TRTCRemoteStatistics

Constructor

new TRTCRemoteStatistics(userId, audioPacketLoss, videoPacketLoss, width, height, frameRate, videoBitrate, audioSampleRate, audioBitrate, jitterBufferDelay, point2PointDelay, audioTotalBlockTime, audioBlockRate, videoTotalBlockTime, videoBlockRate, finalLoss, remoteNetworkUplinkLoss, remoteNetworkRTT, streamType)

Parameters:

Name	Type	Description
`userId`	String	`required` user ID
`audioPacketLoss`	Number	`required` total packet loss rate (%) of the audio stream `audioPacketLoss` represents the packet loss rate eventually calculated on the audience side after the audio/video stream goes through the complete transfer linkage of "anchor -> cloud -> audience". The smaller the `audioPacketLoss`, the better. The packet loss rate of 0 indicates that all data of the audio stream has entirely reached the audience. If `downLoss` is `0` but `audioPacketLoss` isn't, there is no packet loss on the linkage of "cloud -> audience" for the audiostream, but there are unrecoverable packet losses on the linkage of "anchor -> cloud".
`videoPacketLoss`	Number	`required` total packet loss rate (%) of the video stream `videoPacketLoss` represents the packet loss rate eventually calculated on the audience side after the audio/video stream goes through the complete transfer linkage of "anchor -> cloud -> audience". The smaller the `videoPacketLoss`, the better. The packet loss rate of 0 indicates that all data of the video stream has entirely reached the audience. If `downLoss` is `0` but `videoPacketLoss` isn't, there is no packet loss on the linkage of "cloud -> audience" for the video stream, but there are unrecoverable packet losses on the linkage of "anchor -> cloud".
`width`	Number	`required` remote video width in px
`height`	Number	`required` remote video height in px
`frameRate`	Number	`required` remote video frame rate (fps)
`videoBitrate`	Number	`required` remote video bitrate (Kbps)
`audioSampleRate`	Number	`required` remote audio sample rate (Hz)
`audioBitrate`	Number	`required` remote audio bitrate (Kbps)
`jitterBufferDelay`	Number	`required` playback delay (ms)
`point2PointDelay`	Number	`required` end-to-end delay (ms) `point2PointDelay` represents the delay of "anchor -> cloud -> audience". To be more precise, it represents the delay of the entire linkage of "collection -> encoding -> network transfer -> receiving -> buffering -> decoding -> playback". `point2PointDelay` works only if both the local and remote SDKs are on version 8.5 or above. If the remote SDK is on a version below 8.5, this value will always be 0 and thus meaningless.
`audioTotalBlockTime`	Number	`required` cumulative audio playback lag duration (ms)
`audioBlockRate`	Number	`required` audio playback lag rate (%)
`videoTotalBlockTime`	Number	`required` cumulative video playback lag duration (ms)
`videoBlockRate`	Number	`required` video playback lag rate (%)
`finalLoss`	Number	`required` total packet loss rate (%) of the audio/video stream Deprecated, please use audioPacketLoss and videoPacketLoss instead.
`remoteNetworkUplinkLoss`	Number	`required` upstream packet loss rate (%) from the SDK to cloud The smaller the value, the better. If `remoteNetworkUplinkLoss` is `0%`, the upstream network quality is very good, and the data packets uploaded to the cloud are basically not lost. If `remoteNetworkUplinkLoss` is `30%`, 30% of the audio/video data packets sent to the cloud by the SDK are lost on the transfer linkage.
`remoteNetworkRTT`	Number	`required` round-trip delay (ms) from the SDK to cloud This value represents the total time it takes to send a network packet from the SDK to the cloud and then send a network packet back from the cloud to the SDK, i.e., the total time it takes for a network packet to go through the linkage of "SDK -> cloud -> SDK". The smaller the value, the better. If `remoteNetworkRTT` is below 50 ms, it means a short audio/video call delay; if `remoteNetworkRTT` is above 200 ms, it means a long audio/video call delay. It should be explained that `remoteNetworkRTT` represents the total time spent on the linkage of "SDK -> cloud -> SDK"; therefore, there is no need to distinguish between `remoteNetworkUpRTT` and `remoteNetworkDownRTT`.
`streamType`	TRTCVideoStreamType	`required` video stream type (HD big image \| smooth small image \| substream image)