@tencent-rtc/mcp 1.6.7 → 1.6.8

package/resource/knowledge/trtc/ios/TRTCCloudDef.md

@@ -122,7 +122,6 @@ TRTC provides three different video streams, including:
 -  Substream image: it is generally used for screen sharing. Only one user in the room is allowed to publish the substream video image at any time, while other users must wait for this user to close the substream before they can publish their own substream.
 
 > **Note**
->
 > The SDK does not support enabling the smooth small image alone, which must be enabled together with the big image. It will automatically set the resolution and bitrate of the small image.
 >
 
@@ -674,9 +673,9 @@ You can test the network speed through the startSpeedTest: interface before the
 | expectedDownBandwidth | Expected downstream bandwidth (kbps, value range: 10 to 5000, no downlink bandwidth test when it is 0).<br><strong>Note</strong><br>When the parameter  <code>scene</code>  is set to TRTCSpeedTestScene_OnlineChorusTesting, in order to obtain more accurate information such as  <code>rtt / jitter</code> , the value range is limited to [10, 1000]. |
 | expectedUpBandwidth | Expected upstream bandwidth (kbps, value range: 10 to 5000, no uplink bandwidth test when it is 0).<br><strong>Note</strong><br>When the parameter  <code>scene</code>  is set to TRTCSpeedTestScene_OnlineChorusTesting, in order to obtain more accurate information such as  <code>rtt / jitter</code> , the value range is limited to [10, 1000]. |
 | scene | Speed test scene. |
-| sdkAppId | Application identification, please refer to the relevant instructions in TRTCParams. |
-| userId | User identification, please refer to the relevant instructions in TRTCParams. |
-| userSig | User signature, please refer to the relevant instructions in TRTCParams. |
+| sdkAppId | Application identification, please . |
+| userId | User identification, please . |
+| userSig | User signature, please . |
 
 ## TRTCSpeedTestResult
 
@@ -763,7 +762,7 @@ These parameters are used to specify the layout position information of each vid
 | mixUsers | Field description: specify the position, size, layer, and stream type of each video image in On-Cloud MixTranscoding<br>Recommended value: this field is an array in  <code>TRTCMixUser</code>  type, where each element represents the information of a video image. |
 | mode | Field description: layout mode<br>Recommended value: please choose a value according to your business needs. The preset mode has better applicability. |
 | streamId | Field description: ID of the live stream output to CDN<br>Recommended value: default value: null, that is, the audio/video streams in the room will be mixed into the audio/video stream of the caller of this API.<br>-  If you don't set this parameter, the SDK will execute the default logic, that is, it will mix the multiple audio/video streams in the room into the audio/video stream of the caller of this API, i.e., A + B => A.<br>-  If you set this parameter, the SDK will mix the audio/video streams in the room into the live stream you specify, i.e., A + B => C (C is the  <code>streamId</code>  you specify). |
-| videoBitrate | Field description: specify the target video bitrate (Kbps) of On-Cloud MixTranscoding<br>Recommended value: if you enter 0, TRTC will estimate a reasonable bitrate value based on  <code>videoWidth</code>  and  <code>videoHeight</code> . You can also refer to the recommended bitrate value in the video resolution enumeration definition (in the comment section). |
+| videoBitrate | Field description: specify the target video bitrate (Kbps) of On-Cloud MixTranscoding<br>Recommended value: if you enter 0, TRTC will estimate a reasonable bitrate value based on <code>videoWidth</code> and <code>videoHeight</code> . You can also . |
 | videoFramerate | Field description: specify the target video frame rate (fps) of On-Cloud MixTranscoding<br>Recommended value: default value: 15 fps. Value range: (0,30]. |
 | videoGOP | Field description: specify the target video keyframe interval (GOP) of On-Cloud MixTranscoding<br>Recommended value: default value: 2 (in seconds). Value range: [1,8]. |
 | videoHeight | Field description: specify the target resolution (height) of On-Cloud MixTranscoding<br>Recommended value: 640 px. If you only mix audio streams, please set both  <code>width</code>  and  <code>height</code>  to 0; otherwise, there will be a black background in the live stream after mixtranscoding. |
@@ -813,7 +812,7 @@ The startLocalRecording API is an enhanced version of the startAudioRecording AP
 
 **Sound effect parameter (disused).**
 
-"Sound effects" in TRTC refer to some short audio files (usually only a few seconds), such as "applause" and "laughter".
+"Sound effects" in TRTC .
 
 This parameter is used to specify the path and number of playback times of a sound effect file (short audio file) in the sound effect playback API TRTCCloud#playAudioEffect on legacy versions.
 
@@ -931,7 +930,7 @@ This enum type is used by the On-Cloud MixTranscoding parameter TRTCStreamMixing
 | videoEncodedFPS | <code>Description:</code>  The frame rate (fps) of the stream to publish.<br><code>Value:</code>  Value range: (0,30]. Default: 20. |
 | videoEncodedGOP | <code>Description:</code>  The keyframe interval (GOP) of the stream to publish.<br><code>Value:</code>  Value range: [1,5]. Default: 3 (seconds). |
 | videoEncodedHeight | <code>Description:</code>  The resolution (height) of the stream to publish.<br><code>Value:</code>  Recommended value: 640. If you mix only audio streams, to avoid displaying a black video in the transcoded stream, set both  <code>width</code>  and  <code>height</code>  to  <code>0</code> . |
-| videoEncodedKbps | <code>Description:</code>  The video bitrate (Kbps) of the stream to publish.<br><code>Value:</code>  If you set this parameter to  <code>0</code> , TRTC will work out a bitrate based on  <code>videoWidth</code>  and  <code>videoHeight</code> . For details, refer to the recommended bitrates for the constants of the resolution enum type (see comment). |
+| videoEncodedKbps | <code>Description:</code> The video bitrate (Kbps) of the stream to publish.<br><code>Value:</code> If you set this parameter to <code>0</code> , TRTC will work out a bitrate based on <code>videoWidth</code> and <code>videoHeight</code> . . |
 | videoEncodedWidth | <code>Description:</code>  The resolution (width) of the stream to publish.<br><code>Value:</code>  Recommended value: 368. If you mix only audio streams, to avoid displaying a black video in the transcoded stream, set both  <code>width</code>  and  <code>height</code>  to  <code>0</code> . |
 | videoSeiParams | <code>Description:</code>  SEI parameters. Default: null<br><code>Note:</code>  the parameter is passed in the form of a JSON string. Here is an example to use it:<br>`{`<br>`  "payLoadContent":"xxx",`<br>`  "payloadType":5,`<br>`  "payloadUuid":"1234567890abcdef1234567890abcdef",`<br>`  "interval":1000,`<br>`  "followIdr":false`<br>`}`<br>The currently supported fields and their meanings are as follows:<br>-  payloadContent: Required. The payload content of the passthrough SEI, which cannot be empty.<br>-  payloadType: Required. The type of the SEI message, with a value range of 5 or an integer within the range of [100, 254] (excluding 244, which is an internally defined timestamp SEI).<br>-  payloadUuid: Required when payloadType is 5, and ignored in other cases. The value must be a 32-digit hexadecimal number.<br>-  interval: Optional, default is 1000. The sending interval of the SEI, in milliseconds.<br>-  followIdr: Optional, default is false. When this value is true, the SEI will be ensured to be carried when sending a key frame, otherwise it is not guaranteed. |
 

package/resource/knowledge/trtc/ios/TRTCCloudDelegate.md

@@ -143,7 +143,6 @@ For more information on the error codes for room entry failure, see [Error Codes
 |result|If  `result`  is greater than 0, it indicates the time (in ms) the room entry takes; if  `result`  is less than 0, it represents the error code for room entry.|
 
 > **Note**
->
 > 1. In TRTC versions below 6.6, the  `onEnterRoom(result)`  callback is returned only if room entry succeeds, and the onError callback is returned if room entry fails.
 >
 > 2. In TRTC 6.6 and above, the  `onEnterRoom(result)`  callback is returned regardless of whether room entry succeeds or fails, and the onError callback is also returned if room entry fails.
@@ -245,7 +244,6 @@ Due to performance concerns, this callback works differently in different scenar
 |userId|User ID of the remote user|
 
 > **Note**
->
 > 1. The  `onRemoteUserEnterRoom`  callback indicates that a user entered the room, but it does not necessarily mean that the user enabled audio or video.
 >
 > 2. If you want to know whether a user enabled video, we recommend you use the onUserVideoAvailable callback.
@@ -294,7 +292,6 @@ You can then call startRemoteView to subscribe to the remote user’s video. If
 |userId|User ID of the remote user|
 
 > **Note**
->
 > The API used to display substream images is startRemoteView, not startRemoteSubStreamView, startRemoteSubStreamView is deprecated.
 >
 
@@ -313,7 +310,6 @@ If you receive the  `onUserAudioAvailable(userId, YES)`  callback, it indicates
 |userId|User ID of the remote user|
 
 > **Note**
->
 > The auto-subscription mode is used by default. You can switch to the manual subscription mode by calling setDefaultStreamRecvMode, but it must be called before room entry for the switch to take effect.
 >
 
@@ -334,7 +330,6 @@ The SDK returns this event callback when it starts rendering your first video fr
 |width|Video width|
 
 > **Note**
->
 > 1. The callback of the first local video frame being rendered is triggered only after you call startLocalPreview or startScreenCapture.
 >
 > 2. The callback of the first video frame of a remote user being rendered is triggered only after you call startRemoteView or startRemoteSubStreamView.
@@ -445,7 +440,6 @@ This callback is returned every 2 seconds and notifies you of the statistics on
 |statistics|Statistics, including local statistics and the statistics of remote users. For details, please see TRTCStatistics.|
 
 > **Note**
->
 > If you want to learn about only the current network quality and do not want to spend much time analyzing the statistics returned by this callback, we recommend you use onNetworkQuality.
 >
 
@@ -457,7 +451,7 @@ The callback is triggered by startSpeedTest.
 
 |Param|DESC|
 |---------|---------|
-|result|Speed test data, including loss rates, rtt and bandwidth rates, please refer to TRTCSpeedTestResult for details.|
+|result|Speed test data, including loss rates, rtt and bandwidth rates, please .|
 
 ## onConnectionLost
 
@@ -537,7 +531,6 @@ Note that the SDK returns this callback at the specified interval regardless of
 |userVolumes|An array that represents the volume of all users who are speaking in the room. Value range: [0, 100]|
 
 > **Note**
->
 >  `userVolumes`  is an array. If  `userId`  is empty, the elements in the array represent the volume of the local user’s audio. Otherwise, they represent the volume of a remote user’s audio.
 >
 
@@ -571,7 +564,6 @@ When users set the mic capturing volume via the UI or a keyboard shortcut, the S
 |volume|System audio capturing volume, which users can set in the audio control panel. Value range: [0, 100]|
 
 > **Note**
->
 > You need to call enableAudioVolumeEvaluation and set the callback interval ( `interval`  > 0) to enable the callback. To disable the callback, set  `interval`  to  `0` .
 >
 
@@ -591,7 +583,6 @@ When users set the system’s playback volume via the UI or a keyboard shortcut,
 |volume|The system playback volume, which users can set in the audio control panel. Value range: 0-100|
 
 > **Note**
->
 > You need to call enableAudioVolumeEvaluation and set the callback interval ( `interval`  > 0) to enable the callback. To disable the callback, set  `interval`  to  `0` .
 >
 
@@ -640,7 +631,6 @@ If the sender sets  `reliable`  to  `YES` , the SDK will use this callback to no
 |userId|User ID|
 
 > **Note**
->
 > The recipient receives this callback only if the sender sets  `reliable`  to  `YES` .
 >
 
@@ -695,7 +685,6 @@ The SDK will then receive the execution result from the CVM and return the resul
 |errMsg|Error message|
 
 > **Note**
->
 > If you receive a callback that the command is executed successfully, it only means that your command was sent to Tencent Cloud’s backend server. If the CDN vendor does not accept your streams, the publishing will still fail.
 >
 
@@ -930,7 +919,6 @@ uint32_t onProcessVideoFrame(TRTCVideoFrame * _Nonnull)srcFrame dstFrame:(TRTCVi
 |srcFrame|Used to carry images captured by TRTC via the camera|
 
 > **Note**
->
 > Currently, only the OpenGL texture scheme is supported(PC supports TRTCVideoBufferType_Buffer format Only)
 >
 
@@ -954,7 +942,6 @@ After you configure the callback of custom audio processing, the SDK will return
 |frame|Audio frames in PCM format|
 
 > **Note**
->
 > 1. Please avoid time-consuming operations in this callback function. The SDK processes an audio frame every 20 ms, so if your operation takes more than 20 ms, it will cause audio exceptions.
 >
 > 2. The audio data returned via this callback can be read and modified, but please keep the duration of your operation short.
@@ -986,7 +973,6 @@ Other users in the room can receive the message through the  `TRTCAudioFrame.ext
 |frame|Audio frames in PCM format|
 
 > **Note**
->
 > 1. Please avoid time-consuming operations in this callback function. The SDK processes an audio frame every 20 ms, so if your operation takes more than 20 ms, it will cause audio exceptions.
 >
 > 2. The audio data returned via this callback can be read and modified, but please keep the duration of your operation short.
@@ -1011,7 +997,6 @@ After you configure the callback of custom audio processing, the SDK will return
 |userId|User ID|
 
 > **Note**
->
 > The audio data returned via this callback can be read but not modified.
 >
 
@@ -1031,7 +1016,6 @@ After you configure the callback of custom audio processing, the SDK will return
 |frame|Audio frames in PCM format|
 
 > **Note**
->
 > 1. Please avoid time-consuming operations in this callback function. The SDK processes an audio frame every 20 ms, so if your operation takes more than 20 ms, it will cause audio exceptions.
 >
 > 2. The audio data returned via this callback can be read and modified, but please keep the duration of your operation short.
@@ -1055,7 +1039,6 @@ After you configure the callback of custom audio processing, the SDK will return
 |frame|Audio frames in PCM format|
 
 > **Note**
->
 > 1. This data returned via this callback is mixed from all audio in the SDK, including local audio after pre-processing (ANS, AEC, and AGC), special effects application, and music mixing, as well as all remote audio, but it does not include the in-ear monitoring data.
 >
 > 2. The audio data returned via this callback cannot be modified.
@@ -1077,7 +1060,6 @@ After you configure the callback of custom audio processing, the SDK will return
 |frame|Audio frames in PCM format|
 
 > **Note**
->
 > 1. Please avoid time-consuming operations in this callback function, or it will cause audio exceptions.
 >
 > 2. The audio data returned via this callback can be read and modified, but please keep the duration of your operation short.

package/resource/knowledge/trtc/ios/TXAudioEffectManager.md

@@ -59,7 +59,6 @@ Given that not all phones deliver excellent in-ear monitoring effects, we have b
 |enable| `YES:`  enable;  `NO` : disable|
 
 > **Note**
->
 > In-ear monitoring can be enabled only when earphones are used. Please remind anchors to use wired earphones.
 >
 
@@ -74,7 +73,6 @@ This API is used to set the volume of in-ear monitoring.
 |volume|Volume. Value range: [0, 150]; default: 100|
 
 > **Note**
->
 > The volume of the ear monitor is one of the most important factors affecting the user experience. Considering that different users have different sensitivities to volume, it is strongly recommended to provide a ear monitor volume adjustment slider on the UI interface. The recommended adjustment range is between 0 and 150. The gain effect of 100-150 is very helpful for some Android phones with low recording volume.
 >
 
@@ -85,7 +83,6 @@ This API is used to set the volume of in-ear monitoring.
 This API is used to set reverb effects for human voice. For the effects supported, please see TXVoiceReverbType.
 
 > **Note**
->
 > Effects become invalid after room exit. If you want to use the same effect after you enter the room again, you need to set the effect again using this API.
 >
 
@@ -96,7 +93,6 @@ This API is used to set reverb effects for human voice. For the effects supporte
 This API is used to set voice changing effects. For the effects supported, please see TXVoiceChangeType.
 
 > **Note**
->
 > Effects become invalid after room exit. If you want to use the same effect after you enter the room again, you need to set the effect again using this API.
 >
 
@@ -111,7 +107,6 @@ This API is used to set the volume of speech. It is often used together with the
 |volume|Volume. Value range: [0, 150]; default: 100|
 
 > **Note**
->
 > If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
 >
 
@@ -139,7 +134,6 @@ You must assign an ID to each music track so that you can start, stop, or set th
 |startBlock|Callback of starting music|
 
 > **Note**
->
 > 1. If you play the same music track multiple times, please use the same ID instead of a separate ID for each playback.
 >
 > 2. If you want to play different music tracks at the same time, use different IDs for them.
@@ -185,7 +179,6 @@ This API is used to set the local and remote playback volume of background music
 |volume|Volume. Value range: [0, 150]; default: 60|
 
 > **Note**
->
 > If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
 >
 
@@ -201,7 +194,6 @@ This API is used to control the remote playback volume (the volume heard by audi
 |volume|Volume. Value range: [0, 150]; default: 60|
 
 > **Note**
->
 > If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
 >
 
@@ -217,7 +209,6 @@ This API is used to control the local playback volume (the volume heard by ancho
 |volume|Volume. Value range: [0, 150]. default: 60|
 
 > **Note**
->
 > If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
 >
 
@@ -271,7 +262,6 @@ The length of the specified music file is returned. -1 indicates failure to get
 |pts|Unit: millisecond|
 
 > **Note**
->
 > Do not call this API frequently as the music file may be read and written to each time the API is called, which can be time-consuming.
 >
 > Wait till users finish dragging the progress bar before you call this API.
@@ -289,7 +279,6 @@ The length of the specified music file is returned. -1 indicates failure to get
 |scratchSpeedRate|Scratch disc speed, the default value is 1.0f, the range is: a floating point number between [-12.0 ~ 12.0], the positive/negative speed value indicates the direction is positive/negative, and the absolute value indicates the speed.|
 
 > **Note**
->
 > Precondition preloadMusic succeeds.
 >
 
@@ -304,7 +293,6 @@ You must assign an ID to each music track so that you can start, stop, or set th
 |musicParam|Music parameter|
 
 > **Note**
->
 > 1. Preload supports up to 2 preloads with different IDs at the same time, and the preload time does not exceed 10 minutes,you need to call stopPlayMusic API after use, otherwise the memory will not be released.
 >
 > 2. If the music corresponding to the ID is being played, the preloading fails, and stopPlayMusic must be called first.
@@ -330,7 +318,6 @@ You must assign an ID to each music track so that you can start, stop, or set th
 |index|Specify which track to play (the first track is played by default). Value range 0, total number of tracks).|
 
 > **Note**
->
 > The total number of tracks can be obtained through the [getMusicTrackCount interface.
 >
 

package/resource/knowledge/trtc/ios/TXDeviceManager.md

@@ -125,7 +125,6 @@ This API can be used to achieve the following:
 |position|The spot to focus on. Pass in the coordinates of the spot you want to focus on.|
 
 > **Note**
->
 > Before using this API, you must first disable auto focus using enableCameraAutoFocus.
 >
 
@@ -164,7 +163,6 @@ If the audio route is set to the speaker, the volume is relatively high, and the
 |type|Device type. Set it to the type of device you want to get. For details, please see the definition of TXMediaDeviceType.|
 
 > **Note**
->
 > -  To ensure that the SDK can manage the lifecycle of the  `ITXDeviceCollection`  object, after using this API, please call the  `release`  method to release the resources.
 > -  Do not use  `delete`  to release the Collection object returned as deleting the ITXDeviceCollection* pointer will cause crash.
 > -  The valid values of  `type`  are  `TXMediaDeviceTypeMic` ,  `TXMediaDeviceTypeSpeaker` , and  `TXMediaDeviceTypeCamera` .
@@ -229,7 +227,6 @@ This API is used to set the microphone and speaker types. Camera following the s
 **Starting camera testing (for desktop OS).**
 
 > **Note**
->
 > You can use the setCurrentDevice API to switch between cameras during testing.
 >
 
@@ -248,7 +245,6 @@ This API is used to test whether the mic functions properly. The mic volume dete
 |interval|Interval of volume callbacks, in milliseconds.|
 
 > **Note**
->
 > When this interface is called, the sound recorded by the microphone will be played back to the speakers by default.
 >
 

package/resource/knowledge/trtc/ios/TXLiteAVCode.md

@@ -48,7 +48,7 @@
 | ERR_UNSUPPORTED_SAMPLERATE | -1306 | Unsupported audio sample rate. |
 | ERR_SYSTEM_LOOPBACK_START_FAILED | -1307 | System audio loopback startup encountered an error. Common error reasons and solutions are: 1. The current system version does not support the include or excluded specified process loopback function, which requires Windows system version 10.0.19042 (20H2) or higher. For x64 specified path loopback function, it also requires Windows system version 10.0.19042 (20H2) or higher. Please check the current system version. 2. Invalid input parameters (including empty parameters, non-existent input paths, non-existent process IDs, and non-existent device names). Please check whether the input parameters are legal. 3. No available speaker device, please check whether it is available. |
 | ERR_SYSTEM_LOOPBACK_THIRD_PARTY_APP_EXIT | -1324 | The third party app loopback capture encountered an error. This error occurred after the third party app loopback was successfully started. During the sound collection process, the specified process or application unexpectedly exited. Please check the status of the specified process or application and try again by re-entering the process number |
-| ERR_TRTC_ENTER_ROOM_FAILED | -3301 | Failed to enter the room. For the reason, refer to the error message for -3301 in  <code>onError</code> . |
+| ERR_TRTC_ENTER_ROOM_FAILED | -3301 | Failed to enter the room. For the reason. |
 | ERR_TRTC_REQUEST_IP_TIMEOUT | -3307 | IP and signature request timed out. Check your network connection and whether your firewall allows UDP.<br>Try visiting the IP address 162.14.22.165:8000 or 162.14.6.105:8000 and the domain default-query.trtc.tencent-cloud.com:8000. |
 | ERR_TRTC_CONNECT_SERVER_TIMEOUT | -3308 | Room entry request timed out. Check your network connection and whether VPN is used. You can also switch to 4G to run a test. |
 | ERR_TRTC_ROOM_PARAM_NULL | -3316 | Empty room entry parameters. Please check whether valid parameters were passed in to the  <code>enterRoom:appScene:</code>  API. |

package/server.js

@@ -36,7 +36,7 @@ Workflow:
 export function createServer() {
     const server = new McpServer({
         name: "Tencent RTC MCP Server",
-        version: "1.6.6",
+        version: "1.6.8",
     }, {
         instructions: SERVER_INSTRUCTIONS,
     });

package/tools/common/search_trtc_knowledge.js

@@ -19,6 +19,7 @@ const registrySearchTRTCKnowledgeTool = (mcpServer) => {
             ])).optional().describe(parameter_descriptions.types),
             scenario: z.enum(['live', 'voice']).optional().describe(parameter_descriptions.scenario),
             limit: z.number().min(1).max(16).default(8).describe(parameter_descriptions.limit),
+            from: z.string().default('').describe(parameter_descriptions.from),
         },
     }, (params) => {
         const qid = params.queryID || randomUUID();
@@ -29,6 +30,7 @@ const registrySearchTRTCKnowledgeTool = (mcpServer) => {
             framework: params.frameworks?.join(',') ?? '',
             product: params.product ? params.product.join(',') : '',
             ide: params.ide,
+            from: params.from,
         });
         const response = searchKnowledge({
             prompt: params.prompt,

package/tools/definitions/search_trtc_knowledge_def.js

@@ -31,6 +31,7 @@ const PARAMETER_DESCRIPTIONS = {
     scenario: 'live | voice. Only valid for the live product.',
     queryID: "First call: ''. Follow-up calls in the same consultation MUST pass through the original queryID returned by the server.",
     ide: 'IDE name (cursor|claude-code|vscode|codebuddy|codex|trae｜others). Required.',
+    from: 'Empty string by default.',
     limit: 'Number of fragments. Default 8, maximum 16.',
 };
 export const SEARCH_KNOWLEDGE_DEFINITION = {

package/utils/report-cls-client.js

@@ -23,12 +23,12 @@ function reportCLSClient(reportData) {
 }
 function sendCLSReport(reportData) {
     const { SDKAppID: _SDKAppID } = ServerConfig.getInstance().getConfig();
-    const { SDKAppID = _SDKAppID, method = '', userID = '', framework = '', product = '', prompt = '', queryID = '', answer = '', feedback = '', info = '', ide = '', } = reportData;
+    const { SDKAppID = _SDKAppID, method = '', userID = '', framework = '', product = '', prompt = '', queryID = '', answer = '', feedback = '', info = '', ide = '', from = '', } = reportData;
     const identifier = getDeviceIdentifier();
     const time = Date.now();
     const data = {
         platform: 'mcp',
-        verison: '1.6.7',
+        verison: '1.6.8',
         sdkappid: `${SDKAppID}`,
         method: method,
         userid: queryID || userID,
@@ -41,7 +41,7 @@ function sendCLSReport(reportData) {
         level: product,
         app: info,
         callkitversion: ide,
-        isfromchat: '',
+        isfromchat: from,
     };
     return httpPost({
         hostname: 'ap-nanjing.cls.tencentcs.com',

package/utils/report-es-client.js

@@ -7,7 +7,7 @@ function reportESClient(reportData) {
         report: [
             {
                 platform: 'mcp',
-                version: '1.6.7',
+                version: '1.6.8',
                 sdkappid: `${SDKAppID}`,
                 method: method,
                 userID: `${userID}`,

package/resource/knowledge/roomkit/core-sdk/flutter/call-user-to-room.md

@@ -1,230 +0,0 @@
-`RoomStore` 是 `AtomicXCore` 中专职房间事务管理的模块，深度集成了会中呼叫能力。本章节旨在指导如何利用 `RoomStore` 高效完成会中呼叫相关功能的开发与接入。
-
-## 核心功能
-- **会中呼叫**：房间内任意参与者均可呼叫房间外任意成员加入房间。
-
-- **获取会中呼叫用户列表**：可以获取指定房间的会中呼叫的用户列表。
-
-## 核心概念
-
-在开始集成之前，需要通过下表了解一下 `RoomStore` 中与会中呼叫相关的几个核心概念：
-| <strong>核心概念</strong> | <strong>核心职责与描述</strong> |
-| --- | --- |
-| <a href="https://tencent-rtc.github.io/TUIKit_Flutter/api_room_room_store/RoomCallStatus.html">RoomCallStatus</a> | 代表会中呼叫的状态，包含：<br>- <code>none</code>（默认态）。<br>- <code>calling</code>（呼叫中）。<br>- <code>timeout</code>（呼叫超时）。<br>- <code>rejected</code>（呼叫被拒绝）。 |
-| <a href="https://tencent-rtc.github.io/TUIKit_Flutter/api_room_room_store/RoomCallResult.html">RoomCallResult</a> | 代表会中呼叫的结果，包含：<br>- <code>success</code>（呼叫成功）。<br>- <code>alreadyInCalling</code>（已在呼叫中）。<br>- <code>alreadyInRoom</code>（已在房间中）。 |
-| <a href="https://tencent-rtc.github.io/TUIKit_Flutter/api_room_room_store/CallRejectionReason.html">CallRejectionReason</a> | 代表呼叫被拒绝的原因，目前仅支持：<br>- <code>rejected</code>（主动拒绝）。<br>- <code>inOtherRoom</code>（在其他房间中）。 |
-| <a href="https://tencent-rtc.github.io/TUIKit_Flutter/api_room_room_store/RoomCall-class.html">RoomCall</a> | 代表会中呼叫业务的核心数据结构。负责存储呼叫者信息、被呼叫者信息、以及当前呼叫状态。 |
-| <a href="https://tencent-rtc.github.io/TUIKit_Flutter/api_room_room_store/RoomListener-class.html">RoomListener</a> | 代表房间动态的实时事件。其中包含会中呼叫相关的事件。 |
-| <a href="https://tencent-rtc.github.io/TUIKit_Flutter/api_room_room_store/RoomStore-class.html">RoomStore</a> | 这是与房间全生命周期相关的核心类。功能包含：发起会中呼叫，响应会中呼叫，获取房间的会中呼叫列表，并通过订阅其 <code>RoomListener</code> 来接收与会中呼叫相关的实时动态事件。 |
-
-## 实现步骤
-
-### 步骤1：组件集成
-
-请参考 接入概览 集成 **AtomicXCore SDK**，并已完成 实现登录逻辑 部分接入。
-
-### 步骤2：呼叫方功能
-
-#### **1. 发起呼叫**
-
-作为呼叫方调用 `RoomStore` 的 `callUserToRoom` 接口，可以批量呼叫用户进房。
-``` java
-import 'package:atomic_x_core/atomicxcore.dart';
-
-Future<void> callUserToRoom({
-  required List<String> userIDList,
-  int timeout = 30,
-  String? extensionInfo,
-}) async {
-  final result = await RoomStore.shared.callUserToRoom(
-    roomID: "roomID",
-    userIDList: userIDList,
-    timeout: timeout,
-    extensionInfo: extensionInfo,
-  );
-
-  if (result.isSuccess) {
-    print("呼叫用户请求发送成功");
-  } else {
-    print("呼叫用户失败: [错误码: ${result.errorCode}]: ${result.errorMessage}");
-  }
-}
-```
-
-#### **2. 取消呼叫**
-
-作为呼叫方调用 `RoomStore` 的 `cancelCall` 接口，可以批量取消正在呼叫的申请。
-``` java
-import 'package:atomic_x_core/atomicxcore.dart';
-
-Future<void> cancelCall(List<String> userIDList) async {
-  final result = await RoomStore.shared.cancelCall(
-    roomID: "roomID",
-    userIDList: userIDList,
-  );
-
-  if (result.isSuccess) {
-    print("取消呼叫成功");
-  } else {
-    print("取消呼叫失败: [错误码: ${result.errorCode}]: ${result.errorMessage}");
-  }
-}
-```
-
-#### **3. 获取呼叫列表信息**
-
-进入房间后，调用 `RoomStore` 的 `getPendingCalls` 接口，可以分页查询当前房间正在呼叫的用户列表信息。
-``` java
-import 'package:atomic_x_core/atomicxcore.dart';
-
-Future<void> getPendingCalls(String roomID, String? cursor) async {
-  // cursor 为分页游标，首次调用时传入 null，获取第一页数据，后续调用时传入上次返回的 cursor 值。
-  final result = await RoomStore.shared.getPendingCalls(
-    roomID: roomID,
-    cursor: cursor,
-  );
-
-  if (result.isSuccess) {
-    print("获取呼叫列表成功，列表信息: ${result.data}, 分页游标: ${result.cursor}");
-  } else {
-    print("获取呼叫列表失败: [错误码: ${result.errorCode}]: ${result.errorMessage}");
-  }
-}
-```
-
-#### **4. 订阅会中呼叫事件**
-
-作为呼叫方订阅 `RoomStore` 的 `RoomListener` 能够实时获取呼叫的生命周期事件。通过监听以下回调，可以精准捕捉呼叫的最终结果并据此更新 UI 状态或执行后续业务逻辑。
-``` java
-import 'package:atomic_x_core/atomicxcore.dart';
-
-late RoomListener _listener;
-
-/// 订阅会中呼叫相关事件
-void addRoomCallListener() {
-  _listener = RoomListener(
-    onCallTimeout: (roomInfo, call) {
-      print("呼叫超时 房间信息: ${roomInfo.roomID}, 呼叫者: ${call.caller.userID}");
-    },
-    onCallAccepted: (roomInfo, call) {
-      print("呼叫已被接受 房间信息: ${roomInfo.roomID}, 被呼叫者: ${call.callee.userID}");
-    },
-    onCallRejected: (roomInfo, call, reason) {
-      print("呼叫被拒绝 房间信息: ${roomInfo.roomID}, 被呼叫者: ${call.callee.userID}, 拒绝原因: $reason");
-    },
-  );
-
-  RoomStore.shared.addRoomListener(_listener);
-}
-
-void removeRoomCallListener() {
-  RoomStore.shared.removeRoomListener(_listener);
-}
-```
-
-**呼叫事件函数详细说明**
-
-|事件函数|触发时机与说明|
-|---------|---------|
-|[onCallTimeout](https://tencent-rtc.github.io/TUIKit_Flutter/api_room_room_store/RoomListener/onCallTimeout.html)|呼叫超时。当呼叫请求在预设的时间内（由调用 callUserToRoom 时指定）未得到被呼叫方的响应时触发。此时，呼叫自动结束，呼叫方通常需要提示“对方无人接听”。|
-|[onCallAccepted](https://tencent-rtc.github.io/TUIKit_Flutter/api_room_room_store/RoomListener/onCallAccepted.html)|呼叫被接受。被呼叫方主动点击了“接听”或“接受”按钮。收到此回调后，表示被呼叫者即将进入房间，呼叫方应更新 UI 状态，例如将该用户从“呼叫中”列表移除，并等待后续的成员进房通知来刷新成员列表。|
-|[onCallRejected](https://tencent-rtc.github.io/TUIKit_Flutter/api_room_room_store/RoomListener/onCallRejected.html)|呼叫被拒绝。被呼叫方主动拒绝了通话请求，或者因为被呼叫方当前正处于其他房间中而自动拒绝。回调中通常包含拒绝原因（如主动拒绝或在其他房间中）。|
-
-### 步骤3：被呼叫方功能
-
-#### **1. 订阅会中呼叫事件**
-
-作为被呼叫方订阅 `RoomStore` 的 `RoomListener` 可以实时监听并响应来自呼叫方的呼叫行为。以下是构建呼叫提醒界面以及维护呼叫链路状态的关键事件。
-``` java
-import 'package:atomic_x_core/atomicxcore.dart';
-
-late RoomListener _listener;
-
-/// 订阅会中呼叫相关事件
-void addRoomCallListener() {
-  _listener = RoomListener(
-    onCallReceived: (roomInfo, call, extensionInfo) {
-      print("收到房间呼叫 房间信息: ${roomInfo.roomID}, 呼叫者: ${call.caller.userID}, 扩展信息: $extensionInfo");
-    },
-    onCallCancelled: (roomInfo, call) {
-      print("呼叫被取消 房间信息: ${roomInfo.roomID}, 呼叫者: ${call.caller.userID}");
-    },
-    onCallRevokedByAdmin: (roomInfo, call, operatorUser) {
-      print("呼叫被管理员移除 房间信息: ${roomInfo.roomID}, 呼叫者: ${call.caller.userID}, 移除操作者: ${operatorUser.userID}");
-    },
-    onCallHandledByOtherDevice: (roomInfo, isAccepted) {
-      print("呼叫已被其他设备处理 房间信息: ${roomInfo.roomID}, 处理结果: $isAccepted");
-    },
-  );
-
-  RoomStore.shared.addRoomListener(_listener);
-}
-
-void removeRoomCallListener() {
-  RoomStore.shared.removeRoomListener(_listener);
-}
-```
-
-**事件函数详细说明**
-
-|事件函数|触发时机与说明|
-|---------|---------|
-|[onCallReceived](https://tencent-rtc.github.io/TUIKit_Flutter/api_room_room_store/RoomListener/onCallReceived.html)|收到呼叫邀请。当有其他用户向您发起入会邀请时触发。收到此回调后，被呼叫方通常应立即弹出“呼叫提醒”界面，展示呼叫者信息、房间信息，并提供“接听”和“拒绝”按钮。|
-|[onCallCancelled](https://tencent-rtc.github.io/TUIKit_Flutter/api_room_room_store/RoomListener/onCallCancelled.html)|呼叫被发起方取消。呼叫方在您未做出响应前，主动撤回了呼叫请求。收到此回调后，被呼叫方应立即关闭当前的来电界面，并提示“对方已取消呼叫”，以恢复应用正常操作流程。|
-|[onCallRevokedByAdmin](https://tencent-rtc.github.io/TUIKit_Flutter/api_room_room_store/RoomListener/onCallRevokedByAdmin.html)|呼叫被管理员撤销。房间管理员或后台系统因权限变更、房间解散等原因强行终止了该笔呼叫邀请。收到此通知后，被呼叫方需立即停止呼叫流程并隐藏相关 UI，处理逻辑通常类似于呼叫被取消。|
-|[onCallHandledByOtherDevice](https://tencent-rtc.github.io/TUIKit_Flutter/api_room_room_store/RoomListener/onCallHandledByOtherDevice.html)|呼叫已在其他设备处理。当用户在多个设备（例如手机和 iPad）同时登录时，如果呼叫已在其中一台设备上被处理（接受或拒绝），当前设备会收到此通知，用于同步 UI 状态并自动消除呼叫界面。|
-
-#### **2. 接受呼叫、拒绝呼叫**
-
-作为被呼叫方调用 `RoomStore` 的 `acceptCall` 或者 `rejectCall` 接口，处理接受或者拒绝呼叫邀请。
-``` java
-import 'package:atomic_x_core/atomicxcore.dart';
-
-late RoomListener _listener;
-
-/// 订阅会中呼叫相关事件
-void addRoomCallListener() {
-  _listener = RoomListener(
-    onCallReceived: (roomInfo, call, extensionInfo) {
-      print("收到房间呼叫 房间信息: ${roomInfo.roomID}, 呼叫者: ${call.caller.userID}, 扩展信息: $extensionInfo");
-      // 接受、拒绝呼叫邀请操作
-    },
-  );
-
-  RoomStore.shared.addRoomListener(_listener);
-}
-
-void removeRoomCallListener() {
-  RoomStore.shared.removeRoomListener(_listener);
-}
-
-// 接受呼叫邀请
-Future<void> acceptCall(String roomID) async {
-  final result = await RoomStore.shared.acceptCall(roomID);
-
-  if (result.isSuccess) {
-    print("接受呼叫成功，房间ID: $roomID");
-  } else {
-    print("接受呼叫失败: [错误码: ${result.errorCode}] ${result.errorMessage}");
-  }
-}
-
-// 拒绝呼叫邀请
-Future<void> rejectCall(String roomID, {CallRejectionReason reason = CallRejectionReason.rejected}) async {
-  final result = await RoomStore.shared.rejectCall(
-    roomID: roomID,
-    reason: reason,
-  );
-
-  if (result.isSuccess) {
-    print("拒绝呼叫成功，房间ID: $roomID");
-  } else {
-    print("拒绝呼叫失败: [错误码: ${result.errorCode}] ${result.errorMessage}");
-  }
-}
-```
-
-## API 文档
-| <strong>Store/Component</strong> | <strong>功能描述</strong> | <strong>API 文档</strong> |
-| --- | --- | --- |
-| <strong>RoomStore</strong> | 房间全生命周期管理：创建并加入 / 加入 / 离开 / 结束房间 / 更新、获取房间信息 / 预定房间 / 呼叫房间外成员 / 监听房间内被动事件（例如房间解散，房间信息更新等）。 | <a href="https://tencent-rtc.github.io/TUIKit_Flutter/api_room_room_store/RoomStore-class.html">API 文档</a> |