✨ feat: Implement WebRTC messaging and audio handling in the WebRTC service

api/package.json

@@ -82,6 +82,7 @@
     "mongoose": "^8.9.5",
     "multer": "^1.4.5-lts.1",
     "nanoid": "^3.3.7",
+    "node-pre-gyp": "^0.17.0",
     "nodemailer": "^6.9.15",
     "ollama": "^0.5.0",
     "openai": "^4.47.1",
@@ -102,6 +103,7 @@
     "ua-parser-js": "^1.0.36",
     "winston": "^3.11.0",
     "winston-daily-rotate-file": "^4.7.1",
+    "wrtc": "^0.4.7",
     "ws": "^8.18.0",
     "youtube-transcript": "^1.2.1",
     "zod": "^3.22.4"

api/server/services/WebSocket/WebSocketServer.js

@@ -1,12 +1,15 @@
 const { WebSocketServer } = require('ws');
-const fs = require('fs');
+const { RTCPeerConnection } = require('wrtc');
-const path = require('path');
 
 module.exports.WebSocketService = class {
   constructor(server) {
     this.wss = new WebSocketServer({ server, path: '/ws' });
     this.log('Server initialized');
-    this.clientAudioBuffers = new Map();
+    this.activeClients = new Map();
+    this.iceServers = [
+      { urls: 'stun:stun.l.google.com:19302' },
+      { urls: 'stun:stun1.l.google.com:19302' },
+    ];
     this.setupHandlers();
   }
 
@@ -17,7 +20,13 @@ module.exports.WebSocketService = class {
   setupHandlers() {
     this.wss.on('connection', (ws) => {
       const clientId = Date.now().toString();
-      this.clientAudioBuffers.set(clientId, []);
+      this.activeClients.set(clientId, {
+        ws,
+        state: 'idle',
+        audioBuffer: [],
+        currentTranscription: '',
+        isProcessing: false,
+      });
 
       this.log(`Client connected: ${clientId}`);
 
@@ -29,42 +38,175 @@ module.exports.WebSocketService = class {
           return;
         }
 
-        if (message.type === 'audio-chunk') {
+        switch (message.type) {
-          if (!this.clientAudioBuffers.has(clientId)) {
+          case 'call-start':
-            this.clientAudioBuffers.set(clientId, []);
+            this.handleCallStart(clientId);
-          }
+            break;
-          this.clientAudioBuffers.get(clientId).push(message.data);
-        }
 
-        if (message.type === 'request-response') {
+          case 'audio-chunk':
-          const filePath = path.join(__dirname, './assets/response.mp3');
+            await this.handleAudioChunk(clientId, message.data);
-          const audioFile = fs.readFileSync(filePath);
+            break;
-          ws.send(JSON.stringify({ type: 'audio-response', data: audioFile.toString('base64') }));
-        }
 
-        if (message.type === 'call-ended') {
+          case 'processing-start':
-          const allChunks = this.clientAudioBuffers.get(clientId);
+            await this.processAudioStream(clientId);
-          this.writeAudioFile(clientId, allChunks);
+            break;
-          this.clientAudioBuffers.delete(clientId);
+
+          case 'audio-received':
+            this.confirmAudioReceived(clientId);
+            break;
+
+          case 'call-ended':
+            this.handleCallEnd(clientId);
+            break;
         }
       });
 
       ws.on('close', () => {
+        this.handleCallEnd(clientId);
+        this.activeClients.delete(clientId);
         this.log(`Client disconnected: ${clientId}`);
-        this.clientAudioBuffers.delete(clientId);
+      });
+
+      ws.on('error', (error) => {
+        this.log(`Error for client ${clientId}: ${error.message}`);
+        this.handleCallEnd(clientId);
       });
     });
   }
 
-  writeAudioFile(clientId, base64Chunks) {
+  async handleCallStart(clientId) {
-    if (!base64Chunks || base64Chunks.length === 0) {
+    const client = this.activeClients.get(clientId);
+    if (!client) {
       return;
     }
-    const filePath = path.join(__dirname, `recorded_${clientId}.webm`);
+
-    const buffer = Buffer.concat(
+    try {
-      base64Chunks.map((chunk) => Buffer.from(chunk.split(',')[1], 'base64')),
+      client.state = 'active';
+      client.audioBuffer = [];
+      client.currentTranscription = '';
+      client.isProcessing = false;
+
+      const peerConnection = new RTCPeerConnection({
+        iceServers: this.iceServers,
+        sdpSemantics: 'unified-plan',
+      });
+
+      client.peerConnection = peerConnection;
+      client.dataChannel = peerConnection.createDataChannel('audio', {
+        ordered: true,
+        maxRetransmits: 3,
+      });
+
+      client.dataChannel.onopen = () => this.log(`Data channel opened for ${clientId}`);
+      client.dataChannel.onmessage = async (event) => {
+        await this.handleAudioChunk(clientId, event.data);
+      };
+
+      peerConnection.onicecandidate = (event) => {
+        if (event.candidate) {
+          client.ws.send(
+            JSON.stringify({
+              type: 'ice-candidate',
+              candidate: event.candidate,
+            }),
+          );
+        }
+      };
+
+      peerConnection.onnegotiationneeded = async () => {
+        try {
+          const offer = await peerConnection.createOffer();
+          await peerConnection.setLocalDescription(offer);
+          client.ws.send(
+            JSON.stringify({
+              type: 'webrtc-offer',
+              sdp: peerConnection.localDescription,
+            }),
+          );
+        } catch (error) {
+          this.log(`Negotiation failed for ${clientId}: ${error}`);
+        }
+      };
+
+      this.log(`Call started for client ${clientId}`);
+    } catch (error) {
+      this.log(`Error starting call for ${clientId}: ${error.message}`);
+      this.handleCallEnd(clientId);
+    }
+  }
+
+  async handleAudioChunk(clientId, data) {
+    const client = this.activeClients.get(clientId);
+    if (!client || client.state !== 'active') {
+      return;
+    }
+
+    client.audioBuffer.push(data);
+    client.ws.send(JSON.stringify({ type: 'audio-received' }));
+  }
+
+  async processAudioStream(clientId) {
+    const client = this.activeClients.get(clientId);
+    if (!client || client.state !== 'active' || client.isProcessing) {
+      return;
+    }
+
+    client.isProcessing = true;
+
+    try {
+      // Process transcription
+      client.ws.send(
+        JSON.stringify({
+          type: 'transcription',
+          data: 'Processing audio...',
+        }),
+      );
+
+      // Stream LLM response
+      client.ws.send(
+        JSON.stringify({
+          type: 'llm-response',
+          data: 'Processing response...',
+        }),
+      );
+
+      // Stream TTS chunks
+      client.ws.send(
+        JSON.stringify({
+          type: 'tts-chunk',
+          data: 'audio_data_here',
+        }),
+      );
+    } catch (error) {
+      this.log(`Processing error for client ${clientId}: ${error.message}`);
+    } finally {
+      client.isProcessing = false;
+      client.audioBuffer = [];
+    }
+  }
+
+  confirmAudioReceived(clientId) {
+    const client = this.activeClients.get(clientId);
+    if (!client) {
+      return;
+    }
+
+    client.ws.send(
+      JSON.stringify({
+        type: 'audio-received',
+        data: null,
+      }),
     );
-    fs.writeFileSync(filePath, buffer);
+  }
-    this.log(`Saved audio to ${filePath}`);
+
+  handleCallEnd(clientId) {
+    const client = this.activeClients.get(clientId);
+    if (!client) {
+      return;
+    }
+
+    client.state = 'idle';
+    client.audioBuffer = [];
+    client.currentTranscription = '';
   }
 };

client/src/common/types.ts

@@ -48,6 +48,17 @@ export type AudioChunk = {
   };
 };
 
+export interface RTCMessage {
+  type:
+    | 'audio-chunk'
+    | 'audio-received'
+    | 'transcription'
+    | 'llm-response'
+    | 'tts-chunk'
+    | 'call-ended';
+  data?: string | ArrayBuffer | null;
+}
+
 export type AssistantListItem = {
   id: string;
   name: string;

client/src/hooks/useCall.ts

@@ -1,67 +1,106 @@
 import { useState, useRef, useCallback } from 'react';
-import useWebSocket from './useWebSocket';
 import { WebRTCService } from '../services/WebRTC/WebRTCService';
+import type { RTCMessage } from '~/common';
+import useWebSocket from './useWebSocket';
 
 const SILENCE_THRESHOLD = -50;
 const SILENCE_DURATION = 1000;
 
 const useCall = () => {
-  const { sendMessage } = useWebSocket();
+  const { sendMessage: wsMessage } = useWebSocket();
   const [isCalling, setIsCalling] = useState(false);
+  const [isProcessing, setIsProcessing] = useState(false);
   const audioContextRef = useRef<AudioContext | null>(null);
   const analyserRef = useRef<AnalyserNode | null>(null);
+  const audioChunksRef = useRef<Blob[]>([]);
   const silenceStartRef = useRef<number | null>(null);
   const intervalRef = useRef<number | null>(null);
   const webrtcServiceRef = useRef<WebRTCService | null>(null);
 
-  const checkSilence = useCallback(() => {
+  const sendAudioChunk = useCallback(() => {
-    if (!analyserRef.current || !isCalling) {
+    if (audioChunksRef.current.length === 0) {
       return;
     }
-    const data = new Float32Array(analyserRef.current.frequencyBinCount);
+
-    analyserRef.current.getFloatFrequencyData(data);
+    const audioBlob = new Blob(audioChunksRef.current, { type: 'audio/webm' });
-    const avg = data.reduce((a, b) => a + b) / data.length;
+    // Send audio through WebRTC data channel
-    if (avg < SILENCE_THRESHOLD) {
+    webrtcServiceRef.current?.sendAudioChunk(audioBlob);
-      if (!silenceStartRef.current) {
+    // Signal processing start via WebSocket
-        silenceStartRef.current = Date.now();
+    wsMessage({ type: 'processing-start' });
-      } else if (Date.now() - silenceStartRef.current > SILENCE_DURATION) {
+
-        sendMessage({ type: 'request-response' });
+    audioChunksRef.current = [];
-        silenceStartRef.current = null;
+    setIsProcessing(true);
-      }
+  }, [wsMessage]);
-    } else {
+
-      silenceStartRef.current = null;
+  const handleRTCMessage = useCallback((message: RTCMessage) => {
+    if (message.type === 'audio-received') {
+      // Backend confirmed audio receipt
+      setIsProcessing(true);
     }
-  }, [isCalling, sendMessage]);
+  }, []);
 
   const startCall = useCallback(async () => {
-    webrtcServiceRef.current = new WebRTCService(sendMessage);
+    // Initialize WebRTC with message handler
+    webrtcServiceRef.current = new WebRTCService(handleRTCMessage);
     await webrtcServiceRef.current.initializeCall();
 
+    // Signal call start via WebSocket
+    wsMessage({ type: 'call-start' });
+
     const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
     audioContextRef.current = new AudioContext();
     const source = audioContextRef.current.createMediaStreamSource(stream);
     analyserRef.current = audioContextRef.current.createAnalyser();
     source.connect(analyserRef.current);
 
-    intervalRef.current = window.setInterval(checkSilence, 100);
+    // Start VAD monitoring
+    intervalRef.current = window.setInterval(() => {
+      if (!analyserRef.current || !isCalling) {
+        return;
+      }
+
+      const data = new Float32Array(analyserRef.current.frequencyBinCount);
+      analyserRef.current.getFloatFrequencyData(data);
+      const avg = data.reduce((a, b) => a + b) / data.length;
+
+      if (avg < SILENCE_THRESHOLD) {
+        if (silenceStartRef.current === null) {
+          silenceStartRef.current = Date.now();
+        } else if (Date.now() - silenceStartRef.current > SILENCE_DURATION) {
+          sendAudioChunk();
+          silenceStartRef.current = null;
+        }
+      } else {
+        silenceStartRef.current = null;
+      }
+    }, 100);
+
     setIsCalling(true);
-  }, [checkSilence, sendMessage]);
+  }, [handleRTCMessage, wsMessage, sendAudioChunk]);
 
   const hangUp = useCallback(async () => {
     if (intervalRef.current) {
       clearInterval(intervalRef.current);
     }
+
     analyserRef.current = null;
     audioContextRef.current?.close();
     audioContextRef.current = null;
 
     await webrtcServiceRef.current?.endCall();
     webrtcServiceRef.current = null;
-    setIsCalling(false);
-    sendMessage({ type: 'call-ended' });
-  }, [sendMessage]);
 
-  return { isCalling, startCall, hangUp };
+    setIsCalling(false);
+    setIsProcessing(false);
+    wsMessage({ type: 'call-ended' });
+  }, [wsMessage]);
+
+  return {
+    isCalling,
+    isProcessing,
+    startCall,
+    hangUp,
+  };
 };
 
 export default useCall;

client/src/hooks/useWebRTC.ts

@@ -1,75 +1,44 @@
 import { useRef, useCallback } from 'react';
+import { WebRTCService } from '../services/WebRTC/WebRTCService';
+import type { RTCMessage } from '~/common';
 import useWebSocket from './useWebSocket';
 
-const SILENCE_THRESHOLD = -50;
-const SILENCE_DURATION = 1000;
-
 const useWebRTC = () => {
   const { sendMessage } = useWebSocket();
-  const localStreamRef = useRef<MediaStream | null>(null);
+  const webrtcServiceRef = useRef<WebRTCService | null>(null);
-  const audioContextRef = useRef<AudioContext | null>(null);
-  const analyserRef = useRef<AnalyserNode | null>(null);
-  const silenceStartTime = useRef<number | null>(null);
-  const isProcessingRef = useRef(false);
 
-  const log = (msg: string) => console.log(`[WebRTC ${new Date().toISOString()}] ${msg}`);
+  const handleRTCMessage = useCallback(
-
+    (message: RTCMessage) => {
-  const processAudioLevel = () => {
+      switch (message.type) {
-    if (!analyserRef.current || !isProcessingRef.current) {
+        case 'audio-chunk':
-      return;
+          sendMessage({ type: 'processing-start' });
-    }
+          break;
-
+        case 'transcription':
-    const dataArray = new Float32Array(analyserRef.current.frequencyBinCount);
+        case 'llm-response':
-    analyserRef.current.getFloatFrequencyData(dataArray);
+        case 'tts-chunk':
-    const average = dataArray.reduce((a, b) => a + b) / dataArray.length;
+          // TODO: Handle streaming responses
-
+          break;
-    if (average < SILENCE_THRESHOLD) {
-      if (!silenceStartTime.current) {
-        silenceStartTime.current = Date.now();
-        log(`Silence started: ${average}dB`);
-      } else if (Date.now() - silenceStartTime.current > SILENCE_DURATION) {
-        log('Silence threshold reached - requesting response');
-        sendMessage({ type: 'request-response' });
-        silenceStartTime.current = null;
       }
-    } else {
+    },
-      silenceStartTime.current = null;
+    [sendMessage],
-    }
+  );
-
-    requestAnimationFrame(processAudioLevel);
-  };
 
   const startLocalStream = async () => {
     try {
-      log('Starting audio capture');
+      webrtcServiceRef.current = new WebRTCService(handleRTCMessage);
-      localStreamRef.current = await navigator.mediaDevices.getUserMedia({ audio: true });
+      await webrtcServiceRef.current.initializeCall();
-
+      sendMessage({ type: 'call-start' });
-      audioContextRef.current = new AudioContext();
-      const source = audioContextRef.current.createMediaStreamSource(localStreamRef.current);
-      analyserRef.current = audioContextRef.current.createAnalyser();
-
-      source.connect(analyserRef.current);
-      isProcessingRef.current = true;
-      processAudioLevel();
-
-      log('Audio capture started');
     } catch (error) {
-      log(`Error: ${error instanceof Error ? error.message : 'Unknown error'}`);
+      console.error(error);
       throw error;
     }
   };
 
   const stopLocalStream = useCallback(() => {
-    log('Stopping audio capture');
+    webrtcServiceRef.current?.endCall();
-    isProcessingRef.current = false;
+    webrtcServiceRef.current = null;
-    audioContextRef.current?.close();
+    sendMessage({ type: 'call-ended' });
-    localStreamRef.current?.getTracks().forEach((track) => track.stop());
+  }, [sendMessage]);
-
-    localStreamRef.current = null;
-    audioContextRef.current = null;
-    analyserRef.current = null;
-    silenceStartTime.current = null;
-  }, []);
 
   return { startLocalStream, stopLocalStream };
 };

client/src/hooks/useWebSocket.ts

@@ -1,39 +1,47 @@
 import { useEffect, useRef, useState, useCallback } from 'react';
 import { useGetWebsocketUrlQuery } from 'librechat-data-provider/react-query';
+import type { RTCMessage } from '~/common';
 
 const useWebSocket = () => {
-  const { data: url } = useGetWebsocketUrlQuery();
+  const { data: data } = useGetWebsocketUrlQuery();
   const [isConnected, setIsConnected] = useState(false);
   const wsRef = useRef<WebSocket | null>(null);
 
-  console.log('wsConfig:', url?.url);
-
   const connect = useCallback(() => {
-    if (!url?.url) {
+    if (!data || !data.url) {
       return;
     }
 
-    wsRef.current = new WebSocket(url?.url);
+    wsRef.current = new WebSocket(data.url);
     wsRef.current.onopen = () => setIsConnected(true);
     wsRef.current.onclose = () => setIsConnected(false);
     wsRef.current.onerror = (err) => console.error('WebSocket error:', err);
 
     wsRef.current.onmessage = (event) => {
-      const msg = JSON.parse(event.data);
+      const msg: RTCMessage = JSON.parse(event.data);
-      if (msg.type === 'audio-response') {
+      switch (msg.type) {
-        const audioData = msg.data;
+        case 'transcription':
-        const audio = new Audio(`data:audio/mp3;base64,${audioData}`);
+          // TODO: Handle transcription update
-        audio.play().catch(console.error);
+          break;
+        case 'llm-response':
+          // TODO: Handle LLM streaming response
+          break;
+        case 'tts-chunk':
+          if (typeof msg.data === 'string') {
+            const audio = new Audio(`data:audio/mp3;base64,${msg.data}`);
+            audio.play().catch(console.error);
+          }
+          break;
       }
     };
-  }, [url?.url]);
+  }, [data?.url]);
 
   useEffect(() => {
     connect();
     return () => wsRef.current?.close();
   }, [connect]);
 
-  const sendMessage = useCallback((message: any) => {
+  const sendMessage = useCallback((message: Record<string, unknown>) => {
     if (wsRef.current?.readyState === WebSocket.OPEN) {
       wsRef.current.send(JSON.stringify(message));
     }

client/src/services/WebRTC/WebRTCService.ts

@@ -1,36 +1,55 @@
+import type { RTCMessage } from '~/common';
 export class WebRTCService {
   private peerConnection: RTCPeerConnection | null = null;
+  private dataChannel: RTCDataChannel | null = null;
   private mediaRecorder: MediaRecorder | null = null;
-  private sendMessage: (msg: any) => void;
+  private onMessage: (msg: RTCMessage) => void;
 
-  constructor(sendMessage: (msg: any) => void) {
+  constructor(onMessage: (msg: RTCMessage) => void) {
-    this.sendMessage = sendMessage;
+    this.onMessage = onMessage;
   }
 
   async initializeCall() {
-    const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
     this.peerConnection = new RTCPeerConnection();
-    stream.getTracks().forEach((track) => this.peerConnection?.addTrack(track, stream));
+    this.dataChannel = this.peerConnection.createDataChannel('audio');
 
+    const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
     this.mediaRecorder = new MediaRecorder(stream);
+
     this.mediaRecorder.ondataavailable = (e) => {
-      if (e.data.size > 0) {
+      if (e.data.size > 0 && this.dataChannel?.readyState === 'open') {
-        const reader = new FileReader();
+        e.data.arrayBuffer().then((buffer) => {
-        reader.onload = () => {
+          this.dataChannel?.send(buffer);
-          this.sendMessage({
+        });
-            type: 'audio-chunk',
-            data: reader.result,
-          });
-        };
-        reader.readAsDataURL(e.data);
       }
     };
-    this.mediaRecorder.start();
+
+    this.mediaRecorder.start(100);
+    this.setupDataChannel();
+  }
+
+  private setupDataChannel() {
+    if (!this.dataChannel) {
+      return;
+    }
+
+    this.dataChannel.onmessage = (event) => {
+      this.onMessage({
+        type: 'audio-chunk',
+        data: event.data,
+      });
+    };
+  }
+
+  public async sendAudioChunk(audioBlob: Blob) {
+    if (this.dataChannel && this.dataChannel.readyState === 'open') {
+      this.dataChannel.send(await audioBlob.arrayBuffer());
+    }
   }
 
   async endCall() {
     this.mediaRecorder?.stop();
+    this.dataChannel?.close();
     this.peerConnection?.close();
-    this.peerConnection = null;
   }
 }

package-lock.json

@@ -98,6 +98,7 @@
         "mongoose": "^8.9.5",
         "multer": "^1.4.5-lts.1",
         "nanoid": "^3.3.7",
+        "node-pre-gyp": "^0.17.0",
         "nodemailer": "^6.9.15",
         "ollama": "^0.5.0",
         "openai": "^4.47.1",
@@ -118,6 +119,7 @@
         "ua-parser-js": "^1.0.36",
         "winston": "^3.11.0",
         "winston-daily-rotate-file": "^4.7.1",
+        "wrtc": "^0.4.7",
         "ws": "^8.18.0",
         "youtube-transcript": "^1.2.1",
         "zod": "^3.22.4"
@@ -17171,8 +17173,7 @@
     "node_modules/abbrev": {
       "version": "1.1.1",
       "resolved": "https://registry.npmjs.org/abbrev/-/abbrev-1.1.1.tgz",
-      "integrity": "sha512-nne9/IiQ/hzIhY6pdDnbBtz7DjPTKrY00P/zvPSm5pOFkl6xuGrGnXn/VtTNNfNtAfZ9/1RtehkszU9qcTii0Q==",
+      "integrity": "sha512-nne9/IiQ/hzIhY6pdDnbBtz7DjPTKrY00P/zvPSm5pOFkl6xuGrGnXn/VtTNNfNtAfZ9/1RtehkszU9qcTii0Q=="
-      "dev": true
     },
     "node_modules/abort-controller": {
       "version": "3.0.0",
@@ -17379,6 +17380,21 @@
       "resolved": "https://registry.npmjs.org/append-field/-/append-field-1.0.0.tgz",
       "integrity": "sha512-klpgFSWLW1ZEs8svjfb7g4qWY0YS5imI82dTg+QahUvJ8YqAY0P10Uk8tTyh9ZGuYEZEMaeJYCF5BFuX552hsw=="
     },
+    "node_modules/aproba": {
+      "version": "1.2.0",
+      "resolved": "https://registry.npmjs.org/aproba/-/aproba-1.2.0.tgz",
+      "integrity": "sha512-Y9J6ZjXtoYh8RnXVCMOU/ttDmk1aBjunq9vO0ta5x85WDQiQfUF9sIPBITdbiiIVcBo03Hi3jMxigBtsddlXRw=="
+    },
+    "node_modules/are-we-there-yet": {
+      "version": "1.1.7",
+      "resolved": "https://registry.npmjs.org/are-we-there-yet/-/are-we-there-yet-1.1.7.tgz",
+      "integrity": "sha512-nxwy40TuMiUGqMyRHgCSWZ9FM4VAoRP4xUYSTv5ImRog+h9yISPbVH7H8fASCIzYn9wlEv4zvFL7uKDMCFQm3g==",
+      "deprecated": "This package is no longer supported.",
+      "dependencies": {
+        "delegates": "^1.0.0",
+        "readable-stream": "^2.0.6"
+      }
+    },
     "node_modules/arg": {
       "version": "5.0.2",
       "resolved": "https://registry.npmjs.org/arg/-/arg-5.0.2.tgz",
@@ -18861,6 +18877,14 @@
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@@ -37092,6 +37500,40 @@
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