilter("always")
-            warnings.warn(
-                "Configuration helpers are deprecated. If your application needs this behavior, use custom server and client messages.",
-                DeprecationWarning,
-            )
-
         if interrupt:
             await self.interrupt_bot()
         for service_config in data.config:
@@ -1561,33 +1248,6 @@ class RTVIProcessor(FrameProcessor):
         await self._update_config(RTVIConfig(config=data.config), data.interrupt)
         await self._handle_get_config(request_id)
 
-    async def _handle_update_context(self, data: RTVIAppendToContextData):
-        if data.run_immediately:
-            await self.interrupt_bot()
-        frame = LLMMessagesAppendFrame(
-            messages=[{"role": data.role, "content": data.content}],
-            run_llm=data.run_immediately,
-        )
-        await self.push_frame(frame)
-
-    async def _handle_client_message(self, msg_id: str, data: RTVIRawClientMessageData):
-        """Handle a client message frame."""
-        if not data:
-            await self._send_error_response(msg_id, "Malformed client message")
-            return
-
-        # Create a RTVIClientMessageFrame to push the message
-        frame = RTVIClientMessageFrame(msg_id=msg_id, type=data.t, data=data.d)
-        await self.push_frame(frame)
-        await self._call_event_handler(
-            "on_client_message",
-            RTVIClientMessage(
-                msg_id=msg_id,
-                type=data.t,
-                data=data.d,
-            ),
-        )
-
     async def _handle_function_call_result(self, data):
         """Handle a function call result from the client."""
         frame = FunctionCallResultFrame(
@@ -1618,19 +1278,12 @@ class RTVIProcessor(FrameProcessor):
 
     async def _send_bot_ready(self):
         """Send the bot-ready message to the client."""
-        config = None
-        if self._client_version[0] < 1:
-            config = self._config.config
         message = RTVIBotReady(
             id=self._client_ready_id,
-            data=RTVIBotReadyData(version=RTVI_PROTOCOL_VERSION, config=config),
+            data=RTVIBotReadyData(version=RTVI_PROTOCOL_VERSION, config=self._config.config),
         )
         await self._push_transport_message(message)
 
-    async def _send_server_message(self, message: RTVIServerMessage | RTVIServerResponse):
-        """Send a message or response to the client."""
-        await self._push_transport_message(message)
-
     async def _send_error_frame(self, frame: ErrorFrame):
         """Send an error frame as an RTVI error message."""
         if self._errors_enabled:

src/pipecat/serializers/exotel.py

@@ -108,10 +108,6 @@ class ExotelFrameSerializer(FrameSerializer):
             serialized_data = await self._output_resampler.resample(
                 data, frame.sample_rate, self._exotel_sample_rate
             )
-            if serialized_data is None or len(serialized_data) == 0:
-                # Ignoring in case we don't have audio
-                return None
-
             payload = base64.b64encode(serialized_data).decode("ascii")
 
             answer = {
@@ -148,9 +144,6 @@ class ExotelFrameSerializer(FrameSerializer):
                 self._exotel_sample_rate,
                 self._sample_rate,
             )
-            if deserialized_data is None or len(deserialized_data) == 0:
-                # Ignoring in case we don't have audio
-                return None
 
             # Input: Exotel takes PCM data, so just resample to match sample_rate
             audio_frame = InputAudioRawFrame(

src/pipecat/serializers/plivo.py

@@ -132,10 +132,6 @@ class PlivoFrameSerializer(FrameSerializer):
             serialized_data = await pcm_to_ulaw(
                 data, frame.sample_rate, self._plivo_sample_rate, self._output_resampler
             )
-            if serialized_data is None or len(serialized_data) == 0:
-                # Ignoring in case we don't have audio
-                return None
-
             payload = base64.b64encode(serialized_data).decode("utf-8")
             answer = {
                 "event": "playAudio",
@@ -231,10 +227,6 @@ class PlivoFrameSerializer(FrameSerializer):
             deserialized_data = await ulaw_to_pcm(
                 payload, self._plivo_sample_rate, self._sample_rate, self._input_resampler
             )
-            if deserialized_data is None or len(deserialized_data) == 0:
-                # Ignoring in case we don't have audio
-                return None
-
             audio_frame = InputAudioRawFrame(
                 audio=deserialized_data, num_channels=1, sample_rate=self._sample_rate
             )

src/pipecat/serializers/telnyx.py

@@ -155,10 +155,6 @@ class TelnyxFrameSerializer(FrameSerializer):
             else:
                 raise ValueError(f"Unsupported encoding: {self._params.inbound_encoding}")
 
-            if serialized_data is None or len(serialized_data) == 0:
-                # Ignoring in case we don't have audio
-                return None
-
             payload = base64.b64encode(serialized_data).decode("utf-8")
             answer = {
                 "event": "media",
@@ -266,10 +262,6 @@ class TelnyxFrameSerializer(FrameSerializer):
             else:
                 raise ValueError(f"Unsupported encoding: {self._params.outbound_encoding}")
 
-            if deserialized_data is None or len(deserialized_data) == 0:
-                # Ignoring in case we don't have audio
-                return None
-
             audio_frame = InputAudioRawFrame(
                 audio=deserialized_data, num_channels=1, sample_rate=self._sample_rate
             )

src/pipecat/serializers/twilio.py

@@ -132,10 +132,6 @@ class TwilioFrameSerializer(FrameSerializer):
             serialized_data = await pcm_to_ulaw(
                 data, frame.sample_rate, self._twilio_sample_rate, self._output_resampler
             )
-            if serialized_data is None or len(serialized_data) == 0:
-                # Ignoring in case we don't have audio
-                return None
-
             payload = base64.b64encode(serialized_data).decode("utf-8")
             answer = {
                 "event": "media",
@@ -239,10 +235,6 @@ class TwilioFrameSerializer(FrameSerializer):
             deserialized_data = await ulaw_to_pcm(
                 payload, self._twilio_sample_rate, self._sample_rate, self._input_resampler
             )
-            if deserialized_data is None or len(deserialized_data) == 0:
-                # Ignoring in case we don't have audio
-                return None
-
             audio_frame = InputAudioRawFrame(
                 audio=deserialized_data, num_channels=1, sample_rate=self._sample_rate
             )

src/pipecat/services/aws/llm.py

@@ -55,7 +55,7 @@ from pipecat.services.llm_service import LLMService
 from pipecat.utils.tracing.service_decorators import traced_llm
 
 try:
-    import aioboto3
+    import boto3
     import httpx
     from botocore.config import Config
 except ModuleNotFoundError as e:
@@ -749,17 +749,13 @@ class AWSBedrockLLMService(LLMService):
                 read_timeout=300,  # 5 minutes
                 retries={"max_attempts": 3},
             )
-
-        self._aws_session = aioboto3.Session()
-
-        # Store AWS session parameters for creating client in async context
-        self._aws_params = {
-            "aws_access_key_id": aws_access_key,
-            "aws_secret_access_key": aws_secret_key,
-            "aws_session_token": aws_session_token,
-            "region_name": aws_region,
-            "config": client_config,
-        }
+        session = boto3.Session(
+            aws_access_key_id=aws_access_key,
+            aws_secret_access_key=aws_secret_key,
+            aws_session_token=aws_session_token,
+            region_name=aws_region,
+        )
+        self._client = session.client(service_name="bedrock-runtime", config=client_config)
 
         self.set_model_name(model)
         self._settings = {
@@ -907,74 +903,70 @@ class AWSBedrockLLMService(LLMService):
 
             logger.debug(f"Calling AWS Bedrock model with: {request_params}")
 
-            async with self._aws_session.client(
-                service_name="bedrock-runtime", **self._aws_params
-            ) as client:
-                # Call AWS Bedrock with streaming
-                response = await client.converse_stream(**request_params)
+            # Call AWS Bedrock with streaming
+            response = self._client.converse_stream(**request_params)
 
-                await self.stop_ttfb_metrics()
+            await self.stop_ttfb_metrics()
 
-                # Process the streaming response
-                tool_use_block = None
-                json_accumulator = ""
+            # Process the streaming response
+            tool_use_block = None
+            json_accumulator = ""
 
-                function_calls = []
+            function_calls = []
+            for event in response["stream"]:
+                self.reset_watchdog()
 
-                async for event in response["stream"]:
-                    self.reset_watchdog()
+                # Handle text content
+                if "contentBlockDelta" in event:
+                    delta = event["contentBlockDelta"]["delta"]
+                    if "text" in delta:
+                        await self.push_frame(LLMTextFrame(delta["text"]))
+                        completion_tokens_estimate += self._estimate_tokens(delta["text"])
+                    elif "toolUse" in delta and "input" in delta["toolUse"]:
+                        # Handle partial JSON for tool use
+                        json_accumulator += delta["toolUse"]["input"]
+                        completion_tokens_estimate += self._estimate_tokens(
+                            delta["toolUse"]["input"]
+                        )
 
-                    # Handle text content
-                    if "contentBlockDelta" in event:
-                        delta = event["contentBlockDelta"]["delta"]
-                        if "text" in delta:
-                            await self.push_frame(LLMTextFrame(delta["text"]))
-                            completion_tokens_estimate += self._estimate_tokens(delta["text"])
-                        elif "toolUse" in delta and "input" in delta["toolUse"]:
-                            # Handle partial JSON for tool use
-                            json_accumulator += delta["toolUse"]["input"]
-                            completion_tokens_estimate += self._estimate_tokens(
-                                delta["toolUse"]["input"]
-                            )
+                # Handle tool use start
+                elif "contentBlockStart" in event:
+                    content_block_start = event["contentBlockStart"]["start"]
+                    if "toolUse" in content_block_start:
+                        tool_use_block = {
+                            "id": content_block_start["toolUse"].get("toolUseId", ""),
+                            "name": content_block_start["toolUse"].get("name", ""),
+                        }
+                        json_accumulator = ""
 
-                    # Handle tool use start
-                    elif "contentBlockStart" in event:
-                        content_block_start = event["contentBlockStart"]["start"]
-                        if "toolUse" in content_block_start:
-                            tool_use_block = {
-                                "id": content_block_start["toolUse"].get("toolUseId", ""),
-                                "name": content_block_start["toolUse"].get("name", ""),
-                            }
-                            json_accumulator = ""
+                # Handle message completion with tool use
+                elif "messageStop" in event and "stopReason" in event["messageStop"]:
+                    if event["messageStop"]["stopReason"] == "tool_use" and tool_use_block:
+                        try:
+                            arguments = json.loads(json_accumulator) if json_accumulator else {}
 
-                    # Handle message completion with tool use
-                    elif "messageStop" in event and "stopReason" in event["messageStop"]:
-                        if event["messageStop"]["stopReason"] == "tool_use" and tool_use_block:
-                            try:
-                                arguments = json.loads(json_accumulator) if json_accumulator else {}
-
-                                # Only call function if it's not the no_operation tool
-                                if not using_noop_tool:
-                                    function_calls.append(
-                                        FunctionCallFromLLM(
-                                            context=context,
-                                            tool_call_id=tool_use_block["id"],
-                                            function_name=tool_use_block["name"],
-                                            arguments=arguments,
-                                        )
+                            # Only call function if it's not the no_operation tool
+                            if not using_noop_tool:
+                                function_calls.append(
+                                    FunctionCallFromLLM(
+                                        context=context,
+                                        tool_call_id=tool_use_block["id"],
+                                        function_name=tool_use_block["name"],
+                                        arguments=arguments,
                                     )
-                                else:
-                                    logger.debug("Ignoring no_operation tool call")
-                            except json.JSONDecodeError:
-                                logger.error(f"Failed to parse tool arguments: {json_accumulator}")
+                                )
+                            else:
+                                logger.debug("Ignoring no_operation tool call")
+                        except json.JSONDecodeError:
+                            logger.error(f"Failed to parse tool arguments: {json_accumulator}")
 
-                    # Handle usage metrics if available
-                    if "metadata" in event and "usage" in event["metadata"]:
-                        usage = event["metadata"]["usage"]
-                        prompt_tokens += usage.get("inputTokens", 0)
-                        completion_tokens += usage.get("outputTokens", 0)
-                        cache_read_input_tokens += usage.get("cacheReadInputTokens", 0)
-                        cache_creation_input_tokens += usage.get("cacheWriteInputTokens", 0)
+                # Handle usage metrics if available
+                if "metadata" in event and "usage" in event["metadata"]:
+                    usage = event["metadata"]["usage"]
+                    prompt_tokens += usage.get("inputTokens", 0)
+                    completion_tokens += usage.get("outputTokens", 0)
+                    cache_read_input_tokens += usage.get("cacheReadInputTokens", 0)
+                    cache_creation_input_tokens += usage.get("cacheWriteInputTokens", 0)
 
             await self.run_function_calls(function_calls)
         except asyncio.CancelledError:

src/pipecat/services/aws/tts.py

@@ -30,7 +30,7 @@ from pipecat.transcriptions.language import Language
 from pipecat.utils.tracing.service_decorators import traced_tts
 
 try:
-    import aioboto3
+    import boto3
     from botocore.exceptions import BotoCoreError, ClientError
 except ModuleNotFoundError as e:
     logger.error(f"Exception: {e}")
@@ -177,25 +177,13 @@ class AWSPollyTTSService(TTSService):
 
         params = params or AWSPollyTTSService.InputParams()
 
-        # Get credentials from environment variables if not provided
-        self._aws_params = {
-            "aws_access_key_id": aws_access_key_id or os.getenv("AWS_ACCESS_KEY_ID"),
-            "aws_secret_access_key": api_key or os.getenv("AWS_SECRET_ACCESS_KEY"),
-            "aws_session_token": aws_session_token or os.getenv("AWS_SESSION_TOKEN"),
-            "region_name": region or os.getenv("AWS_REGION", "us-east-1"),
-        }
-
-        # Validate that we have the required credentials
-        if (
-            not self._aws_params["aws_access_key_id"]
-            or not self._aws_params["aws_secret_access_key"]
-        ):
-            raise ValueError(
-                "AWS credentials not found. Please provide them either through constructor parameters "
-                "or set AWS_ACCESS_KEY_ID and AWS_SECRET_ACCESS_KEY environment variables."
-            )
-
-        self._aws_session = aioboto3.Session()
+        self._polly_client = boto3.client(
+            "polly",
+            aws_access_key_id=aws_access_key_id,
+            aws_secret_access_key=api_key,
+            aws_session_token=aws_session_token,
+            region_name=region,
+        )
         self._settings = {
             "engine": params.engine,
             "language": self.language_to_service_language(params.language)
@@ -211,6 +199,24 @@ class AWSPollyTTSService(TTSService):
 
         self.set_voice(voice_id)
 
+        # Get credentials from environment variables if not provided
+        self._credentials = {
+            "aws_access_key_id": aws_access_key_id or os.getenv("AWS_ACCESS_KEY_ID"),
+            "aws_secret_access_key": api_key or os.getenv("AWS_SECRET_ACCESS_KEY"),
+            "aws_session_token": aws_session_token or os.getenv("AWS_SESSION_TOKEN"),
+            "region": region or os.getenv("AWS_REGION", "us-east-1"),
+        }
+
+        # Validate that we have the required credentials
+        if (
+            not self._credentials["aws_access_key_id"]
+            or not self._credentials["aws_secret_access_key"]
+        ):
+            raise ValueError(
+                "AWS credentials not found. Please provide them either through constructor parameters "
+                "or set AWS_ACCESS_KEY_ID and AWS_SECRET_ACCESS_KEY environment variables."
+            )
+
     def can_generate_metrics(self) -> bool:
         """Check if this service can generate processing metrics.
 
@@ -273,6 +279,14 @@ class AWSPollyTTSService(TTSService):
         Yields:
             Frame: Audio frames containing the synthesized speech.
         """
+
+        def read_audio_data(**args):
+            response = self._polly_client.synthesize_speech(**args)
+            if "AudioStream" in response:
+                audio_data = response["AudioStream"].read()
+                return audio_data
+            return None
+
         logger.debug(f"{self}: Generating TTS [{text}]")
 
         try:
@@ -295,32 +309,30 @@ class AWSPollyTTSService(TTSService):
             # Filter out None values
             filtered_params = {k: v for k, v in params.items() if v is not None}
 
-            async with self._aws_session.client("polly", **self._aws_params) as polly:
-                response = await polly.synthesize_speech(**filtered_params)
-                if "AudioStream" in response:
-                    # Get the streaming body and read it
-                    stream = response["AudioStream"]
-                    audio_data = await stream.read()
-                else:
-                    logger.error(f"{self} No audio stream in response")
-                    audio_data = None
+            audio_data = await asyncio.to_thread(read_audio_data, **filtered_params)
 
-                audio_data = await self._resampler.resample(audio_data, 16000, self.sample_rate)
+            if not audio_data:
+                logger.error(f"{self} No audio data returned")
+                yield None
+                return
 
-                await self.start_tts_usage_metrics(text)
+            audio_data = await self._resampler.resample(audio_data, 16000, self.sample_rate)
 
-                yield TTSStartedFrame()
+            await self.start_tts_usage_metrics(text)
 
-                CHUNK_SIZE = self.chunk_size
+            yield TTSStartedFrame()
 
-                for i in range(0, len(audio_data), CHUNK_SIZE):
-                    chunk = audio_data[i : i + CHUNK_SIZE]
-                    if len(chunk) > 0:
-                        await self.stop_ttfb_metrics()
-                        frame = TTSAudioRawFrame(chunk, self.sample_rate, 1)
-                        yield frame
+            CHUNK_SIZE = self.chunk_size
+
+            for i in range(0, len(audio_data), CHUNK_SIZE):
+                chunk = audio_data[i : i + CHUNK_SIZE]
+                if len(chunk) > 0:
+                    await self.stop_ttfb_metrics()
+                    frame = TTSAudioRawFrame(chunk, self.sample_rate, 1)
+                    yield frame
+
+            yield TTSStoppedFrame()
 
-                yield TTSStoppedFrame()
         except (BotoCoreError, ClientError) as error:
             logger.exception(f"{self} error generating TTS: {error}")
             error_message = f"AWS Polly TTS error: {str(error)}"

src/pipecat/services/aws_nova_sonic/aws.py

@@ -474,6 +474,7 @@ class AWSNovaSonicLLMService(LLMService):
         # If we need to, send assistant response trigger (depends on self._connected_time)
         if self._triggering_assistant_response:
             await self._send_assistant_response_trigger()
+            self._triggering_assistant_response = False
 
     async def _disconnect(self):
         try:
@@ -1104,6 +1105,7 @@ class AWSNovaSonicLLMService(LLMService):
         # Send the trigger audio, if we're fully connected and set up
         if self._connected_time is not None:
             await self._send_assistant_response_trigger()
+            self._triggering_assistant_response = False
 
     async def _send_assistant_response_trigger(self):
         if (
@@ -1111,51 +1113,46 @@ class AWSNovaSonicLLMService(LLMService):
         ):  # should never happen
             return
 
-        try:
-            logger.debug("Sending assistant response trigger...")
+        logger.debug("Sending assistant response trigger...")
 
-            chunk_duration = 0.02  # what we might get from InputAudioRawFrame
-            chunk_size = int(
-                chunk_duration
-                * self._params.input_sample_rate
-                * self._params.input_channel_count
-                * (self._params.input_sample_size / 8)
-            )  # e.g. 0.02 seconds of 16-bit (2-byte) PCM mono audio at 16kHz is 640 bytes
+        chunk_duration = 0.02  # what we might get from InputAudioRawFrame
+        chunk_size = int(
+            chunk_duration
+            * self._params.input_sample_rate
+            * self._params.input_channel_count
+            * (self._params.input_sample_size / 8)
+        )  # e.g. 0.02 seconds of 16-bit (2-byte) PCM mono audio at 16kHz is 640 bytes
 
-            # Lead with a bit of blank audio, if needed.
-            # It seems like the LLM can't quite "hear" the first little bit of audio sent on a
-            # connection.
-            current_time = time.time()
-            max_blank_audio_duration = 0.5
-            blank_audio_duration = (
-                max_blank_audio_duration - (current_time - self._connected_time)
-                if self._connected_time is not None
-                and (current_time - self._connected_time) < max_blank_audio_duration
-                else None
+        # Lead with a bit of blank audio, if needed.
+        # It seems like the LLM can't quite "hear" the first little bit of audio sent on a
+        # connection.
+        current_time = time.time()
+        max_blank_audio_duration = 0.5
+        blank_audio_duration = (
+            max_blank_audio_duration - (current_time - self._connected_time)
+            if self._connected_time is not None
+            and (current_time - self._connected_time) < max_blank_audio_duration
+            else None
+        )
+        if blank_audio_duration:
+            logger.debug(
+                f"Leading assistant response trigger with {blank_audio_duration}s of blank audio"
             )
-            if blank_audio_duration:
-                logger.debug(
-                    f"Leading assistant response trigger with {blank_audio_duration}s of blank audio"
-                )
-                blank_audio_chunk = b"\x00" * chunk_size
-                num_chunks = int(blank_audio_duration / chunk_duration)
-                for _ in range(num_chunks):
-                    await self._send_user_audio_event(blank_audio_chunk)
-                    await asyncio.sleep(chunk_duration)
-
-            # Send trigger audio
-            # NOTE: this audio *will* be transcribed and eventually make it into the context. That's OK:
-            # if we ever need to seed this service again with context it would make sense to include it
-            # since the instruction (i.e. the "wait for the trigger" instruction) will be part of the
-            # context as well.
-            audio_chunks = [
-                self._assistant_response_trigger_audio[i : i + chunk_size]
-                for i in range(0, len(self._assistant_response_trigger_audio), chunk_size)
-            ]
-            for chunk in audio_chunks:
-                await self._send_user_audio_event(chunk)
+            blank_audio_chunk = b"\x00" * chunk_size
+            num_chunks = int(blank_audio_duration / chunk_duration)
+            for _ in range(num_chunks):
+                await self._send_user_audio_event(blank_audio_chunk)
                 await asyncio.sleep(chunk_duration)
-        finally:
-            # We need to clean up in case sending the trigger was cancelled, e.g. in the case of a user interruption.
-            # (An asyncio.CancelledError would be raised in that case.)
-            self._triggering_assistant_response = False
+
+        # Send trigger audio
+        # NOTE: this audio *will* be transcribed and eventually make it into the context. That's OK:
+        # if we ever need to seed this service again with context it would make sense to include it
+        # since the instruction (i.e. the "wait for the trigger" instruction) will be part of the
+        # context as well.
+        audio_chunks = [
+            self._assistant_response_trigger_audio[i : i + chunk_size]
+            for i in range(0, len(self._assistant_response_trigger_audio), chunk_size)
+        ]
+        for chunk in audio_chunks:
+            await self._send_user_audio_event(chunk)
+            await asyncio.sleep(chunk_duration)

src/pipecat/services/cartesia/tts.py

@@ -121,7 +121,6 @@ class CartesiaTTSService(AudioContextWordTTSService):
         container: str = "raw",
         params: Optional[InputParams] = None,
         text_aggregator: Optional[BaseTextAggregator] = None,
-        aggregate_sentences: Optional[bool] = True,
         **kwargs,
     ):
         """Initialize the Cartesia TTS service.
@@ -137,7 +136,6 @@ class CartesiaTTSService(AudioContextWordTTSService):
             container: Audio container format.
             params: Additional input parameters for voice customization.
             text_aggregator: Custom text aggregator for processing input text.
-            aggregate_sentences: Whether to aggregate sentences within the TTSService.
             **kwargs: Additional arguments passed to the parent service.
         """
         # Aggregating sentences still gives cleaner-sounding results and fewer
@@ -151,7 +149,7 @@ class CartesiaTTSService(AudioContextWordTTSService):
         # can use those to generate text frames ourselves aligned with the
         # playout timing of the audio!
         super().__init__(
-            aggregate_sentences=aggregate_sentences,
+            aggregate_sentences=True,
             push_text_frames=False,
             pause_frame_processing=True,
             sample_rate=sample_rate,

src/pipecat/services/elevenlabs/tts.py

@@ -238,7 +238,6 @@ class ElevenLabsTTSService(AudioContextWordTTSService):
         url: str = "wss://api.elevenlabs.io",
         sample_rate: Optional[int] = None,
         params: Optional[InputParams] = None,
-        aggregate_sentences: Optional[bool] = True,
         **kwargs,
     ):
         """Initialize the ElevenLabs TTS service.
@@ -250,7 +249,6 @@ class ElevenLabsTTSService(AudioContextWordTTSService):
             url: WebSocket URL for ElevenLabs TTS API.
             sample_rate: Audio sample rate. If None, uses default.
             params: Additional input parameters for voice customization.
-            aggregate_sentences: Whether to aggregate sentences within the TTSService.
             **kwargs: Additional arguments passed to the parent service.
         """
         # Aggregating sentences still gives cleaner-sounding results and fewer
@@ -268,7 +266,7 @@ class ElevenLabsTTSService(AudioContextWordTTSService):
         # speaking for a while, so we want the parent class to send TTSStopFrame
         # after a short period not receiving any audio.
         super().__init__(
-            aggregate_sentences=aggregate_sentences,
+            aggregate_sentences=True,
             push_text_frames=False,
             push_stop_frames=True,
             pause_frame_processing=True,

src/pipecat/services/neuphonic/tts.py

@@ -106,11 +106,10 @@ class NeuphonicTTSService(InterruptibleTTSService):
         *,
         api_key: str,
         voice_id: Optional[str] = None,
-        url: str = "wss://eu-west-1.api.neuphonic.com",
+        url: str = "wss://api.neuphonic.com",
         sample_rate: Optional[int] = 22050,
         encoding: str = "pcm_linear",
         params: Optional[InputParams] = None,
-        aggregate_sentences: Optional[bool] = True,
         **kwargs,
     ):
         """Initialize the Neuphonic TTS service.
@@ -122,11 +121,10 @@ class NeuphonicTTSService(InterruptibleTTSService):
             sample_rate: Audio sample rate in Hz. Defaults to 22050.
             encoding: Audio encoding format. Defaults to "pcm_linear".
             params: Additional input parameters for TTS configuration.
-            aggregate_sentences: Whether to aggregate sentences within the TTSService.
             **kwargs: Additional arguments passed to parent InterruptibleTTSService.
         """
         super().__init__(
-            aggregate_sentences=aggregate_sentences,
+            aggregate_sentences=True,
             push_text_frames=False,
             push_stop_frames=True,
             stop_frame_timeout_s=2.0,
@@ -281,18 +279,14 @@ class NeuphonicTTSService(InterruptibleTTSService):
                 "voice_id": self._voice_id,
             }
 
-            query_params = []
+            query_params = [f"api_key={self._api_key}"]
             for key, value in tts_config.items():
                 if value is not None:
                     query_params.append(f"{key}={value}")
 
-            url = f"{self._url}/speak/{self._settings['lang_code']}"
-            if query_params:
-                url += f"?{'&'.join(query_params)}"
+            url = f"{self._url}/speak/{self._settings['lang_code']}?{'&'.join(query_params)}"
 
-            headers = {"x-api-key": self._api_key}
-
-            self._websocket = await websockets.connect(url, extra_headers=headers)
+            self._websocket = await websockets.connect(url)
         except Exception as e:
             logger.error(f"{self} initialization error: {e}")
             self._websocket = None
@@ -317,7 +311,7 @@ class NeuphonicTTSService(InterruptibleTTSService):
         async for message in WatchdogAsyncIterator(self._websocket, manager=self.task_manager):
             if isinstance(message, str):
                 msg = json.loads(message)
-                if msg.get("data") and msg["data"].get("audio"):
+                if msg.get("data", {}).get("audio") is not None:
                     await self.stop_ttfb_metrics()
 
                     audio = base64.b64decode(msg["data"]["audio"])
@@ -330,19 +324,12 @@ class NeuphonicTTSService(InterruptibleTTSService):
         while True:
             self.reset_watchdog()
             await asyncio.sleep(KEEPALIVE_SLEEP)
-            await self._send_keepalive()
-
-    async def _send_keepalive(self):
-        """Send keepalive message to maintain connection."""
-        if self._websocket:
-            # Send empty text for keepalive
-            msg = {"text": ""}
-            await self._websocket.send(json.dumps(msg))
+            await self._send_text("")
 
     async def _send_text(self, text: str):
         """Send text to Neuphonic WebSocket for synthesis."""
         if self._websocket:
-            msg = {"text": f"{text} <STOP>"}
+            msg = {"text": text}
             logger.debug(f"Sending text to websocket: {msg}")
             await self._websocket.send(json.dumps(msg))
 

src/pipecat/services/ollama/llm.py

@@ -6,8 +6,6 @@
 
 """OLLama LLM service implementation for Pipecat AI framework."""
 
-from loguru import logger
-
 from pipecat.services.openai.llm import OpenAILLMService
 
 
@@ -18,28 +16,12 @@ class OLLamaLLMService(OpenAILLMService):
     providing a compatible interface for running large language models locally.
     """
 
-    def __init__(
-        self, *, model: str = "llama2", base_url: str = "http://localhost:11434/v1", **kwargs
-    ):
+    def __init__(self, *, model: str = "llama2", base_url: str = "http://localhost:11434/v1"):
         """Initialize OLLama LLM service.
 
         Args:
             model: The OLLama model to use. Defaults to "llama2".
             base_url: The base URL for the OLLama API endpoint.
                     Defaults to "http://localhost:11434/v1".
-            **kwargs: Additional keyword arguments passed to OpenAILLMService.
         """
-        super().__init__(model=model, base_url=base_url, api_key="ollama", **kwargs)
-
-    def create_client(self, base_url=None, **kwargs):
-        """Create OpenAI-compatible client for Ollama.
-
-        Args:
-            base_url: The base URL for the API. If None, uses instance base_url.
-            **kwargs: Additional keyword arguments passed to the parent create_client method.
-
-        Returns:
-            An OpenAI-compatible client configured for Ollama.
-        """
-        logger.debug(f"Creating Ollama client with api {base_url}")
-        return super().create_client(base_url, **kwargs)
+        super().__init__(model=model, base_url=base_url, api_key="ollama")

src/pipecat/services/rime/tts.py

@@ -99,7 +99,6 @@ class RimeTTSService(AudioContextWordTTSService):
         sample_rate: Optional[int] = None,
         params: Optional[InputParams] = None,
         text_aggregator: Optional[BaseTextAggregator] = None,
-        aggregate_sentences: Optional[bool] = True,
         **kwargs,
     ):
         """Initialize Rime TTS service.
@@ -112,12 +111,11 @@ class RimeTTSService(AudioContextWordTTSService):
             sample_rate: Audio sample rate in Hz.
             params: Additional configuration parameters.
             text_aggregator: Custom text aggregator for processing input text.
-            aggregate_sentences: Whether to aggregate sentences within the TTSService.
             **kwargs: Additional arguments passed to parent class.
         """
         # Initialize with parent class settings for proper frame handling
         super().__init__(
-            aggregate_sentences=aggregate_sentences,
+            aggregate_sentences=True,
             push_text_frames=False,
             push_stop_frames=True,
             pause_frame_processing=True,

src/pipecat/services/riva/stt.py

@@ -279,6 +279,7 @@ class RivaSTTService(STTService):
             streaming_config=self._config,
         )
         for response in responses:
+            self.reset_watchdog()
             if not response.results:
                 continue
             asyncio.run_coroutine_threadsafe(

src/pipecat/services/stt_service.py

@@ -152,13 +152,6 @@ class STTService(AIService):
         else:
             self._user_id = ""
 
-        if not frame.audio:
-            # Ignoring in case we don't have audio to transcribe.
-            logger.warning(
-                f"Empty audio frame received for STT service: {self.name} {frame.num_frames}"
-            )
-            return
-
         await self.process_generator(self.run_stt(frame.audio))
 
     async def process_frame(self, frame: Frame, direction: FrameDirection):

src/pipecat/services/whisper/stt.py

@@ -49,10 +49,8 @@ class Model(Enum):
     Parameters:
         TINY: Smallest multilingual model, fastest inference.
         BASE: Basic multilingual model, good speed/quality balance.
-        SMALL: Small multilingual model, better speed/quality balance than BASE.
         MEDIUM: Medium-sized multilingual model, better quality.
         LARGE: Best quality multilingual model, slower inference.
-        LARGE_V3_TURBO: Fast multilingual model, slightly lower quality than LARGE.
         DISTIL_LARGE_V2: Fast multilingual distilled model.
         DISTIL_MEDIUM_EN: Fast English-only distilled model.
     """
@@ -60,10 +58,8 @@ class Model(Enum):
     # Multilingual models
     TINY = "tiny"
     BASE = "base"
-    SMALL = "small"
     MEDIUM = "medium"
     LARGE = "large-v3"
-    LARGE_V3_TURBO = "deepdml/faster-whisper-large-v3-turbo-ct2"
     DISTIL_LARGE_V2 = "Systran/faster-distil-whisper-large-v2"
 
     # English-only models

src/pipecat/transports/base_input.py

@@ -34,7 +34,6 @@ from pipecat.frames.frames import (
     InputAudioRawFrame,
     InputImageRawFrame,
     MetricsFrame,
-    SpeechControlParamsFrame,
     StartFrame,
     StartInterruptionFrame,
     StopFrame,
@@ -196,13 +195,6 @@ class BaseInputTransport(FrameProcessor):
         if self._params.turn_analyzer:
             self._params.turn_analyzer.set_sample_rate(self._sample_rate)
 
-        if self._params.vad_analyzer or self._params.turn_analyzer:
-            vad_params = self._params.vad_analyzer.params if self._params.vad_analyzer else None
-            turn_params = self._params.turn_analyzer.params if self._params.turn_analyzer else None
-
-            speech_frame = SpeechControlParamsFrame(vad_params=vad_params, turn_params=turn_params)
-            await self.push_frame(speech_frame)
-
         # Start audio filter.
         if self._params.audio_in_filter:
             await self._params.audio_in_filter.start(self._sample_rate)
@@ -318,13 +310,6 @@ class BaseInputTransport(FrameProcessor):
         elif isinstance(frame, VADParamsUpdateFrame):
             if self.vad_analyzer:
                 self.vad_analyzer.set_params(frame.params)
-                speech_frame = SpeechControlParamsFrame(
-                    vad_params=frame.params,
-                    turn_params=self._params.turn_analyzer.params
-                    if self._params.turn_analyzer
-                    else None,
-                )
-                await self.push_frame(speech_frame)
         elif isinstance(frame, FilterUpdateSettingsFrame) and self._params.audio_in_filter:
             await self._params.audio_in_filter.process_frame(frame)
         # Other frames

src/pipecat/transports/services/tavus.py

@@ -127,6 +127,35 @@ class TavusApi:
             logger.debug(f"Fetched Tavus persona: {response}")
             return response["persona_name"]
 
+    async def _validate_persona(self, persona_id: str):
+        """Validate that the persona's microphone is enabled.
+
+        Args:
+            persona_id: ID of the persona to validate.
+        """
+        if self._dev_room_url is not None:
+            return
+
+        url = f"{self.BASE_URL}/personas/{persona_id}"
+        async with self._session.get(url, headers=self._headers) as r:
+            r.raise_for_status()
+            response = await r.json()
+            logger.debug(f"Fetched Tavus persona: {response}")
+            try:
+                transport_settings = response.get("layers", {}).get("transport", {})
+                microphone_enabled = transport_settings.get("input_settings", {}).get(
+                    "microphone", ""
+                )
+                if microphone_enabled != "enabled":
+                    raise Exception(
+                        "Microphone is not enabled for this persona. Please update the persona or use the persona pipecat-stream."
+                    )
+            except Exception as e:
+                logger.error(f"Error validating persona {persona_id}: {e}")
+                raise e
+            logger.info(f"Persona {persona_id} is valid")
+            return True
+
 
 class TavusCallbacks(BaseModel):
     """Callback handlers for Tavus events.
@@ -200,6 +229,7 @@ class TavusTransportClient:
 
     async def _initialize(self) -> str:
         """Initialize the conversation and return the room URL."""
+        await self._api._validate_persona(self._persona_id)
         response = await self._api.create_conversation(self._replica_id, self._persona_id)
         self._conversation_id = response["conversation_id"]
         return response["conversation_url"]

tests/test_context_aggregators.py

@@ -8,8 +8,6 @@ import json
 import unittest
 from typing import Any
 
-from pipecat.audio.turn.smart_turn.base_smart_turn import SmartTurnParams
-from pipecat.audio.vad.vad_analyzer import VADParams
 from pipecat.frames.frames import (
     EmulateUserStartedSpeakingFrame,
     EmulateUserStoppedSpeakingFrame,
@@ -20,7 +18,6 @@ from pipecat.frames.frames import (
     LLMFullResponseEndFrame,
     LLMFullResponseStartFrame,
     OpenAILLMContextAssistantTimestampFrame,
-    SpeechControlParamsFrame,
     StartInterruptionFrame,
     TextFrame,
     TranscriptionFrame,
@@ -287,7 +284,6 @@ class BaseTestUserContextAggregator:
             context, params=LLMUserAggregatorParams(aggregation_timeout=AGGREGATION_TIMEOUT)
         )
         frames_to_send = [
-            SpeechControlParamsFrame(vad_params=VADParams(stop_secs=AGGREGATION_TIMEOUT)),
             UserStartedSpeakingFrame(),
             TranscriptionFrame(text="Hello Pipecat!", user_id="cat", timestamp=""),
             SleepFrame(),
@@ -296,7 +292,6 @@ class BaseTestUserContextAggregator:
             SleepFrame(sleep=AGGREGATION_SLEEP),
         ]
         expected_down_frames = [
-            SpeechControlParamsFrame,
             UserStartedSpeakingFrame,
             UserStoppedSpeakingFrame,
             *self.EXPECTED_CONTEXT_FRAMES,
@@ -373,51 +368,14 @@ class BaseTestUserContextAggregator:
 
         context = self.CONTEXT_CLASS()
         aggregator = self.AGGREGATOR_CLASS(
-            context
-        )  # No aggregation timeout; this tests VAD emulation
-
+            context, params=LLMUserAggregatorParams(aggregation_timeout=AGGREGATION_TIMEOUT)
+        )
         frames_to_send = [
-            SpeechControlParamsFrame(vad_params=VADParams(stop_secs=AGGREGATION_TIMEOUT)),
             TranscriptionFrame(text="Hello!", user_id="cat", timestamp=""),
             SleepFrame(sleep=AGGREGATION_SLEEP),
         ]
-        expected_down_frames = [
-            SpeechControlParamsFrame,
-            *self.EXPECTED_CONTEXT_FRAMES,
-        ]
+        expected_down_frames = [*self.EXPECTED_CONTEXT_FRAMES]
         expected_up_frames = [EmulateUserStartedSpeakingFrame, EmulateUserStoppedSpeakingFrame]
-
-        await run_test(
-            aggregator,
-            frames_to_send=frames_to_send,
-            expected_down_frames=expected_down_frames,
-            expected_up_frames=expected_up_frames,
-        )
-        self.check_message_content(context, 0, "Hello!")
-
-    async def test_t_with_turn_analyzer(self):
-        assert self.CONTEXT_CLASS is not None, "CONTEXT_CLASS must be set in a subclass"
-        assert self.AGGREGATOR_CLASS is not None, "AGGREGATOR_CLASS must be set in a subclass"
-
-        context = self.CONTEXT_CLASS()
-        aggregator = self.AGGREGATOR_CLASS(
-            context, params=LLMUserAggregatorParams(turn_emulated_vad_timeout=AGGREGATION_TIMEOUT)
-        )
-
-        frames_to_send = [
-            SpeechControlParamsFrame(
-                vad_params=VADParams(stop_secs=0.2),
-                turn_params=SmartTurnParams(stop_secs=3.0),  # Turn analyzer present
-            ),
-            TranscriptionFrame(text="Hello!", user_id="cat", timestamp=""),
-            SleepFrame(sleep=AGGREGATION_SLEEP),
-        ]
-        expected_down_frames = [
-            SpeechControlParamsFrame,
-            *self.EXPECTED_CONTEXT_FRAMES,
-        ]
-        expected_up_frames = [EmulateUserStartedSpeakingFrame, EmulateUserStoppedSpeakingFrame]
-
         await run_test(
             aggregator,
             frames_to_send=frames_to_send,
@@ -432,16 +390,15 @@ class BaseTestUserContextAggregator:
 
         context = self.CONTEXT_CLASS()
         aggregator = self.AGGREGATOR_CLASS(
-            context
-        )  # No aggregation timeout; this tests VAD emulation
+            context, params=LLMUserAggregatorParams(aggregation_timeout=AGGREGATION_TIMEOUT)
+        )
         frames_to_send = [
-            SpeechControlParamsFrame(vad_params=VADParams(stop_secs=AGGREGATION_TIMEOUT)),
             InterimTranscriptionFrame(text="Hello ", user_id="cat", timestamp=""),
             SleepFrame(),
             TranscriptionFrame(text="Hello Pipecat!", user_id="cat", timestamp=""),
             SleepFrame(sleep=AGGREGATION_SLEEP),
         ]
-        expected_down_frames = [SpeechControlParamsFrame, *self.EXPECTED_CONTEXT_FRAMES]
+        expected_down_frames = [*self.EXPECTED_CONTEXT_FRAMES]
         expected_up_frames = [EmulateUserStartedSpeakingFrame, EmulateUserStoppedSpeakingFrame]
         await run_test(
             aggregator,