Zoom Rtms
OpenAIOfficialCreated by ZoomDistributed by OpenAI
- Skill
- Tool integration
- Automation and integrations
- MIT
- No issues found
AutoGineer summary
Written by AutoGineer from the skill’s own files, not by the author
The zoom-rtms skill provides the necessary protocols and architectural guidance for integrating Zoom Realtime Media Streams. It covers the WebSocket-based connection lifecycle, including signaling and media handshakes, heartbeat management, and reconnection strategies. The skill supports processing various media types such as audio, video, screen sharing, transcripts, and chat, while providing reference data for message types and event enums.
What it does
- Establishes WebSocket connections for signaling and media streams.
- Handles media handshake requests for audio, video, screen share, transcript, and chat data.
- Provides reference tables for RTMS message types and event enums.
- Defines mandatory heartbeat and reconnection logic for stream stability.
- Supports configuration for transcript language identification.
- Outlines the process for extracting stream IDs and generating HMAC-SHA256 signatures.
When to use it
- When you need to ingest raw meeting or contact center media into a backend pipeline.
- When building services for transcription, AI analysis, or recording of Zoom sessions.
- When you need to subscribe to specific meeting events or participant media streams.
- When you require programmatic access to media data without a visible participant bot.
Advantages
- Provides a structured approach to managing the complex WebSocket lifecycle.
- Includes comprehensive references for message types and event handling.
- Offers clear guidance on signature generation and security requirements.
- Enables granular control over media subscription and transcript settings.
Usage examples
- “How do I initiate a connection to an RTMS stream?”
- The agent will guide you through receiving the rtms_started webhook, generating an HMAC-SHA256 signature, and performing the signaling and media handshakes.
- “What is the correct way to handle heartbeats in RTMS?”
- The agent will explain that you must respond to msg_type 12 (Keep Alive Request) with msg_type 13 (Keep Alive Response) to maintain the connection.
Install
Download the skill as a ZIP:
https://mp-staging.autogineer.com/agent-skills/openai/zoom-rtms/downloadUnzip it into the cross-client skills folder of your project:
mkdir -p .agents/skills && unzip zoom-rtms.zip -d .agents/skills
Documentation by Zoom
Zoom Realtime Media Streams
Use this skill when the integration needs real-time meeting or contact-center media as a backend stream. If the workflow needs a visible participant bot, compare against build-zoom-bot and zoom-meeting-sdk-linux before choosing RTMS.
Workflow
Confirm the source surface: Meetings RTMS, Contact Center voice media, or another documented RTMS stream.
Decide whether RTMS is sufficient or whether a Meeting SDK bot is required for participant identity, meeting controls, or UI-visible behavior.
Implement the WebSocket lifecycle first: event subscription, connection validation, media start, heartbeat, reconnect, and shutdown.
Process media types intentionally: audio, video, screen share, chat, live transcript, and metadata have different payload and timing constraints.
Design downstream pipelines for buffering, transcription, AI analysis, recording, or tool invocation.
Debug by isolating app setup, webhook/event delivery, stream authorization, network connectivity, and media payload decoding.
References
Full preserved guide: references/full-guide.md
Connection architecture: concepts/connection-architecture.md
Lifecycle flow: concepts/lifecycle-flow.md
Media types: references/media-types.md
Data types: references/data-types.md
RTMS bot example: examples/rtms-bot.md
Common issues: troubleshooting/common-issues.md
Files
- ATTRIBUTION.mdAdded by AutoGineer860 B
- LICENSELicence file added by AutoGineer1.1 KiB
- RUNBOOK.md2.6 KiB
- SKILL.md1.6 KiB
- agents/openai.yaml140 B
- concepts/connection-architecture.md9.1 KiB
- concepts/lifecycle-flow.md13.4 KiB
- examples/ai-integration.md11.3 KiB
- examples/manual-websocket.md16.8 KiB
- examples/rtms-bot.md25.9 KiB
- examples/sdk-quickstart.md9.3 KiB
- references/connection.md8.0 KiB
- references/data-types.md13.7 KiB
- references/environment-variables.md1.3 KiB
- references/full-guide.md22.9 KiB
- references/media-types.md6.0 KiB
- references/quickstart.md6.6 KiB
- references/webhooks.md7.4 KiB
- troubleshooting/common-issues.md10.4 KiB
concepts/connection-architecture.md
RTMS Connection Architecture
RTMS uses a two-phase WebSocket design to separate control plane from data plane.
Overview
Multi-Product Note: The two-phase WebSocket design described here is identical for all RTMS products (meetings, webinars, and Video SDK sessions). The only difference is the initial webhook event name and payload ID field. Once connected, the signaling and media protocols are the same.
┌─────────────────────────────────────────────────────────────┐
│ Zoom Meeting │
└─────────────────────────────────────────────────────────────┘
│
▼
┌─────────────────────────────────────────────────────────────┐
│ Zoom RTMS Backend │
│ ┌─────────────────────┐ ┌─────────────────────────────┐ │
│ │ Signaling Server │ │ Media Server │ │
│ │ (Control Plane) │ │ (Data Plane) │ │
│ └──────────┬──────────┘ └──────────────┬──────────────┘ │
└─────────────┼───────────────────────────────┼───────────────┘
│ │
▼ ▼
┌─────────────────────────────────────────────────────────────┐
│ Your Server │
│ ┌─────────────────────┐ ┌─────────────────────────────┐ │
│ │ Signaling Socket │ │ Media Socket │ │
│ │ - Handshake │ │ - Audio data │ │
│ │ - Start/Stop │ │ - Video data │ │
│ │ - Heartbeat │ │ - Transcript │ │
│ └─────────────────────┘ └─────────────────────────────┘ │
└─────────────────────────────────────────────────────────────┘Two-Phase Design
Phase 1: Signaling WebSocket (Control Plane)
Purpose: Authentication, session control, heartbeats
| Responsibility | Description |
|---|---|
| Authentication | Validate signature, establish session |
| Media Server Discovery | Returns media server URL in handshake response |
| Stream Control | Start/stop streaming commands |
| Heartbeat | Keep connection alive (msg_type 12/13) |
| Event Notifications | Participant join/leave, sharing start/stop |
URL Source: From server_urls in webhook payload
Message Flow:
Client Signaling Server
│ │
│──── Handshake Request (1) ────────>│
│<─── Handshake Response (2) ────────│ <- Contains media_server.server_urls
│ │
│──── Client Ready (7) ─────────────>│ <- After media handshake complete
│ │
│<─── Keep Alive Request (12) ───────│
│──── Keep Alive Response (13) ─────>│
│ │Phase 2: Media WebSocket (Data Plane)
Purpose: Actual audio, video, transcript, chat, screen share data
| Responsibility | Description |
|---|---|
| Media Configuration | Set audio/video parameters (codec, resolution, fps) |
| Media Streaming | Receive binary media data |
| Heartbeat | Keep connection alive (msg_type 12/13) |
URL Source: From signaling handshake response (media_server.server_urls.all)
Message Flow:
Client Media Server
│ │
│──── Media Handshake Request (3) ──>│ <- With media_params
│<─── Media Handshake Response (4) ──│
│ │
│<─── Audio Data (14) ───────────────│
│<─── Video Data (15) ───────────────│
│<─── Screen Share Data (16) ────────│
│<─── Transcript Data (17) ──────────│
│<─── Chat Data (18) ────────────────│
│ │
│<─── Keep Alive Request (12) ───────│
│──── Keep Alive Response (13) ─────>│
│ │Why Two Connections?
| Benefit | Explanation |
|---|---|
| Separation of Concerns | Control logic doesn't interfere with media streaming |
| Independent Scaling | Signaling and media servers scale differently |
| Fault Isolation | Media reconnection doesn't require re-auth |
| Split Mode Support | Each media type can have its own connection |
Connection Modes
Split Mode (Recommended)
Each media type gets its own dedicated WebSocket connection:
Signaling WS ─────┬───> Audio WS
├───> Video WS
├───> Transcript WS
└───> Screen Share WSAdvantages:
Independent reconnection per media type
Better reliability
Fault isolation
Unified Mode
One media WebSocket for all media types:
Signaling WS ─────> Media WS (all types)When to use:
Real-time audio+video muxing where sync matters
Simpler implementation for small projects
Signature Generation
Both signaling and media handshakes require HMAC-SHA256 signature:
// For meetings and webinars: use meeting_uuid
const message = `${clientId},${meetingUuid},${streamId}`;
// For Video SDK: use session_id
const message = `${clientId},${sessionId},${streamId}`;
// Generic approach: use whichever ID is present
const idValue = payload.meeting_uuid || payload.session_id;
const message = `${clientId},${idValue},${streamId}`;
const signature = crypto.createHmac('sha256', clientSecret)
.update(message)
.digest('hex');Important: Webinars use
meeting_uuid(notwebinar_uuid). Video SDK usessession_id.
Components:
clientId: OAuth Client ID (General App) or SDK Key (Video SDK App)meetingUuid/sessionId: From webhook payload (meeting_uuidfor meetings/webinars,session_idfor Video SDK)streamId: From webhook payload (rtms_stream_id)clientSecret: OAuth Client Secret (General App) or SDK Secret (Video SDK App)
Heartbeat Protocol
CRITICAL: Both connections require heartbeat responses.
When you receive msg_type: 12 (Keep Alive Request):
// Immediately respond with msg_type: 13
ws.send(JSON.stringify({
msg_type: 13,
timestamp: receivedMessage.timestamp
}));Timeout:
Signaling: ~60 seconds without heartbeat response
Media: ~65 seconds without heartbeat response
Failure to respond = connection closed!
Reconnection
RTMS does NOT auto-reconnect. You must implement:
ws.on('close', (code, reason) => {
console.log(`Connection closed: ${code} ${reason}`);
// Implement exponential backoff
setTimeout(() => {
reconnect();
}, retryDelay);
retryDelay = Math.min(retryDelay * 2, 30000);
});Timeouts:
| Connection | Reconnection Window |
|---|---|
| Signaling | 60 seconds |
| Media | 65 seconds |
Server URL Geo-Routing
Server URLs contain region codes:
| Code | Location |
|---|---|
sjc | San Jose, California |
iad | Washington DC |
sin | Singapore |
fra | Frankfurt, Germany |
syd | Sydney, Australia |
Example: wss://rtms-sjc1.zoom.us/...
For production, route to workers in the same region as the Zoom server for lower latency.
Next Steps
Lifecycle Flow - Complete webhook-to-streaming sequence
SDK Quickstart - SDK handles all this for you
Manual WebSocket - Full protocol implementation
Requirements and compatibility
Written for: OpenAI Codex
Works with: Claude Code, Claude apps, OpenAI Codex, GitHub Copilot, Cursor, Gemini CLI, Kiro, OpenCode, Goose, Amp, Roo Code, JetBrains Junie, Factory Droid, Devin (ex-Windsurf), Mistral Vibe, OpenHands, Letta
“Works with” means the tool reads the open Agent Skills format; a skill written for one tool may use features others lack.
Security
No issues found by our static scan· No issues found
A static scan is not a guarantee. Skills are third-party content: review a skill before you let an agent use it.
Attribution and licence
- Created by
- Zoom
- Publisher
- OpenAI (Official)
- Distributed by
- OpenAI
- Source
- openai/plugins · plugins/zoom/skills/rtms · commit d420d13
- Licence
- MIT License (redistribution allowed)
- LICENSE (added by AutoGineer from plugins/zoom/LICENSE)
- ATTRIBUTION.md (written by AutoGineer)
- Synced
AutoGineer is not affiliated with or endorsed by Zoom or OpenAI.
Download ZIP
The ZIP holds the skill’s files, its licence files and an ATTRIBUTION.md. Download zoom-rtms.zip