Voice JS SDK v1: Twilio.PreflightTest

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{
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  "callSid": "CAa6a7a187a9cba2714d6fdccf472cc7b1",
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  /**
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   * The quality of the call, determined by the MOS (Mean Opinion Score) of the audio stream. Possible values include
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   * PreflightTest.CallQuality.Excellent - If the average mos is over 4.2
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   * PreflightTest.CallQuality.Great - If the average mos is between 4.1 and 4.2 both inclusive
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   * PreflightTest.CallQuality.Good - If the average mos is between 3.7 and 4.0 both inclusive
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   * PreflightTest.CallQuality.Fair - If the average mos is between 3.1 and 3.6 both inclusive
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   * PreflightTest.CallQuality.Degraded - If the average mos is 3.0 or below
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   */
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  "callQuality": "excellent",
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  /**
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   * An array of WebRTC stats for the ICE candidates gathered when connecting to media.
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   * Each item is an RTCIceCandidateStats object which provides information related to an ICE candidate.
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   * See RTCIceCandidateStats for more details https://developer.mozilla.org/en-US/docs/Web/API/RTCIceCandidateStats
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   */
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  "iceCandidateStats": [...]
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  /**
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   * A WebRTC stats for the ICE candidate pair used to connect to media, if candidates were selected.
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   * Each item is an RTCIceCandidateStats object which provides information related to an ICE candidate.
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   * See RTCIceCandidateStats for more details https://developer.mozilla.org/en-US/docs/Web/API/RTCIceCandidateStats
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   */
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  "selectedIceCandidatePairStats": {
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    "localCandidate": {...},
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    "remoteCandidate": {...}
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  }
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  /**
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   * Whether a TURN server is required to connect to media.
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   * This is dependent on the selected ICE candidates, and will be true if either is of type "relay",
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   * false if both are of another type, or undefined if there are no selected ICE candidates.
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   * See `PreflightTest.Options.iceServers` for more details.
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   */
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  "isTurnRequired": false,
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  /**
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   * Network related time measurements which includes millisecond timestamps
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   * and duration for each type of connection.
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   */
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  "networkTiming": {
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    /**
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     * Measurements for establishing DTLS connection.
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     * This is measured from RTCDtlsTransport `connecting` to `connected` state.
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     * See RTCDtlsTransport state
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     * https://developer.mozilla.org/en-US/docs/Web/API/RTCDtlsTransport/state.
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     */
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    "dtls": {
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      "start": 1584573229981,
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      "end": 1584573230166,
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      "duration": 185
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    },
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    /**
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     * Measurements for establishing ICE connection.
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     * This is measured from ICE connection `checking` to `connected` state.
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     * See RTCPeerConnection.iceConnectionState
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     * https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/iceConnectionState.
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     */
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    "ice": {
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      "start": 1584573229898,
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      "end": 1584573229982,
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      "duration": 84
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    },
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    /**
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     * Measurements for establishing a PeerConnection.
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     * This is measured from PeerConnection `connecting` to `connected` state.
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     * See RTCPeerConnection.connectionState
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     * https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/connectionState.
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     */
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    "peerConnection": {
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      "start": 1584573229902,
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      "end": 1584573230167,
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      "duration": 265
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    },
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    /**
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     * Measurements for establishing Signaling connection.
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     * This is measured from initiating a connection using `device.connect()`,
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     * up to when RTCPeerConnection.signalingState transitions to `stable` state.
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     * See RTCPeerConnection.signalingState
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     * https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/signalingState
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     */
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    "signaling": {
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      "start": 1595885835227,
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      "end": 1595885835573,
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      "duration": 346
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    }
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  },
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  /**
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   * RTC related stats that are extracted from WebRTC samples.
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   * This information includes maximum, minimum, and average values calculated for each stat.
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   */
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  "stats": {
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    "jitter": {
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      "average": 35,
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      "max": 35,
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      "min": 35
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    },
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    "mos": {
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      "average": 2,
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      "max": 2,
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      "min": 2
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    },
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    "rtt": {
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      "average": 80.33,
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      "max": 88,
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      "min": 77
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    }
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  },
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  /**
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   * Timing measurement related to the test.
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   * Includes millisecond timestamps and duration.
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   */
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  "testTiming": {
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    "start": 1584573229085,
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    "end": 1584573242279,
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    "duration": 13194
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  },
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  /**
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   * Calculated totals in RTC statistics samples.
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   */
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  "totals": {
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    "bytesReceived": 62720,
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    "bytesSent": 93760,
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    "packetsLost": 0,
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    "packetsLostFraction": 0,
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    "packetsReceived": 392,
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    "packetsSent": 586
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  },
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  /**
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   * Array of samples collected during the test.
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   * See sample object format here
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   * https://www.twilio.com/docs/voice/sdks/javascript/v1/connection#sample
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   */
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  "samples": [...],
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  /**
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   * The edge passed to `Device.runPreflight`.
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   */
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  "selectedEdge": "roaming",
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  /**
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   * The edge that the test call was connected to.
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   */
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  "edge": "ashburn",
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  /**
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   * PreflightTest.Warnings detected during the test.
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   */
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  "warnings": [...]
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}

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{
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  /**
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   * Name of the warning.
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   * See https://www.twilio.com/docs/voice/insights/call-quality-events-twilio-client-sdk
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   */
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  name: 'insights-connection-error',
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  /**
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   * Description of the Warning.
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   */
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  description: 'Received an error when attempting to connect to Insights gateway',
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  /**
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   * Optional RTCWarning data coming from Connection.on('warning')
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   * See https://www.twilio.com/docs/voice/sdks/javascript/v1/connection#onwarning-handlerwarningname
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   */
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  rtcWarning: {...}
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}

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<?xml version="1.0" encoding="UTF-8"?>
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<Response>
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  <Say>You said:</Say>
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  <Play loop="1">{{RecordingUrl}}</Play>
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  <Say>Now waiting for a few seconds to gather audio performance metrics.</Say>
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  <Pause length="3"/>
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  <Say>Hanging up now.</Say>
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</Response>

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<?xml version="1.0" encoding="UTF-8"?>
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<Response>
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  <Say>Record a message in 3, 2, 1</Say>
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  <Record maxLength="5" action="https://my-record-twiml-url"></Record>
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  <Say>Did not detect a message to record</Say>
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</Response>

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<?xml version="1.0" encoding="UTF-8"?>
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<Response>
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  <Echo/>
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</Response>