.default})
-
-The visible video is selected based on this priority:
-
-- Participant is pinned
-- Participant is screen-sharing
-- Participant is the dominant speaker
-- Participant has a video track
-
-### Grid
-
-This layout shows a configurable number of tracks in an equally sized grid.
-
-.default})
-
-### Spotlight
-
-This layout shows a video in a spotlight and the rest of the participants in a separate list or grid.
-
-.default})
-
-## Layout options
-
-Each layout has a number of options that you can configure. Here is an example:
-
-.default})
-
-.default})
-
-While this can be configured from the dashboard, you can also set it for individual calls:
-
-.default})
-
-#### Optimal OBS Video Encoding Settings
-
-To ensure optimal quality and performance when transcoding media from RTMP, follow these recommended H.264 encoding settings:
-
-Go to `Settings`:
-
-**Video**:
-- Base Resolution: 1920x1080
-- Output Resolution: 1920x1080 (Same as input resolution)
-
-**Output**:
-- Encoder: `x264`
-- Rate Control: `CBR` (Constant Bit Rate)
-- Bitrate: 3,000-6,000 kbps (adjust based on your network and type of media being streamed, higher movement will require higher bitrate)
-- Keyframe Interval: 2 seconds
-- CPU Usage Preset: `veryfast` (balance between quality and encoding speed)
-- Tune: `zerolatency`
-
-.default})
-
-_The example is for OBS, but any other RTMP client should support adding the same settings_
-
-### Streaming into a call with restream.io
-
-1. Go to [restream.io](https://restream.io/) and create an account.
-2. For streaming to a custom RTMP server, you need to have a paid account.
-3. Add a new channel and select `Custom RTMP`.
-4. Set the `Server` to the `rtmp_url` you got from the API.
-5. Set the `Stream Key` to the `stream_key` you got from the API.
-
-.default})
-
-
-### Streaming a file into a call with FFmpeg
-
-With your call's RTMP url `$CALL_RTMP_URL` and your token streamKey `$STREAM_KEY` from [publishing](#rtmp-publishing).
-Run:
-
-```bash
-ffmpeg -re -stream_loop 400 -i YourVideoFile1080p30fps.mp4 \
- -c:v libx264 -preset veryfast -tune zerolatency \
- -pix_fmt yuv420p -g 50 -c:a aac -b:a 160k \
- -ac 2 -f flv "$CALL_RTMP_URL/$STREAM_KEY"
-```
-
-### FAQ
-
-#### What is the typical latency introduced by RTMP?
-
-RTMP input typically introduces a latency of 2-5 seconds. This can vary based on the network conditions, encoder settings (as the stream will be transcoded), and the performance of the RTMP server.
-Optimizing [encoding settings](#optimal-obs-video-encoding-settings) and ensuring a stable network connection can help minimize this latency.
-
-#### What are the best practices for setting up OBS for RTMP streaming?
-
-Refer to [Optimal OBS Video Encoding Settings](#optimal-obs-video-encoding-settings).
-
-#### Do we support both RTMP and RTMPS?
-
-RTMP input service supports exclusively RTMPS. RTMPS is the recommended for secure streaming as it uses SSL/TLS to encrypt the data, providing an additional layer of security over RTMP.
-
-#### What are the recommended internet connection settings for optimal streaming performance?
-
-- **Upload Speed**: At least double the target bitrate of your stream. For example, if streaming at 5,000 kbps, ensure an upload speed of at least 10 Mbps.
-- **Connection Type**: Wired Ethernet connection is preferred over Wi-Fi for stability.
-- **Network Quality**: Ensure low packet loss (<1%) and low jitter (<30ms).
diff --git a/docusaurus/video/docusaurus/docs/api/streaming/rtmp_broadcast.mdx b/docusaurus/video/docusaurus/docs/api/streaming/rtmp_broadcast.mdx
deleted file mode 100644
index ccb4c8964..000000000
--- a/docusaurus/video/docusaurus/docs/api/streaming/rtmp_broadcast.mdx
+++ /dev/null
@@ -1,174 +0,0 @@
----
-id: streaming_rtmp_out
-sidebar_position: 2
-slug: /streaming/rtmp_broadcasts
-title: RTMP broadcasts
----
-
-import Tabs from '@theme/Tabs';
-import TabItem from '@theme/TabItem';
-import RtmpBroadcastStart from '../_common_/rtmp-broadcast-start.mdx';
-import CallTypeRtmpSettings from '../_common_/call-type-rtmp-settings.mdx';
-import RtmpBroadcastStop from '../_common_/rtmp-broadcast-stop.mdx';
-import RtmpBroadcastStopAll from '../_common_/rtmp-broadcast-stop-all.mdx';
-import OpenApiModels from '../_common_/OpenApiModels';
-
-
-**RTMP broadcasting** allows you to forward a call to any external streaming service that supports RTMP(s) as input. All major platforms like Twitch, Youtube, Facebook, … support it.
-Multiple broadcasts can be added for the same call, allowing it to send a live call to multiple providers at the same time.
-
-## Understanding RTMP broadcasting
-[RTMP](https://en.wikipedia.org/wiki/Real-Time_Messaging_Protocol) is a protocol initially developed by Adobe for streaming audio, video, and data over
-the internet. Although RTMP is no longer the dominant streaming protocol, it's still widely used for live streaming because of its low latency and
-compatibility with platforms.
-
-**RTMP broadcasting** refers to the process of sending a live video stream from a source (like Stream server) to an external streaming platform (like Youtube).
-
-#### How It Works
-- Stream API gets the media streams from your call and [composes them into a single stream.](#configuring-the-layout-for-your-rtmp-broadcast)
-- Stream API encodes and packages the stream into RTMP with settings suitable for live streaming.
-- The target platform takes this RTMP stream, and processes it. Typically, services like Youtube
-re-encode it (if necessary) and broadcast the stream to viewers in various formats and resolutions,
-using protocols like HLS or DASH.
-
-## Pre-requisites
-
-### Getting a stream url and key
-
-Platforms that support RTMP ingestion provide a stream url (the server where the stream will be sent) and a stream key (unique identifier for your stream).
-
-### Configuring the layout for your rtmp broadcast
-
-As previously mentioned, the call will be composed into a single stream following a layout:
-- You can choose from one of our [pre-defined layouts](./../../recording/calls/#recording-layouts).
-- Theming/branding is possible by changing layout parameters (colors, logos, positioning or elements, …)
-- For more advanced / complex theming customization, you can build your own webapp and use it as the layout of the call.
-
-.default})
-
-On that page you'll find your **Stream key** and the suggested **Stream URL**. [Copy both and add them to the start request.](#starting-an-rtmp-broadcast)
-
-On YouTube side, check the status of the stream from the Livestreaming Dashboard, you should see that your call is live!
-
-To stop it then at any given point in time, use the [stop by name](#stopping-an-rtmp-broadcast-by-name) or
-[stop all](#stopping-all-rtmp-broadcasts-for-a-call) endpoints.
-
-> **Note:** YouTube also provides a way to stop it from their end, although it will stop the broadcast it
-won't stop the RTMP connection from Stream to Youtube, so you should call always stop, or end the call.
-
-
-## FAQ
-
-### How much latency does RTMP broadcast add?
-
-- RTMP itself doesn't add much latency, as it is a 1-1 protocol. Another thing then is how is the livestream content
-processed, for example, in the case of platforms like Youtube the latency is increased when the content is
-broadcasted through HLS/DASH protocol. [Check Youtube's options](https://support.google.com/youtube/answer/7444635?sjid=10230171077221482676-EU)
diff --git a/docusaurus/video/docusaurus/docs/api/streaming/webrtc.mdx b/docusaurus/video/docusaurus/docs/api/streaming/webrtc.mdx
deleted file mode 100644
index 0159f887c..000000000
--- a/docusaurus/video/docusaurus/docs/api/streaming/webrtc.mdx
+++ /dev/null
@@ -1,16 +0,0 @@
----
-id: webrtc
-sidebar_position: 4
-slug: /streaming/webrtc
-title: WebRTC
----
-
-[WebRTC](https://webrtc.org/) allows you to have low-latency streaming (hundreds of milliseconds).
-
-## Start and stop WebRTC broadcast
-
-The WebRTC broadcast is always on. Viewers can access it by joining the call. You can control who can join the call with [backstage mode](../../streaming/backstage).
-
-## Consuming WebRTC broadcast
-
-Your viewers must join the call to access the stream. You can use regular, guest or anonymous users to join the livestream. It's also possible to create call tokens that only allow access for specific calls. For more information, see the [authentication page](../../authentication).
diff --git a/docusaurus/video/docusaurus/docs/api/transcription/_category_.json b/docusaurus/video/docusaurus/docs/api/transcription/_category_.json
deleted file mode 100644
index 43bbe3cd6..000000000
--- a/docusaurus/video/docusaurus/docs/api/transcription/_category_.json
+++ /dev/null
@@ -1,4 +0,0 @@
-{
- "label": "Transcription",
- "position": 5
-}
\ No newline at end of file
diff --git a/docusaurus/video/docusaurus/docs/api/transcription/storage.mdx b/docusaurus/video/docusaurus/docs/api/transcription/storage.mdx
deleted file mode 100644
index aeabbf21f..000000000
--- a/docusaurus/video/docusaurus/docs/api/transcription/storage.mdx
+++ /dev/null
@@ -1,14 +0,0 @@
----
-id: storage
-sidebar_position: 2
-slug: /transcribing/storage
-title: Storage
----
-
-import Storage from '../_common_/storage.mdx';
-
-By default, transcriptions are stored in an AWS S3 bucket managed by Stream, located in the same region as your application.
-Transcription files are retained for two weeks before being automatically deleted.
-If you need to keep transcriptions longer or prefer not to store this data with Stream, you can opt to use your own storage solution.
-
-