Zoom RTMS Samples Repository

This repository contains sample projects demonstrating how to work with Zoom's Realtime Media Streams (RTMS) in JavaScript, Python, Go, Java, C++, .NET, and SDK implementations.

What is RTMS?

Zoom Realtime Media Streams (RTMS) allows developers to access realtime media data from Zoom meetings, including:

• Audio streams - Raw PCM audio (L16, 16kHz/24kHz)
• Video streams - H.264 encoded video
• Transcripts - Real-time speech-to-text
• Screen shares - JPEG/PNG/H.264 frames
• Chat messages - In-meeting chat

Note: RTMS is built on standard WebSocket technology. You do not need the SDK or library to access RTMS streams—you can connect directly using any WebSocket client in any language. The library/ and SDK are provided for convenience, offering helper classes, reconnection managers, and event handling. Feel free to use them as-is, modify them, or implement your own logic for advanced use cases. See RTMS_CONNECTION_FLOW.md for the raw protocol details.

Quick Start

import { RTMSManager } from './library/javascript/rtmsManager/RTMSManager.js';
import WebhookManager from './library/javascript/webhookManager/WebhookManager.js';
import express from 'express';

const app = express();

// Initialize
await RTMSManager.init({
  credentials: {
    meeting: {
      clientId: process.env.ZOOM_CLIENT_ID,
      clientSecret: process.env.ZOOM_CLIENT_SECRET,
      zoomSecretToken: process.env.ZOOM_SECRET_TOKEN,
    }
  }
});

// Setup webhook
const webhookManager = new WebhookManager({
  config: { webhookPath: '/', zoomSecretToken: process.env.ZOOM_SECRET_TOKEN },
  app
});
webhookManager.on('event', (event, payload) => RTMSManager.handleEvent(event, payload));
webhookManager.setup();

// Handle media
RTMSManager.on('audio', ({ buffer, userName }) => console.log(`Audio from ${userName}`));
RTMSManager.on('transcript', ({ text, userName }) => console.log(`${userName}: ${text}`));

// Start
await RTMSManager.start();
app.listen(3000);

→ Full examples: boilerplate/ | Library docs: library/javascript/readme.md

Repository Structure

.
├── audio/                          # Audio processing & transcription samples
│   ├── send_audio_to_assemblyai_transcribe_service_js/
│   ├── send_audio_to_assemblyai_transcribe_service_sdk/
│   ├── send_audio_to_aws_transcribe_service_js/
│   ├── send_audio_to_aws_transcribe_service_sdk/
│   ├── send_audio_to_azure_speech_to_text_service_js/
│   ├── send_audio_to_azure_speech_to_text_service_sdk/
│   ├── send_audio_to_openai_realtime_api/
│   ├── send_audio_to_whisper_local_transcribe_service_js/
│   └── send_audio_to_zoom_scribe_transcribe_service_js/
├── boilerplate/                    # Starter templates for various languages
│   ├── working_cplusplus_wss/
│   ├── working_dotnetcore/
│   ├── working_go/
│   ├── working_java/
│   ├── working_js/
│   ├── working_python/
│   ├── working_python_wss/
│   ├── working_python_wss_zoom_room_screenshot/
│   └── working_sdk/
├── chat/                           # In-meeting chat processing samples
│   └── print_chat_messages_js/
├── library/                        # Shared libraries
│   ├── javascript/                 # RTMSManager, WebhookManager, helpers
│   └── python/                     # Python RTMS utilities
├── rtms_api/                       # Manual RTMS start/stop control
│   ├── manual_start_stop_using_js/
│   ├── manual_start_stop_using_python/
│   └── reconnection_and_chaos_mode_js/
├── rtms-distributed-sample/         # Regional fanout/fanin architecture sample
├── rtms_mcp_client/                # Model Context Protocol integration
├── screen_share/                   # Screen share capture samples
│   ├── save_screen_share_js/
│   └── save_screen_share_pdf_js/
├── storage/                        # Recording & cloud storage samples
│   ├── save_audio_and_video_to_aws_s3_storage_js/
│   ├── save_audio_and_video_to_aws_s3_storage_sdk/
│   ├── save_audio_and_video_to_azure_blob_storage_js/
│   ├── save_audio_and_video_to_azure_blob_storage_sdk/
│   ├── save_audio_and_video_to_local_storage_js/
│   └── save_audio_and_video_to_local_storage_sdk/
├── streaming/                      # Live streaming samples
│   ├── stream_audio_and_video_to_custom_frontend_passthru_js/
│   ├── stream_audio_and_video_to_custom_frontend_sdk/
│   ├── stream_audio_and_video_to_youtube_greedy_gap_filler_js/
│   ├── stream_to_aws_ivs_gap_filler_js/
│   ├── stream_to_aws_ivs_jitter_buffer_js/
│   └── stream_to_aws_kinesis_passthru_js/
├── transcript/                     # Transcript processing samples
│   ├── save_transcript_js/
│   ├── save_transcript_sdk/
│   ├── send_transcript_to_claude_js/
│   ├── send_transcript_to_openai_js/
│   └── send_transcript_to_openrouter_js/
├── video/                          # Video analysis samples
│   ├── detect_emotion_using_amazon_rekognition_js/
│   ├── detect_object_using_tensorflow_js/
│   └── individual_video_js/
├── video-sdk/                      # Video SDK integration samples
│   ├── vsdk_working_java/
│   ├── vsdk_working_js/
│   └── vsdk_working_python/
└── zoom_apps/                      # Complete Zoom App examples
    ├── ai_chat_with_audio_playback_js/
    ├── ai_dnd_game_js/
    ├── ai_industry_specific_notetaker_js/
    ├── ai_rag_customer_support_js/
    ├── ai_transcript_analysis_js/
    ├── prompt_for_user_consent_js/
    ├── start_stop_rtms_control_js/
    └── stream_audio_and_video_deepfake_detection_js/

Sample Categories

About the Library & SDK

RTMS streams are delivered over standard WebSocket connections—no SDK or library is required. The library/ and SDK are provided purely for convenience:

• Helper classes for audio/video processing
• Reconnection managers for handling network interruptions
• Event routing and connection lifecycle management

For advanced use cases requiring performance optimization or unique customization, you can modify the library code or implement your own WebSocket handling directly. See RTMS_CONNECTION_FLOW.md for the complete protocol specification.

Documentation

License

MIT License - Copyright (c) 2025 Zoom Video Communications, Inc.

See LICENSE.md for full text.