Interface RtcStats

WebRTC Stats of a Voice Call

interface RtcStats {
    get audioLevel(): any;
    get audioRecvBytes(): any;
    get audioRecvJitter(): any;
    get audioRecvPackets(): any;
    get audioRecvPacketsLost(): any;
    get audioRtt(): any;
    get audioSentBytes(): any;
    get audioSentJitter(): any;
    get audioSentPackets(): any;
    get audioSentPacketsLost(): any;
    get concealedSamples(): any;
    get concealmentEvents(): any;
    get echoReturnLoss(): any;
    get echoReturnLossEnhancement(): any;
    get totalAudioEnergy(): any;
    get totalSamplesSent(): any;
}

Accessors

  • get audioLevel(): any
  • Returns any

  • get audioRecvBytes(): any
  • Returns any

  • get audioRecvJitter(): any
  • Returns any

  • get audioRecvPackets(): any
  • Returns any

  • get audioRecvPacketsLost(): any
  • Returns any

  • get audioRtt(): any
  • Returns any

  • get audioSentBytes(): any
  • Returns any

  • get audioSentJitter(): any
  • Returns any

  • get audioSentPackets(): any
  • Returns any

  • get audioSentPacketsLost(): any
  • Returns any

  • get concealedSamples(): any
  • Returns any

  • get concealmentEvents(): any
  • Returns any

  • get echoReturnLoss(): any
  • Returns any

  • get echoReturnLossEnhancement(): any
  • Returns any

  • get totalAudioEnergy(): any
  • Returns any

  • get totalSamplesSent(): any
  • Returns any