zoom-rtms

用于实时处理 Zoom 音视频、聊天、转录与客服语音流的工作流技能

来源

GitHub

更新于

2026-06-07

// 安全评估低风险

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正在进行安全审计…

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代码执行
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// 安装

复制安装指令，让 AI 自动完成配置 · 推荐新手

请帮我安装 askskill 上的 "zoom-rtms" 技能：
1. 下载 https://raw.githubusercontent.com/anthropics/knowledge-work-plugins/main/partner-built/zoom-plugin/skills/rtms/SKILL.md
2. 保存为 ~/.claude/skills/rtms/SKILL.md
3. 装好后重载技能，告诉我可以用了

// 下载

下载 SKILL.md机读安装清单 ↗

// 用法示例

会议实时转录与摘要

输入

接入 Zoom 实时会议流，持续处理音频与聊天内容，生成带时间戳的中文转录，并每 5 分钟输出一次会议摘要和待办事项。

预期产出

持续更新的转录文本，以及分阶段会议摘要、决策点和待办清单。

客服语音流质检

输入

处理 Zoom 联络中心的实时语音流，识别客户情绪、关键问题与违规话术，并在检测到高风险对话时立即告警。

预期产出

实时风险告警、情绪分析结果、问题标签，以及可审计的对话片段记录。

屏幕共享内容提取

输入

在 Zoom 屏幕共享会话中实时分析画面与语音，提取演示主题、页面切换要点和关键指标，并同步生成简报草稿。

预期产出

结构化的演示要点、关键数据摘录，以及可直接复用的简报初稿。

// 文档

Zoom Realtime Media Streams (RTMS)

Background reference for live Zoom media pipelines. Prefer build-zoom-bot first, then use this skill for stream types, capabilities, and RTMS-specific implementation constraints.

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:

Connection Architecture - Two-phase WebSocket design
SDK Quickstart - Fastest way to receive media (recommended)
Manual WebSocket - Full protocol control without SDK
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
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`	General App
Webinars	`webinar.rtms_started` / `webinar.rtms_stopped`	`meeting_uuid` (same!)	General App
Video SDK	`session.rtms_started` / `session.rtms_stopped`	`session_id`	Video SDK App
Zoom Contact Center Voice	Product-specific RTMS/ZCC Voice events	Product-specific stream/session identifiers	Contact Center / approved RTMS integration

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.