@tencent-rtc/mcp 1.6.7 → 1.7.0

package/resource/knowledge/roomkit/preset/react/integration.md

@@ -60,7 +60,6 @@ export default function App() {
 Authentication is required to access all TUIRoomKit features. Call **conference.login** immediately after your business system completes authentication (when the user logs in to your web page). After calling `conference.login`, call `conference.setSelfInfo` to set the user's display name and avatar, which appear in the participant video area and member list.
 
 > **Note:**
->
 > Make sure to use the generated UserSig in your configuration. In production, generate UserSig on your server and have the client request a dynamic UserSig from your business server as needed. See [How to Calculate UserSig in Production](https://trtc.io/document/35166).
 >
 
@@ -305,7 +304,6 @@ const enterDualMeeting = async (bizOrderId: string) => {
 ```
 
 > **Tip:**
->
 > The `conference.createAndJoinRoom` API automatically handles "create if not exists (become Host), join if exists (become Participant)", resolving state conflicts under high concurrency.
 >
 
@@ -325,7 +323,6 @@ await conference.leaveRoom();
 ```
 
 > **Note:**
->
 > When the Host leaves or closes the webpage, the room is not automatically ended. Other Participants can continue using TUIRoomKit. The system will reclaim room resources 6 hours after the scheduled end time if there are zero Participants.
 >
 
@@ -339,7 +336,6 @@ await conference.endRoom();
 ```
 
 > **Note:**
->
 > Only the Host can call `endRoom()` after successfully entering the room. Calling as a non-host or before entering will result in an error.
 >
 
@@ -404,8 +400,6 @@ pnpm run dev
 yarn run dev
 ```
 
-After the project starts, open your browser to the debug address (usually http://localhost:5173) to access the app. You should see the meeting interface as shown below:
-
 ## Next Steps
 
 After completing the basic integration, you can further customize the UI to fit your business requirements.
@@ -498,17 +492,13 @@ No. TUIRoomKit does not allow the same userId to join the same room on multiple
 
 ### Is iframe integration supported?
 
-Yes. When integrating TUIRoomKit via iframe, set the allow attribute to grant necessary browser permissions (microphone, camera, screen sharing, fullscreen, etc.) as shown below:
 ``` typescript
 // Enable microphone, camera, screen sharing, fullscreen permissions
 <iframe src="https://your-domain.com/index.html" allow="microphone; camera; display-capture; display; fullscreen;">
 ```
 
-For more information, see Embedded Pages (iframe).
-
 ### Is intranet proxy supported?
 
-Yes. Set intranet proxy parameters as shown below. For more details, see [Dealing with Firewall Restrictions](https://trtc.io/document/59667).
 ``` typescript
 // Set intranet proxy parameters before entering the room
 import TUIRoomEngine from '@tencentcloud/tuiroom-engine-js';
@@ -540,8 +530,6 @@ TUIRoomKit offers rich customization out of the box—custom themes and language
 We strongly discourage exporting the source code. Doing so removes your project from TUIRoomKit's standard npm upgrade path, increases maintenance costs, and prevents you from benefiting from new features and performance improvements in the underlying audio/video engine.
 - Recommended: Explore standard customization options in the UI customization documentation.
 
-- Feedback: If the available APIs do not meet your needs, contact us via [Contact Us](https://trtc.io/contact) to submit your requirements. We will evaluate and strive to provide stable, standardized API support.
-
    If you have fully considered the maintenance risks and require deep customization, you may export the source code as follows:
 
 【React】
@@ -566,7 +554,3 @@ import { ConferenceMainView, conference } from './components/RoomKit/index.ts';
 // Replace with the actual TUIRoomKit source path
 src/components/RoomKit
 ```
-
-## Contact Us
-
-If you have any questions or suggestions during integration or usage, you are welcome to join our [Telegram technical group](https://t.me/+EPk6TMZEZMM5OGY1) or [contact us](https://trtc.io/contact) for support.

package/resource/knowledge/rtcengine/web/api.md

@@ -12,8 +12,6 @@ The TRTC object is created using [TRTC.create()](#create) and provides core real
 - Stop watching remote video [stopRemoteVideo()](#stopremotevideo)
 - Mute/unmute remote audio [muteRemoteAudio()](#muteremoteaudio)
 
-The TRTC lifecycle is shown in the following figure:
-
 ## Methods
 
 ### create()
@@ -110,7 +108,6 @@ await trtc.exitRoom();
 Switch a video call room.
 
 - Switch the video call room that the audience is watching in live scene. It is faster than exit old room and then entering the new room, and can optimize the opening time in live broadcast and other scenarios.
-- [Contact us](https://trtc.io/contact) to enable this API.
 
 #### Example
 
@@ -1222,7 +1219,6 @@ TRTC.setLogLevel(1);
 | Name | Type | Default | Description |
 |------|------|---------|-------------|
 | level | 0-5 | | Log output level 0: TRACE 1: DEBUG 2: INFO 3: WARN 4: ERROR 5: NONE |
-| enableUploadLog | boolean | true | Whether to enable log upload, which is enabled by default. It is not recommended to turn it off, which will affect problem troubleshooting.<br>- Logs are uploaded to Tencent Cloud servers and are only used by developers to troubleshoot problems. If you need to obtain log information, you can [contact us](https://trtc.io/contact). |
 
 ### isSupported
 
@@ -2271,7 +2267,6 @@ The following interfaces will throw this error code when an exception occurs: `e
 
 **Handling suggestions:**
 - Confirm whether the domain name and port required for communication meet your network environment requirements, refer to the document [Handle Firewall Restriction](tutorial-34-advanced-proxy.html)
-- Other issues need to be handled by engineers. [Contact us on telegram](https://t.me/+EPk6TMZEZMM5OGY1)
 
 | extraCode | Description |
 |-----------|-------------|
@@ -2280,7 +2275,7 @@ The following interfaces will throw this error code when an exception occurs: `e
 | 5503 | Re-entering the room failed: When the user experiences a network outage of more than 30s, the SDK will try to re-enter the room to restore the call, but the re-entry may fail due to the expiration of userSig, and this error will be thrown.<br />Handling suggestions: When encountering this error, you can use the latest userSig to call [TRTC.enterRoom](TRTC.html#enterRoom) to enter the room again |
 | 5505 | Video encoding failed. Usually, the device does not support h264 encoding. <br />Solution: Android devices from some manufacturers do not have H264 encoding capabilities (i.e., they cannot push streams). If you want to use the TRTC Web SDK to push streams in the browsers of these devices, please submit [Web SDK User Capability Support Application](https://console.tencentcloud.com/workorder/category) to enable VP8 encoding and decoding capabilities. |
 | 5506 | Audio encoding failed. <br />Solution: When encountering this error, you can upgrade to the latest SDK version, or prompt the user to upgrade to the latest system version. |
-| 5507 | Video decoding failed. Usually, the device does not support h264 decoding. <br />Solution: Refer to the solution for video encoding failure. |
+| 5507 | Video decoding failed. Usually, the device does not support h264 decoding. <br />Solution: . |
 | 5508 | Audio decoding failed. <br />Solution: When encountering this error, you can upgrade to the latest SDK version, or prompt the user to upgrade to the latest system version. |
 
 #### OPERATION_ABORT
@@ -2299,8 +2294,6 @@ The following interfaces will throw this error code when an exception occurs: `e
 
 **Description:** Unknown error or undefined error
 
-**Handling suggestions:** [Contact us on telegram](https://t.me/+EPk6TMZEZMM5OGY1)
-
 ## TYPE
 
 **TRTC Constants**

package/resource/knowledge/rtcengine/web/faq.md

@@ -75,7 +75,7 @@ function showCustomDialog(event) {
 
 ## Disable Autoplay Policy in Webview
 
-The default autoplay policy in Android and iOS Webview may be different to the browser. You can refer to the following document to turn off autoplay restrictions in your App.
+The default autoplay policy in Android and iOS Webview may be different to the browser. You can .
 
 - **Android Webview**: Call [setMediaPlaybackRequiresUserGesture(false)](https://developer.android.com/reference/android/webkit/WebSettings#setMediaPlaybackRequiresUserGesture(boolean)).
 - **iOS Webview**: Set [mediaTypesRequiringUserActionForPlayback](https://developer.apple.com/documentation/webkit/wkwebviewconfiguration/1851524-mediatypesrequiringuseractionfor) to WKAudiovisualMediaTypeNone.
@@ -133,7 +133,7 @@ trtc.on(TRTC.EVENT.ERROR, error => {
 
 ### Setting up a proxy server
 
-After you have set up Nginx and Turn server, you can refer to the following example to set up a proxy server.
+After you have set up Nginx and Turn server, you can .
 
 ```javascript
 const trtc = TRTC.create();
@@ -159,7 +159,7 @@ await trtc.enterRoom({
 
 1. **Deploy Nginx server**
 
-   Refer to the Nginx server deployment tutorial found on the Internet for deployment. If the enterprise has already deployed the Nginx service, it can be configured directly.
+ . If the enterprise has already deployed the Nginx service, it can be configured directly.
 
 2. **Configure Nginx server**
 
@@ -278,8 +278,6 @@ You can search for turn server setup tutorials on the Internet for setup, or you
 
 7. **Test the turn server**
 
-   Use this [test page](https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/) to test whether the turn server is accessible. If the result shows "done" as shown in the screenshot below, the turn server is working properly.
-
 ### Solution 2-B
 
 Solution B builds the Nginx proxy in the same way as Scenario A. There are two main differences:

package/resource/knowledge/rtcengine/web/how_to_integrated_web_sdk.md

@@ -58,8 +58,6 @@ When you use the TRTC Web SDK, you will encounter the following concepts:
    - Listen for the [TRTC.EVENT.REMOTE_VIDEO_AVAILABLE](module-EVENT.html#.REMOTE_VIDEO_AVAILABLE) event before entering the room to receive all remote user video publishing events.
    - In the event callback function, call the [trtc.startRemoteVideo()](TRTC.html#startRemoteVideo) method to play the remote video.
 
-The following figure shows the basic API call process for implementing audio and video communication:
-
 ## Creating a TRTC object
 
 Create a [TRTC](TRTC.html) object through the [TRTC.create()](TRTC.html#.create) method
@@ -159,10 +157,6 @@ trtc.on(TRTC.EVENT.KICKED_OUT, error => {
 });
 ```
 
-## Contact us
-
-If you encounter any problems during the implementation process, please feel free to create an issue on [GitHub issue](https://github.com/LiteAVSDK/TRTC_Web/issues), and we will deal with it as soon as possible.
-
 # Tutorial: Switching Camera/Mic
 
 This article mainly introduces how to switch cameras and microphones during audio and video calls.

package/resource/knowledge/rtcengine/web/plugin/20-advanced-customized-capture-rendering.md

@@ -66,7 +66,7 @@ trtc.startLocalVideo({ option:{ videoTrack } });
 
 By default, when calling {@link TRTC#startLocalVideo startLocalVideo()} or {@link TRTC#startRemoteVideo startRemoteVideo()}, you need to pass in the view parameter. The SDK will create a `video` tag under the specified element tag to play the video.
 
-If you need to customize the rendering, and do not need the SDK to play the video, you can refer to the following steps:
+If you need to customize the rendering, and do not need the SDK to play the video, you can
 
 - Do not fill in the view parameter or pass in null when calling the startLocalVideo or startRemoteVideo method.
 - Listen the TRTC.EVENT.TRACK, SDK will fires this event when there has a new MediaStreamTrack, then you can get the MediaStreamTrack for custom rendering.

package/resource/knowledge/rtcengine/web/plugin/21-advanced-auto-play-policy.md

@@ -85,7 +85,7 @@ function showCustomDialog(event) {
 
 ## Disable Autoplay Policy in Webview
 
-The default autoplay policy in Android and iOS Webview may be different to the browser. You can refer to the following document to turn off autoplay restrictions in your App.
+The default autoplay policy in Android and iOS Webview may be different to the browser. You can .
 
 - Android Webview: Call [setMediaPlaybackRequiresUserGesture(false)](https://developer.android.com/reference/android/webkit/WebSettings#setMediaPlaybackRequiresUserGesture(boolean)).
 - iOS Webview: Set [mediaTypesRequiringUserActionForPlayback](https://developer.apple.com/documentation/webkit/wkwebviewconfiguration/1851524-mediatypesrequiringuseractionfor) to WKAudiovisualMediaTypeNone.

package/resource/knowledge/rtcengine/web/plugin/26-advanced-publish-cdn-stream.md

@@ -444,4 +444,4 @@ You can view the current stream status through the [Stream Management Interface]
 
 Q. Screen sharing is blurry?
 
-The SDK uses `1080p` parameter configuration by default to capture screen sharing. Please refer to the interface: {@link TRTC#startScreenShare startScreenShare}
+The SDK uses `1080p` parameter configuration by default to capture screen sharing. Please

package/resource/knowledge/rtcengine/web/plugin/27-advanced-small-stream.md

@@ -118,7 +118,7 @@ After enabling this feature, when the user scrolls through the video list:
 - When the view changes from visible to invisible, the SDK will stop streaming.
 - When the view changes from invisible to visible, the SDK will resume streaming.
 
-Since resuming streaming until the video playback is successful is an asynchronous process, a black screen may appear during this process. To provide a better user experience, we recommend adding a loading effect to the video area during this asynchronous process. You can refer to the following code to determine whether to display the loading effect.
+Since resuming streaming until the video playback is successful is an asynchronous process, a black screen may appear during this process. To provide a better user experience, we recommend adding a loading effect to the video area during this asynchronous process. You can .
 
 ```js
 let isLoading = true;

package/resource/knowledge/rtcengine/web/plugin/30-advanced-cross-room-link.md

@@ -8,6 +8,8 @@ Cross-room connections support two modes:
 - **Room-level connection**: Connect with all anchors in the target room.
 - **User-level connection**: Connect with a specific anchor in the target room.
 
+> **Note**: Room-level and user-level connections cannot be used simultaneously.
+
 ## How It Works
 
 - **Room-level connection**: Push the audio and video streams of all anchors in the current room to the target room, and push the audio and video streams of all anchors in the target room to the current room.

package/resource/knowledge/rtcengine/web/plugin/31-advanced-plugins-overview.md

@@ -28,7 +28,6 @@ TRTC Web SDK v5 provides a plugin system that supports on-demand loading of vari
 | **AIDenoiser** | AI noise reduction with support for far-field cancellation mode | - | Exclusive Edition | Required | {@tutorial 35-advanced-ai-denoiser} |
 | **VoiceChanger** | Voice effects with 11 voice types (child, loli, uncle, etc.) | ≥ 5.10.0 | Premium Edition | - | {@tutorial 37-advanced-voice-changer} |
 | **RealtimeTranscriber** | Real-time speech transcription and translation, converting audio to text with multi-language translation | ≥ 5.15.0 | None | - | {@tutorial 42-advanced-realtime-transcriber} |
-| **Chorus** | KTV chorus with lead/backup singer roles, lyrics sync, and scoring | - | [Contact us](https://trtc.io/contact) | - | - |
 
 ### Streaming & Mixing
 
@@ -43,7 +42,6 @@ TRTC Web SDK v5 provides a plugin system that supports on-demand loading of vari
 |---------|---------|---------|---------|---------|---------|
 | **DeviceDetector** | Device detection for camera, microphone, speaker, and network with default UI | ≥ 5.8.0 | None | - | {@tutorial 23-advanced-support-detection} |
 | **Debug** | Debug mode with full log upload/export and audio/video dump | ≥ 5.8.0 | None | - | {@tutorial 18-basic-debug} |
-| **TRTCVideoDecoder** | Video decoder fallback to software decoding on failure | ≥ 5.9.0 | [Contact us](https://trtc.io/contact) | Required | {@tutorial 39-advanced-video-decoder} |
 | **SmallStreamAutoSwitcher** | Automatic dual-stream switching for smooth streaming in poor network | ≥ 5.11.0 | None | - | {@tutorial 41-advanced-small-stream-auto-switcher} |
 <!-- | **CustomEncryption** | Custom encryption for end-to-end audio/video encryption with built-in or custom algorithms | - | Contact us | - | - | -->
 
@@ -321,7 +319,7 @@ const trtc = TRTC.create({
 
 - Enable CORS policy if asset domain differs from app domain
 - Assets must be deployed on HTTPS (HTTP blocked by browser security policy)
-- SDK versions below v5.10.0 may require deploying additional files (refer to plugin docs)
+- SDK versions below v5.10.0 may require deploying additional files (
 
 ## Related Links
 

package/resource/knowledge/rtcengine/web/plugin/34-advanced-proxy.md

@@ -59,7 +59,7 @@ trtc.on(TRTC.EVENT.ERROR, error => {
 
 ### Setting up a proxy server
 
-After you have set up Nginx and Turn server, you can refer to the following example to set up a proxy server.
+After you have set up Nginx and Turn server, you can .
 
 ```javascript
 const trtc = TRTC.create();
@@ -86,7 +86,7 @@ await trtc.enterRoom({
 
 1. Deploy Nginx server
 
-   Refer to the Nginx server deployment tutorial found on the Internet for deployment. If the enterprise has already deployed the Nginx service, it can be configured directly.
+ . If the enterprise has already deployed the Nginx service, it can be configured directly.
 
 2. Configure Nginx server
 
@@ -220,8 +220,6 @@ You can search for turn server setup tutorials on the Internet for setup, or you
 
 7. Test the turn server
 
-   Use this [test page](https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/) to test whether the turn server is accessible. If the result shows "done" as shown in the screenshot below, the turn server is working properly.
-
 ### Solution 2-B
 
 Solution B builds the Nginx proxy in the same way as Scenario A. There are two main differences:

package/resource/knowledge/rtcengine/web/plugin/36-advanced-virtual-background.md

@@ -2,10 +2,6 @@
 
 This document will describe how to implement the virtual background feature during a call.
 
-| Original Camera | Background Blur | Background Image |
-| --- | --- | --- |
-|  |  |  |
-
 ## Prerequisites
 
 - Pricing see [RTC-Engine Packages](https://trtc.io/document/56025).

package/resource/knowledge/rtcengine/web/plugin/39-advanced-video-decoder.md

@@ -6,7 +6,7 @@ This document introduces how to use the TRTCVideoDecoder plugin to enable video
 
 - TRTC Web SDK version >= 5.9.0
 
-- [Contact us](https://trtc.io/contact) to enable decoder fallback capability
+- Browser version: Chrome 141+, Edge 141+, and other Chromium-based browsers 141+ (Safari and Firefox are not supported)
 
 ## Implementation Steps