Zoom Realtime Media Streams (RTMS)

Expert guidance for accessing live audio, video, transcript, chat, and screen share data from Zoom meetings, webinars, Video SDK sessions, and Zoom Contact Center Voice in real-time. RTMS uses a WebSocket-based protocol with open standards and does not require a meeting bot to capture the media plane.

Read This First (Critical)

RTMS is primarily a backend media ingestion service.

• Your backend receives and processes live media: audio, video, screen share, chat, transcript.
• RTMS is not a frontend UI SDK by itself.
• Processing is event-triggered: backend waits for RTMS start webhook events before stream handling begins.

Optional architecture (common):

• Add a Zoom App SDK frontend for in-client UI/controls.
• Stream backend RTMS outputs to frontend via WebSocket (or SSE, gRPC, queue workers, etc.).

Use RTMS for media/data plane, and use frontend frameworks/Zoom Apps for presentation + user interactions.

Official Documentation: https://developers.zoom.us/docs/rtms/ SDK Reference (JS): https://zoom.github.io/rtms/js/ SDK Reference (Python): https://zoom.github.io/rtms/py/ Sample Repository: https://github.com/zoom/rtms-samples

Quick Links

New to RTMS? Follow this path:

1. Connection Architecture - Two-phase WebSocket design
2. SDK Quickstart - Fastest way to receive media (recommended)
3. Manual WebSocket - Full protocol control without SDK
4. Media Types - Audio, video, transcript, chat, screen share

Complete Implementation:

• RTMS Bot - End-to-end bot implementation guide

Reference:

• Lifecycle Flow - Complete webhook-to-streaming flow
• Data Types - All enums and constants
• Webhooks - Event subscription details
• Environment Variables - credential modes and runtime knobs
• Quickstart Notes - Secondary quickstart guide
• Changelog - RTMS skill documentation changes
• Integrated Index - see the section below in this file

Having issues?

• Connection fails -> Common Issues
• Duplicate connections -> Webhook Gotchas
• No audio/video -> Media Configuration
• Start with preflight checks -> 5-Minute Runbook

Supported Products

Product	Webhook Event	Payload ID	App Type
Meetings	meeting.rtms_started / meeting.rtms_stopped	meeting_uuid	User-managed General App
Webinars	webinar.rtms_started / webinar.rtms_stopped	meeting_uuid (same!)	User-managed General App
Video SDK	session.rtms_started / session.rtms_stopped	session_id	Video SDK App
Zoom Contact Center Voice	contact_center.voice_rtms_started / contact_center.voice_rtms_stopped	Product-specific stream/session identifiers	General user/admin or S2S, based on scopes

Once connected, the core signaling/media socket model is shared across products. Meetings, webinars, and Video SDK sessions use the familiar start/stop webhooks. Zoom Contact Center Voice adds its own RTMS/ZCC Voice event family and should be treated as the same transport model with product-specific event payloads.

RTMS Overview

RTMS is a data pipeline that gives your app access to live media from Zoom meetings, webinars, and Video SDK sessions without participant bots. Instead of having automated clients join meetings, use RTMS to collect media data directly from Zoom's infrastructure.

What RTMS Provides

Media Type	Format	Use Cases
Audio	PCM (L16), G.711, G.722, Opus	Transcription, voice analysis, recording
Video	H.264, JPG, PNG	Recording, AI vision, thumbnails, active participant selection
Screen Share	H.264, JPG, PNG	Content capture, slide extraction
Transcript	JSON text	Meeting notes, search, compliance
Chat	JSON text	Archive, sentiment analysis

Recent Protocol Changes

• Zoom Contact Center Voice support: RTMS now covers Contact Center Voice audio and transcript scenarios.
• Transcript Language Identification control: transcript media handshakes now support src_language and enable_lid. Default behavior is LID enabled. Set enable_lid: false to force a fixed language.
• Single individual video stream subscription: RTMS can now stream one participant's camera feed at a time when data_opt is set to VIDEO_SINGLE_INDIVIDUAL_STREAM.
• Graceful client-initiated shutdown: backends can send STREAM_CLOSE_REQ over the signaling socket and wait for STREAM_CLOSE_RESP.
• Media keep-alive tolerance increased: media socket keep-alive timeout is now 65 seconds, not 35.
• Audio buffer control in signaling handshake: signaling handshakes now support optional buffer_data. Default is true, which delivers up to 60 seconds of initial audio buffered while signaling/media sockets are established. For live streaming pipelines, set buffer_data: false to drop the startup backlog and start from live audio once connected.
• Media payload length field: audio, video, and screen share media data events can include content.length, the original binary length before base64 encoding.
• ZCC Voice enum update: CONFERENCE_PARTICIPANT_LEFT = 29 was added to RTMS_ZCC_VOICE_EVENT_TYPE.

Media Socket Deprecation

Deprecated: Unified media socket mode and media_type: 32 must not be used. Open a separate data connection for each required media type and send exactly one concrete media type per data handshake: audio (1), video (2), screen share (4), transcript (8), or chat (16). Do not send combined values such as 3, 5, or 9 on the wire.

Two Approaches

Approach	Best For	Complexity
SDK (@zoom/rtms)	Most use cases	Low - handles WebSocket complexity
Manual WebSocket	Custom protocols, other languages	High - full protocol implementation

Prerequisites

• Node.js 20.3.0+ (24 LTS recommended) for JavaScript SDK
• Python 3.10+ for Python SDK
• User-managed General App (meetings/webinars), General or S2S app (Contact Center Voice), or Video SDK App (Video SDK) with RTMS enabled
• Webhook endpoint for RTMS events
• Server to receive WebSocket streams

Need to create/configure the Marketplace app first? Use Marketplace app management before RTMS implementation. It documents General App manifests, event subscription API behavior, app-owned client_credentials, and RTMS event validation drift. Use the user-managed Meeting/Webinar RTMS template, or the General App or S2S Contact Center Voice templates in the template selector.

Need RTMS access? Post in Zoom Developer Forum requesting RTMS access with your use case.

Quick Start (SDK - Recommended)

import rtms from "@zoom/rtms";

// All RTMS start/stop events across products
const RTMS_EVENTS = ["meeting.rtms_started", "webinar.rtms_started", "session.rtms_started"];

// Handle webhook events
rtms.onWebhookEvent(({ event, payload }) => {
  if (!RTMS_EVENTS.includes(event)) return;

  const client = new rtms.Client();

  client.onAudioData((data, timestamp, metadata) => {
    console.log(`Audio from ${metadata.userName}: ${data.length} bytes`);
  });

  client.onTranscriptData((data, timestamp, metadata) => {
    const text = data.toString('utf8');
    console.log(`${metadata.userName}: ${text}`);
  });

  client.onJoinConfirm((reason) => {
    console.log(`Joined session: ${reason}`);
  });

  // SDK handles all WebSocket connections automatically
  // Accepts both meeting_uuid and session_id transparently
  client.join(payload);
});

Quick Start (Manual WebSocket)

For full control or non-SDK languages, implement the two-phase WebSocket protocol:

const WebSocket = require('ws');
const crypto = require('crypto');

const RTMS_EVENTS = ['meeting.rtms_started', 'webinar.rtms_started', 'session.rtms_started'];

// 1. Generate signature
// For meetings/webinars: uses meeting_uuid. For Video SDK: uses session_id.
function generateSignature(clientId, idValue, streamId, clientSecret) {
  const message = `${clientId},${idValue},${streamId}`;
  return crypto.createHmac('sha256', clientSecret).update(message).digest('hex');
}

// 2. Handle webhook
app.post('/webhook', (req, res) => {
  res.status(200).send();  // CRITICAL: Respond immediately!
  
  const { event, payload } = req.body;
  if (RTMS_EVENTS.includes(event)) {
    connectToRTMS(payload);
  }
});

// 3. Connect to signaling WebSocket
function connectToRTMS(payload) {
  const { server_urls, rtms_stream_id } = payload;
  // meeting_uuid for meetings/webinars, session_id for Video SDK
  const idValue = payload.meeting_uuid || payload.session_id;
  const signature = generateSignature(CLIENT_ID, idValue, rtms_stream_id, CLIENT_SECRET);
  
  const signalingWs = new WebSocket(server_urls);
  
  signalingWs.on('open', () => {
    signalingWs.send(JSON.stringify({
      msg_type: 1,  // Handshake request
      protocol_version: 1,
      meeting_uuid: idValue,
      rtms_stream_id,
      signature,
      buffer_data: false // Use false for live streaming; true preserves startup audio
    }));
  });
  
  // ... handle responses, connect to media WebSocket
}

See: Manual WebSocket Guide for complete implementation.

Media Data Connections

Select the media needed by the application, then open one data WebSocket for each selected type using the matching URL from the signaling response:

Type	Wire value	Server URL
Audio	1	server_urls.audio
Video	2	server_urls.video
Screen Share	4	server_urls.deskshare
Transcript	8	server_urls.transcript
Chat	16	server_urls.chat

For generic sample configuration, prefer readable names such as RTMS_MEDIA_TYPES=audio,video,transcript. An SDK or library may accept an internal selection bitmask, but it must expand that selection into separate connections. Never send a selection mask, a combined value, or 32 as the data-handshake media_type.

Critical Gotchas

Issue	Solution
Only 1 signaling session allowed	A duplicate signaling connection can kick out the existing session. Track active sessions; each session still needs one media socket per selected type
Split media sockets required	Use one data WebSocket per concrete media type. Unified mode, server_urls.all, combined media_type values, and media_type: 32 are deprecated
Respond 200 immediately	If webhook delays, Zoom retries creating duplicate connections
Heartbeat mandatory	Respond to msg_type 12 with msg_type 13, or connection dies
Reconnection is YOUR job	RTMS doesn't auto-reconnect. Media keep-alive tolerance is now about 65s; signaling remains around 60s
Transcript language drift	Use src_language plus enable_lid: false when you want fixed-language transcription instead of automatic language switching
Audio params must match the support matrix	Do not mix arbitrary content_type, codec, sample_rate, channel, and send_rate. The media server rejects unsupported combinations
Single participant video only	VIDEO_SINGLE_INDIVIDUAL_STREAM supports one participant at a time. A new VIDEO_SUBSCRIPTION_REQ overrides the previous selection
Graceful close is explicit now	Use STREAM_CLOSE_REQ / STREAM_CLOSE_RESP when your backend wants to terminate the stream cleanly
RTMS manifest validation can drift	Create-time validation can reject RTMS event names that PUT /manifest persists on an existing draft app. Confirm with GET /manifest after updates

Environment Variables

SDK Environment Variables

# Required - Authentication
ZM_RTMS_CLIENT=your_client_id          # Zoom OAuth Client ID
ZM_RTMS_SECRET=your_client_secret      # Zoom OAuth Client Secret

# Optional - Webhook server
ZM_RTMS_PORT=8080                      # Default: 8080
ZM_RTMS_PATH=/webhook                  # Default: /

# Optional - Logging
ZM_RTMS_LOG_LEVEL=info                 # error, warn, info, debug, trace
ZM_RTMS_LOG_FORMAT=progressive         # progressive or json
ZM_RTMS_LOG_ENABLED=true

Manual Implementation Variables

ZOOM_CLIENT_ID=your_client_id
ZOOM_CLIENT_SECRET=your_client_secret
ZOOM_SECRET_TOKEN=your_webhook_token   # For webhook validation

Zoom App Setup

For Meetings and Webinars (General App)

1. Go to marketplace.zoom.us -> Develop -> Build App
2. Choose General App -> User-Managed
3. Features -> Access -> Enable Event Subscription
4. Add Events -> Search "rtms" -> Select:
   • meeting.rtms_started
   • meeting.rtms_stopped
   • webinar.rtms_started (if using webinars)
   • webinar.rtms_stopped (if using webinars)
5. Scopes -> Add Scopes -> Search "rtms" -> Add:
   • meeting:read:meeting_audio
   • meeting:read:meeting_video
   • meeting:read:meeting_transcript
   • meeting:read:meeting_chat
   • webinar:read:webinar_audio (if using webinars)
   • webinar:read:webinar_video (if using webinars)
   • webinar:read:webinar_transcript (if using webinars)
   • webinar:read:webinar_chat (if using webinars)

For Video SDK (Video SDK App)

1. Go to marketplace.zoom.us -> Develop -> Build App
2. Choose Video SDK App
3. Use your SDK Key and SDK Secret (not OAuth Client ID/Secret)
4. Add Events:
   • session.rtms_started
   • session.rtms_stopped

Sample Repositories

Official Samples

Repository	Description
rtms-samples	RTMSManager, boilerplates, AI samples
rtms-quickstart-js	JavaScript SDK quickstart
rtms-quickstart-py	Python SDK quickstart
rtms-sdk-cpp	C++ SDK
zoom-rtms	Main SDK repository

AI Integration Samples

Sample	Description
rtms-meeting-assistant-starter-kit	AI meeting assistant with summaries
arlo-meeting-assistant	Production meeting assistant with DB
videosdk-rtms-transcribe-audio	Whisper transcription

Complete Documentation

Concepts

• Connection Architecture - Two-phase WebSocket design
• Lifecycle Flow - Webhook to streaming flow

Examples

• SDK Quickstart - Using @zoom/rtms SDK
• Manual WebSocket - Raw protocol implementation
• RTMS Bot - Complete bot implementation guide
• AI Integration - Transcription and analysis patterns

References

• Media Types - Audio, video, transcript, chat, screen share
• Data Types - All enums and constants
• Connection - WebSocket protocol details
• Webhooks - Event subscription

Troubleshooting

• Common Issues - FAQ and solutions

Resources

• Official docs: https://developers.zoom.us/docs/rtms/
• Data types: https://developers.zoom.us/docs/rtms/data-types/
• Media params: https://developers.zoom.us/docs/rtms/media-parameter-definition/
• Developer forum: https://devforum.zoom.us/

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Need help? Start with Integrated Index section below for complete navigation.

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Integrated Index

This section was migrated from SKILL.md.

RTMS provides real-time access to live audio, video, transcript, chat, and screen share from Zoom meetings, webinars, and Video SDK sessions.

Critical Positioning

Treat RTMS as a backend service for receiving and processing media streams.

• Backend role: ingest audio/video/share/chat/transcript, run AI/analytics, persist/forward data.
• Optional frontend role: Zoom App SDK or web dashboard that consumes processed stream data from backend transport (WebSocket/SSE/other).
• Kickoff model: backend waits for RTMS start webhook events, then starts stream processing.

Do not model RTMS as a frontend-only SDK.

Quick Start Path

If you're new to RTMS, follow this order:

1. Run preflight checks first -> RUNBOOK.md

2. Understand the architecture -> concepts/connection-architecture.md

   • Two-phase WebSocket: Signaling + Media
   • Why RTMS doesn't use bots

3. Choose your approach -> SDK or Manual

   • SDK (recommended): examples/sdk-quickstart.md
   • Manual WebSocket: examples/manual-websocket.md

4. Understand the lifecycle -> concepts/lifecycle-flow.md

   • Webhook -> Signaling -> Media -> Streaming

5. Configure media types -> references/media-types.md

   • Audio, video, transcript, chat, screen share

6. Troubleshoot issues -> troubleshooting/common-issues.md

   • Connection problems, duplicate webhooks, missing data

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Documentation Structure

rtms/
├── SKILL.md                           # Main skill overview
├── SKILL.md                           # This file - navigation guide
│
├── concepts/                          # Core architectural patterns
│   ├── connection-architecture.md     # Two-phase WebSocket design
│   └── lifecycle-flow.md              # Webhook to streaming flow
│
├── examples/                          # Complete working code
│   ├── sdk-quickstart.md              # Using @zoom/rtms SDK
│   ├── manual-websocket.md            # Raw protocol implementation
│   ├── rtms-bot.md                    # Complete RTMS bot implementation
│   └── ai-integration.md              # Transcription and analysis
│
├── references/                        # Reference documentation
│   ├── media-types.md                 # Audio, video, transcript, chat, share
│   ├── data-types.md                  # All enums and constants
│   ├── connection.md                  # WebSocket protocol details
│   └── webhooks.md                    # Event subscription
│
└── troubleshooting/                   # Problem solving guides
    └── common-issues.md               # FAQ and solutions

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By Use Case

I want to get meeting transcripts

1. SDK Quickstart - Fastest approach
2. Media Types - Transcript configuration
3. AI Integration - Whisper, Deepgram, AssemblyAI

I want to record meetings

1. Media Types - Audio + Video configuration
2. SDK Quickstart - Receiving media
3. AI Integration - Gap-filled recording

I want to build an AI meeting assistant

1. AI Integration - Complete patterns
2. SDK Quickstart - Media ingestion
3. Lifecycle Flow - Event handling

I want to build a complete RTMS bot

1. RTMS Bot - Complete implementation guide
2. Lifecycle Flow - Webhook to streaming flow
3. Connection Architecture - Two-phase design

I need full protocol control

1. Manual WebSocket - START HERE
2. Connection Architecture - Two-phase design
3. Data Types - All message types and enums
4. Connection - Protocol details

I'm getting connection errors

1. Common Issues - Diagnostic checklist
2. Connection Architecture - Verify flow
3. Webhooks - Validation and timing

I want to understand the architecture

1. Connection Architecture - Two-phase WebSocket
2. Lifecycle Flow - Complete flow diagram
3. Data Types - Protocol constants

---

By Product

I'm building for Zoom Meetings

• Standard RTMS setup. Webhook event: meeting.rtms_started. Uses General App with OAuth.
• Start with SDK Quickstart or Manual WebSocket.

I'm building for Zoom Webinars

• Same as meetings, but webhook event is webinar.rtms_started. Payload still uses meeting_uuid (NOT webinar_uuid).
• Add webinar scopes and event subscriptions. See Webhooks.
• Only panelist streams are confirmed available. Attendee streams may not be individual.

I'm building for Zoom Video SDK

• Webhook event: session.rtms_started. Payload uses session_id (NOT meeting_uuid).
• Requires a Video SDK App with SDK Key/Secret (not OAuth Client ID/Secret).
• Once connected, the protocol is identical to meetings.
• See Webhooks for payload details.

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Key Documents

1. Connection Architecture (CRITICAL)

concepts/connection-architecture.md

RTMS uses one signaling WebSocket plus one media WebSocket per selected media type:

• Signaling WebSocket: Authentication, control, heartbeats
• Media WebSockets: Separate audio/video/share/transcript/chat data connections

2. SDK vs Manual (DECISION POINT)

examples/sdk-quickstart.md vs examples/manual-websocket.md

SDK	Manual
Handles WebSocket complexity	Full protocol control
Automatic reconnection	DIY reconnection
Less code	More code
Best for most use cases	Best for custom requirements

3. Critical Gotchas (MOST COMMON ISSUES)

troubleshooting/common-issues.md

1. Respond 200 immediately - Delayed webhook responses cause duplicates
2. Only 1 signaling session per stream - Duplicate signaling joins can kick out the existing session; split media sockets are still required
3. Heartbeat required - Must respond to keep-alive or connection dies
4. Track active sessions - Prevent duplicate join attempts

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Key Learnings

Critical Discoveries:

1. Two-Phase WebSocket Design

   • Signaling: Control plane (handshake, heartbeat, start/stop)
   • Media: Data plane (audio, video, transcript, chat, share)
   • See: Connection Architecture

2. Webhook Response Timing

   • MUST respond 200 BEFORE any processing
   • Delayed response -> Zoom retries -> duplicate connections
   • See: Common Issues

3. Heartbeat is Mandatory

   • Signaling: Receive msg_type 12, respond with msg_type 13
   • Media: Same pattern
   • Failure to respond = connection closed
   • See: Connection

4. Signature Generation

   • Format: HMAC-SHA256(clientSecret, "clientId,meetingUuid,streamId")
   • For Video SDK, use session_id in place of meetingUuid
   • Webinars still use meeting_uuid (not webinar_uuid)
   • Required for both signaling and media handshakes
   • See: Manual WebSocket

5. Media Uses Split Data Connections

   • Audio=1, Video=2, Share=4, Transcript=8, Chat=16
   • Open one data WebSocket per selected media type using its matching URL
   • Never send combined values or the deprecated media_type: 32 in a data handshake
   • See: Media Types

6. Startup Audio Buffering is a Signaling Choice

   • buffer_data belongs in SIGNALING_HAND_SHAKE_REQ
   • default true can replay up to 60s of startup audio
   • set false for live streaming or live-only startup behavior
   • See: Connection

7. Screen Share is SEPARATE from Video

   • Different msg_type (16 vs 15)
   • Different media flag (4 vs 2)
   • Must subscribe separately
   • See: Media Types

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Quick Reference

"Connection fails"

-> Common Issues

"Duplicate connections"

-> Webhook timing

"No audio/video data"

-> Media Types - Check configuration

"How do I implement manually?"

-> Manual WebSocket

"What message types exist?"

-> Data Types

"How do I integrate AI?"

-> AI Integration

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Document Version

Based on Zoom RTMS SDK v1.x and official documentation as of 2026.

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Happy coding!

Remember: Start with SDK Quickstart for the fastest path, or Manual WebSocket if you need full control.