@webex/event-dictionary-ts 1.0.1544 → 1.0.1545
This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
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@@ -298,6 +298,7 @@
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"enum": [
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"SIP",
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"H323",
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"H323_IP",
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"Locus",
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"WebRTC"
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]
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@@ -2166,6 +2167,16 @@
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"sessionJoinCount": {
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"type": "integer"
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},
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"transportType": {
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"type": "string",
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"description": "indicates transport type used",
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"enum": [
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"UDP",
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"TCP",
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"xTLS",
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"TLS"
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]
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},
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"name": {
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"enum": [
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"client.abort.join",
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@@ -797,6 +797,7 @@
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"enum": [
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"SIP",
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"H323",
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"H323_IP",
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"Locus",
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"WebRTC"
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]
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@@ -2665,6 +2666,16 @@
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"sessionJoinCount": {
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"type": "integer"
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},
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"transportType": {
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"type": "string",
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"description": "indicates transport type used",
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"enum": [
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"UDP",
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"TCP",
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"xTLS",
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"TLS"
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]
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},
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"name": {
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"enum": [
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"client.abort.join",
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@@ -8019,6 +8030,7 @@
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"enum": [
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"SIP",
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"H323",
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"H323_IP",
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"Locus",
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"WebRTC"
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]
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@@ -9933,6 +9945,16 @@
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"sessionJoinCount": {
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"type": "integer"
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},
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"transportType": {
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"type": "string",
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"description": "indicates transport type used",
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"enum": [
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"UDP",
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"TCP",
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"xTLS",
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"TLS"
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]
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},
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"remoteAgent": {
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"type": "string"
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},
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@@ -12302,6 +12324,7 @@
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"enum": [
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"SIP",
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"H323",
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"H323_IP",
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"Locus",
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"WebRTC"
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]
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@@ -14216,6 +14239,16 @@
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"sessionJoinCount": {
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"type": "integer"
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},
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"transportType": {
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"type": "string",
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"description": "indicates transport type used",
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"enum": [
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"UDP",
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"TCP",
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"xTLS",
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"TLS"
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]
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},
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"trigger": {
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"enum": [
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"dummyTrigger1",
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@@ -14271,6 +14304,7 @@
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"enum": [
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"SIP",
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"H323",
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"H323_IP",
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"Locus",
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"WebRTC"
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]
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@@ -21208,6 +21242,7 @@
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"enum": [
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"SIP",
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"H323",
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"H323_IP",
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"Locus",
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"WebRTC"
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]
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@@ -23122,6 +23157,16 @@
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"sessionJoinCount": {
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"type": "integer"
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},
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"transportType": {
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"type": "string",
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"description": "indicates transport type used",
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"enum": [
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"UDP",
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"TCP",
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"xTLS",
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"TLS"
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]
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},
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"name": {
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"enum": [
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"client.feature.audio.noise.removal",
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@@ -25153,6 +25198,7 @@
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"enum": [
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"SIP",
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"H323",
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"H323_IP",
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"Locus",
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"WebRTC"
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]
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@@ -27067,6 +27113,16 @@
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"sessionJoinCount": {
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"type": "integer"
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},
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"transportType": {
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"type": "string",
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"description": "indicates transport type used",
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"enum": [
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"UDP",
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"TCP",
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"xTLS",
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"TLS"
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]
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},
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"trigger": {
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"enum": [
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"dummyTrigger1",
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@@ -298,6 +298,7 @@
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"enum": [
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"SIP",
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"H323",
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"H323_IP",
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"Locus",
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"WebRTC"
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]
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@@ -2212,6 +2213,16 @@
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"sessionJoinCount": {
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"type": "integer"
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},
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"transportType": {
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"type": "string",
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"description": "indicates transport type used",
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"enum": [
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"UDP",
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"TCP",
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"xTLS",
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"TLS"
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]
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},
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"name": {
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"enum": [
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"client.feature.audio.noise.removal",
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@@ -299,6 +299,7 @@
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"enum": [
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"SIP",
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"H323",
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"H323_IP",
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"Locus",
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"WebRTC"
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]
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"sessionJoinCount": {
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"type": "integer"
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},
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"transportType": {
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"type": "string",
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"description": "indicates transport type used",
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"enum": [
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"UDP",
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"TCP",
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"xTLS",
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"TLS"
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]
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},
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"trigger": {
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"enum": [
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"dummyTrigger1",
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"enum": [
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"SIP",
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"H323",
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"H323_IP",
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"Locus",
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"WebRTC"
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]
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@@ -360,7 +360,7 @@ export interface Event {
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/**
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* protocols used to help determine how the events are processed as well as how the reports are aggregated and sliced
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*/
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protocol?: "SIP" | "H323" | "Locus" | "WebRTC";
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protocol?: "SIP" | "H323" | "H323_IP" | "Locus" | "WebRTC";
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/**
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* The underlying service provider of the call.
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*/
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additionalProperties?: false;
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};
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sessionJoinCount?: number;
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/**
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* indicates transport type used
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*/
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transportType?: "UDP" | "TCP" | "xTLS" | "TLS";
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name:
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| "client.abort.join"
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| "client.alert.displayed"
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/**
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* protocols used to help determine how the events are processed as well as how the reports are aggregated and sliced
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*/
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protocol?: "SIP" | "H323" | "Locus" | "WebRTC";
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protocol?: "SIP" | "H323" | "H323_IP" | "Locus" | "WebRTC";
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/**
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* The underlying service provider of the call.
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*/
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additionalProperties?: false;
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};
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sessionJoinCount?: number;
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/**
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* indicates transport type used
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*/
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transportType?: "UDP" | "TCP" | "xTLS" | "TLS";
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remoteAgent?: string;
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name:
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| "addin.scheduling.request"
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/**
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* protocols used to help determine how the events are processed as well as how the reports are aggregated and sliced
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*/
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protocol?: "SIP" | "H323" | "Locus" | "WebRTC";
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protocol?: "SIP" | "H323" | "H323_IP" | "Locus" | "WebRTC";
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/**
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* The underlying service provider of the call.
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*/
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additionalProperties?: false;
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};
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sessionJoinCount?: number;
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/**
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* indicates transport type used
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*/
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transportType?: "UDP" | "TCP" | "xTLS" | "TLS";
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trigger?: "dummyTrigger1" | "dummyTrigger2" | "media-quality";
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name: "server.mediaquality.event" | "client.mediaquality.event";
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/**
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/**
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* protocols used to help determine how the events are processed as well as how the reports are aggregated and sliced
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*/
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clientSignallingProtocol?: "SIP" | "H323" | "Locus" | "WebRTC";
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clientSignallingProtocol?: "SIP" | "H323" | "H323_IP" | "Locus" | "WebRTC";
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additionalProperties?: false;
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};
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/**
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/**
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* protocols used to help determine how the events are processed as well as how the reports are aggregated and sliced
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*/
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protocol?: "SIP" | "H323" | "Locus" | "WebRTC";
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protocol?: "SIP" | "H323" | "H323_IP" | "Locus" | "WebRTC";
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/**
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* The underlying service provider of the call.
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*/
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additionalProperties?: false;
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};
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sessionJoinCount?: number;
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/**
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* indicates transport type used
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*/
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transportType?: "UDP" | "TCP" | "xTLS" | "TLS";
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name:
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| "client.feature.embedded-object-info"
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/**
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* protocols used to help determine how the events are processed as well as how the reports are aggregated and sliced
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*/
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protocol?: "SIP" | "H323" | "Locus" | "WebRTC";
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protocol?: "SIP" | "H323" | "H323_IP" | "Locus" | "WebRTC";
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/**
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* The underlying service provider of the call.
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*/
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additionalProperties?: false;
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};
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sessionJoinCount?: number;
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/**
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* indicates transport type used
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*/
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transportType?: "UDP" | "TCP" | "xTLS" | "TLS";
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trigger?: "dummyTrigger1" | "dummyTrigger2" | "edge-mediaquality";
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name: "edge.mediaquality.event";
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/**
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/**
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* protocols used to help determine how the events are processed as well as how the reports are aggregated and sliced
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*/
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protocol?: "SIP" | "H323" | "Locus" | "WebRTC";
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protocol?: "SIP" | "H323" | "H323_IP" | "Locus" | "WebRTC";
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/**
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* The underlying service provider of the call.
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*/
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@@ -12461,6 +12481,10 @@ export interface ClientEvent {
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additionalProperties?: false;
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|
};
|
|
12463
12483
|
sessionJoinCount?: number;
|
|
12484
|
+
/**
|
|
12485
|
+
* indicates transport type used
|
|
12486
|
+
*/
|
|
12487
|
+
transportType?: "UDP" | "TCP" | "xTLS" | "TLS";
|
|
12464
12488
|
name:
|
|
12465
12489
|
| "client.abort.join"
|
|
12466
12490
|
| "client.alert.displayed"
|
|
@@ -15076,7 +15100,7 @@ export interface FeatureEvent {
|
|
|
15076
15100
|
/**
|
|
15077
15101
|
* protocols used to help determine how the events are processed as well as how the reports are aggregated and sliced
|
|
15078
15102
|
*/
|
|
15079
|
-
protocol?: "SIP" | "H323" | "Locus" | "WebRTC";
|
|
15103
|
+
protocol?: "SIP" | "H323" | "H323_IP" | "Locus" | "WebRTC";
|
|
15080
15104
|
/**
|
|
15081
15105
|
* The underlying service provider of the call.
|
|
15082
15106
|
*/
|
|
@@ -15701,6 +15725,10 @@ export interface FeatureEvent {
|
|
|
15701
15725
|
additionalProperties?: false;
|
|
15702
15726
|
};
|
|
15703
15727
|
sessionJoinCount?: number;
|
|
15728
|
+
/**
|
|
15729
|
+
* indicates transport type used
|
|
15730
|
+
*/
|
|
15731
|
+
transportType?: "UDP" | "TCP" | "xTLS" | "TLS";
|
|
15704
15732
|
name:
|
|
15705
15733
|
| "client.feature.audio.noise.removal"
|
|
15706
15734
|
| "client.feature.embedded-object-info"
|
|
@@ -16729,7 +16757,7 @@ export interface MediaQualityEvent {
|
|
|
16729
16757
|
/**
|
|
16730
16758
|
* protocols used to help determine how the events are processed as well as how the reports are aggregated and sliced
|
|
16731
16759
|
*/
|
|
16732
|
-
protocol?: "SIP" | "H323" | "Locus" | "WebRTC";
|
|
16760
|
+
protocol?: "SIP" | "H323" | "H323_IP" | "Locus" | "WebRTC";
|
|
16733
16761
|
/**
|
|
16734
16762
|
* The underlying service provider of the call.
|
|
16735
16763
|
*/
|
|
@@ -17354,6 +17382,10 @@ export interface MediaQualityEvent {
|
|
|
17354
17382
|
additionalProperties?: false;
|
|
17355
17383
|
};
|
|
17356
17384
|
sessionJoinCount?: number;
|
|
17385
|
+
/**
|
|
17386
|
+
* indicates transport type used
|
|
17387
|
+
*/
|
|
17388
|
+
transportType?: "UDP" | "TCP" | "xTLS" | "TLS";
|
|
17357
17389
|
trigger?: "dummyTrigger1" | "dummyTrigger2" | "media-quality";
|
|
17358
17390
|
name: "server.mediaquality.event" | "client.mediaquality.event";
|
|
17359
17391
|
/**
|
|
@@ -17374,7 +17406,7 @@ export interface MediaQualityEvent {
|
|
|
17374
17406
|
/**
|
|
17375
17407
|
* protocols used to help determine how the events are processed as well as how the reports are aggregated and sliced
|
|
17376
17408
|
*/
|
|
17377
|
-
clientSignallingProtocol?: "SIP" | "H323" | "Locus" | "WebRTC";
|
|
17409
|
+
clientSignallingProtocol?: "SIP" | "H323" | "H323_IP" | "Locus" | "WebRTC";
|
|
17378
17410
|
additionalProperties?: false;
|
|
17379
17411
|
};
|
|
17380
17412
|
/**
|
package/package.json
CHANGED