WebRtc StatsReport parameters

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Александр Барон

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Dec 15, 2017, 1:54:03 AM12/15/17
to discuss-webrtc
In my android application I use WebRTC. At the moment, I asked myself the question of collecting statistics. From the PeerConnection object, I get statistics, but there are a lot of values. Some of them are obvious, they are completely incomprehensible to me. Help to deal with everyone. Mind it could not prompt the change of which parameters will indicate a worsening of communication.
ssrc : 2661238870</br>
transportId
: Channel-audio-1
googAccelerateRate
: 0
googCaptureStartNtpTimeMs
: 3722144040007
googCodecName
: opus
googCurrentDelayMs
: 156
googDecodingCNG
: 0
googDecodingCTN
: 34275
 googDecodingCTSG
: 0
googDecodingNormal
: 33904
googDecodingPLC
: 154
googDecodingPLCCNG
: 217
googExpandRate
: 0.750183
googJitterBufferMs
: 11
googJitterReceived
: 11
googPreemptiveExpandRate
: 0
googPreferredJitterBufferMs
: 60
googSecondaryDecodedRate
: 0
googSpeechExpandRate
: 0.750183
googTrackId
: ARDAMSa0
audioInputLevel
: 128
bytesSent
: 1618298
mediaType
: audio
packetsLost
: -1
packetsSent
: 17057
ssrc
: 1827811287
transportId
: Channel-audio-1
googCodecName
: opus
googEchoCancellationQualityMin
: -1
googEchoCancellationEchoDelayMedian
: -1
googEchoCancellationEchoDelayStdDev
: -1
googEchoCancellationReturnLoss
: -100
googEchoCancellationReturnLossEnhancement
: -100
googJitterReceived
: -1
googRtt
: -1
googTrackId
: ARDAMSa0
googTypingNoiseState
: false
googInitiator
: false
googTrackId
: ARDAMSa0
audioOutputLevel
: 760
bytesReceived
: 1701441
mediaType
: audio
packetsLost
: 11
packetsReceived
: 17102


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