@webex/plugin-meetings 2.60.0-next.1 → 2.60.0-next.11

package/dist/statsAnalyzer/mqaUtil.js.map

@@ -1 +1 @@
-{"version":3,"names":["_lodash","require","_constants","getAudioReceiverMqa","exports","_ref","_lastMqaDataSent$medi","_lastMqaDataSent$medi2","_lastMqaDataSent$medi3","_lastMqaDataSent$medi4","_lastMqaDataSent$medi5","_lastMqaDataSent$medi6","_lastMqaDataSent$medi7","_lastMqaDataSent$medi8","audioReceiver","statsResults","lastMqaDataSent","mediaType","sendrecvType","STATS","RECEIVE_DIRECTION","lastPacketsReceived","totalPacketsReceived","lastPacketsLost","totalPacketsLost","lastPacketsDecoded","totalSamplesDecoded","lastSamplesReceived","totalSamplesReceived","lastConcealedSamples","concealedSamples","lastBytesReceived","totalBytesReceived","lastFecPacketsReceived","fecPacketsReceived","lastFecPacketsDiscarded","fecPacketsDiscarded","csi","streams","common","includes","push","direction","transportType","connectionType","local","transport","rtpPackets","mediaHopByHopLost","rtpHopByHopLost","maxRtpJitter","max","meanRtpJitter","mean","rtpJitter","fecRecovered","rtpEndToEndLost","framesDropped","renderedFrameRate","framesReceived","concealedFrames","receivedBitrate","rtpBitrate","getAudioSenderMqa","_ref2","_lastMqaDataSent$medi9","_lastMqaDataSent$medi10","_lastMqaDataSent$medi11","_lastMqaDataSent$medi12","_lastMqaDataSent$medi13","audioSender","SEND_DIRECTION","lastPacketsSent","totalPacketsSent","totalPacketsLostOnReceiver","lastBytesSent","totalBytesSent","lastFramesEncoded","totalKeyFramesEncoded","lastFirCount","totalFirCount","maxRemoteJitter","meanRemoteJitter","availableBitrate","availableOutgoingBitrate","totalpacketsLostForaMin","remoteLossRate","maxRoundTripTime","meanRoundTripTime","roundTripTime","totalBytesSentInaMin","transmittedBitrate","transmittedKeyFrames","requestedKeyFrames","getVideoReceiverMqa","_ref3","_lastMqaDataSent$medi14","_lastMqaDataSent$medi15","_lastMqaDataSent$medi16","_lastMqaDataSent$medi17","_lastMqaDataSent$medi18","_lastMqaDataSent$medi19","_lastMqaDataSent$medi20","_lastMqaDataSent$medi21","videoReceiver","lastFramesReceived","lastFramesDecoded","framesDecoded","lastFramesDropped","lastKeyFramesDecoded","keyFramesDecoded","lastPliCount","totalPliCount","totalPacketLoss","totalBytesReceivedInaMin","totalFrameReceivedInaMin","resolutions","totalFrameDecodedInaMin","receivedFrameRate","receivedHeight","height","receivedWidth","width","receivedFrameSize","receivedKeyFrames","getVideoSenderMqa","_ref4","_lastMqaDataSent$medi22","_lastMqaDataSent$medi23","_lastMqaDataSent$medi24","_lastMqaDataSent$medi25","_lastMqaDataSent$medi26","_lastMqaDataSent$medi27","videoSender","lastKeyFramesEncoded","lastFramesSent","framesSent","totalFrameSentInaMin","transmittedFrameRate","transmittedHeight","transmittedWidth","transmittedFrameSize"],"sources":["mqaUtil.ts"],"sourcesContent":["/* eslint-disable no-param-reassign, prefer-destructuring */\n\nimport {mean, max} from 'lodash';\n\nimport {STATS} from '../constants';\n\nexport const getAudioReceiverMqa = ({audioReceiver, statsResults, lastMqaDataSent, mediaType}) => {\n  const sendrecvType = STATS.RECEIVE_DIRECTION;\n\n  const lastPacketsReceived = lastMqaDataSent[mediaType]?.[sendrecvType].totalPacketsReceived || 0;\n  const lastPacketsLost = lastMqaDataSent[mediaType]?.[sendrecvType].totalPacketsLost || 0;\n  const lastPacketsDecoded = lastMqaDataSent[mediaType]?.[sendrecvType].totalSamplesDecoded || 0;\n  const lastSamplesReceived = lastMqaDataSent[mediaType]?.[sendrecvType].totalSamplesReceived || 0;\n  const lastConcealedSamples = lastMqaDataSent[mediaType]?.[sendrecvType].concealedSamples || 0;\n  const lastBytesReceived = lastMqaDataSent[mediaType]?.[sendrecvType].totalBytesReceived || 0;\n  const lastFecPacketsReceived = lastMqaDataSent[mediaType]?.[sendrecvType].fecPacketsReceived || 0;\n  const lastFecPacketsDiscarded =\n    lastMqaDataSent[mediaType]?.[sendrecvType].fecPacketsDiscarded || 0;\n\n  const {csi} = statsResults[mediaType];\n  if (csi && !audioReceiver.streams[0].common.csi.includes(csi)) {\n    audioReceiver.streams[0].common.csi.push(csi);\n  }\n\n  audioReceiver.common.common.direction = statsResults[mediaType].direction;\n  audioReceiver.common.transportType = statsResults.connectionType.local.transport;\n\n  // add rtpPacket info inside common as also for call analyzer\n  audioReceiver.common.rtpPackets =\n    statsResults[mediaType][sendrecvType].totalPacketsReceived - lastPacketsReceived || 0;\n  // Hop by hop are numbers and not percentage so we compare on what we sent the last min\n  // collect the packets received for the last min\n  audioReceiver.streams[0].common.rtpPackets = audioReceiver.common.rtpPackets;\n  audioReceiver.common.mediaHopByHopLost =\n    statsResults[mediaType][sendrecvType].totalPacketsLost - lastPacketsLost || 0;\n  audioReceiver.common.rtpHopByHopLost =\n    statsResults[mediaType][sendrecvType].totalPacketsLost - lastPacketsLost || 0;\n\n  audioReceiver.streams[0].common.maxRtpJitter =\n    // @ts-ignore\n    max(statsResults[mediaType][sendrecvType].meanRtpJitter) * 1000 || 0;\n  audioReceiver.streams[0].common.meanRtpJitter =\n    mean(statsResults[mediaType][sendrecvType].meanRtpJitter) * 1000 || 0;\n  audioReceiver.streams[0].common.rtpJitter = audioReceiver.streams[0].common.maxRtpJitter;\n\n  // Fec packets do come in as part of the FEC only for audio\n  const fecRecovered =\n    statsResults[mediaType][sendrecvType].fecPacketsReceived -\n    lastFecPacketsReceived -\n    (statsResults[mediaType][sendrecvType].fecPacketsDiscarded - lastFecPacketsDiscarded);\n\n  audioReceiver.streams[0].common.rtpEndToEndLost =\n    statsResults[mediaType][sendrecvType].totalPacketsLost - lastPacketsLost - fecRecovered || 0;\n\n  audioReceiver.streams[0].common.framesDropped =\n    statsResults[mediaType][sendrecvType].totalSamplesDecoded - lastPacketsDecoded || 0;\n  audioReceiver.streams[0].common.renderedFrameRate =\n    (audioReceiver.streams[0].common.framesDropped * 100) / 60 || 0;\n\n  audioReceiver.streams[0].common.framesReceived =\n    statsResults[mediaType][sendrecvType].totalSamplesReceived - lastSamplesReceived || 0;\n  audioReceiver.streams[0].common.concealedFrames =\n    statsResults[mediaType][sendrecvType].concealedSamples - lastConcealedSamples || 0;\n  audioReceiver.streams[0].common.receivedBitrate =\n    ((statsResults[mediaType][sendrecvType].totalBytesReceived - lastBytesReceived) * 8) / 60 || 0;\n\n  audioReceiver.common.rtpBitrate = audioReceiver.streams[0].common.receivedBitrate;\n};\n\nexport const getAudioSenderMqa = ({audioSender, statsResults, lastMqaDataSent, mediaType}) => {\n  const sendrecvType = STATS.SEND_DIRECTION;\n\n  const lastPacketsSent = lastMqaDataSent[mediaType]?.[sendrecvType].totalPacketsSent || 0;\n  const lastPacketsLost =\n    lastMqaDataSent[mediaType]?.[sendrecvType].totalPacketsLostOnReceiver || 0;\n  const lastBytesSent = lastMqaDataSent[mediaType]?.[sendrecvType].totalBytesSent || 0;\n  const lastFramesEncoded = lastMqaDataSent[mediaType]?.[sendrecvType].totalKeyFramesEncoded || 0;\n  const lastFirCount = lastMqaDataSent[mediaType]?.[sendrecvType].totalFirCount || 0;\n\n  const {csi} = statsResults[mediaType];\n  if (csi && !audioSender.streams[0].common.csi.includes(csi)) {\n    audioSender.streams[0].common.csi.push(csi);\n  }\n\n  audioSender.common.common.direction = statsResults[mediaType].direction;\n  audioSender.common.transportType = statsResults.connectionType.local.transport;\n\n  audioSender.common.maxRemoteJitter =\n    // @ts-ignore\n    max(statsResults[mediaType][sendrecvType].meanRemoteJitter) * 1000 || 0;\n  audioSender.common.meanRemoteJitter =\n    mean(statsResults[mediaType][sendrecvType].meanRemoteJitter) * 1000 || 0;\n\n  audioSender.common.rtpPackets =\n    statsResults[mediaType][sendrecvType].totalPacketsSent - lastPacketsSent || 0;\n  audioSender.streams[0].common.rtpPackets = audioSender.common.rtpPackets;\n  // From candidate-pair\n  audioSender.common.availableBitrate =\n    statsResults[mediaType][sendrecvType].availableOutgoingBitrate || 0;\n  // Calculate based on how much packets lost of received compated to how to the client sent\n\n  const totalpacketsLostForaMin =\n    statsResults[mediaType][sendrecvType].totalPacketsLostOnReceiver - lastPacketsLost;\n\n  audioSender.common.remoteLossRate =\n    totalpacketsLostForaMin > 0\n      ? (totalpacketsLostForaMin * 100) / audioSender.common.rtpPackets\n      : 0; // This is the packets sent with in last min || 0;\n\n  audioSender.common.maxRoundTripTime =\n    // @ts-ignore\n    max(statsResults[mediaType][sendrecvType].meanRoundTripTime) * 1000 || 0;\n  audioSender.common.meanRoundTripTime =\n    mean(statsResults[mediaType][sendrecvType].meanRoundTripTime) * 1000 || 0;\n  audioSender.common.roundTripTime = audioSender.common.maxRoundTripTime;\n\n  // Calculate the outgoing bitrate\n  const totalBytesSentInaMin = statsResults[mediaType][sendrecvType].totalBytesSent - lastBytesSent;\n\n  audioSender.streams[0].common.transmittedBitrate = totalBytesSentInaMin\n    ? (totalBytesSentInaMin * 8) / 60\n    : 0;\n  audioSender.common.rtpBitrate = audioSender.streams[0].common.transmittedBitrate;\n\n  audioSender.streams[0].transmittedKeyFrames =\n    statsResults[mediaType][sendrecvType].totalKeyFramesEncoded - lastFramesEncoded || 0;\n  audioSender.streams[0].requestedKeyFrames =\n    statsResults[mediaType][sendrecvType].totalFirCount - lastFirCount || 0;\n};\n\nexport const getVideoReceiverMqa = ({videoReceiver, statsResults, lastMqaDataSent, mediaType}) => {\n  const sendrecvType = STATS.RECEIVE_DIRECTION;\n\n  const lastPacketsReceived = lastMqaDataSent[mediaType]?.[sendrecvType].totalPacketsReceived || 0;\n  const lastPacketsLost = lastMqaDataSent[mediaType]?.[sendrecvType].totalPacketsLost || 0;\n  const lastBytesReceived = lastMqaDataSent[mediaType]?.[sendrecvType].totalBytesReceived || 0;\n  const lastFramesReceived = lastMqaDataSent[mediaType]?.[sendrecvType].framesReceived || 0;\n  const lastFramesDecoded = lastMqaDataSent[mediaType]?.[sendrecvType].framesDecoded || 0;\n  const lastFramesDropped = lastMqaDataSent[mediaType]?.[sendrecvType].framesDropped || 0;\n  const lastKeyFramesDecoded = lastMqaDataSent[mediaType]?.[sendrecvType].keyFramesDecoded || 0;\n  const lastPliCount = lastMqaDataSent[mediaType]?.[sendrecvType].totalPliCount || 0;\n\n  const {csi} = statsResults[mediaType];\n  if (csi && !videoReceiver.streams[0].common.csi.includes(csi)) {\n    videoReceiver.streams[0].common.csi.push(csi);\n  }\n\n  videoReceiver.common.common.direction = statsResults[mediaType].direction;\n  videoReceiver.common.transportType = statsResults.connectionType.local.transport;\n\n  // collect the packets received for the last min\n  videoReceiver.common.rtpPackets =\n    statsResults[mediaType][sendrecvType].totalPacketsReceived - lastPacketsReceived || 0;\n  videoReceiver.streams[0].common.rtpPackets = videoReceiver.common.rtpPackets;\n\n  const totalPacketLoss =\n    statsResults[mediaType][sendrecvType].totalPacketsLost - lastPacketsLost || 0;\n\n  // Hope by hop are numbers and not percentage so we compare on what we sent the last min\n  // this is including packet lost\n  videoReceiver.common.mediaHopByHopLost = totalPacketLoss;\n  videoReceiver.common.rtpHopByHopLost = totalPacketLoss;\n\n  // End to end packetloss is after recovery\n  videoReceiver.streams[0].common.rtpEndToEndLost = totalPacketLoss;\n\n  // calculate this values\n\n  videoReceiver.common.maxRemoteJitter =\n    // @ts-ignore\n    max(statsResults[mediaType][sendrecvType].meanRemoteJitter) * 1000 || 0;\n  videoReceiver.common.meanRemoteJitter =\n    mean(statsResults[mediaType][sendrecvType].meanRemoteJitter) * 1000 || 0;\n\n  videoReceiver.streams[0].common.rtpJitter = videoReceiver.common.maxRemoteJitter;\n  // videoReceiver.streams[0].common.rtpJitter = (statsResults.resolutions[mediaType][sendrecvType].jitterBufferDelay - lastMqaDataSent.resolutions[mediaType]?.[sendrecvType].jitterBufferDelay) / (statsResults.resolutions[mediaType][sendrecvType].jitterBufferEmittedCount - lastMqaDataSent.resolutions[mediaType]?.[sendrecvType].jitterBufferEmittedCount) * 1000 || 0;\n\n  // Calculate the outgoing bitrate\n  const totalBytesReceivedInaMin =\n    statsResults[mediaType][sendrecvType].totalBytesReceived - lastBytesReceived;\n\n  videoReceiver.streams[0].common.receivedBitrate = totalBytesReceivedInaMin\n    ? (totalBytesReceivedInaMin * 8) / 60\n    : 0;\n  videoReceiver.common.rtpBitrate = videoReceiver.streams[0].common.receivedBitrate;\n\n  // From tracks //TODO: calculate a proper one\n  const totalFrameReceivedInaMin =\n    statsResults.resolutions[mediaType][sendrecvType].framesReceived - lastFramesReceived;\n  const totalFrameDecodedInaMin =\n    statsResults.resolutions[mediaType][sendrecvType].framesDecoded - lastFramesDecoded;\n\n  videoReceiver.streams[0].common.receivedFrameRate = totalFrameReceivedInaMin\n    ? (totalFrameReceivedInaMin * 100) / 60\n    : 0;\n  videoReceiver.streams[0].common.renderedFrameRate = totalFrameDecodedInaMin\n    ? (totalFrameDecodedInaMin * 100) / 60\n    : 0;\n\n  videoReceiver.streams[0].common.framesDropped =\n    statsResults.resolutions[mediaType][sendrecvType].framesDropped - lastFramesDropped;\n  videoReceiver.streams[0].receivedHeight =\n    statsResults.resolutions[mediaType][sendrecvType].height || 0;\n  videoReceiver.streams[0].receivedWidth =\n    statsResults.resolutions[mediaType][sendrecvType].width || 0;\n  videoReceiver.streams[0].receivedFrameSize =\n    (videoReceiver.streams[0].receivedHeight * videoReceiver.streams[0].receivedWidth) / 256;\n\n  videoReceiver.streams[0].receivedKeyFrames =\n    statsResults[mediaType][sendrecvType].keyFramesDecoded - lastKeyFramesDecoded || 0;\n  videoReceiver.streams[0].requestedKeyFrames =\n    statsResults[mediaType][sendrecvType].totalPliCount - lastPliCount || 0;\n};\n\nexport const getVideoSenderMqa = ({videoSender, statsResults, lastMqaDataSent, mediaType}) => {\n  const sendrecvType = STATS.SEND_DIRECTION;\n\n  const lastPacketsSent = lastMqaDataSent[mediaType]?.[sendrecvType].totalPacketsSent || 0;\n  const lastPacketsLost =\n    lastMqaDataSent[mediaType]?.[sendrecvType].totalPacketsLostOnReceiver || 0;\n  const lastBytesSent = lastMqaDataSent[mediaType]?.[sendrecvType].totalBytesSent || 0;\n  const lastKeyFramesEncoded =\n    lastMqaDataSent[mediaType]?.[sendrecvType].totalKeyFramesEncoded || 0;\n  const lastFirCount = lastMqaDataSent[mediaType]?.[sendrecvType].totalFirCount || 0;\n  const lastFramesSent = lastMqaDataSent[mediaType]?.[sendrecvType].framesSent || 0;\n  const {csi} = statsResults[mediaType];\n  if (csi && !videoSender.streams[0].common.csi.includes(csi)) {\n    videoSender.streams[0].common.csi.push(csi);\n  }\n\n  videoSender.common.common.direction = statsResults[mediaType].direction;\n  videoSender.common.transportType = statsResults.connectionType.local.transport;\n\n  // @ts-ignore\n  videoSender.common.maxRemoteJitter =\n    // @ts-ignore\n    max(statsResults[mediaType][sendrecvType].meanRemoteJitter) * 1000 || 0;\n  videoSender.common.meanRemoteJitter =\n    mean(statsResults[mediaType][sendrecvType].meanRemoteJitter) * 1000 || 0;\n\n  videoSender.common.rtpPackets =\n    statsResults[mediaType][sendrecvType].totalPacketsSent - lastPacketsSent || 0;\n  videoSender.common.availableBitrate =\n    statsResults[mediaType][sendrecvType].availableOutgoingBitrate || 0;\n  // Calculate based on how much packets lost of received compated to how to the client sent\n\n  const totalpacketsLostForaMin =\n    statsResults[mediaType][sendrecvType].totalPacketsLostOnReceiver - lastPacketsLost;\n\n  videoSender.common.remoteLossRate =\n    totalpacketsLostForaMin > 0\n      ? (totalpacketsLostForaMin * 100) / (videoSender.common.rtpPackets + totalpacketsLostForaMin)\n      : 0; // This is the packets sent with in last min || 0;\n\n  videoSender.common.maxRoundTripTime =\n    // @ts-ignore\n    max(statsResults[mediaType][sendrecvType].meanRoundTripTime) * 1000 || 0;\n  videoSender.common.meanRoundTripTime =\n    mean(statsResults[mediaType][sendrecvType].meanRoundTripTime) * 1000 || 0;\n  videoSender.common.roundTripTime = videoSender.common.maxRoundTripTime;\n\n  videoSender.streams[0].common.rtpPackets =\n    statsResults[mediaType][sendrecvType].totalPacketsSent - lastPacketsSent || 0;\n\n  // Calculate the outgoing bitrate\n  const totalBytesSentInaMin = statsResults[mediaType][sendrecvType].totalBytesSent - lastBytesSent;\n\n  videoSender.streams[0].common.transmittedBitrate = totalBytesSentInaMin\n    ? (totalBytesSentInaMin * 8) / 60\n    : 0;\n\n  videoSender.common.rtpBitrate = videoSender.streams[0].common.transmittedBitrate;\n\n  videoSender.streams[0].transmittedKeyFrames =\n    statsResults[mediaType][sendrecvType].totalKeyFramesEncoded - lastKeyFramesEncoded || 0;\n  videoSender.streams[0].requestedKeyFrames =\n    statsResults[mediaType][sendrecvType].totalFirCount - lastFirCount || 0;\n\n  // From tracks //TODO: calculate a proper one\n  const totalFrameSentInaMin =\n    statsResults.resolutions[mediaType][sendrecvType].framesSent - (lastFramesSent || 0);\n\n  videoSender.streams[0].common.transmittedFrameRate = totalFrameSentInaMin\n    ? (totalFrameSentInaMin * 100) / 60\n    : 0;\n  videoSender.streams[0].transmittedHeight =\n    statsResults.resolutions[mediaType][sendrecvType].height || 0;\n  videoSender.streams[0].transmittedWidth =\n    statsResults.resolutions[mediaType][sendrecvType].width || 0;\n  videoSender.streams[0].transmittedFrameSize =\n    (videoSender.streams[0].transmittedHeight * videoSender.streams[0].transmittedWidth) / 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lastPacketsReceived || 0;\n  // Hop by hop are numbers and not percentage so we compare on what we sent the last min\n  // collect the packets received for the last min\n  audioReceiver.streams[0].common.rtpPackets = audioReceiver.common.rtpPackets;\n  audioReceiver.common.mediaHopByHopLost =\n    statsResults[mediaType][sendrecvType].totalPacketsLost - lastPacketsLost || 0;\n  audioReceiver.common.rtpHopByHopLost =\n    statsResults[mediaType][sendrecvType].totalPacketsLost - lastPacketsLost || 0;\n\n  audioReceiver.streams[0].common.maxRtpJitter =\n    // @ts-ignore\n    max(statsResults[mediaType][sendrecvType].meanRtpJitter) * 1000 || 0;\n  audioReceiver.streams[0].common.meanRtpJitter =\n    mean(statsResults[mediaType][sendrecvType].meanRtpJitter) * 1000 || 0;\n  audioReceiver.streams[0].common.rtpJitter = audioReceiver.streams[0].common.maxRtpJitter;\n\n  // Fec packets do come in as part of the FEC only for audio\n  const fecRecovered =\n    statsResults[mediaType][sendrecvType].fecPacketsReceived -\n    lastFecPacketsReceived -\n    (statsResults[mediaType][sendrecvType].fecPacketsDiscarded - lastFecPacketsDiscarded);\n\n  audioReceiver.streams[0].common.rtpEndToEndLost =\n    statsResults[mediaType][sendrecvType].totalPacketsLost - lastPacketsLost - fecRecovered || 0;\n\n  audioReceiver.streams[0].common.framesDropped =\n    statsResults[mediaType][sendrecvType].totalSamplesDecoded - lastPacketsDecoded || 0;\n  audioReceiver.streams[0].common.renderedFrameRate =\n    (audioReceiver.streams[0].common.framesDropped * 100) / 60 || 0;\n\n  audioReceiver.streams[0].common.framesReceived =\n    statsResults[mediaType][sendrecvType].totalSamplesReceived - lastSamplesReceived || 0;\n  audioReceiver.streams[0].common.concealedFrames =\n    statsResults[mediaType][sendrecvType].concealedSamples - lastConcealedSamples || 0;\n  audioReceiver.streams[0].common.receivedBitrate =\n    ((statsResults[mediaType][sendrecvType].totalBytesReceived - lastBytesReceived) * 8) / 60 || 0;\n\n  audioReceiver.common.rtpBitrate = audioReceiver.streams[0].common.receivedBitrate;\n};\n\nexport const getAudioSenderMqa = ({audioSender, statsResults, lastMqaDataSent, mediaType}) => {\n  const sendrecvType = STATS.SEND_DIRECTION;\n\n  const lastPacketsSent = lastMqaDataSent[mediaType]?.[sendrecvType].totalPacketsSent || 0;\n  const lastPacketsLost =\n    lastMqaDataSent[mediaType]?.[sendrecvType].totalPacketsLostOnReceiver || 0;\n  const lastBytesSent = lastMqaDataSent[mediaType]?.[sendrecvType].totalBytesSent || 0;\n  const lastFramesEncoded = lastMqaDataSent[mediaType]?.[sendrecvType].totalKeyFramesEncoded || 0;\n  const lastFirCount = lastMqaDataSent[mediaType]?.[sendrecvType].totalFirCount || 0;\n\n  const {csi} = statsResults[mediaType];\n  if (csi && !audioSender.streams[0].common.csi.includes(csi)) {\n    audioSender.streams[0].common.csi.push(csi);\n  }\n\n  audioSender.common.common.direction = statsResults[mediaType].direction;\n  audioSender.common.transportType = statsResults.connectionType.local.transport;\n\n  audioSender.common.maxRemoteJitter =\n    // @ts-ignore\n    max(statsResults[mediaType][sendrecvType].meanRemoteJitter) * 1000 || 0;\n  audioSender.common.meanRemoteJitter =\n    mean(statsResults[mediaType][sendrecvType].meanRemoteJitter) * 1000 || 0;\n\n  audioSender.common.rtpPackets =\n    statsResults[mediaType][sendrecvType].totalPacketsSent - lastPacketsSent || 0;\n  audioSender.streams[0].common.rtpPackets = audioSender.common.rtpPackets;\n  // From candidate-pair\n  audioSender.common.availableBitrate =\n    statsResults[mediaType][sendrecvType].availableOutgoingBitrate || 0;\n  // Calculate based on how much packets lost of received compated to how to the client sent\n\n  const totalpacketsLostForaMin =\n    statsResults[mediaType][sendrecvType].totalPacketsLostOnReceiver - lastPacketsLost;\n\n  audioSender.common.remoteLossRate =\n    totalpacketsLostForaMin > 0\n      ? (totalpacketsLostForaMin * 100) / audioSender.common.rtpPackets\n      : 0; // This is the packets sent with in last min || 0;\n\n  audioSender.common.maxRoundTripTime =\n    // @ts-ignore\n    max(statsResults[mediaType][sendrecvType].meanRoundTripTime) * 1000 || 0;\n  audioSender.common.meanRoundTripTime =\n    mean(statsResults[mediaType][sendrecvType].meanRoundTripTime) * 1000 || 0;\n  audioSender.common.roundTripTime = audioSender.common.maxRoundTripTime;\n\n  // Calculate the outgoing bitrate\n  const totalBytesSentInaMin = statsResults[mediaType][sendrecvType].totalBytesSent - lastBytesSent;\n\n  audioSender.streams[0].common.transmittedBitrate = totalBytesSentInaMin\n    ? (totalBytesSentInaMin * 8) / 60\n    : 0;\n  audioSender.common.rtpBitrate = audioSender.streams[0].common.transmittedBitrate;\n\n  audioSender.streams[0].transmittedKeyFrames =\n    statsResults[mediaType][sendrecvType].totalKeyFramesEncoded - lastFramesEncoded || 0;\n  audioSender.streams[0].requestedKeyFrames =\n    statsResults[mediaType][sendrecvType].totalFirCount - lastFirCount || 0;\n};\n\nexport const getVideoReceiverMqa = ({videoReceiver, statsResults, lastMqaDataSent, mediaType}) => {\n  const sendrecvType = STATS.RECEIVE_DIRECTION;\n\n  const lastPacketsReceived = lastMqaDataSent[mediaType]?.[sendrecvType].totalPacketsReceived || 0;\n  const lastPacketsLost = lastMqaDataSent[mediaType]?.[sendrecvType].totalPacketsLost || 0;\n  const lastBytesReceived = lastMqaDataSent[mediaType]?.[sendrecvType].totalBytesReceived || 0;\n  const lastFramesReceived = lastMqaDataSent[mediaType]?.[sendrecvType].framesReceived || 0;\n  const lastFramesDecoded = lastMqaDataSent[mediaType]?.[sendrecvType].framesDecoded || 0;\n  const lastFramesDropped = lastMqaDataSent[mediaType]?.[sendrecvType].framesDropped || 0;\n  const lastKeyFramesDecoded = lastMqaDataSent[mediaType]?.[sendrecvType].keyFramesDecoded || 0;\n  const lastPliCount = lastMqaDataSent[mediaType]?.[sendrecvType].totalPliCount || 0;\n\n  const {csi} = statsResults[mediaType];\n  if (csi && !videoReceiver.streams[0].common.csi.includes(csi)) {\n    videoReceiver.streams[0].common.csi.push(csi);\n  }\n\n  videoReceiver.common.common.direction = statsResults[mediaType].direction;\n  videoReceiver.common.transportType = statsResults.connectionType.local.transport;\n\n  // collect the packets received for the last min\n  videoReceiver.common.rtpPackets =\n    statsResults[mediaType][sendrecvType].totalPacketsReceived - lastPacketsReceived || 0;\n  videoReceiver.streams[0].common.rtpPackets = videoReceiver.common.rtpPackets;\n\n  const totalPacketLoss =\n    statsResults[mediaType][sendrecvType].totalPacketsLost - lastPacketsLost || 0;\n\n  // Hope by hop are numbers and not percentage so we compare on what we sent the last min\n  // this is including packet lost\n  videoReceiver.common.mediaHopByHopLost = totalPacketLoss;\n  videoReceiver.common.rtpHopByHopLost = totalPacketLoss;\n\n  // End to end packetloss is after recovery\n  videoReceiver.streams[0].common.rtpEndToEndLost = totalPacketLoss;\n\n  // calculate this values\n\n  videoReceiver.common.maxRemoteJitter =\n    // @ts-ignore\n    max(statsResults[mediaType][sendrecvType].meanRemoteJitter) * 1000 || 0;\n  videoReceiver.common.meanRemoteJitter =\n    mean(statsResults[mediaType][sendrecvType].meanRemoteJitter) * 1000 || 0;\n\n  videoReceiver.streams[0].common.rtpJitter = videoReceiver.common.maxRemoteJitter;\n\n  // Calculate the outgoing bitrate\n  const totalBytesReceivedInaMin =\n    statsResults[mediaType][sendrecvType].totalBytesReceived - lastBytesReceived;\n\n  videoReceiver.streams[0].common.receivedBitrate = totalBytesReceivedInaMin\n    ? (totalBytesReceivedInaMin * 8) / 60\n    : 0;\n  videoReceiver.common.rtpBitrate = videoReceiver.streams[0].common.receivedBitrate;\n\n  // From tracks //TODO: calculate a proper one\n  const totalFrameReceivedInaMin =\n    statsResults[mediaType][sendrecvType].framesReceived - lastFramesReceived;\n  const totalFrameDecodedInaMin =\n    statsResults[mediaType][sendrecvType].framesDecoded - lastFramesDecoded;\n\n  videoReceiver.streams[0].common.receivedFrameRate = Math.round(\n    totalFrameReceivedInaMin ? totalFrameReceivedInaMin / 60 : 0\n  );\n  videoReceiver.streams[0].common.renderedFrameRate = Math.round(\n    totalFrameDecodedInaMin ? totalFrameDecodedInaMin / 60 : 0\n  );\n\n  videoReceiver.streams[0].common.framesDropped =\n    statsResults[mediaType][sendrecvType].framesDropped - lastFramesDropped;\n  videoReceiver.streams[0].receivedHeight = statsResults[mediaType][sendrecvType].height || 0;\n  videoReceiver.streams[0].receivedWidth = statsResults[mediaType][sendrecvType].width || 0;\n  videoReceiver.streams[0].receivedFrameSize =\n    (videoReceiver.streams[0].receivedHeight * videoReceiver.streams[0].receivedWidth) / 256;\n\n  videoReceiver.streams[0].receivedKeyFrames =\n    statsResults[mediaType][sendrecvType].keyFramesDecoded - lastKeyFramesDecoded || 0;\n  videoReceiver.streams[0].requestedKeyFrames =\n    statsResults[mediaType][sendrecvType].totalPliCount - lastPliCount || 0;\n};\n\nexport const getVideoSenderMqa = ({videoSender, statsResults, lastMqaDataSent, mediaType}) => {\n  const sendrecvType = STATS.SEND_DIRECTION;\n\n  const lastPacketsSent = lastMqaDataSent[mediaType]?.[sendrecvType].totalPacketsSent || 0;\n  const lastPacketsLost =\n    lastMqaDataSent[mediaType]?.[sendrecvType].totalPacketsLostOnReceiver || 0;\n  const lastBytesSent = lastMqaDataSent[mediaType]?.[sendrecvType].totalBytesSent || 0;\n  const lastKeyFramesEncoded =\n    lastMqaDataSent[mediaType]?.[sendrecvType].totalKeyFramesEncoded || 0;\n  const lastFirCount = lastMqaDataSent[mediaType]?.[sendrecvType].totalFirCount || 0;\n  const lastFramesSent = lastMqaDataSent[mediaType]?.[sendrecvType].framesSent || 0;\n  const {csi} = statsResults[mediaType];\n  if (csi && !videoSender.streams[0].common.csi.includes(csi)) {\n    videoSender.streams[0].common.csi.push(csi);\n  }\n\n  videoSender.common.common.direction = statsResults[mediaType].direction;\n  videoSender.common.transportType = statsResults.connectionType.local.transport;\n\n  // @ts-ignore\n  videoSender.common.maxRemoteJitter =\n    // @ts-ignore\n    max(statsResults[mediaType][sendrecvType].meanRemoteJitter) * 1000 || 0;\n  videoSender.common.meanRemoteJitter =\n    mean(statsResults[mediaType][sendrecvType].meanRemoteJitter) * 1000 || 0;\n\n  videoSender.common.rtpPackets =\n    statsResults[mediaType][sendrecvType].totalPacketsSent - lastPacketsSent || 0;\n  videoSender.common.availableBitrate =\n    statsResults[mediaType][sendrecvType].availableOutgoingBitrate || 0;\n  // Calculate based on how much packets lost of received compated to how to the client sent\n\n  const totalpacketsLostForaMin =\n    statsResults[mediaType][sendrecvType].totalPacketsLostOnReceiver - lastPacketsLost;\n\n  videoSender.common.remoteLossRate =\n    totalpacketsLostForaMin > 0\n      ? (totalpacketsLostForaMin * 100) / (videoSender.common.rtpPackets + totalpacketsLostForaMin)\n      : 0; // This is the packets sent with in last min || 0;\n\n  videoSender.common.maxRoundTripTime =\n    // @ts-ignore\n    max(statsResults[mediaType][sendrecvType].meanRoundTripTime) * 1000 || 0;\n  videoSender.common.meanRoundTripTime =\n    mean(statsResults[mediaType][sendrecvType].meanRoundTripTime) * 1000 || 0;\n  videoSender.common.roundTripTime = videoSender.common.maxRoundTripTime;\n\n  videoSender.streams[0].common.rtpPackets =\n    statsResults[mediaType][sendrecvType].totalPacketsSent - lastPacketsSent || 0;\n\n  // Calculate the outgoing bitrate\n  const totalBytesSentInaMin = statsResults[mediaType][sendrecvType].totalBytesSent - lastBytesSent;\n\n  videoSender.streams[0].common.transmittedBitrate = totalBytesSentInaMin\n    ? (totalBytesSentInaMin * 8) / 60\n    : 0;\n\n  videoSender.common.rtpBitrate = videoSender.streams[0].common.transmittedBitrate;\n\n  videoSender.streams[0].transmittedKeyFrames =\n    statsResults[mediaType][sendrecvType].totalKeyFramesEncoded - lastKeyFramesEncoded || 0;\n  videoSender.streams[0].requestedKeyFrames =\n    statsResults[mediaType][sendrecvType].totalFirCount - lastFirCount || 0;\n\n  // From tracks //TODO: calculate a proper one\n  const totalFrameSentInaMin =\n    statsResults[mediaType][sendrecvType].framesSent - (lastFramesSent || 0);\n\n  videoSender.streams[0].common.transmittedFrameRate = Math.round(\n    totalFrameSentInaMin ? totalFrameSentInaMin / 60 : 0\n  );\n  videoSender.streams[0].transmittedHeight = statsResults[mediaType][sendrecvType].height || 0;\n  videoSender.streams[0].transmittedWidth = statsResults[mediaType][sendrecvType].width || 0;\n  videoSender.streams[0].transmittedFrameSize =\n    (videoSender.streams[0].transmittedHeight * 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package/dist/transcription/index.d.ts

@@ -0,0 +1,64 @@
+/**
+ * @description Meeting Webex assistance transcription feature.
+ * @exports
+ * @class Transcription
+ */
+export default class Transcription {
+    webSocketUrl: any;
+    sessionID: any;
+    members: any;
+    memberCSIs: any;
+    webSocket: any;
+    /**
+     * @param {string} webSocketUrl
+     * @param {sessionID} sessionId
+     * @param {object} members
+     * @constructor
+     * @memberof Transcription
+     */
+    constructor(webSocketUrl: string, sessionId: any, members: object);
+    /**
+     * open the Low Latency Mercury connection
+     * An event parameter will be send to the callback.
+     *
+     * @param {string} token
+     * @param {string} sessionID
+     * @returns {void}
+     */
+    connect(token: string): void;
+    /**
+     * Sets callback to invoke when the web socket connection is closed.
+     *
+     * @param {Function} callback
+     * @returns {void}
+     */
+    onCloseSocket(callback: any): void;
+    /**
+     * Sets callback to invoke when a web socket connection error occurs.
+     * An event parameter will be send to the callback.
+     *
+     * @param {Function} callback
+     * @returns {void}
+     */
+    onErrorSocket(callback: any): void;
+    /**
+     * Get current speaker from the given csis payload and update the global csis map
+     *
+     * @private
+     * @param {object} csis
+     * @returns {object}
+     */
+    private getSpeaker;
+    /**
+     * Sends transcription data to given callback as it arrives.
+     *
+     * @param {Function} callback
+     * @returns {void}
+     */
+    subscribe(callback: any): void;
+    /**
+     * Close the LLM web socket connection
+     * @returns {void}
+     */
+    closeSocket(): void;
+}

package/dist/webinar/collection.d.ts

@@ -0,0 +1,16 @@
+/*!
+ * Copyright (c) 2015-2023 Cisco Systems, Inc. See LICENSE file.
+ */
+declare class WebinarCollection {
+    webinarInfo: any;
+    namespace: string;
+    mainIndex: string;
+    constructor();
+    set(id: any, info: any): void;
+    /**
+     * @param {String} id
+     * @returns {Member}
+     */
+    get(id: string): any;
+}
+export default WebinarCollection;

package/dist/webinar/index.d.ts

@@ -0,0 +1,5 @@
+/**
+ * @class Webinar
+ */
+declare const Webinar: any;
+export default Webinar;

package/dist/webinar/index.js

@@ -62,7 +62,7 @@ var Webinar = _webexCore.WebexPlugin.extend({
   updateCanManageWebcast: function updateCanManageWebcast(canManageWebcast) {
     this.set('canManageWebcast', canManageWebcast);
   },
-  version: "2.60.0-next.1"
+  version: "2.60.0-next.11"
 });
 var _default = exports.default = Webinar;
 //# sourceMappingURL=index.js.map

package/package.json

@@ -29,7 +29,7 @@
   },
   "scripts": {
     "build": "yarn run -T tsc --declaration true --declarationDir ./dist",
-    "build:src": "webex-legacy-tools build -dest \"./dist\" -src \"./src\" -js -ts -maps",
+    "build:src": "webex-legacy-tools build -dest \"./dist\" -src \"./src\" -js -ts -maps && yarn build",
     "deploy:npm": "yarn npm publish",
     "test:broken": "yarn test:style && yarn test:unit && yarn test:integration && yarn test:browser",
     "test:browser:broken": "webex-legacy-tools test --integration --unit --runner karma",
@@ -40,17 +40,17 @@
   "devDependencies": {
     "@babel/core": "^7.17.10",
     "@types/jsdom": "^21",
-    "@webex/babel-config-legacy": "^0.0.0",
-    "@webex/eslint-config-legacy": "^0.0.0",
-    "@webex/jest-config-legacy": "^0.0.0",
-    "@webex/legacy-tools": "^0.0.0",
-    "@webex/plugin-meetings": "^2.60.0-next.1",
-    "@webex/plugin-rooms": "^2.60.0-next.1",
-    "@webex/test-helper-chai": "^2.60.0-next.1",
-    "@webex/test-helper-mocha": "^2.60.0-next.1",
-    "@webex/test-helper-mock-webex": "^2.60.0-next.1",
-    "@webex/test-helper-retry": "^2.60.0-next.1",
-    "@webex/test-helper-test-users": "^2.60.0-next.1",
+    "@webex/babel-config-legacy": "0.0.0",
+    "@webex/eslint-config-legacy": "0.0.0",
+    "@webex/jest-config-legacy": "0.0.0",
+    "@webex/legacy-tools": "0.0.0",
+    "@webex/plugin-meetings": "2.60.0-next.11",
+    "@webex/plugin-rooms": "2.60.0-next.8",
+    "@webex/test-helper-chai": "2.60.0-next.8",
+    "@webex/test-helper-mocha": "2.60.0-next.8",
+    "@webex/test-helper-mock-webex": "2.60.0-next.8",
+    "@webex/test-helper-retry": "2.60.0-next.8",
+    "@webex/test-helper-test-users": "2.60.0-next.8",
     "chai": "^4.3.4",
     "chai-as-promised": "^7.1.1",
     "eslint": "^8.24.0",
@@ -62,20 +62,19 @@
     "typescript": "^4.7.4"
   },
   "dependencies": {
-    "@webex/common": "^2.60.0-next.1",
-    "@webex/internal-media-core": "2.2.1",
-    "@webex/internal-plugin-conversation": "^2.60.0-next.1",
-    "@webex/internal-plugin-device": "^2.60.0-next.1",
-    "@webex/internal-plugin-llm": "^2.60.0-next.1",
-    "@webex/internal-plugin-mercury": "^2.60.0-next.1",
-    "@webex/internal-plugin-metrics": "^2.60.0-next.1",
-    "@webex/internal-plugin-support": "^2.60.0-next.1",
-    "@webex/internal-plugin-user": "^2.60.0-next.1",
-    "@webex/media-helpers": "^3.0.0-next.11",
-    "@webex/plugin-people": "^2.60.0-next.1",
-    "@webex/plugin-rooms": "^2.60.0-next.1",
-    "@webex/ts-sdp": "1.0.1",
-    "@webex/webex-core": "^2.60.0-next.1",
+    "@webex/common": "2.60.0-next.8",
+    "@webex/internal-media-core": "2.2.3",
+    "@webex/internal-plugin-conversation": "2.60.0-next.8",
+    "@webex/internal-plugin-device": "2.60.0-next.8",
+    "@webex/internal-plugin-llm": "2.60.0-next.8",
+    "@webex/internal-plugin-mercury": "2.60.0-next.8",
+    "@webex/internal-plugin-metrics": "2.60.0-next.8",
+    "@webex/internal-plugin-support": "2.60.0-next.8",
+    "@webex/internal-plugin-user": "2.60.0-next.8",
+    "@webex/media-helpers": "3.0.0-next.19",
+    "@webex/plugin-people": "2.60.0-next.8",
+    "@webex/plugin-rooms": "2.60.0-next.8",
+    "@webex/webex-core": "2.60.0-next.8",
     "ampersand-collection": "^2.0.2",
     "bowser": "^2.11.0",
     "btoa": "^1.2.1",
@@ -89,5 +88,5 @@
     "uuid": "^3.3.2",
     "webrtc-adapter": "^8.1.2"
   },
-  "version": "2.60.0-next.1"
+  "version": "2.60.0-next.11"
 }

package/src/constants.ts

@@ -1,6 +1,8 @@
 // @ts-ignore
 import {hydraTypes} from '@webex/common';
 
+type Enum<T extends Record<string, unknown>> = T[keyof T];
+
 // *********** LOWERCASE / CAMELCASE STRINGS ************
 
 export const AUDIO = 'audio';
@@ -193,7 +195,7 @@ export const ICE_FAIL_TIMEOUT = 3000;
 export const RETRY_TIMEOUT = 3000;
 
 export const ICE_AND_DTLS_CONNECTION_TIMEOUT = 10000;
-export const ROAP_OFFER_ANSWER_EXCHANGE_TIMEOUT = 10000;
+export const ROAP_OFFER_ANSWER_EXCHANGE_TIMEOUT = 35000;
 
 // ******************** REGEX **********************
 // Please alphabetize
@@ -1036,7 +1038,9 @@ export const RECONNECTION = {
     DEFAULT_TRY_COUNT: 0,
     DEFAULT_STATUS: '',
   },
-};
+} as const;
+
+export type RECONNECTION_STATE = Enum<typeof RECONNECTION.STATE>;
 
 export const RESOURCE = {
   CLUSTERS: 'clusters',
@@ -1072,7 +1076,9 @@ export const NETWORK_STATUS = {
   DISCONNECTED: 'DISCONNECTED',
   RECONNECTING: 'RECONNECTING',
   CONNECTED: 'CONNECTED',
-};
+} as const;
+
+export type NETWORK_STATUS = Enum<typeof NETWORK_STATUS>;
 
 export const NETWORK_TYPE = {
   VPN: 'vpn',
@@ -1288,7 +1294,7 @@ export const IP_VERSION = {
   ipv4_and_ipv6: 1,
 } as const;
 
-export type IP_VERSION = (typeof IP_VERSION)[keyof typeof IP_VERSION];
+export type IP_VERSION = Enum<typeof IP_VERSION>;
 
 // constant for if the permissionToken is about to expire in the next 30 seconds, refresh it
 export const MEETING_PERMISSION_TOKEN_REFRESH_THRESHOLD_IN_SEC = 30;

package/src/controls-options-manager/enums.ts

@@ -14,3 +14,5 @@ enum Control {
 }
 
 export {Control, Setting};
+
+export default Setting;

package/src/locus-info/parser.ts

@@ -53,7 +53,7 @@ export default class Parser {
 
   queue: SortedQueue<LocusDeltaDto>;
   workingCopy: any;
-  syncTimer: null | number | NodeJS.Timeout;
+  syncTimer?: ReturnType<typeof setTimeout>;
 
   /**
    * @constructs Parser
@@ -85,7 +85,7 @@ export default class Parser {
     this.status = 'IDLE';
     this.onDeltaAction = null;
     this.workingCopy = null;
-    this.syncTimer = null;
+    this.syncTimer = undefined;
   }
 
   /**
@@ -673,11 +673,11 @@ export default class Parser {
    * @returns {undefined}
    */
   private startSyncTimer() {
-    if (this.syncTimer === null) {
+    if (this.syncTimer === undefined) {
       const timeout = OOO_DELTA_WAIT_TIME + Math.random() * OOO_DELTA_WAIT_TIME_RANDOM_DELAY;
 
       this.syncTimer = setTimeout(() => {
-        this.syncTimer = null;
+        this.syncTimer = undefined;
         this.triggerSync('timer expired, blocked on out-of-order delta');
       }, timeout);
     }
@@ -689,9 +689,9 @@ export default class Parser {
    * @returns {undefined}
    */
   private stopSyncTimer() {
-    if (this.syncTimer !== null) {
+    if (this.syncTimer !== undefined) {
       clearTimeout(this.syncTimer);
-      this.syncTimer = null;
+      this.syncTimer = undefined;
     }
   }
 

package/src/media/index.ts

@@ -217,10 +217,10 @@ Media.createMediaConnection = (
     {
       // TODO: RoapMediaConnection is not ready to use stream classes yet, so we pass the raw MediaStreamTrack for now SPARK-460530
       localTracks: {
-        audio: audioStream?.outputTrack,
-        video: videoStream?.outputTrack,
-        screenShareVideo: shareVideoStream?.outputTrack,
-        screenShareAudio: shareAudioStream?.outputTrack, // TODO: add type for screenShareAudio in internal-media-core SPARK-446923
+        audio: audioStream?.outputStream?.getTracks()[0],
+        video: videoStream?.outputStream?.getTracks()[0],
+        screenShareVideo: shareVideoStream?.outputStream?.getTracks()[0],
+        screenShareAudio: shareAudioStream?.outputStream?.getTracks()[0], // TODO: add type for screenShareAudio in internal-media-core SPARK-446923
       } as unknown,
       direction: {
         audio: Media.getDirection(true, mediaDirection.receiveAudio, mediaDirection.sendAudio),
@@ -399,7 +399,7 @@ Media.stopStream = (stream: LocalStream) => {
       stream.stop();
     } catch (e) {
       LoggerProxy.logger.error(
-        `Media:index#stopStream --> Unable to stop the stream with ready state of the output track => ${stream.outputTrack.readyState} & input track => ${stream.inputTrack.readyState}, error: ${e}`
+        `Media:index#stopStream --> Unable to stop the stream with ready state => ${stream.readyState}, error: ${e}`
       );
     }
   });

package/src/mediaQualityMetrics/config.ts

@@ -247,138 +247,3 @@ export const emptyVideoTransmit = {
     },
   ],
 };
-
-export const emptyVideoShareReceive = {
-  common: {
-    common: {
-      direction: 'inactive',
-      isMain: false, // Not avaliable
-      mariFecEnabled: true, // Not avaliable
-      mariQosEnabled: true, // Not avaliable
-      multistreamEnabled: true, // Not avaliable
-    },
-    dtlsBitrate: 0, // Not avaliable
-    dtlsPackets: 0, // Not avaliable
-    fecBitrate: 0, // Not avaliable
-    fecPackets: 0, // Not avaliable
-    maxBitrate: 0,
-    mediaHopByHopLost: 0,
-    rtcpBitrate: 0, // Not avaliable
-    rtcpPackets: 0, // Not avaliable
-    rtpBitrate: 0,
-    rtpHopByHopLost: 0,
-    rtpPackets: 0,
-    rtpRecovered: 0, // Not avaliable
-    rtxBitrate: 0, // Not avaliable
-    rtxPackets: 0, // Not avaliable
-    srtcpUnprotectErrors: 0, // Not avaliable
-    srtpUnprotectErrors: 0, // Not avaliable
-    stunBitrate: 0, // Not avaliable
-    stunPackets: 0, // Not avaliable
-    transportType: 'UDP',
-  },
-  streams: [
-    {
-      common: {
-        codec: 'H264',
-        concealedFrames: 0, // Not avaliable
-        csi: [],
-        maxConcealRunLength: 0, // Not avaliable
-        optimalBitrate: 0,
-        optimalFrameRate: 0,
-        receivedBitrate: 0,
-        receivedFrameRate: 0,
-        renderedFrameRate: 0, // Not avaliable
-        requestedBitrate: 0,
-        requestedFrameRate: 0,
-        rtpEndToEndLost: 0,
-        rtpJitter: 0,
-        rtpPackets: 0,
-        ssci: 0, // Not avaliable
-      },
-      h264CodecProfile: 'BP',
-      isActiveSpeaker: true,
-      optimalFrameSize: 0, // Not avaliable
-      receivedFrameSize: 0,
-      receivedHeight: 0,
-      receivedKeyFrames: 0,
-      receivedKeyFramesForRequest: 0,
-      receivedKeyFramesSourceChange: 0,
-      receivedKeyFramesUnknown: 0,
-      receivedWidth: 0,
-      requestedFrameSize: 0,
-      requestedKeyFrames: 0,
-    },
-  ],
-};
-
-export const emptyVideoShareTransmit = {
-  // sceen sharing
-  common: {
-    availableBitrate: 0, // Not avaliable currently hardcoded
-    common: {
-      direction: 'inactive',
-      isMain: false,
-      mariFecEnabled: false, // Not avaliable
-      mariQosEnabled: false, // Not avaliable
-      multistreamEnabled: false, // Not avaliable
-    },
-    dtlsBitrate: 0, // Not avaliable
-    dtlsPackets: 0, // Not avaliable
-    fecBitrate: 0, // Not avaliable
-    fecPackets: 0, // TODO: check inbound-rtp// Not avaliable
-    maxBitrate: 0, // Currently hardcoded
-    queueDelay: 0, // outboundRtp.totalPacketSentDelay  // TODO: check if totalInterFrameDelay/ packetSentDelay/ jitterBufferDalay
-    remoteJitter: 0, // remoteInboundRtp.jitter
-    remoteLossRate: 0, // comparedResults.lossRate
-    remoteReceiveRate: 0, // compareResults.packetsLost
-    roundTripTime: 0, // compareResults.roundTripTime
-    rtcpBitrate: 0, // Dont have access to it
-    rtcpPackets: 0, // Dont have access to rtcp
-    rtpBitrate: 0, // Dont have access
-    rtpPackets: 0, // outboundRtp.packetsSent
-    rtxBitrate: 0, // Dont have access to it
-    rtxPackets: 0, // Dont have access to it
-    stunBitrate: 0, // Dont have access to it
-    stunPackets: 0, // Dont have access to it
-    transportType: 'UDP', // TODO: need to calculate
-  },
-  streams: [
-    {
-      common: {
-        codec: 'H264',
-        csi: [],
-        duplicateSsci: 0, // Not Avaliable
-        requestedBitrate: 0, // TODO: from remote SDP
-        requestedFrames: 0, // TODO: from remote SDP
-        rtpPackets: 0, // same as rtp packets
-        ssci: 0,
-        transmittedBitrate: 0, // TODO: get in the candidate pair
-        transmittedFrameRate: 0, // TODO: from track info
-      },
-      h264CodecProfile: 'BP', // TODO: from localSDP
-      isAvatar: false, // Not Avaliable
-      isHardwareEncoded: false, // Not Avaliable
-      localConfigurationChanges: 2, // Not Avaliable
-      maxFrameQp: 0, // Not Avaliable
-      maxNoiseLevel: 0, // Not Avaliable
-      minRegionQp: 0, // Not Avaliable
-      remoteConfigurationChanges: 0, // Not Avaliable
-      requestedFrameSize: 0, // TODO: from remote SDP
-      requestedKeyFrames: 0, // outbound Fir request
-      transmittedFrameSize: 0, // Not Avaliable
-      transmittedHeight: 0,
-      transmittedKeyFrames: 0, // Key frames encoded
-      transmittedKeyFramesClient: 0, // Not Avaliable
-      transmittedKeyFramesConfigurationChange: 0, // Not Avaliable
-      transmittedKeyFramesFeedback: 0, // Not Avaliable
-      transmittedKeyFramesLocalDrop: 0, // Not Avaliable
-      transmittedKeyFramesOtherLayer: 0, // Not Avaliable
-      transmittedKeyFramesPeriodic: 0, // Not Avaliable
-      transmittedKeyFramesSceneChange: 0, // Not Avaliable
-      transmittedKeyFramesStartup: 0, // Not Avaliable
-      transmittedKeyFramesUnknown: 0, // Not Avaliable
-      transmittedWidth: 0,
-    },
-  ],
-};

package/src/meeting/in-meeting-actions.ts

@@ -71,12 +71,14 @@ interface IInMeetingActions {
   canShareDesktop?: boolean;
   canShareContent?: boolean;
   canTransferFile?: boolean;
+  canChat?: boolean;
   canDoVideo?: boolean;
   canAnnotate?: boolean;
   canUseVoip?: boolean;
   supportHQV?: boolean;
   supportHDV?: boolean;
   canShareWhiteBoard?: boolean;
+  enforceVirtualBackground?: boolean;
 }
 
 /**
@@ -209,6 +211,8 @@ export default class InMeetingActions implements IInMeetingActions {
 
   canTransferFile = null;
 
+  canChat = null;
+
   canDoVideo = null;
 
   canAnnotate = null;
@@ -217,6 +221,8 @@ export default class InMeetingActions implements IInMeetingActions {
 
   supportHQV = null;
 
+  enforceVirtualBackground = null;
+
   supportHDV = null;
 
   canShareWhiteBoard = null;
@@ -288,9 +294,11 @@ export default class InMeetingActions implements IInMeetingActions {
     canShareDesktop: this.canShareDesktop,
     canShareContent: this.canShareContent,
     canTransferFile: this.canTransferFile,
+    canChat: this.canChat,
     canDoVideo: this.canDoVideo,
     canAnnotate: this.canAnnotate,
     canUseVoip: this.canUseVoip,
+    enforceVirtualBackground: this.enforceVirtualBackground,
     supportHQV: this.supportHQV,
     supportHDV: this.supportHDV,
     canShareWhiteBoard: this.canShareWhiteBoard,