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Zoom Rtms

OpenAIOfficialCreated by ZoomDistributed by OpenAI

  • Skill
  • Tool integration
  • Automation and integrations
  • MIT
  • No issues found

From the author

Use when using RTMS.
— the description by Zoom

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

  1. Download the skill as a ZIP:

    https://mp-staging.autogineer.com/agent-skills/openai/zoom-rtms/download
  2. Unzip 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

  1. Confirm the source surface: Meetings RTMS, Contact Center voice media, or another documented RTMS stream.

  2. Decide whether RTMS is sufficient or whether a Meeting SDK bot is required for participant identity, meeting controls, or UI-visible behavior.

  3. Implement the WebSocket lifecycle first: event subscription, connection validation, media start, heartbeat, reconnect, and shutdown.

  4. Process media types intentionally: audio, video, screen share, chat, live transcript, and metadata have different payload and timing constraints.

  5. Design downstream pipelines for buffering, transcription, AI analysis, recording, or tool invocation.

  6. Debug by isolating app setup, webhook/event delivery, stream authorization, network connectivity, and media payload decoding.

References

Files

examples/sdk-quickstart.md

SDK Quickstart

The fastest way to receive RTMS media using the official @zoom/rtms SDK.

Installation

# Requires Node.js 20.3.0+ (24 LTS recommended)
npm install @zoom/rtms express

Environment Setup

# .env
ZM_RTMS_CLIENT=your_client_id
ZM_RTMS_SECRET=your_client_secret

Multi-Product Support

The SDK accepts both meeting_uuid (meetings/webinars) and session_id (Video SDK) via client.join(payload) transparently. You only need to handle the different webhook event names -- the rest of the protocol is identical.

// These constants cover all RTMS products
const RTMS_EVENTS = ["meeting.rtms_started", "webinar.rtms_started", "session.rtms_started"];
const RTMS_STOP_EVENTS = ["meeting.rtms_stopped", "webinar.rtms_stopped", "session.rtms_stopped"];

Minimal Example

import rtms from "@zoom/rtms";

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

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

  const client = new rtms.Client();

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

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

Complete Example with All Media Types

import rtms from "@zoom/rtms";
import fs from 'fs';

const RTMS_EVENTS = ["meeting.rtms_started", "webinar.rtms_started", "session.rtms_started"];
const RTMS_STOP_EVENTS = ["meeting.rtms_stopped", "webinar.rtms_stopped", "session.rtms_stopped"];

const clients = new Map();

rtms.onWebhookEvent(({ event, payload }) => {
  const streamId = payload?.rtms_stream_id;

  // Handle session end (meetings, webinars, and Video SDK)
  if (RTMS_STOP_EVENTS.includes(event)) {
    const client = clients.get(streamId);
    if (client) {
      client.leave();
      clients.delete(streamId);
    }
    return;
  }

  if (!RTMS_EVENTS.includes(event)) return;

  // Prevent duplicate connections
  if (clients.has(streamId)) {
    console.log('Already connected to this stream');
    return;
  }

  const client = new rtms.Client();
  clients.set(streamId, client);

  // Join confirmation
  client.onJoinConfirm((reason) => {
    console.log(`Joined meeting: ${reason}`);
  });

  // Audio data
  client.onAudioData((buffer, timestamp, metadata) => {
    console.log(`Audio from ${metadata.userName}: ${buffer.length} bytes`);
    // Save to file, send to transcription service, etc.
  });

  // Video data
  client.onVideoData((buffer, timestamp, trackId, metadata) => {
    console.log(`Video from ${metadata.userName}: ${buffer.length} bytes`);
    // H.264 NAL units or JPG/PNG frames
  });

  // Transcript (real-time speech-to-text from Zoom)
  client.onTranscriptData((buffer, timestamp, metadata) => {
    const text = buffer.toString('utf8');
    console.log(`[${metadata.userName}]: ${text}`);
  });

  // Chat messages
  client.onChatData((buffer, timestamp, metadata) => {
    const text = buffer.toString('utf8');
    console.log(`[Chat] ${metadata.userName}: ${text}`);
  });

  // Screen share
  client.onShareData((buffer, timestamp, metadata) => {
    console.log(`Screen share from ${metadata.userName}: ${buffer.length} bytes`);
  });

  // Participant events
  client.onParticipantEvent((event, timestamp, participants) => {
    participants.forEach(p => {
      console.log(`Participant ${event}: ${p.userName}`);
    });
  });

  // Active speaker changed
  client.onActiveSpeakerEvent((timestamp, userId, userName) => {
    console.log(`Active speaker: ${userName}`);
  });

  // Screen sharing started/stopped
  client.onSharingEvent((event, timestamp, userId, userName) => {
    console.log(`Sharing ${event}: ${userName}`);
  });

  // Session ended
  client.onLeave((reason) => {
    console.log(`Left meeting: ${reason}`);
    clients.delete(streamId);
  });

  // Join the meeting
  client.join(payload);
});

Configuring Audio Parameters

import rtms from "@zoom/rtms";

const client = new rtms.Client();

// Set audio parameters before joining
client.setAudioParams({
  contentType: 2,    // RAW_AUDIO
  codec: 4,          // OPUS (default)
  sampleRate: 3,     // 48kHz
  channel: 2,        // Stereo (only with OPUS)
  dataOpt: 2,        // AUDIO_MULTI_STREAMS (per-participant)
  duration: 20,      // 20ms chunks
  frameSize: 960     // Samples per frame
});

client.join(payload);
Audio Parameter Options
ParameterOptions
contentType1=RTP, 2=RAW_AUDIO
codec1=L16 (PCM), 2=G.711, 3=G.722, 4=OPUS
sampleRate0=8kHz, 1=16kHz, 2=32kHz, 3=48kHz
channel1=Mono, 2=Stereo (OPUS only!)
dataOpt1=Mixed stream, 2=Multi-streams (per participant)
durationChunk size in ms (multiple of 20, max 1000)

Configuring Video Parameters

client.setVideoParams({
  contentType: 3,    // RAW_VIDEO
  codec: 7,          // H.264
  resolution: 2,     // HD (720p)
  fps: 25,
  dataOpt: 3         // Single active speaker
});
Video Parameter Options
ParameterOptions
codec5=JPG, 6=PNG, 7=H.264
resolution1=SD (480p), 2=HD (720p), 3=FHD (1080p), 4=QHD (1440p)
fps1-30 (JPG/PNG max 5, H.264 max 30)
dataOpt3=Single active speaker

With Express Webhook Handler

import rtms from "@zoom/rtms";
import express from "express";

const app = express();
app.use(express.json());

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

// Use SDK's webhook handler
app.post('/webhook', rtms.createWebhookHandler(({ event, payload }) => {
  if (!RTMS_EVENTS.includes(event)) return;

  const client = new rtms.Client();

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

  client.join(payload);
}, '/webhook'));

app.listen(3000, () => {
  console.log('Server running on port 3000');
});

Class-Based Approach (Multiple Connections)

For applications needing multiple concurrent connections:

import rtms from "@zoom/rtms";

// Initialize SDK once
rtms.Client.initialize();

// Create multiple clients
const client1 = new rtms.Client();
const client2 = new rtms.Client();

client1.onTranscriptData((data, ts, meta) => {
  console.log(`[Meeting 1] ${meta.userName}: ${data.toString('utf8')}`);
});

client2.onTranscriptData((data, ts, meta) => {
  console.log(`[Meeting 2] ${meta.userName}: ${data.toString('utf8')}`);
});

// Join different meetings
client1.join(meeting1Payload);
client2.join(meeting2Payload);

Error Handling

client.onJoinConfirm((reason) => {
  if (reason !== 0) {
    console.error(`Join failed with reason: ${reason}`);
    // Handle error
  }
});

client.onLeave((reason) => {
  console.log(`Left meeting with reason: ${reason}`);

  // Cleanup
  clients.delete(streamId);

  // Optionally reconnect
  if (reason === /* unexpected disconnect */) {
    setTimeout(() => reconnect(), 2000);
  }
});

Python SDK

import rtms
from dotenv import load_dotenv

load_dotenv()

RTMS_EVENTS = ['meeting.rtms_started', 'webinar.rtms_started', 'session.rtms_started']
RTMS_STOP_EVENTS = ['meeting.rtms_stopped', 'webinar.rtms_stopped', 'session.rtms_stopped']

clients = {}

@rtms.onWebhookEvent
def handle_webhook(webhook):
    event = webhook.get('event')
    payload = webhook.get('payload', {})
    stream_id = payload.get('rtms_stream_id')

    if event in RTMS_STOP_EVENTS:
        if stream_id in clients:
            clients[stream_id].leave()
            del clients[stream_id]
        return

    if event not in RTMS_EVENTS:
        return

    client = rtms.Client()
    clients[stream_id] = client

    @client.onTranscriptData
    def on_transcript(data, size, timestamp, metadata):
        text = data.decode('utf-8')
        print(f'[{metadata.userName}]: {text}')

    @client.onJoinConfirm
    def on_join(reason):
        print(f'Joined: {reason}')

    @client.onLeave
    def on_leave(reason):
        print(f'Left: {reason}')

    # SDK accepts both meeting_uuid and session_id transparently
    client.join(payload)

# Main loop
if __name__ == '__main__':
    print('Webhook server running...')
    rtms.run()

Environment Variables Reference

# Required
ZM_RTMS_CLIENT=your_client_id
ZM_RTMS_SECRET=your_client_secret

# Optional
ZM_RTMS_PORT=8080           # Webhook server port
ZM_RTMS_PATH=/webhook       # Webhook endpoint path

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

Common Issues

IssueSolution
Segmentation faultUpgrade to Node.js 20.3.0+ (24 LTS recommended)
Audio metadata missing userIdUse onActiveSpeakerEvent for speaker identification with mixed stream
Video params ignoredCall setVideoParams BEFORE setAudioParams

Next Steps

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

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Download ZIP

The ZIP holds the skill’s files, its licence files and an ATTRIBUTION.md. Download zoom-rtms.zip