diff --git a/.github/workflows/main.yml b/.github/workflows/main.yml
index ffe8f25..7e2e4ab 100644
--- a/.github/workflows/main.yml
+++ b/.github/workflows/main.yml
@@ -196,6 +196,58 @@ jobs:
   # TEST JOBS - Test against pre-built artifacts
   # ============================================================================
 
+  # C++ unit tests — mock SDK replaces the binary; real headers downloaded via check-deps
+  test-cpp-linux:
+    name: Test C++ Core (Linux)
+    runs-on: ubuntu-latest
+
+    steps:
+    - name: Checkout code
+      uses: actions/checkout@v4
+
+    - name: Setup Node.js (for check-deps.js SDK download)
+      uses: actions/setup-node@v4
+      with:
+        node-version: '24.x'
+
+    - name: Install CMake
+      run: sudo apt-get install -y cmake
+
+    - name: Download SDK headers
+      env:
+        GITHUB_TOKEN: ${{ secrets.GITHUB_TOKEN }}
+      run: node scripts/check-deps.js
+
+    - name: Build and run C++ tests
+      run: |
+        cmake -B build/tests -DRTMS_BUILD_TESTS=ON -DCMAKE_BUILD_TYPE=Debug
+        cmake --build build/tests --target rtms_tests -j$(nproc)
+        cd build/tests && ctest --output-on-failure
+
+  test-cpp-macos:
+    name: Test C++ Core (macOS)
+    runs-on: macos-latest
+
+    steps:
+    - name: Checkout code
+      uses: actions/checkout@v4
+
+    - name: Setup Node.js (for check-deps.js SDK download)
+      uses: actions/setup-node@v4
+      with:
+        node-version: '24.x'
+
+    - name: Download SDK headers
+      env:
+        GITHUB_TOKEN: ${{ secrets.GITHUB_TOKEN }}
+      run: node scripts/check-deps.js
+
+    - name: Build and run C++ tests
+      run: |
+        cmake -B build/tests -DRTMS_BUILD_TESTS=ON -DCMAKE_BUILD_TYPE=Debug
+        cmake --build build/tests --target rtms_tests -j$(sysctl -n hw.logicalcpu)
+        cd build/tests && ctest --output-on-failure
+
   # Test Node.js SDK on Linux using pre-built prebuilds
   test-nodejs-linux:
     name: Test Node.js SDK on Linux (Node ${{ matrix.node-version }})
@@ -203,7 +255,7 @@ jobs:
     needs: [build-nodejs-linux]
     strategy:
       matrix:
-        node-version: ['20.3.0', '24.x']  # Test minimum (N-API v9) and latest LTS
+        node-version: ['22.x', '24.x']  # Test current LTS and latest LTS
       fail-fast: false  # Continue testing other versions if one fails
 
     steps:
@@ -224,8 +276,16 @@ jobs:
     - name: Install dependencies
       run: npm install
 
+    - name: Extract native prebuild
+      # install.js skips extraction when .git exists (dev mode detection).
+      # --force bypasses that check so the full install flow runs in CI.
+      run: node scripts/install.js --force
+
+    - name: Compile TypeScript
+      run: npx tsc
+
     - name: Run Node.js tests
-      run: npx jest tests/rtms.test.ts
+      run: npx jest tests/ts/rtms.test.ts tests/ts/rtms.wrapper.test.ts
 
   # Test Node.js SDK on macOS using pre-built prebuilds
   test-nodejs-macos:
@@ -251,8 +311,16 @@ jobs:
     - name: Install dependencies
       run: npm install
 
+    - name: Extract native prebuild
+      # Use install.js so macOS framework archives (.framework.tar.gz) are
+      # extracted after prebuild-install runs. --force bypasses dev-mode check.
+      run: node scripts/install.js --force
+
+    - name: Compile TypeScript
+      run: npx tsc
+
     - name: Run Node.js tests
-      run: npx jest tests/rtms.test.ts
+      run: npx jest tests/ts/rtms.test.ts tests/ts/rtms.wrapper.test.ts
 
   # Integration test: npm pack → install → load (validates end-user install flow)
   test-nodejs-integration-macos:
@@ -279,7 +347,7 @@ jobs:
       run: npm install
 
     - name: Run integration test (npm pack → install → load)
-      run: npx jest tests/pack-install.test.js --testTimeout=120000
+      run: npx jest tests/ts/pack-install.test.js --testTimeout=120000
 
   # Test Python SDK on Linux using pre-built wheels
   test-python-linux:
@@ -323,7 +391,7 @@ jobs:
       run: pip install pytest python-dotenv
 
     - name: Run Python tests
-      run: pytest tests/test_rtms.py -v
+      run: pytest tests/py/test_rtms.py -v
 
   # Test Python SDK on macOS using pre-built wheels
   test-python-macos:
@@ -367,7 +435,7 @@ jobs:
       run: pip install pytest python-dotenv
 
     - name: Run Python tests
-      run: pytest tests/test_rtms.py -v
+      run: pytest tests/py/test_rtms.py -v
 
   # ============================================================================
   # VERSION CHECK & PUBLISH TRIGGERS
@@ -377,7 +445,7 @@ jobs:
   check-version-change:
     name: Check for Version Changes
     runs-on: ubuntu-latest
-    needs: [test-nodejs-linux, test-nodejs-macos, test-nodejs-integration-macos, test-python-linux, test-python-macos]
+    needs: [test-cpp-linux, test-cpp-macos, test-nodejs-linux, test-nodejs-macos, test-nodejs-integration-macos, test-python-linux, test-python-macos]
     if: |
       success() &&
       github.event_name == 'push' &&
@@ -467,7 +535,7 @@ jobs:
   build-docs:
     name: Build Documentation
     runs-on: ubuntu-latest
-    needs: [test-nodejs-linux, test-nodejs-macos, test-python-linux, test-python-macos]
+    needs: [test-cpp-linux, test-cpp-macos, test-nodejs-linux, test-nodejs-macos, test-python-linux, test-python-macos]
     # Only run on main/master branches and if tests pass, or via workflow_dispatch
     if: |
       success() &&
diff --git a/.gitignore b/.gitignore
index e5b0646..8e2209b 100644
--- a/.gitignore
+++ b/.gitignore
@@ -12,6 +12,7 @@ __pycache__/
 ### Dev Only
 docs/*
 !docs/*.md
+spec.md
 
 ### Build
 dist
diff --git a/CHANGELOG.md b/CHANGELOG.md
index 68c78ab..c982b81 100644
--- a/CHANGELOG.md
+++ b/CHANGELOG.md
@@ -5,6 +5,75 @@ All notable changes to the Zoom RTMS SDK will be documented in this file.
 The format is based on [Keep a Changelog](https://keepachangelog.com/en/1.1.0/),
 and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0.html).
 
+## [1.1.0] - 2026-04-15
+
+### Added
+
+#### ZCC (Zoom Contact Center) Support
+- **`engagement_id` in `join()`**: ZCC voice engagements identify sessions with an `engagement_id` instead of `meeting_uuid`; `join()` now accepts `engagement_id` as an alternative identifier (all bindings)
+- **ZCC event constants**: `EVENT_CONSUMER_ANSWERED`, `EVENT_CONSUMER_END`, `EVENT_USER_ANSWERED`, `EVENT_USER_END`, `EVENT_USER_HOLD`, `EVENT_USER_UNHOLD` exported as flat module-level constants in both Node.js and Python
+
+#### Transcript Configuration
+- **`TranscriptParams`**: New parameter class for configuring transcript streams (language hint, content type, language detection)
+- **`TranscriptLanguage` constants**: Full set of language constants (`ENGLISH`, `SPANISH`, `JAPANESE`, `CHINESE_SIMPLIFIED`, and many more) for skipping auto-detection (~30 s delay) by hinting the source language
+- **`setTranscriptParams()`/`set_transcript_params()`**: New method on Client to apply transcript configuration before joining (Node.js and Python)
+
+#### HTTP Proxy Support
+- **`setProxy()`/`set_proxy()`**: New method on Client for configuring HTTP and HTTPS proxies for RTMS connections (Node.js and Python)
+
+#### Individual Video Streams
+- **`subscribeVideo()`/`subscribe_video()`**: Subscribe or unsubscribe per-participant video when using `VIDEO_SINGLE_INDIVIDUAL_STREAM` mode
+- **`onParticipantVideo()`/`on_participant_video()`**: Callback fired when a participant's video turns on or off
+- **`onVideoSubscribed()`/`on_video_subscribed()`**: Callback fired with the result of each `subscribeVideo()` call
+- **`EVENT_PARTICIPANT_VIDEO_ON` / `EVENT_PARTICIPANT_VIDEO_OFF`**: Exported as flat module-level constants in both Node.js and Python
+
+#### Python — Concurrency & Scaling
+- **`EventLoop` / `EventLoopPool`**: New primitives for fine-grained control over polling threads; `EventLoop` wraps a dedicated background thread, `EventLoopPool` balances clients across a pool of threads
+- **`run_async()`**: Drop-in async replacement for `run()` using `asyncio.sleep()` between polls; composes naturally with aiohttp, FastAPI, asyncpg
+- **Executor dispatch**: `Client(executor=...)` and `run(executor=...)` / `run_async(executor=...)` dispatch data callbacks to a `concurrent.futures.Executor`, keeping the poll loop fast for CPU-bound or I/O-heavy callbacks
+- **Async callback detection**: Coroutine callbacks are detected automatically and scheduled on the running event loop via `asyncio.run_coroutine_threadsafe`
+- **GIL release in `poll()`**: The C++ poll call now releases the GIL, allowing other Python threads (e.g. the webhook HTTP server) to run freely during polling
+- **Context manager support**: `with rtms.Client() as client:` guarantees `leave()` is called on exit
+- **Lazy `alloc()` / thread affinity**: `Client()` construction is safe from any thread; the C SDK handle is allocated lazily on the EventLoop's own thread to satisfy the C SDK's thread-affinity constraint
+- **Snake_case aliases**: All `camelCase` methods have `snake_case` aliases (`on_audio_data`, `set_audio_params`, `subscribe_event`, etc.)
+
+#### Metadata & AI Interpreter Fields
+- **Full metadata fields**: `Metadata` now exposes `userId`, `userName`, `startTs`, `endTs` in both bindings
+- **`AiInterpreter` fields**: `Metadata` now exposes `aiInterpreter` with language pair targets from the AI language interpreter SDK feature
+
+#### Constants & Enums
+- **`IntEnum` for Python codec/rate/option types**: `AudioCodec`, `VideoCodec`, `AudioSampleRate`, `AudioChannel`, `DataOption` are now proper Python `IntEnum` values (comparisons with integers still work)
+- **`DataOption`**: Unified stream delivery mode enum covering both audio (`AUDIO_MIXED_STREAM`, `AUDIO_MULTI_STREAMS`) and video (`VIDEO_SINGLE_ACTIVE_STREAM`, `VIDEO_SINGLE_INDIVIDUAL_STREAM`, `VIDEO_MIXED_GALLERY_VIEW`) options — mirrors the C++ `MEDIA_DATA_OPTION` enum directly; `AudioDataOption` and `VideoDataOption` kept as backward-compat aliases
+- **Bidirectional param aliases**: All four param structs (`AudioParams`, `VideoParams`, `DeskshareParams`, `TranscriptParams`) now accept both `snake_case` and `camelCase` field names so existing code continues to work regardless of convention
+
+### Fixed
+
+- **`setOnParticipantVideo` missing auto-subscription**: Registering the participant video callback now automatically subscribes to `EVENT_PARTICIPANT_VIDEO_ON` and `EVENT_PARTICIPANT_VIDEO_OFF` events, matching the pattern used by `setOnUserUpdate` ([#108](https://github.com/zoom/rtms/issues/108))
+- **`config()` called before `open()`**: Video/audio/transcript params are now applied after the SDK session is open; calling `setVideoParams()` before `join()` no longer fails silently
+- **`AiInterpreter.target_size` out-of-bounds**: Guard against uninitialized or negative `target_size` values from the C SDK to prevent array over-reads
+- **`_run_executor` undefined**: Python's `_wrap_callback` referenced a module-level `_run_executor` that was never initialised; added the initialisation and wired the `executor` kwarg through `run()` / `run_async()`
+- **Redundant `RTMS_` prefix on C++ enum names**: All enum classes inside the `rtms` namespace drop the redundant prefix (`RTMS_EVENT_TYPE` → `EVENT_TYPE`, `RTMS_SESSION_STATE` → `SESSION_STATE`, etc.) — internal refactor, no public API change
+- **Spurious configure warnings on leave**: `updateMediaConfiguration` was called during callback teardown after the session was already closed, printing 4 `"Failed to update media configuration"` warnings on every clean leave; suppressed by tracking `sdk_opened_` state and calling `markClosed()` before stopping callbacks
+- **SIGSEGV on meeting end (Python)**: Race condition between the EventLoop thread (inside `poll()`) and the webhook thread (calling `release()`) caused a use-after-free crash with exit status 139; fixed with `poll_mutex_` — `poll()` releases the GIL before acquiring the mutex so neither thread can deadlock the other
+
+### Changed
+
+- **Protocol enums moved to `rtms.h`**: `EVENT_TYPE`, `ZCC_VOICE_EVENT_TYPE`, `SESSION_STATE`, `STREAM_STATE`, `MESSAGE_TYPE`, `STOP_REASON`, and `TRANSCRIPT_LANGUAGE` are now defined in the shared C++ header, making them automatically available to both Node.js and Python bindings
+
+### Documentation
+
+- **Node.js API reference**: New "Media Configuration" section covering `setVideoParams`, `setAudioParams`, `setDeskshareParams`, `setTranscriptParams` with codec/resolution/option constants; updated "Individual Video Streams" section
+- **Python API reference**: New "Media Callbacks", "Media Configuration", and "Individual Video Streams" sections; asyncio integration and executor dispatch examples; scaling strategy guide (4-layer progression from single-process to multi-process)
+- **ZCC product guide**: Added Zoom Contact Center as a supported product with webhook event names and `engagement_id` usage
+
+### Testing
+
+- **C++ unit tests** (70 tests): Full mock-SDK test suite covering Client lifecycle, session/user events, media callbacks, proxy, transcript params, individual video subscriptions, metadata construction, and `AiInterpreter` bounds guarding
+- **Python tests** (122 tests): Coverage for asyncio, executor dispatch, context manager, GIL release, ZCC `engagement_id` routing, individual video methods, transcript params, and all new constants
+- **Node.js wrapper tests** (98 tests): Integration tests against the real built ESM module covering all Client methods, callbacks, event subscriptions, constants, and utility functions
+- **Test reorganisation**: Tests moved from `tests/` root into `tests/ts/`, `tests/py/`, `tests/cpp/` subdirectories for clarity
+- **CI test path fixes**: CI/CD workflows updated to reference new test locations; C++ unit tests added as dedicated `test-cpp-linux` and `test-cpp-macos` jobs using the mock SDK (no real Zoom SDK binary required in CI)
+
 ## [1.0.3] - 2026-02-17
 
 ### Fixed
diff --git a/CMakeLists.txt b/CMakeLists.txt
index e4e6988..9193dce 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -1,5 +1,5 @@
 cmake_minimum_required(VERSION 3.25.1)
-project(rtms VERSION 0.0.1 LANGUAGES CXX)
+project(rtms VERSION 1.1.0 LANGUAGES CXX)
 
 # ===== Build settings =====
 set(CMAKE_CXX_STANDARD 20)
@@ -57,7 +57,7 @@ set(RTMS_SEARCH_PATHS
 
 # ===== Locate library and headers =====
 find_path(RTMS_HEADER_DIR
-  NAMES rtms_csdk.h rtms_common.h
+  NAMES rtms_sdk.h rtms_common.h
   PATHS ${RTMS_INCLUDE_DIR} ${RTMS_SEARCH_PATHS}
   DOC "RTMS SDK headers"
 )
@@ -68,8 +68,13 @@ find_library(RTMS_LIBRARY
   DOC "RTMS SDK library"
 )
 
-# Error if not found
-if(NOT RTMS_LIBRARY)
+# ===== Test-only build option =====
+# When RTMS_BUILD_TESTS=ON the SDK binary is replaced by tests/mock_sdk.cpp,
+# so the real library is not required.
+option(RTMS_BUILD_TESTS "Build C++ unit tests using the mock SDK (no real SDK binary needed)" OFF)
+
+# Error if not found (skip library check for pure test builds)
+if(NOT RTMS_LIBRARY AND NOT RTMS_BUILD_TESTS)
   message(FATAL_ERROR "RTMS SDK library not found. Please install it or specify its location.")
 endif()
 
@@ -297,3 +302,41 @@ elseif(DEFINED SKBUILD_PROJECT_NAME)
 elseif(GO)
   message(STATUS "Go bindings not yet implemented")
 endif()
+
+# ============================================================================
+# C++ Unit Tests
+# Uses mock_sdk.cpp in place of the real Zoom SDK binary.
+# Configure with: cmake -B build/tests -DRTMS_BUILD_TESTS=ON
+# ============================================================================
+if(RTMS_BUILD_TESTS)
+  message(STATUS "Configuring C++ unit tests (mock SDK)")
+
+  include(FetchContent)
+  FetchContent_Declare(
+    Catch2
+    GIT_REPOSITORY https://github.com/catchorg/Catch2.git
+    GIT_TAG        v3.5.3
+    GIT_SHALLOW    TRUE
+  )
+  FetchContent_MakeAvailable(Catch2)
+
+  add_executable(rtms_tests
+    "${RTMS_SOURCE_DIR}/rtms.cpp"
+    "${CMAKE_SOURCE_DIR}/tests/cpp/mock_sdk.cpp"
+    "${CMAKE_SOURCE_DIR}/tests/cpp/test_cpp_wrapper.cpp"
+  )
+
+  target_include_directories(rtms_tests PRIVATE
+    ${RTMS_HEADER_DIR}
+    ${RTMS_SOURCE_DIR}
+    "${CMAKE_SOURCE_DIR}/tests/cpp"
+  )
+
+  target_compile_features(rtms_tests PRIVATE cxx_std_20)
+  target_link_libraries(rtms_tests PRIVATE Catch2::Catch2WithMain)
+
+  include(CTest)
+  list(APPEND CMAKE_MODULE_PATH ${catch2_SOURCE_DIR}/extras)
+  include(Catch)
+  catch_discover_tests(rtms_tests)
+endif()
diff --git a/Dockerfile b/Dockerfile
index 7c50fec..41dce1e 100644
--- a/Dockerfile
+++ b/Dockerfile
@@ -31,31 +31,40 @@ RUN apt update && apt install -y \
 # Install nvm (Node Version Manager) and Node.js
 ENV NVM_DIR="/root/.nvm"
 ENV NODE_VERSION="24"
-RUN curl -o- https://raw.githubusercontent.com/nvm-sh/nvm/v0.40.3/install.sh | bash \
-    && . "$NVM_DIR/nvm.sh" \
-    && nvm install $NODE_VERSION \
-    && nvm alias default $NODE_VERSION \
-    && nvm use default \
-    && npm config set update-notifier false \
-    && ln -sf "$NVM_DIR/versions/node/$(nvm version default)" /usr/local/node
-
-# Add Node.js to PATH
 ENV PATH="/usr/local/node/bin:$PATH"
 
+RUN if [ "$TARGET" = "js" ] || [ "$TARGET" = "all" ]; then \
+        echo "Installing NVM..." \
+        && curl -o- https://raw.githubusercontent.com/nvm-sh/nvm/v0.40.3/install.sh | bash \
+        && . "$NVM_DIR/nvm.sh" \
+        && nvm install $NODE_VERSION \
+        && nvm alias default $NODE_VERSION \
+        && nvm use default \
+        && npm config set update-notifier false \
+        && ln -sf "$NVM_DIR/versions/node/$(nvm version default)" /usr/local/node; \
+    fi
+
+
 # Install pyenv (Python Version Manager)
 ENV PYENV_ROOT="/root/.pyenv"
 ENV PYTHON_VERSION="3.13"
-RUN git clone https://github.com/pyenv/pyenv.git $PYENV_ROOT \
-    && cd $PYENV_ROOT && src/configure && make -C src \
-    && echo 'export PYENV_ROOT="$HOME/.pyenv"' >> ~/.bashrc \
-    && echo 'export PATH="$PYENV_ROOT/bin:$PATH"' >> ~/.bashrc \
-    && echo 'eval "$(pyenv init -)"' >> ~/.bashrc
+ENV PATH="$PYENV_ROOT/shims:$PYENV_ROOT/bin:$PATH"
+RUN if [ "$TARGET" = "py" ] || [ "$TARGET" = "all" ]; then \
+        echo "Installing pyenv version..." \ 
+        && git clone https://github.com/pyenv/pyenv.git $PYENV_ROOT \
+        && cd $PYENV_ROOT && src/configure && make -C src \
+        && echo 'export PYENV_ROOT="$HOME/.pyenv"' >> ~/.bashrc \
+        && echo 'export PATH="$PYENV_ROOT/bin:$PATH"' >> ~/.bashrc \
+        && echo 'eval "$(pyenv init -)"' >> ~/.bashrc; \
+    fi
 
 # Add pyenv to PATH and install Python
-ENV PATH="$PYENV_ROOT/shims:$PYENV_ROOT/bin:$PATH"
-RUN pyenv install $PYTHON_VERSION \
-    && pyenv global $PYTHON_VERSION \
-    && python --version
+RUN if [ "$TARGET" = "py" ] || [ "$TARGET" = "all" ]; then \
+        echo "Installing pyenv version..." \ 
+        && pyenv install $PYTHON_VERSION \
+        && pyenv global $PYTHON_VERSION \
+        && python --version; \
+    fi
 
 # Install Task (go-task)
 RUN sh -c "$(curl --location https://taskfile.dev/install.sh)" -- -d -b /usr/local/bin
@@ -66,10 +75,11 @@ RUN if [ "$TARGET" = "js" ] || [ "$TARGET" = "all" ]; then \
         npm install -g prebuild; \
     fi
 
+COPY requirements-dev.txt .
 RUN if [ "$TARGET" = "py" ] || [ "$TARGET" = "all" ]; then \
         echo "Installing Python build tools..." && \
         pip install --no-cache-dir --upgrade pip && \
-        pip install --no-cache-dir build "pybind11[global]" python-dotenv pdoc3 auditwheel twine; \
+        pip install --no-cache-dir -r requirements-dev.txt; \
     fi
 
 RUN if [ "$TARGET" = "go" ] || [ "$TARGET" = "all" ]; then \
diff --git a/README.md b/README.md
index 7a5c9fc..dff5d57 100644
--- a/README.md
+++ b/README.md
@@ -13,6 +13,7 @@ The RTMS SDK works with multiple Zoom products:
 - **[Zoom Meetings](examples/meetings.md)** - Real-time streams from Zoom Meetings
 - **[Zoom Webinars](examples/webinars.md)** - Broadcast-quality streams from Zoom Webinars
 - **[Zoom Video SDK](examples/videosdk.md)** - Custom video experiences with RTMS access
+- **[Zoom Contact Center](examples/zcc.md)** - Real-time streams from Zoom Contact Center voice engagements
 
 See [examples/](examples/) for complete guides and code samples.
 
@@ -40,9 +41,9 @@ The RTMS SDK allows developers to:
 
 ### Node.js
 
-**⚠️ Requirements: Node.js >= 20.3.0 (Node.js 24 LTS recommended)**
+**⚠️ Requirements: Node.js >= 22.0.0 (Node.js 24 LTS recommended)**
 
-The RTMS SDK uses N-API versions 9 and 10, which require Node.js 20.3.0 or higher.
+The RTMS SDK uses N-API versions 9 and 10, which require Node.js 22.0.0 or higher.
 
 ```bash
 # Check your Node.js version
@@ -52,24 +53,21 @@ node --version
 npm install @zoom/rtms
 ```
 
+#### Using NVM
 **If you're using an older version of Node.js:**
 ```bash
 # Using nvm (recommended)
 nvm install 24       # Install Node.js 24 LTS (recommended)
 nvm use 24
 
-# Or install Node.js 20 LTS (minimum)
-nvm install 20
-nvm use 20
+# Or install Node.js 22 LTS (minimum supported)
+nvm install 22
+nvm use 22
 
 # Reinstall the package
 npm install @zoom/rtms
 ```
 
-**Download Node.js:** https://nodejs.org/
-
-The Node.js package provides both class-based and singleton APIs for connecting to RTMS streams.
-
 ### Python
 
 **⚠️ Requirements: Python >= 3.10 (Python 3.10, 3.11, 3.12, or 3.13)**
@@ -95,59 +93,37 @@ pyenv local 3.12
 # macOS: brew install python@3.12
 ```
 
-**Download Python:** https://www.python.org/downloads/
+## Usage
 
-The Python package provides a Pythonic decorator-based API with full feature parity to Node.js.
+Regardless of the language that you use the RTMS SDK follows the same basic steps: 
 
-## For Contributors
+ - Receive and validate the webhook event
+ - Create an RTMS SDK Client 
+ - Assign data callbacks to that client
+ - Join the meeting with the event payload
 
-**This project uses [Task](https://taskfile.dev) (go-task) for development builds and testing.**
+### Configure
 
-If you're an **end user** installing via npm or pip, you don't need Task - the installation will work automatically using prebuilt binaries.
+All SDK languages read from the environment or a `.env` file:
 
-If you're a **contributor** building from source, you'll need to install Task:
-
-**macOS:**
 ```bash
-brew install go-task
-```
-
-**Linux:**
-```bash
-sh -c "$(curl --location https://taskfile.dev/install.sh)" -- -d -b ~/.local/bin
-```
-
-**Quick Start for Contributors:**
-```bash
-# Verify your environment meets requirements
-task doctor
-
-# Setup the project (fetch SDK, install dependencies)
-task setup
-
-# Build for your platform
-task build:js          # Build Node.js bindings
-task build:py          # Build Python bindings
+# Required - Your Zoom OAuth credentials
+ZM_RTMS_CLIENT=your_client_id
+ZM_RTMS_SECRET=your_client_secret
 
-# Run tests
-task test:js           # Test Node.js
-task test:py           # Test Python
+# Optional - Webhook server configuration
+ZM_RTMS_PORT=8080
+ZM_RTMS_PATH=/webhook
 
-# See all available commands
-task --list
+# Optional - Logging configuration
+ZM_RTMS_LOG_LEVEL=debug          # error, warn, info, debug, trace
+ZM_RTMS_LOG_FORMAT=progressive    # progressive or json
+ZM_RTMS_LOG_ENABLED=true          # true or false
 ```
 
-For detailed contribution guidelines, build instructions, and troubleshooting, see [CONTRIBUTING.md](CONTRIBUTING.md).
-
-## Usage
-
-> **Speaker Identification with Mixed Audio**
->
-> When using `AUDIO_MIXED_STREAM` (the default), all participants are mixed into a single audio stream and the audio callback's metadata will **not** identify the current speaker. To identify who is speaking, register an `onActiveSpeakerEvent` callback — it fires whenever the active speaker changes. See [Troubleshooting #7](#7-identifying-speakers-with-mixed-audio-streams) and [#80](https://github.com/zoom/rtms/issues/80) for details.
-
-### Node.js - Webhook Integration
+### Node.js
 
-Easily respond to Zoom webhooks and connect to RTMS streams:
+For more details, see our [Quickstart App](https://github.com/zoom/rtms-quickstart-js)
 
 ```javascript
 import rtms from "@zoom/rtms";
@@ -168,163 +144,32 @@ rtms.onWebhookEvent(({event, payload}) => {
 });
 ```
 
-### Node.js - Advanced Webhook Handling
-
-For advanced use cases requiring custom webhook validation or response handling (e.g., Zoom's webhook validation challenge), you can use the enhanced callback with raw HTTP access:
-
-```javascript
-import rtms from "@zoom/rtms";
-
-rtms.onWebhookEvent((payload, req, res) => {
-    // Access request headers for webhook validation
-    const signature = req.headers['x-zoom-signature'];
-    
-    // Handle Zoom's webhook validation challenge
-    if (req.headers['x-zoom-webhook-validator']) {
-        const validationToken = req.headers['x-zoom-webhook-validator'];
-        
-        // Echo back the validation token
-        res.writeHead(200, { 'Content-Type': 'application/json' });
-        res.end(JSON.stringify({ plainToken: validationToken }));
-        return;
-    }
-    
-    // Custom validation logic
-    if (!validateWebhookSignature(payload, signature)) {
-        res.writeHead(401, { 'Content-Type': 'application/json' });
-        res.end(JSON.stringify({ error: 'Invalid signature' }));
-        return;
-    }
-    
-    // Process the webhook payload
-    if (payload.event === "meeting.rtms_started") {
-        const client = new rtms.Client();
-        
-        client.onAudioData((data, timestamp, metadata) => {
-            console.log(`Received audio from ${metadata.userName}`);
-        });
-        
-        client.join(payload.payload);
-    }
-    
-    // Send custom response
-    res.writeHead(200, { 'Content-Type': 'application/json' });
-    res.end(JSON.stringify({ status: 'ok' }));
-});
-```
-
-### Node.js - Sharing Ports with Existing Servers
-
-If you need to integrate webhook handling with your existing Express, Fastify, or other HTTP server (useful for Cloud Run, Kubernetes, or any deployment requiring a single port), use `createWebhookHandler`:
-
-```javascript
-import express from 'express';
-import rtms from '@zoom/rtms';
-
-const app = express();
-app.use(express.json());
+> **Production note:** The example above does not validate the incoming webhook signature. Zoom cryptographically signs every webhook — production apps must verify signatures before processing. See [Webhook Validation](examples/node.md#webhook-validation)
 
-// Your existing application routes
-app.get('/health', (req, res) => {
-    res.json({ status: 'healthy' });
-});
+### Python 
 
-app.get('/admin', (req, res) => {
-    res.json({ admin: 'panel' });
-});
+For more details, see our [Quickstart App](https://github.com/zoom/rtms-quickstart-py)
 
-// Create a webhook handler that can be mounted on your existing server
-const webhookHandler = rtms.createWebhookHandler(
-    (payload) => {
-        console.log(`Received webhook: ${payload.event}`);
-
-        if (payload.event === "meeting.rtms_started") {
-            const client = new rtms.Client();
-            client.onAudioData((data, timestamp, metadata) => {
-                console.log(`Audio from ${metadata.userName}`);
-            });
-            client.join(payload.payload);
-        }
-    },
-    '/zoom/webhook'  // Path to handle
-);
-
-// Mount the webhook handler on your Express app
-app.post('/zoom/webhook', webhookHandler);
-
-// Single port for all routes
-const PORT = process.env.PORT || 8080;
-app.listen(PORT, () => {
-    console.log(`Server running on port ${PORT}`);
-    console.log(`Webhook endpoint: http://localhost:${PORT}/zoom/webhook`);
-    console.log(`Health check: http://localhost:${PORT}/health`);
-});
-```
+#### Environment Setup
 
-You can also use `RawWebhookCallback` with `createWebhookHandler` for custom validation:
-
-```javascript
-const webhookHandler = rtms.createWebhookHandler(
-    (payload, req, res) => {
-        // Custom validation with raw HTTP access
-        const signature = req.headers['x-zoom-signature'];
-
-        if (!validateSignature(payload, signature)) {
-            res.writeHead(401);
-            res.end('Unauthorized');
-            return;
-        }
-
-        // Process webhook...
-
-        res.writeHead(200, { 'Content-Type': 'application/json' });
-        res.end(JSON.stringify({ status: 'ok' }));
-    },
-    '/zoom/webhook'
-);
-
-app.post('/zoom/webhook', webhookHandler);
-```
-
-### Node.js - Class-Based Approach
-
-For greater control or connecting to multiple streams simultaneously:
+Create a virtual environment and install dependencies:
 
-```javascript
-import rtms from "@zoom/rtms";
+```bash
+# Create virtual environment
+python3 -m venv .venv
 
-const client = new rtms.Client();
+# Activate virtual environment
+source .venv/bin/activate  # On Windows: .venv\Scripts\activate
 
-client.onAudioData((data, timestamp, metadata) => {
-    console.log(`Received audio: ${data.length} bytes`);
-});
+# Install dependencies
+pip install python-dotenv
 
-client.join({
-    meeting_uuid: "your_meeting_uuid",
-    rtms_stream_id: "your_stream_id",
-    server_urls: "wss://example.zoom.us",
-});
+# Install RTMS SDK
+pip install rtms
 ```
 
-### Node.js - Global Singleton
+#### Quickstart
 
-When you only need to connect to a single RTMS stream:
-
-```javascript
-import rtms from "@zoom/rtms";
-
-rtms.onAudioData((data, timestamp, metadata) => {
-    console.log(`Received audio from ${metadata.userName}`);
-});
-
-rtms.join({
-    meeting_uuid: "your_meeting_uuid",
-    rtms_stream_id: "your_stream_id",
-    server_urls: "wss://rtms.zoom.us"
-});
-```
-
-### Python - Basic Usage
 
 ```python
 #!/usr/bin/env python3
@@ -382,77 +227,63 @@ if __name__ == '__main__':
         time.sleep(0.01)
 ```
 
-### Python - Advanced Webhook Validation
+> **Production note:** The example above does not validate the incoming webhook signature. Zoom cryptographically signs every webhook — production apps must verify signatures before processing. See [Webhook Validation](examples/python.md#webhook-validation) 
 
-For production use cases requiring custom webhook validation:
+## Troubleshooting
 
-```python
-import rtms
-import hmac
-import hashlib
+If you encounter issues some of these steps may help.
 
-client = rtms.Client()
+### 1. Segmentation Fault / Crash on Startup
 
-@client.on_webhook_event()
-def handle_webhook(payload, request, response):
-    # Access request headers for validation
-    signature = request.headers.get('x-zoom-signature')
-
-    # Handle Zoom's webhook validation challenge
-    if request.headers.get('x-zoom-webhook-validator'):
-        validator = request.headers['x-zoom-webhook-validator']
-        response.set_status(200)
-        response.send({'plainToken': validator})
-        return
-
-    # Custom signature validation
-    if not validate_signature(payload, signature):
-        response.set_status(401)
-        response.send({'error': 'Invalid signature'})
-        return
-
-    # Process valid webhook
-    if payload.get('event') == 'meeting.rtms_started':
-        client.join(payload.get('payload'))
+**Symptoms:**
+- Immediate crash when requiring/importing the module
+- Error message: `Segmentation fault (core dumped)`
+- Stack trace shows `napi_module_register_by_symbol`
 
-    response.send({'status': 'ok'})
-```
+**Root Cause:** Using Node.js version < 22.0.0
 
-### Python - Environment Setup
+**Solution:** See [Using NVM](#using-nvm)
 
-Create a virtual environment and install dependencies:
 
-```bash
-# Create virtual environment
-python3 -m venv .venv
+**Prevention:**
+- Always use Node.js 22.0.0 or higher
+- Use recommended version with `.nvmrc`: `nvm use` (Node.js 24 LTS)
+- Check version before installing: `node --version`
 
-# Activate virtual environment
-source .venv/bin/activate  # On Windows: .venv\Scripts\activate
+### 2. Platform Support
+Verify you're using a supported platform (darwin-arm64 or linux-x64)
 
-# Install dependencies
-pip install python-dotenv
+### 3. SDK Files
+Ensure RTMS C++ SDK files are correctly placed in the appropriate lib directory
 
-# Install RTMS SDK
-pip install rtms
-```
+### 4. Build Mode
+Try both debug and release modes (`npm run debug` or `npm run release`)
 
-Create a `.env` file:
+### 5. Dependencies
+Verify all prerequisites are installed
 
-```bash
-# Required - Your Zoom OAuth credentials
-ZM_RTMS_CLIENT=your_client_id
-ZM_RTMS_SECRET=your_client_secret
+### 6. Audio Defaults Mismatch
+This SDK uses different default audio parameters than the raw RTMS WebSocket protocol for better out-of-the-box quality. If you need to match the WebSocket protocol defaults, see [#92](https://github.com/zoom/rtms/issues/92) for details.
 
-# Optional - Webhook server configuration
-ZM_RTMS_PORT=8080
-ZM_RTMS_PATH=/webhook
+### 7. Identifying Speakers with Mixed Audio Streams
+When using `AUDIO_MIXED_STREAM`, the audio callback's metadata does not identify the current speaker since all participants are mixed into a single stream. To identify who is speaking, use the `onActiveSpeakerEvent` callback:
 
-# Optional - Logging configuration
-ZM_RTMS_LOG_LEVEL=debug          # error, warn, info, debug, trace
-ZM_RTMS_LOG_FORMAT=progressive    # progressive or json
-ZM_RTMS_LOG_ENABLED=true          # true or false
+**Node.js:**
+```javascript
+client.onActiveSpeakerEvent((timestamp, userId, userName) => {
+    console.log(`Active speaker: ${userName} (${userId})`);
+});
 ```
 
+**Python:**
+```python
+@client.onActiveSpeakerEvent
+def on_active_speaker(timestamp, user_id, user_name):
+    print(f"Active speaker: {user_name} ({user_id})")
+```
+
+This callback notifies your application whenever the active speaker changes in the meeting. You can also use the lower-level `onEventEx` function with the active speaker event type directly. See [#80](https://github.com/zoom/rtms/issues/80) for more details.
+
 ## Building from Source
 
 The RTMS SDK can be built from source using either Docker (recommended) or local build tools.
@@ -461,7 +292,7 @@ The RTMS SDK can be built from source using either Docker (recommended) or local
 
 #### Prerequisites
 - Docker and Docker Compose
-- Zoom RTMS C SDK files (contact Zoom for access)
+- Zoom RTMS C++ SDK files (contact Zoom for access)
 - Task installed (or use Docker's Task installation)
 
 #### Steps
@@ -494,12 +325,12 @@ Docker Compose creates **distributable packages** for linux-x64 (prebuilds for N
 ### Building Locally
 
 #### Prerequisites
-- Node.js (>= 20.3.0, LTS recommended)
+- Node.js (>= 22.0.0, LTS recommended)
 - Python 3.10+ with pip (for Python build)
 - CMake 3.25+
 - C/C++ build tools
 - Task (go-task) - https://taskfile.dev/installation/
-- Zoom RTMS C SDK files (contact Zoom for access)
+- Zoom RTMS C++ SDK files (contact Zoom for access)
 
 #### Steps
 ```bash
@@ -553,95 +384,17 @@ task setup                    # Fetch SDK and install dependencies
 BUILD_TYPE=Debug task build:js    # Build in debug mode
 BUILD_TYPE=Release task build:js  # Build in release mode (default)
 
-# Debug logging for C SDK callbacks
+# Debug logging for C++ SDK callbacks
 RTMS_DEBUG=ON task build:js       # Enable verbose callback logging
 ```
 
-## Troubleshooting
-
-If you encounter issues:
-
-### 1. Segmentation Fault / Crash on Startup
-
-**Symptoms:**
-- Immediate crash when requiring/importing the module
-- Error message: `Segmentation fault (core dumped)`
-- Stack trace shows `napi_module_register_by_symbol`
-
-**Root Cause:** Using Node.js version < 20.3.0
-
-**Solution:**
-```bash
-# 1. Check your Node.js version
-node --version
-
-# 2. If < 20.3.0, upgrade to a supported version
-
-# Using nvm (recommended):
-nvm install 24           # Install Node.js 24 LTS (recommended)
-nvm use 24
-
-# Or install minimum version:
-nvm install 20
-nvm use 20
-
-# Or download from: https://nodejs.org/
-
-# 3. Clear npm cache and reinstall
-npm cache clean --force
-rm -rf node_modules package-lock.json
-npm install
-```
-
-**Prevention:**
-- Always use Node.js 20.3.0 or higher
-- Use recommended version with `.nvmrc`: `nvm use` (Node.js 24 LTS)
-- Check version before installing: `node --version`
-
-### 2. Platform Support
-Verify you're using a supported platform (darwin-arm64 or linux-x64)
-
-### 3. SDK Files
-Ensure RTMS C SDK files are correctly placed in the appropriate lib directory
-
-### 4. Build Mode
-Try both debug and release modes (`npm run debug` or `npm run release`)
-
-### 5. Dependencies
-Verify all prerequisites are installed
-
-### 6. Audio Defaults Mismatch
-This SDK uses different default audio parameters than the raw RTMS WebSocket protocol for better out-of-the-box quality. If you need to match the WebSocket protocol defaults, see [#92](https://github.com/zoom/rtms/issues/92) for details.
-
-### 7. Identifying Speakers with Mixed Audio Streams
-When using `AUDIO_MIXED_STREAM`, the audio callback's metadata does not identify the current speaker since all participants are mixed into a single stream. To identify who is speaking, use the `onActiveSpeakerEvent` callback:
-
-**Node.js:**
-```javascript
-client.onActiveSpeakerEvent((timestamp, userId, userName) => {
-    console.log(`Active speaker: ${userName} (${userId})`);
-});
-```
-
-**Python:**
-```python
-@client.onActiveSpeakerEvent
-def on_active_speaker(timestamp, user_id, user_name):
-    print(f"Active speaker: {user_name} ({user_id})")
-```
+## For Contributors
 
-This callback notifies your application whenever the active speaker changes in the meeting. You can also use the lower-level `onEventEx` function with the active speaker event type directly. See [#80](https://github.com/zoom/rtms/issues/80) for more details.
+For detailed contribution guidelines, build instructions, and troubleshooting, see [CONTRIBUTING.md](CONTRIBUTING.md).
 
 ## For Maintainers
 
-If you're a maintainer looking to build, test, or publish new releases of the RTMS SDK, please refer to [PUBLISHING.md](PUBLISHING.md) for comprehensive documentation on:
-
-- Building platform-specific wheels and prebuilds
-- Publishing to npm and PyPI
-- GitHub Actions CI/CD workflow
-- Testing procedures
-- Troubleshooting common issues
-- Release workflows for Node.js and Python
+If you're a maintainer looking to build, test, or publish new releases of the RTMS SDK, please refer to [PUBLISHING.md](PUBLISHING.md)
 
 ## License
 
diff --git a/Taskfile.yml b/Taskfile.yml
index f0174e8..9fb6945 100644
--- a/Taskfile.yml
+++ b/Taskfile.yml
@@ -107,7 +107,7 @@ tasks:
           for wheel in dist/py/*linux_x86_64*.whl; do
             if [ -f "$wheel" ]; then
               echo "Repairing: $wheel"
-              auditwheel repair "$wheel" --exclude librtmsdk.so.0 -w dist/py
+              auditwheel repair "$wheel" --plat manylinux_2_34_x86_64 --exclude librtmsdk.so.0 -w dist/py
               rm "$wheel"
               echo "Wheel repaired successfully"
             fi
@@ -216,13 +216,13 @@ tasks:
     desc: "Run Node.js tests (local)"
     deps: [build:js]
     cmds:
-      - npx jest tests/rtms.test.ts
+      - npx jest tests/ts/rtms.test.ts tests/ts/rtms.wrapper.test.ts
 
   test:js:integration:
     desc: "Run npm pack → install → load integration test"
     deps: [build:js]
     cmds:
-      - npx jest tests/pack-install.test.js --testTimeout=120000
+      - npx jest tests/ts/pack-install.test.js --testTimeout=120000
 
   test:js:linux:
     desc: "Run Node.js tests in Linux Docker"
@@ -230,17 +230,24 @@ tasks:
       - docker compose run --rm test-js
 
   test:py:
-    desc: "Run Python tests (local)"
+    desc: "Run Python tests (local darwin)"
     deps: [build:py]
     cmds:
-      - python -m pip install --find-links=dist/py rtms
-      - pytest tests/test_rtms.py -v
+      - .venv/bin/pip install --force-reinstall --find-links=dist/py rtms --quiet
+      - .venv/bin/pytest tests/py/test_rtms.py -v
 
   test:py:linux:
     desc: "Run Python tests in Linux Docker"
     cmds:
       - docker compose run --rm test-py
 
+  test:cpp:
+    desc: "Build and run C++ unit tests (no SDK binary required)"
+    cmds:
+      - cmake -B build/tests -DRTMS_BUILD_TESTS=ON -DCMAKE_BUILD_TYPE=Debug
+      - cmake --build build/tests --target rtms_tests -j$(nproc 2>/dev/null || sysctl -n hw.logicalcpu)
+      - cd build/tests && ctest --output-on-failure
+
   test:local:
     desc: "Run all tests locally"
     cmds:
diff --git a/compose.yaml b/compose.yaml
index 76ba139..031d99b 100644
--- a/compose.yaml
+++ b/compose.yaml
@@ -52,7 +52,7 @@ services:
         TARGET: js
     environment:
       - GITHUB_TOKEN
-    command: sh -c "task setup && task build:js && npx jest tests/"
+    command: sh -c "task setup && task build:js && npx jest tests/ts/"
 
   test-py:
     <<: *base
@@ -60,7 +60,7 @@ services:
       context: .
       args:
         TARGET: py
-    command: sh -c "task setup && task build:py && python -m pip install --find-links=dist/py rtms && pytest tests/test_rtms.py -v"
+    command: sh -c "task build:py && pip install --find-links=dist/py rtms --quiet && pytest tests/py/test_rtms.py -v"
 
   # =============================================================================
   # Docs Service - Documentation generation
diff --git a/examples/node.md b/examples/node.md
new file mode 100644
index 0000000..a86f8c7
--- /dev/null
+++ b/examples/node.md
@@ -0,0 +1,334 @@
+# Node.js API Reference
+
+Complete API reference and examples for the `@zoom/rtms` Node.js package.
+
+**Requirements:** Node.js >= 22.0.0 (Node.js 24 LTS recommended)
+
+For product-specific webhook events and payload fields, see the [product guides](../README.md#supported-products).
+
+## Installation
+
+```bash
+npm install @zoom/rtms
+```
+
+## Webhook Integration
+
+`onWebhookEvent` sets up an HTTP server that receives Zoom webhook deliveries and passes the parsed payload to your callback. The SDK starts polling for RTMS events automatically after `join()` succeeds.
+
+```javascript
+import rtms from "@zoom/rtms";
+
+const clients = new Map();
+
+rtms.onWebhookEvent(({ event, payload }) => {
+    const streamId = payload?.rtms_stream_id;
+
+    if (event.includes("rtms_stopped")) {
+        clients.get(streamId)?.leave();
+        clients.delete(streamId);
+        return;
+    }
+
+    if (!event.includes("rtms_started")) return;
+
+    const client = new rtms.Client();
+    clients.set(streamId, client);
+
+    client.onTranscriptData((data, size, timestamp, metadata) => {
+        console.log(`[${timestamp}] ${metadata.userName}: ${data}`);
+    });
+
+    client.join(payload);
+});
+```
+
+For the specific event names for your product, see the [product guides](../README.md#supported-products).
+
+## Webhook Validation
+
+> **⚠️ Required for production.** The example above processes all incoming requests without verification. In production, Zoom cryptographically signs every webhook — you must validate the signature to reject forged requests.
+
+Use the raw callback form `(payload, req, res)` to access headers directly:
+
+```javascript
+import rtms from "@zoom/rtms";
+import { createHmac } from "crypto";
+
+function verifySignature(body, timestamp, signature) {
+    const message = `v0:${timestamp}:${body}`;
+    const expected = "v0=" + createHmac("sha256", process.env.ZM_RTMS_WEBHOOK_SECRET)
+        .update(message)
+        .digest("hex");
+    return expected === signature;
+}
+
+rtms.onWebhookEvent((payload, req, res) => {
+    const signature  = req.headers["x-zm-signature"];
+    const timestamp  = req.headers["x-zm-request-timestamp"];
+
+    // Zoom endpoint validation challenge — respond before processing events
+    if (req.headers["x-zm-webhook-validator"]) {
+        const token = req.headers["x-zm-webhook-validator"];
+        res.writeHead(200, { "Content-Type": "application/json" });
+        res.end(JSON.stringify({ plainToken: token }));
+        return;
+    }
+
+    if (!signature || !timestamp || !verifySignature(JSON.stringify(payload), timestamp, signature)) {
+        res.writeHead(401);
+        res.end(JSON.stringify({ error: "Unauthorized" }));
+        return;
+    }
+
+    res.writeHead(200, { "Content-Type": "application/json" });
+    res.end(JSON.stringify({ status: "ok" }));
+
+    if (payload.event?.includes("rtms_started")) {
+        const client = new rtms.Client();
+        client.onTranscriptData((data, size, timestamp, metadata) => {
+            console.log(`[${timestamp}] ${metadata.userName}: ${data}`);
+        });
+        client.join(payload.payload);
+    }
+});
+```
+
+## Sharing a Port with an Existing Server
+
+If your app already runs an HTTP server (Express, Fastify, Cloud Run, etc.), mount the webhook handler on your existing port with `createWebhookHandler`:
+
+```javascript
+import express from "express";
+import rtms from "@zoom/rtms";
+
+const app = express();
+app.use(express.json());
+
+app.get("/health", (req, res) => res.json({ status: "ok" }));
+
+const webhookHandler = rtms.createWebhookHandler(
+    ({ event, payload }) => {
+        if (!event.includes("rtms_started")) return;
+        const client = new rtms.Client();
+        client.onAudioData((data, size, timestamp, metadata) => {
+            console.log(`Audio from ${metadata.userName}: ${data.length}B`);
+        });
+        client.join(payload);
+    },
+    "/zoom/webhook"
+);
+
+app.post("/zoom/webhook", webhookHandler);
+app.listen(8080);
+```
+
+`createWebhookHandler` also accepts the raw `(payload, req, res)` form for custom validation.
+
+## Class-Based API
+
+Use `new rtms.Client()` directly for more control or to connect to multiple streams simultaneously:
+
+```javascript
+import rtms from "@zoom/rtms";
+
+const client = new rtms.Client();
+
+client.onAudioData((data, size, timestamp, metadata) => {
+    console.log(`Received ${data.length} bytes from ${metadata.userName}`);
+});
+
+client.join({
+    meeting_uuid: "your_meeting_uuid",
+    rtms_stream_id: "your_stream_id",
+    server_urls:   "wss://example.zoom.us",
+});
+```
+
+## Media Callbacks
+
+All callbacks receive a `metadata` object with `userId` and `userName`:
+
+```javascript
+// Transcript — text data with speaker info
+client.onTranscriptData((data, size, timestamp, metadata) => {
+    console.log(`[${timestamp}] ${metadata.userName}: ${data}`);
+});
+
+// Audio — raw PCM / Opus frames
+client.onAudioData((data, size, timestamp, metadata) => {
+    console.log(`Audio: ${data.length}B from ${metadata.userName}`);
+});
+
+// Video — H.264 / raw frames
+client.onVideoData((data, size, timestamp, metadata) => {
+    console.log(`Video: ${size}B from ${metadata.userName}`);
+});
+
+// Desktop share
+client.onDeskshareData((data, size, timestamp, metadata) => {
+    console.log(`Deskshare: ${size}B from ${metadata.userName}`);
+});
+```
+
+> **Speaker identification with mixed audio:** When using the default `AUDIO_MIXED_STREAM`, audio metadata does not identify the current speaker. Use `onActiveSpeakerEvent` to track who is speaking:
+>
+> ```javascript
+> client.onActiveSpeakerEvent((timestamp, userId, userName) => {
+>     console.log(`Active speaker: ${userName} (${userId})`);
+> });
+> ```
+
+## Transcript Language Configuration
+
+By default the SDK auto-detects the spoken language before enabling transcription (~30 seconds). Providing a language hint with `setTranscriptParams` lets transcription begin immediately:
+
+```javascript
+// Hint the source language — skips auto-detect, transcription starts immediately
+client.setTranscriptParams({ srcLanguage: rtms.TranscriptLanguage.ENGLISH });
+```
+
+`TranscriptLanguage` constants: `ENGLISH`, `SPANISH`, `JAPANESE`, `CHINESE_SIMPLIFIED`, and many more. To use auto-detection, omit `setTranscriptParams` or pass `srcLanguage: rtms.TranscriptLanguage.NONE`.
+
+## Media Configuration
+
+By default each stream type uses sensible settings (OPUS audio at 48 kHz, H.264 video at HD/30 fps). Call the relevant `set*Params` method before `join()` to override any field — unspecified fields keep their defaults.
+
+### Video
+
+```javascript
+// Switch from the default composite active-speaker stream to per-participant streams
+client.setVideoParams({
+    dataOpt: rtms.VideoDataOption.VIDEO_SINGLE_INDIVIDUAL_STREAM,
+});
+
+// Full control — all fields optional
+client.setVideoParams({
+    codec:      rtms.VideoCodec.H264,
+    resolution: rtms.VideoResolution.HD,
+    fps:        30,
+    dataOpt:    rtms.VideoDataOption.VIDEO_SINGLE_ACTIVE_STREAM,
+});
+```
+
+`VideoCodec` constants: `H264`, `JPG`, `PNG`. `VideoResolution` constants: `SD`, `HD`, `FHD`, `QHD`. `VideoDataOption` constants: `VIDEO_SINGLE_ACTIVE_STREAM` (default composite), `VIDEO_SINGLE_INDIVIDUAL_STREAM` (per-participant), `VIDEO_MIXED_GALLERY_VIEW`.
+
+### Audio
+
+```javascript
+// Receive a single mixed stream instead of the default per-participant streams
+client.setAudioParams({
+    dataOpt: rtms.AudioDataOption.AUDIO_MIXED_STREAM,
+});
+```
+
+`AudioSampleRate` constants: `SR_8K`, `SR_16K`, `SR_32K`, `SR_48K` (default). `AudioChannel` constants: `MONO`, `STEREO` (default). `AudioDataOption` constants: `AUDIO_MULTI_STREAMS` (default, per-participant), `AUDIO_MIXED_STREAM`.
+
+### Desktop Share
+
+```javascript
+client.setDeskshareParams({
+    codec:      rtms.VideoCodec.H264,
+    resolution: rtms.VideoResolution.FHD,
+    fps:        5,
+});
+```
+
+Uses the same `codec`, `resolution`, `fps`, and `dataOpt` fields as video.
+
+## HTTP Proxy
+
+Route RTMS WebSocket traffic through an HTTP proxy. Call `setProxy` before `join()` — it returns `true` on success:
+
+```javascript
+import rtms from "@zoom/rtms";
+
+const clients = new Map();
+
+rtms.onWebhookEvent(({ event, payload }) => {
+    const streamId = payload?.rtms_stream_id;
+
+    if (event.includes("rtms_stopped")) {
+        clients.get(streamId)?.leave();
+        clients.delete(streamId);
+        return;
+    }
+
+    if (!event.includes("rtms_started")) return;
+
+    const client = new rtms.Client();
+    clients.set(streamId, client);
+
+    // Route WebSocket traffic through an HTTP proxy.
+    // Must be called before join(). Returns true on success.
+    const ok = client.setProxy("http", process.env.RTMS_PROXY_URL);
+    if (!ok) console.warn("setProxy failed — joining without proxy");
+
+    client.onTranscriptData((data, size, timestamp, metadata) => {
+        console.log(`[${timestamp}] ${metadata.userName}: ${data}`);
+    });
+
+    client.join(payload);
+});
+```
+
+The first argument is the proxy type (`"http"`). The second argument is the full proxy URL including host and port.
+
+## Individual Video Streams
+
+By default you receive a single composite stream of the active speaker. To receive per-participant video, first configure `VIDEO_SINGLE_INDIVIDUAL_STREAM`, then subscribe per participant as they join:
+
+```javascript
+// Must be called before join() — switches from composite to per-participant streams
+client.setVideoParams({
+    dataOpt: rtms.VideoDataOption.VIDEO_SINGLE_INDIVIDUAL_STREAM,
+});
+
+// Subscribe when a participant joins, unsubscribe when they leave
+client.onUserUpdate((op, participant) => {
+    if (op === rtms.USER_JOIN && participant?.id) {
+        client.subscribeVideo(participant.id, true);
+    }
+    if (op === rtms.USER_LEAVE && participant?.id) {
+        client.subscribeVideo(participant.id, false);
+    }
+});
+
+// Fires when a participant's video turns on or off
+client.onParticipantVideo((userIds, isOn) => {
+    console.log(`Video ${isOn ? "on" : "off"} for users: ${userIds}`);
+});
+
+// Fires with the subscription result for each subscribeVideo() call
+client.onVideoSubscribed((userId, status, error) => {
+    console.log(`subscribeVideo(${userId}): status=${status}${error ? " error=" + error : ""}`);
+});
+```
+
+## Participant Events
+
+```javascript
+client.onParticipantEvent((event, timestamp, participants) => {
+    for (const p of participants) {
+        console.log(`${event} ts=${timestamp} userId=${p.userId} name="${p.userName}"`);
+    }
+});
+```
+
+## Environment Variables
+
+| Variable | Required | Default | Description |
+|---|---|---|---|
+| `ZM_RTMS_CLIENT` | Yes | — | Your Zoom OAuth Client ID |
+| `ZM_RTMS_SECRET` | Yes | — | Your Zoom OAuth Client Secret |
+| `ZM_RTMS_PORT` | No | `8080` | Webhook server port |
+| `ZM_RTMS_PATH` | No | `/` | Webhook endpoint path |
+| `ZM_RTMS_CA` | No | system CA | Path to CA certificate file |
+| `ZM_RTMS_LOG_LEVEL` | No | `info` | Log level: `error`, `warn`, `info`, `debug`, `trace` |
+| `ZM_RTMS_LOG_FORMAT` | No | `progressive` | Log format: `progressive` or `json` |
+| `ZM_RTMS_LOG_ENABLED` | No | `true` | Enable/disable SDK logging |
+
+## Related
+
+- [Python API Reference](python.md)
+- [Full API Reference](https://zoom.github.io/rtms/js/)
diff --git a/examples/python.md b/examples/python.md
new file mode 100644
index 0000000..80c1fd2
--- /dev/null
+++ b/examples/python.md
@@ -0,0 +1,445 @@
+# Python API Reference
+
+Complete API reference and examples for the `rtms` Python package.
+
+**Requirements:** Python >= 3.10
+
+For product-specific webhook events and payload fields, see the [product guides](../README.md#supported-products).
+
+## Installation
+
+```bash
+pip install rtms
+```
+
+## Webhook Integration
+
+The `@rtms.on_webhook_event` decorator sets up an HTTP server that receives Zoom webhook deliveries. The SDK starts polling for RTMS events automatically after `join()` succeeds.
+
+```python
+import rtms
+
+@rtms.on_webhook_event
+def handle_webhook(payload):
+    if 'rtms_started' not in payload.get('event', ''):
+        return
+
+    client = rtms.Client()
+
+    @client.on_transcript_data
+    def on_transcript(data, size, timestamp, metadata):
+        print(f'[{timestamp}] {metadata.userName}: {data.decode()}')
+
+    client.join(payload['payload'])
+
+rtms.run()  # blocks; Ctrl-C to stop
+```
+
+For the specific event names for your product, see the [product guides](../README.md#supported-products).
+
+## Webhook Validation
+
+> **⚠️ Required for production.** The example above processes all incoming requests without verification. In production, Zoom cryptographically signs every webhook — you must validate the signature to reject forged requests.
+
+```python
+import rtms
+import hmac
+import hashlib
+import os
+
+def verify_signature(body: str, timestamp: str, signature: str) -> bool:
+    message = f"v0:{timestamp}:{body}"
+    expected = "v0=" + hmac.new(
+        os.environ["ZM_RTMS_WEBHOOK_SECRET"].encode(),
+        message.encode(),
+        hashlib.sha256
+    ).hexdigest()
+    return hmac.compare_digest(expected, signature)
+
+@rtms.on_webhook_event
+def handle_webhook(payload, request, response):
+    signature = request.headers.get('x-zm-signature', '')
+    timestamp = request.headers.get('x-zm-request-timestamp', '')
+
+    # Zoom endpoint validation challenge
+    validator = request.headers.get('x-zm-webhook-validator')
+    if validator:
+        response.set_status(200)
+        response.send({'plainToken': validator})
+        return
+
+    if not verify_signature(str(payload), timestamp, signature):
+        response.set_status(401)
+        response.send({'error': 'Unauthorized'})
+        return
+
+    response.send({'status': 'ok'})
+
+    if 'rtms_started' in payload.get('event', ''):
+        client = rtms.Client()
+        client.on_transcript_data(
+            lambda data, size, ts, meta: print(f'{meta.userName}: {data.decode()}')
+        )
+        client.join(payload['payload'])
+
+rtms.run()
+```
+
+## Context Manager
+
+Use `with rtms.Client() as client:` to ensure `leave()` is always called — even if an exception occurs:
+
+```python
+import rtms
+
+with rtms.Client() as client:
+    client.on_audio_data(lambda data, size, ts, meta: process(data))
+    client.join(
+        meeting_uuid=...,
+        rtms_stream_id=...,
+        server_urls=...
+    )
+    rtms.run(stop_on_empty=True)
+# client.leave() called automatically on exit
+```
+
+## Media Callbacks
+
+All callbacks receive a `metadata` object with `userId` and `userName`:
+
+```python
+# Transcript — text data with speaker info
+@client.on_transcript_data
+def on_transcript(data, size, timestamp, metadata):
+    print(f'[{timestamp}] {metadata.userName}: {data.decode()}')
+
+# Audio — raw PCM / Opus frames
+@client.on_audio_data
+def on_audio(data, size, timestamp, metadata):
+    print(f'Audio: {len(data)}B from {metadata.userName}')
+
+# Video — H.264 / raw frames
+@client.on_video_data
+def on_video(data, size, timestamp, metadata):
+    print(f'Video: {size}B from {metadata.userName}')
+
+# Desktop share
+@client.on_deskshare_data
+def on_deskshare(data, size, timestamp, metadata):
+    print(f'Deskshare: {size}B from {metadata.userName}')
+```
+
+> **Speaker identification with mixed audio:** When using the default `AUDIO_MIXED_STREAM`, audio metadata does not identify the current speaker. Use `on_active_speaker_event` to track who is speaking:
+>
+> ```python
+> @client.on_active_speaker_event
+> def on_speaker(timestamp, user_id, user_name):
+>     print(f'Active speaker: {user_name} ({user_id})')
+> ```
+
+## Media Configuration
+
+By default each stream type uses sensible settings (OPUS audio at 48 kHz, H.264 video at HD/30 fps). Call the relevant `set_*_params` method before `join()` to override any field — unspecified fields keep their defaults.
+
+### Video
+
+```python
+# Switch from the default composite active-speaker stream to per-participant streams
+params = rtms.VideoParams()
+params.data_opt = rtms.DataOption.VIDEO_SINGLE_INDIVIDUAL_STREAM
+client.set_video_params(params)
+
+# Full control — set only the fields you want to change
+params = rtms.VideoParams()
+params.codec      = rtms.VideoCodec.H264
+params.resolution = rtms.VideoResolution.HD
+params.fps        = 30
+params.data_opt   = rtms.DataOption.VIDEO_SINGLE_ACTIVE_STREAM
+client.set_video_params(params)
+```
+
+`VideoCodec` constants: `H264`, `JPG`, `PNG`. `VideoResolution` constants: `SD`, `HD`, `FHD`, `QHD`. `DataOption` video constants: `VIDEO_SINGLE_ACTIVE_STREAM` (default composite), `VIDEO_SINGLE_INDIVIDUAL_STREAM` (per-participant), `VIDEO_MIXED_GALLERY_VIEW`.
+
+### Audio
+
+```python
+# Receive a single mixed stream instead of the default per-participant streams
+params = rtms.AudioParams()
+params.data_opt = rtms.DataOption.AUDIO_MIXED_STREAM
+client.set_audio_params(params)
+```
+
+`AudioSampleRate` constants: `SR_8K`, `SR_16K`, `SR_32K`, `SR_48K` (default). `AudioChannel` constants: `MONO`, `STEREO` (default). `DataOption` audio constants: `AUDIO_MULTI_STREAMS` (default, per-participant), `AUDIO_MIXED_STREAM`.
+
+### Desktop Share
+
+```python
+params = rtms.DeskshareParams()
+params.codec      = rtms.VideoCodec.H264
+params.resolution = rtms.VideoResolution.FHD
+params.fps        = 5
+client.set_deskshare_params(params)
+```
+
+Uses the same `codec`, `resolution`, `fps`, and `data_opt` fields as video.
+
+### Transcript Language
+
+By default the SDK auto-detects the spoken language before enabling transcription (~30 seconds). Providing a language hint lets transcription begin immediately:
+
+```python
+# Hint the source language — skips auto-detect, transcription starts immediately
+params = rtms.TranscriptParams()
+params.src_language = rtms.TranscriptLanguage.ENGLISH
+client.set_transcript_params(params)
+```
+
+`TranscriptLanguage` constants: `ENGLISH`, `SPANISH`, `JAPANESE`, `CHINESE_SIMPLIFIED`, and many more. To use auto-detection, omit `set_transcript_params` or set `src_language = rtms.TranscriptLanguage.NONE`.
+
+## Individual Video Streams
+
+By default you receive a single composite stream of the active speaker. To receive per-participant video, first configure `VIDEO_SINGLE_INDIVIDUAL_STREAM`, then subscribe per participant as they join:
+
+```python
+# Must be called before join() — switches from composite to per-participant streams
+params = rtms.VideoParams()
+params.data_opt = rtms.DataOption.VIDEO_SINGLE_INDIVIDUAL_STREAM
+client.set_video_params(params)
+
+# Subscribe when a participant joins, unsubscribe when they leave
+@client.on_user_update
+def on_user(op, participant):
+    if op == rtms.USER_JOIN and participant.id:
+        client.subscribe_video(participant.id, True)
+    if op == rtms.USER_LEAVE and participant.id:
+        client.subscribe_video(participant.id, False)
+
+# Fires when a participant's video turns on or off
+@client.on_participant_video
+def on_participant_video(user_ids, is_on):
+    print(f'Video {"on" if is_on else "off"} for users: {user_ids}')
+
+# Fires with the subscription result for each subscribe_video() call
+@client.on_video_subscribed
+def on_video_subscribed(user_id, status, error):
+    print(f'subscribe_video({user_id}): status={status}' + (f' error={error}' if error else ''))
+
+@client.on_video_data
+def on_video(data, size, timestamp, metadata):
+    print(f'Video: {size}B from {metadata.userName}')
+```
+
+## asyncio Integration
+
+`run_async()` is a drop-in replacement for `run()` that uses `asyncio.sleep()` between polls, so it composes naturally with aiohttp, FastAPI, asyncpg, and any other async framework on a shared event loop:
+
+```python
+import asyncio
+import rtms
+from aiohttp import web
+
+routes = web.RouteTableDef()
+
+@routes.post('/webhook')
+async def webhook(request):
+    payload = await request.json()
+    if 'rtms_started' in payload.get('event', ''):
+        client = rtms.Client()
+        client.on_transcript_data(
+            lambda d, s, t, m: print(m.userName, d.decode())
+        )
+        client.join(payload['payload'])
+    return web.Response(text='ok')
+
+async def main():
+    app = web.Application()
+    app.add_routes(routes)
+    runner = web.AppRunner(app)
+    await runner.setup()
+    await web.TCPSite(runner, port=8080).start()
+    await rtms.run_async()   # yields control between polls — never blocks
+
+asyncio.run(main())
+```
+
+Async callbacks are detected automatically and scheduled on the running event loop:
+
+```python
+client = rtms.Client()
+
+async def save_audio(data, size, timestamp, metadata):
+    await db.insert('audio', data)   # fully non-blocking
+
+client.on_audio_data(save_audio)     # coroutine detected, dispatched via loop
+```
+
+## Executor-Based Callback Dispatch
+
+For CPU-bound or I/O-heavy callbacks that should not block the poll loop, pass a `concurrent.futures.Executor`:
+
+```python
+from concurrent.futures import ThreadPoolExecutor
+import rtms
+
+# Per-client executor
+client = rtms.Client(executor=ThreadPoolExecutor(max_workers=8))
+client.on_audio_data(run_asr_model)          # dispatched to thread pool
+client.on_transcript_data(write_to_database) # dispatched to thread pool
+
+# Global default for all clients in the event loop
+rtms.run(executor=ThreadPoolExecutor(max_workers=32))
+```
+
+Callbacks without an executor continue to run inline — identical to v1.0 behavior.
+
+## Scaling Strategy
+
+The Python SDK provides a layered set of primitives for scaling from a single meeting to hundreds of concurrent streams.
+
+### Layer 1 — Single process, inline callbacks (default, zero config)
+
+Appropriate for < ~20 concurrent meetings with lightweight callbacks (logging, forwarding, simple storage).
+
+```python
+import rtms
+
+@rtms.on_webhook_event
+def handle(payload):
+    if payload.get('event') != 'meeting.rtms_started':
+        return
+    client = rtms.Client()
+    client.on_transcript_data(lambda d, s, t, m: print(m.userName, d.decode()))
+    client.join(payload['payload'])
+
+rtms.run()
+```
+
+- **Event loop**: `rtms.run()` — synchronous, single-threaded
+- **Callbacks**: inline, no dispatch overhead
+- **GIL**: released during `poll()`, so other Python threads (e.g. the webhook HTTP server) stay responsive
+
+### Layer 2 — Single process, executor dispatch
+
+Appropriate for CPU-bound callbacks (ML inference, audio processing) or I/O callbacks (database writes, S3 uploads) that would otherwise block the poll loop.
+
+```python
+from concurrent.futures import ThreadPoolExecutor
+import rtms
+
+executor = ThreadPoolExecutor(max_workers=32)
+
+@rtms.on_webhook_event
+def handle(payload):
+    if payload.get('event') != 'meeting.rtms_started':
+        return
+    client = rtms.Client(executor=executor)
+    client.on_audio_data(run_asr_model)          # dispatched to thread pool
+    client.on_transcript_data(write_to_database) # dispatched to thread pool
+    client.join(payload['payload'])
+
+rtms.run()
+```
+
+- **Event loop**: `rtms.run()` — poll loop stays fast; callbacks run in the pool
+- **Thread pool size**: tune `max_workers` to match your callback latency profile
+- **Back-pressure**: executor queue depth limits unbounded growth
+
+### Layer 3 — asyncio native
+
+Appropriate when the rest of your stack is already async (aiohttp, FastAPI, asyncpg). Callbacks declared as `async def` are auto-scheduled on the running event loop.
+
+```python
+import asyncio
+import rtms
+
+async def on_transcript(data, size, timestamp, metadata):
+    await db.execute('INSERT INTO transcripts VALUES ($1, $2)', metadata.userId, data)
+
+@rtms.on_webhook_event
+def handle(payload):
+    if payload.get('event') != 'meeting.rtms_started':
+        return
+    client = rtms.Client()
+    client.on_transcript_data(on_transcript)   # coroutine auto-detected
+    client.join(payload['payload'])
+
+async def main():
+    await asyncio.gather(
+        rtms.run_async(),
+        start_web_server(),
+    )
+
+asyncio.run(main())
+```
+
+- **Event loop**: `rtms.run_async()` — `await asyncio.sleep()` between polls, never blocks
+- **Coroutine dispatch**: `asyncio.run_coroutine_threadsafe` bridges SDK callbacks to the loop
+- **Composable**: runs alongside any other async service on the same loop
+
+### Layer 4 — Multi-process
+
+Appropriate for 50+ concurrent meetings. Each process runs its own `rtms.run()` loop; a load balancer routes webhook events to available workers.
+
+```
+Zoom → Load Balancer (nginx/HAProxy)
+         ├── Worker 0 (rtms.run(), meetings 0..N)
+         ├── Worker 1 (rtms.run(), meetings N..2N)
+         └── Worker 2 (rtms.run(), meetings 2N..3N)
+```
+
+Use a message queue (Redis pub/sub, RabbitMQ, Kafka) to distribute join requests:
+
+```python
+# worker.py — one process per worker
+import rtms, redis, json
+
+r = redis.Redis()
+sub = r.pubsub()
+sub.subscribe('rtms:join')
+
+@rtms.on_webhook_event
+def handle(payload):
+    # Publish to queue; a free worker picks it up
+    r.publish('rtms:join', json.dumps(payload))
+
+rtms.run()
+```
+
+### Choosing the Right Layer
+
+| Workload | Recommended Layer |
+|---|---|
+| Prototyping / light callbacks | Layer 1 — inline |
+| CPU-bound (ASR, video) | Layer 2 — executor |
+| Async framework (FastAPI, aiohttp) | Layer 3 — run_async |
+| 50+ concurrent meetings | Layer 4 — multi-process |
+
+All layers use the same `Client` API. You can mix layers — e.g. Layer 3 for the web server plus Layer 2 executors for heavy callbacks — without any API changes.
+
+## Environment Setup
+
+```bash
+python3 -m venv .venv
+source .venv/bin/activate      # Windows: .venv\Scripts\activate
+pip install python-dotenv rtms
+```
+
+Create a `.env` file:
+
+```bash
+# Required
+ZM_RTMS_CLIENT=your_client_id
+ZM_RTMS_SECRET=your_client_secret
+
+# Optional
+ZM_RTMS_PORT=8080
+ZM_RTMS_PATH=/webhook
+ZM_RTMS_LOG_LEVEL=info          # error, warn, info, debug, trace
+ZM_RTMS_LOG_FORMAT=progressive  # progressive or json
+ZM_RTMS_LOG_ENABLED=true
+```
+
+## Related
+
+- [Node.js API Reference](node.md)
+- [Full API Reference](https://zoom.github.io/rtms/py/)
diff --git a/examples/zcc.md b/examples/zcc.md
new file mode 100644
index 0000000..2086494
--- /dev/null
+++ b/examples/zcc.md
@@ -0,0 +1,105 @@
+# Zoom Contact Center (ZCC)
+
+Real-time media streaming from Zoom Contact Center voice engagements.
+
+## Overview
+
+The API is identical to Meetings. See the [Meetings Quick Start](meetings.md) for full examples.
+
+Key differences:
+- Webhook event: `contact_center.voice_rtms_started` instead of `meeting.rtms_started`
+- Stop event: `contact_center.voice_rtms_stopped` instead of `meeting.rtms_stopped`
+
+> **Audio Configuration:** This SDK uses different default audio parameters than the raw RTMS WebSocket protocol. See [Audio Configuration](meetings.md#audio-configuration) in the Meetings documentation for details.
+
+## Quick Start
+
+### Node.js
+
+The API is identical to Meetings. See the [Meetings Example](meetings.md) for full examples.
+
+```javascript
+import rtms from "@zoom/rtms";
+
+const clients = new Map();
+
+rtms.onWebhookEvent(({ event, payload }) => {
+    const streamId = payload?.rtms_stream_id;
+
+    if (event === "contact_center.voice_rtms_stopped") {
+        clients.get(streamId)?.leave();
+        clients.delete(streamId);
+        return;
+    }
+
+    if (event !== "contact_center.voice_rtms_started") return;
+
+    const client = new rtms.Client();
+    clients.set(streamId, client);
+
+    client.onAudioData((data, size, timestamp, metadata) => {
+        console.log(`Contact center audio from ${metadata.userName}`);
+    });
+
+    client.onTranscriptData((data, size, timestamp, metadata) => {
+        console.log(`[${timestamp}] ${metadata.userName}: ${data}`);
+    });
+
+    client.join(payload);
+});
+```
+
+### Python
+
+The API is identical to Meetings. See the [Meetings Python Quick Start](meetings.md) for full examples.
+
+```python
+import rtms
+
+clients = {}
+
+@rtms.on_webhook_event
+def handle_webhook(payload):
+    event = payload.get('event', '')
+    rtms_payload = payload.get('payload', {})
+    stream_id = rtms_payload.get('rtms_stream_id')
+
+    if event == 'contact_center.voice_rtms_stopped':
+        client = clients.pop(stream_id, None)
+        if client:
+            client.leave()
+        return
+
+    if event != 'contact_center.voice_rtms_started':
+        return
+
+    client = rtms.Client()
+    clients[stream_id] = client
+
+    @client.on_audio_data
+    def on_audio(data, size, timestamp, metadata):
+        print(f'Contact center audio from {metadata.userName}: {size} bytes')
+
+    @client.on_transcript_data
+    def on_transcript(data, size, timestamp, metadata):
+        print(f'[{timestamp}] {metadata.userName}: {data.decode()}')
+
+    client.join(rtms_payload)
+
+rtms.run()
+```
+
+## Use Cases
+
+- **Compliance recording** — Archive voice engagement audio and transcripts for regulatory requirements
+- **Real-time transcription** — Provide live captions or searchable transcripts for agents and supervisors
+- **Sentiment analysis** — Detect customer sentiment during calls to surface coaching opportunities
+- **Agent assist** — Feed transcripts to an LLM to suggest responses or retrieve knowledge base articles in real-time
+- **Quality assurance** — Automatically score call quality against defined criteria
+
+## Related Documentation
+
+- [Meetings Examples](meetings.md) — Full API examples (same API)
+- [Node.js API Reference](node.md) — Webhook validation and advanced patterns
+- [Python API Reference](python.md) — asyncio, executor dispatch, and more
+- [Main README](../README.md)
diff --git a/index.ts b/index.ts
index 1634554..f829719 100644
--- a/index.ts
+++ b/index.ts
@@ -5,9 +5,9 @@ import { createHmac } from 'crypto';
 import { createRequire } from 'module';
 import { IncomingMessage, Server, ServerResponse } from 'http';
 
-import type { 
+import type {
   JoinParams, SignatureParams, WebhookCallback, RawWebhookCallback,
-  VideoParams, AudioParams, DeskshareParams
+  VideoParams, AudioParams, DeskshareParams, TranscriptParams
 } from "./rtms.d.ts";
 
 const require = createRequire(import.meta.url);
@@ -975,6 +975,7 @@ class Client extends nativeRtms.Client {
       meeting_uuid,
       webinar_uuid,
       session_id,
+      engagement_id,
       rtms_stream_id,
       server_urls,
       signature: providedSignature,
@@ -984,12 +985,14 @@ class Client extends nativeRtms.Client {
       pollInterval = 0
     } = options;
 
-    // Use meeting_uuid for Meeting SDK, webinar_uuid for Webinar, session_id for Video SDK
-    const instance_id = meeting_uuid || webinar_uuid || session_id;
+    // Use meeting_uuid for Meeting SDK, webinar_uuid for Webinar,
+    // session_id for Video SDK, engagement_id for ZCC
+    const instance_id = meeting_uuid || webinar_uuid || session_id || engagement_id;
 
     this.pollRate = pollInterval;
 
-    Logger.info('client', `Joining ${meeting_uuid ? 'meeting' : webinar_uuid ? 'webinar' : 'session'}: ${instance_id}`, {
+    const sessionType = meeting_uuid ? 'meeting' : webinar_uuid ? 'webinar' : engagement_id ? 'engagement' : 'session';
+    Logger.info('client', `Joining ${sessionType}: ${instance_id}`, {
       streamId: rtms_stream_id,
       serverUrls: server_urls,
       timeout: providedTimeout,
@@ -997,7 +1000,7 @@ class Client extends nativeRtms.Client {
     });
 
     if (!instance_id) {
-      throw new Error('Either meeting_uuid, webinar_uuid, or session_id must be provided');
+      throw new Error('Either meeting_uuid, webinar_uuid, session_id, or engagement_id must be provided');
     }
 
     const finalSignature = providedSignature || generateSignature({
@@ -1052,17 +1055,45 @@ class Client extends nativeRtms.Client {
     );
   }
 
-  /** 
-  * Sets deskshare parameters for the client
-  */
- setDeskshareParams(params: DeskshareParams): boolean {
-   return setParameters<DeskshareParams>(
-     'client',
-     'deskshare',
-     params,
-     (p) => super.setDeskshareParams(p)
-   );
- }
+  /**
+   * Sets deskshare parameters for the client
+   */
+  setDeskshareParams(params: DeskshareParams): boolean {
+    return setParameters<DeskshareParams>(
+      'client',
+      'deskshare',
+      params,
+      (p) => super.setDeskshareParams(p)
+    );
+  }
+
+  /**
+   * Sets transcript parameters for the client
+   */
+  setTranscriptParams(params: TranscriptParams): boolean {
+    return setParameters<TranscriptParams>(
+      'client',
+      'transcript',
+      params,
+      (p) => super.setTranscriptParams(p)
+    );
+  }
+
+  setProxy(proxy_type: string, proxy_url: string): boolean {
+    return super.setProxy(proxy_type, proxy_url);
+  }
+
+  subscribeVideo(userId: number, subscribe: boolean): boolean {
+    return super.subscribeVideo(userId, subscribe);
+  }
+
+  onParticipantVideo(callback: (users: number[], isOn: boolean) => void): boolean {
+    return super.onParticipantVideo(callback);
+  }
+
+  onVideoSubscribed(callback: (userId: number, status: number, error: string) => void): boolean {
+    return super.onVideoSubscribed(callback);
+  }
 
   /**
    * Register a callback for participant join/leave events
diff --git a/jest.config.cjs b/jest.config.cjs
new file mode 100644
index 0000000..973db0b
--- /dev/null
+++ b/jest.config.cjs
@@ -0,0 +1,8 @@
+/** @type {import('jest').Config} */
+module.exports = {
+  transform: {
+    '^.+\\.tsx?$': ['ts-jest', { tsconfig: { module: 'commonjs' } }],
+    '^.+\\.jsx?$': 'babel-jest',
+  },
+  testEnvironment: 'node',
+};
diff --git a/package-lock.json b/package-lock.json
index 72fdfc8..bc229cb 100644
--- a/package-lock.json
+++ b/package-lock.json
@@ -1,12 +1,12 @@
 {
   "name": "@zoom/rtms",
-  "version": "1.0.2",
+  "version": "1.1.0",
   "lockfileVersion": 3,
   "requires": true,
   "packages": {
     "": {
       "name": "@zoom/rtms",
-      "version": "1.0.2",
+      "version": "1.1.0",
       "cpu": [
         "x64",
         "arm64"
@@ -26,6 +26,7 @@
       "devDependencies": {
         "@types/jest": "^30.0.0",
         "cmake-js": "^8.0.0",
+        "ts-jest": "^29.4.6",
         "typedoc": "^0.28.14",
         "typedoc-plugin-missing-exports": "^4.1.2",
         "typescript": "^5.9.3"
@@ -35,9 +36,9 @@
       }
     },
     "node_modules/@babel/code-frame": {
-      "version": "7.28.6",
-      "resolved": "https://registry.npmjs.org/@babel/code-frame/-/code-frame-7.28.6.tgz",
-      "integrity": "sha512-JYgintcMjRiCvS8mMECzaEn+m3PfoQiyqukOMCCVQtoJGYJw8j/8LBJEiqkHLkfwCcs74E3pbAUFNg7d9VNJ+Q==",
+      "version": "7.29.0",
+      "resolved": "https://registry.npmjs.org/@babel/code-frame/-/code-frame-7.29.0.tgz",
+      "integrity": "sha512-9NhCeYjq9+3uxgdtp20LSiJXJvN0FeCtNGpJxuMFZ1Kv3cWUNb6DOhJwUvcVCzKGR66cw4njwM6hrJLqgOwbcw==",
       "dev": true,
       "license": "MIT",
       "dependencies": {
@@ -49,885 +50,3507 @@
         "node": ">=6.9.0"
       }
     },
-    "node_modules/@babel/helper-validator-identifier": {
-      "version": "7.28.5",
-      "resolved": "https://registry.npmjs.org/@babel/helper-validator-identifier/-/helper-validator-identifier-7.28.5.tgz",
-      "integrity": "sha512-qSs4ifwzKJSV39ucNjsvc6WVHs6b7S03sOh2OcHF9UHfVPqWWALUsNUVzhSBiItjRZoLHx7nIarVjqKVusUZ1Q==",
+    "node_modules/@babel/compat-data": {
+      "version": "7.29.0",
+      "resolved": "https://registry.npmjs.org/@babel/compat-data/-/compat-data-7.29.0.tgz",
+      "integrity": "sha512-T1NCJqT/j9+cn8fvkt7jtwbLBfLC/1y1c7NtCeXFRgzGTsafi68MRv8yzkYSapBnFA6L3U2VSc02ciDzoAJhJg==",
       "dev": true,
       "license": "MIT",
       "engines": {
         "node": ">=6.9.0"
       }
     },
-    "node_modules/@gerrit0/mini-shiki": {
-      "version": "3.21.0",
-      "resolved": "https://registry.npmjs.org/@gerrit0/mini-shiki/-/mini-shiki-3.21.0.tgz",
-      "integrity": "sha512-9PrsT5DjZA+w3lur/aOIx3FlDeHdyCEFlv9U+fmsVyjPZh61G5SYURQ/1ebe2U63KbDmI2V8IhIUegWb8hjOyg==",
+    "node_modules/@babel/core": {
+      "version": "7.29.0",
+      "resolved": "https://registry.npmjs.org/@babel/core/-/core-7.29.0.tgz",
+      "integrity": "sha512-CGOfOJqWjg2qW/Mb6zNsDm+u5vFQ8DxXfbM09z69p5Z6+mE1ikP2jUXw+j42Pf1XTYED2Rni5f95npYeuwMDQA==",
       "dev": true,
       "license": "MIT",
+      "peer": true,
       "dependencies": {
-        "@shikijs/engine-oniguruma": "^3.21.0",
-        "@shikijs/langs": "^3.21.0",
-        "@shikijs/themes": "^3.21.0",
-        "@shikijs/types": "^3.21.0",
-        "@shikijs/vscode-textmate": "^10.0.2"
+        "@babel/code-frame": "^7.29.0",
+        "@babel/generator": "^7.29.0",
+        "@babel/helper-compilation-targets": "^7.28.6",
+        "@babel/helper-module-transforms": "^7.28.6",
+        "@babel/helpers": "^7.28.6",
+        "@babel/parser": "^7.29.0",
+        "@babel/template": "^7.28.6",
+        "@babel/traverse": "^7.29.0",
+        "@babel/types": "^7.29.0",
+        "@jridgewell/remapping": "^2.3.5",
+        "convert-source-map": "^2.0.0",
+        "debug": "^4.1.0",
+        "gensync": "^1.0.0-beta.2",
+        "json5": "^2.2.3",
+        "semver": "^6.3.1"
+      },
+      "engines": {
+        "node": ">=6.9.0"
+      },
+      "funding": {
+        "type": "opencollective",
+        "url": "https://opencollective.com/babel"
       }
     },
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+      "resolved": "https://registry.npmjs.org/semver/-/semver-6.3.1.tgz",
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       "license": "ISC",
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-        "minipass": "^7.0.4"
-      },
-      "engines": {
-        "node": ">=18.0.0"
+      "bin": {
+        "semver": "bin/semver.js"
       }
     },
-    "node_modules/@jest/diff-sequences": {
-      "version": "30.0.1",
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       "license": "MIT",
       "dependencies": {
-        "@jest/get-type": "30.1.0"
+        "@babel/compat-data": "^7.28.6",
+        "@babel/helper-validator-option": "^7.27.1",
+        "browserslist": "^4.24.0",
+        "lru-cache": "^5.1.1",
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       },
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-        "jest-regex-util": "30.0.1"
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+        "@babel/types": "^7.28.6"
       },
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+        "@babel/types": "^7.29.0"
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+      "bin": {
+        "uglifyjs": "bin/uglifyjs"
+      },
+      "engines": {
+        "node": ">=0.8.0"
+      }
+    },
     "node_modules/undici-types": {
       "version": "7.16.0",
       "resolved": "https://registry.npmjs.org/undici-types/-/undici-types-7.16.0.tgz",
@@ -1567,6 +4915,72 @@
         "node": ">= 10.0.0"
       }
     },
+    "node_modules/unrs-resolver": {
+      "version": "1.11.1",
+      "resolved": "https://registry.npmjs.org/unrs-resolver/-/unrs-resolver-1.11.1.tgz",
+      "integrity": "sha512-bSjt9pjaEBnNiGgc9rUiHGKv5l4/TGzDmYw3RhnkJGtLhbnnA/5qJj7x3dNDCRx/PJxu774LlH8lCOlB4hEfKg==",
+      "dev": true,
+      "hasInstallScript": true,
+      "license": "MIT",
+      "dependencies": {
+        "napi-postinstall": "^0.3.0"
+      },
+      "funding": {
+        "url": "https://opencollective.com/unrs-resolver"
+      },
+      "optionalDependencies": {
+        "@unrs/resolver-binding-android-arm-eabi": "1.11.1",
+        "@unrs/resolver-binding-android-arm64": "1.11.1",
+        "@unrs/resolver-binding-darwin-arm64": "1.11.1",
+        "@unrs/resolver-binding-darwin-x64": "1.11.1",
+        "@unrs/resolver-binding-freebsd-x64": "1.11.1",
+        "@unrs/resolver-binding-linux-arm-gnueabihf": "1.11.1",
+        "@unrs/resolver-binding-linux-arm-musleabihf": "1.11.1",
+        "@unrs/resolver-binding-linux-arm64-gnu": "1.11.1",
+        "@unrs/resolver-binding-linux-arm64-musl": "1.11.1",
+        "@unrs/resolver-binding-linux-ppc64-gnu": "1.11.1",
+        "@unrs/resolver-binding-linux-riscv64-gnu": "1.11.1",
+        "@unrs/resolver-binding-linux-riscv64-musl": "1.11.1",
+        "@unrs/resolver-binding-linux-s390x-gnu": "1.11.1",
+        "@unrs/resolver-binding-linux-x64-gnu": "1.11.1",
+        "@unrs/resolver-binding-linux-x64-musl": "1.11.1",
+        "@unrs/resolver-binding-wasm32-wasi": "1.11.1",
+        "@unrs/resolver-binding-win32-arm64-msvc": "1.11.1",
+        "@unrs/resolver-binding-win32-ia32-msvc": "1.11.1",
+        "@unrs/resolver-binding-win32-x64-msvc": "1.11.1"
+      }
+    },
+    "node_modules/update-browserslist-db": {
+      "version": "1.2.3",
+      "resolved": "https://registry.npmjs.org/update-browserslist-db/-/update-browserslist-db-1.2.3.tgz",
+      "integrity": "sha512-Js0m9cx+qOgDxo0eMiFGEueWztz+d4+M3rGlmKPT+T4IS/jP4ylw3Nwpu6cpTTP8R1MAC1kF4VbdLt3ARf209w==",
+      "dev": true,
+      "funding": [
+        {
+          "type": "opencollective",
+          "url": "https://opencollective.com/browserslist"
+        },
+        {
+          "type": "tidelift",
+          "url": "https://tidelift.com/funding/github/npm/browserslist"
+        },
+        {
+          "type": "github",
+          "url": "https://github.com/sponsors/ai"
+        }
+      ],
+      "license": "MIT",
+      "dependencies": {
+        "escalade": "^3.2.0",
+        "picocolors": "^1.1.1"
+      },
+      "bin": {
+        "update-browserslist-db": "cli.js"
+      },
+      "peerDependencies": {
+        "browserslist": ">= 4.21.0"
+      }
+    },
     "node_modules/url-join": {
       "version": "4.0.1",
       "resolved": "https://registry.npmjs.org/url-join/-/url-join-4.0.1.tgz",
@@ -1574,6 +4988,31 @@
       "dev": true,
       "license": "MIT"
     },
+    "node_modules/v8-to-istanbul": {
+      "version": "9.3.0",
+      "resolved": "https://registry.npmjs.org/v8-to-istanbul/-/v8-to-istanbul-9.3.0.tgz",
+      "integrity": "sha512-kiGUalWN+rgBJ/1OHZsBtU4rXZOfj/7rKQxULKlIzwzQSvMJUUNgPwJEEh7gU6xEVxC0ahoOBvN2YI8GH6FNgA==",
+      "dev": true,
+      "license": "ISC",
+      "dependencies": {
+        "@jridgewell/trace-mapping": "^0.3.12",
+        "@types/istanbul-lib-coverage": "^2.0.1",
+        "convert-source-map": "^2.0.0"
+      },
+      "engines": {
+        "node": ">=10.12.0"
+      }
+    },
+    "node_modules/walker": {
+      "version": "1.0.8",
+      "resolved": "https://registry.npmjs.org/walker/-/walker-1.0.8.tgz",
+      "integrity": "sha512-ts/8E8l5b7kY0vlWLewOkDXMmPdLcVV4GmOQLyxuSswIJsweeFZtAsMF7k1Nszz+TYBQrlYRmzOnr398y1JemQ==",
+      "dev": true,
+      "license": "Apache-2.0",
+      "dependencies": {
+        "makeerror": "1.0.12"
+      }
+    },
     "node_modules/which": {
       "version": "6.0.0",
       "resolved": "https://registry.npmjs.org/which/-/which-6.0.0.tgz",
@@ -1590,6 +5029,13 @@
         "node": "^20.17.0 || >=22.9.0"
       }
     },
+    "node_modules/wordwrap": {
+      "version": "1.0.0",
+      "resolved": "https://registry.npmjs.org/wordwrap/-/wordwrap-1.0.0.tgz",
+      "integrity": "sha512-gvVzJFlPycKc5dZN4yPkP8w7Dc37BtP1yczEneOb4uq34pXZcvrtRTmWV8W+Ume+XCxKgbjM+nevkyFPMybd4Q==",
+      "dev": true,
+      "license": "MIT"
+    },
     "node_modules/wrap-ansi": {
       "version": "7.0.0",
       "resolved": "https://registry.npmjs.org/wrap-ansi/-/wrap-ansi-7.0.0.tgz",
@@ -1608,12 +5054,45 @@
         "url": "https://github.com/chalk/wrap-ansi?sponsor=1"
       }
     },
+    "node_modules/wrap-ansi-cjs": {
+      "name": "wrap-ansi",
+      "version": "7.0.0",
+      "resolved": "https://registry.npmjs.org/wrap-ansi/-/wrap-ansi-7.0.0.tgz",
+      "integrity": "sha512-YVGIj2kamLSTxw6NsZjoBxfSwsn0ycdesmc4p+Q21c5zPuZ1pl+NfxVdxPtdHvmNVOQ6XSYG4AUtyt/Fi7D16Q==",
+      "dev": true,
+      "license": "MIT",
+      "dependencies": {
+        "ansi-styles": "^4.0.0",
+        "string-width": "^4.1.0",
+        "strip-ansi": "^6.0.0"
+      },
+      "engines": {
+        "node": ">=10"
+      },
+      "funding": {
+        "url": "https://github.com/chalk/wrap-ansi?sponsor=1"
+      }
+    },
     "node_modules/wrappy": {
       "version": "1.0.2",
       "resolved": "https://registry.npmjs.org/wrappy/-/wrappy-1.0.2.tgz",
       "integrity": "sha512-l4Sp/DRseor9wL6EvV2+TuQn63dMkPjZ/sp9XkghTEbV9KlPS1xUsZ3u7/IQO4wxtcFB4bgpQPRcR3QCvezPcQ==",
       "license": "ISC"
     },
+    "node_modules/write-file-atomic": {
+      "version": "5.0.1",
+      "resolved": "https://registry.npmjs.org/write-file-atomic/-/write-file-atomic-5.0.1.tgz",
+      "integrity": "sha512-+QU2zd6OTD8XWIJCbffaiQeH9U73qIqafo1x6V1snCWYGJf6cVE0cDR4D8xRzcEnfI21IFrUPzPGtcPf8AC+Rw==",
+      "dev": true,
+      "license": "ISC",
+      "dependencies": {
+        "imurmurhash": "^0.1.4",
+        "signal-exit": "^4.0.1"
+      },
+      "engines": {
+        "node": "^14.17.0 || ^16.13.0 || >=18.0.0"
+      }
+    },
     "node_modules/y18n": {
       "version": "5.0.8",
       "resolved": "https://registry.npmjs.org/y18n/-/y18n-5.0.8.tgz",
@@ -1634,9 +5113,9 @@
       }
     },
     "node_modules/yaml": {
-      "version": "2.8.2",
-      "resolved": "https://registry.npmjs.org/yaml/-/yaml-2.8.2.tgz",
-      "integrity": "sha512-mplynKqc1C2hTVYxd0PU2xQAc22TI1vShAYGksCCfxbn/dFwnHTNi1bvYsBTkhdUNtGIf5xNOg938rrSSYvS9A==",
+      "version": "2.8.3",
+      "resolved": "https://registry.npmjs.org/yaml/-/yaml-2.8.3.tgz",
+      "integrity": "sha512-AvbaCLOO2Otw/lW5bmh9d/WEdcDFdQp2Z2ZUH3pX9U2ihyUY0nvLv7J6TrWowklRGPYbB/IuIMfYgxaCPg5Bpg==",
       "dev": true,
       "license": "ISC",
       "bin": {
@@ -1677,6 +5156,19 @@
       "engines": {
         "node": ">=12"
       }
+    },
+    "node_modules/yocto-queue": {
+      "version": "0.1.0",
+      "resolved": "https://registry.npmjs.org/yocto-queue/-/yocto-queue-0.1.0.tgz",
+      "integrity": "sha512-rVksvsnNCdJ/ohGc6xgPwyN8eheCxsiLM8mxuE/t/mOVqJewPuO1miLpTHQiRgTKCLexL4MeAFVagts7HmNZ2Q==",
+      "dev": true,
+      "license": "MIT",
+      "engines": {
+        "node": ">=10"
+      },
+      "funding": {
+        "url": "https://github.com/sponsors/sindresorhus"
+      }
     }
   }
 }
diff --git a/package.json b/package.json
index b0d9248..a8a8d92 100644
--- a/package.json
+++ b/package.json
@@ -1,6 +1,6 @@
 {
   "name": "@zoom/rtms",
-  "version": "1.0.3",
+  "version": "1.1.0",
   "description": "Node.js Wrapper for the Zoom RTMS C SDK",
   "main": "./build/Release/index.js",
   "types": "rtms.d.ts",
@@ -17,6 +17,7 @@
   "devDependencies": {
     "@types/jest": "^30.0.0",
     "cmake-js": "^8.0.0",
+    "ts-jest": "^29.4.6",
     "typedoc": "^0.28.14",
     "typedoc-plugin-missing-exports": "^4.1.2",
     "typescript": "^5.9.3"
@@ -80,6 +81,6 @@
   ],
   "license": "MIT",
   "engines": {
-    "node": ">=20.3.0"
+    "node": ">=22.0.0"
   }
 }
diff --git a/pyproject.toml b/pyproject.toml
index a20ffa2..8154433 100644
--- a/pyproject.toml
+++ b/pyproject.toml
@@ -5,7 +5,7 @@ build-backend = "scikit_build_core.build"
 
 [project]
 name = "rtms"
-version = "1.0.3"
+version = "1.1.0"
 description = "Python bindings for the Zoom RTMS C SDK - Real-Time Media Streaming"
 readme = "README.md"
 requires-python = ">=3.10"
diff --git a/requirements-dev.txt b/requirements-dev.txt
new file mode 100644
index 0000000..91379c0
--- /dev/null
+++ b/requirements-dev.txt
@@ -0,0 +1,29 @@
+# Dev/build dependencies pinned to non-vulnerable versions.
+# Audit: pip-audit found CVE-2026-4539 in pygments<2.20.0
+
+# Build
+build==1.3.0
+scikit_build_core==0.11.6
+pybind11==3.0.1
+pybind11-global==3.0.1
+wheel==0.46.3
+
+# Testing
+pytest==9.0.2
+python-dotenv==1.2.1
+
+# Publishing
+twine==6.2.0
+
+# Multi-version wheel building
+cibuildwheel==3.3.1
+
+# Linux wheel repair (Linux only)
+auditwheel==6.3.0
+
+# Documentation
+pdoc3==0.11.6
+Pygments>=2.20.0  # CVE-2026-4539 fixed in 2.20.0
+
+# Auditing
+pip-audit==2.10.0
diff --git a/rtms.d.ts b/rtms.d.ts
index 1cda6e1..2bb8e3f 100644
--- a/rtms.d.ts
+++ b/rtms.d.ts
@@ -183,9 +183,43 @@ export interface MediaTypes {
   transcript?: boolean;
 }
 
+/**
+ * A single target language entry from an AI interpreter stream
+ *
+ * @category Data Interfaces
+ */
+export interface AiTargetLanguage {
+  /** Language ID of the target language */
+  lid: number;
+  /** Tone ID (dialect/variant) of the target language */
+  toneId: number;
+  /** Voice model identifier used to render this target language */
+  voiceId: string;
+  /** AI engine name (e.g. "zoom_ai") */
+  engine: string;
+}
+
+/**
+ * AI interpreter metadata attached to an audio stream
+ *
+ * @category Data Interfaces
+ */
+export interface AiInterpreter {
+  /** Language ID of the source language being interpreted */
+  lid: number;
+  /** Timestamp of the interpretation event */
+  timestamp: number;
+  /** Number of audio channels in the interpreter stream */
+  channelNum: number;
+  /** Sample rate of the interpreter stream in Hz */
+  sampleRate: number;
+  /** Active target language entries (up to 100) */
+  targets: AiTargetLanguage[];
+}
+
 /**
  * Metadata information about a participant in a Zoom meeting
- * 
+ *
  * @category Data Interfaces
  */
 export interface Metadata {
@@ -193,6 +227,12 @@ export interface Metadata {
   userName: string;
   /** The user ID of the Zoom participant */
   userId: number;
+  /** Stream start timestamp in milliseconds */
+  startTs: number;
+  /** Stream end timestamp in milliseconds */
+  endTs: number;
+  /** AI interpreter metadata (populated when Zoom's AI interpreter is active) */
+  aiInterpreter: AiInterpreter;
 }
 
 /**
@@ -296,6 +336,70 @@ export interface DeskshareParams {
   fps?: number;
 }
 
+/**
+ * Configuration parameters for transcript streams
+ *
+ * @category Media Configuration
+ */
+export interface TranscriptParams {
+  /** Source language ID. Use TranscriptLanguage constants. Default: NONE (auto-detect after 30s) */
+  srcLanguage?: number;
+  /** Enable Language Identification. Default: true */
+  enableLid?: boolean;
+}
+
+/**
+ * Language ID constants for use with TranscriptParams.srcLanguage
+ *
+ * @example
+ * ```typescript
+ * const params = new rtms.TranscriptParams();
+ * params.setSrcLanguage(rtms.TranscriptLanguage.ENGLISH);
+ * ```
+ *
+ * @category Constants
+ */
+export const TranscriptLanguage: {
+  readonly NONE: number;
+  readonly ARABIC: number;
+  readonly BENGALI: number;
+  readonly CANTONESE: number;
+  readonly CATALAN: number;
+  readonly CHINESE_SIMPLIFIED: number;
+  readonly CHINESE_TRADITIONAL: number;
+  readonly CZECH: number;
+  readonly DANISH: number;
+  readonly DUTCH: number;
+  readonly ENGLISH: number;
+  readonly ESTONIAN: number;
+  readonly FINNISH: number;
+  readonly FRENCH_CANADA: number;
+  readonly FRENCH_FRANCE: number;
+  readonly GERMAN: number;
+  readonly HEBREW: number;
+  readonly HINDI: number;
+  readonly HUNGARIAN: number;
+  readonly INDONESIAN: number;
+  readonly ITALIAN: number;
+  readonly JAPANESE: number;
+  readonly KOREAN: number;
+  readonly MALAY: number;
+  readonly PERSIAN: number;
+  readonly POLISH: number;
+  readonly PORTUGUESE: number;
+  readonly ROMANIAN: number;
+  readonly RUSSIAN: number;
+  readonly SPANISH: number;
+  readonly SWEDISH: number;
+  readonly TAGALOG: number;
+  readonly TAMIL: number;
+  readonly TELUGU: number;
+  readonly THAI: number;
+  readonly TURKISH: number;
+  readonly UKRAINIAN: number;
+  readonly VIETNAMESE: number;
+};
+
 
 //-----------------------------------------------------------------------------------
 // Parameter interfaces
@@ -318,6 +422,8 @@ export interface JoinParams {
   webinar_uuid?: string;
   /** The session ID (for Video SDK events) - used when meeting_uuid is not provided */
   session_id?: string;
+  /** The engagement ID (for ZCC events) - used when meeting_uuid is not provided */
+  engagement_id?: string;
   /** The RTMS stream ID for this connection */
   rtms_stream_id: string;
   /** The server URL(s) to connect to */
@@ -761,7 +867,41 @@ export class Client {
    * @returns true if the operation succeeds
    */
     setDeskshareParams(params: DeskshareParams): boolean;
-  
+
+  /**
+   * Configures a proxy for SDK connections.
+   *
+   * @param proxy_type Proxy protocol type (e.g. `'http'`, `'https'`)
+   * @param proxy_url Full proxy URL including host and port
+   * @returns true if the operation succeeds
+   */
+  setProxy(proxy_type: string, proxy_url: string): boolean;
+
+  /**
+   * Subscribe or unsubscribe from an individual participant's video stream.
+   *
+   * @param userId The participant's user ID
+   * @param subscribe true to subscribe, false to unsubscribe
+   * @returns true if the operation succeeds
+   */
+  subscribeVideo(userId: number, subscribe: boolean): boolean;
+
+  /**
+   * Sets a callback fired when participants' video state changes.
+   *
+   * @param callback Invoked with an array of user IDs and a boolean indicating video on/off
+   * @returns true if registration succeeds
+   */
+  onParticipantVideo(callback: (users: number[], isOn: boolean) => void): boolean;
+
+  /**
+   * Sets a callback fired in response to a `subscribeVideo` call.
+   *
+   * @param callback Invoked with userId, status code, and an error string (empty on success)
+   * @returns true if registration succeeds
+   */
+  onVideoSubscribed(callback: (userId: number, status: number, error: string) => void): boolean;
+
   /**
    * Sets a callback for join confirmation events
    * 
@@ -976,16 +1116,8 @@ export class Client {
    * 
    * @example
    * ```typescript
-   * client.onVideoData((buffer, size, timestamp, trackId, metadata) => {
+   * client.onVideoData((buffer, size, timestamp, metadata) => {
    *   console.log(`Received ${size} bytes of video from ${metadata.userName}`);
-   *   console.log(`Track ID: ${trackId}`);
-   *   
-   *   // Process the video data
-   *   // buffer - Raw video data (Buffer)
-   *   // size - Size of the video data in bytes
-   *   // timestamp - Timestamp of the video data
-   *   // trackId - ID of the video track
-   *   // metadata - Information about the sender
    * });
    * ```
    */
diff --git a/scripts/install.js b/scripts/install.js
index 1130af3..3cebbb5 100644
--- a/scripts/install.js
+++ b/scripts/install.js
@@ -87,8 +87,13 @@ function installPrebuild() {
   try {
     log('Attempting to download prebuilt binary...');
 
-    // Use prebuild-install to download from GitHub releases
-    execSync('prebuild-install -r napi', {
+    // Use prebuild-install to download from GitHub releases.
+    // During `npm install`, npm augments PATH with node_modules/.bin/ so
+    // the bare command works. When called directly (e.g. CI with --force),
+    // PATH isn't augmented, so resolve from local node_modules/.bin/ if present.
+    const localBin = join(__dirname, '..', 'node_modules', '.bin', 'prebuild-install');
+    const prebuildCmd = existsSync(localBin) ? localBin : 'prebuild-install';
+    execSync(`${prebuildCmd} -r napi`, {
       stdio: 'inherit',
       env: process.env
     });
@@ -109,10 +114,12 @@ function installPrebuild() {
  * Main install function
  */
 function main() {
-  // Detect development mode (git repo exists)
+  // Detect development mode (git repo exists).
+  // Pass --force to bypass this check (e.g. CI extracting a downloaded prebuild).
   const isDevelopment = existsSync(join(__dirname, '..', '.git'));
+  const isForced = process.argv.includes('--force');
 
-  if (isDevelopment) {
+  if (isDevelopment && !isForced) {
     // In dev mode, skip prebuild - developers use `task build`
     process.exit(0);
   }
diff --git a/src/node.cpp b/src/node.cpp
index a4a4b97..dbc061e 100644
--- a/src/node.cpp
+++ b/src/node.cpp
@@ -11,6 +11,35 @@
 using namespace Napi;
 using namespace std;
 
+static Napi::Object buildMetadataObj(Napi::Env env, const rtms::Metadata& metadata) {
+    Napi::Object obj = Napi::Object::New(env);
+    obj.Set("userName", Napi::String::New(env, metadata.userName()));
+    obj.Set("userId", Napi::Number::New(env, metadata.userId()));
+    obj.Set("startTs", Napi::Number::New(env, static_cast<double>(metadata.startTs())));
+    obj.Set("endTs", Napi::Number::New(env, static_cast<double>(metadata.endTs())));
+
+    const auto& aii = metadata.aiInterpreter();
+    Napi::Object aiObj = Napi::Object::New(env);
+    aiObj.Set("lid", Napi::Number::New(env, aii.lid()));
+    aiObj.Set("timestamp", Napi::Number::New(env, static_cast<double>(aii.timestamp())));
+    aiObj.Set("channelNum", Napi::Number::New(env, aii.channelNum()));
+    aiObj.Set("sampleRate", Napi::Number::New(env, aii.sampleRate()));
+
+    Napi::Array targets = Napi::Array::New(env, aii.targets().size());
+    for (size_t i = 0; i < aii.targets().size(); ++i) {
+        const auto& t = aii.targets()[i];
+        Napi::Object tObj = Napi::Object::New(env);
+        tObj.Set("lid", Napi::Number::New(env, t.lid()));
+        tObj.Set("toneId", Napi::Number::New(env, t.toneId()));
+        tObj.Set("voiceId", Napi::String::New(env, t.voiceId()));
+        tObj.Set("engine", Napi::String::New(env, t.engine()));
+        targets.Set(i, tObj);
+    }
+    aiObj.Set("targets", targets);
+    obj.Set("aiInterpreter", aiObj);
+    return obj;
+}
+
 class NodeClient : public Napi::ObjectWrap<NodeClient> {
 public:
     static Napi::Object init(Napi::Env env, Napi::Object exports);
@@ -34,6 +63,8 @@ class NodeClient : public Napi::ObjectWrap<NodeClient> {
     Napi::Value setDeskshareParams(const Napi::CallbackInfo& info);
     Napi::Value setAudioParams(const Napi::CallbackInfo& info);
     Napi::Value setVideoParams(const Napi::CallbackInfo& info);
+    Napi::Value setTranscriptParams(const Napi::CallbackInfo& info);
+    Napi::Value setProxy(const Napi::CallbackInfo& info);
 
     Napi::Value setOnJoinConfirm(const Napi::CallbackInfo& info);
     Napi::Value setOnSessionUpdate(const Napi::CallbackInfo& info);
@@ -48,6 +79,10 @@ class NodeClient : public Napi::ObjectWrap<NodeClient> {
     Napi::Value subscribeEvent(const Napi::CallbackInfo& info);
     Napi::Value unsubscribeEvent(const Napi::CallbackInfo& info);
 
+    Napi::Value subscribeVideo(const Napi::CallbackInfo& info);
+    Napi::Value setOnParticipantVideo(const Napi::CallbackInfo& info);
+    Napi::Value setOnVideoSubscribed(const Napi::CallbackInfo& info);
+
     unique_ptr<rtms::Client> client_;
     Napi::ThreadSafeFunction tsfn_join_confirm_;
     Napi::ThreadSafeFunction tsfn_session_update_;
@@ -58,6 +93,8 @@ class NodeClient : public Napi::ObjectWrap<NodeClient> {
     Napi::ThreadSafeFunction tsfn_transcript_data_;
     Napi::ThreadSafeFunction tsfn_leave_;
     Napi::ThreadSafeFunction tsfn_event_ex_;
+    Napi::ThreadSafeFunction tsfn_participant_video_;
+    Napi::ThreadSafeFunction tsfn_video_subscribed_;
 };
 
 Napi::Value NodeClient::poll(const Napi::CallbackInfo& info) {
@@ -263,6 +300,64 @@ Napi::Value NodeClient::setVideoParams(const Napi::CallbackInfo& info) {
     return Napi::Boolean::New(env, true);
 }
 
+rtms::TranscriptParams readTranscriptParams(const Napi::Object& params) {
+    rtms::TranscriptParams transcript_params;
+
+    if (params.Has("srcLanguage") && params.Get("srcLanguage").IsNumber()) {
+        transcript_params.setSrcLanguage(params.Get("srcLanguage").As<Napi::Number>().Int32Value());
+    }
+
+    if (params.Has("enableLid") && params.Get("enableLid").IsBoolean()) {
+        transcript_params.setEnableLid(params.Get("enableLid").As<Napi::Boolean>().Value());
+    }
+
+    return transcript_params;
+}
+
+Napi::Value NodeClient::setTranscriptParams(const Napi::CallbackInfo& info) {
+    Napi::Env env = info.Env();
+    Napi::HandleScope scope(env);
+
+    if (info.Length() < 1 || !info[0].IsObject()) {
+        Napi::TypeError::New(env, "Object argument expected").ThrowAsJavaScriptException();
+        return env.Null();
+    }
+
+    Napi::Object params = info[0].As<Napi::Object>();
+    auto transcript_params = readTranscriptParams(params);
+
+    try {
+        client_->setTranscriptParams(transcript_params);
+    } catch (const std::invalid_argument& e) {
+        Napi::Error::New(env, e.what()).ThrowAsJavaScriptException();
+        return env.Null();
+    }
+
+    return Napi::Boolean::New(env, true);
+}
+
+Napi::Value NodeClient::setProxy(const Napi::CallbackInfo& info) {
+    Napi::Env env = info.Env();
+    Napi::HandleScope scope(env);
+
+    if (info.Length() < 2 || !info[0].IsString() || !info[1].IsString()) {
+        Napi::TypeError::New(env, "Two string arguments expected: proxy_type, proxy_url").ThrowAsJavaScriptException();
+        return env.Null();
+    }
+
+    std::string proxy_type = info[0].As<Napi::String>().Utf8Value();
+    std::string proxy_url  = info[1].As<Napi::String>().Utf8Value();
+
+    try {
+        client_->setProxy(proxy_type, proxy_url);
+    } catch (const rtms::Exception& e) {
+        Napi::Error::New(env, e.what()).ThrowAsJavaScriptException();
+        return env.Null();
+    }
+
+    return Napi::Boolean::New(env, true);
+}
+
 
 Napi::Value NodeClient::setOnJoinConfirm(const Napi::CallbackInfo& info) {
     Napi::Env env = info.Env();
@@ -386,15 +481,10 @@ Napi::Value NodeClient::setOnDeskshareData(const Napi::CallbackInfo& info) {
     );
 
     client_->setOnDeskshareData([this](const vector<uint8_t>& data, uint64_t timestamp, const rtms::Metadata& metadata) {
-        auto callback = [data, timestamp, userName = metadata.userName(), userId = metadata.userId()]
+        auto callback = [data, timestamp, metadata]
                        (Napi::Env env, Napi::Function jsCallback) {
             Napi::Buffer<uint8_t> buffer = Napi::Buffer<uint8_t>::Copy(env, data.data(), data.size());
-            
-            Napi::Object metadataObj = Napi::Object::New(env);
-            metadataObj.Set("userName", Napi::String::New(env, userName));
-            metadataObj.Set("userId", Napi::Number::New(env, userId));
-
-            jsCallback.Call({buffer, Napi::Number::New(env, data.size()), Napi::Number::New(env, timestamp), metadataObj});
+            jsCallback.Call({buffer, Napi::Number::New(env, data.size()), Napi::Number::New(env, timestamp), buildMetadataObj(env, metadata)});
         };
         tsfn_ds_data_.BlockingCall(callback);
     });
@@ -422,15 +512,10 @@ Napi::Value NodeClient::setOnAudioData(const Napi::CallbackInfo& info) {
     );
 
     client_->setOnAudioData([this](const vector<uint8_t>& data, uint64_t timestamp, const rtms::Metadata& metadata) {
-        auto callback = [data, timestamp, userName = metadata.userName(), userId = metadata.userId()]
+        auto callback = [data, timestamp, metadata]
                        (Napi::Env env, Napi::Function jsCallback) {
             Napi::Buffer<uint8_t> buffer = Napi::Buffer<uint8_t>::Copy(env, data.data(), data.size());
-            
-            Napi::Object metadataObj = Napi::Object::New(env);
-            metadataObj.Set("userName", Napi::String::New(env, userName));
-            metadataObj.Set("userId", Napi::Number::New(env, userId));
-
-            jsCallback.Call({buffer, Napi::Number::New(env, data.size()), Napi::Number::New(env, timestamp), metadataObj});
+            jsCallback.Call({buffer, Napi::Number::New(env, data.size()), Napi::Number::New(env, timestamp), buildMetadataObj(env, metadata)});
         };
         tsfn_audio_data_.BlockingCall(callback);
     });
@@ -458,15 +543,10 @@ Napi::Value NodeClient::setOnVideoData(const Napi::CallbackInfo& info) {
     );
 
     client_->setOnVideoData([this](const vector<uint8_t>& data, uint64_t timestamp, const rtms::Metadata& metadata) {
-        auto callback = [data, timestamp, userName = metadata.userName(), userId = metadata.userId()]
+        auto callback = [data, timestamp, metadata]
                        (Napi::Env env, Napi::Function jsCallback) {
             Napi::Buffer<uint8_t> buffer = Napi::Buffer<uint8_t>::Copy(env, data.data(), data.size());
-            
-            Napi::Object metadataObj = Napi::Object::New(env);
-            metadataObj.Set("userName", Napi::String::New(env, userName));
-            metadataObj.Set("userId", Napi::Number::New(env, userId));
-
-            jsCallback.Call({buffer, Napi::Number::New(env, data.size()), Napi::Number::New(env, timestamp), metadataObj});
+            jsCallback.Call({buffer, Napi::Number::New(env, data.size()), Napi::Number::New(env, timestamp), buildMetadataObj(env, metadata)});
         };
         tsfn_video_data_.BlockingCall(callback);
     });
@@ -494,15 +574,10 @@ Napi::Value NodeClient::setOnTranscriptData(const Napi::CallbackInfo& info) {
     );
 
     client_->setOnTranscriptData([this](const vector<uint8_t>& data, uint64_t timestamp, const rtms::Metadata& metadata) {
-        auto callback = [data, timestamp, userName = metadata.userName(), userId = metadata.userId()]
+        auto callback = [data, timestamp, metadata]
                        (Napi::Env env, Napi::Function jsCallback) {
             Napi::Buffer<uint8_t> buffer = Napi::Buffer<uint8_t>::Copy(env, data.data(), data.size());
-            
-            Napi::Object metadataObj = Napi::Object::New(env);
-            metadataObj.Set("userName", Napi::String::New(env, userName));
-            metadataObj.Set("userId", Napi::Number::New(env, userId));
-
-            jsCallback.Call({buffer, Napi::Number::New(env, data.size()), Napi::Number::New(env, timestamp), metadataObj});
+            jsCallback.Call({buffer, Napi::Number::New(env, data.size()), Napi::Number::New(env, timestamp), buildMetadataObj(env, metadata)});
         };
         tsfn_transcript_data_.BlockingCall(callback);
     });
@@ -616,6 +691,93 @@ Napi::Value NodeClient::unsubscribeEvent(const Napi::CallbackInfo& info) {
     }
 }
 
+Napi::Value NodeClient::subscribeVideo(const Napi::CallbackInfo& info) {
+    Napi::Env env = info.Env();
+    Napi::HandleScope scope(env);
+
+    if (info.Length() < 2 || !info[0].IsNumber() || !info[1].IsBoolean()) {
+        Napi::TypeError::New(env, "Two arguments expected: userId (number), subscribe (boolean)").ThrowAsJavaScriptException();
+        return env.Null();
+    }
+
+    int user_id = info[0].As<Napi::Number>().Int32Value();
+    bool subscribe = info[1].As<Napi::Boolean>().Value();
+
+    try {
+        client_->subscribeVideo(user_id, subscribe);
+        return Napi::Boolean::New(env, true);
+    } catch (const rtms::Exception& e) {
+        Napi::Error::New(env, e.what()).ThrowAsJavaScriptException();
+        return env.Null();
+    }
+}
+
+Napi::Value NodeClient::setOnParticipantVideo(const Napi::CallbackInfo& info) {
+    Napi::Env env = info.Env();
+    Napi::HandleScope scope(env);
+
+    if (info.Length() < 1 || !info[0].IsFunction()) {
+        Napi::TypeError::New(env, "Function argument expected").ThrowAsJavaScriptException();
+        return env.Null();
+    }
+
+    Napi::Function callback = info[0].As<Napi::Function>();
+
+    if (tsfn_participant_video_) {
+        tsfn_participant_video_.Release();
+    }
+
+    tsfn_participant_video_ = Napi::ThreadSafeFunction::New(
+        env, callback, "ParticipantVideoCallback", 0, 1
+    );
+
+    client_->setOnParticipantVideo([this](const std::vector<int>& users, bool is_on) {
+        auto callback = [users, is_on](Napi::Env env, Napi::Function jsCallback) {
+            Napi::Array arr = Napi::Array::New(env, users.size());
+            for (size_t i = 0; i < users.size(); i++) {
+                arr.Set(static_cast<uint32_t>(i), Napi::Number::New(env, users[i]));
+            }
+            jsCallback.Call({arr, Napi::Boolean::New(env, is_on)});
+        };
+        tsfn_participant_video_.BlockingCall(callback);
+    });
+
+    return Napi::Boolean::New(env, true);
+}
+
+Napi::Value NodeClient::setOnVideoSubscribed(const Napi::CallbackInfo& info) {
+    Napi::Env env = info.Env();
+    Napi::HandleScope scope(env);
+
+    if (info.Length() < 1 || !info[0].IsFunction()) {
+        Napi::TypeError::New(env, "Function argument expected").ThrowAsJavaScriptException();
+        return env.Null();
+    }
+
+    Napi::Function callback = info[0].As<Napi::Function>();
+
+    if (tsfn_video_subscribed_) {
+        tsfn_video_subscribed_.Release();
+    }
+
+    tsfn_video_subscribed_ = Napi::ThreadSafeFunction::New(
+        env, callback, "VideoSubscribedCallback", 0, 1
+    );
+
+    client_->setOnVideoSubscribed([this](int user_id, int status, const std::string& error) {
+        auto callback = [user_id, status, error](Napi::Env env, Napi::Function jsCallback) {
+            jsCallback.Call({
+                Napi::Number::New(env, user_id),
+                Napi::Number::New(env, status),
+                Napi::String::New(env, error)
+            });
+        };
+        tsfn_video_subscribed_.BlockingCall(callback);
+    });
+
+    return Napi::Boolean::New(env, true);
+}
+
 Napi::Value NodeClient::enableVideo(const Napi::CallbackInfo& info) {
     Napi::Env env = info.Env();
     Napi::HandleScope scope(env);
@@ -661,7 +823,7 @@ Napi::Value NodeClient::enableTranscript(const Napi::CallbackInfo& info) {
     return Napi::Boolean::New(env, true);
 }
 
-NodeClient::NodeClient(const Napi::CallbackInfo& info) 
+NodeClient::NodeClient(const Napi::CallbackInfo& info)
     : Napi::ObjectWrap<NodeClient>(info) {
     Napi::Env env = info.Env();
     Napi::HandleScope scope(env);
@@ -774,6 +936,8 @@ Napi::Object NodeClient::init(Napi::Env env, Napi::Object exports) {
         InstanceMethod("setDeskshareParams", &NodeClient::setDeskshareParams),
         InstanceMethod("setAudioParams", &NodeClient::setAudioParams),
         InstanceMethod("setVideoParams", &NodeClient::setVideoParams),
+        InstanceMethod("setTranscriptParams", &NodeClient::setTranscriptParams),
+        InstanceMethod("setProxy", &NodeClient::setProxy),
         InstanceMethod("onJoinConfirm", &NodeClient::setOnJoinConfirm),
         InstanceMethod("onSessionUpdate", &NodeClient::setOnSessionUpdate),
         InstanceMethod("onUserUpdate", &NodeClient::setOnUserUpdate),
@@ -785,6 +949,9 @@ Napi::Object NodeClient::init(Napi::Env env, Napi::Object exports) {
         InstanceMethod("onEventEx", &NodeClient::setOnEventEx),
         InstanceMethod("subscribeEvent", &NodeClient::subscribeEvent),
         InstanceMethod("unsubscribeEvent", &NodeClient::unsubscribeEvent),
+        InstanceMethod("subscribeVideo", &NodeClient::subscribeVideo),
+        InstanceMethod("onParticipantVideo", &NodeClient::setOnParticipantVideo),
+        InstanceMethod("onVideoSubscribed", &NodeClient::setOnVideoSubscribed),
     });
 
     Napi::FunctionReference* constructor = new Napi::FunctionReference();
@@ -810,21 +977,23 @@ Napi::Object NodeClient::init(Napi::Env env, Napi::Object exports) {
     exports.Set(Napi::String::New(env, "USER_LEAVE"), Napi::Number::New(env, USER_LEAVE));
 
     // Event types for subscribeEvent/unsubscribeEvent (used with onEventEx callback)
-    // These match RTMS_EVENT_TYPE from Zoom's C SDK
-    exports.Set(Napi::String::New(env, "EVENT_UNDEFINED"), Napi::Number::New(env, rtms::Client::EVENT_UNDEFINED));
-    exports.Set(Napi::String::New(env, "EVENT_FIRST_PACKET_TIMESTAMP"), Napi::Number::New(env, rtms::Client::EVENT_FIRST_PACKET_TIMESTAMP));
-    exports.Set(Napi::String::New(env, "EVENT_ACTIVE_SPEAKER_CHANGE"), Napi::Number::New(env, rtms::Client::EVENT_ACTIVE_SPEAKER_CHANGE));
-    exports.Set(Napi::String::New(env, "EVENT_PARTICIPANT_JOIN"), Napi::Number::New(env, rtms::Client::EVENT_PARTICIPANT_JOIN));
-    exports.Set(Napi::String::New(env, "EVENT_PARTICIPANT_LEAVE"), Napi::Number::New(env, rtms::Client::EVENT_PARTICIPANT_LEAVE));
-    exports.Set(Napi::String::New(env, "EVENT_SHARING_START"), Napi::Number::New(env, rtms::Client::EVENT_SHARING_START));
-    exports.Set(Napi::String::New(env, "EVENT_SHARING_STOP"), Napi::Number::New(env, rtms::Client::EVENT_SHARING_STOP));
-    exports.Set(Napi::String::New(env, "EVENT_MEDIA_CONNECTION_INTERRUPTED"), Napi::Number::New(env, rtms::Client::EVENT_MEDIA_CONNECTION_INTERRUPTED));
-    exports.Set(Napi::String::New(env, "EVENT_CONSUMER_ANSWERED"), Napi::Number::New(env, rtms::Client::EVENT_CONSUMER_ANSWERED));
-    exports.Set(Napi::String::New(env, "EVENT_CONSUMER_END"), Napi::Number::New(env, rtms::Client::EVENT_CONSUMER_END));
-    exports.Set(Napi::String::New(env, "EVENT_USER_ANSWERED"), Napi::Number::New(env, rtms::Client::EVENT_USER_ANSWERED));
-    exports.Set(Napi::String::New(env, "EVENT_USER_END"), Napi::Number::New(env, rtms::Client::EVENT_USER_END));
-    exports.Set(Napi::String::New(env, "EVENT_USER_HOLD"), Napi::Number::New(env, rtms::Client::EVENT_USER_HOLD));
-    exports.Set(Napi::String::New(env, "EVENT_USER_UNHOLD"), Napi::Number::New(env, rtms::Client::EVENT_USER_UNHOLD));
+    // These match EVENT_TYPE from Zoom's C SDK
+    exports.Set(Napi::String::New(env, "EVENT_UNDEFINED"),                    Napi::Number::New(env, (int)rtms::EVENT_TYPE::UNDEFINED));
+    exports.Set(Napi::String::New(env, "EVENT_FIRST_PACKET_TIMESTAMP"),       Napi::Number::New(env, (int)rtms::EVENT_TYPE::FIRST_PACKET_TIMESTAMP));
+    exports.Set(Napi::String::New(env, "EVENT_ACTIVE_SPEAKER_CHANGE"),        Napi::Number::New(env, (int)rtms::EVENT_TYPE::ACTIVE_SPEAKER_CHANGE));
+    exports.Set(Napi::String::New(env, "EVENT_PARTICIPANT_JOIN"),             Napi::Number::New(env, (int)rtms::EVENT_TYPE::PARTICIPANT_JOIN));
+    exports.Set(Napi::String::New(env, "EVENT_PARTICIPANT_LEAVE"),            Napi::Number::New(env, (int)rtms::EVENT_TYPE::PARTICIPANT_LEAVE));
+    exports.Set(Napi::String::New(env, "EVENT_SHARING_START"),                Napi::Number::New(env, (int)rtms::EVENT_TYPE::SHARING_START));
+    exports.Set(Napi::String::New(env, "EVENT_SHARING_STOP"),                 Napi::Number::New(env, (int)rtms::EVENT_TYPE::SHARING_STOP));
+    exports.Set(Napi::String::New(env, "EVENT_MEDIA_CONNECTION_INTERRUPTED"), Napi::Number::New(env, (int)rtms::EVENT_TYPE::MEDIA_CONNECTION_INTERRUPTED));
+    exports.Set(Napi::String::New(env, "EVENT_PARTICIPANT_VIDEO_ON"),         Napi::Number::New(env, (int)rtms::EVENT_TYPE::PARTICIPANT_VIDEO_ON));
+    exports.Set(Napi::String::New(env, "EVENT_PARTICIPANT_VIDEO_OFF"),        Napi::Number::New(env, (int)rtms::EVENT_TYPE::PARTICIPANT_VIDEO_OFF));
+    exports.Set(Napi::String::New(env, "EVENT_CONSUMER_ANSWERED"),            Napi::Number::New(env, (int)rtms::ZCC_VOICE_EVENT_TYPE::CONSUMER_ANSWERED));
+    exports.Set(Napi::String::New(env, "EVENT_CONSUMER_END"),                 Napi::Number::New(env, (int)rtms::ZCC_VOICE_EVENT_TYPE::CONSUMER_END));
+    exports.Set(Napi::String::New(env, "EVENT_USER_ANSWERED"),                Napi::Number::New(env, (int)rtms::ZCC_VOICE_EVENT_TYPE::USER_ANSWERED));
+    exports.Set(Napi::String::New(env, "EVENT_USER_END"),                     Napi::Number::New(env, (int)rtms::ZCC_VOICE_EVENT_TYPE::USER_END));
+    exports.Set(Napi::String::New(env, "EVENT_USER_HOLD"),                    Napi::Number::New(env, (int)rtms::ZCC_VOICE_EVENT_TYPE::USER_HOLD));
+    exports.Set(Napi::String::New(env, "EVENT_USER_UNHOLD"),                  Napi::Number::New(env, (int)rtms::ZCC_VOICE_EVENT_TYPE::USER_UNHOLD));
 
     exports.Set(Napi::String::New(env, "RTMS_SDK_FAILURE"), Napi::Number::New(env, RTMS_SDK_FAILURE));
     exports.Set(Napi::String::New(env, "RTMS_SDK_OK"), Napi::Number::New(env, RTMS_SDK_OK));
@@ -839,178 +1008,244 @@ Napi::Object NodeClient::init(Napi::Env env, Napi::Object exports) {
 
     // ===== Audio Parameters =====
 
-    // Audio Content Type
+    // Audio Content Type (MEDIA_CONTENT_TYPE)
     Napi::Object audioContentType = Napi::Object::New(env);
-    audioContentType.Set("UNDEFINED", Napi::Number::New(env, 0));
-    audioContentType.Set("RTP", Napi::Number::New(env, 1));
-    audioContentType.Set("RAW_AUDIO", Napi::Number::New(env, 2));
-    audioContentType.Set("FILE_STREAM", Napi::Number::New(env, 4));
-    audioContentType.Set("TEXT", Napi::Number::New(env, 5));
+    audioContentType.Set("UNDEFINED",   Napi::Number::New(env, (int)rtms::MEDIA_CONTENT_TYPE::UNDEFINED));
+    audioContentType.Set("RTP",         Napi::Number::New(env, (int)rtms::MEDIA_CONTENT_TYPE::RTP));
+    audioContentType.Set("RAW_AUDIO",   Napi::Number::New(env, (int)rtms::MEDIA_CONTENT_TYPE::RAW_AUDIO));
+    audioContentType.Set("FILE_STREAM", Napi::Number::New(env, (int)rtms::MEDIA_CONTENT_TYPE::FILE_STREAM));
+    audioContentType.Set("TEXT",        Napi::Number::New(env, (int)rtms::MEDIA_CONTENT_TYPE::TEXT));
     exports.Set("AudioContentType", audioContentType);
 
-    // Audio Codec
+    // Audio Codec (MEDIA_PAYLOAD_TYPE)
     Napi::Object audioCodec = Napi::Object::New(env);
-    audioCodec.Set("UNDEFINED", Napi::Number::New(env, 0));
-    audioCodec.Set("L16", Napi::Number::New(env, 1));
-    audioCodec.Set("G711", Napi::Number::New(env, 2));
-    audioCodec.Set("G722", Napi::Number::New(env, 3));
-    audioCodec.Set("OPUS", Napi::Number::New(env, 4));
+    audioCodec.Set("UNDEFINED", Napi::Number::New(env, (int)rtms::MEDIA_PAYLOAD_TYPE::UNDEFINED));
+    audioCodec.Set("L16",       Napi::Number::New(env, (int)rtms::MEDIA_PAYLOAD_TYPE::L16));
+    audioCodec.Set("G711",      Napi::Number::New(env, (int)rtms::MEDIA_PAYLOAD_TYPE::G711));
+    audioCodec.Set("G722",      Napi::Number::New(env, (int)rtms::MEDIA_PAYLOAD_TYPE::G722));
+    audioCodec.Set("OPUS",      Napi::Number::New(env, (int)rtms::MEDIA_PAYLOAD_TYPE::OPUS));
     exports.Set("AudioCodec", audioCodec);
 
-    // Audio Sample Rate
+    // Audio Sample Rate (AUDIO_SAMPLE_RATE)
     Napi::Object audioSampleRate = Napi::Object::New(env);
-    audioSampleRate.Set("SR_8K", Napi::Number::New(env, 0));
-    audioSampleRate.Set("SR_16K", Napi::Number::New(env, 1));
-    audioSampleRate.Set("SR_32K", Napi::Number::New(env, 2));
-    audioSampleRate.Set("SR_48K", Napi::Number::New(env, 3));
+    audioSampleRate.Set("SR_8K",  Napi::Number::New(env, (int)rtms::AUDIO_SAMPLE_RATE::SR_8K));
+    audioSampleRate.Set("SR_16K", Napi::Number::New(env, (int)rtms::AUDIO_SAMPLE_RATE::SR_16K));
+    audioSampleRate.Set("SR_32K", Napi::Number::New(env, (int)rtms::AUDIO_SAMPLE_RATE::SR_32K));
+    audioSampleRate.Set("SR_48K", Napi::Number::New(env, (int)rtms::AUDIO_SAMPLE_RATE::SR_48K));
     exports.Set("AudioSampleRate", audioSampleRate);
 
-    // Audio Channel
+    // Audio Channel (AUDIO_CHANNEL)
     Napi::Object audioChannel = Napi::Object::New(env);
-    audioChannel.Set("MONO", Napi::Number::New(env, 1));
-    audioChannel.Set("STEREO", Napi::Number::New(env, 2));
+    audioChannel.Set("MONO",   Napi::Number::New(env, (int)rtms::AUDIO_CHANNEL::MONO));
+    audioChannel.Set("STEREO", Napi::Number::New(env, (int)rtms::AUDIO_CHANNEL::STEREO));
     exports.Set("AudioChannel", audioChannel);
 
-    // Audio Data Option
+    // Audio Data Option (MEDIA_DATA_OPTION)
     Napi::Object audioDataOption = Napi::Object::New(env);
-    audioDataOption.Set("UNDEFINED", Napi::Number::New(env, 0));
-    audioDataOption.Set("AUDIO_MIXED_STREAM", Napi::Number::New(env, 1));
-    audioDataOption.Set("AUDIO_MULTI_STREAMS", Napi::Number::New(env, 2));
+    audioDataOption.Set("UNDEFINED",          Napi::Number::New(env, (int)rtms::MEDIA_DATA_OPTION::UNDEFINED));
+    audioDataOption.Set("AUDIO_MIXED_STREAM",  Napi::Number::New(env, (int)rtms::MEDIA_DATA_OPTION::AUDIO_MIXED_STREAM));
+    audioDataOption.Set("AUDIO_MULTI_STREAMS", Napi::Number::New(env, (int)rtms::MEDIA_DATA_OPTION::AUDIO_MULTI_STREAMS));
     exports.Set("AudioDataOption", audioDataOption);
 
     // ===== Video Parameters =====
 
-    // Video Content Type
+    // Video Content Type (MEDIA_CONTENT_TYPE)
     Napi::Object videoContentType = Napi::Object::New(env);
-    videoContentType.Set("UNDEFINED", Napi::Number::New(env, 0));
-    videoContentType.Set("RTP", Napi::Number::New(env, 1));
-    videoContentType.Set("RAW_VIDEO", Napi::Number::New(env, 3));
-    videoContentType.Set("FILE_STREAM", Napi::Number::New(env, 4));
-    videoContentType.Set("TEXT", Napi::Number::New(env, 5));
+    videoContentType.Set("UNDEFINED",   Napi::Number::New(env, (int)rtms::MEDIA_CONTENT_TYPE::UNDEFINED));
+    videoContentType.Set("RTP",         Napi::Number::New(env, (int)rtms::MEDIA_CONTENT_TYPE::RTP));
+    videoContentType.Set("RAW_VIDEO",   Napi::Number::New(env, (int)rtms::MEDIA_CONTENT_TYPE::RAW_VIDEO));
+    videoContentType.Set("FILE_STREAM", Napi::Number::New(env, (int)rtms::MEDIA_CONTENT_TYPE::FILE_STREAM));
+    videoContentType.Set("TEXT",        Napi::Number::New(env, (int)rtms::MEDIA_CONTENT_TYPE::TEXT));
     exports.Set("VideoContentType", videoContentType);
 
-    // Video Codec
+    // Video Codec (MEDIA_PAYLOAD_TYPE)
     Napi::Object videoCodec = Napi::Object::New(env);
-    videoCodec.Set("UNDEFINED", Napi::Number::New(env, 0));
-    videoCodec.Set("JPG", Napi::Number::New(env, 5));
-    videoCodec.Set("PNG", Napi::Number::New(env, 6));
-    videoCodec.Set("H264", Napi::Number::New(env, 7));
+    videoCodec.Set("UNDEFINED", Napi::Number::New(env, (int)rtms::MEDIA_PAYLOAD_TYPE::UNDEFINED));
+    videoCodec.Set("JPG",       Napi::Number::New(env, (int)rtms::MEDIA_PAYLOAD_TYPE::JPG));
+    videoCodec.Set("PNG",       Napi::Number::New(env, (int)rtms::MEDIA_PAYLOAD_TYPE::PNG));
+    videoCodec.Set("H264",      Napi::Number::New(env, (int)rtms::MEDIA_PAYLOAD_TYPE::H264));
     exports.Set("VideoCodec", videoCodec);
 
-    // Video Resolution
+    // Video Resolution (MEDIA_RESOLUTION)
     Napi::Object videoResolution = Napi::Object::New(env);
-    videoResolution.Set("SD", Napi::Number::New(env, 1));
-    videoResolution.Set("HD", Napi::Number::New(env, 2));
-    videoResolution.Set("FHD", Napi::Number::New(env, 3));
-    videoResolution.Set("QHD", Napi::Number::New(env, 4));
+    videoResolution.Set("SD",  Napi::Number::New(env, (int)rtms::MEDIA_RESOLUTION::SD));
+    videoResolution.Set("HD",  Napi::Number::New(env, (int)rtms::MEDIA_RESOLUTION::HD));
+    videoResolution.Set("FHD", Napi::Number::New(env, (int)rtms::MEDIA_RESOLUTION::FHD));
+    videoResolution.Set("QHD", Napi::Number::New(env, (int)rtms::MEDIA_RESOLUTION::QHD));
     exports.Set("VideoResolution", videoResolution);
 
-    // Video Data Option
+    // Video Data Option (MEDIA_DATA_OPTION)
     Napi::Object videoDataOption = Napi::Object::New(env);
-    videoDataOption.Set("UNDEFINED", Napi::Number::New(env, 0));
-    videoDataOption.Set("VIDEO_SINGLE_ACTIVE_STREAM", Napi::Number::New(env, 3));
-    videoDataOption.Set("VIDEO_MIXED_SPEAKER_VIEW", Napi::Number::New(env, 4));
-    videoDataOption.Set("VIDEO_MIXED_GALLERY_VIEW", Napi::Number::New(env, 5));
+    videoDataOption.Set("UNDEFINED",                      Napi::Number::New(env, (int)rtms::MEDIA_DATA_OPTION::UNDEFINED));
+    videoDataOption.Set("VIDEO_SINGLE_ACTIVE_STREAM",     Napi::Number::New(env, (int)rtms::MEDIA_DATA_OPTION::VIDEO_SINGLE_ACTIVE_STREAM));
+    videoDataOption.Set("VIDEO_MIXED_SPEAKER_VIEW",       Napi::Number::New(env, (int)rtms::MEDIA_DATA_OPTION::VIDEO_SINGLE_INDIVIDUAL_STREAM));
+    videoDataOption.Set("VIDEO_SINGLE_INDIVIDUAL_STREAM", Napi::Number::New(env, (int)rtms::MEDIA_DATA_OPTION::VIDEO_SINGLE_INDIVIDUAL_STREAM));
+    videoDataOption.Set("VIDEO_MIXED_GALLERY_VIEW",       Napi::Number::New(env, (int)rtms::MEDIA_DATA_OPTION::VIDEO_MIXED_GALLERY_VIEW));
     exports.Set("VideoDataOption", videoDataOption);
 
     // ===== Media Types =====
 
-    // Media Data Type 
+    // Media Data Type (MEDIA_DATA_TYPE)
     Napi::Object mediaDataType = Napi::Object::New(env);
-    mediaDataType.Set("UNDEFINED", Napi::Number::New(env, 0));
-    mediaDataType.Set("AUDIO", Napi::Number::New(env, 1));
-    mediaDataType.Set("VIDEO", Napi::Number::New(env, 2));
-    mediaDataType.Set("DESKSHARE", Napi::Number::New(env, 4));
-    mediaDataType.Set("TRANSCRIPT", Napi::Number::New(env, 8));
-    mediaDataType.Set("CHAT", Napi::Number::New(env, 16));
-    mediaDataType.Set("ALL", Napi::Number::New(env, 32));
+    mediaDataType.Set("UNDEFINED",  Napi::Number::New(env, (int)rtms::MEDIA_DATA_TYPE::UNDEFINED));
+    mediaDataType.Set("AUDIO",      Napi::Number::New(env, (int)rtms::MEDIA_DATA_TYPE::AUDIO));
+    mediaDataType.Set("VIDEO",      Napi::Number::New(env, (int)rtms::MEDIA_DATA_TYPE::VIDEO));
+    mediaDataType.Set("DESKSHARE",  Napi::Number::New(env, (int)rtms::MEDIA_DATA_TYPE::DESKSHARE));
+    mediaDataType.Set("TRANSCRIPT", Napi::Number::New(env, (int)rtms::MEDIA_DATA_TYPE::TRANSCRIPT));
+    mediaDataType.Set("CHAT",       Napi::Number::New(env, (int)rtms::MEDIA_DATA_TYPE::CHAT));
+    mediaDataType.Set("ALL",        Napi::Number::New(env, (int)rtms::MEDIA_DATA_TYPE::ALL));
     exports.Set("MediaDataType", mediaDataType);
 
     // ===== Session States =====
 
-    // Session State
+    // Session State (SESSION_STATE)
     Napi::Object sessionState = Napi::Object::New(env);
-    sessionState.Set("INACTIVE", Napi::Number::New(env, 0));
-    sessionState.Set("INITIALIZE", Napi::Number::New(env, 1));
-    sessionState.Set("STARTED", Napi::Number::New(env, 2));
-    sessionState.Set("PAUSED", Napi::Number::New(env, 3));
-    sessionState.Set("RESUMED", Napi::Number::New(env, 4));
-    sessionState.Set("STOPPED", Napi::Number::New(env, 5));
+    sessionState.Set("INACTIVE",   Napi::Number::New(env, (int)rtms::SESSION_STATE::INACTIVE));
+    sessionState.Set("INITIALIZE", Napi::Number::New(env, (int)rtms::SESSION_STATE::INITIALIZE));
+    sessionState.Set("STARTED",    Napi::Number::New(env, (int)rtms::SESSION_STATE::STARTED));
+    sessionState.Set("PAUSED",     Napi::Number::New(env, (int)rtms::SESSION_STATE::PAUSED));
+    sessionState.Set("RESUMED",    Napi::Number::New(env, (int)rtms::SESSION_STATE::RESUMED));
+    sessionState.Set("STOPPED",    Napi::Number::New(env, (int)rtms::SESSION_STATE::STOPPED));
     exports.Set("SessionState", sessionState);
 
-    // Stream State
+    // Stream State (STREAM_STATE)
     Napi::Object streamState = Napi::Object::New(env);
-    streamState.Set("INACTIVE", Napi::Number::New(env, 0));
-    streamState.Set("ACTIVE", Napi::Number::New(env, 1));
-    streamState.Set("INTERRUPTED", Napi::Number::New(env, 2));
-    streamState.Set("TERMINATING", Napi::Number::New(env, 3));
-    streamState.Set("TERMINATED", Napi::Number::New(env, 4));
+    streamState.Set("INACTIVE",    Napi::Number::New(env, (int)rtms::STREAM_STATE::INACTIVE));
+    streamState.Set("ACTIVE",      Napi::Number::New(env, (int)rtms::STREAM_STATE::ACTIVE));
+    streamState.Set("INTERRUPTED", Napi::Number::New(env, (int)rtms::STREAM_STATE::INTERRUPTED));
+    streamState.Set("TERMINATING", Napi::Number::New(env, (int)rtms::STREAM_STATE::TERMINATING));
+    streamState.Set("TERMINATED",  Napi::Number::New(env, (int)rtms::STREAM_STATE::TERMINATED));
+    streamState.Set("PAUSED",      Napi::Number::New(env, (int)rtms::STREAM_STATE::PAUSED));
+    streamState.Set("RESUMED",     Napi::Number::New(env, (int)rtms::STREAM_STATE::RESUMED));
     exports.Set("StreamState", streamState);
 
-    // Event Type (matches RTMS_EVENT_TYPE from Zoom's C SDK)
+    // Event Type (EVENT_TYPE + ZCC_VOICE_EVENT_TYPE for backward compat)
     Napi::Object eventType = Napi::Object::New(env);
-    eventType.Set("UNDEFINED", Napi::Number::New(env, 0));
-    eventType.Set("FIRST_PACKET_TIMESTAMP", Napi::Number::New(env, 1));
-    eventType.Set("ACTIVE_SPEAKER_CHANGE", Napi::Number::New(env, 2));
-    eventType.Set("PARTICIPANT_JOIN", Napi::Number::New(env, 3));
-    eventType.Set("PARTICIPANT_LEAVE", Napi::Number::New(env, 4));
-    eventType.Set("SHARING_START", Napi::Number::New(env, 5));
-    eventType.Set("SHARING_STOP", Napi::Number::New(env, 6));
-    eventType.Set("MEDIA_CONNECTION_INTERRUPTED", Napi::Number::New(env, 7));
-    eventType.Set("CONSUMER_ANSWERED", Napi::Number::New(env, 8));
-    eventType.Set("CONSUMER_END", Napi::Number::New(env, 9));
-    eventType.Set("USER_ANSWERED", Napi::Number::New(env, 10));
-    eventType.Set("USER_END", Napi::Number::New(env, 11));
-    eventType.Set("USER_HOLD", Napi::Number::New(env, 12));
-    eventType.Set("USER_UNHOLD", Napi::Number::New(env, 13));
+    eventType.Set("UNDEFINED",                    Napi::Number::New(env, (int)rtms::EVENT_TYPE::UNDEFINED));
+    eventType.Set("FIRST_PACKET_TIMESTAMP",       Napi::Number::New(env, (int)rtms::EVENT_TYPE::FIRST_PACKET_TIMESTAMP));
+    eventType.Set("ACTIVE_SPEAKER_CHANGE",        Napi::Number::New(env, (int)rtms::EVENT_TYPE::ACTIVE_SPEAKER_CHANGE));
+    eventType.Set("PARTICIPANT_JOIN",             Napi::Number::New(env, (int)rtms::EVENT_TYPE::PARTICIPANT_JOIN));
+    eventType.Set("PARTICIPANT_LEAVE",            Napi::Number::New(env, (int)rtms::EVENT_TYPE::PARTICIPANT_LEAVE));
+    eventType.Set("SHARING_START",                Napi::Number::New(env, (int)rtms::EVENT_TYPE::SHARING_START));
+    eventType.Set("SHARING_STOP",                 Napi::Number::New(env, (int)rtms::EVENT_TYPE::SHARING_STOP));
+    eventType.Set("MEDIA_CONNECTION_INTERRUPTED", Napi::Number::New(env, (int)rtms::EVENT_TYPE::MEDIA_CONNECTION_INTERRUPTED));
+    eventType.Set("PARTICIPANT_VIDEO_ON",         Napi::Number::New(env, (int)rtms::EVENT_TYPE::PARTICIPANT_VIDEO_ON));
+    eventType.Set("PARTICIPANT_VIDEO_OFF",        Napi::Number::New(env, (int)rtms::EVENT_TYPE::PARTICIPANT_VIDEO_OFF));
+    eventType.Set("CONSUMER_ANSWERED",            Napi::Number::New(env, (int)rtms::ZCC_VOICE_EVENT_TYPE::CONSUMER_ANSWERED));
+    eventType.Set("CONSUMER_END",                 Napi::Number::New(env, (int)rtms::ZCC_VOICE_EVENT_TYPE::CONSUMER_END));
+    eventType.Set("USER_ANSWERED",                Napi::Number::New(env, (int)rtms::ZCC_VOICE_EVENT_TYPE::USER_ANSWERED));
+    eventType.Set("USER_END",                     Napi::Number::New(env, (int)rtms::ZCC_VOICE_EVENT_TYPE::USER_END));
+    eventType.Set("USER_HOLD",                    Napi::Number::New(env, (int)rtms::ZCC_VOICE_EVENT_TYPE::USER_HOLD));
+    eventType.Set("USER_UNHOLD",                  Napi::Number::New(env, (int)rtms::ZCC_VOICE_EVENT_TYPE::USER_UNHOLD));
     exports.Set("EventType", eventType);
 
-    // Message Type
+    // Message Type (MESSAGE_TYPE) — complete set of 30 values
     Napi::Object messageType = Napi::Object::New(env);
-    messageType.Set("UNDEFINED", Napi::Number::New(env, 0));
-    messageType.Set("SIGNALING_HAND_SHAKE_REQ", Napi::Number::New(env, 1));
-    messageType.Set("SIGNALING_HAND_SHAKE_RESP", Napi::Number::New(env, 2));
-    messageType.Set("DATA_HAND_SHAKE_REQ", Napi::Number::New(env, 3));
-    messageType.Set("DATA_HAND_SHAKE_RESP", Napi::Number::New(env, 4));
-    messageType.Set("EVENT_SUBSCRIPTION", Napi::Number::New(env, 5));
-    messageType.Set("EVENT_UPDATE", Napi::Number::New(env, 6));
-    messageType.Set("CLIENT_READY_ACK", Napi::Number::New(env, 7));
-    messageType.Set("STREAM_STATE_UPDATE", Napi::Number::New(env, 8));
-    messageType.Set("SESSION_STATE_UPDATE", Napi::Number::New(env, 9));
-    messageType.Set("SESSION_STATE_REQ", Napi::Number::New(env, 10));
-    messageType.Set("SESSION_STATE_RESP", Napi::Number::New(env, 11));
-    messageType.Set("KEEP_ALIVE_REQ", Napi::Number::New(env, 12));
-    messageType.Set("KEEP_ALIVE_RESP", Napi::Number::New(env, 13));
-    messageType.Set("MEDIA_DATA_AUDIO", Napi::Number::New(env, 14));
-    messageType.Set("MEDIA_DATA_VIDEO", Napi::Number::New(env, 15));
-    messageType.Set("MEDIA_DATA_SHARE", Napi::Number::New(env, 16));
-    messageType.Set("MEDIA_DATA_TRANSCRIPT", Napi::Number::New(env, 17));
-    messageType.Set("MEDIA_DATA_CHAT", Napi::Number::New(env, 18));
+    messageType.Set("UNDEFINED",                Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::UNDEFINED));
+    messageType.Set("SIGNALING_HAND_SHAKE_REQ", Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::SIGNALING_HAND_SHAKE_REQ));
+    messageType.Set("SIGNALING_HAND_SHAKE_RESP",Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::SIGNALING_HAND_SHAKE_RESP));
+    messageType.Set("DATA_HAND_SHAKE_REQ",      Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::DATA_HAND_SHAKE_REQ));
+    messageType.Set("DATA_HAND_SHAKE_RESP",     Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::DATA_HAND_SHAKE_RESP));
+    messageType.Set("EVENT_SUBSCRIPTION",       Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::EVENT_SUBSCRIPTION));
+    messageType.Set("EVENT_UPDATE",             Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::EVENT_UPDATE));
+    messageType.Set("CLIENT_READY_ACK",         Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::CLIENT_READY_ACK));
+    messageType.Set("STREAM_STATE_UPDATE",      Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::STREAM_STATE_UPDATE));
+    messageType.Set("SESSION_STATE_UPDATE",     Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::SESSION_STATE_UPDATE));
+    messageType.Set("SESSION_STATE_REQ",        Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::SESSION_STATE_REQ));
+    messageType.Set("SESSION_STATE_RESP",       Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::SESSION_STATE_RESP));
+    messageType.Set("KEEP_ALIVE_REQ",           Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::KEEP_ALIVE_REQ));
+    messageType.Set("KEEP_ALIVE_RESP",          Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::KEEP_ALIVE_RESP));
+    messageType.Set("MEDIA_DATA_AUDIO",         Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::MEDIA_DATA_AUDIO));
+    messageType.Set("MEDIA_DATA_VIDEO",         Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::MEDIA_DATA_VIDEO));
+    messageType.Set("MEDIA_DATA_SHARE",         Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::MEDIA_DATA_SHARE));
+    messageType.Set("MEDIA_DATA_TRANSCRIPT",    Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::MEDIA_DATA_TRANSCRIPT));
+    messageType.Set("MEDIA_DATA_CHAT",          Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::MEDIA_DATA_CHAT));
+    messageType.Set("STREAM_STATE_REQ",         Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::STREAM_STATE_REQ));
+    messageType.Set("STREAM_STATE_RESP",        Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::STREAM_STATE_RESP));
+    messageType.Set("STREAM_CLOSE_REQ",         Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::STREAM_CLOSE_REQ));
+    messageType.Set("STREAM_CLOSE_RESP",        Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::STREAM_CLOSE_RESP));
+    messageType.Set("META_DATA_AUDIO",          Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::META_DATA_AUDIO));
+    messageType.Set("META_DATA_VIDEO",          Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::META_DATA_VIDEO));
+    messageType.Set("META_DATA_SHARE",          Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::META_DATA_SHARE));
+    messageType.Set("META_DATA_TRANSCRIPT",     Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::META_DATA_TRANSCRIPT));
+    messageType.Set("META_DATA_CHAT",           Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::META_DATA_CHAT));
+    messageType.Set("VIDEO_SUBSCRIPTION_REQ",   Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::VIDEO_SUBSCRIPTION_REQ));
+    messageType.Set("VIDEO_SUBSCRIPTION_RESP",  Napi::Number::New(env, (int)rtms::MESSAGE_TYPE::VIDEO_SUBSCRIPTION_RESP));
     exports.Set("MessageType", messageType);
 
-    // Stop Reason
+    // Stop Reason (STOP_REASON) — complete set of 27 values
     Napi::Object stopReason = Napi::Object::New(env);
-    stopReason.Set("UNDEFINED", Napi::Number::New(env, 0));
-    stopReason.Set("STOP_BC_HOST_TRIGGERED", Napi::Number::New(env, 1));
-    stopReason.Set("STOP_BC_USER_TRIGGERED", Napi::Number::New(env, 2));
-    stopReason.Set("STOP_BC_USER_LEFT", Napi::Number::New(env, 3));
-    stopReason.Set("STOP_BC_USER_EJECTED", Napi::Number::New(env, 4));
-    stopReason.Set("STOP_BC_APP_DISABLED_BY_HOST", Napi::Number::New(env, 5));
-    stopReason.Set("STOP_BC_MEETING_ENDED", Napi::Number::New(env, 6));
-    stopReason.Set("STOP_BC_STREAM_CANCELED", Napi::Number::New(env, 7));
-    stopReason.Set("STOP_BC_STREAM_REVOKED", Napi::Number::New(env, 8));
-    stopReason.Set("STOP_BC_ALL_APPS_DISABLED", Napi::Number::New(env, 9));
-    stopReason.Set("STOP_BC_INTERNAL_EXCEPTION", Napi::Number::New(env, 10));
-    stopReason.Set("STOP_BC_CONNECTION_TIMEOUT", Napi::Number::New(env, 11));
-    stopReason.Set("STOP_BC_MEETING_CONNECTION_INTERRUPTED", Napi::Number::New(env, 12));
-    stopReason.Set("STOP_BC_SIGNAL_CONNECTION_INTERRUPTED", Napi::Number::New(env, 13));
-    stopReason.Set("STOP_BC_DATA_CONNECTION_INTERRUPTED", Napi::Number::New(env, 14));
-    stopReason.Set("STOP_BC_SIGNAL_CONNECTION_CLOSED_ABNORMALLY", Napi::Number::New(env, 15));
-    stopReason.Set("STOP_BC_DATA_CONNECTION_CLOSED_ABNORMALLY", Napi::Number::New(env, 16));
-    stopReason.Set("STOP_BC_EXIT_SIGNAL", Napi::Number::New(env, 17));
-    stopReason.Set("STOP_BC_AUTHENTICATION_FAILURE", Napi::Number::New(env, 18));
+    stopReason.Set("UNDEFINED",                               Napi::Number::New(env, (int)rtms::STOP_REASON::UNDEFINED));
+    stopReason.Set("HOST_TRIGGERED",                          Napi::Number::New(env, (int)rtms::STOP_REASON::HOST_TRIGGERED));
+    stopReason.Set("USER_TRIGGERED",                          Napi::Number::New(env, (int)rtms::STOP_REASON::USER_TRIGGERED));
+    stopReason.Set("USER_LEFT",                               Napi::Number::New(env, (int)rtms::STOP_REASON::USER_LEFT));
+    stopReason.Set("USER_EJECTED",                            Napi::Number::New(env, (int)rtms::STOP_REASON::USER_EJECTED));
+    stopReason.Set("HOST_DISABLED_APP",                       Napi::Number::New(env, (int)rtms::STOP_REASON::HOST_DISABLED_APP));
+    stopReason.Set("MEETING_ENDED",                           Napi::Number::New(env, (int)rtms::STOP_REASON::MEETING_ENDED));
+    stopReason.Set("STREAM_CANCELED",                         Napi::Number::New(env, (int)rtms::STOP_REASON::STREAM_CANCELED));
+    stopReason.Set("STREAM_REVOKED",                          Napi::Number::New(env, (int)rtms::STOP_REASON::STREAM_REVOKED));
+    stopReason.Set("ALL_APPS_DISABLED",                       Napi::Number::New(env, (int)rtms::STOP_REASON::ALL_APPS_DISABLED));
+    stopReason.Set("INTERNAL_EXCEPTION",                      Napi::Number::New(env, (int)rtms::STOP_REASON::INTERNAL_EXCEPTION));
+    stopReason.Set("CONNECTION_TIMEOUT",                      Napi::Number::New(env, (int)rtms::STOP_REASON::CONNECTION_TIMEOUT));
+    stopReason.Set("INSTANCE_CONNECTION_INTERRUPTED",         Napi::Number::New(env, (int)rtms::STOP_REASON::INSTANCE_CONNECTION_INTERRUPTED));
+    stopReason.Set("SIGNAL_CONNECTION_INTERRUPTED",           Napi::Number::New(env, (int)rtms::STOP_REASON::SIGNAL_CONNECTION_INTERRUPTED));
+    stopReason.Set("DATA_CONNECTION_INTERRUPTED",             Napi::Number::New(env, (int)rtms::STOP_REASON::DATA_CONNECTION_INTERRUPTED));
+    stopReason.Set("SIGNAL_CONNECTION_CLOSED_ABNORMALLY",     Napi::Number::New(env, (int)rtms::STOP_REASON::SIGNAL_CONNECTION_CLOSED_ABNORMALLY));
+    stopReason.Set("DATA_CONNECTION_CLOSED_ABNORMALLY",       Napi::Number::New(env, (int)rtms::STOP_REASON::DATA_CONNECTION_CLOSED_ABNORMALLY));
+    stopReason.Set("EXIT_SIGNAL",                             Napi::Number::New(env, (int)rtms::STOP_REASON::EXIT_SIGNAL));
+    stopReason.Set("AUTHENTICATION_FAILURE",                  Napi::Number::New(env, (int)rtms::STOP_REASON::AUTHENTICATION_FAILURE));
+    stopReason.Set("AWAIT_RECONNECTION_TIMEOUT",              Napi::Number::New(env, (int)rtms::STOP_REASON::AWAIT_RECONNECTION_TIMEOUT));
+    stopReason.Set("RECEIVER_REQUEST_CLOSE",                  Napi::Number::New(env, (int)rtms::STOP_REASON::RECEIVER_REQUEST_CLOSE));
+    stopReason.Set("CUSTOMER_DISCONNECTED",                   Napi::Number::New(env, (int)rtms::STOP_REASON::CUSTOMER_DISCONNECTED));
+    stopReason.Set("AGENT_DISCONNECTED",                      Napi::Number::New(env, (int)rtms::STOP_REASON::AGENT_DISCONNECTED));
+    stopReason.Set("ADMIN_DISABLED_APP",                      Napi::Number::New(env, (int)rtms::STOP_REASON::ADMIN_DISABLED_APP));
+    stopReason.Set("KEEP_ALIVE_TIMEOUT",                      Napi::Number::New(env, (int)rtms::STOP_REASON::KEEP_ALIVE_TIMEOUT));
+    stopReason.Set("MANUAL_API_TRIGGERED",                    Napi::Number::New(env, (int)rtms::STOP_REASON::MANUAL_API_TRIGGERED));
+    stopReason.Set("STREAMING_NOT_SUPPORTED",                 Napi::Number::New(env, (int)rtms::STOP_REASON::STREAMING_NOT_SUPPORTED));
     exports.Set("StopReason", stopReason);
 
+    // Transcript language constants (mirrors AudioCodec/VideoCodec pattern)
+    Napi::Object transcriptLanguage = Napi::Object::New(env);
+    transcriptLanguage.Set("NONE",                Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::NONE));
+    transcriptLanguage.Set("ARABIC",              Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::ARABIC));
+    transcriptLanguage.Set("BENGALI",             Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::BENGALI));
+    transcriptLanguage.Set("CANTONESE",           Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::CANTONESE));
+    transcriptLanguage.Set("CATALAN",             Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::CATALAN));
+    transcriptLanguage.Set("CHINESE_SIMPLIFIED",  Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::CHINESE_SIMPLIFIED));
+    transcriptLanguage.Set("CHINESE_TRADITIONAL", Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::CHINESE_TRADITIONAL));
+    transcriptLanguage.Set("CZECH",               Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::CZECH));
+    transcriptLanguage.Set("DANISH",              Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::DANISH));
+    transcriptLanguage.Set("DUTCH",               Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::DUTCH));
+    transcriptLanguage.Set("ENGLISH",             Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::ENGLISH));
+    transcriptLanguage.Set("ESTONIAN",            Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::ESTONIAN));
+    transcriptLanguage.Set("FINNISH",             Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::FINNISH));
+    transcriptLanguage.Set("FRENCH_CANADA",       Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::FRENCH_CANADA));
+    transcriptLanguage.Set("FRENCH_FRANCE",       Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::FRENCH_FRANCE));
+    transcriptLanguage.Set("GERMAN",              Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::GERMAN));
+    transcriptLanguage.Set("HEBREW",              Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::HEBREW));
+    transcriptLanguage.Set("HINDI",               Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::HINDI));
+    transcriptLanguage.Set("HUNGARIAN",           Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::HUNGARIAN));
+    transcriptLanguage.Set("INDONESIAN",          Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::INDONESIAN));
+    transcriptLanguage.Set("ITALIAN",             Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::ITALIAN));
+    transcriptLanguage.Set("JAPANESE",            Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::JAPANESE));
+    transcriptLanguage.Set("KOREAN",              Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::KOREAN));
+    transcriptLanguage.Set("MALAY",               Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::MALAY));
+    transcriptLanguage.Set("PERSIAN",             Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::PERSIAN));
+    transcriptLanguage.Set("POLISH",              Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::POLISH));
+    transcriptLanguage.Set("PORTUGUESE",          Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::PORTUGUESE));
+    transcriptLanguage.Set("ROMANIAN",            Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::ROMANIAN));
+    transcriptLanguage.Set("RUSSIAN",             Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::RUSSIAN));
+    transcriptLanguage.Set("SPANISH",             Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::SPANISH));
+    transcriptLanguage.Set("SWEDISH",             Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::SWEDISH));
+    transcriptLanguage.Set("TAGALOG",             Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::TAGALOG));
+    transcriptLanguage.Set("TAMIL",               Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::TAMIL));
+    transcriptLanguage.Set("TELUGU",              Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::TELUGU));
+    transcriptLanguage.Set("THAI",                Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::THAI));
+    transcriptLanguage.Set("TURKISH",             Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::TURKISH));
+    transcriptLanguage.Set("UKRAINIAN",           Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::UKRAINIAN));
+    transcriptLanguage.Set("VIETNAMESE",          Napi::Number::New(env, (int)rtms::TRANSCRIPT_LANGUAGE::VIETNAMESE));
+    exports.Set("TranscriptLanguage", transcriptLanguage);
+
     return exports;
 }
 
diff --git a/src/python.cpp b/src/python.cpp
index 70c731f..cd7a93c 100644
--- a/src/python.cpp
+++ b/src/python.cpp
@@ -22,15 +22,46 @@ using namespace rtms;
  */
 class PyClient {
 public:
-    PyClient() : client_(std::make_unique<Client>()) {
-        // Client created successfully
-    }
+    // Phase 1: safe to call from any thread — no C SDK interaction.
+    PyClient() : client_(nullptr) {}
 
     ~PyClient() {
-        // Clear all callbacks before destruction
         clearCallbacks();
     }
 
+    // Phase 2: allocates the C SDK handle. Must be called from the thread that
+    // will own this client for its entire lifetime (join/poll/release must all
+    // run on the same OS thread as alloc).
+    //
+    // Replays all callbacks and params that were buffered before alloc() so
+    // that the caller can set up the client fully before calling alloc()+join().
+    void alloc() {
+        if (client_) return;  // idempotent
+        client_ = std::make_unique<Client>();
+
+        // Replay buffered callbacks
+        if (!join_confirm_callback_.is_none())   _registerJoinConfirm();
+        if (!session_update_callback_.is_none()) _registerSessionUpdate();
+        if (!user_update_callback_.is_none())    _registerUserUpdate();
+        if (!audio_data_callback_.is_none())     _registerAudioData();
+        if (!video_data_callback_.is_none())     _registerVideoData();
+        if (!deskshare_data_callback_.is_none()) _registerDeskshareData();
+        if (!transcript_data_callback_.is_none()) _registerTranscriptData();
+        if (!leave_callback_.is_none())          _registerLeave();
+        if (!event_ex_callback_.is_none())       _registerEventEx();
+        if (!participant_video_callback_.is_none()) _registerParticipantVideo();
+        if (!video_subscribed_callback_.is_none())  _registerVideoSubscribed();
+
+        // Replay buffered params
+        if (pending_audio_params_)      client_->setAudioParams(*pending_audio_params_);
+        if (pending_video_params_)      client_->setVideoParams(*pending_video_params_);
+        if (pending_deskshare_params_)  client_->setDeskshareParams(*pending_deskshare_params_);
+        if (pending_transcript_params_) client_->setTranscriptParams(*pending_transcript_params_);
+        if (!pending_proxy_type_.empty()) client_->setProxy(pending_proxy_type_, pending_proxy_url_);
+    }
+
+    bool isAllocated() const { return client_ != nullptr; }
+
     // ========================================================================
     // Core Methods
     // ========================================================================
@@ -38,24 +69,40 @@ class PyClient {
     void join(const std::string& uuid, const std::string& stream_id,
              const std::string& signature, const std::string& server_urls,
              int timeout = -1) {
+        if (!client_) throw std::runtime_error("alloc() must be called before join()");
         client_->join(uuid, stream_id, signature, server_urls, timeout);
     }
 
     void poll() {
-        client_->poll();
+        // Release the GIL *before* acquiring poll_mutex_ to prevent deadlock:
+        // release() holds poll_mutex_ while the webhook thread holds the GIL.
+        // If poll() tried to acquire poll_mutex_ while holding the GIL, the two
+        // threads would deadlock. Releasing the GIL first breaks the cycle.
+        py::gil_scoped_release release;
+        std::lock_guard<std::mutex> lk(poll_mutex_);
+        if (client_) client_->poll();
     }
 
     void release() {
+        if (!client_) return;
+        // Hold poll_mutex_ for the entire release sequence so that any in-flight
+        // poll() completes before we tear down the C SDK handle.
+        std::lock_guard<std::mutex> lk(poll_mutex_);
+        // markClosed() sets sdk_opened_=false so that stopCallbacks() calls
+        // setOnAudioData/Video/etc. with empty lambdas without triggering
+        // configure() on an already-dead session (avoids 4 spurious warnings).
+        client_->markClosed();
         stopCallbacks();
         client_->release();
+        client_.reset();  // prevent subsequent poll() from calling into released SDK
     }
 
     std::string uuid() const {
-        return client_->uuid();
+        return client_ ? client_->uuid() : "";
     }
 
     std::string streamId() const {
-        return client_->streamId();
+        return client_ ? client_->streamId() : "";
     }
 
     // ========================================================================
@@ -63,197 +110,286 @@ class PyClient {
     // ========================================================================
 
     void enableAudio(bool enable) {
-        client_->enableAudio(enable);
+        if (client_) client_->enableAudio(enable);
     }
 
     void enableVideo(bool enable) {
-        client_->enableVideo(enable);
+        if (client_) client_->enableVideo(enable);
     }
 
     void enableTranscript(bool enable) {
-        client_->enableTranscript(enable);
+        if (client_) client_->enableTranscript(enable);
     }
 
     void enableDeskshare(bool enable) {
-        client_->enableDeskshare(enable);
+        if (client_) client_->enableDeskshare(enable);
     }
 
     // ========================================================================
     // Parameter Setting Methods
+    // Buffer params pre-alloc; apply immediately post-alloc.
     // ========================================================================
 
     void setAudioParams(const AudioParams& params) {
-        client_->setAudioParams(params);
+        pending_audio_params_ = std::make_unique<AudioParams>(params);
+        if (client_) client_->setAudioParams(params);
     }
 
     void setVideoParams(const VideoParams& params) {
-        client_->setVideoParams(params);
+        pending_video_params_ = std::make_unique<VideoParams>(params);
+        if (client_) client_->setVideoParams(params);
     }
 
     void setDeskshareParams(const DeskshareParams& params) {
-        client_->setDeskshareParams(params);
+        pending_deskshare_params_ = std::make_unique<DeskshareParams>(params);
+        if (client_) client_->setDeskshareParams(params);
+    }
+
+    void setTranscriptParams(const TranscriptParams& params) {
+        pending_transcript_params_ = std::make_unique<TranscriptParams>(params);
+        if (client_) client_->setTranscriptParams(params);
+    }
+
+    void setProxy(const std::string& proxy_type, const std::string& proxy_url) {
+        pending_proxy_type_ = proxy_type;
+        pending_proxy_url_ = proxy_url;
+        if (client_) client_->setProxy(proxy_type, proxy_url);
     }
 
     // ========================================================================
     // Callback Registration Methods
+    // Buffer callback pre-alloc; register with C SDK immediately post-alloc.
     // ========================================================================
 
     void onJoinConfirm(py::function callback) {
         join_confirm_callback_ = callback;
+        if (client_) _registerJoinConfirm();
+    }
+
+    void onSessionUpdate(py::function callback) {
+        session_update_callback_ = callback;
+        if (client_) _registerSessionUpdate();
+    }
+
+    void onUserUpdate(py::function callback) {
+        user_update_callback_ = callback;
+        if (client_) _registerUserUpdate();
+    }
+
+    void onAudioData(py::function callback) {
+        audio_data_callback_ = callback;
+        if (client_) _registerAudioData();
+    }
+
+    void onVideoData(py::function callback) {
+        video_data_callback_ = callback;
+        if (client_) _registerVideoData();
+    }
+
+    void onDeskshareData(py::function callback) {
+        deskshare_data_callback_ = callback;
+        if (client_) _registerDeskshareData();
+    }
+
+    void onTranscriptData(py::function callback) {
+        transcript_data_callback_ = callback;
+        if (client_) _registerTranscriptData();
+    }
+
+    void onLeave(py::function callback) {
+        leave_callback_ = callback;
+        if (client_) _registerLeave();
+    }
+
+    void onEventEx(py::function callback) {
+        event_ex_callback_ = callback;
+        if (client_) _registerEventEx();
+    }
+
+    // ========================================================================
+    // Individual Video Subscription
+    // ========================================================================
+
+    void subscribeVideo(int user_id, bool subscribe) {
+        if (!client_) throw std::runtime_error("alloc() must be called before subscribeVideo()");
+        client_->subscribeVideo(user_id, subscribe);
+    }
+
+    void onParticipantVideo(py::function callback) {
+        participant_video_callback_ = callback;
+        if (client_) _registerParticipantVideo();
+    }
+
+    void onVideoSubscribed(py::function callback) {
+        video_subscribed_callback_ = callback;
+        if (client_) _registerVideoSubscribed();
+    }
+
+    // ========================================================================
+    // Event Subscription Methods
+    // ========================================================================
+
+    void subscribeEvent(const std::vector<int>& events) {
+        if (!client_) {
+            // Buffer for replay after alloc
+            pending_subscriptions_.insert(pending_subscriptions_.end(), events.begin(), events.end());
+            return;
+        }
+        client_->subscribeEvent(events);
+    }
+
+    void unsubscribeEvent(const std::vector<int>& events) {
+        if (client_) client_->unsubscribeEvent(events);
+    }
+
+private:
+    std::unique_ptr<Client> client_;
+    std::mutex poll_mutex_;  // guards poll() vs release() race
+
+    // Python callback storage (buffered pre-alloc, registered post-alloc)
+    py::object join_confirm_callback_ = py::none();
+    py::object session_update_callback_ = py::none();
+    py::object user_update_callback_ = py::none();
+    py::object audio_data_callback_ = py::none();
+    py::object video_data_callback_ = py::none();
+    py::object deskshare_data_callback_ = py::none();
+    py::object transcript_data_callback_ = py::none();
+    py::object leave_callback_ = py::none();
+    py::object event_ex_callback_ = py::none();
+    py::object participant_video_callback_ = py::none();
+    py::object video_subscribed_callback_ = py::none();
+
+    // Param buffers (applied on alloc)
+    std::unique_ptr<AudioParams>      pending_audio_params_;
+    std::unique_ptr<VideoParams>      pending_video_params_;
+    std::unique_ptr<DeskshareParams>  pending_deskshare_params_;
+    std::unique_ptr<TranscriptParams> pending_transcript_params_;
+    std::string pending_proxy_type_;
+    std::string pending_proxy_url_;
+    std::vector<int> pending_subscriptions_;
+
+    // ── Private registration helpers ────────────────────────────────────────
+    // Each helper wires one stored py::object into the C++ Client.
+    // Called from alloc() (replay) and from the public setter (live update).
+
+    void _registerJoinConfirm() {
         client_->setOnJoinConfirm([this](int reason) {
             if (!join_confirm_callback_.is_none()) {
                 py::gil_scoped_acquire acquire;
-                try {
-                    join_confirm_callback_(reason);
-                } catch (const py::error_already_set& e) {
-                    py::print("Error in join confirm callback:", e.what());
-                }
+                try { join_confirm_callback_(reason); }
+                catch (const py::error_already_set& e) { py::print("Error in join_confirm callback:", e.what()); }
             }
         });
     }
 
-    void onSessionUpdate(py::function callback) {
-        session_update_callback_ = callback;
+    void _registerSessionUpdate() {
         client_->setOnSessionUpdate([this](int op, const Session& session) {
             if (!session_update_callback_.is_none()) {
                 py::gil_scoped_acquire acquire;
-                try {
-                    session_update_callback_(op, session);
-                } catch (const py::error_already_set& e) {
-                    py::print("Error in session update callback:", e.what());
-                }
+                try { session_update_callback_(op, session); }
+                catch (const py::error_already_set& e) { py::print("Error in session_update callback:", e.what()); }
             }
         });
     }
 
-    void onUserUpdate(py::function callback) {
-        user_update_callback_ = callback;
+    void _registerUserUpdate() {
         client_->setOnUserUpdate([this](int op, const Participant& participant) {
             if (!user_update_callback_.is_none()) {
                 py::gil_scoped_acquire acquire;
-                try {
-                    user_update_callback_(op, participant);
-                } catch (const py::error_already_set& e) {
-                    py::print("Error in user update callback:", e.what());
-                }
+                try { user_update_callback_(op, participant); }
+                catch (const py::error_already_set& e) { py::print("Error in user_update callback:", e.what()); }
             }
         });
     }
 
-    void onAudioData(py::function callback) {
-        audio_data_callback_ = callback;
+    void _registerAudioData() {
         client_->setOnAudioData([this](const std::vector<uint8_t>& data, uint64_t timestamp, const Metadata& metadata) {
             if (!audio_data_callback_.is_none()) {
                 py::gil_scoped_acquire acquire;
                 try {
                     py::bytes py_data(reinterpret_cast<const char*>(data.data()), data.size());
                     audio_data_callback_(py_data, data.size(), timestamp, metadata);
-                } catch (const py::error_already_set& e) {
-                    py::print("Error in audio data callback:", e.what());
-                }
+                } catch (const py::error_already_set& e) { py::print("Error in audio_data callback:", e.what()); }
             }
         });
     }
 
-    void onVideoData(py::function callback) {
-        video_data_callback_ = callback;
+    void _registerVideoData() {
         client_->setOnVideoData([this](const std::vector<uint8_t>& data, uint64_t timestamp, const Metadata& metadata) {
             if (!video_data_callback_.is_none()) {
                 py::gil_scoped_acquire acquire;
                 try {
                     py::bytes py_data(reinterpret_cast<const char*>(data.data()), data.size());
                     video_data_callback_(py_data, data.size(), timestamp, metadata);
-                } catch (const py::error_already_set& e) {
-                    py::print("Error in video data callback:", e.what());
-                }
+                } catch (const py::error_already_set& e) { py::print("Error in video_data callback:", e.what()); }
             }
         });
     }
 
-    void onDeskshareData(py::function callback) {
-        deskshare_data_callback_ = callback;
+    void _registerDeskshareData() {
         client_->setOnDeskshareData([this](const std::vector<uint8_t>& data, uint64_t timestamp, const Metadata& metadata) {
             if (!deskshare_data_callback_.is_none()) {
                 py::gil_scoped_acquire acquire;
                 try {
                     py::bytes py_data(reinterpret_cast<const char*>(data.data()), data.size());
                     deskshare_data_callback_(py_data, data.size(), timestamp, metadata);
-                } catch (const py::error_already_set& e) {
-                    py::print("Error in deskshare data callback:", e.what());
-                }
+                } catch (const py::error_already_set& e) { py::print("Error in deskshare_data callback:", e.what()); }
             }
         });
     }
 
-    void onTranscriptData(py::function callback) {
-        transcript_data_callback_ = callback;
+    void _registerTranscriptData() {
         client_->setOnTranscriptData([this](const std::vector<uint8_t>& data, uint64_t timestamp, const Metadata& metadata) {
             if (!transcript_data_callback_.is_none()) {
                 py::gil_scoped_acquire acquire;
                 try {
                     py::bytes py_data(reinterpret_cast<const char*>(data.data()), data.size());
                     transcript_data_callback_(py_data, data.size(), timestamp, metadata);
-                } catch (const py::error_already_set& e) {
-                    py::print("Error in transcript data callback:", e.what());
-                }
+                } catch (const py::error_already_set& e) { py::print("Error in transcript_data callback:", e.what()); }
             }
         });
     }
 
-    void onLeave(py::function callback) {
-        leave_callback_ = callback;
+    void _registerLeave() {
         client_->setOnLeave([this](int reason) {
             if (!leave_callback_.is_none()) {
                 py::gil_scoped_acquire acquire;
-                try {
-                    leave_callback_(reason);
-                } catch (const py::error_already_set& e) {
-                    py::print("Error in leave callback:", e.what());
-                }
+                try { leave_callback_(reason); }
+                catch (const py::error_already_set& e) { py::print("Error in leave callback:", e.what()); }
             }
         });
     }
 
-    void onEventEx(py::function callback) {
-        event_ex_callback_ = callback;
+    void _registerEventEx() {
         client_->setOnEventEx([this](const std::string& event_data) {
             if (!event_ex_callback_.is_none()) {
                 py::gil_scoped_acquire acquire;
-                try {
-                    event_ex_callback_(event_data);
-                } catch (const py::error_already_set& e) {
-                    py::print("Error in event ex callback:", e.what());
-                }
+                try { event_ex_callback_(event_data); }
+                catch (const py::error_already_set& e) { py::print("Error in event_ex callback:", e.what()); }
             }
         });
     }
 
-    // ========================================================================
-    // Event Subscription Methods
-    // ========================================================================
-
-    void subscribeEvent(const std::vector<int>& events) {
-        client_->subscribeEvent(events);
+    void _registerParticipantVideo() {
+        client_->setOnParticipantVideo([this](const std::vector<int>& users, bool is_on) {
+            if (!participant_video_callback_.is_none()) {
+                py::gil_scoped_acquire acquire;
+                try { participant_video_callback_(users, is_on); }
+                catch (const py::error_already_set& e) { py::print("Error in participant_video callback:", e.what()); }
+            }
+        });
     }
 
-    void unsubscribeEvent(const std::vector<int>& events) {
-        client_->unsubscribeEvent(events);
+    void _registerVideoSubscribed() {
+        client_->setOnVideoSubscribed([this](int user_id, int status, const std::string& error) {
+            if (!video_subscribed_callback_.is_none()) {
+                py::gil_scoped_acquire acquire;
+                try { video_subscribed_callback_(user_id, status, error); }
+                catch (const py::error_already_set& e) { py::print("Error in video_subscribed callback:", e.what()); }
+            }
+        });
     }
 
-private:
-    std::unique_ptr<Client> client_;
-
-    // Python callback storage
-    py::object join_confirm_callback_ = py::none();
-    py::object session_update_callback_ = py::none();
-    py::object user_update_callback_ = py::none();
-    py::object audio_data_callback_ = py::none();
-    py::object video_data_callback_ = py::none();
-    py::object deskshare_data_callback_ = py::none();
-    py::object transcript_data_callback_ = py::none();
-    py::object leave_callback_ = py::none();
-    py::object event_ex_callback_ = py::none();
-
     void clearCallbacks() {
         join_confirm_callback_ = py::none();
         session_update_callback_ = py::none();
@@ -264,10 +400,11 @@ class PyClient {
         transcript_data_callback_ = py::none();
         leave_callback_ = py::none();
         event_ex_callback_ = py::none();
+        participant_video_callback_ = py::none();
+        video_subscribed_callback_ = py::none();
     }
 
     void stopCallbacks() {
-        // Replace callbacks with no-ops to prevent calling Python during shutdown
         if (client_) {
             client_->setOnJoinConfirm([](int) {});
             client_->setOnSessionUpdate([](int, const Session&) {});
@@ -278,6 +415,8 @@ class PyClient {
             client_->setOnTranscriptData([](const std::vector<uint8_t>&, uint64_t, const Metadata&) {});
             client_->setOnLeave([](int) {});
             client_->setOnEventEx([](const std::string&) {});
+            client_->setOnParticipantVideo([](const std::vector<int>&, bool) {});
+            client_->setOnVideoSubscribed([](int, int, const std::string&) {});
         }
     }
 };
@@ -312,9 +451,25 @@ PYBIND11_MODULE(_rtms, m) {
         .def_property_readonly("id", &Participant::id)
         .def_property_readonly("name", &Participant::name);
 
+    py::class_<AiTargetLanguage>(m, "AiTargetLanguage")
+        .def_property_readonly("lid", &AiTargetLanguage::lid)
+        .def_property_readonly("toneId", &AiTargetLanguage::toneId)
+        .def_property_readonly("voiceId", &AiTargetLanguage::voiceId)
+        .def_property_readonly("engine", &AiTargetLanguage::engine);
+
+    py::class_<AiInterpreter>(m, "AiInterpreter")
+        .def_property_readonly("lid", &AiInterpreter::lid)
+        .def_property_readonly("timestamp", &AiInterpreter::timestamp)
+        .def_property_readonly("channelNum", &AiInterpreter::channelNum)
+        .def_property_readonly("sampleRate", &AiInterpreter::sampleRate)
+        .def_property_readonly("targets", &AiInterpreter::targets);
+
     py::class_<Metadata>(m, "Metadata")
         .def_property_readonly("userName", &Metadata::userName)
-        .def_property_readonly("userId", &Metadata::userId);
+        .def_property_readonly("userId", &Metadata::userId)
+        .def_property_readonly("startTs", &Metadata::startTs)
+        .def_property_readonly("endTs", &Metadata::endTs)
+        .def_property_readonly("aiInterpreter", &Metadata::aiInterpreter);
 
     // ========================================================================
     // Parameter Classes
@@ -326,34 +481,102 @@ PYBIND11_MODULE(_rtms, m) {
              py::arg("content_type"), py::arg("codec"), py::arg("sample_rate"),
              py::arg("channel"), py::arg("data_opt"), py::arg("duration"),
              py::arg("frame_size"))
+        // camelCase (primary — kept for backwards compat)
         .def_property("contentType", &AudioParams::contentType, &AudioParams::setContentType)
-        .def_property("codec", &AudioParams::codec, &AudioParams::setCodec)
-        .def_property("sampleRate", &AudioParams::sampleRate, &AudioParams::setSampleRate)
-        .def_property("channel", &AudioParams::channel, &AudioParams::setChannel)
-        .def_property("dataOpt", &AudioParams::dataOpt, &AudioParams::setDataOpt)
-        .def_property("duration", &AudioParams::duration, &AudioParams::setDuration)
-        .def_property("frameSize", &AudioParams::frameSize, &AudioParams::setFrameSize);
+        .def_property("codec",       &AudioParams::codec,       &AudioParams::setCodec)
+        .def_property("sampleRate",  &AudioParams::sampleRate,  &AudioParams::setSampleRate)
+        .def_property("channel",     &AudioParams::channel,     &AudioParams::setChannel)
+        .def_property("dataOpt",     &AudioParams::dataOpt,     &AudioParams::setDataOpt)
+        .def_property("duration",    &AudioParams::duration,    &AudioParams::setDuration)
+        .def_property("frameSize",   &AudioParams::frameSize,   &AudioParams::setFrameSize)
+        // snake_case aliases
+        .def_property("content_type", &AudioParams::contentType, &AudioParams::setContentType)
+        .def_property("sample_rate",  &AudioParams::sampleRate,  &AudioParams::setSampleRate)
+        .def_property("data_opt",     &AudioParams::dataOpt,     &AudioParams::setDataOpt)
+        .def_property("frame_size",   &AudioParams::frameSize,   &AudioParams::setFrameSize);
 
     py::class_<VideoParams>(m, "VideoParams")
         .def(py::init<>())
         .def(py::init<int, int, int, int, int>(),
              py::arg("content_type"), py::arg("codec"), py::arg("resolution"),
              py::arg("data_opt"), py::arg("fps"))
+        // camelCase (primary — kept for backwards compat)
         .def_property("contentType", &VideoParams::contentType, &VideoParams::setContentType)
-        .def_property("codec", &VideoParams::codec, &VideoParams::setCodec)
-        .def_property("resolution", &VideoParams::resolution, &VideoParams::setResolution)
-        .def_property("dataOpt", &VideoParams::dataOpt, &VideoParams::setDataOpt)
-        .def_property("fps", &VideoParams::fps, &VideoParams::setFps);
+        .def_property("codec",       &VideoParams::codec,       &VideoParams::setCodec)
+        .def_property("resolution",  &VideoParams::resolution,  &VideoParams::setResolution)
+        .def_property("dataOpt",     &VideoParams::dataOpt,     &VideoParams::setDataOpt)
+        .def_property("fps",         &VideoParams::fps,         &VideoParams::setFps)
+        // snake_case aliases
+        .def_property("content_type", &VideoParams::contentType, &VideoParams::setContentType)
+        .def_property("data_opt",     &VideoParams::dataOpt,     &VideoParams::setDataOpt);
 
     py::class_<DeskshareParams>(m, "DeskshareParams")
         .def(py::init<>())
         .def(py::init<int, int, int, int>(),
              py::arg("content_type"), py::arg("codec"), py::arg("resolution"),
              py::arg("fps"))
+        // camelCase (primary — kept for backwards compat)
         .def_property("contentType", &DeskshareParams::contentType, &DeskshareParams::setContentType)
-        .def_property("codec", &DeskshareParams::codec, &DeskshareParams::setCodec)
-        .def_property("resolution", &DeskshareParams::resolution, &DeskshareParams::setResolution)
-        .def_property("fps", &DeskshareParams::fps, &DeskshareParams::setFps);
+        .def_property("codec",       &DeskshareParams::codec,       &DeskshareParams::setCodec)
+        .def_property("resolution",  &DeskshareParams::resolution,  &DeskshareParams::setResolution)
+        .def_property("fps",         &DeskshareParams::fps,         &DeskshareParams::setFps)
+        // snake_case aliases
+        .def_property("content_type", &DeskshareParams::contentType, &DeskshareParams::setContentType)
+        .def_property("data_opt",     &DeskshareParams::dataOpt,     &DeskshareParams::setDataOpt);
+
+    py::class_<TranscriptParams>(m, "TranscriptParams")
+        .def(py::init<>())
+        // snake_case (primary)
+        .def_property("content_type", &TranscriptParams::contentType,  &TranscriptParams::setContentType)
+        .def_property("src_language", &TranscriptParams::srcLanguage,  &TranscriptParams::setSrcLanguage)
+        .def_property("enable_lid",   &TranscriptParams::enableLid,    &TranscriptParams::setEnableLid)
+        // camelCase aliases
+        .def_property("contentType",  &TranscriptParams::contentType,  &TranscriptParams::setContentType)
+        .def_property("srcLanguage",  &TranscriptParams::srcLanguage,  &TranscriptParams::setSrcLanguage)
+        .def_property("enableLid",    &TranscriptParams::enableLid,    &TranscriptParams::setEnableLid);
+
+    // TranscriptLanguage constants dict (matches pattern of AudioCodec, VideoCodec, etc.)
+    // Values sourced from the TRANSCRIPT_LANGUAGE enum in rtms.h
+    py::dict transcriptLanguage;
+    transcriptLanguage["NONE"]                = (int)TRANSCRIPT_LANGUAGE::NONE;
+    transcriptLanguage["ARABIC"]              = (int)TRANSCRIPT_LANGUAGE::ARABIC;
+    transcriptLanguage["BENGALI"]             = (int)TRANSCRIPT_LANGUAGE::BENGALI;
+    transcriptLanguage["CANTONESE"]           = (int)TRANSCRIPT_LANGUAGE::CANTONESE;
+    transcriptLanguage["CATALAN"]             = (int)TRANSCRIPT_LANGUAGE::CATALAN;
+    transcriptLanguage["CHINESE_SIMPLIFIED"]  = (int)TRANSCRIPT_LANGUAGE::CHINESE_SIMPLIFIED;
+    transcriptLanguage["CHINESE_TRADITIONAL"] = (int)TRANSCRIPT_LANGUAGE::CHINESE_TRADITIONAL;
+    transcriptLanguage["CZECH"]               = (int)TRANSCRIPT_LANGUAGE::CZECH;
+    transcriptLanguage["DANISH"]              = (int)TRANSCRIPT_LANGUAGE::DANISH;
+    transcriptLanguage["DUTCH"]               = (int)TRANSCRIPT_LANGUAGE::DUTCH;
+    transcriptLanguage["ENGLISH"]             = (int)TRANSCRIPT_LANGUAGE::ENGLISH;
+    transcriptLanguage["ESTONIAN"]            = (int)TRANSCRIPT_LANGUAGE::ESTONIAN;
+    transcriptLanguage["FINNISH"]             = (int)TRANSCRIPT_LANGUAGE::FINNISH;
+    transcriptLanguage["FRENCH_CANADA"]       = (int)TRANSCRIPT_LANGUAGE::FRENCH_CANADA;
+    transcriptLanguage["FRENCH_FRANCE"]       = (int)TRANSCRIPT_LANGUAGE::FRENCH_FRANCE;
+    transcriptLanguage["GERMAN"]              = (int)TRANSCRIPT_LANGUAGE::GERMAN;
+    transcriptLanguage["HEBREW"]              = (int)TRANSCRIPT_LANGUAGE::HEBREW;
+    transcriptLanguage["HINDI"]               = (int)TRANSCRIPT_LANGUAGE::HINDI;
+    transcriptLanguage["HUNGARIAN"]           = (int)TRANSCRIPT_LANGUAGE::HUNGARIAN;
+    transcriptLanguage["INDONESIAN"]          = (int)TRANSCRIPT_LANGUAGE::INDONESIAN;
+    transcriptLanguage["ITALIAN"]             = (int)TRANSCRIPT_LANGUAGE::ITALIAN;
+    transcriptLanguage["JAPANESE"]            = (int)TRANSCRIPT_LANGUAGE::JAPANESE;
+    transcriptLanguage["KOREAN"]              = (int)TRANSCRIPT_LANGUAGE::KOREAN;
+    transcriptLanguage["MALAY"]               = (int)TRANSCRIPT_LANGUAGE::MALAY;
+    transcriptLanguage["PERSIAN"]             = (int)TRANSCRIPT_LANGUAGE::PERSIAN;
+    transcriptLanguage["POLISH"]              = (int)TRANSCRIPT_LANGUAGE::POLISH;
+    transcriptLanguage["PORTUGUESE"]          = (int)TRANSCRIPT_LANGUAGE::PORTUGUESE;
+    transcriptLanguage["ROMANIAN"]            = (int)TRANSCRIPT_LANGUAGE::ROMANIAN;
+    transcriptLanguage["RUSSIAN"]             = (int)TRANSCRIPT_LANGUAGE::RUSSIAN;
+    transcriptLanguage["SPANISH"]             = (int)TRANSCRIPT_LANGUAGE::SPANISH;
+    transcriptLanguage["SWEDISH"]             = (int)TRANSCRIPT_LANGUAGE::SWEDISH;
+    transcriptLanguage["TAGALOG"]             = (int)TRANSCRIPT_LANGUAGE::TAGALOG;
+    transcriptLanguage["TAMIL"]               = (int)TRANSCRIPT_LANGUAGE::TAMIL;
+    transcriptLanguage["TELUGU"]              = (int)TRANSCRIPT_LANGUAGE::TELUGU;
+    transcriptLanguage["THAI"]                = (int)TRANSCRIPT_LANGUAGE::THAI;
+    transcriptLanguage["TURKISH"]             = (int)TRANSCRIPT_LANGUAGE::TURKISH;
+    transcriptLanguage["UKRAINIAN"]           = (int)TRANSCRIPT_LANGUAGE::UKRAINIAN;
+    transcriptLanguage["VIETNAMESE"]          = (int)TRANSCRIPT_LANGUAGE::VIETNAMESE;
+    m.attr("TranscriptLanguage") = transcriptLanguage;
 
     // ========================================================================
     // Client Class
@@ -368,6 +591,12 @@ PYBIND11_MODULE(_rtms, m) {
              py::arg("ca_path"), py::arg("is_verify_cert") = 1, py::arg("agent") = nullptr)
         .def_static("uninitialize", &Client::uninitialize,
              "Uninitialize the RTMS SDK")
+        .def("alloc", &PyClient::alloc,
+             "Allocate the C SDK handle. Must be called from the thread that will own this "
+             "client (same thread must call join/poll/release). Replays any callbacks and "
+             "params registered before alloc().")
+        .def("is_allocated", &PyClient::isAllocated,
+             "Return True if alloc() has been called for this client")
         .def("join", &PyClient::join,
              "Join an RTMS session",
              py::arg("uuid"), py::arg("stream_id"), py::arg("signature"),
@@ -378,43 +607,107 @@ PYBIND11_MODULE(_rtms, m) {
              "Release client resources")
         .def("uuid", &PyClient::uuid,
              "Get meeting UUID")
+        .def("stream_id", &PyClient::streamId,
+             "Get stream ID")
         .def("streamId", &PyClient::streamId,
              "Get stream ID")
+        .def("enable_audio", &PyClient::enableAudio,
+             "Enable/disable audio streaming")
         .def("enableAudio", &PyClient::enableAudio,
              "Enable/disable audio streaming")
+        .def("enable_video", &PyClient::enableVideo,
+             "Enable/disable video streaming")
         .def("enableVideo", &PyClient::enableVideo,
              "Enable/disable video streaming")
+        .def("enable_transcript", &PyClient::enableTranscript,
+             "Enable/disable transcript streaming")
         .def("enableTranscript", &PyClient::enableTranscript,
              "Enable/disable transcript streaming")
+        .def("enable_deskshare", &PyClient::enableDeskshare,
+             "Enable/disable deskshare streaming")
         .def("enableDeskshare", &PyClient::enableDeskshare,
              "Enable/disable deskshare streaming")
+        .def("set_audio_params", &PyClient::setAudioParams,
+             "Set audio parameters")
         .def("setAudioParams", &PyClient::setAudioParams,
              "Set audio parameters")
+        .def("set_video_params", &PyClient::setVideoParams,
+             "Set video parameters")
         .def("setVideoParams", &PyClient::setVideoParams,
              "Set video parameters")
+        .def("set_deskshare_params", &PyClient::setDeskshareParams,
+             "Set deskshare parameters")
         .def("setDeskshareParams", &PyClient::setDeskshareParams,
              "Set deskshare parameters")
+        .def("set_transcript_params", &PyClient::setTranscriptParams,
+             "Set transcript parameters")
+        .def("setTranscriptParams", &PyClient::setTranscriptParams,
+             "Set transcript parameters")
+        .def("set_proxy", &PyClient::setProxy,
+             "Set proxy for SDK connections",
+             py::arg("proxy_type"), py::arg("proxy_url"))
+        .def("setProxy", &PyClient::setProxy,
+             "Set proxy for SDK connections",
+             py::arg("proxy_type"), py::arg("proxy_url"))
+        .def("on_join_confirm", &PyClient::onJoinConfirm,
+             "Register join confirm callback")
         .def("onJoinConfirm", &PyClient::onJoinConfirm,
              "Register join confirm callback")
+        .def("on_session_update", &PyClient::onSessionUpdate,
+             "Register session update callback")
         .def("onSessionUpdate", &PyClient::onSessionUpdate,
              "Register session update callback")
+        .def("on_user_update", &PyClient::onUserUpdate,
+             "Register user update callback")
         .def("onUserUpdate", &PyClient::onUserUpdate,
              "Register user update callback")
+        .def("on_audio_data", &PyClient::onAudioData,
+             "Register audio data callback")
         .def("onAudioData", &PyClient::onAudioData,
              "Register audio data callback")
+        .def("on_video_data", &PyClient::onVideoData,
+             "Register video data callback")
         .def("onVideoData", &PyClient::onVideoData,
              "Register video data callback")
+        .def("on_deskshare_data", &PyClient::onDeskshareData,
+             "Register deskshare data callback")
         .def("onDeskshareData", &PyClient::onDeskshareData,
              "Register deskshare data callback")
+        .def("on_transcript_data", &PyClient::onTranscriptData,
+             "Register transcript data callback")
         .def("onTranscriptData", &PyClient::onTranscriptData,
              "Register transcript data callback")
+        .def("on_leave", &PyClient::onLeave,
+             "Register leave callback")
         .def("onLeave", &PyClient::onLeave,
              "Register leave callback")
+        .def("on_event_ex", &PyClient::onEventEx,
+             "Register extended event callback")
         .def("onEventEx", &PyClient::onEventEx,
              "Register extended event callback")
+        .def("subscribe_video", &PyClient::subscribeVideo,
+             "Subscribe or unsubscribe from an individual participant's video stream",
+             py::arg("user_id"), py::arg("subscribe"))
+        .def("subscribeVideo", &PyClient::subscribeVideo,
+             "Subscribe or unsubscribe from an individual participant's video stream",
+             py::arg("user_id"), py::arg("subscribe"))
+        .def("on_participant_video", &PyClient::onParticipantVideo,
+             "Register callback for participant video state changes")
+        .def("onParticipantVideo", &PyClient::onParticipantVideo,
+             "Register callback for participant video state changes")
+        .def("on_video_subscribed", &PyClient::onVideoSubscribed,
+             "Register callback for video subscription responses")
+        .def("onVideoSubscribed", &PyClient::onVideoSubscribed,
+             "Register callback for video subscription responses")
+        .def("subscribe_event", &PyClient::subscribeEvent,
+             "Subscribe to receive specific event types",
+             py::arg("events"))
         .def("subscribeEvent", &PyClient::subscribeEvent,
              "Subscribe to receive specific event types",
              py::arg("events"))
+        .def("unsubscribe_event", &PyClient::unsubscribeEvent,
+             "Unsubscribe from specific event types",
+             py::arg("events"))
         .def("unsubscribeEvent", &PyClient::unsubscribeEvent,
              "Unsubscribe from specific event types",
              py::arg("events"));
@@ -450,23 +743,25 @@ PYBIND11_MODULE(_rtms, m) {
     // Constants - Event Types
     // ========================================================================
     // Event Types (for subscribeEvent/unsubscribeEvent - used with onEventEx callback)
-    // These match RTMS_EVENT_TYPE from Zoom's C SDK
-    // ========================================================================
-
-    m.attr("EVENT_UNDEFINED") = py::int_(static_cast<int>(Client::EVENT_UNDEFINED));
-    m.attr("EVENT_FIRST_PACKET_TIMESTAMP") = py::int_(static_cast<int>(Client::EVENT_FIRST_PACKET_TIMESTAMP));
-    m.attr("EVENT_ACTIVE_SPEAKER_CHANGE") = py::int_(static_cast<int>(Client::EVENT_ACTIVE_SPEAKER_CHANGE));
-    m.attr("EVENT_PARTICIPANT_JOIN") = py::int_(static_cast<int>(Client::EVENT_PARTICIPANT_JOIN));
-    m.attr("EVENT_PARTICIPANT_LEAVE") = py::int_(static_cast<int>(Client::EVENT_PARTICIPANT_LEAVE));
-    m.attr("EVENT_SHARING_START") = py::int_(static_cast<int>(Client::EVENT_SHARING_START));
-    m.attr("EVENT_SHARING_STOP") = py::int_(static_cast<int>(Client::EVENT_SHARING_STOP));
-    m.attr("EVENT_MEDIA_CONNECTION_INTERRUPTED") = py::int_(static_cast<int>(Client::EVENT_MEDIA_CONNECTION_INTERRUPTED));
-    m.attr("EVENT_CONSUMER_ANSWERED") = py::int_(static_cast<int>(Client::EVENT_CONSUMER_ANSWERED));
-    m.attr("EVENT_CONSUMER_END") = py::int_(static_cast<int>(Client::EVENT_CONSUMER_END));
-    m.attr("EVENT_USER_ANSWERED") = py::int_(static_cast<int>(Client::EVENT_USER_ANSWERED));
-    m.attr("EVENT_USER_END") = py::int_(static_cast<int>(Client::EVENT_USER_END));
-    m.attr("EVENT_USER_HOLD") = py::int_(static_cast<int>(Client::EVENT_USER_HOLD));
-    m.attr("EVENT_USER_UNHOLD") = py::int_(static_cast<int>(Client::EVENT_USER_UNHOLD));
+    // These match EVENT_TYPE from Zoom's C SDK
+    // ========================================================================
+
+    m.attr("EVENT_UNDEFINED") = py::int_(static_cast<int>(EVENT_TYPE::UNDEFINED));
+    m.attr("EVENT_FIRST_PACKET_TIMESTAMP") = py::int_(static_cast<int>(EVENT_TYPE::FIRST_PACKET_TIMESTAMP));
+    m.attr("EVENT_ACTIVE_SPEAKER_CHANGE") = py::int_(static_cast<int>(EVENT_TYPE::ACTIVE_SPEAKER_CHANGE));
+    m.attr("EVENT_PARTICIPANT_JOIN") = py::int_(static_cast<int>(EVENT_TYPE::PARTICIPANT_JOIN));
+    m.attr("EVENT_PARTICIPANT_LEAVE") = py::int_(static_cast<int>(EVENT_TYPE::PARTICIPANT_LEAVE));
+    m.attr("EVENT_SHARING_START") = py::int_(static_cast<int>(EVENT_TYPE::SHARING_START));
+    m.attr("EVENT_SHARING_STOP") = py::int_(static_cast<int>(EVENT_TYPE::SHARING_STOP));
+    m.attr("EVENT_MEDIA_CONNECTION_INTERRUPTED") = py::int_(static_cast<int>(EVENT_TYPE::MEDIA_CONNECTION_INTERRUPTED));
+    m.attr("EVENT_PARTICIPANT_VIDEO_ON")  = py::int_(static_cast<int>(EVENT_TYPE::PARTICIPANT_VIDEO_ON));
+    m.attr("EVENT_PARTICIPANT_VIDEO_OFF") = py::int_(static_cast<int>(EVENT_TYPE::PARTICIPANT_VIDEO_OFF));
+    m.attr("EVENT_CONSUMER_ANSWERED") = py::int_(static_cast<int>(ZCC_VOICE_EVENT_TYPE::CONSUMER_ANSWERED));
+    m.attr("EVENT_CONSUMER_END") = py::int_(static_cast<int>(ZCC_VOICE_EVENT_TYPE::CONSUMER_END));
+    m.attr("EVENT_USER_ANSWERED") = py::int_(static_cast<int>(ZCC_VOICE_EVENT_TYPE::USER_ANSWERED));
+    m.attr("EVENT_USER_END") = py::int_(static_cast<int>(ZCC_VOICE_EVENT_TYPE::USER_END));
+    m.attr("EVENT_USER_HOLD") = py::int_(static_cast<int>(ZCC_VOICE_EVENT_TYPE::USER_HOLD));
+    m.attr("EVENT_USER_UNHOLD") = py::int_(static_cast<int>(ZCC_VOICE_EVENT_TYPE::USER_UNHOLD));
 
     // ========================================================================
     // Constants - Error Codes
@@ -492,84 +787,87 @@ PYBIND11_MODULE(_rtms, m) {
     // ========================================================================
 
     py::dict audioContentType;
-    audioContentType["UNDEFINED"] = 0;
-    audioContentType["RTP"] = 1;
-    audioContentType["RAW_AUDIO"] = 2;
-    audioContentType["FILE_STREAM"] = 4;
-    audioContentType["TEXT"] = 5;
+    audioContentType["UNDEFINED"]   = (int)MEDIA_CONTENT_TYPE::UNDEFINED;
+    audioContentType["RTP"]         = (int)MEDIA_CONTENT_TYPE::RTP;
+    audioContentType["RAW_AUDIO"]   = (int)MEDIA_CONTENT_TYPE::RAW_AUDIO;
+    audioContentType["FILE_STREAM"] = (int)MEDIA_CONTENT_TYPE::FILE_STREAM;
+    audioContentType["TEXT"]        = (int)MEDIA_CONTENT_TYPE::TEXT;
     m.attr("AudioContentType") = audioContentType;
 
     py::dict audioCodec;
-    audioCodec["UNDEFINED"] = 0;
-    audioCodec["L16"] = 1;
-    audioCodec["G711"] = 2;
-    audioCodec["G722"] = 3;
-    audioCodec["OPUS"] = 4;
+    audioCodec["UNDEFINED"] = (int)MEDIA_PAYLOAD_TYPE::UNDEFINED;
+    audioCodec["L16"]       = (int)MEDIA_PAYLOAD_TYPE::L16;
+    audioCodec["G711"]      = (int)MEDIA_PAYLOAD_TYPE::G711;
+    audioCodec["G722"]      = (int)MEDIA_PAYLOAD_TYPE::G722;
+    audioCodec["OPUS"]      = (int)MEDIA_PAYLOAD_TYPE::OPUS;
     m.attr("AudioCodec") = audioCodec;
 
     py::dict audioSampleRate;
-    audioSampleRate["SR_8K"] = 0;
-    audioSampleRate["SR_16K"] = 1;
-    audioSampleRate["SR_32K"] = 2;
-    audioSampleRate["SR_48K"] = 3;
+    audioSampleRate["SR_8K"]  = (int)AUDIO_SAMPLE_RATE::SR_8K;
+    audioSampleRate["SR_16K"] = (int)AUDIO_SAMPLE_RATE::SR_16K;
+    audioSampleRate["SR_32K"] = (int)AUDIO_SAMPLE_RATE::SR_32K;
+    audioSampleRate["SR_48K"] = (int)AUDIO_SAMPLE_RATE::SR_48K;
     m.attr("AudioSampleRate") = audioSampleRate;
 
     py::dict audioChannel;
-    audioChannel["MONO"] = 1;
-    audioChannel["STEREO"] = 2;
+    audioChannel["MONO"]   = (int)AUDIO_CHANNEL::MONO;
+    audioChannel["STEREO"] = (int)AUDIO_CHANNEL::STEREO;
     m.attr("AudioChannel") = audioChannel;
 
-    py::dict audioDataOption;
-    audioDataOption["UNDEFINED"] = 0;
-    audioDataOption["AUDIO_MIXED_STREAM"] = 1;
-    audioDataOption["AUDIO_MULTI_STREAMS"] = 2;
-    m.attr("AudioDataOption") = audioDataOption;
+    // DataOption — unified dict for all audio and video stream delivery modes.
+    // Mirrors the C++ MEDIA_DATA_OPTION enum directly.
+    // AudioDataOption and VideoDataOption are kept as backward-compat aliases.
+    py::dict dataOption;
+    dataOption["UNDEFINED"]                    = (int)MEDIA_DATA_OPTION::UNDEFINED;
+    dataOption["AUDIO_MIXED_STREAM"]           = (int)MEDIA_DATA_OPTION::AUDIO_MIXED_STREAM;
+    dataOption["AUDIO_MULTI_STREAMS"]          = (int)MEDIA_DATA_OPTION::AUDIO_MULTI_STREAMS;
+    dataOption["VIDEO_SINGLE_ACTIVE_STREAM"]   = (int)MEDIA_DATA_OPTION::VIDEO_SINGLE_ACTIVE_STREAM;
+    dataOption["VIDEO_SINGLE_INDIVIDUAL_STREAM"] = (int)MEDIA_DATA_OPTION::VIDEO_SINGLE_INDIVIDUAL_STREAM;
+    dataOption["VIDEO_MIXED_SPEAKER_VIEW"]     = (int)MEDIA_DATA_OPTION::VIDEO_SINGLE_INDIVIDUAL_STREAM;  // legacy alias
+    dataOption["VIDEO_MIXED_GALLERY_VIEW"]     = (int)MEDIA_DATA_OPTION::VIDEO_MIXED_GALLERY_VIEW;
+    m.attr("DataOption")      = dataOption;
+    m.attr("AudioDataOption") = dataOption;  // legacy alias
+    m.attr("VideoDataOption") = dataOption;  // legacy alias
 
     // ========================================================================
     // Constants - Video Parameters
     // ========================================================================
 
     py::dict videoContentType;
-    videoContentType["UNDEFINED"] = 0;
-    videoContentType["RTP"] = 1;
-    videoContentType["RAW_VIDEO"] = 3;
-    videoContentType["FILE_STREAM"] = 4;
-    videoContentType["TEXT"] = 5;
+    videoContentType["UNDEFINED"]   = (int)MEDIA_CONTENT_TYPE::UNDEFINED;
+    videoContentType["RTP"]         = (int)MEDIA_CONTENT_TYPE::RTP;
+    videoContentType["RAW_VIDEO"]   = (int)MEDIA_CONTENT_TYPE::RAW_VIDEO;
+    videoContentType["FILE_STREAM"] = (int)MEDIA_CONTENT_TYPE::FILE_STREAM;
+    videoContentType["TEXT"]        = (int)MEDIA_CONTENT_TYPE::TEXT;
     m.attr("VideoContentType") = videoContentType;
 
     py::dict videoCodec;
-    videoCodec["UNDEFINED"] = 0;
-    videoCodec["JPG"] = 5;
-    videoCodec["PNG"] = 6;
-    videoCodec["H264"] = 7;
+    videoCodec["UNDEFINED"] = (int)MEDIA_PAYLOAD_TYPE::UNDEFINED;
+    videoCodec["JPG"]       = (int)MEDIA_PAYLOAD_TYPE::JPG;
+    videoCodec["PNG"]       = (int)MEDIA_PAYLOAD_TYPE::PNG;
+    videoCodec["H264"]      = (int)MEDIA_PAYLOAD_TYPE::H264;
     m.attr("VideoCodec") = videoCodec;
 
     py::dict videoResolution;
-    videoResolution["SD"] = 1;
-    videoResolution["HD"] = 2;
-    videoResolution["FHD"] = 3;
-    videoResolution["QHD"] = 4;
+    videoResolution["SD"]  = (int)MEDIA_RESOLUTION::SD;
+    videoResolution["HD"]  = (int)MEDIA_RESOLUTION::HD;
+    videoResolution["FHD"] = (int)MEDIA_RESOLUTION::FHD;
+    videoResolution["QHD"] = (int)MEDIA_RESOLUTION::QHD;
     m.attr("VideoResolution") = videoResolution;
 
-    py::dict videoDataOption;
-    videoDataOption["UNDEFINED"] = 0;
-    videoDataOption["VIDEO_SINGLE_ACTIVE_STREAM"] = 3;
-    videoDataOption["VIDEO_MIXED_SPEAKER_VIEW"] = 4;
-    videoDataOption["VIDEO_MIXED_GALLERY_VIEW"] = 5;
-    m.attr("VideoDataOption") = videoDataOption;
 
     // ========================================================================
     // Constants - Media Data Types
     // ========================================================================
 
     py::dict mediaDataType;
-    mediaDataType["UNDEFINED"] = 0;
-    mediaDataType["AUDIO"] = 1;
-    mediaDataType["VIDEO"] = 2;
-    mediaDataType["DESKSHARE"] = 4;
-    mediaDataType["TRANSCRIPT"] = 8;
-    mediaDataType["CHAT"] = 16;
-    mediaDataType["ALL"] = 31;
+    mediaDataType["UNDEFINED"]  = (int)MEDIA_DATA_TYPE::UNDEFINED;
+    mediaDataType["AUDIO"]      = (int)MEDIA_DATA_TYPE::AUDIO;
+    mediaDataType["VIDEO"]      = (int)MEDIA_DATA_TYPE::VIDEO;
+    mediaDataType["DESKSHARE"]  = (int)MEDIA_DATA_TYPE::DESKSHARE;
+    mediaDataType["TRANSCRIPT"] = (int)MEDIA_DATA_TYPE::TRANSCRIPT;
+    mediaDataType["CHAT"]       = (int)MEDIA_DATA_TYPE::CHAT;
+    mediaDataType["ALL"]        = (int)MEDIA_DATA_TYPE::ALL;
     m.attr("MediaDataType") = mediaDataType;
 
     // ========================================================================
@@ -577,12 +875,12 @@ PYBIND11_MODULE(_rtms, m) {
     // ========================================================================
 
     py::dict sessionState;
-    sessionState["INACTIVE"] = 0;
-    sessionState["INITIALIZE"] = 1;
-    sessionState["STARTED"] = 2;
-    sessionState["PAUSED"] = 3;
-    sessionState["RESUMED"] = 4;
-    sessionState["STOPPED"] = 5;
+    sessionState["INACTIVE"]   = (int)SESSION_STATE::INACTIVE;
+    sessionState["INITIALIZE"] = (int)SESSION_STATE::INITIALIZE;
+    sessionState["STARTED"]    = (int)SESSION_STATE::STARTED;
+    sessionState["PAUSED"]     = (int)SESSION_STATE::PAUSED;
+    sessionState["RESUMED"]    = (int)SESSION_STATE::RESUMED;
+    sessionState["STOPPED"]    = (int)SESSION_STATE::STOPPED;
     m.attr("SessionState") = sessionState;
 
     // ========================================================================
@@ -590,32 +888,37 @@ PYBIND11_MODULE(_rtms, m) {
     // ========================================================================
 
     py::dict streamState;
-    streamState["INACTIVE"] = 0;
-    streamState["ACTIVE"] = 1;
-    streamState["INTERRUPTED"] = 2;
-    streamState["TERMINATING"] = 3;
-    streamState["TERMINATED"] = 4;
+    streamState["INACTIVE"]    = (int)STREAM_STATE::INACTIVE;
+    streamState["ACTIVE"]      = (int)STREAM_STATE::ACTIVE;
+    streamState["INTERRUPTED"] = (int)STREAM_STATE::INTERRUPTED;
+    streamState["TERMINATING"] = (int)STREAM_STATE::TERMINATING;
+    streamState["TERMINATED"]  = (int)STREAM_STATE::TERMINATED;
+    streamState["PAUSED"]      = (int)STREAM_STATE::PAUSED;
+    streamState["RESUMED"]     = (int)STREAM_STATE::RESUMED;
     m.attr("StreamState") = streamState;
 
     // ========================================================================
-    // Constants - Event Type (matches RTMS_EVENT_TYPE from Zoom's C SDK)
+    // Constants - Event Type (matches EVENT_TYPE from Zoom's C SDK)
     // ========================================================================
 
     py::dict eventType;
-    eventType["UNDEFINED"] = 0;
-    eventType["FIRST_PACKET_TIMESTAMP"] = 1;
-    eventType["ACTIVE_SPEAKER_CHANGE"] = 2;
-    eventType["PARTICIPANT_JOIN"] = 3;
-    eventType["PARTICIPANT_LEAVE"] = 4;
-    eventType["SHARING_START"] = 5;
-    eventType["SHARING_STOP"] = 6;
-    eventType["MEDIA_CONNECTION_INTERRUPTED"] = 7;
-    eventType["CONSUMER_ANSWERED"] = 8;
-    eventType["CONSUMER_END"] = 9;
-    eventType["USER_ANSWERED"] = 10;
-    eventType["USER_END"] = 11;
-    eventType["USER_HOLD"] = 12;
-    eventType["USER_UNHOLD"] = 13;
+    eventType["UNDEFINED"]                    = (int)EVENT_TYPE::UNDEFINED;
+    eventType["FIRST_PACKET_TIMESTAMP"]       = (int)EVENT_TYPE::FIRST_PACKET_TIMESTAMP;
+    eventType["ACTIVE_SPEAKER_CHANGE"]        = (int)EVENT_TYPE::ACTIVE_SPEAKER_CHANGE;
+    eventType["PARTICIPANT_JOIN"]             = (int)EVENT_TYPE::PARTICIPANT_JOIN;
+    eventType["PARTICIPANT_LEAVE"]            = (int)EVENT_TYPE::PARTICIPANT_LEAVE;
+    eventType["SHARING_START"]                = (int)EVENT_TYPE::SHARING_START;
+    eventType["SHARING_STOP"]                 = (int)EVENT_TYPE::SHARING_STOP;
+    eventType["MEDIA_CONNECTION_INTERRUPTED"] = (int)EVENT_TYPE::MEDIA_CONNECTION_INTERRUPTED;
+    eventType["PARTICIPANT_VIDEO_ON"]         = (int)EVENT_TYPE::PARTICIPANT_VIDEO_ON;
+    eventType["PARTICIPANT_VIDEO_OFF"]        = (int)EVENT_TYPE::PARTICIPANT_VIDEO_OFF;
+    // ZCC voice events
+    eventType["CONSUMER_ANSWERED"] = (int)ZCC_VOICE_EVENT_TYPE::CONSUMER_ANSWERED;
+    eventType["CONSUMER_END"]      = (int)ZCC_VOICE_EVENT_TYPE::CONSUMER_END;
+    eventType["USER_ANSWERED"]     = (int)ZCC_VOICE_EVENT_TYPE::USER_ANSWERED;
+    eventType["USER_END"]          = (int)ZCC_VOICE_EVENT_TYPE::USER_END;
+    eventType["USER_HOLD"]         = (int)ZCC_VOICE_EVENT_TYPE::USER_HOLD;
+    eventType["USER_UNHOLD"]       = (int)ZCC_VOICE_EVENT_TYPE::USER_UNHOLD;
     m.attr("EventType") = eventType;
 
     // ========================================================================
@@ -623,25 +926,36 @@ PYBIND11_MODULE(_rtms, m) {
     // ========================================================================
 
     py::dict messageType;
-    messageType["UNDEFINED"] = 0;
-    messageType["SIGNALING_HAND_SHAKE_REQ"] = 1;
-    messageType["SIGNALING_HAND_SHAKE_RESP"] = 2;
-    messageType["DATA_HAND_SHAKE_REQ"] = 3;
-    messageType["DATA_HAND_SHAKE_RESP"] = 4;
-    messageType["EVENT_SUBSCRIPTION"] = 5;
-    messageType["EVENT_UPDATE"] = 6;
-    messageType["CLIENT_READY_ACK"] = 7;
-    messageType["STREAM_STATE_UPDATE"] = 8;
-    messageType["SESSION_STATE_UPDATE"] = 9;
-    messageType["SESSION_STATE_REQ"] = 10;
-    messageType["SESSION_STATE_RESP"] = 11;
-    messageType["KEEP_ALIVE_REQ"] = 12;
-    messageType["KEEP_ALIVE_RESP"] = 13;
-    messageType["MEDIA_DATA_AUDIO"] = 14;
-    messageType["MEDIA_DATA_VIDEO"] = 15;
-    messageType["MEDIA_DATA_SHARE"] = 16;
-    messageType["MEDIA_DATA_TRANSCRIPT"] = 17;
-    messageType["MEDIA_DATA_CHAT"] = 18;
+    messageType["UNDEFINED"]                 = (int)MESSAGE_TYPE::UNDEFINED;
+    messageType["SIGNALING_HAND_SHAKE_REQ"]  = (int)MESSAGE_TYPE::SIGNALING_HAND_SHAKE_REQ;
+    messageType["SIGNALING_HAND_SHAKE_RESP"] = (int)MESSAGE_TYPE::SIGNALING_HAND_SHAKE_RESP;
+    messageType["DATA_HAND_SHAKE_REQ"]       = (int)MESSAGE_TYPE::DATA_HAND_SHAKE_REQ;
+    messageType["DATA_HAND_SHAKE_RESP"]      = (int)MESSAGE_TYPE::DATA_HAND_SHAKE_RESP;
+    messageType["EVENT_SUBSCRIPTION"]        = (int)MESSAGE_TYPE::EVENT_SUBSCRIPTION;
+    messageType["EVENT_UPDATE"]              = (int)MESSAGE_TYPE::EVENT_UPDATE;
+    messageType["CLIENT_READY_ACK"]          = (int)MESSAGE_TYPE::CLIENT_READY_ACK;
+    messageType["STREAM_STATE_UPDATE"]       = (int)MESSAGE_TYPE::STREAM_STATE_UPDATE;
+    messageType["SESSION_STATE_UPDATE"]      = (int)MESSAGE_TYPE::SESSION_STATE_UPDATE;
+    messageType["SESSION_STATE_REQ"]         = (int)MESSAGE_TYPE::SESSION_STATE_REQ;
+    messageType["SESSION_STATE_RESP"]        = (int)MESSAGE_TYPE::SESSION_STATE_RESP;
+    messageType["KEEP_ALIVE_REQ"]            = (int)MESSAGE_TYPE::KEEP_ALIVE_REQ;
+    messageType["KEEP_ALIVE_RESP"]           = (int)MESSAGE_TYPE::KEEP_ALIVE_RESP;
+    messageType["MEDIA_DATA_AUDIO"]          = (int)MESSAGE_TYPE::MEDIA_DATA_AUDIO;
+    messageType["MEDIA_DATA_VIDEO"]          = (int)MESSAGE_TYPE::MEDIA_DATA_VIDEO;
+    messageType["MEDIA_DATA_SHARE"]          = (int)MESSAGE_TYPE::MEDIA_DATA_SHARE;
+    messageType["MEDIA_DATA_TRANSCRIPT"]     = (int)MESSAGE_TYPE::MEDIA_DATA_TRANSCRIPT;
+    messageType["MEDIA_DATA_CHAT"]           = (int)MESSAGE_TYPE::MEDIA_DATA_CHAT;
+    messageType["STREAM_STATE_REQ"]          = (int)MESSAGE_TYPE::STREAM_STATE_REQ;
+    messageType["STREAM_STATE_RESP"]         = (int)MESSAGE_TYPE::STREAM_STATE_RESP;
+    messageType["STREAM_CLOSE_REQ"]          = (int)MESSAGE_TYPE::STREAM_CLOSE_REQ;
+    messageType["STREAM_CLOSE_RESP"]         = (int)MESSAGE_TYPE::STREAM_CLOSE_RESP;
+    messageType["META_DATA_AUDIO"]           = (int)MESSAGE_TYPE::META_DATA_AUDIO;
+    messageType["META_DATA_VIDEO"]           = (int)MESSAGE_TYPE::META_DATA_VIDEO;
+    messageType["META_DATA_SHARE"]           = (int)MESSAGE_TYPE::META_DATA_SHARE;
+    messageType["META_DATA_TRANSCRIPT"]      = (int)MESSAGE_TYPE::META_DATA_TRANSCRIPT;
+    messageType["META_DATA_CHAT"]            = (int)MESSAGE_TYPE::META_DATA_CHAT;
+    messageType["VIDEO_SUBSCRIPTION_REQ"]    = (int)MESSAGE_TYPE::VIDEO_SUBSCRIPTION_REQ;
+    messageType["VIDEO_SUBSCRIPTION_RESP"]   = (int)MESSAGE_TYPE::VIDEO_SUBSCRIPTION_RESP;
     m.attr("MessageType") = messageType;
 
     // ========================================================================
@@ -649,24 +963,32 @@ PYBIND11_MODULE(_rtms, m) {
     // ========================================================================
 
     py::dict stopReason;
-    stopReason["UNDEFINED"] = 0;
-    stopReason["STOP_BC_HOST_TRIGGERED"] = 1;
-    stopReason["STOP_BC_USER_TRIGGERED"] = 2;
-    stopReason["STOP_BC_USER_LEFT"] = 3;
-    stopReason["STOP_BC_USER_EJECTED"] = 4;
-    stopReason["STOP_BC_APP_DISABLED_BY_HOST"] = 5;
-    stopReason["STOP_BC_MEETING_ENDED"] = 6;
-    stopReason["STOP_BC_STREAM_CANCELED"] = 7;
-    stopReason["STOP_BC_STREAM_REVOKED"] = 8;
-    stopReason["STOP_BC_ALL_APPS_DISABLED"] = 9;
-    stopReason["STOP_BC_INTERNAL_EXCEPTION"] = 10;
-    stopReason["STOP_BC_CONNECTION_TIMEOUT"] = 11;
-    stopReason["STOP_BC_MEETING_CONNECTION_INTERRUPTED"] = 12;
-    stopReason["STOP_BC_SIGNAL_CONNECTION_INTERRUPTED"] = 13;
-    stopReason["STOP_BC_DATA_CONNECTION_INTERRUPTED"] = 14;
-    stopReason["STOP_BC_SIGNAL_CONNECTION_CLOSED_ABNORMALLY"] = 15;
-    stopReason["STOP_BC_DATA_CONNECTION_CLOSED_ABNORMALLY"] = 16;
-    stopReason["STOP_BC_EXIT_SIGNAL"] = 17;
-    stopReason["STOP_BC_AUTHENTICATION_FAILURE"] = 18;
+    stopReason["UNDEFINED"]                               = (int)STOP_REASON::UNDEFINED;
+    stopReason["HOST_TRIGGERED"]                          = (int)STOP_REASON::HOST_TRIGGERED;
+    stopReason["USER_TRIGGERED"]                          = (int)STOP_REASON::USER_TRIGGERED;
+    stopReason["USER_LEFT"]                               = (int)STOP_REASON::USER_LEFT;
+    stopReason["USER_EJECTED"]                            = (int)STOP_REASON::USER_EJECTED;
+    stopReason["HOST_DISABLED_APP"]                       = (int)STOP_REASON::HOST_DISABLED_APP;
+    stopReason["MEETING_ENDED"]                           = (int)STOP_REASON::MEETING_ENDED;
+    stopReason["STREAM_CANCELED"]                         = (int)STOP_REASON::STREAM_CANCELED;
+    stopReason["STREAM_REVOKED"]                          = (int)STOP_REASON::STREAM_REVOKED;
+    stopReason["ALL_APPS_DISABLED"]                       = (int)STOP_REASON::ALL_APPS_DISABLED;
+    stopReason["INTERNAL_EXCEPTION"]                      = (int)STOP_REASON::INTERNAL_EXCEPTION;
+    stopReason["CONNECTION_TIMEOUT"]                      = (int)STOP_REASON::CONNECTION_TIMEOUT;
+    stopReason["INSTANCE_CONNECTION_INTERRUPTED"]         = (int)STOP_REASON::INSTANCE_CONNECTION_INTERRUPTED;
+    stopReason["SIGNAL_CONNECTION_INTERRUPTED"]           = (int)STOP_REASON::SIGNAL_CONNECTION_INTERRUPTED;
+    stopReason["DATA_CONNECTION_INTERRUPTED"]             = (int)STOP_REASON::DATA_CONNECTION_INTERRUPTED;
+    stopReason["SIGNAL_CONNECTION_CLOSED_ABNORMALLY"]     = (int)STOP_REASON::SIGNAL_CONNECTION_CLOSED_ABNORMALLY;
+    stopReason["DATA_CONNECTION_CLOSED_ABNORMALLY"]       = (int)STOP_REASON::DATA_CONNECTION_CLOSED_ABNORMALLY;
+    stopReason["EXIT_SIGNAL"]                             = (int)STOP_REASON::EXIT_SIGNAL;
+    stopReason["AUTHENTICATION_FAILURE"]                  = (int)STOP_REASON::AUTHENTICATION_FAILURE;
+    stopReason["AWAIT_RECONNECTION_TIMEOUT"]              = (int)STOP_REASON::AWAIT_RECONNECTION_TIMEOUT;
+    stopReason["RECEIVER_REQUEST_CLOSE"]                  = (int)STOP_REASON::RECEIVER_REQUEST_CLOSE;
+    stopReason["CUSTOMER_DISCONNECTED"]                   = (int)STOP_REASON::CUSTOMER_DISCONNECTED;
+    stopReason["AGENT_DISCONNECTED"]                      = (int)STOP_REASON::AGENT_DISCONNECTED;
+    stopReason["ADMIN_DISABLED_APP"]                      = (int)STOP_REASON::ADMIN_DISABLED_APP;
+    stopReason["KEEP_ALIVE_TIMEOUT"]                      = (int)STOP_REASON::KEEP_ALIVE_TIMEOUT;
+    stopReason["MANUAL_API_TRIGGERED"]                    = (int)STOP_REASON::MANUAL_API_TRIGGERED;
+    stopReason["STREAMING_NOT_SUPPORTED"]                 = (int)STOP_REASON::STREAMING_NOT_SUPPORTED;
     m.attr("StopReason") = stopReason;
 }
diff --git a/src/rtms.cpp b/src/rtms.cpp
index 04f1910..d642e25 100644
--- a/src/rtms.cpp
+++ b/src/rtms.cpp
@@ -5,9 +5,6 @@
 
 namespace rtms {
 
-unordered_map<struct rtms_csdk*, Client*> Client::sdk_registry_;
-mutex Client::registry_mutex_;
-
 Exception::Exception(int error_code, const string& message)
     : runtime_error(message), error_code_(error_code) {}
 
@@ -15,18 +12,47 @@ int Exception::code() const noexcept {
     return error_code_;
 }
 
-Metadata::Metadata(const rtms_metadata& metadata)
-    : user_name_(metadata.user_name ? metadata.user_name : ""),
-      user_id_(metadata.user_id) {}
+AiTargetLanguage::AiTargetLanguage(const ai_target_lan& atl)
+    : lid_(atl.lid),
+      tone_id_(atl.toneid),
+      voice_id_(atl.voice_id),
+      engine_(atl.engine) {}
 
-string Metadata::userName() const {
-    return user_name_;
-}
+int AiTargetLanguage::lid() const { return lid_; }
+int AiTargetLanguage::toneId() const { return tone_id_; }
+string AiTargetLanguage::voiceId() const { return voice_id_; }
+string AiTargetLanguage::engine() const { return engine_; }
 
-int Metadata::userId() const {
-    return user_id_;
+AiInterpreter::AiInterpreter(const ai_interpreter& aii)
+    : lid_(aii.lid),
+      timestamp_(aii.timestamp),
+      channel_num_(aii.channel_num),
+      sample_rate_(aii.sample_rate) {
+    int count = (aii.target_size > 0 && aii.target_size < 100) ? aii.target_size : 0;
+    targets_.reserve(count);
+    for (int i = 0; i < count; ++i)
+        targets_.emplace_back(aii.atl[i]);
 }
 
+int AiInterpreter::lid() const { return lid_; }
+uint64_t AiInterpreter::timestamp() const { return timestamp_; }
+int AiInterpreter::channelNum() const { return channel_num_; }
+int AiInterpreter::sampleRate() const { return sample_rate_; }
+const vector<AiTargetLanguage>& AiInterpreter::targets() const { return targets_; }
+
+Metadata::Metadata(const rtms_metadata& metadata)
+    : user_name_(metadata.user_name ? metadata.user_name : ""),
+      user_id_(metadata.user_id),
+      start_ts_(metadata.start_ts),
+      end_ts_(metadata.end_ts),
+      ai_interpreter_(metadata.aii) {}
+
+string Metadata::userName() const { return user_name_; }
+int Metadata::userId() const { return user_id_; }
+uint64_t Metadata::startTs() const { return start_ts_; }
+uint64_t Metadata::endTs() const { return end_ts_; }
+const AiInterpreter& Metadata::aiInterpreter() const { return ai_interpreter_; }
+
 Session::Session(const session_info& info)
     : stat_time_(info.stat_time),
       status_(info.status) {
@@ -126,7 +152,7 @@ AudioParams::AudioParams()
     setDataOpt(2);      // AUDIO_MULTI_STREAMS - enables per-participant audio with userId
 }
 
-AudioParams::AudioParams(int content_type, int codec, int sample_rate, 
+AudioParams::AudioParams(int content_type, int codec, int sample_rate,
                                int channel, int data_opt, int duration, int frame_size)
     : BaseMediaParams(), sample_rate_(sample_rate), channel_(channel),
       duration_(duration), frame_size_(frame_size) {
@@ -225,7 +251,11 @@ void AudioParams::validate() const {
 }
 
 VideoParams::VideoParams()
-    : BaseMediaParams(), resolution_(0), fps_(0) {}
+    : BaseMediaParams(), resolution_((int)MEDIA_RESOLUTION::HD), fps_(30) {
+    setContentType((int)MEDIA_CONTENT_TYPE::RAW_VIDEO);
+    setCodec((int)MEDIA_PAYLOAD_TYPE::H264);
+    setDataOpt((int)MEDIA_DATA_OPTION::VIDEO_SINGLE_ACTIVE_STREAM);
+}
 
 VideoParams::VideoParams(int content_type, int codec, int resolution, int data_opt, int fps)
     : BaseMediaParams(), resolution_(resolution), fps_(fps) {
@@ -251,8 +281,41 @@ int VideoParams::fps() const {
     return fps_;
 }
 
-DeskshareParams::DeskshareParams() 
-    : BaseMediaParams(), resolution_(0), fps_(0) {}
+TranscriptParams::TranscriptParams()
+    : BaseMediaParams(), src_language_(-1), enable_lid_(true) {
+    setContentType(5);  // TEXT
+}
+
+void TranscriptParams::setSrcLanguage(int src_language) {
+    src_language_ = src_language;
+}
+
+void TranscriptParams::setEnableLid(bool enable_lid) {
+    enable_lid_ = enable_lid;
+}
+
+int TranscriptParams::srcLanguage() const {
+    return src_language_;
+}
+
+bool TranscriptParams::enableLid() const {
+    return enable_lid_;
+}
+
+transcript_parameters TranscriptParams::toNative() const {
+    transcript_parameters p{};
+    p.content_type  = contentType();
+    p.src_language  = src_language_;
+    p.enable_lid    = enable_lid_;
+    return p;
+}
+
+DeskshareParams::DeskshareParams()
+    : BaseMediaParams(), resolution_((int)MEDIA_RESOLUTION::HD), fps_(30) {
+    setContentType((int)MEDIA_CONTENT_TYPE::RAW_VIDEO);
+    setCodec((int)MEDIA_PAYLOAD_TYPE::H264);
+    setDataOpt((int)MEDIA_DATA_OPTION::VIDEO_SINGLE_ACTIVE_STREAM);
+}
 
 DeskshareParams::DeskshareParams(int content_type, int codec, int resolution, int fps)
     : BaseMediaParams(), resolution_(resolution), fps_(fps) {
@@ -391,18 +454,35 @@ bool MediaParams::hasVideoParams() const {
     return video_params_ != nullptr;
 }
 
+void MediaParams::setTranscriptParams(const TranscriptParams& transcript_params) {
+    transcript_params_ = make_unique<TranscriptParams>(transcript_params);
+}
+
+const TranscriptParams& MediaParams::transcriptParams() const {
+    if (!transcript_params_) {
+        throw Exception(RTMS_SDK_NOT_EXIST, "Transcript parameters not set");
+    }
+    return *transcript_params_;
+}
+
+bool MediaParams::hasTranscriptParams() const {
+    return transcript_params_ != nullptr;
+}
+
 media_parameters MediaParams::toNative() const {
     media_parameters params;
     memset(&params, 0, sizeof(params));
     params.audio_param = nullptr;
     params.video_param = nullptr;
-    
+    params.ds_param = nullptr;
+    params.tr_param = nullptr;
+
     if (audio_params_) {
         audio_parameters* audio_params = new audio_parameters();
         *audio_params = audio_params_->toNative();
         params.audio_param = audio_params;
     }
-    
+
     if (video_params_) {
         video_parameters* video_params = new video_parameters();
         *video_params = video_params_->toNative();
@@ -414,7 +494,13 @@ media_parameters MediaParams::toNative() const {
         *ds_params = ds_params_->toNative();
         params.ds_param = ds_params;
     }
-    
+
+    if (transcript_params_) {
+        transcript_parameters* tr_params = new transcript_parameters();
+        *tr_params = transcript_params_->toNative();
+        params.tr_param = tr_params;
+    }
+
     return params;
 }
 
@@ -422,25 +508,18 @@ Client::Client()
     : sdk_(nullptr),
       enabled_media_types_(0),
       media_params_updated_(false),
+      sdk_opened_(false),
       join_confirmed_(false) {
-    sdk_ = rtms_alloc();
+    sdk_ = rtms_sdk_provider::instance()->create_sdk();
     if (!sdk_) {
         throw Exception(RTMS_SDK_FAILURE, "Failed to allocate RTMS SDK instance");
     }
-    
-    lock_guard<mutex> lock(registry_mutex_);
-    sdk_registry_[sdk_] = this;
 }
 
 Client::~Client() {
     try {
         if (sdk_) {
-            {
-                lock_guard<mutex> lock(registry_mutex_);
-                sdk_registry_.erase(sdk_);
-            }
-            
-            rtms_release(sdk_);
+            rtms_sdk_provider::instance()->release_sdk(sdk_);
             sdk_ = nullptr;
         }
     } catch (const exception& e) {
@@ -449,46 +528,50 @@ Client::~Client() {
 }
 
 void Client::initialize(const string& ca_path, int is_verify_cert, const char* agent) {
-    // SDK will crash if agent is nullptr, so pass empty string instead
-    const char* agent_str = (agent != nullptr) ? agent : "";
-    int result = rtms_init(ca_path.empty() ? nullptr : ca_path.c_str(), is_verify_cert, agent_str);
+    if (agent && agent[0] != '\0') {
+        g_agent = agent;
+    }
+    int result = rtms_sdk_provider::instance()->init(
+        ca_path.empty() ? nullptr : ca_path.c_str(),
+        is_verify_cert != 0
+    );
     if (result != RTMS_SDK_OK) {
         throw Exception(result, "Failed to initialize RTMS SDK");
     }
 }
 
 void Client::uninitialize() {
-    rtms_uninit();
+    rtms_sdk_provider::instance()->uninit();
 }
 
 void Client::configure(const MediaParams& params, int media_types, bool enable_application_layer_encryption, bool apply_defaults) {
-    // Store the media parameters for future use
+    // Always store the configuration so join() can apply it later
     media_params_ = params;
     enabled_media_types_ = media_types;
     media_params_updated_ = true;
 
-    // If media types are enabled but no params were set, use sensible defaults
-    // This ensures proper metadata attribution (e.g., AUDIO_MULTI_STREAMS for audio)
-    // Skip defaults when called from setAudioParams/setVideoParams/setDeskshareParams
-    // to avoid prematurely filling in default params for other media types
+    // Don't call sdk_->config() until sdk_->open() has been called in join().
+    // Calling config() before open() (e.g. from setOnVideoData) hangs for VIDEO.
+    if (!sdk_opened_) return;
+
+    // If a media type is enabled but no params were provided, fill in defaults.
+    // Skip when called from setAudioParams/setVideoParams/setDeskshareParams (apply_defaults=false).
     if (apply_defaults) {
         if ((media_types & MediaType::AUDIO) && !media_params_.hasAudioParams()) {
 #ifdef RTMS_DEBUG
-            cerr << "[DEBUG CONFIG] Audio enabled but no params set, using defaults (AUDIO_MULTI_STREAMS)" << endl;
+            cerr << "[DEBUG CONFIG] Audio enabled but no params set, using defaults (OPUS/AUDIO_MULTI_STREAMS)" << endl;
 #endif
             media_params_.setAudioParams(AudioParams());
         }
-
         if ((media_types & MediaType::VIDEO) && !media_params_.hasVideoParams()) {
 #ifdef RTMS_DEBUG
-            cerr << "[DEBUG CONFIG] Video enabled but no params set, using defaults" << endl;
+            cerr << "[DEBUG CONFIG] Video enabled but no params set, using defaults (H264/HD/30fps)" << endl;
 #endif
             media_params_.setVideoParams(VideoParams());
         }
-
         if ((media_types & MediaType::DESKSHARE) && !media_params_.hasDeskshareParams()) {
 #ifdef RTMS_DEBUG
-            cerr << "[DEBUG CONFIG] Deskshare enabled but no params set, using defaults" << endl;
+            cerr << "[DEBUG CONFIG] Deskshare enabled but no params set, using defaults (H264/HD/30fps)" << endl;
 #endif
             media_params_.setDeskshareParams(DeskshareParams());
         }
@@ -496,17 +579,17 @@ void Client::configure(const MediaParams& params, int media_types, bool enable_a
 
     if (!media_params_.hasAudioParams() && !media_params_.hasVideoParams() && !media_params_.hasDeskshareParams()) {
 #ifdef RTMS_DEBUG
-        cerr << "[DEBUG CONFIG] Calling rtms_config with NULL params, media_types=" << media_types << endl;
+        cerr << "[DEBUG CONFIG] Calling config with NULL params, media_types=" << media_types << endl;
 #endif
-        int result = rtms_config(sdk_, NULL, media_types, enable_application_layer_encryption ? 1 : 0);
+        int result = sdk_->config(nullptr, media_types, enable_application_layer_encryption ? 1 : 0);
         throwIfError(result, "configure with null params");
         return;
     }
-    
+
     media_parameters native_params = media_params_.toNative();
 
 #ifdef RTMS_DEBUG
-    cerr << "[DEBUG CONFIG] Calling rtms_config with params, media_types=" << media_types << endl;
+    cerr << "[DEBUG CONFIG] Calling config with params, media_types=" << media_types << endl;
     if (native_params.audio_param) {
         cerr << "[DEBUG CONFIG] audio_param: data_opt=" << native_params.audio_param->data_opt
              << " content_type=" << native_params.audio_param->content_type
@@ -516,15 +599,18 @@ void Client::configure(const MediaParams& params, int media_types, bool enable_a
     }
 #endif
 
-    int result = rtms_config(sdk_, &native_params, media_types, enable_application_layer_encryption ? 1 : 0);
-    
+    int result = sdk_->config(&native_params, media_types, enable_application_layer_encryption ? 1 : 0);
+
     if (native_params.audio_param) {
         delete native_params.audio_param;
     }
     if (native_params.video_param) {
         delete native_params.video_param;
     }
-    
+    if (native_params.ds_param) {
+        delete native_params.ds_param;
+    }
+
     throwIfError(result, "configure");
 }
 
@@ -546,13 +632,13 @@ void Client::enableDeskshare(bool enable) {
 
 void Client::updateMediaConfiguration(int mediaType, bool enable) {
 
-    if (enable) { 
+    if (enable) {
         enabled_media_types_ |= mediaType;
     } else {
         enabled_media_types_ &= mediaType;
     }
-    
-    if (sdk_) {
+
+    if (sdk_ && sdk_opened_) {
         try {
             configure(media_params_, enabled_media_types_, false);
         } catch (const Exception& e) {
@@ -576,7 +662,11 @@ void Client::setOnUserUpdate(UserUpdateFn callback) {
         lock_guard<mutex> lock(mutex_);
         user_update_callback_ = std::move(callback);
     }
-    subscribeEvent({EVENT_ACTIVE_SPEAKER_CHANGE, EVENT_PARTICIPANT_JOIN, EVENT_PARTICIPANT_LEAVE});
+    subscribeEvent({
+        (int)EVENT_TYPE::ACTIVE_SPEAKER_CHANGE,
+        (int)EVENT_TYPE::PARTICIPANT_JOIN,
+        (int)EVENT_TYPE::PARTICIPANT_LEAVE,
+    });
 }
 
 void Client::setOnDeskshareData(DsDataFn callback){
@@ -589,21 +679,21 @@ void Client::setOnDeskshareData(DsDataFn callback){
 void Client::setOnAudioData(AudioDataFn callback) {
     lock_guard<mutex> lock(mutex_);
     audio_data_callback_ = std::move(callback);
-    
+
     updateMediaConfiguration(MediaType::AUDIO);
 }
 
 void Client::setOnVideoData(VideoDataFn callback) {
     lock_guard<mutex> lock(mutex_);
     video_data_callback_ = std::move(callback);
-    
+
     updateMediaConfiguration(MediaType::VIDEO);
 }
 
 void Client::setOnTranscriptData(TranscriptDataFn callback) {
     lock_guard<mutex> lock(mutex_);
     transcript_data_callback_ = std::move(callback);
-    
+
     updateMediaConfiguration(MediaType::TRANSCRIPT);
 }
 
@@ -639,7 +729,7 @@ void Client::subscribeEvent(const std::vector<int>& events) {
 
     // Already joined - subscribe immediately (release lock first to avoid holding during SDK call)
     mutex_.unlock();
-    int result = rtms_subscribe_event(sdk_, const_cast<int*>(events.data()), static_cast<int>(events.size()));
+    int result = sdk_->subscribe_event(const_cast<int*>(events.data()), static_cast<int>(events.size()));
     mutex_.lock();
 
     if (result != RTMS_SDK_OK) {
@@ -660,7 +750,7 @@ void Client::unsubscribeEvent(const std::vector<int>& events) {
         throw Exception(RTMS_SDK_INVALID_STATUS, "SDK not initialized");
     }
 
-    int result = rtms_unsubscribe_event(sdk_, const_cast<int*>(events.data()), static_cast<int>(events.size()));
+    int result = sdk_->unsubscribe_event(const_cast<int*>(events.data()), static_cast<int>(events.size()));
     throwIfError(result, "unsubscribe_event");
 
     // Remove from tracked events
@@ -727,33 +817,73 @@ void Client::setAudioParams(const AudioParams& audio_params)
     }
 }
 
-void Client::join(const string& meeting_uuid, const string& rtms_stream_id, 
+void Client::setTranscriptParams(const TranscriptParams& transcript_params)
+{
+    media_params_.setTranscriptParams(transcript_params);
+
+    // If transcript is already enabled, reconfigure with the new params
+    if (enabled_media_types_ & MediaType::TRANSCRIPT) {
+        try {
+            configure(media_params_, enabled_media_types_, false, false);
+        } catch (const Exception& e) {
+            cerr << "Warning: Failed to reconfigure transcript params: " << e.what() << endl;
+        }
+    }
+}
+
+void Client::setProxy(const string& proxy_type, const string& proxy_url)
+{
+    int result = sdk_->set_proxy(proxy_type.c_str(), proxy_url.c_str());
+    if (result != 0) {
+        throw Exception(result, "setProxy failed: Operation failed");
+    }
+}
+
+void Client::subscribeVideo(int user_id, bool subscribe)
+{
+    int result = sdk_->send_subscript_video(user_id, subscribe);
+    throwIfError(result, "subscribeVideo");
+}
+
+void Client::setOnParticipantVideo(ParticipantVideoFn callback)
+{
+    {
+        lock_guard<mutex> lock(mutex_);
+        participant_video_callback_ = std::move(callback);
+    }
+    subscribeEvent({
+        (int)EVENT_TYPE::PARTICIPANT_VIDEO_ON,
+        (int)EVENT_TYPE::PARTICIPANT_VIDEO_OFF,
+    });
+}
+
+void Client::setOnVideoSubscribed(VideoSubscribedFn callback)
+{
+    lock_guard<mutex> lock(mutex_);
+    video_subscribed_callback_ = std::move(callback);
+}
+
+void Client::join(const string& meeting_uuid, const string& rtms_stream_id,
                     const string& signature, const string& server_url, int timeout) {
-    struct rtms_csdk_ops ops;
-    memset(&ops, 0, sizeof(ops));
-    ops.on_join_confirm = &Client::handleJoinConfirm;
-    ops.on_session_update = &Client::handleSessionUpdate;
-    ops.on_user_update = &Client::handleUserUpdate;
-    ops.on_ds_data = &Client::handleDsData;
-    ops.on_audio_data = &Client::handleAudioData;
-    ops.on_video_data = &Client::handleVideoData;
-    ops.on_transcript_data = &Client::handleTranscriptData;
-    ops.on_leave = &Client::handleLeave;
-    ops.on_event_ex = &Client::handleEventEx;
-    
-    int result = rtms_set_callbacks(sdk_, &ops);
-    throwIfError(result, "set_callbacks");
-    
-    if (enabled_media_types_ > 0 && !media_params_updated_) {
+    // Register this client as the sink — replaces the old static callback registry
+    int result = sdk_->open(this);
+    throwIfError(result, "open");
+    sdk_opened_ = true;
+
+    // Apply any media configuration that was registered before join() via
+    // setOnAudioData / setOnVideoData / setVideoParams etc. Those calls stored
+    // the state in media_params_ / enabled_media_types_ but deferred the actual
+    // sdk_->config() call until now (after open).
+    if (enabled_media_types_ > 0) {
         try {
             configure(media_params_, enabled_media_types_, false);
         } catch (const Exception& e) {
             cerr << "Warning: Failed to configure media types before join: " << e.what() << endl;
         }
     }
-    
-    result = rtms_join(sdk_, meeting_uuid.c_str(), rtms_stream_id.c_str(),
-                      signature.c_str(), server_url.c_str(), timeout);
+
+    result = sdk_->join(meeting_uuid.c_str(), rtms_stream_id.c_str(),
+                        signature.c_str(), server_url.c_str(), timeout);
     throwIfError(result, "join");
 
     lock_guard<mutex> lock(mutex_);
@@ -762,33 +892,28 @@ void Client::join(const string& meeting_uuid, const string& rtms_stream_id,
 }
 
 void Client::poll() {
-    int result = rtms_poll(sdk_);
+    int result = sdk_->poll();
     throwIfError(result, "poll");
 }
 
-void Client::release() {
-    int result = rtms_release(sdk_);
+void Client::markClosed() {
+    lock_guard<mutex> lock(mutex_);
+    sdk_opened_ = false;
+}
 
-    // Always clean up, even if release returned an error
-    {
-        lock_guard<mutex> lock(registry_mutex_);
-        sdk_registry_.erase(sdk_);
-    }
+void Client::release() {
+    sdk_->leave(0);
 
-    // Reset join state
     {
         lock_guard<mutex> lock(mutex_);
+        sdk_opened_ = false;
         join_confirmed_ = false;
         pending_event_subscriptions_.clear();
         subscribed_events_.clear();
     }
 
+    rtms_sdk_provider::instance()->release_sdk(sdk_);
     sdk_ = nullptr;
-
-    // RTMS_SDK_NOT_EXIST means the resource was already released — that's fine
-    if (result != RTMS_SDK_OK && result != RTMS_SDK_NOT_EXIST) {
-        throwIfError(result, "release");
-    }
 }
 
 string Client::uuid() const {
@@ -831,23 +956,15 @@ void Client::throwIfError(int result, const string& operation) const {
     }
 }
 
-Client* Client::getClient(struct rtms_csdk* sdk) {
-    lock_guard<mutex> lock(registry_mutex_);
-    auto it = sdk_registry_.find(sdk);
-    if (it == sdk_registry_.end()) {
-        return nullptr;
-    }
-    return it->second;
-}
-
 void Client::processPendingSubscriptions() {
     // Called with mutex_ already held
     if (pending_event_subscriptions_.empty()) return;
 
     // Process all pending subscriptions now that we're joined
-    int result = rtms_subscribe_event(sdk_,
-                                       const_cast<int*>(pending_event_subscriptions_.data()),
-                                       static_cast<int>(pending_event_subscriptions_.size()));
+    int result = sdk_->subscribe_event(
+        const_cast<int*>(pending_event_subscriptions_.data()),
+        static_cast<int>(pending_event_subscriptions_.size())
+    );
 
     if (result == RTMS_SDK_OK) {
         // Move pending to subscribed
@@ -861,121 +978,125 @@ void Client::processPendingSubscriptions() {
     pending_event_subscriptions_.clear();
 }
 
-void Client::handleJoinConfirm(struct rtms_csdk* sdk, int reason) {
-    Client* client = getClient(sdk);
-    if (client) {
-        lock_guard<mutex> lock(client->mutex_);
+// ============================================================================
+// rtms_sdk_sink virtual overrides
+// ============================================================================
 
-        // Mark as joined FIRST
-        client->join_confirmed_ = true;
+void Client::on_join_confirm(int reason) {
+    lock_guard<mutex> lock(mutex_);
 
-        // Process any pending event subscriptions before user callback
-        client->processPendingSubscriptions();
+    // Mark as joined FIRST
+    join_confirmed_ = true;
 
-        // Then invoke user callback
-        if (client->join_confirm_callback_) {
-            client->join_confirm_callback_(reason);
-        }
+    // Process any pending event subscriptions before user callback
+    processPendingSubscriptions();
+
+    // Then invoke user callback
+    if (join_confirm_callback_) {
+        join_confirm_callback_(reason);
     }
 }
 
-void Client::handleSessionUpdate(struct rtms_csdk* sdk, int op, struct session_info* sess) {
-    Client* client = getClient(sdk);
-    if (client && sess) {
-        lock_guard<mutex> lock(client->mutex_);
-        if (client->session_update_callback_) {
+void Client::on_session_update(int op, struct session_info* sess) {
+    if (sess) {
+        lock_guard<mutex> lock(mutex_);
+        if (session_update_callback_) {
             Session session(*sess);
-            client->session_update_callback_(op, session);
+            session_update_callback_(op, session);
         }
     }
 }
 
-void Client::handleUserUpdate(struct rtms_csdk* sdk, int op, struct participant_info* pi) {
-    Client* client = getClient(sdk);
-    if (client && pi) {
-        lock_guard<mutex> lock(client->mutex_);
-        if (client->user_update_callback_) {
+void Client::on_user_update(int op, struct participant_info* pi) {
+    if (pi) {
+        lock_guard<mutex> lock(mutex_);
+        if (user_update_callback_) {
             Participant participant(*pi);
-            client->user_update_callback_(op, participant);
+            user_update_callback_(op, participant);
         }
     }
 }
 
-void Client::handleDsData(rtms_csdk* sdk, unsigned char* buf, int size, uint64_t timestamp, rtms_metadata* md) {
-    Client* client = getClient(sdk);
-    if (client && buf && size > 0 && md) {
-        lock_guard<mutex> lock(client->mutex_);
-        if (client->ds_data_callback_) {
-            vector<uint8_t> data(buf, buf + size);
+void Client::on_ds_data(unsigned char* data_buf, int size, uint64_t timestamp, struct rtms_metadata* md) {
+    if (data_buf && size > 0 && md) {
+        lock_guard<mutex> lock(mutex_);
+        if (ds_data_callback_) {
+            vector<uint8_t> data(data_buf, data_buf + size);
             Metadata metadata(*md);
-            client->ds_data_callback_(data, timestamp, metadata);
+            ds_data_callback_(data, timestamp, metadata);
         }
     }
 }
 
-void Client::handleAudioData(struct rtms_csdk* sdk, unsigned char* buf, int size, uint64_t timestamp, struct rtms_metadata* md) {
-    Client* client = getClient(sdk);
-    if (client && buf && size > 0 && md) {
+void Client::on_audio_data(unsigned char* data_buf, int size, uint64_t timestamp, struct rtms_metadata* md) {
+    if (data_buf && size > 0 && md) {
 #ifdef RTMS_DEBUG
         cerr << "[DEBUG AUDIO] md->user_id=" << md->user_id
              << " md->user_name=" << (md->user_name ? md->user_name : "(null)") << endl;
 #endif
-        lock_guard<mutex> lock(client->mutex_);
-        if (client->audio_data_callback_) {
-            vector<uint8_t> data(buf, buf + size);
+        lock_guard<mutex> lock(mutex_);
+        if (audio_data_callback_) {
+            vector<uint8_t> data(data_buf, data_buf + size);
             Metadata metadata(*md);
-            client->audio_data_callback_(data, timestamp, metadata);
+            audio_data_callback_(data, timestamp, metadata);
         }
     }
 }
 
-void Client::handleVideoData(struct rtms_csdk* sdk, unsigned char* buf, int size, uint64_t timestamp, struct rtms_metadata* md) {
-    Client* client = getClient(sdk);
-    if (client && buf && size > 0 && md) {
-        lock_guard<mutex> lock(client->mutex_);
-        if (client->video_data_callback_) {
-            vector<uint8_t> data(buf, buf + size);
+void Client::on_video_data(unsigned char* data_buf, int size, uint64_t timestamp, struct rtms_metadata* md) {
+    if (data_buf && size > 0 && md) {
+        lock_guard<mutex> lock(mutex_);
+        if (video_data_callback_) {
+            vector<uint8_t> data(data_buf, data_buf + size);
             Metadata metadata(*md);
-            client->video_data_callback_(data, timestamp, metadata);
+            video_data_callback_(data, timestamp, metadata);
         }
     }
 }
 
-void Client::handleTranscriptData(struct rtms_csdk* sdk, unsigned char* buf, int size, uint64_t timestamp, struct rtms_metadata* md) {
-    Client* client = getClient(sdk);
-    if (client && buf && size > 0 && md) {
+void Client::on_transcript_data(unsigned char* data_buf, int size, uint64_t timestamp, struct rtms_metadata* md) {
+    if (data_buf && size > 0 && md) {
 #ifdef RTMS_DEBUG
         cerr << "[DEBUG TRANSCRIPT] md->user_id=" << md->user_id
              << " md->user_name=" << (md->user_name ? md->user_name : "(null)") << endl;
 #endif
-        lock_guard<mutex> lock(client->mutex_);
-        if (client->transcript_data_callback_) {
-            vector<uint8_t> data(buf, buf + size);
+        lock_guard<mutex> lock(mutex_);
+        if (transcript_data_callback_) {
+            vector<uint8_t> data(data_buf, data_buf + size);
             Metadata metadata(*md);
-            client->transcript_data_callback_(data, timestamp, metadata);
+            transcript_data_callback_(data, timestamp, metadata);
         }
     }
 }
 
-void Client::handleLeave(struct rtms_csdk* sdk, int reason) {
-    Client* client = getClient(sdk);
-    if (client) {
-        lock_guard<mutex> lock(client->mutex_);
-        if (client->leave_callback_) {
-            client->leave_callback_(reason);
-        }
+void Client::on_leave(int reason) {
+    lock_guard<mutex> lock(mutex_);
+    if (leave_callback_) {
+        leave_callback_(reason);
     }
 }
 
-void Client::handleEventEx(struct rtms_csdk* sdk, const char* buf, int size) {
-    Client* client = getClient(sdk);
-    if (client && buf && size > 0) {
-        lock_guard<mutex> lock(client->mutex_);
-        if (client->event_ex_callback_) {
-            string event_data(buf, size);
-            client->event_ex_callback_(event_data);
+void Client::on_event_ex(const std::string& compact_str) {
+    if (!compact_str.empty()) {
+        lock_guard<mutex> lock(mutex_);
+        if (event_ex_callback_) {
+            event_ex_callback_(compact_str);
         }
     }
 }
 
-} // namespace rtms
\ No newline at end of file
+void Client::on_participant_video(std::vector<int> users, bool is_on) {
+    lock_guard<mutex> lock(mutex_);
+    if (participant_video_callback_) {
+        participant_video_callback_(users, is_on);
+    }
+}
+
+void Client::on_video_subscript_resp(int user_id, int status, std::string error) {
+    lock_guard<mutex> lock(mutex_);
+    if (video_subscribed_callback_) {
+        video_subscribed_callback_(user_id, status, error);
+    }
+}
+
+} // namespace rtms
diff --git a/src/rtms.h b/src/rtms.h
index f5e6ee1..10b326a 100644
--- a/src/rtms.h
+++ b/src/rtms.h
@@ -1,12 +1,11 @@
 #ifndef RTMS_H
 #define RTMS_H
 
-#include "rtms_csdk.h"
+#include "rtms_sdk.h"
 #include <functional>
 #include <sstream>
 #include <thread>
 #include <mutex>
-#include <unordered_map>
 #include <vector>
 
 using namespace std;
@@ -54,16 +53,56 @@ class Participant {
     string name_;
 };
 
+class AiTargetLanguage {
+public:
+    explicit AiTargetLanguage(const ai_target_lan& atl);
+
+    int lid() const;
+    int toneId() const;
+    string voiceId() const;
+    string engine() const;
+
+private:
+    int lid_;
+    int tone_id_;
+    string voice_id_;
+    string engine_;
+};
+
+class AiInterpreter {
+public:
+    explicit AiInterpreter(const ai_interpreter& aii);
+
+    int lid() const;
+    uint64_t timestamp() const;
+    int channelNum() const;
+    int sampleRate() const;
+    const vector<AiTargetLanguage>& targets() const;
+
+private:
+    int lid_;
+    uint64_t timestamp_;
+    int channel_num_;
+    int sample_rate_;
+    vector<AiTargetLanguage> targets_;
+};
+
 class Metadata {
 public:
     explicit Metadata(const rtms_metadata& metadata);
 
     string userName() const;
     int userId() const;
+    uint64_t startTs() const;
+    uint64_t endTs() const;
+    const AiInterpreter& aiInterpreter() const;
 
 private:
     string user_name_;
     int user_id_;
+    uint64_t start_ts_;
+    uint64_t end_ts_;
+    AiInterpreter ai_interpreter_;
 };
 
 class BaseMediaParams {
@@ -84,6 +123,272 @@ class BaseMediaParams {
     int data_opt_;
 };
 
+// ============================================================================
+// Protocol-level enums — single source of truth for all bindings.
+// Values sourced from https://developers.zoom.us/docs/rtms/data-types/
+// and verified against rtms_common.h where applicable.
+// ============================================================================
+
+// MEDIA_CONTENT_TYPE — shared by audio, video, deskshare, transcript
+enum class MEDIA_CONTENT_TYPE {
+    UNDEFINED   = 0,
+    RTP         = 1,
+    RAW_AUDIO   = 2,
+    RAW_VIDEO   = 3,
+    FILE_STREAM = 4,
+    TEXT        = 5,
+};
+
+// MEDIA_PAYLOAD_TYPE — codec identifiers (shared by audio and video)
+enum class MEDIA_PAYLOAD_TYPE {
+    UNDEFINED = 0,
+    L16       = 1,
+    G711      = 2,
+    G722      = 3,
+    OPUS      = 4,
+    JPG       = 5,
+    PNG       = 6,
+    H264      = 7,
+};
+
+// AUDIO_SAMPLE_RATE
+enum class AUDIO_SAMPLE_RATE {
+    SR_8K  = 0,
+    SR_16K = 1,
+    SR_32K = 2,
+    SR_48K = 3,
+};
+
+// AUDIO_CHANNEL
+enum class AUDIO_CHANNEL {
+    MONO   = 1,
+    STEREO = 2,
+};
+
+// MEDIA_RESOLUTION — for video and deskshare
+enum class MEDIA_RESOLUTION {
+    SD  = 1,
+    HD  = 2,
+    FHD = 3,
+    QHD = 4,
+};
+
+// MEDIA_DATA_OPTION — audio and video stream delivery modes
+// Note: VIDEO_SINGLE_INDIVIDUAL_STREAM is also known as VIDEO_MIXED_SPEAKER_VIEW in older SDK docs.
+enum class MEDIA_DATA_OPTION {
+    UNDEFINED                      = 0,
+    AUDIO_MIXED_STREAM             = 1,
+    AUDIO_MULTI_STREAMS            = 2,
+    VIDEO_SINGLE_ACTIVE_STREAM     = 3,
+    VIDEO_SINGLE_INDIVIDUAL_STREAM = 4,
+    VIDEO_MIXED_GALLERY_VIEW       = 5,
+};
+
+// MEDIA_DATA_TYPE — bitmask of media stream types
+// SDK_ALL = 0x1<<5 = 32 (matches rtms_common.h sdk_type::SDK_ALL)
+enum class MEDIA_DATA_TYPE {
+    UNDEFINED  = 0,
+    AUDIO      = 1,
+    VIDEO      = 2,
+    DESKSHARE  = 4,
+    TRANSCRIPT = 8,
+    CHAT       = 16,
+    ALL        = 32,
+};
+
+// SESSION_STATE
+enum class SESSION_STATE {
+    INACTIVE   = 0,
+    INITIALIZE = 1,
+    STARTED    = 2,
+    PAUSED     = 3,
+    RESUMED    = 4,
+    STOPPED    = 5,
+};
+
+// STREAM_STATE
+enum class STREAM_STATE {
+    INACTIVE    = 0,
+    ACTIVE      = 1,
+    INTERRUPTED = 2,
+    TERMINATING = 3,
+    TERMINATED  = 4,
+    PAUSED      = 5,
+    RESUMED     = 6,
+};
+
+// EVENT_TYPE — standard meeting events for subscribeEvent
+enum class EVENT_TYPE {
+        UNDEFINED                    = 0,
+        FIRST_PACKET_TIMESTAMP       = 1,
+        ACTIVE_SPEAKER_CHANGE        = 2,
+        PARTICIPANT_JOIN             = 3,
+        PARTICIPANT_LEAVE            = 4,
+        SHARING_START                = 5,
+        SHARING_STOP                 = 6,
+        MEDIA_CONNECTION_INTERRUPTED = 7,
+        PARTICIPANT_VIDEO_ON         = 8,
+        PARTICIPANT_VIDEO_OFF        = 9,
+        // ZCC voice events (kept here for backward compatibility)
+        CONSUMER_ANSWERED            = 8,
+        CONSUMER_END                 = 9,
+        USER_ANSWERED                = 10,
+        USER_END                     = 11,
+        USER_HOLD                    = 12,
+        USER_UNHOLD                  = 13,
+};
+
+
+// ZCC_VOICE_EVENT_TYPE — Zoom Contact Center voice events
+enum class ZCC_VOICE_EVENT_TYPE {
+    UNDEFINED                      = 0,
+    CONSUMER_ANSWERED              = 8,
+    CONSUMER_END                   = 9,
+    USER_ANSWERED                  = 10,
+    USER_END                       = 11,
+    USER_HOLD                      = 12,
+    USER_UNHOLD                    = 13,
+    MONITOR_STARTED                = 14,
+    MONITOR_TRANSITIONED           = 15,
+    MONITOR_ENDED                  = 16,
+    TAKEOVER_STARTED               = 17,
+    TRANSFER_INITIATED             = 18,
+    TRANSFER_CANCELED              = 19,
+    TRANSFER_ACCEPTED              = 20,
+    TRANSFER_COMPLETED             = 21,
+    TRANSFER_REJECTED              = 22,
+    TRANSFER_TIMEOUT               = 23,
+    CONFERENCE_CANCELED            = 24,
+    CONFERENCE_PARTICIPANT_CANCELED = 25,
+    CONFERENCE_PARTICIPANT_INVITED = 26,
+    CONFERENCE_PARTICIPANT_REJECTED = 27,
+    CONFERENCE_PARTICIPANT_TIMEOUT = 28,
+};
+
+// MESSAGE_TYPE — WebSocket protocol message types
+enum class MESSAGE_TYPE {
+    UNDEFINED                = 0,
+    SIGNALING_HAND_SHAKE_REQ = 1,
+    SIGNALING_HAND_SHAKE_RESP = 2,
+    DATA_HAND_SHAKE_REQ      = 3,
+    DATA_HAND_SHAKE_RESP     = 4,
+    EVENT_SUBSCRIPTION       = 5,
+    EVENT_UPDATE             = 6,
+    CLIENT_READY_ACK         = 7,
+    STREAM_STATE_UPDATE      = 8,
+    SESSION_STATE_UPDATE     = 9,
+    SESSION_STATE_REQ        = 10,
+    SESSION_STATE_RESP       = 11,
+    KEEP_ALIVE_REQ           = 12,
+    KEEP_ALIVE_RESP          = 13,
+    MEDIA_DATA_AUDIO         = 14,
+    MEDIA_DATA_VIDEO         = 15,
+    MEDIA_DATA_SHARE         = 16,
+    MEDIA_DATA_TRANSCRIPT    = 17,
+    MEDIA_DATA_CHAT          = 18,
+    STREAM_STATE_REQ         = 19,
+    STREAM_STATE_RESP        = 20,
+    STREAM_CLOSE_REQ         = 21,
+    STREAM_CLOSE_RESP        = 22,
+    META_DATA_AUDIO          = 23,
+    META_DATA_VIDEO          = 24,
+    META_DATA_SHARE          = 25,
+    META_DATA_TRANSCRIPT     = 26,
+    META_DATA_CHAT           = 27,
+    VIDEO_SUBSCRIPTION_REQ   = 28,
+    VIDEO_SUBSCRIPTION_RESP  = 29,
+};
+
+// STOP_REASON — reasons for stream termination
+enum class STOP_REASON {
+    UNDEFINED                              = 0,
+    HOST_TRIGGERED                         = 1,
+    USER_TRIGGERED                         = 2,
+    USER_LEFT                              = 3,
+    USER_EJECTED                           = 4,
+    HOST_DISABLED_APP                      = 5,
+    MEETING_ENDED                          = 6,
+    STREAM_CANCELED                        = 7,
+    STREAM_REVOKED                         = 8,
+    ALL_APPS_DISABLED                      = 9,
+    INTERNAL_EXCEPTION                     = 10,
+    CONNECTION_TIMEOUT                     = 11,
+    INSTANCE_CONNECTION_INTERRUPTED        = 12,
+    SIGNAL_CONNECTION_INTERRUPTED          = 13,
+    DATA_CONNECTION_INTERRUPTED            = 14,
+    SIGNAL_CONNECTION_CLOSED_ABNORMALLY    = 15,
+    DATA_CONNECTION_CLOSED_ABNORMALLY      = 16,
+    EXIT_SIGNAL                            = 17,
+    AUTHENTICATION_FAILURE                 = 18,
+    AWAIT_RECONNECTION_TIMEOUT             = 19,
+    RECEIVER_REQUEST_CLOSE                 = 20,
+    CUSTOMER_DISCONNECTED                  = 21,
+    AGENT_DISCONNECTED                     = 22,
+    ADMIN_DISABLED_APP                     = 23,
+    KEEP_ALIVE_TIMEOUT                     = 24,
+    MANUAL_API_TRIGGERED                   = 25,
+    STREAMING_NOT_SUPPORTED                = 26,
+};
+
+// TRANSCRIPT_LANGUAGE — source language IDs for transcript configuration.
+// Use NONE (-1) to enable automatic language detection (LID).
+enum class TRANSCRIPT_LANGUAGE {
+    NONE                = -1,
+    ARABIC              = 0,
+    BENGALI             = 1,
+    CANTONESE           = 2,
+    CATALAN             = 3,
+    CHINESE_SIMPLIFIED  = 4,
+    CHINESE_TRADITIONAL = 5,
+    CZECH               = 6,
+    DANISH              = 7,
+    DUTCH               = 8,
+    ENGLISH             = 9,
+    ESTONIAN            = 10,
+    FINNISH             = 11,
+    FRENCH_CANADA       = 12,
+    FRENCH_FRANCE       = 13,
+    GERMAN              = 14,
+    HEBREW              = 15,
+    HINDI               = 16,
+    HUNGARIAN           = 17,
+    INDONESIAN          = 18,
+    ITALIAN             = 19,
+    JAPANESE            = 20,
+    KOREAN              = 21,
+    MALAY               = 22,
+    PERSIAN             = 23,
+    POLISH              = 24,
+    PORTUGUESE          = 25,
+    ROMANIAN            = 26,
+    RUSSIAN             = 27,
+    SPANISH             = 28,
+    SWEDISH             = 29,
+    TAGALOG             = 30,
+    TAMIL               = 31,
+    TELUGU              = 32,
+    THAI                = 33,
+    TURKISH             = 34,
+    UKRAINIAN           = 35,
+    VIETNAMESE          = 36,
+};
+
+class TranscriptParams : public BaseMediaParams {
+public:
+    TranscriptParams();
+
+    void setSrcLanguage(int src_language);
+    void setEnableLid(bool enable_lid);
+    int srcLanguage() const;
+    bool enableLid() const;
+
+    transcript_parameters toNative() const;
+
+private:
+    int src_language_;
+    bool enable_lid_;
+};
+
 class DeskshareParams : public BaseMediaParams {
 public:
         DeskshareParams();
@@ -182,7 +487,7 @@ class MediaParams {
                 } else {
                     audio_params_.reset();
                 }
-                
+
                 if (other.hasVideoParams()) {
                     video_params_ = std::make_unique<VideoParams>(other.videoParams());
                 } else {
@@ -194,31 +499,41 @@ class MediaParams {
                 } else {
                     ds_params_.reset();
                 }
+
+                if (other.hasTranscriptParams()) {
+                    transcript_params_ = std::make_unique<TranscriptParams>(other.transcriptParams());
+                } else {
+                    transcript_params_.reset();
+                }
             }
             return *this;
         }
-        
+
         void setDeskshareParams(const DeskshareParams& ds_params);
         void setAudioParams(const AudioParams& audio_params);
         void setVideoParams(const VideoParams& video_params);
-        
+        void setTranscriptParams(const TranscriptParams& transcript_params);
+
         const DeskshareParams& deskshareParams() const;
         const AudioParams& audioParams() const;
         const VideoParams& videoParams() const;
-        
+        const TranscriptParams& transcriptParams() const;
+
         bool hasDeskshareParams() const;
         bool hasAudioParams() const;
         bool hasVideoParams() const;
-        
+        bool hasTranscriptParams() const;
+
         media_parameters toNative() const;
-    
+
     private:
         std::unique_ptr<DeskshareParams> ds_params_;
         std::unique_ptr<AudioParams> audio_params_;
         std::unique_ptr<VideoParams> video_params_;
+        std::unique_ptr<TranscriptParams> transcript_params_;
     };
 
-class Client {
+class Client : public rtms_sdk_sink {
 
 public:
     using JoinConfirmFn = function<void(int)>;
@@ -230,36 +545,21 @@ class Client {
     using TranscriptDataFn = function<void(const vector<uint8_t>&, uint64_t, const Metadata&)>;
     using LeaveFn = function<void(int)>;
     using EventExFn = function<void(const string&)>;
+    using ParticipantVideoFn = function<void(const vector<int>&, bool)>;
+    using VideoSubscribedFn = function<void(int, int, const string&)>;
 
-    // Media type constants
+    // Media type bitmask constants (matches SDK media_type enum in rtms_common.h)
+    // ALL = SDK_ALL = 0x1<<5 = 32
     enum MediaType {
-        AUDIO = 1,
-        VIDEO = 2,
-        DESKSHARE = 4,
+        AUDIO      = 1,
+        VIDEO      = 2,
+        DESKSHARE  = 4,
         TRANSCRIPT = 8,
-        CHAT = 16,
-        ALL = 31  // Sum of all types (1+2+4+8+16)
+        CHAT       = 16,
+        ALL        = 32,
     };
 
-    // Event types for subscribeEvent/unsubscribeEvent
-    // These match the RTMS_EVENT_TYPE enum from Zoom's C SDK
-    // Used with on_event_ex callback (JSON events)
-    enum EventType {
-        EVENT_UNDEFINED = 0,
-        EVENT_FIRST_PACKET_TIMESTAMP = 1,
-        EVENT_ACTIVE_SPEAKER_CHANGE = 2,
-        EVENT_PARTICIPANT_JOIN = 3,
-        EVENT_PARTICIPANT_LEAVE = 4,
-        EVENT_SHARING_START = 5,
-        EVENT_SHARING_STOP = 6,
-        EVENT_MEDIA_CONNECTION_INTERRUPTED = 7,
-        EVENT_CONSUMER_ANSWERED = 8,
-        EVENT_CONSUMER_END = 9,
-        EVENT_USER_ANSWERED = 10,
-        EVENT_USER_END = 11,
-        EVENT_USER_HOLD = 12,
-        EVENT_USER_UNHOLD = 13
-    };
+
 
     Client();
     ~Client();
@@ -289,26 +589,47 @@ class Client {
     void setDeskshareParams(const DeskshareParams& ds_params);
     void setVideoParams(const VideoParams& video_params);
     void setAudioParams(const AudioParams& audio_params);
+    void setTranscriptParams(const TranscriptParams& transcript_params);
+    void setProxy(const string& proxy_type, const string& proxy_url);
+    void subscribeVideo(int user_id, bool subscribe);
+
+    void setOnParticipantVideo(ParticipantVideoFn callback);
+    void setOnVideoSubscribed(VideoSubscribedFn callback);
 
     void join(const string& meeting_uuid, const string& rtms_stream_id, const string& signature, const string& server_url, int timeout = -1);
 
     void poll();
+    void markClosed();   // mark sdk_opened_=false before teardown so configure() becomes a no-op
     void release();
-    
+
     string uuid() const;
     string streamId() const;
+
+    // rtms_sdk_sink overrides — called by the SDK from within poll()
+    void on_join_confirm(int reason) override;
+    void on_session_update(int op, struct session_info* sess) override;
+    void on_user_update(int op, struct participant_info* pi) override;
+    void on_audio_data(unsigned char* data_buf, int size, uint64_t timestamp, struct rtms_metadata* md) override;
+    void on_video_data(unsigned char* data_buf, int size, uint64_t timestamp, struct rtms_metadata* md) override;
+    void on_ds_data(unsigned char* data_buf, int size, uint64_t timestamp, struct rtms_metadata* md) override;
+    void on_transcript_data(unsigned char* data_buf, int size, uint64_t timestamp, struct rtms_metadata* md) override;
+    void on_leave(int reason) override;
+    void on_event_ex(const std::string& compact_str) override;
+    void on_participant_video(std::vector<int> users, bool is_on) override;
+    void on_video_subscript_resp(int user_id, int status, std::string error) override;
+
 private:
     mutable mutex mutex_;
-    rtms_csdk* sdk_;
+    rtms_sdk* sdk_;
 
     string meeting_uuid_;
     string rtms_stream_id_;
-    
+
     int enabled_media_types_;
     bool media_params_updated_;
+    bool sdk_opened_;
     MediaParams media_params_;
 
-
     JoinConfirmFn join_confirm_callback_;
     SessionUpdateFn session_update_callback_;
     UserUpdateFn user_update_callback_;
@@ -318,6 +639,8 @@ class Client {
     TranscriptDataFn transcript_data_callback_;
     LeaveFn leave_callback_;
     EventExFn event_ex_callback_;
+    ParticipantVideoFn participant_video_callback_;
+    VideoSubscribedFn video_subscribed_callback_;
 
     std::vector<int> subscribed_events_;
 
@@ -326,24 +649,9 @@ class Client {
     std::vector<int> pending_event_subscriptions_;
     void processPendingSubscriptions();
 
-    static void handleJoinConfirm(struct rtms_csdk* sdk, int reason);
-    static void handleSessionUpdate(struct rtms_csdk* sdk, int op, struct session_info* sess);
-    static void handleUserUpdate(struct rtms_csdk* sdk, int op, struct participant_info* pi);
-    static void handleDsData(struct rtms_csdk* sdk, unsigned char* buf, int size, uint64_t timestamp, struct rtms_metadata* md);
-    static void handleAudioData(struct rtms_csdk* sdk, unsigned char* buf, int size, uint64_t timestamp, struct rtms_metadata* md);
-    static void handleVideoData(struct rtms_csdk* sdk, unsigned char* buf, int size, uint64_t timestamp, struct rtms_metadata* md);
-    static void handleTranscriptData(struct rtms_csdk* sdk, unsigned char* buf, int size, uint64_t timestamp, struct rtms_metadata* md);
-    static void handleLeave(struct rtms_csdk* sdk, int reason);
-    static void handleEventEx(struct rtms_csdk* sdk, const char* buf, int size);
-
-    static unordered_map<struct rtms_csdk*, Client*> sdk_registry_;
-    static mutex registry_mutex_;
-
     void throwIfError(int result, const std::string& operation) const;
-    static Client* getClient(struct rtms_csdk* sdk);
-    
     void updateMediaConfiguration(int mediaType, bool enable = true);
 };
 }
 
-#endif // RTMS_H
\ No newline at end of file
+#endif // RTMS_H
diff --git a/src/rtms/__init__.py b/src/rtms/__init__.py
index 24f23e4..ab4a7fd 100644
--- a/src/rtms/__init__.py
+++ b/src/rtms/__init__.py
@@ -6,6 +6,8 @@
 import time
 import sys
 import traceback
+import asyncio
+import inspect
 from http.server import BaseHTTPRequestHandler, HTTPServer
 from typing import Callable, Dict, Any, Optional, Union, List, Tuple
 from enum import IntEnum, Enum
@@ -14,7 +16,8 @@
 from ._rtms import (
     # Classes
     Client as _ClientBase, Session, Participant, Metadata,
-    AudioParams, VideoParams, DeskshareParams,
+    AiTargetLanguage, AiInterpreter,
+    AudioParams, VideoParams, DeskshareParams, TranscriptParams,
 
     # Media type constants
     MEDIA_TYPE_AUDIO, MEDIA_TYPE_VIDEO, MEDIA_TYPE_DESKSHARE,
@@ -27,11 +30,12 @@
     USER_JOIN, USER_LEAVE,
 
     # Event types for subscribeEvent/unsubscribeEvent (used with onEventEx callback)
-    # These match RTMS_EVENT_TYPE from Zoom's C SDK
+    # These match EVENT_TYPE from Zoom's C SDK
     EVENT_UNDEFINED, EVENT_FIRST_PACKET_TIMESTAMP,
     EVENT_ACTIVE_SPEAKER_CHANGE, EVENT_PARTICIPANT_JOIN, EVENT_PARTICIPANT_LEAVE,
     EVENT_SHARING_START, EVENT_SHARING_STOP,
     EVENT_MEDIA_CONNECTION_INTERRUPTED,
+    EVENT_PARTICIPANT_VIDEO_ON, EVENT_PARTICIPANT_VIDEO_OFF,
     EVENT_CONSUMER_ANSWERED, EVENT_CONSUMER_END,
     EVENT_USER_ANSWERED, EVENT_USER_END, EVENT_USER_HOLD, EVENT_USER_UNHOLD,
 
@@ -41,12 +45,43 @@
     RTMS_SDK_INVALID_STATUS, RTMS_SDK_INVALID_ARGS,
     SESS_STATUS_ACTIVE, SESS_STATUS_PAUSED,
 
-    # Parameter dictionaries - import directly with their original names
-    AudioContentType, AudioCodec, AudioSampleRate, AudioChannel, AudioDataOption,
-    VideoContentType, VideoCodec, VideoResolution, VideoDataOption,
-    MediaDataType, SessionState, StreamState, EventType, MessageType, StopReason
+    # Parameter dictionaries - imported as private names, converted to IntEnum below
+    AudioContentType as _AudioContentType,
+    AudioCodec as _AudioCodec,
+    AudioSampleRate as _AudioSampleRate,
+    AudioChannel as _AudioChannel,
+    DataOption as _DataOption,
+    VideoContentType as _VideoContentType,
+    VideoCodec as _VideoCodec,
+    VideoResolution as _VideoResolution,
+    MediaDataType as _MediaDataType,
+    SessionState as _SessionState,
+    StreamState as _StreamState,
+    EventType as _EventType,
+    MessageType as _MessageType,
+    StopReason as _StopReason,
+    TranscriptLanguage as _TranscriptLanguage,
 )
 
+# Convert raw C++ dicts to IntEnum for Pythonic dot-notation access
+AudioContentType  = IntEnum("AudioContentType",  _AudioContentType)
+AudioCodec        = IntEnum("AudioCodec",         _AudioCodec)
+AudioSampleRate   = IntEnum("AudioSampleRate",    _AudioSampleRate)
+AudioChannel      = IntEnum("AudioChannel",       _AudioChannel)
+DataOption        = IntEnum("DataOption",         _DataOption)
+AudioDataOption   = DataOption  # legacy alias
+VideoContentType  = IntEnum("VideoContentType",   _VideoContentType)
+VideoCodec        = IntEnum("VideoCodec",         _VideoCodec)
+VideoResolution   = IntEnum("VideoResolution",    _VideoResolution)
+VideoDataOption   = DataOption  # legacy alias
+MediaDataType     = IntEnum("MediaDataType",      _MediaDataType)
+SessionState      = IntEnum("SessionState",       _SessionState)
+StreamState       = IntEnum("StreamState",        _StreamState)
+EventType         = IntEnum("EventType",          _EventType)
+MessageType       = IntEnum("MessageType",        _MessageType)
+StopReason        = IntEnum("StopReason",         _StopReason)
+TranscriptLanguage = IntEnum("TranscriptLanguage", _TranscriptLanguage)
+
 
 # Set up logging
 _log_level = os.getenv('ZM_RTMS_LOG_LEVEL', 'info').lower()
@@ -70,14 +105,13 @@ class LogFormat(str, Enum):
 # Global webhook server
 _webhook_server = None
 
-# Global client registry for run() event loop
+# Global client registry (for status/monitoring — len(rtms._clients))
 _clients: Dict[int, 'Client'] = {}
 _clients_lock = threading.Lock()
-_pending_operations: List[Tuple[Callable, tuple, dict]] = []
-_pending_lock = threading.Lock()
-_main_thread_id: Optional[int] = None
+_sdk_init_lock = threading.Lock()
 _running = False
 _stop_event = threading.Event()
+_run_executor: Optional['Executor'] = None
 
 def _log(level, component, message, details=None):
     """Log a message at the specified level"""
@@ -445,11 +479,11 @@ def _validate_audio_params(params):
 
     # Check data option
     if hasattr(params, 'dataOpt') and params.dataOpt is not None:
-        valid_opts = [v for k, v in AudioDataOption.__dict__.items() if not k.startswith('_')]
+        valid_opts = [v for k, v in DataOption.__dict__.items() if not k.startswith('_')]
         if params.dataOpt not in valid_opts:
             errors.append(
                 f"Invalid audio dataOpt: {params.dataOpt}. "
-                f"Use rtms.AudioDataOption constants (e.g., rtms.AudioDataOption.AUDIO_MULTI_STREAMS)"
+                f"Use rtms.DataOption constants (e.g., rtms.DataOption.AUDIO_MULTI_STREAMS)"
             )
 
     # Validate codec-specific requirements
@@ -521,11 +555,11 @@ def _validate_video_params(params):
 
     # Check data option
     if hasattr(params, 'dataOpt') and params.dataOpt is not None:
-        valid_opts = [v for k, v in VideoDataOption.__dict__.items() if not k.startswith('_')]
+        valid_opts = [v for k, v in DataOption.__dict__.items() if not k.startswith('_')]
         if params.dataOpt not in valid_opts:
             errors.append(
                 f"Invalid video dataOpt: {params.dataOpt}. "
-                f"Use rtms.VideoDataOption constants (e.g., rtms.VideoDataOption.VIDEO_SINGLE_ACTIVE_STREAM)"
+                f"Use rtms.DataOption constants (e.g., rtms.DataOption.VIDEO_SINGLE_ACTIVE_STREAM)"
             )
 
     if errors:
@@ -538,6 +572,313 @@ def _validate_deskshare_params(params):
     _validate_video_params(params)
 
 
+# ============================================================================
+# EventLoop
+# ============================================================================
+
+class EventLoop:
+    """
+    An SDK I/O thread that owns one or more Client lifecycles.
+
+    The Zoom C SDK requires that alloc(), join(), poll(), and release() all run
+    on the same OS thread. EventLoop is that thread. Clients assigned to a loop
+    via add() will have their entire lifecycle managed on the loop's thread.
+
+    Usage::
+
+        loop = rtms.EventLoop()
+
+        @rtms.on_webhook_event
+        def handle(payload):
+            client = rtms.Client(executor=EXECUTOR)
+            client.on_audio_data(on_audio)
+            loop.add(client)
+            client.join(payload['payload'])
+
+        await loop.run_async()   # or loop.run() to block
+
+    Callbacks are dispatched according to the executor set on each Client:
+    - No executor: callback runs inline on the loop's thread (simple, low latency)
+    - executor=ThreadPoolExecutor(...): heavy work offloaded to worker pool
+    - async def callback: bridged to the asyncio event loop via run_coroutine_threadsafe
+    """
+
+    def __init__(self, poll_interval: float = 0.01, name: str = None):
+        """
+        Args:
+            poll_interval: Seconds between poll cycles (default: 0.01 = 10ms)
+            name: Optional thread name for debugging
+        """
+        self._poll_interval = poll_interval
+        self._name = name
+        self._clients: List['Client'] = []
+        self._clients_lock = threading.Lock()
+        self._pending: List['Client'] = []   # clients waiting for alloc+join on this thread
+        self._pending_lock = threading.Lock()
+        self._running = False
+        self._stop_event = threading.Event()
+        self._thread: Optional[threading.Thread] = None
+
+    @property
+    def client_count(self) -> int:
+        """Number of clients currently owned by this loop."""
+        with self._clients_lock:
+            return len(self._clients)
+
+    def add(self, client: 'Client') -> None:
+        """
+        Assign a client to this loop's thread.
+
+        Must be called before client.join(). The loop's thread will call
+        alloc() and join() on behalf of the client.
+
+        Args:
+            client: A Client whose join() will be deferred to this loop's thread
+        """
+        client._assigned_loop = self
+        with self._pending_lock:
+            self._pending.append(client)
+
+    def _drain_pending(self) -> None:
+        """Called from the loop's thread — alloc and join all waiting clients."""
+        with self._pending_lock:
+            pending = self._pending[:]
+            self._pending.clear()
+
+        for client in pending:
+            try:
+                client._do_alloc_and_join()
+                with self._clients_lock:
+                    self._clients.append(client)
+            except Exception as e:
+                log_error('eventloop', f'Failed to alloc/join client: {e}')
+                traceback.print_exc()
+
+    def _poll_all(self) -> None:
+        """Poll all active clients. Removes clients that have left."""
+        with self._clients_lock:
+            active = self._clients[:]
+
+        to_remove = []
+        for client in active:
+            if client._running:
+                try:
+                    client.poll()
+                except Exception as e:
+                    log_error('eventloop', f'Error polling client: {e}')
+                    to_remove.append(client)
+            else:
+                to_remove.append(client)
+
+        if to_remove:
+            with self._clients_lock:
+                for c in to_remove:
+                    self._clients.discard(c) if hasattr(self._clients, 'discard') else None
+            with self._clients_lock:
+                self._clients = [c for c in self._clients if c not in to_remove]
+
+    def run(self, stop_on_empty: bool = False) -> None:
+        """
+        Run the event loop on the current thread (blocking).
+
+        The current thread becomes the SDK I/O thread. Use this when you want
+        explicit control of which thread drives the loop.
+
+        Args:
+            stop_on_empty: Stop automatically when all clients have left
+        """
+        self._running = True
+        self._stop_event.clear()
+        log_info('eventloop', f'Starting event loop{" (" + self._name + ")" if self._name else ""} '
+                              f'(poll_interval={self._poll_interval}s)')
+        try:
+            while self._running and not self._stop_event.is_set():
+                self._drain_pending()
+                self._poll_all()
+                if stop_on_empty:
+                    with self._clients_lock:
+                        if not self._clients and not self._pending:
+                            break
+                time.sleep(self._poll_interval)
+        except KeyboardInterrupt:
+            pass
+        finally:
+            self._running = False
+            log_debug('eventloop', 'Event loop stopped')
+
+    async def run_async(self, stop_on_empty: bool = False) -> None:
+        """
+        Run the event loop as an asyncio coroutine.
+
+        Yields to the asyncio event loop between poll cycles so other coroutines
+        (aiohttp, FastAPI, asyncpg, etc.) run freely. Async callbacks registered
+        on clients are automatically bridged to this event loop.
+
+        Args:
+            stop_on_empty: Stop automatically when all clients have left
+
+        Example::
+
+            async def main():
+                loop = rtms.EventLoop()
+                await asyncio.gather(loop.run_async(), aiohttp_app.start())
+
+            asyncio.run(main())
+        """
+        self._running = True
+        self._stop_event.clear()
+        log_info('eventloop', f'Starting async event loop{" (" + self._name + ")" if self._name else ""} '
+                              f'(poll_interval={self._poll_interval}s)')
+        try:
+            while self._running and not self._stop_event.is_set():
+                self._drain_pending()
+                self._poll_all()
+                if stop_on_empty:
+                    with self._clients_lock:
+                        if not self._clients and not self._pending:
+                            break
+                await asyncio.sleep(self._poll_interval)
+        except asyncio.CancelledError:
+            log_info('eventloop', 'Async event loop cancelled')
+        finally:
+            self._running = False
+            log_debug('eventloop', 'Async event loop stopped')
+
+    def start(self) -> 'EventLoop':
+        """
+        Start the event loop in a background daemon thread.
+
+        Returns self for chaining::
+
+            loop = rtms.EventLoop().start()
+
+        The thread runs until stop() is called or the process exits.
+        """
+        self._thread = threading.Thread(
+            target=self.run,
+            name=self._name or 'rtms-eventloop',
+            daemon=True,
+        )
+        self._thread.start()
+        log_debug('eventloop', f'Background thread started: {self._thread.name}')
+        return self
+
+    def stop(self) -> None:
+        """Signal the event loop to stop after the current poll cycle."""
+        self._running = False
+        self._stop_event.set()
+
+    def join(self, timeout: float = None) -> None:
+        """Wait for the background thread to finish (only valid after start())."""
+        if self._thread:
+            self._thread.join(timeout=timeout)
+
+
+# ============================================================================
+# EventLoopPool
+# ============================================================================
+
+class EventLoopPool:
+    """
+    A pool of EventLoop threads that distributes clients across N SDK I/O threads.
+
+    Use this for high-concurrency deployments where many clients share a fixed
+    number of threads. Each client is permanently assigned to one loop for its
+    entire lifetime.
+
+    Usage::
+
+        pool = rtms.EventLoopPool(threads=4)
+
+        @rtms.on_webhook_event
+        def handle(payload):
+            client = rtms.Client(executor=EXECUTOR)
+            client.on_audio_data(on_audio)
+            pool.add(client)          # routed to least-loaded loop
+            client.join(payload['payload'])
+
+        await pool.run_async()        # or pool.run()
+
+    Scaling guidance:
+    - 1 thread per ~25 clients is a reasonable starting point
+    - Use executor= on Client for CPU/IO-heavy callbacks
+    - Monitor loop.client_count to tune thread count
+    """
+
+    def __init__(
+        self,
+        threads: int = 4,
+        poll_interval: float = 0.01,
+        strategy: str = 'least_loaded',
+    ):
+        """
+        Args:
+            threads: Number of SDK I/O threads (default: 4)
+            poll_interval: Seconds between poll cycles per loop (default: 0.01)
+            strategy: Client routing strategy — 'least_loaded' or 'round_robin'
+        """
+        if threads < 1:
+            raise ValueError("threads must be >= 1")
+        if strategy not in ('least_loaded', 'round_robin'):
+            raise ValueError("strategy must be 'least_loaded' or 'round_robin'")
+        self._loops = [
+            EventLoop(poll_interval=poll_interval, name=f'rtms-pool-{i}')
+            for i in range(threads)
+        ]
+        self._strategy = strategy
+        self._rr_index = 0
+        self._rr_lock = threading.Lock()
+
+    @property
+    def loops(self) -> List[EventLoop]:
+        """The underlying EventLoop list."""
+        return self._loops
+
+    @property
+    def client_count(self) -> int:
+        """Total clients across all loops."""
+        return sum(l.client_count for l in self._loops)
+
+    def add(self, client: 'Client') -> EventLoop:
+        """
+        Assign a client to a loop according to the routing strategy.
+
+        Returns the EventLoop the client was assigned to.
+        """
+        if self._strategy == 'least_loaded':
+            loop = min(self._loops, key=lambda l: l.client_count)
+        else:  # round_robin
+            with self._rr_lock:
+                loop = self._loops[self._rr_index % len(self._loops)]
+                self._rr_index += 1
+        loop.add(client)
+        return loop
+
+    def run(self, stop_on_empty: bool = False) -> None:
+        """
+        Run all loops. Starts N-1 loops as background daemon threads and runs
+        the last one on the current thread (blocking).
+        """
+        for loop in self._loops[:-1]:
+            loop.start()
+        self._loops[-1].run(stop_on_empty=stop_on_empty)
+
+    async def run_async(self, stop_on_empty: bool = False) -> None:
+        """
+        Run all loops as asyncio coroutines concurrently.
+        """
+        await asyncio.gather(*[l.run_async(stop_on_empty=stop_on_empty) for l in self._loops])
+
+    def stop(self) -> None:
+        """Stop all loops."""
+        for loop in self._loops:
+            loop.stop()
+
+
+# Module-level default EventLoop used by rtms.run() / rtms.run_async()
+_default_loop: Optional[EventLoop] = None
+
+
 class Client(_ClientBase):
     """
     RTMS Client - provides real-time media streaming capabilities
@@ -548,33 +889,34 @@ class Client(_ClientBase):
 
     _sdk_initialized = False
 
-    def __init__(self):
-        """Initialize a new RTMS client"""
-        # Ensure SDK is initialized before creating client instance
-        if not Client._sdk_initialized:
-            try:
-                ca_path = find_ca_certificate()
-                log_debug("client", f"Initializing SDK with CA: {ca_path}")
-                _ClientBase.initialize(ca_path, 1, "python-rtms")
-                Client._sdk_initialized = True
-                log_debug("client", "SDK initialized successfully")
-            except Exception as e:
-                log_error("client", f"SDK initialization failed: {e}")
-                # Try with empty path as fallback
-                try:
-                    log_debug("client", "Trying SDK initialization with empty CA path")
-                    _ClientBase.initialize("", 1, "python-rtms")
-                    Client._sdk_initialized = True
-                    log_debug("client", "SDK initialized with empty CA path")
-                except Exception as e2:
-                    log_error("client", f"SDK initialization failed completely: {e2}")
-                    raise RuntimeError(f"Failed to initialize RTMS SDK: {e2}")
+    def __init__(self, executor=None):
+        """Initialize a new RTMS client.
 
+        Args:
+            executor: Optional concurrent.futures.Executor for dispatching data callbacks
+                (audio, video, transcript, deskshare) to a thread pool. When set, callbacks
+                are submitted via executor.submit() instead of running inline. Pass
+                concurrent.futures.ThreadPoolExecutor(n) for CPU-bound or I/O-heavy callbacks.
+        """
+        # super().__init__() is PyClient() — intentionally a no-op at construction
+        # time. The C SDK handle is allocated lazily in _do_alloc_and_join(), which
+        # runs on the owning EventLoop's thread. This satisfies the C SDK's thread
+        # affinity requirement: alloc/join/poll/release must share one OS thread.
         super().__init__()
-        self._polling_thread = None
         self._polling_interval = 10  # milliseconds
         self._running = False
         self._webhook_server = None
+        self._executor = executor  # concurrent.futures.Executor or None
+        self._loop = None           # asyncio event loop captured at callback-registration time
+
+        # EventLoop that owns this client's lifecycle (set by loop.add() or auto-created)
+        self._assigned_loop: Optional['EventLoop'] = None
+        # Pending join params — stored until the loop's thread calls _do_alloc_and_join()
+        self._pending_join_params: Optional[dict] = None
+
+        # Individual video subscription callbacks
+        self._participant_video_callback = None
+        self._video_subscribed_callback = None
 
         # Shared event dispatcher state (matches Node.js setupEventHandler pattern)
         self._event_handler_registered = False
@@ -592,6 +934,7 @@ def join(self,
              meeting_uuid: str = None,
              webinar_uuid: str = None,
              session_id: str = None,
+             engagement_id: str = None,
              rtms_stream_id: str = None,
              server_urls: str = None,
              signature: str = None,
@@ -610,6 +953,7 @@ def join(self,
             meeting_uuid (str): Meeting UUID (for Meeting SDK events)
             webinar_uuid (str): Webinar UUID (for Webinar events)
             session_id (str): Session ID (for Video SDK events) - used when meeting_uuid is not provided
+            engagement_id (str): Engagement ID (for ZCC events) - used when meeting_uuid is not provided
             rtms_stream_id (str): RTMS stream ID
             server_urls (str): Server URLs (comma-separated)
             signature (str, optional): Authentication signature. If not provided, will be generated
@@ -624,123 +968,121 @@ def join(self,
             bool: True if joined successfully, False otherwise
         """
 
-        try:
-            # Support for both dictionary-style and parameter-style calls
-            if len(kwargs) > 0:
-                # If additional kwargs are provided, merge them with the named parameters
-                params = {
-                    'meeting_uuid': meeting_uuid,
-                    'webinar_uuid': webinar_uuid,
-                    'session_id': session_id,
-                    'rtms_stream_id': rtms_stream_id,
-                    'server_urls': server_urls,
-                    'signature': signature,
-                    'timeout': timeout,
-                    'ca': ca,
-                    'client': client,
-                    'secret': secret,
-                    'poll_interval': poll_interval
-                }
-                # Update with any additional kwargs
-                params.update(kwargs)
-                return self._join_with_params(**params)
-
-            # Check if uuid is actually a dictionary (first param)
-            if isinstance(meeting_uuid, dict):
-                return self._join_with_params(**meeting_uuid)
-
-            # Otherwise, use the parameters directly
-            return self._join_with_params(
-                meeting_uuid=meeting_uuid,
-                webinar_uuid=webinar_uuid,
-                session_id=session_id,
-                rtms_stream_id=rtms_stream_id,
-                server_urls=server_urls,
-                signature=signature,
-                timeout=timeout,
-                ca=ca,
-                client=client,
-                secret=secret,
-                poll_interval=poll_interval
-            )
-        except Exception as e:
-            log_error("client", f"Error in join: {e}")
-            traceback.print_exc()
-            return False
+        # Normalise all call forms into a single params dict
+        params = {
+            'meeting_uuid': meeting_uuid,
+            'webinar_uuid': webinar_uuid,
+            'session_id': session_id,
+            'engagement_id': engagement_id,
+            'rtms_stream_id': rtms_stream_id,
+            'server_urls': server_urls,
+            'signature': signature,
+            'timeout': timeout,
+            'ca': ca,
+            'client': client,
+            'secret': secret,
+            'poll_interval': poll_interval,
+        }
+        if kwargs:
+            params.update(kwargs)
+        if isinstance(meeting_uuid, dict):
+            params = dict(meeting_uuid)
+
+        # Store params for the loop thread to consume
+        self._pending_join_params = params
+
+        # If the client has been assigned to an explicit EventLoop, it will be
+        # picked up by that loop's _drain_pending() call — nothing else to do.
+        if self._assigned_loop is not None:
+            log_debug("client", "join() deferred to assigned EventLoop thread")
+            return True
+
+        # If rtms.run() / rtms.run_async() is active, route to the default loop.
+        if _default_loop is not None:
+            log_debug("client", "join() routed to default EventLoop")
+            _default_loop.add(self)
+            return True
 
-    def _join_with_params(self, **params):
+        # Zero-config (Tier 0): no loop assigned and no run() active —
+        # create an implicit single-client EventLoop as a background daemon thread.
+        log_debug("client", "No EventLoop assigned — creating implicit single-client loop")
+        implicit_loop = EventLoop(
+            poll_interval=params.get('poll_interval', 10) / 1000.0,
+            name='rtms-implicit',
+        )
+        implicit_loop.add(self)
+        implicit_loop.start()
+        return True
+
+    def _do_alloc_and_join(self) -> None:
         """
-        Internal method to join with parameter dictionary.
+        Called by EventLoop._drain_pending() on the loop's own thread.
 
-        IMPORTANT: Due to SDK threading constraints, the actual join() must be called
-        from the same thread that runs the event loop. If rtms.run() hasn't been called,
-        we proceed directly (backwards compatible). Otherwise, we queue for main thread.
+        Performs the two operations that must share an OS thread:
+          1. alloc()  — creates the C SDK handle (rtms_alloc)
+          2. join()   — registers callbacks and connects (rtms_set_callbacks + rtms_join)
         """
-        # If rtms.run() hasn't been called yet, proceed directly (backwards compatible)
-        if _main_thread_id is None:
-            return self._do_join(**params)
-
-        # If on main thread (where rtms.run() is executing), join directly
-        if threading.get_ident() == _main_thread_id:
-            return self._do_join(**params)
-
-        # Queue the join for main thread execution
-        log_debug("client", "Join called from non-main thread, queuing request")
-        with _pending_lock:
-            _pending_operations.append((self._do_join, (), params))
-        log_debug("client", "Join request queued, will be processed by rtms.run()")
-        return True  # Return immediately; actual join happens later
-
-    def _do_join(self, **params):
-        """Actually perform the join - always called on main thread or when no event loop"""
+        params = self._pending_join_params
+        if params is None:
+            raise RuntimeError("_do_alloc_and_join called with no pending join params")
+
         try:
-            # Extract parameters with defaults
-            meeting_uuid = params.get('meeting_uuid')
-            webinar_uuid = params.get('webinar_uuid')
-            session_id = params.get('session_id')
+            meeting_uuid  = params.get('meeting_uuid')
+            webinar_uuid  = params.get('webinar_uuid')
+            session_id    = params.get('session_id')
+            engagement_id = params.get('engagement_id')
             rtms_stream_id = params.get('rtms_stream_id')
-            server_urls = params.get('server_urls')
-            signature = params.get('signature')
-            timeout = params.get('timeout', -1)
-            ca = params.get('ca')
-            client = params.get('client', os.getenv('ZM_RTMS_CLIENT'))
-            secret = params.get('secret', os.getenv('ZM_RTMS_SECRET'))
+            server_urls   = params.get('server_urls')
+            signature     = params.get('signature')
+            timeout       = params.get('timeout', -1)
+            client_id     = params.get('client', os.getenv('ZM_RTMS_CLIENT'))
+            secret        = params.get('secret', os.getenv('ZM_RTMS_SECRET'))
             poll_interval = params.get('poll_interval', 10)
 
-            # Use meeting_uuid for Meeting SDK, webinar_uuid for Webinar, session_id for Video SDK
-            instance_id = meeting_uuid or webinar_uuid or session_id
-
+            instance_id = meeting_uuid or webinar_uuid or session_id or engagement_id
             if not instance_id:
-                raise ValueError("Either meeting_uuid, webinar_uuid, or session_id is required")
+                raise ValueError("meeting_uuid, webinar_uuid, session_id, or engagement_id is required")
             if not rtms_stream_id:
-                raise ValueError("RTMS Stream ID is required")
+                raise ValueError("rtms_stream_id is required")
             if not server_urls:
-                raise ValueError("Server URLs is required")
+                raise ValueError("server_urls is required")
 
-            # Generate signature if not provided
             if not signature:
-                try:
-                    signature = generate_signature(client, secret, instance_id, rtms_stream_id)
-                except Exception as e:
-                    log_error("client", f"Error generating signature: {e}")
-                    raise
+                signature = generate_signature(client_id, secret, instance_id, rtms_stream_id)
 
-            # Store polling interval
             self._polling_interval = poll_interval
 
-            # Join the meeting/webinar/session
-            log_info("client", f"Joining {'meeting' if meeting_uuid else 'webinar' if webinar_uuid else 'session'}: {instance_id}")
-            super().join(instance_id, rtms_stream_id, signature, server_urls, timeout)
+            # Phase 1: ensure SDK is initialized on THIS thread (same thread as alloc/join).
+            # The lock ensures init() runs exactly once even if multiple EventLoops start
+            # simultaneously, but the actual C call only happens on the first client's thread.
+            with _sdk_init_lock:
+                if not Client._sdk_initialized:
+                    ca_path = find_ca_certificate()
+                    log_debug("client", f"Initializing SDK with CA: {ca_path}")
+                    try:
+                        _ClientBase.initialize(ca_path, 1, "python-rtms")
+                    except Exception:
+                        log_debug("client", "Trying SDK initialization with empty CA path")
+                        _ClientBase.initialize("", 1, "python-rtms")
+                    Client._sdk_initialized = True
+                    log_debug("client", "SDK initialized successfully")
 
-            # Start polling thread
-            self._start_polling()
+            # Phase 2: allocate C SDK handle on this thread
+            super().alloc()
+
+            session_type = 'meeting' if meeting_uuid else 'webinar' if webinar_uuid else 'engagement' if engagement_id else 'session'
+            log_info("client", f"Joining {session_type}: {instance_id}")
+
+            # join() on the same thread as alloc() — C SDK constraint satisfied
+            super().join(instance_id, rtms_stream_id, signature, server_urls, timeout)
 
+            self._running = True
             log_info("client", "Successfully joined")
-            return True
+
         except Exception as e:
-            log_error("client", f"Error joining: {e}")
+            log_error("client", f"Error in _do_alloc_and_join: {e}")
             traceback.print_exc()
-            return False
+            raise
 
     def _initialize_rtms(self, ca_path=None):
         """Initialize the RTMS SDK with the best available CA certificate"""
@@ -764,29 +1106,91 @@ def _initialize_rtms(self, ca_path=None):
                 log_error("client", f"Failed to initialize with empty CA path: {e2}")
                 raise e  # Raise the original error
 
-    def _poll_if_needed(self):
-        """
-        Poll the RTMS client if needed.
-
-        IMPORTANT: Due to SDK threading constraints, poll() must be called from the
-        main thread (same thread that initialized the SDK). This should be called
-        periodically from the main loop.
-        """
+    def poll(self):
+        """Poll the C SDK for pending events. Called by the owning EventLoop's thread."""
         if self._running:
             try:
                 super().poll()
             except Exception as e:
                 log_error("client", f"Error during polling: {e}")
 
-    def _start_polling(self):
-        """Mark that polling should begin (will be done from main thread)"""
-        self._running = True
-        log_debug("client", "Polling enabled - call _poll_if_needed() from main loop")
+    # ========================================================================
+    # Callback Dispatch
+    # ========================================================================
 
-    def _stop_polling(self):
-        """Stop polling"""
-        self._running = False
-        log_debug("client", "Polling stopped")
+    def _wrap_callback(self, callback):
+        """Wrap a callback for executor or asyncio dispatch.
+
+        - sync + no executor  → returned unchanged (v1.0 inline behavior)
+        - sync + executor     → submitted to executor.submit() on each call
+        - async coroutine     → scheduled on the captured asyncio event loop
+        """
+        if callback is None:
+            return None
+        if inspect.iscoroutinefunction(callback):
+            # Capture the running loop now (at registration time) if one exists.
+            try:
+                loop = asyncio.get_running_loop()
+            except RuntimeError:
+                loop = None
+            self._loop = loop
+            def async_wrapper(*args):
+                _loop = self._loop
+                if _loop and _loop.is_running():
+                    asyncio.run_coroutine_threadsafe(callback(*args), _loop)
+                else:
+                    try:
+                        asyncio.run(callback(*args))
+                    except RuntimeError:
+                        pass
+            return async_wrapper
+        executor = self._executor or _run_executor
+        if executor is not None:
+            def executor_wrapper(*args):
+                executor.submit(callback, *args)
+            return executor_wrapper
+        return callback
+
+    # ========================================================================
+    # Data Callbacks (Python-level so _wrap_callback applies and aliases work)
+    # ========================================================================
+
+    def on_audio_data(self, callback) -> None:
+        """Register audio data callback. Supports executor and async coroutines."""
+        super().on_audio_data(self._wrap_callback(callback))
+
+    onAudioData = on_audio_data
+
+    def on_video_data(self, callback) -> None:
+        """Register video data callback. Supports executor and async coroutines."""
+        super().on_video_data(self._wrap_callback(callback))
+
+    onVideoData = on_video_data
+
+    def on_deskshare_data(self, callback) -> None:
+        """Register deskshare data callback. Supports executor and async coroutines."""
+        super().on_deskshare_data(self._wrap_callback(callback))
+
+    onDeskshareData = on_deskshare_data
+
+    def on_transcript_data(self, callback) -> None:
+        """Register transcript data callback. Supports executor and async coroutines."""
+        super().on_transcript_data(self._wrap_callback(callback))
+
+    onTranscriptData = on_transcript_data
+
+    # ========================================================================
+    # Context Manager
+    # ========================================================================
+
+    def __enter__(self):
+        """Support `with rtms.Client() as client:` usage."""
+        return self
+
+    def __exit__(self, *_):
+        """Call leave() on context exit. Exceptions are not suppressed."""
+        self.leave()
+        return False
 
     def stop(self):
         """
@@ -796,7 +1200,7 @@ def stop(self):
         """
         return self.leave()
 
-    def setAudioParams(self, params):
+    def set_audio_params(self, params):
         """
         Set audio parameters with validation.
 
@@ -810,9 +1214,12 @@ def setAudioParams(self, params):
             ValueError: If parameters are invalid
         """
         _validate_audio_params(params)
-        return super().setAudioParams(params)
+        return super().set_audio_params(params)
 
-    def setVideoParams(self, params):
+    # camelCase legacy alias
+    setAudioParams = set_audio_params
+
+    def set_video_params(self, params):
         """
         Set video parameters with validation.
 
@@ -826,9 +1233,12 @@ def setVideoParams(self, params):
             ValueError: If parameters are invalid
         """
         _validate_video_params(params)
-        return super().setVideoParams(params)
+        return super().set_video_params(params)
+
+    # camelCase legacy alias
+    setVideoParams = set_video_params
 
-    def setDeskshareParams(self, params):
+    def set_deskshare_params(self, params):
         """
         Set deskshare parameters with validation.
 
@@ -842,13 +1252,74 @@ def setDeskshareParams(self, params):
             ValueError: If parameters are invalid
         """
         _validate_deskshare_params(params)
-        return super().setDeskshareParams(params)
+        return super().set_deskshare_params(params)
 
-    def subscribeEvent(self, events):
+    # camelCase legacy alias
+    setDeskshareParams = set_deskshare_params
+
+    def set_transcript_params(self, params):
+        """
+        Set transcript parameters.
+
+        Args:
+            params (TranscriptParams): Transcript parameters object
+
+        Returns:
+            bool: True if parameters were set successfully
+        """
+        return super().set_transcript_params(params)
+
+    # camelCase legacy alias
+    setTranscriptParams = set_transcript_params
+
+    def set_proxy(self, proxy_type: str, proxy_url: str) -> None:
+        """Configure a proxy for SDK connections.
+
+        Args:
+            proxy_type (str): Proxy protocol type (e.g. 'http', 'https').
+            proxy_url (str): Full proxy URL including host and port.
+        """
+        return super().set_proxy(proxy_type, proxy_url)
+
+    # camelCase legacy alias
+    setProxy = set_proxy
+
+    def subscribe_video(self, user_id: int, subscribe: bool) -> None:
+        """Subscribe or unsubscribe from an individual participant's video stream.
+
+        Args:
+            user_id (int): The participant's user ID.
+            subscribe (bool): True to subscribe, False to unsubscribe.
+        """
+        return super().subscribe_video(user_id, subscribe)
+
+    subscribeVideo = subscribe_video
+
+    def on_participant_video(self, callback) -> None:
+        """Register a callback for participant video state changes.
+
+        The callback receives (users: list[int], is_on: bool).
+        """
+        self._participant_video_callback = callback
+        super().on_participant_video(callback)
+
+    onParticipantVideo = on_participant_video
+
+    def on_video_subscribed(self, callback) -> None:
+        """Register a callback for video subscription responses.
+
+        The callback receives (user_id: int, status: int, error: str).
+        """
+        self._video_subscribed_callback = callback
+        super().on_video_subscribed(callback)
+
+    onVideoSubscribed = on_video_subscribed
+
+    def subscribe_event(self, events):
         """
         Subscribe to receive specific event types.
 
-        Note: Calling onParticipantEvent() automatically subscribes to
+        Note: Calling on_participant_event() automatically subscribes to
         EVENT_PARTICIPANT_JOIN and EVENT_PARTICIPANT_LEAVE events.
 
         Args:
@@ -857,9 +1328,12 @@ def subscribeEvent(self, events):
         Returns:
             bool: True if subscription was successful
         """
-        return super().subscribeEvent(events)
+        return super().subscribe_event(events)
 
-    def unsubscribeEvent(self, events):
+    # camelCase legacy alias
+    subscribeEvent = subscribe_event
+
+    def unsubscribe_event(self, events):
         """
         Unsubscribe from specific event types.
 
@@ -869,7 +1343,10 @@ def unsubscribeEvent(self, events):
         Returns:
             bool: True if unsubscription was successful
         """
-        return super().unsubscribeEvent(events)
+        return super().unsubscribe_event(events)
+
+    # camelCase legacy alias
+    unsubscribeEvent = unsubscribe_event
 
     def _setup_event_handler(self):
         """
@@ -936,9 +1413,9 @@ def event_dispatcher(event_data: str):
             except Exception as e:
                 log_error('client', f'Failed to parse event: {e}')
 
-        super().onEventEx(event_dispatcher)
+        super().on_event_ex(event_dispatcher)
 
-    def onParticipantEvent(self, callback: Callable[[str, int, list], None]) -> bool:
+    def on_participant_event(self, callback: Callable[[str, int, list], None]) -> bool:
         """
         Register a callback for participant join/leave events.
 
@@ -957,17 +1434,20 @@ def onParticipantEvent(self, callback: Callable[[str, int, list], None]) -> bool
         Example:
             >>> def on_participant(event, timestamp, participants):
             ...     print(f"Participant {event}: {participants}")
-            >>> client.onParticipantEvent(on_participant)
+            >>> client.on_participant_event(on_participant)
         """
         self._participant_event_callback = callback
         self._setup_event_handler()
         try:
-            self.subscribeEvent([EVENT_PARTICIPANT_JOIN, EVENT_PARTICIPANT_LEAVE])
+            self.subscribe_event([EVENT_PARTICIPANT_JOIN, EVENT_PARTICIPANT_LEAVE])
         except Exception as e:
             log_warn('client', f'Failed to auto-subscribe to participant events: {e}')
         return True
 
-    def onActiveSpeakerEvent(self, callback: Callable[[int, int, str], None]) -> bool:
+    # camelCase legacy alias
+    onParticipantEvent = on_participant_event
+
+    def on_active_speaker_event(self, callback: Callable[[int, int, str], None]) -> bool:
         """
         Register a callback for active speaker change events.
 
@@ -982,17 +1462,20 @@ def onActiveSpeakerEvent(self, callback: Callable[[int, int, str], None]) -> boo
         Example:
             >>> def on_speaker(timestamp, user_id, user_name):
             ...     print(f"Active speaker: {user_name} ({user_id})")
-            >>> client.onActiveSpeakerEvent(on_speaker)
+            >>> client.on_active_speaker_event(on_speaker)
         """
         self._active_speaker_callback = callback
         self._setup_event_handler()
         try:
-            self.subscribeEvent([EVENT_ACTIVE_SPEAKER_CHANGE])
+            self.subscribe_event([EVENT_ACTIVE_SPEAKER_CHANGE])
         except Exception as e:
             log_warn('client', f'Failed to auto-subscribe to active speaker events: {e}')
         return True
 
-    def onSharingEvent(self, callback: Callable[[str, int, Optional[int], Optional[str]], None]) -> bool:
+    # camelCase legacy alias
+    onActiveSpeakerEvent = on_active_speaker_event
+
+    def on_sharing_event(self, callback: Callable[[str, int, Optional[int], Optional[str]], None]) -> bool:
         """
         Register a callback for sharing start/stop events.
 
@@ -1012,17 +1495,20 @@ def onSharingEvent(self, callback: Callable[[str, int, Optional[int], Optional[s
         Example:
             >>> def on_sharing(event, timestamp, user_id, user_name):
             ...     print(f"Sharing {event} by {user_name}")
-            >>> client.onSharingEvent(on_sharing)
+            >>> client.on_sharing_event(on_sharing)
         """
         self._sharing_callback = callback
         self._setup_event_handler()
         try:
-            self.subscribeEvent([EVENT_SHARING_START, EVENT_SHARING_STOP])
+            self.subscribe_event([EVENT_SHARING_START, EVENT_SHARING_STOP])
         except Exception as e:
             log_warn('client', f'Failed to auto-subscribe to sharing events: {e}')
         return True
 
-    def onMediaConnectionInterrupted(self, callback: Callable[[int], None]) -> bool:
+    # camelCase legacy alias
+    onSharingEvent = on_sharing_event
+
+    def on_media_connection_interrupted(self, callback: Callable[[int], None]) -> bool:
         """
         Register a callback for media connection interrupted events.
 
@@ -1037,17 +1523,20 @@ def onMediaConnectionInterrupted(self, callback: Callable[[int], None]) -> bool:
         Example:
             >>> def on_interrupted(timestamp):
             ...     print(f"Media connection interrupted at {timestamp}")
-            >>> client.onMediaConnectionInterrupted(on_interrupted)
+            >>> client.on_media_connection_interrupted(on_interrupted)
         """
         self._media_interrupted_callback = callback
         self._setup_event_handler()
         try:
-            self.subscribeEvent([EVENT_MEDIA_CONNECTION_INTERRUPTED])
+            self.subscribe_event([EVENT_MEDIA_CONNECTION_INTERRUPTED])
         except Exception as e:
             log_warn('client', f'Failed to auto-subscribe to media connection interrupted events: {e}')
         return True
 
-    def onEventEx(self, callback: Callable[[str], None]) -> bool:
+    # camelCase legacy alias
+    onMediaConnectionInterrupted = on_media_connection_interrupted
+
+    def on_event_ex(self, callback: Callable[[str], None]) -> bool:
         """
         Register a callback for raw event data.
 
@@ -1065,17 +1554,23 @@ def onEventEx(self, callback: Callable[[str], None]) -> bool:
         self._setup_event_handler()
         return True
 
+    # camelCase legacy alias
+    onEventEx = on_event_ex
+
     def leave(self):
         """
-        Leave the RTMS session and stop all threads.
+        Leave the RTMS session.
+
+        Signals the owning EventLoop to stop polling this client and releases
+        the C SDK handle. Safe to call from any thread.
 
         Returns:
             bool: True if left successfully
         """
         log_info("client", "Leaving RTMS session")
 
-        # Stop polling thread
-        self._stop_polling()
+        # Signal the EventLoop to stop polling this client
+        self._running = False
 
         # Unregister from global client registry
         with _clients_lock:
@@ -1234,120 +1729,103 @@ def decorator(func):
 on_webhook_event = onWebhookEvent
 
 
-def _process_pending_operations():
-    """Process operations queued from other threads"""
-    with _pending_lock:
-        if not _pending_operations:
-            return
-        operations = _pending_operations[:]
-        _pending_operations.clear()
-
-    for func, args, kwargs in operations:
-        try:
-            func(*args, **kwargs)
-        except Exception as e:
-            log_error('rtms', f'Error processing pending operation: {e}')
-            traceback.print_exc()
+def run(poll_interval: float = 0.01, stop_on_empty: bool = False, executor=None):
+    """
+    Start the default RTMS event loop (blocking).
 
+    Clients that call join() without an explicit EventLoop are automatically
+    routed to this default loop. Blocks until interrupted or stop() is called.
 
-def _cleanup_all_clients():
-    """Clean up all clients on shutdown"""
-    with _clients_lock:
-        for client in list(_clients.values()):
-            try:
-                client.leave()
-            except Exception:
-                pass
-        _clients.clear()
-
+    Args:
+        poll_interval: Seconds between poll cycles (default: 0.01 = 10ms)
+        stop_on_empty: Stop automatically when all clients have left
+        executor: Optional concurrent.futures.Executor for dispatching data
+            callbacks on all clients that don't have their own executor set.
 
-def run(poll_interval: float = 0.01, stop_on_empty: bool = False):
-    """
-    Start the RTMS event loop.
+    Example::
 
-    This function blocks and handles:
-    - Polling all active clients
-    - Processing pending operations from other threads (like webhook handlers)
-    - Graceful shutdown on KeyboardInterrupt
+        @rtms.on_webhook_event
+        def handle(payload):
+            client = rtms.Client()
+            client.on_transcript_data(lambda d,s,t,m: print(m.userName, d))
+            client.join(payload['payload'])
 
-    Args:
-        poll_interval: Time in seconds between poll cycles (default: 0.01 = 10ms)
-        stop_on_empty: If True, stop when no clients remain (default: False)
-
-    Example:
-        >>> import rtms
-        >>>
-        >>> clients = {}
-        >>>
-        >>> @rtms.onWebhookEvent
-        >>> def handle(payload):
-        >>>     client = rtms.Client()
-        >>>     clients[payload['payload']['rtms_stream_id']] = client
-        >>>     client.onTranscriptData(lambda d,s,t,m: print(m.userName, d))
-        >>>     client.join(payload['payload'])
-        >>>
-        >>> rtms.run()  # Blocks until interrupted
+        rtms.run()   # blocks until Ctrl-C
     """
-    global _main_thread_id, _running
-
-    _main_thread_id = threading.get_ident()
+    global _default_loop, _running, _run_executor
+    _run_executor = executor
+    _default_loop = EventLoop(poll_interval=poll_interval, name='rtms-default')
     _running = True
     _stop_event.clear()
+    try:
+        _default_loop.run(stop_on_empty=stop_on_empty)
+    finally:
+        _running = False
+        _default_loop = None
+        _run_executor = None
 
-    log_info('rtms', f'Starting RTMS event loop (poll_interval={poll_interval}s)')
 
-    try:
-        while _running and not _stop_event.is_set():
-            # Process pending operations from other threads
-            _process_pending_operations()
+async def run_async(poll_interval: float = 0.01, stop_on_empty: bool = False, executor=None):
+    """
+    Start the default RTMS event loop as an asyncio coroutine.
 
-            # Poll all active clients
-            with _clients_lock:
-                clients_to_poll = list(_clients.values())
+    Drop-in async replacement for rtms.run(). Yields to the asyncio event loop
+    between poll cycles so other coroutines (aiohttp, FastAPI, asyncpg) run freely.
 
-            for client in clients_to_poll:
-                if client._running:
-                    try:
-                        client.poll()
-                    except Exception as e:
-                        log_error('rtms', f'Error polling client: {e}')
+    Args:
+        poll_interval: Seconds between poll cycles (default: 0.01 = 10ms)
+        stop_on_empty: Stop automatically when all clients have left
+        executor: Optional concurrent.futures.Executor for dispatching data
+            callbacks on all clients that don't have their own executor set.
 
-            # Check stop_on_empty condition
-            if stop_on_empty and not clients_to_poll:
-                log_info('rtms', 'No active clients, stopping event loop')
-                break
+    Example::
 
-            time.sleep(poll_interval)
+        async def main():
+            await asyncio.gather(rtms.run_async(), aiohttp_app.start())
 
-    except KeyboardInterrupt:
-        log_info('rtms', 'Received interrupt, shutting down...')
+        asyncio.run(main())
+    """
+    global _default_loop, _running, _run_executor
+    _run_executor = executor
+    _default_loop = EventLoop(poll_interval=poll_interval, name='rtms-default')
+    _running = True
+    _stop_event.clear()
+    try:
+        await _default_loop.run_async(stop_on_empty=stop_on_empty)
     finally:
         _running = False
-        _main_thread_id = None
-        _cleanup_all_clients()
+        _default_loop = None
+        _run_executor = None
 
 
 def stop():
     """
-    Signal the event loop to stop.
+    Stop the default event loop (rtms.run() or rtms.run_async()).
 
-    Call this from another thread to gracefully stop the rtms.run() loop.
+    For EventLoop or EventLoopPool instances, call their .stop() method directly.
     """
     global _running
     _running = False
     _stop_event.set()
+    if _default_loop:
+        _default_loop.stop()
     log_info('rtms', 'Stop signal received')
 
 
 __all__ = [
     # Classes
     "Client",
+    "EventLoop",
+    "EventLoopPool",
     "Session",
     "Participant",
+    "AiTargetLanguage",
+    "AiInterpreter",
     "Metadata",
     "AudioParams",
     "VideoParams",
     "DeskshareParams",
+    "TranscriptParams",
     "LogLevel",
     "LogFormat",
 
@@ -1370,7 +1848,7 @@ def stop():
     "USER_LEAVE",
 
     # Constants - Event Types (for subscribeEvent/onEventEx)
-    # These match RTMS_EVENT_TYPE from Zoom's C SDK
+    # These match EVENT_TYPE from Zoom's C SDK
     "EVENT_UNDEFINED",
     "EVENT_FIRST_PACKET_TIMESTAMP",
     "EVENT_ACTIVE_SPEAKER_CHANGE",
@@ -1379,6 +1857,8 @@ def stop():
     "EVENT_SHARING_START",
     "EVENT_SHARING_STOP",
     "EVENT_MEDIA_CONNECTION_INTERRUPTED",
+    "EVENT_PARTICIPANT_VIDEO_ON",
+    "EVENT_PARTICIPANT_VIDEO_OFF",
     "EVENT_CONSUMER_ANSWERED",
     "EVENT_CONSUMER_END",
     "EVENT_USER_ANSWERED",
@@ -1397,16 +1877,20 @@ def stop():
     "SESS_STATUS_ACTIVE",
     "SESS_STATUS_PAUSED",
 
+    # Transcript language constants dict (mirrors AudioCodec/VideoCodec pattern)
+    "TranscriptLanguage",
+
     # Parameter dictionaries
     "AudioContentType",
     "AudioCodec",
     "AudioSampleRate",
     "AudioChannel",
-    "AudioDataOption",
+    "DataOption",
+    "AudioDataOption",  # legacy alias for DataOption
     "VideoContentType",
     "VideoCodec",
     "VideoResolution",
-    "VideoDataOption",
+    "VideoDataOption",  # legacy alias for DataOption
     "MediaDataType",
     "SessionState",
     "StreamState",
@@ -1427,6 +1911,7 @@ def stop():
 
     # Event loop functions
     "run",
+    "run_async",
     "stop",
 
     # Logging functions
diff --git a/src/rtms/__init__.pyi b/src/rtms/__init__.pyi
index 72d0a2c..f1912ea 100644
--- a/src/rtms/__init__.pyi
+++ b/src/rtms/__init__.pyi
@@ -4,6 +4,8 @@ Real-Time Media Streaming SDK for Python
 """
 
 from typing import Callable, Dict, Any, Optional, List, Literal, TypedDict
+from concurrent.futures import Executor
+from typing import Awaitable, Coroutine
 
 # ============================================================================
 # Data Classes
@@ -33,12 +35,42 @@ class Participant:
     @property
     def name(self) -> str: ...
 
+class AiTargetLanguage:
+    """A single target language entry from an AI interpreter stream"""
+    @property
+    def lid(self) -> int: ...
+    @property
+    def toneId(self) -> int: ...
+    @property
+    def voiceId(self) -> str: ...
+    @property
+    def engine(self) -> str: ...
+
+class AiInterpreter:
+    """AI interpreter metadata attached to an audio stream"""
+    @property
+    def lid(self) -> int: ...
+    @property
+    def timestamp(self) -> int: ...
+    @property
+    def channelNum(self) -> int: ...
+    @property
+    def sampleRate(self) -> int: ...
+    @property
+    def targets(self) -> list[AiTargetLanguage]: ...
+
 class Metadata:
     """Metadata about a participant in a Zoom meeting"""
     @property
     def userName(self) -> str: ...
     @property
     def userId(self) -> int: ...
+    @property
+    def startTs(self) -> int: ...
+    @property
+    def endTs(self) -> int: ...
+    @property
+    def aiInterpreter(self) -> AiInterpreter: ...
 
 # ============================================================================
 # Parameter Classes
@@ -198,12 +230,103 @@ EventExCallback = Callable[[str], None]
 
 # ============================================================================
 # Client Class
+# ============================================================================
+# EventLoop and EventLoopPool
+# ============================================================================
+
+class EventLoop:
+    """
+    An SDK I/O thread that owns one or more Client lifecycles.
+
+    Manages alloc/join/poll/release on a single dedicated OS thread, satisfying
+    the C SDK's thread-affinity requirement.
+    """
+    def __init__(self, poll_interval: float = 0.01, name: Optional[str] = None) -> None: ...
+
+    @property
+    def client_count(self) -> int: ...
+
+    def add(self, client: 'Client') -> None:
+        """Assign a client to this loop. Must be called before client.join()."""
+        ...
+
+    def run(self, stop_on_empty: bool = False) -> None:
+        """Run the event loop on the current thread (blocking)."""
+        ...
+
+    async def run_async(self, stop_on_empty: bool = False) -> None:
+        """Run the event loop as an asyncio coroutine."""
+        ...
+
+    def start(self) -> 'EventLoop':
+        """Start the event loop in a background daemon thread. Returns self."""
+        ...
+
+    def stop(self) -> None:
+        """Signal the loop to stop after the current poll cycle."""
+        ...
+
+    def join(self, timeout: Optional[float] = None) -> None:
+        """Wait for the background thread to finish (only valid after start())."""
+        ...
+
+
+class EventLoopPool:
+    """
+    A pool of EventLoop threads for distributing clients across N SDK I/O threads.
+
+    Example::
+
+        pool = rtms.EventLoopPool(threads=4)
+
+        @rtms.on_webhook_event
+        def handle(payload):
+            client = rtms.Client(executor=EXECUTOR)
+            client.on_audio_data(on_audio)
+            pool.add(client)
+            client.join(payload['payload'])
+
+        await pool.run_async()
+    """
+    def __init__(
+        self,
+        threads: int = 4,
+        poll_interval: float = 0.01,
+        strategy: Literal['least_loaded', 'round_robin'] = 'least_loaded',
+    ) -> None: ...
+
+    @property
+    def loops(self) -> List[EventLoop]: ...
+
+    @property
+    def client_count(self) -> int: ...
+
+    def add(self, client: 'Client') -> EventLoop:
+        """Route client to a loop per the strategy. Returns the assigned EventLoop."""
+        ...
+
+    def run(self, stop_on_empty: bool = False) -> None:
+        """Run all loops. Starts N-1 as background threads, runs last on current thread."""
+        ...
+
+    async def run_async(self, stop_on_empty: bool = False) -> None:
+        """Run all loops as concurrent asyncio coroutines."""
+        ...
+
+    def stop(self) -> None:
+        """Stop all loops."""
+        ...
+
+
 # ============================================================================
 
 class Client:
     """RTMS Client for connecting to Zoom real-time media streams"""
 
-    def __init__(self) -> None: ...
+    def __init__(self, executor: Optional[Executor] = None) -> None: ...
+
+    def __enter__(self) -> "Client": ...
+    def __exit__(self, *_: Any) -> bool: ...
 
     @staticmethod
     def initialize(ca_path: str, is_verify_cert: int = 1, agent: Optional[str] = None) -> None:
@@ -220,6 +343,7 @@ class Client:
         meeting_uuid: Optional[str] = None,
         webinar_uuid: Optional[str] = None,
         session_id: Optional[str] = None,
+        engagement_id: Optional[str] = None,
         rtms_stream_id: Optional[str] = None,
         server_urls: Optional[str] = None,
         signature: Optional[str] = None,
@@ -235,6 +359,7 @@ class Client:
         For Meeting SDK events (meeting.rtms_started), use meeting_uuid.
         For Webinar events (webinar.rtms_started), use webinar_uuid.
         For Video SDK events (session.rtms_started), use session_id.
+        For ZCC events (engagement.rtms_started), use engagement_id.
         """
         ...
 
@@ -258,47 +383,129 @@ class Client:
         """Get meeting UUID"""
         ...
 
-    def streamId(self) -> str:
+    def stream_id(self) -> str:
         """Get stream ID"""
         ...
+    def streamId(self) -> str:
+        """Get stream ID (legacy camelCase alias)"""
+        ...
 
-    def enableAudio(self, enable: bool) -> None:
+    def enable_audio(self, enable: bool) -> None:
         """Enable/disable audio streaming"""
         ...
+    def enableAudio(self, enable: bool) -> None:
+        """Enable/disable audio streaming (legacy camelCase alias)"""
+        ...
 
-    def enableVideo(self, enable: bool) -> None:
+    def enable_video(self, enable: bool) -> None:
         """Enable/disable video streaming"""
         ...
+    def enableVideo(self, enable: bool) -> None:
+        """Enable/disable video streaming (legacy camelCase alias)"""
+        ...
 
-    def enableTranscript(self, enable: bool) -> None:
+    def enable_transcript(self, enable: bool) -> None:
         """Enable/disable transcript streaming"""
         ...
+    def enableTranscript(self, enable: bool) -> None:
+        """Enable/disable transcript streaming (legacy camelCase alias)"""
+        ...
 
-    def enableDeskshare(self, enable: bool) -> None:
+    def enable_deskshare(self, enable: bool) -> None:
         """Enable/disable deskshare streaming"""
         ...
+    def enableDeskshare(self, enable: bool) -> None:
+        """Enable/disable deskshare streaming (legacy camelCase alias)"""
+        ...
 
-    def setAudioParams(self, params: AudioParams) -> None:
+    def set_audio_params(self, params: AudioParams) -> None:
         """Set audio parameters"""
         ...
+    def setAudioParams(self, params: AudioParams) -> None:
+        """Set audio parameters (legacy camelCase alias)"""
+        ...
 
-    def setVideoParams(self, params: VideoParams) -> None:
+    def set_video_params(self, params: VideoParams) -> None:
         """Set video parameters"""
         ...
+    def setVideoParams(self, params: VideoParams) -> None:
+        """Set video parameters (legacy camelCase alias)"""
+        ...
 
-    def setDeskshareParams(self, params: DeskshareParams) -> None:
+    def set_deskshare_params(self, params: DeskshareParams) -> None:
         """Set deskshare parameters"""
         ...
+    def setDeskshareParams(self, params: DeskshareParams) -> None:
+        """Set deskshare parameters (legacy camelCase alias)"""
+        ...
 
-    def onJoinConfirm(self, callback: Callable[[int], None]) -> None:
+    def set_transcript_params(self, params: TranscriptParams) -> None:
+        """Set transcript parameters"""
+        ...
+    def setTranscriptParams(self, params: TranscriptParams) -> None:
+        """Set transcript parameters (legacy camelCase alias)"""
+        ...
+
+    def set_proxy(self, proxy_type: str, proxy_url: str) -> None:
+        """Configure a proxy for SDK connections"""
+        ...
+    def setProxy(self, proxy_type: str, proxy_url: str) -> None:
+        """Configure a proxy for SDK connections (legacy camelCase alias)"""
+        ...
+
+    def on_join_confirm(self, callback: Callable[[int], None]) -> None:
         """Register join confirm callback"""
         ...
+    def onJoinConfirm(self, callback: Callable[[int], None]) -> None:
+        """Register join confirm callback (legacy camelCase alias)"""
+        ...
 
-    def onSessionUpdate(self, callback: Callable[[int, Session], None]) -> None:
+    def on_session_update(self, callback: Callable[[int, Session], None]) -> None:
         """Register session update callback"""
         ...
+    def onSessionUpdate(self, callback: Callable[[int, Session], None]) -> None:
+        """Register session update callback (legacy camelCase alias)"""
+        ...
 
-    def onParticipantEvent(self, callback: ParticipantEventCallback) -> bool:
+    def on_user_update(self, callback: Callable[[int, Participant], None]) -> None:
+        """Register user update callback"""
+        ...
+    def onUserUpdate(self, callback: Callable[[int, Participant], None]) -> None:
+        """Register user update callback (legacy camelCase alias)"""
+        ...
+
+    def _wrap_callback(self, callback: Callable) -> Callable:
+        """Wrap a callback for executor or asyncio dispatch."""
+        ...
+
+    def on_audio_data(self, callback: Callable[[bytes, int, int, Metadata], None]) -> None:
+        """Register audio data callback. Supports executor and async coroutines."""
+        ...
+    onAudioData: Callable  # camelCase alias
+
+    def on_video_data(self, callback: Callable[[bytes, int, int, Metadata], None]) -> None:
+        """Register video data callback. Supports executor and async coroutines."""
+        ...
+    onVideoData: Callable  # camelCase alias
+
+    def on_deskshare_data(self, callback: Callable[[bytes, int, int, Metadata], None]) -> None:
+        """Register deskshare data callback. Supports executor and async coroutines."""
+        ...
+    onDeskshareData: Callable  # camelCase alias
+
+    def on_transcript_data(self, callback: Callable[[bytes, int, int, Metadata], None]) -> None:
+        """Register transcript data callback. Supports executor and async coroutines."""
+        ...
+    onTranscriptData: Callable  # camelCase alias
+
+    def on_leave(self, callback: Callable[[int], None]) -> None:
+        """Register leave callback"""
+        ...
+    def onLeave(self, callback: Callable[[int], None]) -> None:
+        """Register leave callback (legacy camelCase alias)"""
+        ...
+
+    def on_participant_event(self, callback: ParticipantEventCallback) -> bool:
         """
         Register participant join/leave event callback.
 
@@ -314,8 +521,11 @@ class Client:
             True if callback was set successfully
         """
         ...
+    def onParticipantEvent(self, callback: ParticipantEventCallback) -> bool:
+        """Register participant join/leave event callback (legacy camelCase alias)"""
+        ...
 
-    def onActiveSpeakerEvent(self, callback: ActiveSpeakerEventCallback) -> bool:
+    def on_active_speaker_event(self, callback: ActiveSpeakerEventCallback) -> bool:
         """
         Register active speaker change event callback.
 
@@ -331,8 +541,11 @@ class Client:
             True if callback was set successfully
         """
         ...
+    def onActiveSpeakerEvent(self, callback: ActiveSpeakerEventCallback) -> bool:
+        """Register active speaker change event callback (legacy camelCase alias)"""
+        ...
 
-    def onSharingEvent(self, callback: SharingEventCallback) -> bool:
+    def on_sharing_event(self, callback: SharingEventCallback) -> bool:
         """
         Register sharing start/stop event callback.
 
@@ -350,8 +563,11 @@ class Client:
             True if callback was set successfully
         """
         ...
+    def onSharingEvent(self, callback: SharingEventCallback) -> bool:
+        """Register sharing start/stop event callback (legacy camelCase alias)"""
+        ...
 
-    def onMediaConnectionInterrupted(self, callback: Callable[[int], None]) -> bool:
+    def on_media_connection_interrupted(self, callback: Callable[[int], None]) -> bool:
         """
         Register media connection interrupted event callback.
 
@@ -364,8 +580,11 @@ class Client:
             True if callback was set successfully
         """
         ...
+    def onMediaConnectionInterrupted(self, callback: Callable[[int], None]) -> bool:
+        """Register media connection interrupted event callback (legacy camelCase alias)"""
+        ...
 
-    def onEventEx(self, callback: EventExCallback) -> bool:
+    def on_event_ex(self, callback: EventExCallback) -> bool:
         """
         Register raw JSON event callback.
 
@@ -380,32 +599,15 @@ class Client:
             True if callback was set successfully
         """
         ...
-
-    def onAudioData(self, callback: Callable[[bytes, int, int, Metadata], None]) -> None:
-        """Register audio data callback"""
-        ...
-
-    def onVideoData(self, callback: Callable[[bytes, int, int, Metadata], None]) -> None:
-        """Register video data callback"""
-        ...
-
-    def onDeskshareData(self, callback: Callable[[bytes, int, int, Metadata], None]) -> None:
-        """Register deskshare data callback"""
-        ...
-
-    def onTranscriptData(self, callback: Callable[[bytes, int, int, Metadata], None]) -> None:
-        """Register transcript data callback"""
-        ...
-
-    def onLeave(self, callback: Callable[[int], None]) -> None:
-        """Register leave callback"""
+    def onEventEx(self, callback: EventExCallback) -> bool:
+        """Register raw JSON event callback (legacy camelCase alias)"""
         ...
 
-    def subscribeEvent(self, events: List[int]) -> bool:
+    def subscribe_event(self, events: List[int]) -> bool:
         """
         Subscribe to receive specific event types.
 
-        Note: Typed event callbacks (onParticipantEvent, onActiveSpeakerEvent, etc.)
+        Note: Typed event callbacks (on_participant_event, on_active_speaker_event, etc.)
         automatically subscribe to their respective events.
 
         Args:
@@ -415,8 +617,11 @@ class Client:
             True if subscription was successful
         """
         ...
+    def subscribeEvent(self, events: List[int]) -> bool:
+        """Subscribe to event types (legacy camelCase alias)"""
+        ...
 
-    def unsubscribeEvent(self, events: List[int]) -> bool:
+    def unsubscribe_event(self, events: List[int]) -> bool:
         """
         Unsubscribe from specific event types.
 
@@ -427,6 +632,9 @@ class Client:
             True if unsubscription was successful
         """
         ...
+    def unsubscribeEvent(self, events: List[int]) -> bool:
+        """Unsubscribe from event types (legacy camelCase alias)"""
+        ...
 
     def on_webhook_event(
         self,
@@ -466,7 +674,7 @@ USER_LEAVE: int
 
 # ============================================================================
 # Constants - Event Types (for subscribeEvent/unsubscribeEvent)
-# These match RTMS_EVENT_TYPE from Zoom's C SDK
+# These match EVENT_TYPE from Zoom's C SDK
 # ============================================================================
 
 EVENT_UNDEFINED: int
@@ -477,6 +685,8 @@ EVENT_PARTICIPANT_LEAVE: int
 EVENT_SHARING_START: int
 EVENT_SHARING_STOP: int
 EVENT_MEDIA_CONNECTION_INTERRUPTED: int
+EVENT_PARTICIPANT_VIDEO_ON: int
+EVENT_PARTICIPANT_VIDEO_OFF: int
 EVENT_CONSUMER_ANSWERED: int
 EVENT_CONSUMER_END: int
 EVENT_USER_ANSWERED: int
@@ -500,26 +710,38 @@ SESS_STATUS_ACTIVE: int
 SESS_STATUS_PAUSED: int
 
 # ============================================================================
-# Parameter Dictionaries
+# Parameter Enums
 # ============================================================================
 
-AudioContentType: Dict[str, int]
-AudioCodec: Dict[str, int]
-AudioSampleRate: Dict[str, int]
-AudioChannel: Dict[str, int]
-AudioDataOption: Dict[str, int]
+from enum import IntEnum
 
-VideoContentType: Dict[str, int]
-VideoCodec: Dict[str, int]
-VideoResolution: Dict[str, int]
-VideoDataOption: Dict[str, int]
+class AudioContentType(IntEnum): ...
+class AudioCodec(IntEnum): ...
+class AudioSampleRate(IntEnum): ...
+class AudioChannel(IntEnum): ...
 
-MediaDataType: Dict[str, int]
-SessionState: Dict[str, int]
-StreamState: Dict[str, int]
-EventType: Dict[str, int]
-MessageType: Dict[str, int]
-StopReason: Dict[str, int]
+class DataOption(IntEnum):
+    UNDEFINED: int
+    AUDIO_MIXED_STREAM: int
+    AUDIO_MULTI_STREAMS: int
+    VIDEO_SINGLE_ACTIVE_STREAM: int
+    VIDEO_SINGLE_INDIVIDUAL_STREAM: int
+    VIDEO_MIXED_GALLERY_VIEW: int
+
+AudioDataOption = DataOption  # legacy alias
+VideoDataOption = DataOption  # legacy alias
+
+class VideoContentType(IntEnum): ...
+class VideoCodec(IntEnum): ...
+class VideoResolution(IntEnum): ...
+
+class MediaDataType(IntEnum): ...
+class SessionState(IntEnum): ...
+class StreamState(IntEnum): ...
+class EventType(IntEnum): ...
+class MessageType(IntEnum): ...
+class StopReason(IntEnum): ...
+class TranscriptLanguage(IntEnum): ...
 
 # ============================================================================
 # SDK Initialization Functions
@@ -605,35 +827,56 @@ on_webhook_event = onWebhookEvent
 # Event Loop Functions
 # ============================================================================
 
-def run(poll_interval: float = 0.01, stop_on_empty: bool = False) -> None:
+def run(
+    poll_interval: float = 0.01,
+    stop_on_empty: bool = False,
+    executor: Optional[Executor] = None,
+) -> None:
     """
-    Start the RTMS event loop.
+    Start the RTMS event loop (blocking).
 
-    This function blocks and handles:
-    - Polling all active clients
-    - Processing pending operations from other threads (like webhook handlers)
-    - Graceful shutdown on KeyboardInterrupt
-
-    With this function, you can create clients and call join() directly from
-    webhook handlers without manual queue management.
+    Polls all active clients in a loop, processing webhook-triggered joins and
+    dispatching callbacks. Blocks until KeyboardInterrupt or rtms.stop() is called.
 
     Args:
         poll_interval: Time in seconds between poll cycles (default: 0.01 = 10ms)
         stop_on_empty: If True, stop when no clients remain (default: False)
+        executor: Optional global executor for callback dispatch on all clients that
+            do not have their own executor set.
 
     Example:
         >>> import rtms
-        >>>
-        >>> clients = {}
-        >>>
         >>> @rtms.onWebhookEvent
         >>> def handle(payload):
         >>>     client = rtms.Client()
-        >>>     clients[payload['payload']['rtms_stream_id']] = client
-        >>>     client.onTranscriptData(lambda d,s,t,m: print(m.userName, d))
+        >>>     client.on_transcript_data(lambda d,s,t,m: print(m.userName, d))
         >>>     client.join(payload['payload'])
-        >>>
-        >>> rtms.run()  # Blocks until interrupted
+        >>> rtms.run()
+    """
+    ...
+
+async def run_async(
+    poll_interval: float = 0.01,
+    stop_on_empty: bool = False,
+    executor: Optional[Executor] = None,
+) -> None:
+    """
+    Start the RTMS event loop as an asyncio coroutine.
+
+    Drop-in async replacement for rtms.run(). Uses asyncio.sleep() instead of
+    time.sleep(), yielding control back to the event loop between poll cycles so
+    it composes naturally with aiohttp, FastAPI, asyncpg, and other async frameworks.
+
+    Args:
+        poll_interval: Time in seconds between poll cycles (default: 0.01 = 10ms)
+        stop_on_empty: If True, stop when no clients remain (default: False)
+        executor: Optional global executor for callback dispatch.
+
+    Example:
+        >>> import asyncio, rtms
+        >>> async def main():
+        >>>     await asyncio.gather(rtms.run_async(), my_server.start())
+        >>> asyncio.run(main())
     """
     ...
 
@@ -641,7 +884,8 @@ def stop() -> None:
     """
     Signal the event loop to stop.
 
-    Call this from another thread to gracefully stop the rtms.run() loop.
+    Call this from another thread or coroutine to gracefully stop rtms.run()
+    or rtms.run_async().
     """
     ...
 
diff --git a/tests/cpp/mock_sdk.cpp b/tests/cpp/mock_sdk.cpp
new file mode 100644
index 0000000..3e6a3b7
--- /dev/null
+++ b/tests/cpp/mock_sdk.cpp
@@ -0,0 +1,184 @@
+/**
+ * Mock implementations of rtms_sdk, rtms_sdk_provider, and g_agent.
+ * Linked in place of the real Zoom RTMS shared library for unit tests.
+ */
+
+#include "mock_sdk.h"
+#include <string>
+#include <mutex>
+
+// Required by rtms_sdk.h extern declaration
+std::string g_agent;
+
+// Global mock state — reset in each test
+MockSdkState g_mock_state;
+
+// ============================================================================
+// rtms_sdk stubs
+// rtms_sdk_impl is only forward-declared; m_impl is never used in the mock.
+// ============================================================================
+
+rtms_sdk::rtms_sdk() : m_impl(nullptr) {}
+rtms_sdk::~rtms_sdk() {}
+
+int rtms_sdk::open(rtms_sdk_sink* sink) {
+    ++g_mock_state.open_calls;
+    g_mock_state.last_sink = sink;
+    return g_mock_state.open_result;
+}
+
+int rtms_sdk::config(struct media_parameters* params, int media_types, int ale, int feature_ids) {
+    ++g_mock_state.config_calls;
+    g_mock_state.last_media_types = media_types;
+    g_mock_state.last_ale         = ale;
+    return g_mock_state.config_result;
+}
+
+int rtms_sdk::set_proxy(std::string proxy_type, std::string proxy_url) {
+    ++g_mock_state.proxy_calls;
+    g_mock_state.last_proxy_type = std::move(proxy_type);
+    g_mock_state.last_proxy_url  = std::move(proxy_url);
+    return g_mock_state.proxy_result;
+}
+
+int rtms_sdk::join(const char* meeting_uuid, const char* rtms_stream_id,
+                   const char* signature, const char* server_url, int timeout) {
+    ++g_mock_state.join_calls;
+    g_mock_state.last_meeting_uuid = meeting_uuid  ? meeting_uuid  : "";
+    g_mock_state.last_stream_id    = rtms_stream_id ? rtms_stream_id : "";
+    g_mock_state.last_signature    = signature     ? signature     : "";
+    g_mock_state.last_server_url   = server_url    ? server_url    : "";
+    g_mock_state.last_timeout      = timeout;
+    return g_mock_state.join_result;
+}
+
+int rtms_sdk::send_data(int, unsigned char*, int, uint64_t, void*) {
+    return RTMS_SDK_OK;
+}
+
+int rtms_sdk::leave(int) {
+    ++g_mock_state.leave_calls;
+    return g_mock_state.leave_result;
+}
+
+int rtms_sdk::poll() {
+    ++g_mock_state.poll_calls;
+    return g_mock_state.poll_result;
+}
+
+int rtms_sdk::subscribe_event(int events[], int len) {
+    ++g_mock_state.subscribe_calls;
+    g_mock_state.last_subscribed_events.assign(events, events + len);
+    return g_mock_state.subscribe_result;
+}
+
+int rtms_sdk::unsubscribe_event(int events[], int len) {
+    ++g_mock_state.unsubscribe_calls;
+    g_mock_state.last_unsubscribed_events.assign(events, events + len);
+    return g_mock_state.unsubscribe_result;
+}
+
+int rtms_sdk::send_subscript_video(int user_id, bool is_sub) {
+    ++g_mock_state.subscript_video_calls;
+    g_mock_state.last_subscript_user_id = user_id;
+    g_mock_state.last_subscript_is_sub  = is_sub;
+    return g_mock_state.subscript_video_result;
+}
+
+// ============================================================================
+// rtms_sdk_provider stubs
+// ============================================================================
+
+rtms_sdk_provider* rtms_sdk_provider::s_inst = nullptr;
+std::mutex         rtms_sdk_provider::mtx;
+
+rtms_sdk_provider::rtms_sdk_provider()  {}
+rtms_sdk_provider::~rtms_sdk_provider() {}
+
+rtms_sdk_provider* rtms_sdk_provider::instance() {
+    std::lock_guard<std::mutex> lock(mtx);
+    if (!s_inst) s_inst = new rtms_sdk_provider();
+    return s_inst;
+}
+
+int rtms_sdk_provider::init(const char* ca_path, bool is_verify_cert) {
+    ++g_mock_state.init_calls;
+    g_mock_state.last_ca_path      = ca_path ? ca_path : "";
+    g_mock_state.last_verify_cert  = is_verify_cert;
+    return g_mock_state.init_result;
+}
+
+rtms_sdk* rtms_sdk_provider::create_sdk() {
+    ++g_mock_state.create_calls;
+    if (g_mock_state.create_result != 0) return nullptr;
+    return new rtms_sdk();
+}
+
+int rtms_sdk_provider::release_sdk(rtms_sdk* sdk) {
+    ++g_mock_state.release_calls;
+    delete sdk;
+    return RTMS_SDK_OK;
+}
+
+void rtms_sdk_provider::uninit() {
+    ++g_mock_state.uninit_calls;
+}
+
+// ============================================================================
+// Callback trigger helpers
+// ============================================================================
+
+void mock_trigger_join_confirm(int reason) {
+    if (g_mock_state.last_sink)
+        g_mock_state.last_sink->on_join_confirm(reason);
+}
+
+void mock_trigger_session_update(int op, session_info* sess) {
+    if (g_mock_state.last_sink)
+        g_mock_state.last_sink->on_session_update(op, sess);
+}
+
+void mock_trigger_user_update(int op, participant_info* pi) {
+    if (g_mock_state.last_sink)
+        g_mock_state.last_sink->on_user_update(op, pi);
+}
+
+void mock_trigger_audio_data(unsigned char* buf, int size, uint64_t ts, rtms_metadata* md) {
+    if (g_mock_state.last_sink)
+        g_mock_state.last_sink->on_audio_data(buf, size, ts, md);
+}
+
+void mock_trigger_video_data(unsigned char* buf, int size, uint64_t ts, rtms_metadata* md) {
+    if (g_mock_state.last_sink)
+        g_mock_state.last_sink->on_video_data(buf, size, ts, md);
+}
+
+void mock_trigger_ds_data(unsigned char* buf, int size, uint64_t ts, rtms_metadata* md) {
+    if (g_mock_state.last_sink)
+        g_mock_state.last_sink->on_ds_data(buf, size, ts, md);
+}
+
+void mock_trigger_transcript_data(unsigned char* buf, int size, uint64_t ts, rtms_metadata* md) {
+    if (g_mock_state.last_sink)
+        g_mock_state.last_sink->on_transcript_data(buf, size, ts, md);
+}
+
+void mock_trigger_leave(int reason) {
+    if (g_mock_state.last_sink)
+        g_mock_state.last_sink->on_leave(reason);
+}
+
+void mock_trigger_event_ex(const std::string& compact_str) {
+    if (g_mock_state.last_sink)
+        g_mock_state.last_sink->on_event_ex(compact_str);
+}
+
+void mock_trigger_participant_video(std::vector<int> users, bool is_on) {
+    if (g_mock_state.last_sink)
+        g_mock_state.last_sink->on_participant_video(std::move(users), is_on);
+}
+
+void mock_trigger_video_subscript_resp(int user_id, int status, std::string error) {
+    if (g_mock_state.last_sink)
+        g_mock_state.last_sink->on_video_subscript_resp(user_id, status, std::move(error));
+}
diff --git a/tests/cpp/mock_sdk.h b/tests/cpp/mock_sdk.h
new file mode 100644
index 0000000..fafee76
--- /dev/null
+++ b/tests/cpp/mock_sdk.h
@@ -0,0 +1,99 @@
+/**
+ * Mock SDK state for C++ unit tests.
+ *
+ * Provides:
+ *   - g_mock_state  — per-test configurable return values and call records
+ *   - mock_trigger_*  — helpers to fire SDK callbacks on the registered sink
+ *
+ * Usage in tests:
+ *   g_mock_state.reset();           // called in each test fixture
+ *   g_mock_state.join_result = RTMS_SDK_FAILURE;   // inject failure
+ *   mock_trigger_join_confirm(0);   // simulate SDK firing a callback
+ */
+
+#pragma once
+
+#include "rtms_sdk.h"
+#include <string>
+#include <vector>
+#include <cstdint>
+
+struct MockSdkState {
+    // --- Configurable return values ---
+    int create_result   = 0;        // 0 = return real sdk, non-zero = return nullptr
+    int init_result     = RTMS_SDK_OK;
+    int open_result     = RTMS_SDK_OK;
+    int config_result   = RTMS_SDK_OK;
+    int join_result     = RTMS_SDK_OK;
+    int poll_result     = RTMS_SDK_OK;
+    int leave_result    = RTMS_SDK_OK;
+    int subscribe_result   = RTMS_SDK_OK;
+    int unsubscribe_result = RTMS_SDK_OK;
+    int proxy_result       = RTMS_SDK_OK;
+    int subscript_video_result = RTMS_SDK_OK;
+
+    // --- Call counters ---
+    int init_calls       = 0;
+    int uninit_calls     = 0;
+    int create_calls     = 0;
+    int release_calls    = 0;
+    int open_calls       = 0;
+    int config_calls     = 0;
+    int join_calls       = 0;
+    int poll_calls       = 0;
+    int leave_calls      = 0;
+    int subscribe_calls  = 0;
+    int unsubscribe_calls = 0;
+    int proxy_calls      = 0;
+    int subscript_video_calls = 0;
+
+    // --- Recorded arguments ---
+    rtms_sdk_sink* last_sink = nullptr;
+
+    // init()
+    std::string last_ca_path;
+    bool        last_verify_cert = true;
+
+    // join()
+    std::string last_meeting_uuid;
+    std::string last_stream_id;
+    std::string last_signature;
+    std::string last_server_url;
+    int         last_timeout = 0;
+
+    // config()
+    int last_media_types = 0;
+    int last_ale         = 0;           // application layer encryption flag
+
+    // set_proxy()
+    std::string last_proxy_type;
+    std::string last_proxy_url;
+
+    // subscribe_event() / unsubscribe_event()
+    std::vector<int> last_subscribed_events;
+    std::vector<int> last_unsubscribed_events;
+
+    // send_subscript_video()
+    int  last_subscript_user_id = 0;
+    bool last_subscript_is_sub  = false;
+
+    void reset() { *this = MockSdkState{}; }
+};
+
+extern MockSdkState g_mock_state;
+
+// ============================================================================
+// Callback trigger helpers — simulate the SDK firing events on the Client
+// ============================================================================
+
+void mock_trigger_join_confirm(int reason);
+void mock_trigger_session_update(int op, session_info* sess);
+void mock_trigger_user_update(int op, participant_info* pi);
+void mock_trigger_audio_data(unsigned char* buf, int size, uint64_t ts, rtms_metadata* md);
+void mock_trigger_video_data(unsigned char* buf, int size, uint64_t ts, rtms_metadata* md);
+void mock_trigger_ds_data(unsigned char* buf, int size, uint64_t ts, rtms_metadata* md);
+void mock_trigger_transcript_data(unsigned char* buf, int size, uint64_t ts, rtms_metadata* md);
+void mock_trigger_leave(int reason);
+void mock_trigger_event_ex(const std::string& compact_str);
+void mock_trigger_participant_video(std::vector<int> users, bool is_on);
+void mock_trigger_video_subscript_resp(int user_id, int status, std::string error);
diff --git a/tests/cpp/test_cpp_wrapper.cpp b/tests/cpp/test_cpp_wrapper.cpp
new file mode 100644
index 0000000..83a8728
--- /dev/null
+++ b/tests/cpp/test_cpp_wrapper.cpp
@@ -0,0 +1,1127 @@
+/**
+ * C++ unit tests for the RTMS SDK wrapper (src/rtms.h / src/rtms.cpp).
+ *
+ * Linked against tests/mock_sdk.cpp instead of the real Zoom SDK binary,
+ * so these run in CI without credentials or the proprietary library.
+ *
+ * Test coverage:
+ *   - AudioParams / VideoParams / DeskshareParams validation
+ *   - Session / Participant / Metadata data classes
+ *   - MediaParams composition and toNative()
+ *   - Client lifecycle (create, initialize, join, poll, release)
+ *   - Callback dispatch (via mock_trigger_* helpers)
+ *   - Event subscription deferral / on-confirm flush
+ *   - Media type auto-enable on callback registration
+ *   - setProxy forwarding and error handling
+ */
+
+#include <catch2/catch_test_macros.hpp>
+#include <catch2/matchers/catch_matchers_string.hpp>
+
+#include "rtms.h"
+#include "mock_sdk.h"
+
+#include <cstring>
+#include <string>
+#include <vector>
+
+using namespace rtms;
+using Catch::Matchers::ContainsSubstring;
+
+// Reset global mock state before every test case
+struct R { R() { g_mock_state.reset(); } };
+
+// ============================================================================
+// AudioParams
+// ============================================================================
+
+TEST_CASE("AudioParams default constructor", "[params][audio]") {
+    R _;
+    AudioParams p;
+    CHECK(p.contentType() == 2);  // RAW_AUDIO
+    CHECK(p.codec()       == 4);  // OPUS
+    CHECK(p.sampleRate()  == 3);  // SR_48K
+    CHECK(p.channel()     == 2);  // STEREO
+    CHECK(p.dataOpt()     == 2);  // AUDIO_MULTI_STREAMS
+    CHECK(p.duration()    == 20);
+    CHECK(p.frameSize()   == 960);
+}
+
+TEST_CASE("AudioParams validate() — valid configurations", "[params][audio]") {
+    R _;
+
+    SECTION("default params pass validation") {
+        AudioParams p;
+        REQUIRE_NOTHROW(p.validate());
+    }
+
+    SECTION("PCM/L16 at 16kHz with correct frame size passes") {
+        // L16=1, SR_16K=1, MONO=1, AUDIO_MULTI_STREAMS=2, 20ms → 320 samples
+        AudioParams p(2, 1, 1, 1, 2, 20, 320);
+        REQUIRE_NOTHROW(p.validate());
+    }
+
+    SECTION("PCM/L16 at 8kHz with correct frame size passes") {
+        // L16=1, SR_8K=0, MONO=1, AUDIO_MULTI_STREAMS=2, 20ms → 160 samples
+        AudioParams p(2, 1, 0, 1, 2, 20, 160);
+        REQUIRE_NOTHROW(p.validate());
+    }
+}
+
+TEST_CASE("AudioParams validate() — invalid configurations throw", "[params][audio]") {
+    R _;
+
+    SECTION("contentType 0 throws") {
+        AudioParams p;
+        p.setContentType(0);
+        REQUIRE_THROWS_AS(p.validate(), std::invalid_argument);
+    }
+
+    SECTION("codec 0 throws") {
+        AudioParams p;
+        p.setCodec(0);
+        REQUIRE_THROWS_AS(p.validate(), std::invalid_argument);
+    }
+
+    SECTION("channel 0 throws") {
+        AudioParams p;
+        p.setChannel(0);
+        REQUIRE_THROWS_AS(p.validate(), std::invalid_argument);
+    }
+
+    SECTION("dataOpt 0 throws") {
+        AudioParams p;
+        p.setDataOpt(0);
+        REQUIRE_THROWS_AS(p.validate(), std::invalid_argument);
+    }
+
+    SECTION("OPUS codec with 16kHz sample rate throws") {
+        AudioParams p;
+        p.setSampleRate(1); // SR_16K — OPUS requires SR_48K
+        REQUIRE_THROWS_AS(p.validate(), std::invalid_argument);
+    }
+
+    SECTION("OPUS codec with 8kHz sample rate throws") {
+        AudioParams p;
+        p.setSampleRate(0); // SR_8K
+        REQUIRE_THROWS_AS(p.validate(), std::invalid_argument);
+    }
+
+    SECTION("frameSize mismatch for 48kHz 20ms throws") {
+        AudioParams p;                 // default: 48kHz, 20ms → expects 960
+        p.setFrameSize(480);           // wrong
+        REQUIRE_THROWS_AS(p.validate(), std::invalid_argument);
+    }
+
+    SECTION("frameSize mismatch error message names the values") {
+        AudioParams p;
+        p.setFrameSize(480);
+        REQUIRE_THROWS_WITH(p.validate(), ContainsSubstring("960"));
+    }
+
+    SECTION("OPUS error message mentions 48kHz") {
+        AudioParams p;
+        p.setSampleRate(1);
+        REQUIRE_THROWS_WITH(p.validate(), ContainsSubstring("48kHz"));
+    }
+}
+
+TEST_CASE("AudioParams toNative() maps fields correctly", "[params][audio]") {
+    R _;
+    AudioParams p(1, 2, 0, 1, 1, 20, 160); // RTP, G711, 8K, MONO, MIXED, 20ms, 160
+    auto n = p.toNative();
+    CHECK(n.content_type == 1);
+    CHECK(n.codec        == 2);
+    CHECK(n.sample_rate  == 0);
+    CHECK(n.channel      == 1);
+    CHECK(n.data_opt     == 1);
+    CHECK(n.duration     == 20);
+    CHECK(n.frame_size   == 160);
+}
+
+// ============================================================================
+// VideoParams
+// ============================================================================
+
+TEST_CASE("VideoParams default constructor uses H264/HD/30fps defaults", "[params][video]") {
+    R _;
+    VideoParams p;
+    CHECK(p.contentType() == (int)MEDIA_CONTENT_TYPE::RAW_VIDEO);
+    CHECK(p.codec()       == (int)MEDIA_PAYLOAD_TYPE::H264);
+    CHECK(p.resolution()  == (int)MEDIA_RESOLUTION::HD);
+    CHECK(p.dataOpt()     == (int)MEDIA_DATA_OPTION::VIDEO_SINGLE_ACTIVE_STREAM);
+    CHECK(p.fps()         == 30);
+}
+
+TEST_CASE("VideoParams validate() — invalid configurations throw", "[params][video]") {
+    R _;
+
+    SECTION("resolution 0 throws") {
+        VideoParams p(3, 7, 0, 5, 30); // RAW_VIDEO, H264, no resolution
+        REQUIRE_THROWS_AS(p.validate(), std::invalid_argument);
+    }
+
+    SECTION("JPG codec with fps > 5 throws") {
+        VideoParams p(3, 5, 2, 5, 10); // JPG=5, HD=2, fps=10
+        REQUIRE_THROWS_AS(p.validate(), std::invalid_argument);
+    }
+
+    SECTION("PNG codec with fps > 5 throws") {
+        VideoParams p(3, 6, 2, 5, 6);  // PNG=6, fps=6
+        REQUIRE_THROWS_AS(p.validate(), std::invalid_argument);
+    }
+
+    SECTION("JPG codec with fps <= 5 passes") {
+        VideoParams p(3, 5, 2, 5, 5);  // JPG=5, fps=5
+        REQUIRE_NOTHROW(p.validate());
+    }
+
+    SECTION("H264 with fps > 30 throws") {
+        VideoParams p(3, 7, 2, 5, 31); // H264=7, fps=31
+        REQUIRE_THROWS_AS(p.validate(), std::invalid_argument);
+    }
+
+    SECTION("H264 with fps 0 throws") {
+        VideoParams p(3, 7, 2, 5, 0);
+        REQUIRE_THROWS_AS(p.validate(), std::invalid_argument);
+    }
+
+    SECTION("H264 with fps 30 passes") {
+        VideoParams p(3, 7, 2, 5, 30);
+        REQUIRE_NOTHROW(p.validate());
+    }
+}
+
+// ============================================================================
+// DeskshareParams
+// ============================================================================
+
+TEST_CASE("DeskshareParams validate() — fps/codec rules match VideoParams", "[params][ds]") {
+    R _;
+
+    SECTION("JPG with fps > 5 throws") {
+        DeskshareParams p(3, 5, 2, 10); // JPG=5, fps=10
+        REQUIRE_THROWS_AS(p.validate(), std::invalid_argument);
+    }
+
+    SECTION("H264 with fps 30 passes") {
+        DeskshareParams p(3, 7, 2, 30);
+        REQUIRE_NOTHROW(p.validate());
+    }
+}
+
+// ============================================================================
+// Session
+// ============================================================================
+
+TEST_CASE("Session normal construction", "[data][session]") {
+    R _;
+    session_info info;
+    char sid[]  = "sess-abc";
+    char strid[] = "stream-xyz";
+    char mid[MAX_MEETING_ID_LEN] = "meeting-123";
+    info.session_id = sid;
+    info.stream_id  = strid;
+    memcpy(info.meeting_id, mid, sizeof(info.meeting_id));
+    info.stat_time  = 42;
+    info.status     = SESS_STATUS_ACTIVE;
+
+    Session s(info);
+    CHECK(s.sessionId() == "sess-abc");
+    CHECK(s.streamId()  == "stream-xyz");
+    CHECK(s.meetingId() == "meeting-123");
+    CHECK(s.statTime()  == 42);
+    CHECK(s.isActive());
+    CHECK_FALSE(s.isPaused());
+}
+
+TEST_CASE("Session null pointer safety", "[data][session]") {
+    R _;
+
+    SECTION("null session_id becomes empty string") {
+        session_info info{};
+        info.session_id = nullptr;
+        info.stream_id  = nullptr;
+        Session s(info);
+        CHECK(s.sessionId().empty());
+        CHECK(s.streamId().empty());
+    }
+
+    SECTION("low-address session_id treated as invalid") {
+        session_info info{};
+        // Anything <= 0xFFFF is treated as an invalid pointer (not a real address)
+        info.session_id = reinterpret_cast<char*>(0x100);
+        info.stream_id  = nullptr;
+        Session s(info);
+        CHECK(s.sessionId().empty());
+    }
+
+    SECTION("meeting_id buffer without null terminator is safe") {
+        session_info info{};
+        info.session_id = nullptr;
+        info.stream_id  = nullptr;
+        // Fill entire buffer with 'X' — no null terminator
+        memset(info.meeting_id, 'X', MAX_MEETING_ID_LEN);
+        Session s(info);
+        // Should read exactly MAX_MEETING_ID_LEN 'X' characters, not overflow
+        CHECK(s.meetingId().size() == MAX_MEETING_ID_LEN);
+        CHECK(s.meetingId() == std::string(MAX_MEETING_ID_LEN, 'X'));
+    }
+}
+
+TEST_CASE("Session pause status", "[data][session]") {
+    R _;
+    session_info info{};
+    info.status = SESS_STATUS_PAUSED;
+    Session s(info);
+    CHECK(s.isPaused());
+    CHECK_FALSE(s.isActive());
+}
+
+// ============================================================================
+// Participant
+// ============================================================================
+
+TEST_CASE("Participant construction", "[data][participant]") {
+    R _;
+
+    SECTION("normal construction copies id and name") {
+        participant_info pi;
+        char name[] = "Alice";
+        pi.participant_id   = 99;
+        pi.participant_name = name;
+        Participant p(pi);
+        CHECK(p.id()   == 99);
+        CHECK(p.name() == "Alice");
+    }
+
+    SECTION("null participant_name becomes empty string") {
+        participant_info pi;
+        pi.participant_id   = 1;
+        pi.participant_name = nullptr;
+        Participant p(pi);
+        CHECK(p.name().empty());
+    }
+}
+
+// ============================================================================
+// Metadata
+// ============================================================================
+
+TEST_CASE("Metadata construction", "[data][metadata]") {
+    R _;
+
+    SECTION("normal construction copies user_name and user_id") {
+        rtms_metadata md{};
+        char name[] = "Bob";
+        md.user_name = name;
+        md.user_id   = 42;
+        Metadata m(md);
+        CHECK(m.userName() == "Bob");
+        CHECK(m.userId()   == 42);
+    }
+
+    SECTION("null user_name becomes empty string") {
+        rtms_metadata md{};
+        md.user_name = nullptr;
+        md.user_id   = 7;
+        Metadata m(md);
+        CHECK(m.userName().empty());
+        CHECK(m.userId() == 7);
+    }
+}
+
+// ============================================================================
+// MediaParams
+// ============================================================================
+
+TEST_CASE("MediaParams — empty by default", "[params][media]") {
+    R _;
+    MediaParams mp;
+    CHECK_FALSE(mp.hasAudioParams());
+    CHECK_FALSE(mp.hasVideoParams());
+    CHECK_FALSE(mp.hasDeskshareParams());
+}
+
+TEST_CASE("MediaParams — set/get round-trips", "[params][media]") {
+    R _;
+    MediaParams mp;
+
+    AudioParams ap;
+    ap.setSampleRate(1); // SR_16K
+    mp.setAudioParams(ap);
+    REQUIRE(mp.hasAudioParams());
+    CHECK(mp.audioParams().sampleRate() == 1);
+
+    VideoParams vp;
+    vp.setResolution(2); // HD
+    mp.setVideoParams(vp);
+    REQUIRE(mp.hasVideoParams());
+    CHECK(mp.videoParams().resolution() == 2);
+}
+
+TEST_CASE("MediaParams toNative() — correct pointers", "[params][media]") {
+    R _;
+
+    SECTION("empty MediaParams produces all-null native struct") {
+        MediaParams mp;
+        auto n = mp.toNative();
+        CHECK(n.audio_param == nullptr);
+        CHECK(n.video_param == nullptr);
+        CHECK(n.ds_param    == nullptr);
+        CHECK(n.tr_param    == nullptr);
+        // No leak — nothing was allocated
+    }
+
+    SECTION("set audio produces non-null audio_param, others null") {
+        MediaParams mp;
+        mp.setAudioParams(AudioParams());
+        auto n = mp.toNative();
+        REQUIRE(n.audio_param != nullptr);
+        CHECK(n.video_param == nullptr);
+        CHECK(n.ds_param    == nullptr);
+        CHECK(n.tr_param    == nullptr);
+        CHECK(n.audio_param->codec == 4); // OPUS default
+        delete n.audio_param;
+    }
+
+    SECTION("set video produces non-null video_param") {
+        MediaParams mp;
+        VideoParams vp(3, 7, 2, 5, 30); // RAW_VIDEO, H264, HD, GALLERY, 30fps
+        mp.setVideoParams(vp);
+        auto n = mp.toNative();
+        REQUIRE(n.video_param != nullptr);
+        CHECK(n.audio_param == nullptr);
+        CHECK(n.video_param->codec == 7);
+        delete n.video_param;
+    }
+}
+
+TEST_CASE("MediaParams assignment copies all params", "[params][media]") {
+    R _;
+    MediaParams a;
+    a.setAudioParams(AudioParams());
+    VideoParams vp; vp.setResolution(2);
+    a.setVideoParams(vp);
+
+    MediaParams b;
+    b = a;
+    REQUIRE(b.hasAudioParams());
+    REQUIRE(b.hasVideoParams());
+    CHECK_FALSE(b.hasDeskshareParams());
+    CHECK(b.videoParams().resolution() == 2);
+}
+
+// ============================================================================
+// Client lifecycle
+// ============================================================================
+
+TEST_CASE("Client constructor calls create_sdk", "[client][lifecycle]") {
+    R _;
+    { Client c; }
+    CHECK(g_mock_state.create_calls == 1);
+    // Destructor should release
+    CHECK(g_mock_state.release_calls == 1);
+}
+
+TEST_CASE("Client constructor throws when create_sdk returns null", "[client][lifecycle]") {
+    R _;
+    g_mock_state.create_result = 1; // make create_sdk return nullptr
+    REQUIRE_THROWS_AS(Client(), Exception);
+}
+
+TEST_CASE("Client::initialize calls provider->init with correct args", "[client][lifecycle]") {
+    R _;
+    Client::initialize("/path/to/ca.pem", 1, nullptr);
+    CHECK(g_mock_state.init_calls        == 1);
+    CHECK(g_mock_state.last_ca_path      == "/path/to/ca.pem");
+    CHECK(g_mock_state.last_verify_cert  == true);
+}
+
+TEST_CASE("Client::initialize with verify=0 passes false to provider", "[client][lifecycle]") {
+    R _;
+    Client::initialize("", 0);
+    CHECK(g_mock_state.last_verify_cert == false);
+}
+
+TEST_CASE("Client::initialize sets g_agent when provided", "[client][lifecycle]") {
+    R _;
+    Client::initialize("", 1, "my-agent/1.0");
+    CHECK(g_agent == "my-agent/1.0");
+    g_agent.clear(); // cleanup
+}
+
+TEST_CASE("Client::uninitialize calls provider->uninit", "[client][lifecycle]") {
+    R _;
+    Client::uninitialize();
+    CHECK(g_mock_state.uninit_calls == 1);
+}
+
+TEST_CASE("Client::join calls open() then join() in order", "[client][join]") {
+    R _;
+    Client c;
+    c.join("uuid-1", "stream-1", "sig-1", "wss://server", 5000);
+
+    CHECK(g_mock_state.open_calls  == 1);
+    CHECK(g_mock_state.join_calls  == 1);
+    // open() must be called before join() — verified by sequential counts
+    CHECK(g_mock_state.last_meeting_uuid == "uuid-1");
+    CHECK(g_mock_state.last_stream_id    == "stream-1");
+    CHECK(g_mock_state.last_signature    == "sig-1");
+    CHECK(g_mock_state.last_server_url   == "wss://server");
+    CHECK(g_mock_state.last_timeout      == 5000);
+}
+
+TEST_CASE("Client::join passes itself as the sink to open()", "[client][join]") {
+    R _;
+    Client c;
+    c.join("uuid", "stream", "sig", "url");
+    CHECK(g_mock_state.last_sink == &c);
+}
+
+TEST_CASE("Client::join throws when open() fails", "[client][join]") {
+    R _;
+    g_mock_state.open_result = RTMS_SDK_FAILURE;
+    Client c;
+    REQUIRE_THROWS_AS(c.join("uuid", "stream", "sig", "url"), Exception);
+    CHECK(g_mock_state.join_calls == 0); // join() should not be called after open() fails
+}
+
+TEST_CASE("Client::join throws when join() fails", "[client][join]") {
+    R _;
+    g_mock_state.join_result = RTMS_SDK_FAILURE;
+    Client c;
+    REQUIRE_THROWS_AS(c.join("uuid", "stream", "sig", "url"), Exception);
+}
+
+TEST_CASE("Client::poll calls sdk->poll()", "[client][poll]") {
+    R _;
+    Client c;
+    c.poll();
+    CHECK(g_mock_state.poll_calls == 1);
+}
+
+TEST_CASE("Client::poll throws on SDK error", "[client][poll]") {
+    R _;
+    g_mock_state.poll_result = RTMS_SDK_TIMEOUT;
+    Client c;
+    REQUIRE_THROWS_AS(c.poll(), Exception);
+}
+
+TEST_CASE("Client::release calls leave() then release_sdk()", "[client][release]") {
+    R _;
+    {
+        Client c;
+        c.release();
+        CHECK(g_mock_state.leave_calls   == 1);
+        CHECK(g_mock_state.release_calls == 1);
+    }
+    // Destructor should NOT double-release (sdk_ is nullptr after release())
+    CHECK(g_mock_state.release_calls == 1);
+}
+
+TEST_CASE("Client::uuid and streamId return joined values", "[client][join]") {
+    R _;
+    Client c;
+    c.join("my-uuid", "my-stream", "sig", "url");
+    CHECK(c.uuid()     == "my-uuid");
+    CHECK(c.streamId() == "my-stream");
+}
+
+// ============================================================================
+// Callback dispatch
+// ============================================================================
+
+TEST_CASE("on_join_confirm fires registered callback", "[client][callbacks]") {
+    R _;
+    Client c;
+    c.join("u", "s", "sig", "url");
+
+    int received_reason = -99;
+    c.setOnJoinConfirm([&](int r) { received_reason = r; });
+    mock_trigger_join_confirm(0);
+
+    CHECK(received_reason == 0);
+}
+
+TEST_CASE("on_leave fires registered callback with reason", "[client][callbacks]") {
+    R _;
+    Client c;
+    c.join("u", "s", "sig", "url");
+
+    int received = -1;
+    c.setOnLeave([&](int r) { received = r; });
+    mock_trigger_leave(3);
+
+    CHECK(received == 3);
+}
+
+TEST_CASE("on_event_ex fires registered callback with string", "[client][callbacks]") {
+    R _;
+    Client c;
+    c.join("u", "s", "sig", "url");
+
+    std::string received;
+    c.setOnEventEx([&](const std::string& s) { received = s; });
+    mock_trigger_event_ex(R"({"event":"test"})");
+
+    CHECK(received == R"({"event":"test"})");
+}
+
+TEST_CASE("on_event_ex does not fire callback on empty string", "[client][callbacks]") {
+    R _;
+    Client c;
+    c.join("u", "s", "sig", "url");
+
+    bool called = false;
+    c.setOnEventEx([&](const std::string&) { called = true; });
+    mock_trigger_event_ex("");
+
+    CHECK_FALSE(called);
+}
+
+TEST_CASE("on_audio_data fires callback with correct data", "[client][callbacks]") {
+    R _;
+    Client c;
+    c.join("u", "s", "sig", "url");
+
+    std::vector<uint8_t> received_data;
+    uint64_t received_ts = 0;
+    int received_uid = -1;
+    c.setOnAudioData([&](const std::vector<uint8_t>& d, uint64_t ts, const Metadata& md) {
+        received_data = d;
+        received_ts   = ts;
+        received_uid  = md.userId();
+    });
+
+    unsigned char buf[] = {0x01, 0x02, 0x03};
+    rtms_metadata md{}; md.user_id = 42; md.user_name = nullptr;
+    mock_trigger_audio_data(buf, 3, 999ULL, &md);
+
+    REQUIRE(received_data.size() == 3);
+    CHECK(received_data[0] == 0x01);
+    CHECK(received_data[2] == 0x03);
+    CHECK(received_ts      == 999ULL);
+    CHECK(received_uid     == 42);
+}
+
+TEST_CASE("on_audio_data does not fire callback on null buf", "[client][callbacks]") {
+    R _;
+    Client c;
+    c.join("u", "s", "sig", "url");
+
+    bool called = false;
+    c.setOnAudioData([&](const std::vector<uint8_t>&, uint64_t, const Metadata&) { called = true; });
+
+    rtms_metadata md{};
+    mock_trigger_audio_data(nullptr, 3, 0, &md);
+    CHECK_FALSE(called);
+}
+
+TEST_CASE("on_audio_data does not fire callback on zero size", "[client][callbacks]") {
+    R _;
+    Client c;
+    c.join("u", "s", "sig", "url");
+
+    bool called = false;
+    c.setOnAudioData([&](const std::vector<uint8_t>&, uint64_t, const Metadata&) { called = true; });
+
+    unsigned char buf[] = {0x01};
+    rtms_metadata md{};
+    mock_trigger_audio_data(buf, 0, 0, &md);
+    CHECK_FALSE(called);
+}
+
+TEST_CASE("on_audio_data does not fire callback on null metadata", "[client][callbacks]") {
+    R _;
+    Client c;
+    c.join("u", "s", "sig", "url");
+
+    bool called = false;
+    c.setOnAudioData([&](const std::vector<uint8_t>&, uint64_t, const Metadata&) { called = true; });
+
+    unsigned char buf[] = {0x01};
+    mock_trigger_audio_data(buf, 1, 0, nullptr);
+    CHECK_FALSE(called);
+}
+
+TEST_CASE("on_session_update fires with correct Session object", "[client][callbacks]") {
+    R _;
+    Client c;
+    c.join("u", "s", "sig", "url");
+
+    int received_op = -1;
+    std::string received_sid;
+    c.setOnSessionUpdate([&](int op, const Session& sess) {
+        received_op  = op;
+        received_sid = sess.sessionId();
+    });
+
+    session_info info{};
+    char sid[] = "my-session";
+    info.session_id = sid;
+    info.stream_id  = nullptr;
+    info.status     = SESS_STATUS_ACTIVE;
+    mock_trigger_session_update(SESSION_ADD, &info);
+
+    CHECK(received_op  == SESSION_ADD);
+    CHECK(received_sid == "my-session");
+}
+
+TEST_CASE("on_user_update fires with correct Participant object", "[client][callbacks]") {
+    R _;
+    Client c;
+    c.join("u", "s", "sig", "url");
+
+    int received_op = -1;
+    int received_id = -1;
+    c.setOnUserUpdate([&](int op, const Participant& p) {
+        received_op = op;
+        received_id = p.id();
+    });
+
+    participant_info pi;
+    pi.participant_id   = 77;
+    pi.participant_name = nullptr;
+    mock_trigger_user_update(USER_JOIN, &pi);
+
+    CHECK(received_op == USER_JOIN);
+    CHECK(received_id == 77);
+}
+
+// ============================================================================
+// Event subscription
+// ============================================================================
+
+TEST_CASE("subscribeEvent before join queues events for later", "[client][events]") {
+    R _;
+    Client c;
+    // Not joined yet — subscribe should NOT call sdk immediately
+    c.subscribeEvent({1, 2, 3});
+    CHECK(g_mock_state.subscribe_calls == 0);
+}
+
+TEST_CASE("pending subscriptions flushed on join confirm", "[client][events]") {
+    R _;
+    Client c;
+    c.subscribeEvent({4, 5});
+    CHECK(g_mock_state.subscribe_calls == 0);
+
+    c.join("u", "s", "sig", "url");
+    mock_trigger_join_confirm(0);
+
+    CHECK(g_mock_state.subscribe_calls == 1);
+    // The pending events should have been sent
+    REQUIRE(g_mock_state.last_subscribed_events.size() == 2);
+    CHECK(g_mock_state.last_subscribed_events[0] == 4);
+    CHECK(g_mock_state.last_subscribed_events[1] == 5);
+}
+
+TEST_CASE("subscribeEvent after join confirm fires sdk immediately", "[client][events]") {
+    R _;
+    Client c;
+    c.join("u", "s", "sig", "url");
+    mock_trigger_join_confirm(0);
+    g_mock_state.subscribe_calls = 0; // reset counter (was called for pending)
+
+    c.subscribeEvent({7});
+    CHECK(g_mock_state.subscribe_calls == 1);
+    CHECK(g_mock_state.last_subscribed_events[0] == 7);
+}
+
+TEST_CASE("setOnUserUpdate auto-subscribes to participant events", "[client][events]") {
+    R _;
+    Client c;
+    c.setOnUserUpdate([](int, const Participant&) {});
+    // Before join: should be queued (not sent to sdk yet)
+    CHECK(g_mock_state.subscribe_calls == 0);
+
+    c.join("u", "s", "sig", "url");
+    mock_trigger_join_confirm(0);
+
+    // After join confirm, pending subscriptions should include participant events
+    REQUIRE(g_mock_state.subscribe_calls >= 1);
+    auto& evts = g_mock_state.last_subscribed_events;
+    bool has_join  = std::find(evts.begin(), evts.end(), (int)EVENT_TYPE::PARTICIPANT_JOIN)  != evts.end();
+    bool has_leave = std::find(evts.begin(), evts.end(), (int)EVENT_TYPE::PARTICIPANT_LEAVE) != evts.end();
+    CHECK(has_join);
+    CHECK(has_leave);
+}
+
+// ============================================================================
+// Media type auto-enable
+// ============================================================================
+
+TEST_CASE("setOnAudioData enables AUDIO media type via config on join", "[client][media]") {
+    R _;
+    Client c;
+    c.setOnAudioData([](const std::vector<uint8_t>&, uint64_t, const Metadata&) {});
+    // Config is deferred until join() calls open() — no sdk_->config() before open()
+    CHECK(g_mock_state.config_calls == 0);
+    c.join("u", "s", "sig", "url");
+    REQUIRE(g_mock_state.config_calls >= 1);
+    CHECK(g_mock_state.last_media_types & Client::AUDIO);
+}
+
+TEST_CASE("setOnVideoData enables VIDEO media type via config on join", "[client][media]") {
+    R _;
+    Client c;
+    c.setOnVideoData([](const std::vector<uint8_t>&, uint64_t, const Metadata&) {});
+    // Config is deferred until join() — calling sdk_->config() before open()
+    // hangs for VIDEO in the real C++ SDK
+    CHECK(g_mock_state.config_calls == 0);
+    c.join("u", "s", "sig", "url");
+    REQUIRE(g_mock_state.config_calls >= 1);
+    CHECK(g_mock_state.last_media_types & Client::VIDEO);
+}
+
+TEST_CASE("setOnTranscriptData enables TRANSCRIPT media type via config on join", "[client][media]") {
+    R _;
+    Client c;
+    c.setOnTranscriptData([](const std::vector<uint8_t>&, uint64_t, const Metadata&) {});
+    // Config is deferred until join()
+    CHECK(g_mock_state.config_calls == 0);
+    c.join("u", "s", "sig", "url");
+    REQUIRE(g_mock_state.config_calls >= 1);
+    CHECK(g_mock_state.last_media_types & Client::TRANSCRIPT);
+}
+
+// ============================================================================
+// TranscriptParams
+// ============================================================================
+
+TEST_CASE("TranscriptParams default constructor", "[params][transcript]") {
+    R _;
+    TranscriptParams p;
+    CHECK(p.contentType()   == 5);     // TEXT
+    CHECK(p.srcLanguage()   == -1);    // LANGUAGE_ID_NONE — auto-detect
+    CHECK(p.enableLid()     == true);
+}
+
+TEST_CASE("TranscriptParams setters round-trip", "[params][transcript]") {
+    R _;
+    TranscriptParams p;
+    p.setSrcLanguage(9);   // LANGUAGE_ID_ENGLISH
+    p.setEnableLid(false);
+    CHECK(p.srcLanguage() == 9);
+    CHECK(p.enableLid()   == false);
+}
+
+TEST_CASE("TranscriptParams toNative() maps fields correctly", "[params][transcript]") {
+    R _;
+    TranscriptParams p;
+    p.setSrcLanguage(9);
+    p.setEnableLid(false);
+    auto n = p.toNative();
+    CHECK(n.content_type  == 5);
+    CHECK(n.src_language  == 9);
+    CHECK(n.enable_lid    == false);
+}
+
+TEST_CASE("MediaParams setTranscriptParams / hasTranscriptParams", "[params][media][transcript]") {
+    R _;
+    MediaParams mp;
+    CHECK_FALSE(mp.hasTranscriptParams());
+
+    TranscriptParams tp;
+    mp.setTranscriptParams(tp);
+    REQUIRE(mp.hasTranscriptParams());
+    CHECK(mp.transcriptParams().contentType() == 5);
+}
+
+TEST_CASE("MediaParams toNative() produces non-null tr_param when set", "[params][media][transcript]") {
+    R _;
+    MediaParams mp;
+    mp.setTranscriptParams(TranscriptParams());
+    auto n = mp.toNative();
+    REQUIRE(n.tr_param != nullptr);
+    CHECK(n.tr_param->content_type == 5);
+    CHECK(n.tr_param->enable_lid   == true);
+    delete n.tr_param;
+}
+
+TEST_CASE("Client::setTranscriptParams calls config with updated params", "[client][transcript]") {
+    R _;
+    Client c;
+    // Enable transcript first so the reconfigure path runs
+    c.setOnTranscriptData([](const std::vector<uint8_t>&, uint64_t, const Metadata&) {});
+    // Must join first — configure() is deferred until after open() sets sdk_opened_=true
+    c.join("u", "s", "sig", "url");
+    int calls_before = g_mock_state.config_calls;
+
+    TranscriptParams tp;
+    tp.setSrcLanguage(14); // LANGUAGE_ID_GERMAN
+    c.setTranscriptParams(tp);
+
+    // Should have called config() again with the updated params
+    CHECK(g_mock_state.config_calls > calls_before);
+}
+
+// ============================================================================
+// Client::setProxy
+// ============================================================================
+
+TEST_CASE("Client::setProxy forwards proxy_type and proxy_url to SDK", "[client][proxy]") {
+    R _;
+    Client c;
+    c.setProxy("http", "http://proxy.example.com:8080");
+    CHECK(g_mock_state.proxy_calls    == 1);
+    CHECK(g_mock_state.last_proxy_type == "http");
+    CHECK(g_mock_state.last_proxy_url  == "http://proxy.example.com:8080");
+}
+
+TEST_CASE("Client::setProxy works for https proxy", "[client][proxy]") {
+    R _;
+    Client c;
+    c.setProxy("https", "https://proxy.example.com:8080");
+    CHECK(g_mock_state.proxy_calls    == 1);
+    CHECK(g_mock_state.last_proxy_type == "https");
+    CHECK(g_mock_state.last_proxy_url  == "https://proxy.example.com:8080");
+}
+
+TEST_CASE("Client::setProxy throws on SDK failure", "[client][proxy]") {
+    R _;
+    g_mock_state.proxy_result = RTMS_SDK_FAILURE;
+    Client c;
+    REQUIRE_THROWS_WITH(
+        c.setProxy("http", "http://proxy.example.com:8080"),
+        ContainsSubstring("setProxy failed")
+    );
+}
+
+// ============================================================================
+// Client::subscribeVideo
+// ============================================================================
+
+TEST_CASE("Client::subscribeVideo forwards user_id and true to send_subscript_video", "[client][video]") {
+    R _;
+    Client c;
+    c.subscribeVideo(12345, true);
+    CHECK(g_mock_state.subscript_video_calls    == 1);
+    CHECK(g_mock_state.last_subscript_user_id   == 12345);
+    CHECK(g_mock_state.last_subscript_is_sub    == true);
+}
+
+TEST_CASE("Client::subscribeVideo forwards user_id and false to send_subscript_video", "[client][video]") {
+    R _;
+    Client c;
+    c.subscribeVideo(99999, false);
+    CHECK(g_mock_state.subscript_video_calls    == 1);
+    CHECK(g_mock_state.last_subscript_user_id   == 99999);
+    CHECK(g_mock_state.last_subscript_is_sub    == false);
+}
+
+TEST_CASE("Client::subscribeVideo throws on SDK failure", "[client][video]") {
+    R _;
+    g_mock_state.subscript_video_result = RTMS_SDK_FAILURE;
+    Client c;
+    REQUIRE_THROWS_WITH(
+        c.subscribeVideo(12345, true),
+        ContainsSubstring("subscribeVideo failed")
+    );
+}
+
+// ============================================================================
+// on_participant_video callback
+// ============================================================================
+
+TEST_CASE("on_participant_video fires registered callback with users and flag", "[client][video][callbacks]") {
+    R _;
+    Client c;
+    c.join("u", "s", "sig", "url");
+
+    std::vector<int> received_users;
+    bool received_is_on = false;
+    c.setOnParticipantVideo([&](const std::vector<int>& users, bool is_on) {
+        received_users = users;
+        received_is_on = is_on;
+    });
+
+    mock_trigger_participant_video({11111, 22222}, true);
+
+    REQUIRE(received_users.size() == 2);
+    CHECK(received_users[0] == 11111);
+    CHECK(received_users[1] == 22222);
+    CHECK(received_is_on    == true);
+}
+
+TEST_CASE("on_participant_video fires with is_on=false when video turns off", "[client][video][callbacks]") {
+    R _;
+    Client c;
+    c.join("u", "s", "sig", "url");
+
+    bool received_is_on = true;
+    c.setOnParticipantVideo([&](const std::vector<int>&, bool is_on) {
+        received_is_on = is_on;
+    });
+
+    mock_trigger_participant_video({11111}, false);
+    CHECK(received_is_on == false);
+}
+
+TEST_CASE("on_participant_video does not fire when no callback is registered", "[client][video][callbacks]") {
+    R _;
+    Client c;
+    c.join("u", "s", "sig", "url");
+    // No callback registered — should not crash
+    REQUIRE_NOTHROW(mock_trigger_participant_video({11111}, true));
+}
+
+// ============================================================================
+// on_video_subscript_resp callback
+// ============================================================================
+
+TEST_CASE("on_video_subscript_resp fires registered callback with user_id, status, error", "[client][video][callbacks]") {
+    R _;
+    Client c;
+    c.join("u", "s", "sig", "url");
+
+    int received_user_id = -1;
+    int received_status  = -1;
+    std::string received_error;
+    c.setOnVideoSubscribed([&](int user_id, int status, const std::string& error) {
+        received_user_id = user_id;
+        received_status  = status;
+        received_error   = error;
+    });
+
+    mock_trigger_video_subscript_resp(12345, 0, "");
+
+    CHECK(received_user_id == 12345);
+    CHECK(received_status  == 0);
+    CHECK(received_error.empty());
+}
+
+TEST_CASE("on_video_subscript_resp relays error string on failure", "[client][video][callbacks]") {
+    R _;
+    Client c;
+    c.join("u", "s", "sig", "url");
+
+    int received_status = 0;
+    std::string received_error;
+    c.setOnVideoSubscribed([&](int, int status, const std::string& error) {
+        received_status = status;
+        received_error  = error;
+    });
+
+    mock_trigger_video_subscript_resp(12345, RTMS_SDK_FAILURE, "subscription failed");
+
+    CHECK(received_status == RTMS_SDK_FAILURE);
+    CHECK(received_error  == "subscription failed");
+}
+
+TEST_CASE("on_video_subscript_resp does not fire when no callback is registered", "[client][video][callbacks]") {
+    R _;
+    Client c;
+    c.join("u", "s", "sig", "url");
+    REQUIRE_NOTHROW(mock_trigger_video_subscript_resp(12345, 0, ""));
+}
+
+// ============================================================================
+// setOnParticipantVideo auto-subscribe
+// ============================================================================
+
+TEST_CASE("setOnParticipantVideo auto-subscribes to video on/off events", "[client][events]") {
+    R _;
+    Client c;
+    c.setOnParticipantVideo([](const std::vector<int>&, bool) {});
+    // Before join: subscriptions should be queued, not sent to sdk yet
+    CHECK(g_mock_state.subscribe_calls == 0);
+
+    c.join("u", "s", "sig", "url");
+    mock_trigger_join_confirm(0);
+
+    // After join confirm, pending subscriptions must include both video events
+    REQUIRE(g_mock_state.subscribe_calls >= 1);
+    auto& evts = g_mock_state.last_subscribed_events;
+    bool has_on  = std::find(evts.begin(), evts.end(), (int)EVENT_TYPE::PARTICIPANT_VIDEO_ON)  != evts.end();
+    bool has_off = std::find(evts.begin(), evts.end(), (int)EVENT_TYPE::PARTICIPANT_VIDEO_OFF) != evts.end();
+    CHECK(has_on);
+    CHECK(has_off);
+}
+
+// ============================================================================
+// Metadata — startTs / endTs / AiInterpreter
+// ============================================================================
+
+TEST_CASE("Metadata construction — start_ts and end_ts", "[data][metadata]") {
+    R _;
+
+    SECTION("start_ts and end_ts are copied correctly") {
+        rtms_metadata md{};
+        md.start_ts = 1000000ULL;
+        md.end_ts   = 2000000ULL;
+        Metadata m(md);
+        CHECK(m.startTs() == 1000000ULL);
+        CHECK(m.endTs()   == 2000000ULL);
+    }
+
+    SECTION("zero timestamps are preserved") {
+        rtms_metadata md{};
+        md.start_ts = 0;
+        md.end_ts   = 0;
+        Metadata m(md);
+        CHECK(m.startTs() == 0ULL);
+        CHECK(m.endTs()   == 0ULL);
+    }
+}
+
+TEST_CASE("AiInterpreter construction — zero targets when target_size is zero", "[data][metadata]") {
+    R _;
+    rtms_metadata md{};
+    md.aii.lid         = 9;        // ENGLISH
+    md.aii.timestamp   = 12345ULL;
+    md.aii.channel_num = 1;
+    md.aii.sample_rate = 16000;
+    md.aii.target_size = 0;
+
+    Metadata m(md);
+    const auto& ai = m.aiInterpreter();
+    CHECK(ai.lid()        == 9);
+    CHECK(ai.timestamp()  == 12345ULL);
+    CHECK(ai.channelNum() == 1);
+    CHECK(ai.sampleRate() == 16000);
+    CHECK(ai.targets().empty());
+}
+
+TEST_CASE("AiInterpreter construction — target_size guarded against out-of-bounds values", "[data][metadata]") {
+    R _;
+
+    SECTION("target_size larger than atl array is clamped") {
+        rtms_metadata md{};
+        md.aii.target_size = 999;  // atl has only 100 entries
+        // Must not read past bounds — just constructing should be safe
+        Metadata m(md);
+        CHECK(m.aiInterpreter().targets().size() <= 100);
+    }
+
+    SECTION("negative target_size produces empty targets") {
+        rtms_metadata md{};
+        md.aii.target_size = -1;
+        Metadata m(md);
+        CHECK(m.aiInterpreter().targets().empty());
+    }
+}
+
+TEST_CASE("AiInterpreter construction — one target populated correctly", "[data][metadata]") {
+    R _;
+    rtms_metadata md{};
+    md.aii.lid         = 9;
+    md.aii.timestamp   = 100ULL;
+    md.aii.channel_num = 2;
+    md.aii.sample_rate = 48000;
+    md.aii.target_size = 1;
+    md.aii.atl[0].lid    = 14;  // GERMAN
+    md.aii.atl[0].toneid = 0;
+    strncpy(md.aii.atl[0].voice_id, "voice-de-1", MAX_VOICE_ID_LEN - 1);
+    strncpy(md.aii.atl[0].engine,   "engine-A",   MAX_ENGINE_LEN - 1);
+
+    Metadata m(md);
+    const auto& ai = m.aiInterpreter();
+    REQUIRE(ai.targets().size() == 1);
+    const auto& tgt = ai.targets()[0];
+    CHECK(tgt.lid()     == 14);
+    CHECK(tgt.toneId()  == 0);
+    CHECK(tgt.voiceId() == "voice-de-1");
+    CHECK(tgt.engine()  == "engine-A");
+}
diff --git a/tests/py/test_rtms.py b/tests/py/test_rtms.py
new file mode 100644
index 0000000..d6419d4
--- /dev/null
+++ b/tests/py/test_rtms.py
@@ -0,0 +1,1037 @@
+#!/usr/bin/env python3
+"""
+Comprehensive test suite for Python RTMS SDK
+
+Mirrors the Node.js test coverage in tests/rtms.test.ts
+"""
+
+import pytest
+import rtms
+from unittest.mock import Mock, patch, MagicMock
+import json
+import asyncio
+import inspect
+
+
+@pytest.fixture(autouse=True)
+def clean_rtms_state():
+    """Clear global rtms state before each test.
+
+    Prevents clients created in earlier tests from leaking into later tests
+    via the _clients registry, which would cause run_async(stop_on_empty=True)
+    to never exit and _cleanup_all_clients() to block on unjoined C++ clients.
+    """
+    with rtms._clients_lock:
+        rtms._clients.clear()
+    rtms._running = False
+    rtms._stop_event.clear()
+    yield
+
+
+class TestConstants:
+    """Test that all constants are properly defined"""
+
+    def test_media_type_constants(self):
+        """Test media type constants match expected values"""
+        assert rtms.MEDIA_TYPE_AUDIO == 1
+        assert rtms.MEDIA_TYPE_VIDEO == 2
+        assert rtms.MEDIA_TYPE_DESKSHARE == 4
+        assert rtms.MEDIA_TYPE_TRANSCRIPT == 8
+        assert rtms.MEDIA_TYPE_CHAT == 16
+        assert rtms.MEDIA_TYPE_ALL == 32
+
+    def test_session_event_constants(self):
+        """Test session event constants"""
+        assert rtms.SESSION_EVENT_ADD == 1
+        assert rtms.SESSION_EVENT_STOP == 2
+        assert rtms.SESSION_EVENT_PAUSE == 3
+        assert rtms.SESSION_EVENT_RESUME == 4
+
+    def test_event_type_constants(self):
+        """Test event type constants (for subscribeEvent/unsubscribeEvent)"""
+        assert rtms.EVENT_UNDEFINED == 0
+        assert rtms.EVENT_FIRST_PACKET_TIMESTAMP == 1
+        assert rtms.EVENT_ACTIVE_SPEAKER_CHANGE == 2
+        assert rtms.EVENT_PARTICIPANT_JOIN == 3
+        assert rtms.EVENT_PARTICIPANT_LEAVE == 4
+        assert rtms.EVENT_SHARING_START == 5
+        assert rtms.EVENT_SHARING_STOP == 6
+        assert rtms.EVENT_MEDIA_CONNECTION_INTERRUPTED == 7
+        assert rtms.EVENT_PARTICIPANT_VIDEO_ON == 8
+        assert rtms.EVENT_PARTICIPANT_VIDEO_OFF == 9
+        assert rtms.EVENT_CONSUMER_ANSWERED == 8
+        assert rtms.EVENT_CONSUMER_END == 9
+        assert rtms.EVENT_USER_ANSWERED == 10
+        assert rtms.EVENT_USER_END == 11
+        assert rtms.EVENT_USER_HOLD == 12
+        assert rtms.EVENT_USER_UNHOLD == 13
+
+    def test_status_constants(self):
+        """Test SDK status constants"""
+        assert rtms.RTMS_SDK_FAILURE == -1
+        assert rtms.RTMS_SDK_OK == 0
+        assert rtms.RTMS_SDK_TIMEOUT == 1
+        assert rtms.RTMS_SDK_NOT_EXIST == 2
+
+    def test_audio_codec_constants(self):
+        """Test audio codec constants exist"""
+        assert 'OPUS' in rtms.AudioCodec.__members__
+        assert 'L16' in rtms.AudioCodec.__members__
+        assert 'G711' in rtms.AudioCodec.__members__
+
+    def test_video_codec_constants(self):
+        """Test video codec constants exist"""
+        assert 'H264' in rtms.VideoCodec.__members__
+        assert 'JPG' in rtms.VideoCodec.__members__
+
+    def test_sample_rate_constants(self):
+        """Test audio sample rate constants"""
+        assert 'SR_48K' in rtms.AudioSampleRate.__members__
+        assert 'SR_16K' in rtms.AudioSampleRate.__members__
+        assert 'SR_8K' in rtms.AudioSampleRate.__members__
+
+
+class TestLogging:
+    """Test logging functionality"""
+
+    def test_log_levels_enum(self):
+        """Test LogLevel enum values"""
+        assert rtms.LogLevel.ERROR == 0
+        assert rtms.LogLevel.WARN == 1
+        assert rtms.LogLevel.INFO == 2
+        assert rtms.LogLevel.DEBUG == 3
+        assert rtms.LogLevel.TRACE == 4
+
+    def test_log_format_enum(self):
+        """Test LogFormat enum values"""
+        assert rtms.LogFormat.PROGRESSIVE == 'progressive'
+        assert rtms.LogFormat.JSON == 'json'
+
+    def test_configure_logger(self):
+        """Test logger configuration"""
+        # Should not raise errors
+        rtms.configure_logger({
+            'level': 'debug',
+            'format': 'json',
+            'enabled': True
+        })
+
+        rtms.configure_logger({
+            'level': 'info',
+            'format': 'progressive',
+            'enabled': False
+        })
+
+    def test_log_functions_exist(self):
+        """Test that all log functions are available"""
+        assert callable(rtms.log_debug)
+        assert callable(rtms.log_info)
+        assert callable(rtms.log_warn)
+        assert callable(rtms.log_error)
+
+
+class TestClient:
+    """Test Client class functionality"""
+
+    def test_client_instantiation(self):
+        """Test that Client can be instantiated"""
+        client = rtms.Client()
+        assert client is not None
+        assert isinstance(client, rtms.Client)
+
+    def test_client_has_join_method(self):
+        """Test that Client has join method"""
+        client = rtms.Client()
+        assert hasattr(client, 'join')
+        assert callable(client.join)
+
+    def test_client_has_leave_method(self):
+        """Test that Client has leave method"""
+        client = rtms.Client()
+        assert hasattr(client, 'leave')
+        assert callable(client.leave)
+
+    def test_client_has_callback_methods(self):
+        """Test that Client has all callback registration methods"""
+        client = rtms.Client()
+        assert hasattr(client, 'onJoinConfirm')
+        assert hasattr(client, 'onSessionUpdate')
+        assert hasattr(client, 'onParticipantEvent')
+        assert hasattr(client, 'onActiveSpeakerEvent')
+        assert hasattr(client, 'onSharingEvent')
+        assert hasattr(client, 'onEventEx')
+        assert hasattr(client, 'onAudioData')
+        assert hasattr(client, 'onVideoData')
+        assert hasattr(client, 'onDeskshareData')
+        assert hasattr(client, 'onTranscriptData')
+        assert hasattr(client, 'onLeave')
+
+    def test_client_has_param_methods(self):
+        """Test that Client has parameter configuration methods"""
+        client = rtms.Client()
+        assert hasattr(client, 'setAudioParams')
+        assert hasattr(client, 'setVideoParams')
+        assert hasattr(client, 'setDeskshareParams')
+
+    def test_client_has_event_subscription_methods(self):
+        """Test that Client has event subscription methods"""
+        client = rtms.Client()
+        assert hasattr(client, 'subscribeEvent')
+        assert callable(client.subscribeEvent)
+        assert hasattr(client, 'unsubscribeEvent')
+
+    def test_client_has_individual_video_methods(self):
+        """Test that Client has individual video stream methods"""
+        client = rtms.Client()
+        assert hasattr(client, 'subscribe_video')
+        assert callable(client.subscribe_video)
+        assert hasattr(client, 'subscribeVideo')   # camelCase alias
+        assert hasattr(client, 'on_participant_video')
+        assert callable(client.on_participant_video)
+        assert hasattr(client, 'onParticipantVideo')
+        assert hasattr(client, 'on_video_subscribed')
+        assert callable(client.on_video_subscribed)
+        assert hasattr(client, 'onVideoSubscribed')
+
+    def test_on_participant_video_accepts_callback(self):
+        """Test that on_participant_video registers a callback without error"""
+        client = rtms.Client()
+        client.on_participant_video(lambda user_ids, is_on: None)
+
+    def test_on_video_subscribed_accepts_callback(self):
+        """Test that on_video_subscribed registers a callback without error"""
+        client = rtms.Client()
+        client.on_video_subscribed(lambda user_id, status, error: None)
+        assert callable(client.unsubscribeEvent)
+
+    def test_client_callback_registration(self):
+        """Test callback registration doesn't raise errors"""
+        client = rtms.Client()
+
+        @client.onJoinConfirm
+        def on_join(reason):
+            pass
+
+        @client.onTranscriptData
+        def on_transcript(data, size, timestamp, metadata):
+            pass
+
+        # Should complete without errors
+        assert True
+
+
+class TestParameterValidation:
+    """Test parameter validation functionality"""
+
+    def test_audio_params_exist(self):
+        """Test AudioParams class exists"""
+        assert hasattr(rtms, 'AudioParams')
+
+    def test_video_params_exist(self):
+        """Test VideoParams class exists"""
+        assert hasattr(rtms, 'VideoParams')
+
+    def test_deskshare_params_exist(self):
+        """Test DeskshareParams class exists"""
+        assert hasattr(rtms, 'DeskshareParams')
+
+
+class TestUtilityFunctions:
+    """Test utility functions"""
+
+    def test_generate_signature_exists(self):
+        """Test signature generation function exists"""
+        assert hasattr(rtms, 'generate_signature')
+        assert callable(rtms.generate_signature)
+
+    def test_find_ca_certificate_exists(self):
+        """Test CA certificate finder exists"""
+        assert hasattr(rtms, 'find_ca_certificate')
+        assert callable(rtms.find_ca_certificate)
+
+    def test_initialize_exists(self):
+        """Test initialize function exists"""
+        assert hasattr(rtms, 'initialize')
+        assert callable(rtms.initialize)
+
+    def test_uninitialize_exists(self):
+        """Test uninitialize function exists"""
+        assert hasattr(rtms, 'uninitialize')
+        assert callable(rtms.uninitialize)
+
+
+class TestWebhookFunctionality:
+    """Test webhook-related functionality"""
+
+    def test_on_webhook_event_exists(self):
+        """Test webhook event decorator exists"""
+        assert hasattr(rtms, 'onWebhookEvent')
+        assert callable(rtms.onWebhookEvent)
+        # Also test the alias
+        assert hasattr(rtms, 'on_webhook_event')
+        assert callable(rtms.on_webhook_event)
+
+    def test_webhook_response_class_exists(self):
+        """Test WebhookResponse class exists"""
+        # WebhookResponse is internal but should be importable
+        from rtms import WebhookResponse
+        assert WebhookResponse is not None
+
+    def test_client_webhook_decorator(self):
+        """Test client webhook event decorator"""
+        client = rtms.Client()
+        assert hasattr(client, 'on_webhook_event')
+        assert callable(client.on_webhook_event)
+
+        # Test decorator usage
+        @client.on_webhook_event()
+        def handle_webhook(payload):
+            pass
+
+        assert True
+
+
+class TestDataClasses:
+    """Test data classes and structures"""
+
+    def test_session_class_exists(self):
+        """Test Session class exists"""
+        assert hasattr(rtms, 'Session')
+
+    def test_participant_class_exists(self):
+        """Test Participant class exists"""
+        assert hasattr(rtms, 'Participant')
+
+    def test_metadata_class_exists(self):
+        """Test Metadata class exists"""
+        assert hasattr(rtms, 'Metadata')
+
+
+class TestModuleExports:
+    """Test that all expected exports are available"""
+
+    def test_client_export(self):
+        """Test Client is exported"""
+        assert 'Client' in rtms.__all__
+
+    def test_constant_exports(self):
+        """Test constants are exported"""
+        exports_to_check = [
+            'MEDIA_TYPE_AUDIO',
+            'MEDIA_TYPE_VIDEO',
+            'MEDIA_TYPE_TRANSCRIPT',
+            'SESSION_EVENT_ADD',
+            'EVENT_PARTICIPANT_JOIN',
+            'EVENT_PARTICIPANT_LEAVE',
+            'EVENT_ACTIVE_SPEAKER_CHANGE',
+            'EVENT_SHARING_START',
+            'EVENT_SHARING_STOP',
+            'RTMS_SDK_OK'
+        ]
+        for export in exports_to_check:
+            assert export in rtms.__all__
+
+    def test_function_exports(self):
+        """Test functions are exported"""
+        functions = [
+            'initialize',
+            'uninitialize',
+            'generate_signature',
+            'configure_logger'
+        ]
+        for func in functions:
+            assert func in rtms.__all__
+
+    def test_enum_exports(self):
+        """Test enums are exported"""
+        enums = [
+            'AudioCodec',
+            'VideoCodec',
+            'AudioSampleRate',
+            'VideoResolution',
+            'LogLevel',
+            'LogFormat'
+        ]
+        for enum in enums:
+            assert enum in rtms.__all__
+
+
+class TestThreadSafety:
+    """Test thread-safe features"""
+
+    def test_module_has_run_function(self):
+        """Test module has run() function for event loop"""
+        assert hasattr(rtms, 'run')
+        assert callable(rtms.run)
+
+    def test_module_has_stop_function(self):
+        """Test module has stop() function to signal shutdown"""
+        assert hasattr(rtms, 'stop')
+        assert callable(rtms.stop)
+
+    def test_client_has_polling_control(self):
+        """Test client has EventLoop integration attributes"""
+        client = rtms.Client()
+        assert hasattr(client, '_do_alloc_and_join')
+        assert hasattr(client, '_pending_join_params')
+
+
+class TestTranscriptParams:
+    """Test TranscriptParams class and setTranscriptParams/set_transcript_params.
+
+    These tests fail before implementation because TranscriptParams does not exist
+    and Client has no set_transcript_params() method.
+    """
+
+    def test_transcript_params_class_exists(self):
+        """rtms.TranscriptParams should be importable."""
+        assert hasattr(rtms, 'TranscriptParams')
+
+    def test_transcript_params_default_content_type(self):
+        """Default content_type should be 5 (TEXT)."""
+        p = rtms.TranscriptParams()
+        assert p.content_type == 5
+
+    def test_transcript_params_default_src_language(self):
+        """Default src_language should be -1 (LANGUAGE_ID_NONE, auto-detect)."""
+        p = rtms.TranscriptParams()
+        assert p.src_language == -1
+
+    def test_transcript_params_default_enable_lid(self):
+        """Default enable_lid should be True."""
+        p = rtms.TranscriptParams()
+        assert p.enable_lid is True
+
+    def test_transcript_params_setters(self):
+        """src_language and enable_lid setters should round-trip."""
+        p = rtms.TranscriptParams()
+        p.src_language = 9   # LANGUAGE_ID_ENGLISH
+        p.enable_lid = False
+        assert p.src_language == 9
+        assert p.enable_lid is False
+
+    def test_transcript_language_dict_exists(self):
+        """TranscriptLanguage constant dict should exist."""
+        assert hasattr(rtms, 'TranscriptLanguage')
+
+    def test_transcript_language_english(self):
+        """TranscriptLanguage['ENGLISH'] should be 9."""
+        assert rtms.TranscriptLanguage['ENGLISH'] == 9
+
+    def test_transcript_language_none(self):
+        """TranscriptLanguage['NONE'] should be -1 (auto-detect)."""
+        assert rtms.TranscriptLanguage['NONE'] == -1
+
+    def test_client_has_set_transcript_params(self):
+        """Client should expose set_transcript_params() method."""
+        client = rtms.Client()
+        assert hasattr(client, 'set_transcript_params')
+        assert callable(client.set_transcript_params)
+
+    def test_set_transcript_params_does_not_raise(self):
+        """Calling set_transcript_params with a valid TranscriptParams should not raise."""
+        client = rtms.Client()
+        p = rtms.TranscriptParams()
+        p.src_language = rtms.TranscriptLanguage['ENGLISH']
+        client.set_transcript_params(p)
+
+    def test_transcript_params_exported_in_all(self):
+        """TranscriptParams and TranscriptLanguage dict should appear in rtms.__all__."""
+        assert 'TranscriptParams' in rtms.__all__
+        assert 'TranscriptLanguage' in rtms.__all__
+
+
+class TestZccEngagementId:
+    """Test ZCC engagement_id support in join()
+
+    ZCC (Zoom Contact Center) identifies sessions with an engagement_id instead
+    of a meeting_uuid. join() must accept engagement_id and route it as the
+    instance identifier to the native SDK.
+
+    These tests fail before implementation because _do_join() raises:
+        ValueError("Either meeting_uuid, webinar_uuid, or session_id is required")
+    when only engagement_id is supplied.
+    """
+
+    def _make_client_with_mocked_native(self):
+        """Return a Client instance with native join() and polling mocked out."""
+        client = rtms.Client()
+        # Patch the native C++ join on the base class so no real SDK call is made
+        NativeClient = rtms.Client.__bases__[0]
+        return client, NativeClient
+
+    def test_join_with_engagement_id_returns_true(self):
+        """join() called with engagement_id should return True, not False."""
+        client, NativeClient = self._make_client_with_mocked_native()
+        # join() stores params and returns True immediately (defers to EventLoop)
+        with patch.object(client, '_do_alloc_and_join'):
+            result = client.join(
+                engagement_id='engagement-abc-123',
+                rtms_stream_id='stream-xyz',
+                server_urls='wss://rtms.zoom.us',
+                signature='mock-sig',
+            )
+        # Before implementation: False (ValueError caught, swallowed, returns False)
+        # After implementation: True
+        assert result is True
+
+    def test_do_join_with_engagement_id_does_not_raise(self):
+        """_do_alloc_and_join() with engagement_id should not raise a missing-identifier error."""
+        client, NativeClient = self._make_client_with_mocked_native()
+        client._pending_join_params = {
+            'engagement_id': 'engagement-abc-123',
+            'rtms_stream_id': 'stream-xyz',
+            'server_urls': 'wss://rtms.zoom.us',
+            'signature': 'mock-sig',
+            'timeout': -1,
+        }
+        with patch.object(NativeClient, 'join', return_value=None), \
+             patch.object(NativeClient, 'alloc', return_value=None):
+            # Before implementation: raises ValueError
+            # After implementation: should complete cleanly
+            client._do_alloc_and_join()
+
+    def test_engagement_id_passed_as_first_arg_to_native_join(self):
+        """engagement_id should be forwarded as the meeting_uuid positional arg."""
+        client, NativeClient = self._make_client_with_mocked_native()
+        client._pending_join_params = {
+            'engagement_id': 'eng-abc-123',
+            'rtms_stream_id': 'stream-xyz',
+            'server_urls': 'wss://rtms.zoom.us',
+            'signature': 'mock-sig',
+            'timeout': -1,
+        }
+        with patch.object(NativeClient, 'join', return_value=None) as mock_native_join, \
+             patch.object(NativeClient, 'alloc', return_value=None):
+            client._do_alloc_and_join()
+        mock_native_join.assert_called_once_with(
+            'eng-abc-123', 'stream-xyz', 'mock-sig', 'wss://rtms.zoom.us', -1
+        )
+
+    def test_engagement_id_lower_priority_than_meeting_uuid(self):
+        """When both meeting_uuid and engagement_id are supplied, meeting_uuid wins."""
+        client, NativeClient = self._make_client_with_mocked_native()
+        client._pending_join_params = {
+            'meeting_uuid': 'meeting-uuid-wins',
+            'engagement_id': 'engagement-id-loses',
+            'rtms_stream_id': 'stream-xyz',
+            'server_urls': 'wss://rtms.zoom.us',
+            'signature': 'mock-sig',
+            'timeout': -1,
+        }
+        with patch.object(NativeClient, 'join', return_value=None) as mock_native_join, \
+             patch.object(NativeClient, 'alloc', return_value=None):
+            client._do_alloc_and_join()
+        first_arg = mock_native_join.call_args[0][0]
+        assert first_arg == 'meeting-uuid-wins'
+
+
+class TestProxySupport:
+    """Test set_proxy() support.
+
+    Mirrors JS wrapper test coverage for set_proxy:
+    - exists as a callable
+    - does not raise for http proxy
+    - does not raise for https proxy
+
+    Python-specific additions:
+    - set_proxy legacy camelCase alias
+    - native forwarding verified via mock
+    """
+
+    def test_set_proxy_is_callable(self):
+        """set_proxy should be callable on a Client instance."""
+        client = rtms.Client()
+        assert callable(client.set_proxy)
+
+    def test_set_proxy_legacy_alias_is_callable(self):
+        """set_proxy legacy alias should exist for camelCase parity with Node.js."""
+        client = rtms.Client()
+        assert callable(client.setProxy)
+
+    def test_set_proxy_does_not_raise_for_http(self):
+        """set_proxy should not raise for an http proxy."""
+        client = rtms.Client()
+        NativeClient = rtms.Client.__bases__[0]
+        with patch.object(NativeClient, 'set_proxy', return_value=None):
+            client.set_proxy('http', 'http://proxy.example.com:8080')
+
+    def test_set_proxy_does_not_raise_for_https(self):
+        """set_proxy should not raise for an https proxy."""
+        client = rtms.Client()
+        NativeClient = rtms.Client.__bases__[0]
+        with patch.object(NativeClient, 'set_proxy', return_value=None):
+            client.set_proxy('https', 'https://proxy.example.com:8080')
+
+    def test_set_proxy_exported_in_all(self):
+        """set_proxy does not need to be in __all__ but the Client method must be accessible."""
+        client = rtms.Client()
+        assert callable(getattr(client, 'set_proxy', None))
+
+    def test_set_proxy_forwards_to_native(self):
+        """set_proxy should forward proxy_type and proxy_url to the native layer."""
+        client = rtms.Client()
+        NativeClient = rtms.Client.__bases__[0]
+        with patch.object(NativeClient, 'set_proxy', return_value=None) as mock_native:
+            client.set_proxy('http', 'http://proxy.example.com:8080')
+        mock_native.assert_called_once_with('http', 'http://proxy.example.com:8080')
+
+
+class TestIndividualVideoSubscription:
+    """Test subscribe_video() and on_participant_video / on_video_subscribed callbacks.
+
+    These tests fail before implementation because:
+    - Client has no subscribe_video() / subscribeVideo() method
+    - Client has no on_participant_video() / on_video_subscribed() callback registration
+
+    Per spec (branch 5):
+    - subscribe_video(user_id, True)  → subscribe to individual participant stream
+    - subscribe_video(user_id, False) → unsubscribe
+    - on_participant_video(callback)  → fires when participant video state changes
+                                        callback signature: (users: List[int], is_on: bool)
+    - on_video_subscribed(callback)   → fires in response to subscribe_video()
+                                        callback signature: (user_id: int, status: int, error: str)
+    """
+
+    def _native(self):
+        return rtms.Client.__bases__[0]
+
+    # ------------------------------------------------------------------
+    # subscribe_video method
+
+    def test_subscribe_video_is_callable(self):
+        """subscribe_video should be callable on a Client instance."""
+        client = rtms.Client()
+        assert callable(getattr(client, 'subscribe_video', None))
+
+    def test_subscribe_video_camelcase_alias_is_callable(self):
+        """subscribeVideo camelCase alias must exist for parity with Node.js."""
+        client = rtms.Client()
+        assert callable(getattr(client, 'subscribeVideo', None))
+
+    def test_subscribe_video_does_not_raise_for_subscribe(self):
+        """subscribe_video(user_id, True) should not raise."""
+        client = rtms.Client()
+        with patch.object(self._native(), 'subscribe_video', return_value=None):
+            client.subscribe_video(12345, True)
+
+    def test_subscribe_video_does_not_raise_for_unsubscribe(self):
+        """subscribe_video(user_id, False) should not raise."""
+        client = rtms.Client()
+        with patch.object(self._native(), 'subscribe_video', return_value=None):
+            client.subscribe_video(12345, False)
+
+    def test_subscribe_video_forwards_to_native(self):
+        """subscribe_video should forward user_id and subscribe flag to native layer."""
+        client = rtms.Client()
+        with patch.object(self._native(), 'subscribe_video', return_value=None) as mock_native:
+            client.subscribe_video(99999, True)
+        mock_native.assert_called_once_with(99999, True)
+
+    def test_subscribe_video_unsubscribe_forwards_to_native(self):
+        """subscribe_video(user_id, False) should forward False to native layer."""
+        client = rtms.Client()
+        with patch.object(self._native(), 'subscribe_video', return_value=None) as mock_native:
+            client.subscribe_video(99999, False)
+        mock_native.assert_called_once_with(99999, False)
+
+    # ------------------------------------------------------------------
+    # on_participant_video callback
+
+    def test_on_participant_video_is_callable(self):
+        """on_participant_video should be callable on a Client instance."""
+        client = rtms.Client()
+        assert callable(getattr(client, 'on_participant_video', None))
+
+    def test_on_participant_video_camelcase_alias_is_callable(self):
+        """onParticipantVideo camelCase alias must exist for Node.js parity."""
+        client = rtms.Client()
+        assert callable(getattr(client, 'onParticipantVideo', None))
+
+    def test_on_participant_video_does_not_raise(self):
+        """Registering an on_participant_video callback should not raise."""
+        client = rtms.Client()
+        client.on_participant_video(lambda *_: None)
+
+    def test_on_participant_video_callback_receives_users_and_flag(self):
+        """on_participant_video callback must receive a list of user IDs and a bool."""
+        client = rtms.Client()
+        received = {}
+
+        def cb(users, is_on):
+            received['users'] = users
+            received['is_on'] = is_on
+
+        client.on_participant_video(cb)
+        # Simulate the callback being fired by the native layer
+        client._participant_video_callback([11111, 22222], True)
+        assert received['users'] == [11111, 22222]
+        assert received['is_on'] is True
+
+    # ------------------------------------------------------------------
+    # on_video_subscribed callback
+
+    def test_on_video_subscribed_is_callable(self):
+        """on_video_subscribed should be callable on a Client instance."""
+        client = rtms.Client()
+        assert callable(getattr(client, 'on_video_subscribed', None))
+
+    def test_on_video_subscribed_camelcase_alias_is_callable(self):
+        """onVideoSubscribed camelCase alias must exist for Node.js parity."""
+        client = rtms.Client()
+        assert callable(getattr(client, 'onVideoSubscribed', None))
+
+    def test_on_video_subscribed_does_not_raise(self):
+        """Registering an on_video_subscribed callback should not raise."""
+        client = rtms.Client()
+        client.on_video_subscribed(lambda *_: None)
+
+    def test_on_video_subscribed_callback_receives_userid_status_error(self):
+        """on_video_subscribed callback must receive user_id (int), status (int), error (str)."""
+        client = rtms.Client()
+        received = {}
+
+        def cb(user_id, status, error):
+            received['user_id'] = user_id
+            received['status'] = status
+            received['error'] = error
+
+        client.on_video_subscribed(cb)
+        # Simulate the callback being fired by the native layer
+        client._video_subscribed_callback(12345, 0, '')
+        assert received['user_id'] == 12345
+        assert received['status'] == 0
+        assert received['error'] == ''
+
+    def test_on_video_subscribed_callback_reports_error_string(self):
+        """on_video_subscribed callback should relay a non-empty error string."""
+        client = rtms.Client()
+        received = {}
+
+        def cb(user_id, status, error):
+            received['user_id'] = user_id
+            received['status'] = status
+            received['error'] = error
+
+        client.on_video_subscribed(cb)
+        client._video_subscribed_callback(12345, -1, 'subscription failed')
+        assert received['status'] == -1
+        assert received['error'] == 'subscription failed'
+
+
+class TestGilRelease:
+    """Tests that poll() releases the GIL, allowing other Python threads to run concurrently."""
+
+    def test_poll_exists_on_client(self):
+        client = rtms.Client()
+        assert hasattr(client, 'poll')
+
+    def test_poll_callable(self):
+        client = rtms.Client()
+        # Should not raise (SDK errors are OK; what matters is it doesn't hang)
+        try:
+            client.poll()
+        except Exception:
+            pass
+
+    def test_poll_does_not_block_other_threads(self):
+        """Smoke test: poll() on an unjoined client returns without hanging.
+
+        NOTE: On an unjoined client with no active meeting, poll() returns in
+        microseconds — too fast to observe true GIL-release concurrency. This
+        test verifies poll() is callable and doesn't hang; it does NOT prove
+        GIL release in a loaded scenario.
+
+        True GIL-release verification requires either a live meeting (where
+        poll() blocks waiting for SDK events) or a mock SDK with a sleep().
+        The structural guarantee lives in src/python.cpp: py::gil_scoped_release.
+        """
+        import threading
+        results = []
+
+        def worker():
+            results.append(threading.get_ident())
+
+        client = rtms.Client()
+        t = threading.Thread(target=worker)
+        t.start()
+        try:
+            client.poll()
+        except Exception:
+            pass
+        t.join(timeout=2.0)
+        assert len(results) == 1, "Worker thread should have completed"
+
+
+class TestRunAsync:
+    """Tests the new run_async() asyncio-native event loop."""
+
+    def test_run_async_exists(self):
+        assert hasattr(rtms, 'run_async')
+
+    def test_run_async_is_coroutine_function(self):
+        assert inspect.iscoroutinefunction(rtms.run_async)
+
+    def test_run_async_accepts_poll_interval(self):
+        sig = inspect.signature(rtms.run_async)
+        assert 'poll_interval' in sig.parameters
+
+    def test_run_async_accepts_stop_on_empty(self):
+        sig = inspect.signature(rtms.run_async)
+        assert 'stop_on_empty' in sig.parameters
+
+    def test_run_async_stops_when_stop_called(self):
+        async def run_and_stop():
+            loop = asyncio.get_event_loop()
+            loop.call_later(0.05, rtms.stop)
+            await rtms.run_async(poll_interval=0.01)
+
+        asyncio.run(run_and_stop())  # Should complete, not hang
+
+    def test_run_async_stop_on_empty_exits(self):
+        async def run_empty():
+            await rtms.run_async(stop_on_empty=True, poll_interval=0.01)
+
+        asyncio.run(run_empty())  # No clients registered → exits immediately
+
+    def test_run_async_composes_with_gather(self):
+        """run_async() should work inside asyncio.gather without blocking."""
+        results = []
+
+        async def side_task():
+            results.append('side_task_ran')
+
+        async def test():
+            await asyncio.gather(
+                rtms.run_async(stop_on_empty=True, poll_interval=0.01),
+                side_task(),
+            )
+
+        asyncio.run(test())
+        assert 'side_task_ran' in results
+
+
+class TestExecutorSupport:
+    """Tests executor-based callback dispatch."""
+
+    def test_client_accepts_executor_kwarg(self):
+        from concurrent.futures import ThreadPoolExecutor
+        executor = ThreadPoolExecutor(max_workers=2)
+        client = rtms.Client(executor=executor)
+        assert client is not None
+        executor.shutdown(wait=False)
+
+    def test_client_stores_executor(self):
+        from concurrent.futures import ThreadPoolExecutor
+        executor = ThreadPoolExecutor(max_workers=2)
+        client = rtms.Client(executor=executor)
+        assert client._executor is executor
+        executor.shutdown(wait=False)
+
+    def test_client_no_executor_default(self):
+        client = rtms.Client()
+        assert client._executor is None
+
+    def test_run_accepts_executor_kwarg(self):
+        sig = inspect.signature(rtms.run)
+        assert 'executor' in sig.parameters
+
+    def test_run_executor_default_is_none(self):
+        sig = inspect.signature(rtms.run)
+        assert sig.parameters['executor'].default is None
+
+    def test_wrap_callback_exists(self):
+        client = rtms.Client()
+        assert hasattr(client, '_wrap_callback')
+
+    def test_sync_callback_no_executor_passthrough(self):
+        """Sync callback with no executor should be returned unchanged."""
+        client = rtms.Client()
+
+        def sync_cb(*_):
+            pass
+
+        wrapped = client._wrap_callback(sync_cb)
+        assert wrapped is sync_cb, "No executor → callback passed through unchanged"
+
+    def test_executor_wraps_sync_callback(self):
+        """When executor is set, sync callbacks are wrapped for thread pool dispatch."""
+        from concurrent.futures import ThreadPoolExecutor
+        executor = ThreadPoolExecutor(max_workers=1)
+        client = rtms.Client(executor=executor)
+
+        def audio_cb(*_):
+            pass
+
+        wrapped = client._wrap_callback(audio_cb)
+        assert wrapped is not audio_cb, "Executor wrapping should produce a different callable"
+        executor.shutdown(wait=False)
+
+    def test_async_callback_detected_and_wrapped(self):
+        """Async coroutine callbacks should be auto-wrapped for event loop dispatch."""
+        client = rtms.Client()
+
+        async def async_cb(*_):
+            pass
+
+        wrapped = client._wrap_callback(async_cb)
+        assert wrapped is not async_cb, "Async callback should be wrapped"
+
+    def test_executor_applied_to_video_data_callback(self):
+        from concurrent.futures import ThreadPoolExecutor
+        executor = ThreadPoolExecutor(max_workers=1)
+        client = rtms.Client(executor=executor)
+
+        def video_cb(*_):
+            pass
+
+        wrapped = client._wrap_callback(video_cb)
+        assert wrapped is not video_cb
+        executor.shutdown(wait=False)
+
+
+class TestContextManager:
+    """Tests `with rtms.Client() as client:` protocol."""
+
+    def test_client_has_enter(self):
+        assert hasattr(rtms.Client, '__enter__')
+
+    def test_client_has_exit(self):
+        assert hasattr(rtms.Client, '__exit__')
+
+    def test_enter_returns_client(self):
+        client = rtms.Client()
+        result = client.__enter__()
+        assert result is client
+
+    def test_exit_calls_leave(self):
+        client = rtms.Client()
+        with patch.object(client, 'leave') as mock_leave:
+            client.__exit__(None, None, None)
+        mock_leave.assert_called_once()
+
+    def test_with_statement_basic(self):
+        """with rtms.Client() as client: should work without error."""
+        with patch.object(rtms.Client, 'leave'):
+            with rtms.Client() as client:
+                assert client is not None
+
+    def test_exit_called_on_exception(self):
+        """leave() should be called even when an exception is raised inside with block."""
+        leave_called = []
+
+        class TrackingClient(rtms.Client):
+            def leave(self):
+                leave_called.append(True)
+                return True
+
+        client = TrackingClient()
+        try:
+            with client:
+                raise ValueError("test exception")
+        except ValueError:
+            pass
+
+        assert len(leave_called) == 1, "leave() should be called on exception"
+
+    def test_exit_does_not_suppress_exceptions(self):
+        """__exit__ should return False/None so exceptions propagate."""
+        client = rtms.Client()
+        with patch.object(client, 'leave'):
+            result = client.__exit__(ValueError, ValueError("test"), None)
+        assert not result  # False or None — do not suppress
+
+    def test_context_manager_integration(self):
+        """Full with-statement: enter returns client, leave called on exit."""
+        leave_called = []
+
+        with patch.object(rtms.Client, 'leave', side_effect=lambda: leave_called.append(True) or True):
+            with rtms.Client() as client:
+                assert client is not None
+
+        assert len(leave_called) == 1
+
+
+class TestSnakeCaseCallbacks:
+    """snake_case callback methods are canonical; camelCase are backward-compat aliases."""
+
+    def test_on_audio_data_canonical(self):
+        client = rtms.Client()
+        client.on_audio_data(lambda *_: None)
+
+    def test_on_video_data_canonical(self):
+        client = rtms.Client()
+        client.on_video_data(lambda *_: None)
+
+    def test_on_transcript_data_canonical(self):
+        client = rtms.Client()
+        client.on_transcript_data(lambda *_: None)
+
+    def test_on_deskshare_data_canonical(self):
+        client = rtms.Client()
+        client.on_deskshare_data(lambda *_: None)
+
+    def test_on_join_confirm_canonical(self):
+        client = rtms.Client()
+        client.on_join_confirm(lambda _: None)
+
+    def test_on_session_update_canonical(self):
+        client = rtms.Client()
+        client.on_session_update(lambda *_: None)
+
+    def test_on_user_update_canonical(self):
+        client = rtms.Client()
+        client.on_user_update(lambda *_: None)
+
+    def test_on_leave_canonical(self):
+        client = rtms.Client()
+        client.on_leave(lambda _: None)
+
+    def test_on_event_ex_canonical(self):
+        client = rtms.Client()
+        client.on_event_ex(lambda _: None)
+
+    def test_on_participant_event_canonical(self):
+        client = rtms.Client()
+        with patch.object(client, 'subscribeEvent'):
+            client.on_participant_event(lambda *_: None)
+
+    def test_on_active_speaker_event_canonical(self):
+        client = rtms.Client()
+        with patch.object(client, 'subscribeEvent'):
+            client.on_active_speaker_event(lambda *_: None)
+
+    def test_on_sharing_event_canonical(self):
+        client = rtms.Client()
+        with patch.object(client, 'subscribeEvent'):
+            client.on_sharing_event(lambda *_: None)
+
+    def test_on_media_connection_interrupted_canonical(self):
+        client = rtms.Client()
+        with patch.object(client, 'subscribeEvent'):
+            client.on_media_connection_interrupted(lambda *_: None)
+
+    def test_onAudioData_alias_works(self):
+        client = rtms.Client()
+        client.onAudioData(lambda *_: None)
+
+    def test_onVideoData_alias_works(self):
+        client = rtms.Client()
+        client.onVideoData(lambda *_: None)
+
+    def test_onLeave_alias_works(self):
+        client = rtms.Client()
+        client.onLeave(lambda _: None)
+
+    def test_camelCase_is_alias_for_snake_case(self):
+        """camelCase names should be Python-level aliases pointing to the same function as snake_case.
+
+        After fix: on_audio_data is defined in __init__.py and onAudioData = on_audio_data.
+        At the class level, they should be identical objects.
+        """
+        assert rtms.Client.onAudioData is rtms.Client.on_audio_data
+
+
+# Run tests
+if __name__ == '__main__':
+    pytest.main([__file__, '-v'])
diff --git a/tests/test_rtms.py b/tests/test_rtms.py
deleted file mode 100644
index a0612c2..0000000
--- a/tests/test_rtms.py
+++ /dev/null
@@ -1,339 +0,0 @@
-#!/usr/bin/env python3
-"""
-Comprehensive test suite for Python RTMS SDK
-
-Mirrors the Node.js test coverage in tests/rtms.test.ts
-"""
-
-import pytest
-import rtms
-from unittest.mock import Mock, patch, MagicMock
-import json
-
-
-class TestConstants:
-    """Test that all constants are properly defined"""
-
-    def test_media_type_constants(self):
-        """Test media type constants match expected values"""
-        assert rtms.MEDIA_TYPE_AUDIO == 1
-        assert rtms.MEDIA_TYPE_VIDEO == 2
-        assert rtms.MEDIA_TYPE_DESKSHARE == 4
-        assert rtms.MEDIA_TYPE_TRANSCRIPT == 8
-        assert rtms.MEDIA_TYPE_CHAT == 16
-        assert rtms.MEDIA_TYPE_ALL == 32
-
-    def test_session_event_constants(self):
-        """Test session event constants"""
-        assert rtms.SESSION_EVENT_ADD == 1
-        assert rtms.SESSION_EVENT_STOP == 2
-        assert rtms.SESSION_EVENT_PAUSE == 3
-        assert rtms.SESSION_EVENT_RESUME == 4
-
-    def test_event_type_constants(self):
-        """Test event type constants (for subscribeEvent/unsubscribeEvent)"""
-        assert rtms.EVENT_UNDEFINED == 0
-        assert rtms.EVENT_FIRST_PACKET_TIMESTAMP == 1
-        assert rtms.EVENT_ACTIVE_SPEAKER_CHANGE == 2
-        assert rtms.EVENT_PARTICIPANT_JOIN == 3
-        assert rtms.EVENT_PARTICIPANT_LEAVE == 4
-        assert rtms.EVENT_SHARING_START == 5
-        assert rtms.EVENT_SHARING_STOP == 6
-        assert rtms.EVENT_MEDIA_CONNECTION_INTERRUPTED == 7
-        assert rtms.EVENT_CONSUMER_ANSWERED == 8
-        assert rtms.EVENT_CONSUMER_END == 9
-        assert rtms.EVENT_USER_ANSWERED == 10
-        assert rtms.EVENT_USER_END == 11
-        assert rtms.EVENT_USER_HOLD == 12
-        assert rtms.EVENT_USER_UNHOLD == 13
-
-    def test_status_constants(self):
-        """Test SDK status constants"""
-        assert rtms.RTMS_SDK_FAILURE == -1
-        assert rtms.RTMS_SDK_OK == 0
-        assert rtms.RTMS_SDK_TIMEOUT == 1
-        assert rtms.RTMS_SDK_NOT_EXIST == 2
-
-    def test_audio_codec_constants(self):
-        """Test audio codec constants exist"""
-        assert 'OPUS' in rtms.AudioCodec
-        assert 'L16' in rtms.AudioCodec
-        assert 'G711' in rtms.AudioCodec
-
-    def test_video_codec_constants(self):
-        """Test video codec constants exist"""
-        assert 'H264' in rtms.VideoCodec
-        assert 'JPG' in rtms.VideoCodec
-
-    def test_sample_rate_constants(self):
-        """Test audio sample rate constants"""
-        assert 'SR_48K' in rtms.AudioSampleRate
-        assert 'SR_16K' in rtms.AudioSampleRate
-        assert 'SR_8K' in rtms.AudioSampleRate
-
-
-class TestLogging:
-    """Test logging functionality"""
-
-    def test_log_levels_enum(self):
-        """Test LogLevel enum values"""
-        assert rtms.LogLevel.ERROR == 0
-        assert rtms.LogLevel.WARN == 1
-        assert rtms.LogLevel.INFO == 2
-        assert rtms.LogLevel.DEBUG == 3
-        assert rtms.LogLevel.TRACE == 4
-
-    def test_log_format_enum(self):
-        """Test LogFormat enum values"""
-        assert rtms.LogFormat.PROGRESSIVE == 'progressive'
-        assert rtms.LogFormat.JSON == 'json'
-
-    def test_configure_logger(self):
-        """Test logger configuration"""
-        # Should not raise errors
-        rtms.configure_logger({
-            'level': 'debug',
-            'format': 'json',
-            'enabled': True
-        })
-
-        rtms.configure_logger({
-            'level': 'info',
-            'format': 'progressive',
-            'enabled': False
-        })
-
-    def test_log_functions_exist(self):
-        """Test that all log functions are available"""
-        assert callable(rtms.log_debug)
-        assert callable(rtms.log_info)
-        assert callable(rtms.log_warn)
-        assert callable(rtms.log_error)
-
-
-class TestClient:
-    """Test Client class functionality"""
-
-    def test_client_instantiation(self):
-        """Test that Client can be instantiated"""
-        client = rtms.Client()
-        assert client is not None
-        assert isinstance(client, rtms.Client)
-
-    def test_client_has_join_method(self):
-        """Test that Client has join method"""
-        client = rtms.Client()
-        assert hasattr(client, 'join')
-        assert callable(client.join)
-
-    def test_client_has_leave_method(self):
-        """Test that Client has leave method"""
-        client = rtms.Client()
-        assert hasattr(client, 'leave')
-        assert callable(client.leave)
-
-    def test_client_has_callback_methods(self):
-        """Test that Client has all callback registration methods"""
-        client = rtms.Client()
-        assert hasattr(client, 'onJoinConfirm')
-        assert hasattr(client, 'onSessionUpdate')
-        assert hasattr(client, 'onParticipantEvent')
-        assert hasattr(client, 'onActiveSpeakerEvent')
-        assert hasattr(client, 'onSharingEvent')
-        assert hasattr(client, 'onEventEx')
-        assert hasattr(client, 'onAudioData')
-        assert hasattr(client, 'onVideoData')
-        assert hasattr(client, 'onDeskshareData')
-        assert hasattr(client, 'onTranscriptData')
-        assert hasattr(client, 'onLeave')
-
-    def test_client_has_param_methods(self):
-        """Test that Client has parameter configuration methods"""
-        client = rtms.Client()
-        assert hasattr(client, 'setAudioParams')
-        assert hasattr(client, 'setVideoParams')
-        assert hasattr(client, 'setDeskshareParams')
-
-    def test_client_has_event_subscription_methods(self):
-        """Test that Client has event subscription methods"""
-        client = rtms.Client()
-        assert hasattr(client, 'subscribeEvent')
-        assert callable(client.subscribeEvent)
-        assert hasattr(client, 'unsubscribeEvent')
-        assert callable(client.unsubscribeEvent)
-
-    def test_client_callback_registration(self):
-        """Test callback registration doesn't raise errors"""
-        client = rtms.Client()
-
-        @client.onJoinConfirm
-        def on_join(reason):
-            pass
-
-        @client.onTranscriptData
-        def on_transcript(data, size, timestamp, metadata):
-            pass
-
-        # Should complete without errors
-        assert True
-
-
-class TestParameterValidation:
-    """Test parameter validation functionality"""
-
-    def test_audio_params_exist(self):
-        """Test AudioParams class exists"""
-        assert hasattr(rtms, 'AudioParams')
-
-    def test_video_params_exist(self):
-        """Test VideoParams class exists"""
-        assert hasattr(rtms, 'VideoParams')
-
-    def test_deskshare_params_exist(self):
-        """Test DeskshareParams class exists"""
-        assert hasattr(rtms, 'DeskshareParams')
-
-
-class TestUtilityFunctions:
-    """Test utility functions"""
-
-    def test_generate_signature_exists(self):
-        """Test signature generation function exists"""
-        assert hasattr(rtms, 'generate_signature')
-        assert callable(rtms.generate_signature)
-
-    def test_find_ca_certificate_exists(self):
-        """Test CA certificate finder exists"""
-        assert hasattr(rtms, 'find_ca_certificate')
-        assert callable(rtms.find_ca_certificate)
-
-    def test_initialize_exists(self):
-        """Test initialize function exists"""
-        assert hasattr(rtms, 'initialize')
-        assert callable(rtms.initialize)
-
-    def test_uninitialize_exists(self):
-        """Test uninitialize function exists"""
-        assert hasattr(rtms, 'uninitialize')
-        assert callable(rtms.uninitialize)
-
-
-class TestWebhookFunctionality:
-    """Test webhook-related functionality"""
-
-    def test_on_webhook_event_exists(self):
-        """Test webhook event decorator exists"""
-        assert hasattr(rtms, 'onWebhookEvent')
-        assert callable(rtms.onWebhookEvent)
-        # Also test the alias
-        assert hasattr(rtms, 'on_webhook_event')
-        assert callable(rtms.on_webhook_event)
-
-    def test_webhook_response_class_exists(self):
-        """Test WebhookResponse class exists"""
-        # WebhookResponse is internal but should be importable
-        from rtms import WebhookResponse
-        assert WebhookResponse is not None
-
-    def test_client_webhook_decorator(self):
-        """Test client webhook event decorator"""
-        client = rtms.Client()
-        assert hasattr(client, 'on_webhook_event')
-        assert callable(client.on_webhook_event)
-
-        # Test decorator usage
-        @client.on_webhook_event()
-        def handle_webhook(payload):
-            pass
-
-        assert True
-
-
-class TestDataClasses:
-    """Test data classes and structures"""
-
-    def test_session_class_exists(self):
-        """Test Session class exists"""
-        assert hasattr(rtms, 'Session')
-
-    def test_participant_class_exists(self):
-        """Test Participant class exists"""
-        assert hasattr(rtms, 'Participant')
-
-    def test_metadata_class_exists(self):
-        """Test Metadata class exists"""
-        assert hasattr(rtms, 'Metadata')
-
-
-class TestModuleExports:
-    """Test that all expected exports are available"""
-
-    def test_client_export(self):
-        """Test Client is exported"""
-        assert 'Client' in rtms.__all__
-
-    def test_constant_exports(self):
-        """Test constants are exported"""
-        exports_to_check = [
-            'MEDIA_TYPE_AUDIO',
-            'MEDIA_TYPE_VIDEO',
-            'MEDIA_TYPE_TRANSCRIPT',
-            'SESSION_EVENT_ADD',
-            'EVENT_PARTICIPANT_JOIN',
-            'EVENT_PARTICIPANT_LEAVE',
-            'EVENT_ACTIVE_SPEAKER_CHANGE',
-            'EVENT_SHARING_START',
-            'EVENT_SHARING_STOP',
-            'RTMS_SDK_OK'
-        ]
-        for export in exports_to_check:
-            assert export in rtms.__all__
-
-    def test_function_exports(self):
-        """Test functions are exported"""
-        functions = [
-            'initialize',
-            'uninitialize',
-            'generate_signature',
-            'configure_logger'
-        ]
-        for func in functions:
-            assert func in rtms.__all__
-
-    def test_enum_exports(self):
-        """Test enums are exported"""
-        enums = [
-            'AudioCodec',
-            'VideoCodec',
-            'AudioSampleRate',
-            'VideoResolution',
-            'LogLevel',
-            'LogFormat'
-        ]
-        for enum in enums:
-            assert enum in rtms.__all__
-
-
-class TestThreadSafety:
-    """Test thread-safe features"""
-
-    def test_module_has_run_function(self):
-        """Test module has run() function for event loop"""
-        assert hasattr(rtms, 'run')
-        assert callable(rtms.run)
-
-    def test_module_has_stop_function(self):
-        """Test module has stop() function to signal shutdown"""
-        assert hasattr(rtms, 'stop')
-        assert callable(rtms.stop)
-
-    def test_client_has_polling_control(self):
-        """Test client has polling control"""
-        client = rtms.Client()
-        assert hasattr(client, '_poll_if_needed')
-        assert hasattr(client, '_running')
-
-
-# Run tests
-if __name__ == '__main__':
-    pytest.main([__file__, '-v'])
diff --git a/tests/pack-install.test.js b/tests/ts/pack-install.test.js
similarity index 94%
rename from tests/pack-install.test.js
rename to tests/ts/pack-install.test.js
index 3981820..4cc07f2 100644
--- a/tests/pack-install.test.js
+++ b/tests/ts/pack-install.test.js
@@ -168,8 +168,6 @@ describe('npm pack → install → load integration', () => {
       throw new Error(`Native module not found at ${buildDir}/rtms.node`);
     }
 
-    console.log(`Using build artifacts from: ${buildDir}`);
-
     // Create tarball from current package
     const output = execSync('npm pack --json', { encoding: 'utf8' });
     const packages = JSON.parse(output);
@@ -277,4 +275,21 @@ describe('npm pack → install → load integration', () => {
 
     expect(result.trim()).toBe('OK');
   });
+
+  test('Client.setProxy is a function', () => {
+    installWithLocalBuild(tempDir, tarballPath, buildDir);
+
+    const testScript = `
+      import rtms from '@zoom/rtms';
+      const client = new rtms.Client();
+      console.log(typeof client.setProxy === 'function' ? 'OK' : 'FAIL');
+    `;
+
+    const result = execSync(`node -e "${testScript}"`, {
+      cwd: tempDir,
+      encoding: 'utf8'
+    });
+
+    expect(result.trim()).toBe('OK');
+  });
 });
diff --git a/tests/rtms.test.ts b/tests/ts/rtms.test.ts
similarity index 82%
rename from tests/rtms.test.ts
rename to tests/ts/rtms.test.ts
index d3a997b..8c6c3d3 100644
--- a/tests/rtms.test.ts
+++ b/tests/ts/rtms.test.ts
@@ -1,7 +1,7 @@
-const rtms = require("../index.ts")
+const rtms = require("../../index.ts")
 
 // Mock the native addon methods for both Client and global functions
-jest.mock("../index.ts", () => {
+jest.mock("../../index.ts", () => {
   // Create mock functions for all methods
   const mockFunctions = {
     // Client class methods
@@ -26,6 +26,7 @@ jest.mock("../index.ts", () => {
       setAudioParams: jest.fn().mockReturnValue(true),
       setVideoParams: jest.fn().mockReturnValue(true),
       setDeskshareParams: jest.fn().mockReturnValue(true),
+      setTranscriptParams: jest.fn().mockReturnValue(true),
       
       // Callback methods
       onJoinConfirm: jest.fn().mockReturnValue(true),
@@ -43,8 +44,33 @@ jest.mock("../index.ts", () => {
       // Event subscription methods
       subscribeEvent: jest.fn().mockReturnValue(true),
       unsubscribeEvent: jest.fn().mockReturnValue(true),
+
+      // Individual video subscription methods
+      subscribeVideo: jest.fn().mockReturnValue(true),
+      onParticipantVideo: jest.fn().mockReturnValue(true),
+      onVideoSubscribed: jest.fn().mockReturnValue(true),
+    })),
+
+    // TranscriptParams class
+    TranscriptParams: jest.fn().mockImplementation(() => ({
+      contentType: 5,
+      srcLanguage: -1,
+      enableLid: true,
+      setSrcLanguage: jest.fn(),
+      setEnableLid: jest.fn(),
     })),
 
+    // Transcript language constants (mirrors AudioCodec pattern)
+    TranscriptLanguage: {
+      NONE: -1,
+      ARABIC: 0,
+      ENGLISH: 9,
+      FRENCH_FRANCE: 13,
+      GERMAN: 14,
+      JAPANESE: 20,
+      SPANISH: 28,
+    },
+
     // Utility functions
     initialize: jest.fn().mockReturnValue(true),
     uninitialize: jest.fn().mockReturnValue(true),
@@ -197,7 +223,7 @@ describe('RTMS Node.JS Addon Comprehensive Test Suite', () => {
 
   // ---- Class-based approach tests ----
   describe('Class-based Client Approach', () => {
-    let client;
+    let client: InstanceType<typeof rtms.Client>;
 
     beforeEach(() => {
       client = new rtms.Client();
@@ -219,7 +245,24 @@ describe('RTMS Node.JS Addon Comprehensive Test Suite', () => {
           timeout: 5000,
           pollInterval: 100
         };
-        
+
+        const result = client.join(joinParams);
+        expect(client.join).toHaveBeenCalledWith(joinParams);
+        expect(result).toBe(true);
+      });
+
+      // ZCC engagement_id tests
+      // NOTE: The test module is fully mocked so client.join always returns true
+      // regardless of params. These tests document the expected API contract.
+      // The real routing logic is validated in the Python tests and by TypeScript
+      // compilation (engagement_id must appear in the JoinParams interface).
+      test('client.join accepts engagement_id for ZCC sessions', () => {
+        const joinParams = {
+          engagement_id: "engagement-abc-123",
+          rtms_stream_id: "stream-xyz",
+          server_urls: "wss://rtms.zoom.us",
+          signature: "mock-sig",
+        };
         const result = client.join(joinParams);
         expect(client.join).toHaveBeenCalledWith(joinParams);
         expect(result).toBe(true);
@@ -319,6 +362,24 @@ describe('RTMS Node.JS Addon Comprehensive Test Suite', () => {
         expect(client.setDeskshareParams).toHaveBeenCalledWith(deskshareParams);
         expect(result).toBe(true);
       });
+
+      // TranscriptParams tests
+      // NOTE: The module is fully mocked; these document the expected API shape.
+      // The real default values and toNative() mapping are verified in the C++ tests.
+      test('client.setTranscriptParams accepts a TranscriptParams object', () => {
+        const params = new rtms.TranscriptParams();
+        const result = client.setTranscriptParams(params);
+        expect(client.setTranscriptParams).toHaveBeenCalledWith(params);
+        expect(result).toBe(true);
+      });
+
+      test('TranscriptLanguage.ENGLISH equals 9', () => {
+        expect(rtms.TranscriptLanguage.ENGLISH).toBe(9);
+      });
+
+      test('TranscriptLanguage.NONE equals -1', () => {
+        expect(rtms.TranscriptLanguage.NONE).toBe(-1);
+      });
     });
 
     describe('Callback Methods', () => {
@@ -400,6 +461,64 @@ describe('RTMS Node.JS Addon Comprehensive Test Suite', () => {
       });
     });
 
+    describe('Individual Video Subscription', () => {
+      test('client.subscribeVideo subscribes to an individual participant video stream', () => {
+        const result = client.subscribeVideo(12345, true);
+        expect(client.subscribeVideo).toHaveBeenCalledWith(12345, true);
+        expect(result).toBe(true);
+      });
+
+      test('client.subscribeVideo unsubscribes from an individual participant video stream', () => {
+        const result = client.subscribeVideo(12345, false);
+        expect(client.subscribeVideo).toHaveBeenCalledWith(12345, false);
+        expect(result).toBe(true);
+      });
+
+      test('client.onParticipantVideo sets the participant video state callback', () => {
+        const callback = jest.fn();
+        const result = client.onParticipantVideo(callback);
+        expect(client.onParticipantVideo).toHaveBeenCalledWith(callback);
+        expect(result).toBe(true);
+      });
+
+      test('client.onVideoSubscribed sets the video subscription response callback', () => {
+        const callback = jest.fn();
+        const result = client.onVideoSubscribed(callback);
+        expect(client.onVideoSubscribed).toHaveBeenCalledWith(callback);
+        expect(result).toBe(true);
+      });
+
+      test('onParticipantVideo callback receives users array and isOn flag', () => {
+        let capturedUsers: number[] = [];
+        let capturedIsOn: boolean | null = null;
+        const callback = jest.fn().mockImplementation((users: number[], isOn: boolean) => {
+          capturedUsers = users;
+          capturedIsOn = isOn;
+        });
+        client.onParticipantVideo(callback);
+        // Simulate a callback invocation
+        callback([11111, 22222], true);
+        expect(capturedUsers).toEqual([11111, 22222]);
+        expect(capturedIsOn).toBe(true);
+      });
+
+      test('onVideoSubscribed callback receives userId, status, and error string', () => {
+        let capturedUserId: number | null = null;
+        let capturedStatus: number | null = null;
+        let capturedError: string | null = null;
+        const callback = jest.fn().mockImplementation((userId: number, status: number, error: string) => {
+          capturedUserId = userId;
+          capturedStatus = status;
+          capturedError = error;
+        });
+        client.onVideoSubscribed(callback);
+        callback(12345, 0, '');
+        expect(capturedUserId).toBe(12345);
+        expect(capturedStatus).toBe(0);
+        expect(capturedError).toBe('');
+      });
+    });
+
     describe('Event Subscription Methods', () => {
       test('client.subscribeEvent subscribes to events correctly', () => {
         const events = [rtms.EVENT_PARTICIPANT_JOIN, rtms.EVENT_PARTICIPANT_LEAVE];
diff --git a/tests/ts/rtms.wrapper.test.ts b/tests/ts/rtms.wrapper.test.ts
new file mode 100644
index 0000000..4a190b2
--- /dev/null
+++ b/tests/ts/rtms.wrapper.test.ts
@@ -0,0 +1,206 @@
+/**
+ * Integration tests for the index.ts TypeScript wrapper layer.
+ *
+ * Unlike rtms.test.ts (which mocks the entire module), these tests run against
+ * the real built ESM output (build/Release/index.js) via child node processes.
+ * This provides a genuine red-green TDD cycle: methods missing from index.ts
+ * cause failures here even if the mock in rtms.test.ts is already wired.
+ *
+ * Requires build/Release/index.js to exist (task build:js).
+ */
+
+import { execSync } from 'child_process';
+import { existsSync } from 'fs';
+import * as path from 'path';
+
+const BUILD = path.resolve(__dirname, '../../build/Release/index.js');
+
+// The built module uses a default export. Prefix every eval with this preamble.
+const PREAMBLE = `import * as _mod from '${BUILD}'; const rtms = _mod.default;`;
+
+/** Run an expression with a Client instance; returns true if exit code 0. */
+function run(expr: string): boolean {
+  try {
+    execSync(
+      `node --input-type=module --eval "${PREAMBLE} const c = new rtms.Client(); process.exit((${expr}) ? 0 : 1);"`,
+      { stdio: 'pipe' }
+    );
+    return true;
+  } catch {
+    return false;
+  }
+}
+
+/** Run an expression at module level (no Client). */
+function runModule(expr: string): boolean {
+  try {
+    execSync(
+      `node --input-type=module --eval "${PREAMBLE} process.exit((${expr}) ? 0 : 1);"`,
+      { stdio: 'pipe' }
+    );
+    return true;
+  } catch {
+    return false;
+  }
+}
+
+describe('index.ts wrapper — real built module', () => {
+  beforeAll(() => {
+    if (!existsSync(BUILD)) {
+      throw new Error(`Build output not found: ${BUILD}\nRun 'task build:js' first.`);
+    }
+  });
+
+  // --------------------------------------------------------------------------
+  describe('Client — connection methods', () => {
+    test('join is a function', () => {
+      expect(run("typeof c.join === 'function'")).toBe(true);
+    });
+
+    test('leave is a function', () => {
+      expect(run("typeof c.leave === 'function'")).toBe(true);
+    });
+  });
+
+  // --------------------------------------------------------------------------
+  describe('Client — parameter methods', () => {
+    test('setAudioParams is a function', () => {
+      expect(run("typeof c.setAudioParams === 'function'")).toBe(true);
+    });
+
+    test('setVideoParams is a function', () => {
+      expect(run("typeof c.setVideoParams === 'function'")).toBe(true);
+    });
+
+    test('setDeskshareParams is a function', () => {
+      expect(run("typeof c.setDeskshareParams === 'function'")).toBe(true);
+    });
+
+    test('setTranscriptParams is a function', () => {
+      expect(run("typeof c.setTranscriptParams === 'function'")).toBe(true);
+    });
+
+    test('setProxy is a function', () => {
+      expect(run("typeof c.setProxy === 'function'")).toBe(true);
+    });
+
+    test('setProxy does not throw for http', () => {
+      expect(run("(c.setProxy('http', 'http://proxy.example.com:8080'), true)")).toBe(true);
+    });
+
+    test('setProxy does not throw for https', () => {
+      expect(run("(c.setProxy('https', 'https://proxy.example.com:8080'), true)")).toBe(true);
+    });
+  });
+
+  // --------------------------------------------------------------------------
+  describe('Client — callback registration methods', () => {
+    const callbacks = [
+      'onJoinConfirm', 'onSessionUpdate', 'onUserUpdate',
+      'onParticipantEvent', 'onActiveSpeakerEvent', 'onSharingEvent', 'onEventEx',
+      'onAudioData', 'onVideoData', 'onDeskshareData', 'onTranscriptData', 'onLeave',
+    ];
+
+    for (const method of callbacks) {
+      test(`${method} is a function`, () => {
+        expect(run(`typeof c.${method} === 'function'`)).toBe(true);
+      });
+    }
+  });
+
+  // --------------------------------------------------------------------------
+  describe('Client — event subscription methods', () => {
+    test('subscribeEvent is a function', () => {
+      expect(run("typeof c.subscribeEvent === 'function'")).toBe(true);
+    });
+
+    test('unsubscribeEvent is a function', () => {
+      expect(run("typeof c.unsubscribeEvent === 'function'")).toBe(true);
+    });
+  });
+
+  // --------------------------------------------------------------------------
+  describe('Client — individual video subscription', () => {
+    test('subscribeVideo is a function', () => {
+      expect(run("typeof c.subscribeVideo === 'function'")).toBe(true);
+    });
+
+    test('onParticipantVideo is a function', () => {
+      expect(run("typeof c.onParticipantVideo === 'function'")).toBe(true);
+    });
+
+    test('onVideoSubscribed is a function', () => {
+      expect(run("typeof c.onVideoSubscribed === 'function'")).toBe(true);
+    });
+
+    test('onParticipantVideo accepts a callback without throwing', () => {
+      expect(run("(c.onParticipantVideo(() => {}), true)")).toBe(true);
+    });
+
+    test('onVideoSubscribed accepts a callback without throwing', () => {
+      expect(run("(c.onVideoSubscribed(() => {}), true)")).toBe(true);
+    });
+  });
+
+  // --------------------------------------------------------------------------
+  describe('Module — utility functions', () => {
+    test('generateSignature is a function', () => {
+      expect(runModule("typeof rtms.generateSignature === 'function'")).toBe(true);
+    });
+
+    test('configureLogger is a function', () => {
+      expect(runModule("typeof rtms.configureLogger === 'function'")).toBe(true);
+    });
+
+    test('onWebhookEvent is a function', () => {
+      expect(runModule("typeof rtms.onWebhookEvent === 'function'")).toBe(true);
+    });
+
+    test('Client.initialize is a static function', () => {
+      expect(runModule("typeof rtms.Client.initialize === 'function'")).toBe(true);
+    });
+
+    test('Client.uninitialize is a static function', () => {
+      expect(runModule("typeof rtms.Client.uninitialize === 'function'")).toBe(true);
+    });
+  });
+
+  // --------------------------------------------------------------------------
+  describe('Module — constants', () => {
+    test('MEDIA_TYPE_AUDIO === 1', () => {
+      expect(runModule('rtms.MEDIA_TYPE_AUDIO === 1')).toBe(true);
+    });
+
+    test('MEDIA_TYPE_VIDEO === 2', () => {
+      expect(runModule('rtms.MEDIA_TYPE_VIDEO === 2')).toBe(true);
+    });
+
+    test('MEDIA_TYPE_TRANSCRIPT === 8', () => {
+      expect(runModule('rtms.MEDIA_TYPE_TRANSCRIPT === 8')).toBe(true);
+    });
+
+    test('RTMS_SDK_OK === 0', () => {
+      expect(runModule('rtms.RTMS_SDK_OK === 0')).toBe(true);
+    });
+
+    test('SESSION_EVENT_ADD === 1', () => {
+      expect(runModule('rtms.SESSION_EVENT_ADD === 1')).toBe(true);
+    });
+
+    test('TranscriptLanguage.ENGLISH === 9', () => {
+      expect(runModule('rtms.TranscriptLanguage && rtms.TranscriptLanguage.ENGLISH === 9')).toBe(true);
+    });
+
+    test('TranscriptLanguage.NONE === -1', () => {
+      expect(runModule('rtms.TranscriptLanguage && rtms.TranscriptLanguage.NONE === -1')).toBe(true);
+    });
+
+    test('AudioCodec dict is exported', () => {
+      expect(runModule("typeof rtms.AudioCodec === 'object' && rtms.AudioCodec !== null")).toBe(true);
+    });
+
+    test('VideoCodec dict is exported', () => {
+      expect(runModule("typeof rtms.VideoCodec === 'object' && rtms.VideoCodec !== null")).toBe(true);
+    });
+  });
+});