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Changelog

All notable user-facing changes to this repository are documented in this file.

The format is based on Keep a Changelog. This repository contains samples rather than a versioned package, so changes remain under Unreleased until a repository release is published.

[Unreleased]

Added

  • Added zoom_apps/zoom_decisions_compliance_risk_copilot_js, a Zoom App that uses typed decisions to review live transcripts for compliance gaps and risk.

Changed

  • RTMS media connections now use split mode exclusively. Clients open a separate data connection for each required media type and send one concrete media_type per data handshake: audio (1), video (2), screen share (4), transcript (8), or chat (16).
  • Generic boilerplates now require an explicit comma-separated selection such as RTMS_MEDIA_TYPES=audio,video,transcript.
  • Purpose-built samples select only the media required for their use case.
  • Audio/video streaming samples now manage separate audio and video sockets, including independent keep-alive and reconnection handling.
  • JavaScript and Python shared libraries now require an explicit media selection instead of implicitly requesting every media type.

Deprecated

  • Unified media socket mode and aggregate data handshakes using media_type: 32 are deprecated and must not be used by new clients.

Removed

  • Removed server_urls.all usage from runnable samples.
  • Removed the MEDIA_SOCKET_CONNECTION_MODE and MEDIA_TYPES_FLAG environment settings from generic sample configuration.
  • Removed the JavaScript MEDIA.ALL, Python MediaType.ALL, and JavaScript PRESETS.FULL_MEDIA public helpers.

Migration

Replace legacy configuration:

MEDIA_SOCKET_CONNECTION_MODE=unified
MEDIA_TYPES_FLAG=32

with explicit media names:

RTMS_MEDIA_TYPES=audio,video,transcript

For direct WebSocket implementations, use the matching URL and concrete value for each connection. For example, connect to server_urls.audio with media_type: 1, then connect separately to server_urls.transcript with media_type: 8.