@webex/plugin-meetings 3.0.0-beta.276 → 3.0.0-beta.278

package/dist/statsAnalyzer/mqaUtil.js.map

@@ -1 +1 @@
-{"version":3,"names":["getAudioReceiverMqa","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","remote","transport","rtpPackets","mediaHopByHopLost","rtpHopByHopLost","maxRtpJitter","meanRtpJitter","rtpJitter","fecRecovered","rtpEndToEndLost","framesDropped","renderedFrameRate","framesReceived","concealedFrames","receivedBitrate","rtpBitrate","getAudioSenderMqa","audioSender","SEND_DIRECTION","lastPacketsSent","totalPacketsSent","totalPacketsLostOnReceiver","lastBytesSent","totalBytesSent","lastFramesEncoded","totalKeyFramesEncoded","lastFirCount","totalFirCount","local","maxRemoteJitter","meanRemoteJitter","availableBitrate","availableOutgoingBitrate","totalpacketsLostForaMin","remoteLossRate","maxRoundTripTime","meanRoundTripTime","roundTripTime","totalBytesSentInaMin","transmittedBitrate","transmittedKeyFrames","requestedKeyFrames","getVideoReceiverMqa","videoReceiver","lastFramesReceived","lastFramesDecoded","framesDecoded","lastFramesDropped","lastKeyFramesDecoded","keyFramesDecoded","lastPliCount","totalPliCount","totalPacketLoss","totalBytesReceivedInaMin","totalFrameReceivedInaMin","resolutions","totalFrameDecodedInaMin","receivedFrameRate","receivedHeight","height","receivedWidth","width","receivedFrameSize","receivedKeyFrames","getVideoSenderMqa","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.remote.transport[0];\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[0];\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.remote.transport[0];\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[0];\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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+{"version":3,"names":["getAudioReceiverMqa","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","meanRtpJitter","rtpJitter","fecRecovered","rtpEndToEndLost","framesDropped","renderedFrameRate","framesReceived","concealedFrames","receivedBitrate","rtpBitrate","getAudioSenderMqa","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","videoReceiver","lastFramesReceived","lastFramesDecoded","framesDecoded","lastFramesDropped","lastKeyFramesDecoded","keyFramesDecoded","lastPliCount","totalPliCount","totalPacketLoss","totalBytesReceivedInaMin","totalFrameReceivedInaMin","resolutions","totalFrameDecodedInaMin","receivedFrameRate","receivedHeight","height","receivedWidth","width","receivedFrameSize","receivedKeyFrames","getVideoSenderMqa","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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package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@webex/plugin-meetings",
-  "version": "3.0.0-beta.276",
+  "version": "3.0.0-beta.278",
   "description": "",
   "license": "Cisco EULA (https://www.cisco.com/c/en/us/products/end-user-license-agreement.html)",
   "contributors": [
@@ -32,12 +32,12 @@
     "build": "yarn run -T tsc --declaration true --declarationDir ./dist/types"
   },
   "devDependencies": {
-    "@webex/plugin-meetings": "3.0.0-beta.276",
-    "@webex/test-helper-chai": "3.0.0-beta.276",
-    "@webex/test-helper-mocha": "3.0.0-beta.276",
-    "@webex/test-helper-mock-webex": "3.0.0-beta.276",
-    "@webex/test-helper-retry": "3.0.0-beta.276",
-    "@webex/test-helper-test-users": "3.0.0-beta.276",
+    "@webex/plugin-meetings": "3.0.0-beta.278",
+    "@webex/test-helper-chai": "3.0.0-beta.278",
+    "@webex/test-helper-mocha": "3.0.0-beta.278",
+    "@webex/test-helper-mock-webex": "3.0.0-beta.278",
+    "@webex/test-helper-retry": "3.0.0-beta.278",
+    "@webex/test-helper-test-users": "3.0.0-beta.278",
     "chai": "^4.3.4",
     "chai-as-promised": "^7.1.1",
     "jsdom-global": "3.0.2",
@@ -46,19 +46,19 @@
     "typescript": "^4.7.4"
   },
   "dependencies": {
-    "@webex/common": "3.0.0-beta.276",
+    "@webex/common": "3.0.0-beta.278",
     "@webex/internal-media-core": "2.0.6",
-    "@webex/internal-plugin-conversation": "3.0.0-beta.276",
-    "@webex/internal-plugin-device": "3.0.0-beta.276",
-    "@webex/internal-plugin-llm": "3.0.0-beta.276",
-    "@webex/internal-plugin-mercury": "3.0.0-beta.276",
-    "@webex/internal-plugin-metrics": "3.0.0-beta.276",
-    "@webex/internal-plugin-support": "3.0.0-beta.276",
-    "@webex/internal-plugin-user": "3.0.0-beta.276",
-    "@webex/media-helpers": "3.0.0-beta.276",
-    "@webex/plugin-people": "3.0.0-beta.276",
-    "@webex/plugin-rooms": "3.0.0-beta.276",
-    "@webex/webex-core": "3.0.0-beta.276",
+    "@webex/internal-plugin-conversation": "3.0.0-beta.278",
+    "@webex/internal-plugin-device": "3.0.0-beta.278",
+    "@webex/internal-plugin-llm": "3.0.0-beta.278",
+    "@webex/internal-plugin-mercury": "3.0.0-beta.278",
+    "@webex/internal-plugin-metrics": "3.0.0-beta.278",
+    "@webex/internal-plugin-support": "3.0.0-beta.278",
+    "@webex/internal-plugin-user": "3.0.0-beta.278",
+    "@webex/media-helpers": "3.0.0-beta.278",
+    "@webex/plugin-people": "3.0.0-beta.278",
+    "@webex/plugin-rooms": "3.0.0-beta.278",
+    "@webex/webex-core": "3.0.0-beta.278",
     "ampersand-collection": "^2.0.2",
     "bowser": "^2.11.0",
     "btoa": "^1.2.1",

package/src/meeting/index.ts

@@ -5542,8 +5542,8 @@ export default class Meeting extends StatelessWebexPlugin {
           this.mediaProperties.mediaDirection.receiveShare,
       ];
 
-      this.sendSlotManager.createSlot(mc, MediaType.VideoMain, audioEnabled);
-      this.sendSlotManager.createSlot(mc, MediaType.AudioMain, videoEnabled);
+      this.sendSlotManager.createSlot(mc, MediaType.VideoMain, videoEnabled);
+      this.sendSlotManager.createSlot(mc, MediaType.AudioMain, audioEnabled);
       this.sendSlotManager.createSlot(mc, MediaType.VideoSlides, shareEnabled);
       this.sendSlotManager.createSlot(mc, MediaType.AudioSlides, shareEnabled);
     }

package/src/statsAnalyzer/index.ts

@@ -193,7 +193,7 @@ export class StatsAnalyzer extends EventsScope {
       }
     });
 
-    newMqa.intervalMetadata.peerReflexiveIP = this.statsResults.connectionType.local.ipAddress[0];
+    newMqa.intervalMetadata.peerReflexiveIP = this.statsResults.connectionType.local.ipAddress;
 
     // Adding peripheral information
     newMqa.intervalMetadata.peripherals = [];
@@ -1094,73 +1094,35 @@ export class StatsAnalyzer extends EventsScope {
     if (!result || !result.id) {
       return;
     }
-    const RemoteCandidateType = {};
-    const RemoteTransport = {};
-    const RemoteIpAddress = {};
-    const RemoteNetworkType = {};
 
-    if (!result.id) return;
+    let transport;
+    if (result.relayProtocol) {
+      transport = result.relayProtocol.toUpperCase();
+    } else if (result.protocol) {
+      transport = result.protocol.toUpperCase();
+    }
 
     const sendRecvType = isSender ? STATS.SEND_DIRECTION : STATS.RECEIVE_DIRECTION;
     const ipType = isRemote ? STATS.REMOTE : STATS.LOCAL;
 
-    if (!RemoteCandidateType[result.id]) {
-      RemoteCandidateType[result.id] = [];
-    }
-
-    if (!RemoteTransport[result.id]) {
-      RemoteTransport[result.id] = [];
-    }
-
-    if (!RemoteIpAddress[result.id]) {
-      RemoteIpAddress[result.id] = [];
-    }
-    if (!RemoteNetworkType[result.id]) {
-      RemoteNetworkType[result.id] = [];
-    }
-
-    if (
-      result.candidateType &&
-      RemoteCandidateType[result.id].indexOf(result.candidateType) === -1
-    ) {
-      RemoteCandidateType[result.id].push(result.candidateType);
-    }
-
-    if (result.protocol && RemoteTransport[result.id].indexOf(result.protocol) === -1) {
-      RemoteTransport[result.id].push(result.protocol.toUpperCase());
-    }
-
-    if (
-      result.ip &&
-      RemoteIpAddress[result.id].indexOf(`${result.ip}:${result.portNumber}`) === -1
-    ) {
-      RemoteIpAddress[result.id].push(`${result.ip}`); // TODO: Add ports
-    }
-
-    if (result.networkType && RemoteNetworkType[result.id].indexOf(result.networkType) === -1) {
-      RemoteNetworkType[result.id].push(result.networkType);
-    }
-
     this.statsResults.internal.candidates[result.id] = {
-      candidateType: RemoteCandidateType[result.id],
-      ipAddress: RemoteIpAddress[result.id],
+      candidateType: result.candidateType,
+      ipAddress: result.ip, // TODO: add ports
       portNumber: result.port,
-      networkType: RemoteNetworkType[result.id],
+      networkType: result.networkType,
       priority: result.priority,
-      transport: RemoteTransport[result.id],
+      transport,
       timestamp: result.time,
       id: result.id,
       type: result.type,
     };
 
-    this.statsResults.connectionType[ipType].candidateType = RemoteCandidateType[result.id];
-    this.statsResults.connectionType[ipType].ipAddress = RemoteIpAddress[result.id];
+    this.statsResults.connectionType[ipType].candidateType = result.candidateType;
+    this.statsResults.connectionType[ipType].ipAddress = result.ipAddress;
 
     this.statsResults.connectionType[ipType].networkType =
-      RemoteNetworkType[result.id][0] === NETWORK_TYPE.VPN
-        ? NETWORK_TYPE.UNKNOWN
-        : RemoteNetworkType[result.id][0];
-    this.statsResults.connectionType[ipType].transport = RemoteTransport[result.id];
+      result.networkType === NETWORK_TYPE.VPN ? NETWORK_TYPE.UNKNOWN : result.networkType;
+    this.statsResults.connectionType[ipType].transport = transport;
 
     this.statsResults[type][sendRecvType].totalRoundTripTime = result.totalRoundTripTime;
   };

package/src/statsAnalyzer/mqaUtil.ts

@@ -23,7 +23,7 @@ export const getAudioReceiverMqa = ({audioReceiver, statsResults, lastMqaDataSen
   }
 
   audioReceiver.common.common.direction = statsResults[mediaType].direction;
-  audioReceiver.common.transportType = statsResults.connectionType.remote.transport[0];
+  audioReceiver.common.transportType = statsResults.connectionType.local.transport;
 
   // add rtpPacket info inside common as also for call analyzer
   audioReceiver.common.rtpPackets =
@@ -83,7 +83,7 @@ export const getAudioSenderMqa = ({audioSender, statsResults, lastMqaDataSent, m
   }
 
   audioSender.common.common.direction = statsResults[mediaType].direction;
-  audioSender.common.transportType = statsResults.connectionType.local.transport[0];
+  audioSender.common.transportType = statsResults.connectionType.local.transport;
 
   audioSender.common.maxRemoteJitter =
     // @ts-ignore
@@ -146,7 +146,8 @@ export const getVideoReceiverMqa = ({videoReceiver, statsResults, lastMqaDataSen
   }
 
   videoReceiver.common.common.direction = statsResults[mediaType].direction;
-  videoReceiver.common.transportType = statsResults.connectionType.remote.transport[0];
+  videoReceiver.common.transportType = statsResults.connectionType.local.transport;
+
   // collect the packets received for the last min
   videoReceiver.common.rtpPackets =
     statsResults[mediaType][sendrecvType].totalPacketsReceived - lastPacketsReceived || 0;
@@ -228,7 +229,7 @@ export const getVideoSenderMqa = ({videoSender, statsResults, lastMqaDataSent, m
   }
 
   videoSender.common.common.direction = statsResults[mediaType].direction;
-  videoSender.common.transportType = statsResults.connectionType.local.transport[0];
+  videoSender.common.transportType = statsResults.connectionType.local.transport;
 
   // @ts-ignore
   videoSender.common.maxRemoteJitter =

package/test/unit/spec/meeting/index.js

@@ -2081,6 +2081,11 @@ describe('plugin-meetings', () => {
               iceServers,
             }, meetingId);
 
+            assert.calledWith(fakeMultistreamRoapMediaConnection.createSendSlot, MediaType.AudioMain, direction.audio !== 'inactive');
+            assert.calledWith(fakeMultistreamRoapMediaConnection.createSendSlot, MediaType.VideoMain, direction.video !== 'inactive');
+            assert.calledWith(fakeMultistreamRoapMediaConnection.createSendSlot, MediaType.AudioSlides, direction.screenShare !== 'inactive');
+            assert.calledWith(fakeMultistreamRoapMediaConnection.createSendSlot, MediaType.VideoSlides, direction.screenShare !== 'inactive');
+
             for(let type in localStreams){
               const stream = localStreams[type];
               if(stream !== undefined){
@@ -2109,7 +2114,7 @@ describe('plugin-meetings', () => {
                   screenShareVideo: localStreams.screenShareVideo?.outputTrack,
                   screenShareAudio: localStreams.screenShareAudio?.outputTrack,
                 },
-                direction,
+                direction: {audio: direction.audio, video: direction.video, screenShareVideo: direction.screenShare},
                 remoteQualityLevel,
               },
               expectedDebugId
@@ -2133,7 +2138,7 @@ describe('plugin-meetings', () => {
             direction: {
               audio: 'sendrecv',
               video: 'sendrecv',
-              screenShareVideo: 'recvonly',
+              screenShare: 'recvonly',
             },
             remoteQualityLevel: 'HIGH',
             expectedDebugId,
@@ -2163,7 +2168,7 @@ describe('plugin-meetings', () => {
             direction: {
               audio: 'sendrecv',
               video: 'sendrecv',
-              screenShareVideo: 'recvonly',
+              screenShare: 'recvonly',
             },
             remoteQualityLevel: 'HIGH',
             expectedDebugId,
@@ -2195,7 +2200,7 @@ describe('plugin-meetings', () => {
             direction: {
               audio: 'sendrecv',
               video: 'sendrecv',
-              screenShareVideo: 'recvonly',
+              screenShare: 'recvonly',
             },
             remoteQualityLevel: 'HIGH',
             expectedDebugId,
@@ -2224,7 +2229,7 @@ describe('plugin-meetings', () => {
             direction: {
               audio: 'inactive',
               video: 'sendrecv',
-              screenShareVideo: 'recvonly',
+              screenShare: 'recvonly',
             },
             remoteQualityLevel: 'HIGH',
             expectedDebugId,
@@ -2254,7 +2259,68 @@ describe('plugin-meetings', () => {
             direction: {
               audio: 'inactive',
               video: 'sendrecv',
-              screenShareVideo: 'recvonly',
+              screenShare: 'recvonly',
+            },
+            remoteQualityLevel: 'HIGH',
+            expectedDebugId,
+            meetingId: meeting.id
+          });
+
+          // and SDP offer was sent with the right audioMuted/videoMuted values
+          checkSdpOfferSent({audioMuted: true, videoMuted: true});
+
+          // and no other local mute requests were sent to Locus
+          assert.calledOnce(locusMediaRequestStub);
+        });
+
+
+        it('addMedia() works correctly when video is disabled with no tracks to publish', async () => {
+          await meeting.addMedia({videoEnabled: false});
+          await simulateRoapOffer();
+
+          // check RoapMediaConnection was created correctly
+          checkMediaConnectionCreated({
+            mediaConnectionConfig: expectedMediaConnectionConfig,
+            localStreams: {
+              audio: undefined,
+              video: undefined,
+              screenShareVideo: undefined,
+              screenShareAudio: undefined,
+            },
+            direction: {
+              audio: 'sendrecv',
+              video: 'inactive',
+              screenShare: 'recvonly',
+            },
+            remoteQualityLevel: 'HIGH',
+            expectedDebugId,
+            meetingId: meeting.id
+          });
+
+          // and SDP offer was sent with the right audioMuted/videoMuted values
+          checkSdpOfferSent({audioMuted: true, videoMuted: true});
+
+          // and no other local mute requests were sent to Locus
+          assert.calledOnce(locusMediaRequestStub);
+        });
+
+        it('addMedia() works correctly when screen share is disabled with no tracks to publish', async () => {
+          await meeting.addMedia({shareAudioEnabled: false, shareVideoEnabled: false});
+          await simulateRoapOffer();
+
+          // check RoapMediaConnection was created correctly
+          checkMediaConnectionCreated({
+            mediaConnectionConfig: expectedMediaConnectionConfig,
+            localStreams: {
+              audio: undefined,
+              video: undefined,
+              screenShareVideo: undefined,
+              screenShareAudio: undefined,
+            },
+            direction: {
+              audio: 'sendrecv',
+              video: 'sendrecv',
+              screenShare: 'inactive',
             },
             remoteQualityLevel: 'HIGH',
             expectedDebugId,

package/test/unit/spec/stats-analyzer/index.js

@@ -125,6 +125,11 @@ describe('plugin-meetings', () => {
                     packetsSent: 0,
                     bytesSent: 1,
                   },
+                  {
+                    type: 'local-candidate',
+                    id: 'fake-candidate-id',
+                    protocol: 'tcp'
+                  },
                 ],
               },
             ],
@@ -136,6 +141,11 @@ describe('plugin-meetings', () => {
                     packetsReceived: 0,
                     bytesReceived: 1,
                   },
+                  {
+                    type: 'local-candidate',
+                    id: 'fake-candidate-id',
+                    protocol: 'tcp'
+                  },
                 ],
               },
             ],
@@ -149,6 +159,11 @@ describe('plugin-meetings', () => {
                     framesSent: 0,
                     bytesSent: 1,
                   },
+                  {
+                    type: 'local-candidate',
+                    id: 'fake-candidate-id',
+                    protocol: 'tcp'
+                  },
                 ],
               },
             ],
@@ -163,6 +178,11 @@ describe('plugin-meetings', () => {
                     frameWidth: 1280,
                     framesReceived: 1,
                   },
+                  {
+                    type: 'local-candidate',
+                    id: 'fake-candidate-id',
+                    protocol: 'tcp'
+                  },
                 ],
               },
             ],
@@ -242,6 +262,13 @@ describe('plugin-meetings', () => {
         assert.strictEqual(mqeData.videoReceive[0].streams[0].receivedWidth, 1280);
       };
 
+      const checkMqeTransportType = () => {
+        console.log(mqeData.audioTransmit[0])
+        console.log(mqeData.videoReceive[0])
+        assert.strictEqual(mqeData.audioTransmit[0].common.transportType, 'TCP');
+        assert.strictEqual(mqeData.videoReceive[0].common.transportType, 'TCP');
+      }
+
       it('emits LOCAL_MEDIA_STARTED and LOCAL_MEDIA_STOPPED events for audio', async () => {
         await startStatsAnalyzer({expected: {sendAudio: true}});
 
@@ -330,6 +357,14 @@ describe('plugin-meetings', () => {
         // Check that the mqe data has been emitted and is correctly computed.
         checkMqeData();
       });
+
+      it('emits the correct transportType in MEDIA_QUALITY events', async () => {
+        await startStatsAnalyzer({expected: {receiveVideo: true}});
+
+        await progressTime();
+
+        checkMqeTransportType();
+      });
     });
   });
 });