npm - @tencent-rtc/mcp - Versions diffs - 1.3.9 → 1.4.0 - Mend

@tencent-rtc/mcp 1.3.9 → 1.4.0

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package/resource/doc/trtc/C++/ITRTCStatistics.md CHANGED Viewed

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 ## Overview
 `ITRTCStatistics.h` provides read-only statistics for TRTC Real-Time Communication (TRTC SDK) on C++. The TRTC SDK reports real-time audio and video metrics—including frame rate, bitrate, and stutter status—every two seconds.
 ## Struct Type
-<table>
-<tr>
-<td rowspan="1" colSpan="1" >Function List</td>
-<td rowspan="1" colSpan="1" >Description</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >[TRTCLocalStatistics](https://write.woa.com/document/87030057818304512)</td>
-<td rowspan="1" colSpan="1" >Local Audio and Video Statistics</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >[TRTCRemoteStatistics](https://write.woa.com/document/87030057818304512)</td>
-<td rowspan="1" colSpan="1" >Remote Audio and Video Statistics</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >[TRTCStatistics](https://write.woa.com/document/87030057818304512)</td>
-<td rowspan="1" colSpan="1" >Summary Statistics of Network and Performance</td>
-</tr>
-</table>
+| Function List | Description |
+| --- | --- |
+| TRTCLocalStatistics | Local Audio and Video Statistics |
+| TRTCRemoteStatistics | Remote Audio and Video Statistics |
+| TRTCStatistics | Summary Statistics of Network and Performance |
 ## TRTCLocalStatistics
 #### Local Audio and Video Statistics
-<table>
-<tr>
-<td rowspan="1" colSpan="1" >Enumeration Types</td>
-<td rowspan="1" colSpan="1" >Description</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >audioBitrate</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Local audio bitrate, that is, how much new audio data is generated per second, in Kbps.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >audioCaptureState</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Audio device collection status (used to detect the health of audio peripherals).<br>0: Collection Device Status Normal; 1: Long Period of Silence Detected; 2: Crackling Sound Detected; 3: Abnormal Sound Intermittence Detected.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >audioSampleRate</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Local audio sampling rate, in Hz.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >frameRate</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Local video frame rate, that is, how many video frames per second, unit: FPS.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >height</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Local video height, in px.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >streamType</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Video stream type (HD large screen \| Low-definition Small Picture \| Auxiliary Stream Picture).</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >videoBitrate</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Local video bitrate, that is, how much new video data is generated per second, in Kbps.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >width</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Local video width, in pixels (px).</td>
-</tr>
-</table>
+| Enumeration Types | Description |
+| --- | --- |
+| audioBitrate | [Field Meaning] Local audio bitrate, that is, how much new audio data is generated per second, in Kbps. |
+| audioCaptureState | [Field Meaning] Audio device collection status (used to detect the health of audio peripherals).<br>0: Collection Device Status Normal; 1: Long Period of Silence Detected; 2: Crackling Sound Detected; 3: Abnormal Sound Intermittence Detected. |
+| audioSampleRate | [Field Meaning] Local audio sampling rate, in Hz. |
+| frameRate | [Field Meaning] Local video frame rate, that is, how many video frames per second, unit: FPS. |
+| height | [Field Meaning] Local video height, in px. |
+| streamType | [Field Meaning] Video stream type (HD large screen \\| Low-definition Small Picture \\| Auxiliary Stream Picture). |
+| videoBitrate | [Field Meaning] Local video bitrate, that is, how much new video data is generated per second, in Kbps. |
+| width | [Field Meaning] Local video width, in pixels (px). |
 ## TRTCRemoteStatistics
 #### Remote Audio and Video Statistics
-<table>
-<tr>
-<td rowspan="1" colSpan="1" >Enumeration Types</td>
-<td rowspan="1" colSpan="1" >Description</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >audioBitrate</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Local audio bitrate, in Kbps</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >audioBlockRate</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Audio playback stuttering rate, in percentage (%)<br>Audio playback stuttering rate (audioBlockRate) = cumulative stuttering duration during audio playback (audioTotalBlockTime) / total duration of audio playback</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >audioPacketLoss</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Total audio stream packet loss rate (%).<br>audioPacketLoss represents the packet loss rate at the audience end after an audio stream completes the transmission chain from ` Broadcaster > Cloud > Audience `.<br>The smaller the audioPacketLoss, the better. A packet loss rate of 0 means all audio stream data has been completely received by the audience.<br>If downLoss == 0 but audioPacketLoss != 0 occurs, it indicates that the audio stream has no packet loss on the "cloud=>audience" segment, but there is irrecoverable packet loss on the ` host>cloud ` segment.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >audioSampleRate</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Local audio sampling rate, in Hz.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >audioTotalBlockTime</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Cumulative stuttering duration during audio playback, in milliseconds (ms)</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >finalLoss</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Overall packet loss rate of the audio and video stream (%). Deprecated; it is recommended to use audioPacketLoss and videoPacketLoss instead.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >frameRate</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Remote video frame rate, in FPS.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >height</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Remote video height, in pixels (px).</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >jitterBufferDelay</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Playback latency, in milliseconds (ms)<br>To avoid audio and screen freezing caused by network jitter and packet disorder, TRTC manages a playback buffer at the playback end to organize received network packets. The size of this buffer is adaptively adjusted based on current network quality. This buffer size is translated into a time length in milliseconds, referred to as jitterBufferDelay.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >point2PointDelay</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] End-to-end latency, in milliseconds (ms)<br>point2PointDelay represents the delay from "Anchors => Cloud => Audience". More precisely, it represents the full-link latency of "Collection => Encoding => Network Transmission => Receive => Buffering => Decoding => Play".<br>point2PointDelay requires both local and remote SDKs to be version 8.5 or above to be effective. If the remote user is on a version before 8.5, this value will always be 0, meaning it is not applicable.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >remoteNetworkRTT</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] The round-trip time from the SDK to the cloud, in milliseconds (ms)<br>This value represents the total time it takes for a network packet to travel from the SDK to the cloud and back, summing up the "SDK => Cloud => SDK" time.<br>The smaller this value, the better: if remoteNetworkRTT < 50ms, it indicates low audio and video call latency; if remoteNetworkRTT > 200ms, it suggests high audio and video call latency.<br>It is important to note that remoteNetworkRTT represents the total time for "SDK => Cloud => SDK", so there is no need to differentiate between remoteNetworkUpRTT and remoteNetworkDownRTT</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >remoteNetworkUplinkLoss</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] The uplink packet loss rate from the SDK to the cloud, in percentage (%)<br>The smaller the value, the better. If remoteNetworkUplinkLoss is 0%, it means the uplink network quality is good, with almost no data packet loss when uploading to the cloud.<br>If remoteNetworkUplinkLoss is 30%, it means that 30% of the audio and video data packets sent by the SDK to the cloud are lost in the transmission link.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >streamType</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Video stream type (HD large screen \| Low-definition Small Picture \| Auxiliary Stream Picture).</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >userId</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] User ID</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >videoBitrate</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Remote video bitrate, in Kbps.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >videoBlockRate</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Video playback stutter rate, in percentage (%). Video playback stutter rate (videoBlockRate) = cumulative stutter duration during video playback (videoTotalBlockTime) / total duration of video playback.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >videoPacketLoss</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Total packet loss rate of the video stream (%).<br>videoPacketLoss represents the packet loss rate at the audience end after the video stream completes the transmission chain from ` Anchors > Cloud > Audience `.<br>The smaller the videoPacketLoss, the better. A packet loss rate of 0 means all video stream data has been completely received by the audience.<br>If downLoss == 0 but videoPacketLoss != 0 occurs, it indicates that the video stream has no packet loss on the ` Cloud > Audience ` segment, but there is irrecoverable packet loss on the ` Anchors > Cloud ` segment.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >videoTotalBlockTime</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Cumulative stutter duration during video playback, in milliseconds (ms)</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >width</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Remote video width, in pixels (px).</td>
-</tr>
-</table>
+| Enumeration Types | Description |
+| --- | --- |
+| audioBitrate | [Field Meaning] Local audio bitrate, in Kbps |
+| audioBlockRate | [Field Meaning] Audio playback stuttering rate, in percentage (%)<br>Audio playback stuttering rate (audioBlockRate) = cumulative stuttering duration during audio playback (audioTotalBlockTime) / total duration of audio playback |
+| audioPacketLoss | [Field Meaning] Total audio stream packet loss rate (%).<br>audioPacketLoss represents the packet loss rate at the audience end after an audio stream completes the transmission chain from ` Broadcaster > Cloud > Audience `.<br>The smaller the audioPacketLoss, the better. A packet loss rate of 0 means all audio stream data has been completely received by the audience.<br>If downLoss == 0 but audioPacketLoss != 0 occurs, it indicates that the audio stream has no packet loss on the "cloud=>audience" segment, but there is irrecoverable packet loss on the ` host>cloud ` segment. |
+| audioSampleRate | [Field Meaning] Local audio sampling rate, in Hz. |
+| audioTotalBlockTime | [Field Meaning] Cumulative stuttering duration during audio playback, in milliseconds (ms) |
+| finalLoss | [Field Meaning] Overall packet loss rate of the audio and video stream (%). Deprecated; it is recommended to use audioPacketLoss and videoPacketLoss instead. |
+| frameRate | [Field Meaning] Remote video frame rate, in FPS. |
+| height | [Field Meaning] Remote video height, in pixels (px). |
+| jitterBufferDelay | [Field Meaning] Playback latency, in milliseconds (ms)<br>To avoid audio and screen freezing caused by network jitter and packet disorder, TRTC manages a playback buffer at the playback end to organize received network packets. The size of this buffer is adaptively adjusted based on current network quality. This buffer size is translated into a time length in milliseconds, referred to as jitterBufferDelay. |
+| point2PointDelay | [Field Meaning] End-to-end latency, in milliseconds (ms)<br>point2PointDelay represents the delay from "Anchors => Cloud => Audience". More precisely, it represents the full-link latency of "Collection => Encoding => Network Transmission => Receive => Buffering => Decoding => Play".<br>point2PointDelay requires both local and remote SDKs to be version 8.5 or above to be effective. If the remote user is on a version before 8.5, this value will always be 0, meaning it is not applicable. |
+| remoteNetworkRTT | [Field Meaning] The round-trip time from the SDK to the cloud, in milliseconds (ms)<br>This value represents the total time it takes for a network packet to travel from the SDK to the cloud and back, summing up the "SDK => Cloud => SDK" time.<br>The smaller this value, the better: if remoteNetworkRTT < 50ms, it indicates low audio and video call latency; if remoteNetworkRTT > 200ms, it suggests high audio and video call latency.<br>It is important to note that remoteNetworkRTT represents the total time for "SDK => Cloud => SDK", so there is no need to differentiate between remoteNetworkUpRTT and remoteNetworkDownRTT |
+| remoteNetworkUplinkLoss | [Field Meaning] The uplink packet loss rate from the SDK to the cloud, in percentage (%)<br>The smaller the value, the better. If remoteNetworkUplinkLoss is 0%, it means the uplink network quality is good, with almost no data packet loss when uploading to the cloud.<br>If remoteNetworkUplinkLoss is 30%, it means that 30% of the audio and video data packets sent by the SDK to the cloud are lost in the transmission link. |
+| streamType | [Field Meaning] Video stream type (HD large screen \\| Low-definition Small Picture \\| Auxiliary Stream Picture). |
+| userId | [Field Meaning] User ID |
+| videoBitrate | [Field Meaning] Remote video bitrate, in Kbps. |
+| videoBlockRate | [Field Meaning] Video playback stutter rate, in percentage (%). Video playback stutter rate (videoBlockRate) = cumulative stutter duration during video playback (videoTotalBlockTime) / total duration of video playback. |
+| videoPacketLoss | [Field Meaning] Total packet loss rate of the video stream (%).<br>videoPacketLoss represents the packet loss rate at the audience end after the video stream completes the transmission chain from ` Anchors > Cloud > Audience `.<br>The smaller the videoPacketLoss, the better. A packet loss rate of 0 means all video stream data has been completely received by the audience.<br>If downLoss == 0 but videoPacketLoss != 0 occurs, it indicates that the video stream has no packet loss on the ` Cloud > Audience ` segment, but there is irrecoverable packet loss on the ` Anchors > Cloud ` segment. |
+| videoTotalBlockTime | [Field Meaning] Cumulative stutter duration during video playback, in milliseconds (ms) |
+| width | [Field Meaning] Remote video width, in pixels (px). |
 ## TRTCStatistics
 #### Summary Statistics of Network and Performance
-<table>
-<tr>
-<td rowspan="1" colSpan="1" >Enumeration Types</td>
-<td rowspan="1" colSpan="1" >Description</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >appCpu</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Current application CPU utilization, in percentage (%); not supported on Android 8.0 and above.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >appMemoryUsageInMB</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Current application memory usage, in MB.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >downLoss</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Downlink packet loss rate from the cloud to SDK, in percentage (%)<br>The smaller the value, the better. If downLoss is 0%, it means the downlink network quality is good, with almost no data packet loss from the cloud.<br>If downLoss is 30%, it means that 30% of the audio and video data packets transmitted from the cloud to the SDK are lost in the transmission link.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >gatewayRtt</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Round-trip latency from the SDK to the local router, in milliseconds (ms). This value represents the total time it takes for a network packet to travel from the SDK to the local router gateway and back, summing up the [SDK -> Gateway -> SDK] time.<br>The smaller the value, the better. If gatewayRtt < 50ms, it means lower audio and video call latency. If gatewayRtt > 200ms, it means higher audio and video call latency.<br>This value is invalid when the network type is cellular.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >localStatisticsArray</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Local audio and video statistical information<br>Since there may be three local audio and video streams (i.e., high-definition main screen, low-definition small screen, and auxiliary screen), the local audio and video statistical information is an array.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >localStatisticsArraySize</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Size of the array localStatisticsArray</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >receivedBytes</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Total received bytes (including signaling data and audio and video data), in bytes.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >remoteStatisticsArray</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Remote audio and video statistical information<br>Since there may be multiple remote users at the same time, and each remote user may simultaneously have multiple audio and video streams (i.e., high-definition main screen, low-definition small screen, and auxiliary screen), the remote audio and video statistical information is an array.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >remoteStatisticsArraySize</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Size of the array remoteStatisticsArray.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >rtt</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Round-trip latency from the SDK to the cloud, in milliseconds (ms)<br>This value represents the total time it takes for a network packet to travel from the SDK to the cloud and back, summing up the "SDK => Cloud => SDK" time.<br>The smaller the value, the better: if RTT < 50ms, it means lower audio and video call latency; if RTT > 200ms, it means higher audio and video call latency.<br>It needs to be specifically explained that RTT represents the total time for "SDK => Cloud => SDK," so there is no need to distinguish between upRtt and downRtt.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >sentBytes</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Total bytes sent (including signaling data and audio and video data), in bytes (Bytes).</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >systemCpu</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Current system CPU utilization, in percentage (%); not supported on Android 8.0 and above.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >systemMemoryInMB</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Current system memory, in MB.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >systemMemoryUsageInMB</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Current system memory usage, in MB, not supported on iOS/MAC.</td>
-</tr>
-<tr>
-<td rowspan="1" colSpan="1" >upLoss</td>
-<td rowspan="1" colSpan="1" >[Field Meaning] Uplink packet loss rate from the SDK to the cloud, in percentage (%)<br>The smaller the value, the better. If upLoss is 0%, it means the uplink network quality is good, with almost no data packet loss when uploading to the cloud.<br>If upLoss is 30%, it means that 30% of the audio and video data packets sent by the SDK to the cloud are lost in the transmission link.</td>
-</tr>
-</table>
+| Enumeration Types | Description |
+| --- | --- |
+| appCpu | [Field Meaning] Current application CPU utilization, in percentage (%); not supported on Android 8.0 and above. |
+| appMemoryUsageInMB | [Field Meaning] Current application memory usage, in MB. |
+| downLoss | [Field Meaning] Downlink packet loss rate from the cloud to SDK, in percentage (%)<br>The smaller the value, the better. If downLoss is 0%, it means the downlink network quality is good, with almost no data packet loss from the cloud.<br>If downLoss is 30%, it means that 30% of the audio and video data packets transmitted from the cloud to the SDK are lost in the transmission link. |
+| gatewayRtt | [Field Meaning] Round-trip latency from the SDK to the local router, in milliseconds (ms). This value represents the total time it takes for a network packet to travel from the SDK to the local router gateway and back, summing up the [SDK -> Gateway -> SDK] time.<br>The smaller the value, the better. If gatewayRtt < 50ms, it means lower audio and video call latency. If gatewayRtt > 200ms, it means higher audio and video call latency.<br>This value is invalid when the network type is cellular. |
+| localStatisticsArray | [Field Meaning] Local audio and video statistical information<br>Since there may be three local audio and video streams (i.e., high-definition main screen, low-definition small screen, and auxiliary screen), the local audio and video statistical information is an array. |
+| localStatisticsArraySize | [Field Meaning] Size of the array localStatisticsArray |
+| receivedBytes | [Field Meaning] Total received bytes (including signaling data and audio and video data), in bytes. |
+| remoteStatisticsArray | [Field Meaning] Remote audio and video statistical information<br>Since there may be multiple remote users at the same time, and each remote user may simultaneously have multiple audio and video streams (i.e., high-definition main screen, low-definition small screen, and auxiliary screen), the remote audio and video statistical information is an array. |
+| remoteStatisticsArraySize | [Field Meaning] Size of the array remoteStatisticsArray. |
+| rtt | [Field Meaning] Round-trip latency from the SDK to the cloud, in milliseconds (ms)<br>This value represents the total time it takes for a network packet to travel from the SDK to the cloud and back, summing up the "SDK => Cloud => SDK" time.<br>The smaller the value, the better: if RTT < 50ms, it means lower audio and video call latency; if RTT > 200ms, it means higher audio and video call latency.<br>It needs to be specifically explained that RTT represents the total time for "SDK => Cloud => SDK," so there is no need to distinguish between upRtt and downRtt. |
+| sentBytes | [Field Meaning] Total bytes sent (including signaling data and audio and video data), in bytes (Bytes). |
+| systemCpu | [Field Meaning] Current system CPU utilization, in percentage (%); not supported on Android 8.0 and above. |
+| systemMemoryInMB | [Field Meaning] Current system memory, in MB. |
+| systemMemoryUsageInMB | [Field Meaning] Current system memory usage, in MB, not supported on iOS/MAC. |
+| upLoss | [Field Meaning] Uplink packet loss rate from the SDK to the cloud, in percentage (%)<br>The smaller the value, the better. If upLoss is 0%, it means the uplink network quality is good, with almost no data packet loss when uploading to the cloud.<br>If upLoss is 30%, it means that 30% of the audio and video data packets sent by the SDK to the cloud are lost in the transmission link. |