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.
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 videoTranscripts - Real-time speech-to-textScreen shares - JPEG/PNG/H.264 framesChat 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/RTMS_CONNECTION_FLOW.md for the raw protocol details.
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
.
├── 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/
│ └── send_individual_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/
│ └── save_edited_audio_and_video_to_local_storage_js/
├── 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/
├── jev_transcript_analysis_js/
├── prompt_for_user_consent_js/
├── send_audio_to_openai_realtime_api_with_audio_playback_js/
├── start_stop_rtms_control_js/
└── stream_audio_and_video_deepfake_detection_js/
Category
Description
Count
audio/Transcription services (AWS, Azure, OpenAI Realtime, Zoom Scribe, AssemblyAI, Whisper)
10
boilerplate/Starter templates (JS, Python, Go, Java, C++, .NET, SDK)
9
chat/In-meeting chat message processing
1
streaming/Live streaming (AWS IVS, Kinesis, YouTube, custom)
6
storage/Cloud & local storage (S3, Azure Blob, local, edited media)
7
transcript/Transcript processing & LLM integration
5
zoom_apps/Complete Zoom App examples (AI, RAG, games, Jev, deepfake detection)
10
video/Video analysis and individual video stream samples
3
video-sdk/Video SDK integration
3
screen_share/Screen capture & PDF export
2
rtms_api/Manual RTMS session control and reconnection testing
3
rtms-distributed-sample/Distributed RTMS fanout/fanin sample with regional compute, control stores, cache, and artifact storage
1
rtms_mcp_client/Model Context Protocol client
1
library/Shared utilities (RTMSManager, helpers)
2
Each link opens the project directory and its project-level readme.md.
Chat, RTMS Control, And Screen Share Distributed, MCP, And Shared Components
Project
Description
rtms-distributed-sampleDistributed RTMS webhook routing, regional compute, queues, caches, and artifact storage.
zoom-rtms-mcp-clientRoutes RTMS transcript turns through an MCP client and configurable LLM router.
library/javascriptShared JavaScript RTMSManager, webhook, media, and frontend WebSocket helpers.
library/pythonShared Python RTMS utilities and protocol helpers.
RTMS streams are delivered over standard WebSocket connections—no SDK or library is required . The library/
Helper classes for audio/video processingReconnection managers for handling network interruptionsEvent 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.
MIT License - Copyright (c) 2025 Zoom Video Communications, Inc.
See LICENSE.md for full text.