@signalapp/ringrtc 2.59.4 → 2.60.1
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.
- package/dist/acknowledgments.md +28 -1
- package/dist/bin/virtual_audio.sh +4 -2
- package/dist/index.d.ts +2 -1
- package/dist/index.js +6 -3
- package/dist/ringrtc/CallSummary.d.ts +83 -0
- package/dist/ringrtc/CallSummary.js +32 -0
- package/dist/ringrtc/Service.d.ts +51 -45
- package/dist/ringrtc/Service.js +87 -77
- package/package.json +7 -3
package/dist/acknowledgments.md
CHANGED
|
@@ -669,7 +669,7 @@ For more information on this, and how to apply and follow the GNU AGPL, see
|
|
|
669
669
|
|
|
670
670
|
```
|
|
671
671
|
|
|
672
|
-
## libsignal-account-keys 0.1.0, libsignal-core 0.1.0, mrp 2.
|
|
672
|
+
## libsignal-account-keys 0.1.0, libsignal-core 0.1.0, mrp 2.60.1, protobuf 2.60.1, ringrtc 2.60.1, regex-aot 0.1.0, partial-default-derive 0.1.0
|
|
673
673
|
|
|
674
674
|
```
|
|
675
675
|
GNU AFFERO GENERAL PUBLIC LICENSE
|
|
@@ -3470,6 +3470,33 @@ SOFTWARE.
|
|
|
3470
3470
|
|
|
3471
3471
|
```
|
|
3472
3472
|
|
|
3473
|
+
## strum 0.27.2, strum_macros 0.27.2
|
|
3474
|
+
|
|
3475
|
+
```
|
|
3476
|
+
MIT License
|
|
3477
|
+
|
|
3478
|
+
Copyright (c) 2019 Peter Glotfelty
|
|
3479
|
+
|
|
3480
|
+
Permission is hereby granted, free of charge, to any person obtaining a copy
|
|
3481
|
+
of this software and associated documentation files (the "Software"), to deal
|
|
3482
|
+
in the Software without restriction, including without limitation the rights
|
|
3483
|
+
to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
|
|
3484
|
+
copies of the Software, and to permit persons to whom the Software is
|
|
3485
|
+
furnished to do so, subject to the following conditions:
|
|
3486
|
+
|
|
3487
|
+
The above copyright notice and this permission notice shall be included in all
|
|
3488
|
+
copies or substantial portions of the Software.
|
|
3489
|
+
|
|
3490
|
+
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
|
|
3491
|
+
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
|
|
3492
|
+
FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
|
|
3493
|
+
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
|
|
3494
|
+
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
|
|
3495
|
+
OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
|
|
3496
|
+
SOFTWARE.
|
|
3497
|
+
|
|
3498
|
+
```
|
|
3499
|
+
|
|
3473
3500
|
## zeroize_derive 1.4.2
|
|
3474
3501
|
|
|
3475
3502
|
```
|
|
@@ -184,8 +184,10 @@ setup_linux()
|
|
|
184
184
|
teardown_linux()
|
|
185
185
|
{
|
|
186
186
|
# safe to ignore failures here
|
|
187
|
-
|
|
188
|
-
pactl
|
|
187
|
+
# There will be two devices matching this: the INPUT_SINK and the INPUT_SOURCE
|
|
188
|
+
pactl list modules short | grep "${INPUT_SOURCE}" | cut -f1 | xargs -n 1 pactl unload-module || true
|
|
189
|
+
# There will only be one device matching this
|
|
190
|
+
pactl list modules short | grep "${OUTPUT_SINK}" | cut -f1 | xargs -n 1 pactl unload-module || true
|
|
189
191
|
}
|
|
190
192
|
|
|
191
193
|
play_linux()
|
package/dist/index.d.ts
CHANGED
|
@@ -1,4 +1,5 @@
|
|
|
1
1
|
import { RingRTCType } from './ringrtc/Service';
|
|
2
|
-
export { AnswerMessage, AudioDevice, DataMode, BusyMessage, Call,
|
|
2
|
+
export { AnswerMessage, AudioDevice, DataMode, BusyMessage, Call, CallEndReason, CallId, CallLogLevel, CallMessageUrgency, CallRejectReason, CallSettings, CallState, CallingMessage, ConnectionState, DeviceId, GroupCall, GroupCallKind, GroupCallObserver, GroupMemberInfo, HangupMessage, HangupType, HttpMethod, HttpResult, IceCandidateMessage, JoinState, LocalDeviceState, OfferMessage, OfferType, OpaqueMessage, PeekDeviceInfo, PeekInfo, PeekStatusCodes, Reaction, RemoteDeviceState, RingCancelReason, RingRTCType, RingUpdate, SpeechEvent, UserId, VideoFrameSender, VideoFrameSource, VideoPixelFormatEnum, videoPixelFormatToEnum, VideoRequest, callIdFromEra, callIdFromRingId, } from './ringrtc/Service';
|
|
3
3
|
export { CallLinkRootKey, CallLinkRestrictions, CallLinkState, CallLinkEpoch, } from './ringrtc/CallLinks';
|
|
4
|
+
export { CallSummary, QualityStats } from './ringrtc/CallSummary';
|
|
4
5
|
export declare const RingRTC: RingRTCType;
|
package/dist/index.js
CHANGED
|
@@ -4,21 +4,21 @@
|
|
|
4
4
|
// SPDX-License-Identifier: AGPL-3.0-only
|
|
5
5
|
//
|
|
6
6
|
Object.defineProperty(exports, "__esModule", { value: true });
|
|
7
|
-
exports.RingRTC = exports.CallLinkEpoch = exports.CallLinkState = exports.CallLinkRestrictions = exports.CallLinkRootKey = exports.callIdFromRingId = exports.callIdFromEra = exports.VideoRequest = exports.videoPixelFormatToEnum = exports.VideoPixelFormatEnum = exports.SpeechEvent = exports.RingUpdate = exports.RingRTCType = exports.RingCancelReason = exports.RemoteDeviceState = exports.PeekStatusCodes = exports.OpaqueMessage = exports.OfferType = exports.OfferMessage = exports.LocalDeviceState = exports.JoinState = exports.IceCandidateMessage = exports.HttpMethod = exports.HangupType = exports.HangupMessage = exports.GroupMemberInfo = exports.GroupCallKind = exports.
|
|
7
|
+
exports.RingRTC = exports.QualityStats = exports.CallSummary = exports.CallLinkEpoch = exports.CallLinkState = exports.CallLinkRestrictions = exports.CallLinkRootKey = exports.callIdFromRingId = exports.callIdFromEra = exports.VideoRequest = exports.videoPixelFormatToEnum = exports.VideoPixelFormatEnum = exports.SpeechEvent = exports.RingUpdate = exports.RingRTCType = exports.RingCancelReason = exports.RemoteDeviceState = exports.PeekStatusCodes = exports.OpaqueMessage = exports.OfferType = exports.OfferMessage = exports.LocalDeviceState = exports.JoinState = exports.IceCandidateMessage = exports.HttpMethod = exports.HangupType = exports.HangupMessage = exports.GroupMemberInfo = exports.GroupCallKind = exports.GroupCall = exports.ConnectionState = exports.CallingMessage = exports.CallState = exports.CallRejectReason = exports.CallMessageUrgency = exports.CallLogLevel = exports.CallEndReason = exports.Call = exports.BusyMessage = exports.DataMode = exports.AnswerMessage = void 0;
|
|
8
8
|
const Service_1 = require("./ringrtc/Service");
|
|
9
9
|
var Service_2 = require("./ringrtc/Service");
|
|
10
10
|
Object.defineProperty(exports, "AnswerMessage", { enumerable: true, get: function () { return Service_2.AnswerMessage; } });
|
|
11
11
|
Object.defineProperty(exports, "DataMode", { enumerable: true, get: function () { return Service_2.DataMode; } });
|
|
12
12
|
Object.defineProperty(exports, "BusyMessage", { enumerable: true, get: function () { return Service_2.BusyMessage; } });
|
|
13
13
|
Object.defineProperty(exports, "Call", { enumerable: true, get: function () { return Service_2.Call; } });
|
|
14
|
-
Object.defineProperty(exports, "
|
|
14
|
+
Object.defineProperty(exports, "CallEndReason", { enumerable: true, get: function () { return Service_2.CallEndReason; } });
|
|
15
15
|
Object.defineProperty(exports, "CallLogLevel", { enumerable: true, get: function () { return Service_2.CallLogLevel; } });
|
|
16
16
|
Object.defineProperty(exports, "CallMessageUrgency", { enumerable: true, get: function () { return Service_2.CallMessageUrgency; } });
|
|
17
|
+
Object.defineProperty(exports, "CallRejectReason", { enumerable: true, get: function () { return Service_2.CallRejectReason; } });
|
|
17
18
|
Object.defineProperty(exports, "CallState", { enumerable: true, get: function () { return Service_2.CallState; } });
|
|
18
19
|
Object.defineProperty(exports, "CallingMessage", { enumerable: true, get: function () { return Service_2.CallingMessage; } });
|
|
19
20
|
Object.defineProperty(exports, "ConnectionState", { enumerable: true, get: function () { return Service_2.ConnectionState; } });
|
|
20
21
|
Object.defineProperty(exports, "GroupCall", { enumerable: true, get: function () { return Service_2.GroupCall; } });
|
|
21
|
-
Object.defineProperty(exports, "GroupCallEndReason", { enumerable: true, get: function () { return Service_2.GroupCallEndReason; } });
|
|
22
22
|
Object.defineProperty(exports, "GroupCallKind", { enumerable: true, get: function () { return Service_2.GroupCallKind; } });
|
|
23
23
|
Object.defineProperty(exports, "GroupMemberInfo", { enumerable: true, get: function () { return Service_2.GroupMemberInfo; } });
|
|
24
24
|
Object.defineProperty(exports, "HangupMessage", { enumerable: true, get: function () { return Service_2.HangupMessage; } });
|
|
@@ -46,5 +46,8 @@ Object.defineProperty(exports, "CallLinkRootKey", { enumerable: true, get: funct
|
|
|
46
46
|
Object.defineProperty(exports, "CallLinkRestrictions", { enumerable: true, get: function () { return CallLinks_1.CallLinkRestrictions; } });
|
|
47
47
|
Object.defineProperty(exports, "CallLinkState", { enumerable: true, get: function () { return CallLinks_1.CallLinkState; } });
|
|
48
48
|
Object.defineProperty(exports, "CallLinkEpoch", { enumerable: true, get: function () { return CallLinks_1.CallLinkEpoch; } });
|
|
49
|
+
var CallSummary_1 = require("./ringrtc/CallSummary");
|
|
50
|
+
Object.defineProperty(exports, "CallSummary", { enumerable: true, get: function () { return CallSummary_1.CallSummary; } });
|
|
51
|
+
Object.defineProperty(exports, "QualityStats", { enumerable: true, get: function () { return CallSummary_1.QualityStats; } });
|
|
49
52
|
exports.RingRTC = new Service_1.RingRTCType();
|
|
50
53
|
//# sourceMappingURL=index.js.map
|
|
@@ -0,0 +1,83 @@
|
|
|
1
|
+
/**
|
|
2
|
+
* Media quality statistics for audio or video streams.
|
|
3
|
+
*
|
|
4
|
+
* Contains network quality metrics including RTT, jitter, and packet loss.
|
|
5
|
+
*/
|
|
6
|
+
export declare class MediaQualityStats {
|
|
7
|
+
/**
|
|
8
|
+
* Median RTT in milliseconds calculated via RTP/RTCP, or undefined if unavailable.
|
|
9
|
+
*/
|
|
10
|
+
readonly rttMedianMillis: number | undefined;
|
|
11
|
+
/**
|
|
12
|
+
* Median jitter for sent packets as reported by remote peer in milliseconds,
|
|
13
|
+
* or undefined if unavailable.
|
|
14
|
+
*/
|
|
15
|
+
readonly jitterMedianSendMillis: number | undefined;
|
|
16
|
+
/**
|
|
17
|
+
* Median jitter for received packets in milliseconds, or undefined if unavailable.
|
|
18
|
+
*/
|
|
19
|
+
readonly jitterMedianRecvMillis: number | undefined;
|
|
20
|
+
/**
|
|
21
|
+
* Packet loss percentage for sent packets as reported by remote peer,
|
|
22
|
+
* or undefined if unavailable.
|
|
23
|
+
*/
|
|
24
|
+
readonly packetLossPercentageSend: number | undefined;
|
|
25
|
+
/**
|
|
26
|
+
* Packet loss percentage for received packets, or undefined if unavailable.
|
|
27
|
+
*/
|
|
28
|
+
readonly packetLossPercentageRecv: number | undefined;
|
|
29
|
+
}
|
|
30
|
+
/**
|
|
31
|
+
* Overall call quality statistics.
|
|
32
|
+
*
|
|
33
|
+
* Contains connection-level metrics and separate audio/video quality stats.
|
|
34
|
+
*/
|
|
35
|
+
export declare class QualityStats {
|
|
36
|
+
/**
|
|
37
|
+
* Median connection RTT in milliseconds calculated via STUN/ICE,
|
|
38
|
+
* or undefined if unavailable.
|
|
39
|
+
*/
|
|
40
|
+
readonly rttMedianConnection: number | undefined;
|
|
41
|
+
/** Audio quality statistics. */
|
|
42
|
+
readonly audioStats: MediaQualityStats;
|
|
43
|
+
/** Video quality statistics. */
|
|
44
|
+
readonly videoStats: MediaQualityStats;
|
|
45
|
+
}
|
|
46
|
+
/**
|
|
47
|
+
* Summary of call telemetry data providing a synopsis of call quality.
|
|
48
|
+
*
|
|
49
|
+
* Statistics are captured when the call ends and are available for reporting.
|
|
50
|
+
*/
|
|
51
|
+
export declare class CallSummary {
|
|
52
|
+
/**
|
|
53
|
+
* Call start timestamp in milliseconds since January 1, 1970 00:00:00 UTC.
|
|
54
|
+
*/
|
|
55
|
+
readonly startTime: number;
|
|
56
|
+
/**
|
|
57
|
+
* Call end timestamp in milliseconds since January 1, 1970 00:00:00 UTC.
|
|
58
|
+
*/
|
|
59
|
+
readonly endTime: number;
|
|
60
|
+
/**
|
|
61
|
+
* High-level call quality statistics with cumulative metrics for the entire
|
|
62
|
+
* call session, including connection-level stats and separate audio/video
|
|
63
|
+
* quality stats.
|
|
64
|
+
*/
|
|
65
|
+
readonly qualityStats: QualityStats;
|
|
66
|
+
/**
|
|
67
|
+
* Raw call telemetry data containing periodic internal/opaque values for the
|
|
68
|
+
* last few seconds of the call, or undefined if unavailable.
|
|
69
|
+
*/
|
|
70
|
+
readonly rawStats: Uint8Array | undefined;
|
|
71
|
+
/**
|
|
72
|
+
* Textual description of raw telemetry data, or undefined if unavailable.
|
|
73
|
+
*/
|
|
74
|
+
readonly rawStatsText: string | undefined;
|
|
75
|
+
/**
|
|
76
|
+
* Textual representation of the call end reason.
|
|
77
|
+
*/
|
|
78
|
+
readonly callEndReasonText: string;
|
|
79
|
+
/**
|
|
80
|
+
* Whether the call is eligible for user survey (i.e., the call actually connected).
|
|
81
|
+
*/
|
|
82
|
+
readonly isSurveyCandidate: boolean;
|
|
83
|
+
}
|
|
@@ -0,0 +1,32 @@
|
|
|
1
|
+
"use strict";
|
|
2
|
+
//
|
|
3
|
+
// Copyright 2019-2025 Signal Messenger, LLC
|
|
4
|
+
// SPDX-License-Identifier: AGPL-3.0-only
|
|
5
|
+
//
|
|
6
|
+
Object.defineProperty(exports, "__esModule", { value: true });
|
|
7
|
+
exports.CallSummary = exports.QualityStats = exports.MediaQualityStats = void 0;
|
|
8
|
+
/**
|
|
9
|
+
* Media quality statistics for audio or video streams.
|
|
10
|
+
*
|
|
11
|
+
* Contains network quality metrics including RTT, jitter, and packet loss.
|
|
12
|
+
*/
|
|
13
|
+
class MediaQualityStats {
|
|
14
|
+
}
|
|
15
|
+
exports.MediaQualityStats = MediaQualityStats;
|
|
16
|
+
/**
|
|
17
|
+
* Overall call quality statistics.
|
|
18
|
+
*
|
|
19
|
+
* Contains connection-level metrics and separate audio/video quality stats.
|
|
20
|
+
*/
|
|
21
|
+
class QualityStats {
|
|
22
|
+
}
|
|
23
|
+
exports.QualityStats = QualityStats;
|
|
24
|
+
/**
|
|
25
|
+
* Summary of call telemetry data providing a synopsis of call quality.
|
|
26
|
+
*
|
|
27
|
+
* Statistics are captured when the call ends and are available for reporting.
|
|
28
|
+
*/
|
|
29
|
+
class CallSummary {
|
|
30
|
+
}
|
|
31
|
+
exports.CallSummary = CallSummary;
|
|
32
|
+
//# sourceMappingURL=CallSummary.js.map
|
|
@@ -1,4 +1,5 @@
|
|
|
1
1
|
import { CallLinkState, CallLinkRestrictions, CallLinkRootKey, CallLinkEpoch } from './CallLinks';
|
|
2
|
+
import { CallSummary } from './CallSummary';
|
|
2
3
|
export declare const callIdFromEra: (era: string) => CallId;
|
|
3
4
|
export declare function callIdFromRingId(ringId: bigint): CallId;
|
|
4
5
|
declare class Config {
|
|
@@ -97,7 +98,7 @@ export declare class RingRTCType {
|
|
|
97
98
|
handleStartCall: ((call: Call) => Promise<boolean>) | null;
|
|
98
99
|
handleOutputDeviceChanged: ((devices: Array<AudioDevice>) => Promise<void>) | null;
|
|
99
100
|
handleInputDeviceChanged: ((devices: Array<AudioDevice>) => Promise<void>) | null;
|
|
100
|
-
|
|
101
|
+
handleRejectedIncomingCallRequest: ((callId: CallId, remoteUserId: UserId, reason: CallRejectReason, ageSec: number, wasVideoCall: boolean, receivedAtCounter: number | undefined, receivedAtDate: number | undefined) => void) | null;
|
|
101
102
|
handleLogMessage: ((level: CallLogLevel, fileName: string, line: number, message: string) => void) | null;
|
|
102
103
|
handleSendHttpRequest: ((requestId: number, url: string, method: HttpMethod, headers: {
|
|
103
104
|
[name: string]: string;
|
|
@@ -115,7 +116,8 @@ export declare class RingRTCType {
|
|
|
115
116
|
onStartIncomingCall(remoteUserId: UserId, callId: CallId, isVideoCall: boolean): void;
|
|
116
117
|
proceed(callId: CallId, settings: CallSettings): void;
|
|
117
118
|
onCallState(remoteUserId: UserId, state: CallState): void;
|
|
118
|
-
|
|
119
|
+
onCallRejected(remoteUserId: UserId, callId: CallId, reason: CallRejectReason, ageSec: number): void;
|
|
120
|
+
onCallEnded(remoteUserId: UserId, callId: CallId, reason: CallEndReason, summary: CallSummary): void;
|
|
119
121
|
onRemoteAudioEnabled(remoteUserId: UserId, enabled: boolean): void;
|
|
120
122
|
onRemoteVideoEnabled(remoteUserId: UserId, enabled: boolean): void;
|
|
121
123
|
onRemoteSharingScreen(remoteUserId: UserId, enabled: boolean): void;
|
|
@@ -234,6 +236,7 @@ export declare class RingRTCType {
|
|
|
234
236
|
getGroupCall(groupId: GroupId, sfuUrl: string, hkdfExtraInfo: Uint8Array, audioLevelsIntervalMillis: number | undefined, observer: GroupCallObserver): GroupCall | undefined;
|
|
235
237
|
getCallLinkCall(sfuUrl: string, endorsementPublicKey: Uint8Array, authCredentialPresentation: Uint8Array, rootKey: CallLinkRootKey, epoch: CallLinkEpoch | undefined, adminPasskey: Uint8Array | undefined, hkdfExtraInfo: Uint8Array, audioLevelsIntervalMillis: number | undefined, observer: GroupCallObserver): GroupCall | undefined;
|
|
236
238
|
peekGroupCall(sfuUrl: string, membershipProof: Uint8Array, groupMembers: Array<GroupMemberInfo>): Promise<PeekInfo>;
|
|
239
|
+
setMicrophoneWarmupEnabled(warmup: boolean): void;
|
|
237
240
|
peekCallLinkCall(sfuUrl: string, authCredentialPresentation: Uint8Array, rootKey: CallLinkRootKey, epoch: CallLinkEpoch | undefined): Promise<HttpResult<PeekInfo>>;
|
|
238
241
|
requestMembershipProof(clientId: GroupCallClientId): void;
|
|
239
242
|
requestGroupMembers(clientId: GroupCallClientId): void;
|
|
@@ -249,7 +252,7 @@ export declare class RingRTCType {
|
|
|
249
252
|
handlePeekResponse(requestId: number, statusCode: number, rawInfo: RawPeekInfo | undefined): void;
|
|
250
253
|
handleCallLinkResponse(requestId: number, statusCode: number, state: RawCallLinkState | undefined): void;
|
|
251
254
|
handleEmptyResponse(requestId: number, statusCode: number, state: EmptyObj | undefined): void;
|
|
252
|
-
handleEnded(clientId: GroupCallClientId, reason:
|
|
255
|
+
handleEnded(clientId: GroupCallClientId, reason: CallEndReason, summary: CallSummary): void;
|
|
253
256
|
groupCallRingUpdate(groupId: GroupId, ringId: bigint, sender: GroupCallUserId, state: RingUpdate): void;
|
|
254
257
|
handleRtcStatsReportComplete(reportJson: string): void;
|
|
255
258
|
handleSpeechEvent(clientId: GroupCallClientId, event: SpeechEvent): void;
|
|
@@ -369,7 +372,8 @@ export declare class Call {
|
|
|
369
372
|
remoteAudioLevel: NormalizedAudioLevel;
|
|
370
373
|
remoteSharingScreen: boolean;
|
|
371
374
|
networkRoute: NetworkRoute;
|
|
372
|
-
endedReason?:
|
|
375
|
+
endedReason?: CallEndReason;
|
|
376
|
+
summary?: CallSummary;
|
|
373
377
|
handleStateChanged?: () => void;
|
|
374
378
|
handleRemoteAudioEnabled?: () => void;
|
|
375
379
|
handleRemoteVideoEnabled?: () => void;
|
|
@@ -422,24 +426,6 @@ export declare enum JoinState {
|
|
|
422
426
|
Pending = 2,
|
|
423
427
|
Joined = 3
|
|
424
428
|
}
|
|
425
|
-
export declare enum GroupCallEndReason {
|
|
426
|
-
DeviceExplicitlyDisconnected = 0,
|
|
427
|
-
ServerExplicitlyDisconnected = 1,
|
|
428
|
-
DeniedRequestToJoinCall = 2,
|
|
429
|
-
RemovedFromCall = 3,
|
|
430
|
-
CallManagerIsBusy = 4,
|
|
431
|
-
SfuClientFailedToJoin = 5,
|
|
432
|
-
FailedToCreatePeerConnectionFactory = 6,
|
|
433
|
-
FailedToNegotiateSrtpKeys = 7,
|
|
434
|
-
FailedToCreatePeerConnection = 8,
|
|
435
|
-
FailedToStartPeerConnection = 9,
|
|
436
|
-
FailedToUpdatePeerConnection = 10,
|
|
437
|
-
FailedToSetMaxSendBitrate = 11,
|
|
438
|
-
IceFailedWhileConnecting = 12,
|
|
439
|
-
IceFailedAfterConnected = 13,
|
|
440
|
-
ServerChangedDemuxId = 14,
|
|
441
|
-
HasMaxDevices = 15
|
|
442
|
-
}
|
|
443
429
|
export declare enum SpeechEvent {
|
|
444
430
|
StoppedSpeaking = 0,
|
|
445
431
|
LowerHandSuggestion = 1
|
|
@@ -517,7 +503,7 @@ export interface GroupCallObserver {
|
|
|
517
503
|
onReactions(groupCall: GroupCall, reactions: Array<Reaction>): void;
|
|
518
504
|
onRaisedHands(groupCall: GroupCall, raisedHands: Array<number>): void;
|
|
519
505
|
onPeekChanged(groupCall: GroupCall): void;
|
|
520
|
-
onEnded(groupCall: GroupCall, reason:
|
|
506
|
+
onEnded(groupCall: GroupCall, reason: CallEndReason, summary: CallSummary): void;
|
|
521
507
|
onSpeechEvent(groupCall: GroupCall, event: SpeechEvent): void;
|
|
522
508
|
onRemoteMute(groupCall: GroupCall, demuxId: number): void;
|
|
523
509
|
onObservedRemoteMute(groupCall: GroupCall, sourceDemuxId: number, targetDemuxId: number): void;
|
|
@@ -570,7 +556,7 @@ export declare class GroupCall {
|
|
|
570
556
|
handleRaisedHands(raisedHands: Array<number>): void;
|
|
571
557
|
handleRemoteDevicesChanged(remoteDeviceStates: Array<RemoteDeviceState>): void;
|
|
572
558
|
handlePeekChanged(info: PeekInfo): void;
|
|
573
|
-
handleEnded(reason:
|
|
559
|
+
handleEnded(reason: CallEndReason, summary: CallSummary): void;
|
|
574
560
|
sendVideoFrame(width: number, height: number, format: VideoPixelFormatEnum, buffer: Uint8Array): void;
|
|
575
561
|
getVideoSource(remoteDemuxId: number): GroupCallVideoFrameSource;
|
|
576
562
|
setRemoteAspectRatio(remoteDemuxId: number, aspectRatio: number): void;
|
|
@@ -662,6 +648,7 @@ export interface CallManager {
|
|
|
662
648
|
signalingMessageSent(callId: CallId): void;
|
|
663
649
|
signalingMessageSendFailed(callId: CallId): void;
|
|
664
650
|
setOutgoingAudioEnabled(enabled: boolean): void;
|
|
651
|
+
setMicrophoneWarmupEnabled(enabled: boolean): void;
|
|
665
652
|
setOutgoingVideoEnabled(enabled: boolean): void;
|
|
666
653
|
setOutgoingVideoIsScreenShare(enabled: boolean): void;
|
|
667
654
|
updateDataMode(dataMode: DataMode): void;
|
|
@@ -717,7 +704,8 @@ export interface CallManagerCallbacks {
|
|
|
717
704
|
onStartOutgoingCall(remoteUserId: UserId, callId: CallId): void;
|
|
718
705
|
onStartIncomingCall(remoteUserId: UserId, callId: CallId, isVideoCall: boolean): void;
|
|
719
706
|
onCallState(remoteUserId: UserId, state: CallState): void;
|
|
720
|
-
|
|
707
|
+
onCallRejected(remoteUserId: UserId, callId: CallId, rejectReason: CallRejectReason, ageSec: number): void;
|
|
708
|
+
onCallEnded(remoteUserId: UserId, callId: CallId, endedReason: CallEndReason, summary: CallSummary): void;
|
|
721
709
|
onRemoteAudioEnabled(remoteUserId: UserId, enabled: boolean): void;
|
|
722
710
|
onRemoteVideoEnabled(remoteUserId: UserId, enabled: boolean): void;
|
|
723
711
|
onRemoteSharingScreen(remoteUserId: UserId, enabled: boolean): void;
|
|
@@ -738,7 +726,7 @@ export interface CallManagerCallbacks {
|
|
|
738
726
|
handleRemoteDevicesChanged(clientId: GroupCallClientId, remoteDeviceStates: Array<RemoteDeviceState>): void;
|
|
739
727
|
handlePeekChanged(clientId: GroupCallClientId, rawInfo: RawPeekInfo): void;
|
|
740
728
|
handlePeekResponse(requestId: number, statusCode: number, rawInfo: RawPeekInfo | undefined): void;
|
|
741
|
-
handleEnded(clientId: GroupCallClientId, reason:
|
|
729
|
+
handleEnded(clientId: GroupCallClientId, reason: CallEndReason, summary: CallSummary): void;
|
|
742
730
|
onLogMessage(level: number, fileName: string, line: number, message: string): void;
|
|
743
731
|
}
|
|
744
732
|
export declare enum CallState {
|
|
@@ -748,25 +736,43 @@ export declare enum CallState {
|
|
|
748
736
|
Reconnecting = "connecting",
|
|
749
737
|
Ended = "ended"
|
|
750
738
|
}
|
|
751
|
-
export declare enum
|
|
752
|
-
LocalHangup =
|
|
753
|
-
RemoteHangup =
|
|
754
|
-
RemoteHangupNeedPermission =
|
|
755
|
-
|
|
756
|
-
|
|
757
|
-
|
|
758
|
-
|
|
759
|
-
|
|
760
|
-
|
|
761
|
-
|
|
762
|
-
|
|
763
|
-
|
|
764
|
-
ConnectionFailure =
|
|
765
|
-
|
|
766
|
-
|
|
767
|
-
|
|
768
|
-
|
|
769
|
-
|
|
739
|
+
export declare enum CallEndReason {
|
|
740
|
+
LocalHangup = 0,
|
|
741
|
+
RemoteHangup = 1,
|
|
742
|
+
RemoteHangupNeedPermission = 2,
|
|
743
|
+
RemoteHangupAccepted = 3,
|
|
744
|
+
RemoteHangupDeclined = 4,
|
|
745
|
+
RemoteHangupBusy = 5,
|
|
746
|
+
RemoteBusy = 6,
|
|
747
|
+
RemoteGlare = 7,
|
|
748
|
+
RemoteReCall = 8,
|
|
749
|
+
Timeout = 9,
|
|
750
|
+
InternalFailure = 10,
|
|
751
|
+
SignalingFailure = 11,
|
|
752
|
+
ConnectionFailure = 12,
|
|
753
|
+
AppDroppedCall = 13,
|
|
754
|
+
DeviceExplicitlyDisconnected = 14,
|
|
755
|
+
ServerExplicitlyDisconnected = 15,
|
|
756
|
+
DeniedRequestToJoinCall = 16,
|
|
757
|
+
RemovedFromCall = 17,
|
|
758
|
+
CallManagerIsBusy = 18,
|
|
759
|
+
SfuClientFailedToJoin = 19,
|
|
760
|
+
FailedToCreatePeerConnectionFactory = 20,
|
|
761
|
+
FailedToNegotiatedSrtpKeys = 21,
|
|
762
|
+
FailedToCreatePeerConnection = 22,
|
|
763
|
+
FailedToStartPeerConnection = 23,
|
|
764
|
+
FailedToUpdatePeerConnection = 24,
|
|
765
|
+
FailedToSetMaxSendBitrate = 25,
|
|
766
|
+
IceFailedWhileConnecting = 26,
|
|
767
|
+
IceFailedAfterConnected = 27,
|
|
768
|
+
ServerChangedDemuxId = 28,
|
|
769
|
+
HasMaxDevices = 29
|
|
770
|
+
}
|
|
771
|
+
export declare enum CallRejectReason {
|
|
772
|
+
GlareHandlingFailure = 0,
|
|
773
|
+
ReceivedOfferExpired = 1,
|
|
774
|
+
ReceivedOfferWhileActive = 2,
|
|
775
|
+
ReceivedOfferWithGlare = 3
|
|
770
776
|
}
|
|
771
777
|
export declare enum CallLogLevel {
|
|
772
778
|
Off = 0,
|
package/dist/ringrtc/Service.js
CHANGED
|
@@ -7,7 +7,7 @@ var __importDefault = (this && this.__importDefault) || function (mod) {
|
|
|
7
7
|
return (mod && mod.__esModule) ? mod : { "default": mod };
|
|
8
8
|
};
|
|
9
9
|
Object.defineProperty(exports, "__esModule", { value: true });
|
|
10
|
-
exports.CallLogLevel = exports.
|
|
10
|
+
exports.CallLogLevel = exports.CallRejectReason = exports.CallEndReason = exports.CallState = exports.RingCancelReason = exports.DataMode = exports.HangupType = exports.OpaqueMessage = exports.HangupMessage = exports.BusyMessage = exports.IceCandidateMessage = exports.AnswerMessage = exports.OfferType = exports.OfferMessage = exports.CallingMessage = exports.GroupCall = exports.GroupCallKind = exports.VideoRequest = exports.GroupMemberInfo = exports.RemoteDeviceState = exports.LocalDeviceState = exports.HttpMethod = exports.RingUpdate = exports.CallMessageUrgency = exports.SpeechEvent = exports.JoinState = exports.ConnectionState = exports.Call = exports.VideoPixelFormatEnum = exports.RingRTCType = exports.ReceivedAudioLevel = exports.NetworkRoute = exports.PeekStatusCodes = exports.callIdFromEra = void 0;
|
|
11
11
|
exports.callIdFromRingId = callIdFromRingId;
|
|
12
12
|
exports.videoPixelFormatToEnum = videoPixelFormatToEnum;
|
|
13
13
|
/* eslint-disable max-classes-per-file */
|
|
@@ -88,6 +88,8 @@ NativeCallManager.prototype.httpRequestFailed =
|
|
|
88
88
|
Native_1.default.cm_httpRequestFailed;
|
|
89
89
|
NativeCallManager.prototype.setOutgoingAudioEnabled =
|
|
90
90
|
Native_1.default.cm_setOutgoingAudioEnabled;
|
|
91
|
+
NativeCallManager.prototype.setMicrophoneWarmupEnabled =
|
|
92
|
+
Native_1.default.cm_setMicrophoneWarmupEnabled;
|
|
91
93
|
NativeCallManager.prototype.setOutgoingVideoEnabled =
|
|
92
94
|
Native_1.default.cm_setOutgoingVideoEnabled;
|
|
93
95
|
NativeCallManager.prototype.setOutgoingVideoIsScreenShare =
|
|
@@ -270,7 +272,7 @@ class RingRTCType {
|
|
|
270
272
|
this.handleStartCall = null;
|
|
271
273
|
this.handleOutputDeviceChanged = null;
|
|
272
274
|
this.handleInputDeviceChanged = null;
|
|
273
|
-
this.
|
|
275
|
+
this.handleRejectedIncomingCallRequest = null;
|
|
274
276
|
this.handleLogMessage = null;
|
|
275
277
|
this.handleSendHttpRequest = null;
|
|
276
278
|
this.handleSendCallMessage = null;
|
|
@@ -336,8 +338,8 @@ class RingRTCType {
|
|
|
336
338
|
// In case of a glare loser, an incoming call will be generated right
|
|
337
339
|
// after the outgoing call is ended. In that case, ignore it once.
|
|
338
340
|
if (this._call &&
|
|
339
|
-
(this._call.endedReason ===
|
|
340
|
-
this._call.endedReason ===
|
|
341
|
+
(this._call.endedReason === CallEndReason.RemoteGlare ||
|
|
342
|
+
this._call.endedReason === CallEndReason.RemoteReCall)) {
|
|
341
343
|
this._call.endedReason = undefined;
|
|
342
344
|
// EVIL HACK: We are the "loser" of a glare collision and have ended the outgoing call
|
|
343
345
|
// and are now receiving the incoming call from the remote side (the "winner").
|
|
@@ -397,28 +399,20 @@ class RingRTCType {
|
|
|
397
399
|
call.state = state;
|
|
398
400
|
}
|
|
399
401
|
// Called by Rust
|
|
400
|
-
|
|
402
|
+
onCallRejected(remoteUserId, callId, reason, ageSec) {
|
|
401
403
|
const callInfo = this._callInfoByCallId.get(this.getCallInfoKey(callId));
|
|
402
404
|
const { isVideoCall, receivedAtCounter, receivedAtDate } = callInfo || {
|
|
403
405
|
isVideoCall: false,
|
|
404
406
|
receivedAtCounter: undefined,
|
|
405
407
|
receivedAtDate: undefined,
|
|
406
408
|
};
|
|
407
|
-
this._callInfoByCallId.delete(this.getCallInfoKey(callId));
|
|
408
409
|
const call = this._call;
|
|
409
|
-
if (call && reason ==
|
|
410
|
+
if (call && reason == CallRejectReason.ReceivedOfferWithGlare) {
|
|
410
411
|
// The current call is the outgoing call.
|
|
411
|
-
// The
|
|
412
|
+
// The rejected call is the incoming call.
|
|
412
413
|
// We're the "winner", so ignore the incoming call and keep going with the outgoing call.
|
|
413
414
|
return;
|
|
414
415
|
}
|
|
415
|
-
if (call &&
|
|
416
|
-
(reason === CallEndedReason.Glare || reason === CallEndedReason.ReCall)) {
|
|
417
|
-
// The current call is the outgoing call.
|
|
418
|
-
// The ended call is the outgoing call.
|
|
419
|
-
// We're the "loser", so end the outgoing/current call and wait for a new incoming call.
|
|
420
|
-
// (proceeded down to the code below)
|
|
421
|
-
}
|
|
422
416
|
// If there is no call or the remoteUserId doesn't match that of the current
|
|
423
417
|
// call, or if one of the "receive offer while already in a call or because
|
|
424
418
|
// it expired" reasons are provided, don't end the current call, because
|
|
@@ -427,24 +421,25 @@ class RingRTCType {
|
|
|
427
421
|
// update the call history because it is just a replay of messages.
|
|
428
422
|
if (!call ||
|
|
429
423
|
call.remoteUserId !== remoteUserId ||
|
|
430
|
-
reason ===
|
|
431
|
-
reason ===
|
|
424
|
+
reason === CallRejectReason.ReceivedOfferWhileActive ||
|
|
425
|
+
reason === CallRejectReason.ReceivedOfferExpired ||
|
|
432
426
|
(call.state === CallState.Prering && call.isIncoming)) {
|
|
433
|
-
if (this.
|
|
434
|
-
this.
|
|
435
|
-
}
|
|
436
|
-
if (call && call.state === CallState.Prering && call.isIncoming) {
|
|
437
|
-
// Set the state to Ended without triggering a state update since we
|
|
438
|
-
// already notified the client.
|
|
439
|
-
call.endedReason = reason;
|
|
440
|
-
call.setCallEnded();
|
|
427
|
+
if (this.handleRejectedIncomingCallRequest) {
|
|
428
|
+
this.handleRejectedIncomingCallRequest(callId, remoteUserId, reason, ageSec, isVideoCall, receivedAtCounter, receivedAtDate);
|
|
441
429
|
}
|
|
442
|
-
return;
|
|
443
430
|
}
|
|
444
|
-
|
|
445
|
-
|
|
446
|
-
|
|
447
|
-
|
|
431
|
+
}
|
|
432
|
+
// Called by Rust
|
|
433
|
+
onCallEnded(remoteUserId, callId, reason, summary) {
|
|
434
|
+
this._callInfoByCallId.delete(this.getCallInfoKey(callId));
|
|
435
|
+
const call = this._call;
|
|
436
|
+
if (call) {
|
|
437
|
+
// Set the end reason first because setting the state triggers
|
|
438
|
+
// call.handleStateChanged, which may look at call.endedReason.
|
|
439
|
+
call.endedReason = reason;
|
|
440
|
+
call.summary = summary;
|
|
441
|
+
call.state = CallState.Ended;
|
|
442
|
+
}
|
|
448
443
|
}
|
|
449
444
|
onRemoteAudioEnabled(remoteUserId, enabled) {
|
|
450
445
|
const call = this._call;
|
|
@@ -774,6 +769,12 @@ class RingRTCType {
|
|
|
774
769
|
});
|
|
775
770
|
}
|
|
776
771
|
// Called by UX
|
|
772
|
+
setMicrophoneWarmupEnabled(warmup) {
|
|
773
|
+
sillyDeadlockProtection(() => {
|
|
774
|
+
this.callManager.setMicrophoneWarmupEnabled(warmup);
|
|
775
|
+
});
|
|
776
|
+
}
|
|
777
|
+
// Called by UX
|
|
777
778
|
peekCallLinkCall(sfuUrl, authCredentialPresentation, rootKey, epoch) {
|
|
778
779
|
const [requestId, promise] = this._peekRequests.add();
|
|
779
780
|
// Response comes back via handlePeekResponse
|
|
@@ -948,7 +949,7 @@ class RingRTCType {
|
|
|
948
949
|
});
|
|
949
950
|
}
|
|
950
951
|
// Called by Rust
|
|
951
|
-
handleEnded(clientId, reason) {
|
|
952
|
+
handleEnded(clientId, reason, summary) {
|
|
952
953
|
sillyDeadlockProtection(() => {
|
|
953
954
|
const groupCall = this._groupCallByClientId.get(clientId);
|
|
954
955
|
if (!groupCall) {
|
|
@@ -956,7 +957,7 @@ class RingRTCType {
|
|
|
956
957
|
return;
|
|
957
958
|
}
|
|
958
959
|
this._groupCallByClientId.delete(clientId);
|
|
959
|
-
groupCall.handleEnded(reason);
|
|
960
|
+
groupCall.handleEnded(reason, summary);
|
|
960
961
|
});
|
|
961
962
|
}
|
|
962
963
|
// Called by Rust
|
|
@@ -1397,28 +1398,6 @@ var JoinState;
|
|
|
1397
1398
|
JoinState[JoinState["Pending"] = 2] = "Pending";
|
|
1398
1399
|
JoinState[JoinState["Joined"] = 3] = "Joined";
|
|
1399
1400
|
})(JoinState || (exports.JoinState = JoinState = {}));
|
|
1400
|
-
// If not ended purposely by the user, gives the reason why a group call ended.
|
|
1401
|
-
var GroupCallEndReason;
|
|
1402
|
-
(function (GroupCallEndReason) {
|
|
1403
|
-
// Normal events
|
|
1404
|
-
GroupCallEndReason[GroupCallEndReason["DeviceExplicitlyDisconnected"] = 0] = "DeviceExplicitlyDisconnected";
|
|
1405
|
-
GroupCallEndReason[GroupCallEndReason["ServerExplicitlyDisconnected"] = 1] = "ServerExplicitlyDisconnected";
|
|
1406
|
-
GroupCallEndReason[GroupCallEndReason["DeniedRequestToJoinCall"] = 2] = "DeniedRequestToJoinCall";
|
|
1407
|
-
GroupCallEndReason[GroupCallEndReason["RemovedFromCall"] = 3] = "RemovedFromCall";
|
|
1408
|
-
// Things that can go wrong
|
|
1409
|
-
GroupCallEndReason[GroupCallEndReason["CallManagerIsBusy"] = 4] = "CallManagerIsBusy";
|
|
1410
|
-
GroupCallEndReason[GroupCallEndReason["SfuClientFailedToJoin"] = 5] = "SfuClientFailedToJoin";
|
|
1411
|
-
GroupCallEndReason[GroupCallEndReason["FailedToCreatePeerConnectionFactory"] = 6] = "FailedToCreatePeerConnectionFactory";
|
|
1412
|
-
GroupCallEndReason[GroupCallEndReason["FailedToNegotiateSrtpKeys"] = 7] = "FailedToNegotiateSrtpKeys";
|
|
1413
|
-
GroupCallEndReason[GroupCallEndReason["FailedToCreatePeerConnection"] = 8] = "FailedToCreatePeerConnection";
|
|
1414
|
-
GroupCallEndReason[GroupCallEndReason["FailedToStartPeerConnection"] = 9] = "FailedToStartPeerConnection";
|
|
1415
|
-
GroupCallEndReason[GroupCallEndReason["FailedToUpdatePeerConnection"] = 10] = "FailedToUpdatePeerConnection";
|
|
1416
|
-
GroupCallEndReason[GroupCallEndReason["FailedToSetMaxSendBitrate"] = 11] = "FailedToSetMaxSendBitrate";
|
|
1417
|
-
GroupCallEndReason[GroupCallEndReason["IceFailedWhileConnecting"] = 12] = "IceFailedWhileConnecting";
|
|
1418
|
-
GroupCallEndReason[GroupCallEndReason["IceFailedAfterConnected"] = 13] = "IceFailedAfterConnected";
|
|
1419
|
-
GroupCallEndReason[GroupCallEndReason["ServerChangedDemuxId"] = 14] = "ServerChangedDemuxId";
|
|
1420
|
-
GroupCallEndReason[GroupCallEndReason["HasMaxDevices"] = 15] = "HasMaxDevices";
|
|
1421
|
-
})(GroupCallEndReason || (exports.GroupCallEndReason = GroupCallEndReason = {}));
|
|
1422
1401
|
// Matches SpeechEvent in rust.
|
|
1423
1402
|
var SpeechEvent;
|
|
1424
1403
|
(function (SpeechEvent) {
|
|
@@ -1708,9 +1687,9 @@ class GroupCall {
|
|
|
1708
1687
|
this._observer.onPeekChanged(this);
|
|
1709
1688
|
}
|
|
1710
1689
|
// Called by Rust via RingRTC object
|
|
1711
|
-
handleEnded(reason) {
|
|
1690
|
+
handleEnded(reason, summary) {
|
|
1712
1691
|
this._callManager.deleteGroupCallClient(this._clientId);
|
|
1713
|
-
this._observer.onEnded(this, reason);
|
|
1692
|
+
this._observer.onEnded(this, reason, summary);
|
|
1714
1693
|
}
|
|
1715
1694
|
// With this, a GroupCall is a VideoFrameSender
|
|
1716
1695
|
sendVideoFrame(width, height, format, buffer) {
|
|
@@ -1834,27 +1813,58 @@ var CallState;
|
|
|
1834
1813
|
CallState["Reconnecting"] = "connecting";
|
|
1835
1814
|
CallState["Ended"] = "ended";
|
|
1836
1815
|
})(CallState || (exports.CallState = CallState = {}));
|
|
1837
|
-
|
|
1838
|
-
|
|
1839
|
-
|
|
1840
|
-
|
|
1841
|
-
|
|
1842
|
-
|
|
1843
|
-
|
|
1844
|
-
|
|
1845
|
-
|
|
1846
|
-
|
|
1847
|
-
|
|
1848
|
-
|
|
1849
|
-
|
|
1850
|
-
|
|
1851
|
-
|
|
1852
|
-
|
|
1853
|
-
|
|
1854
|
-
|
|
1855
|
-
|
|
1856
|
-
|
|
1857
|
-
|
|
1816
|
+
//
|
|
1817
|
+
// NOTE:
|
|
1818
|
+
//
|
|
1819
|
+
// The ordering of CallEndReason must be kept in sync with the ordering of CallEndReason
|
|
1820
|
+
// in <project-root>/src/rust/common/mod.rs.
|
|
1821
|
+
//
|
|
1822
|
+
var CallEndReason;
|
|
1823
|
+
(function (CallEndReason) {
|
|
1824
|
+
CallEndReason[CallEndReason["LocalHangup"] = 0] = "LocalHangup";
|
|
1825
|
+
CallEndReason[CallEndReason["RemoteHangup"] = 1] = "RemoteHangup";
|
|
1826
|
+
CallEndReason[CallEndReason["RemoteHangupNeedPermission"] = 2] = "RemoteHangupNeedPermission";
|
|
1827
|
+
CallEndReason[CallEndReason["RemoteHangupAccepted"] = 3] = "RemoteHangupAccepted";
|
|
1828
|
+
CallEndReason[CallEndReason["RemoteHangupDeclined"] = 4] = "RemoteHangupDeclined";
|
|
1829
|
+
CallEndReason[CallEndReason["RemoteHangupBusy"] = 5] = "RemoteHangupBusy";
|
|
1830
|
+
CallEndReason[CallEndReason["RemoteBusy"] = 6] = "RemoteBusy";
|
|
1831
|
+
CallEndReason[CallEndReason["RemoteGlare"] = 7] = "RemoteGlare";
|
|
1832
|
+
CallEndReason[CallEndReason["RemoteReCall"] = 8] = "RemoteReCall";
|
|
1833
|
+
CallEndReason[CallEndReason["Timeout"] = 9] = "Timeout";
|
|
1834
|
+
CallEndReason[CallEndReason["InternalFailure"] = 10] = "InternalFailure";
|
|
1835
|
+
CallEndReason[CallEndReason["SignalingFailure"] = 11] = "SignalingFailure";
|
|
1836
|
+
CallEndReason[CallEndReason["ConnectionFailure"] = 12] = "ConnectionFailure";
|
|
1837
|
+
CallEndReason[CallEndReason["AppDroppedCall"] = 13] = "AppDroppedCall";
|
|
1838
|
+
CallEndReason[CallEndReason["DeviceExplicitlyDisconnected"] = 14] = "DeviceExplicitlyDisconnected";
|
|
1839
|
+
CallEndReason[CallEndReason["ServerExplicitlyDisconnected"] = 15] = "ServerExplicitlyDisconnected";
|
|
1840
|
+
CallEndReason[CallEndReason["DeniedRequestToJoinCall"] = 16] = "DeniedRequestToJoinCall";
|
|
1841
|
+
CallEndReason[CallEndReason["RemovedFromCall"] = 17] = "RemovedFromCall";
|
|
1842
|
+
CallEndReason[CallEndReason["CallManagerIsBusy"] = 18] = "CallManagerIsBusy";
|
|
1843
|
+
CallEndReason[CallEndReason["SfuClientFailedToJoin"] = 19] = "SfuClientFailedToJoin";
|
|
1844
|
+
CallEndReason[CallEndReason["FailedToCreatePeerConnectionFactory"] = 20] = "FailedToCreatePeerConnectionFactory";
|
|
1845
|
+
CallEndReason[CallEndReason["FailedToNegotiatedSrtpKeys"] = 21] = "FailedToNegotiatedSrtpKeys";
|
|
1846
|
+
CallEndReason[CallEndReason["FailedToCreatePeerConnection"] = 22] = "FailedToCreatePeerConnection";
|
|
1847
|
+
CallEndReason[CallEndReason["FailedToStartPeerConnection"] = 23] = "FailedToStartPeerConnection";
|
|
1848
|
+
CallEndReason[CallEndReason["FailedToUpdatePeerConnection"] = 24] = "FailedToUpdatePeerConnection";
|
|
1849
|
+
CallEndReason[CallEndReason["FailedToSetMaxSendBitrate"] = 25] = "FailedToSetMaxSendBitrate";
|
|
1850
|
+
CallEndReason[CallEndReason["IceFailedWhileConnecting"] = 26] = "IceFailedWhileConnecting";
|
|
1851
|
+
CallEndReason[CallEndReason["IceFailedAfterConnected"] = 27] = "IceFailedAfterConnected";
|
|
1852
|
+
CallEndReason[CallEndReason["ServerChangedDemuxId"] = 28] = "ServerChangedDemuxId";
|
|
1853
|
+
CallEndReason[CallEndReason["HasMaxDevices"] = 29] = "HasMaxDevices";
|
|
1854
|
+
})(CallEndReason || (exports.CallEndReason = CallEndReason = {}));
|
|
1855
|
+
//
|
|
1856
|
+
// NOTE:
|
|
1857
|
+
//
|
|
1858
|
+
// The ordering of CallRejectReason must be kept in sync with the ordering of RejectReason
|
|
1859
|
+
// in <project-root>/src/rust/src/native.rs.
|
|
1860
|
+
//
|
|
1861
|
+
var CallRejectReason;
|
|
1862
|
+
(function (CallRejectReason) {
|
|
1863
|
+
CallRejectReason[CallRejectReason["GlareHandlingFailure"] = 0] = "GlareHandlingFailure";
|
|
1864
|
+
CallRejectReason[CallRejectReason["ReceivedOfferExpired"] = 1] = "ReceivedOfferExpired";
|
|
1865
|
+
CallRejectReason[CallRejectReason["ReceivedOfferWhileActive"] = 2] = "ReceivedOfferWhileActive";
|
|
1866
|
+
CallRejectReason[CallRejectReason["ReceivedOfferWithGlare"] = 3] = "ReceivedOfferWithGlare";
|
|
1867
|
+
})(CallRejectReason || (exports.CallRejectReason = CallRejectReason = {}));
|
|
1858
1868
|
var CallLogLevel;
|
|
1859
1869
|
(function (CallLogLevel) {
|
|
1860
1870
|
CallLogLevel[CallLogLevel["Off"] = 0] = "Off";
|
package/package.json
CHANGED
|
@@ -1,7 +1,11 @@
|
|
|
1
1
|
{
|
|
2
2
|
"name": "@signalapp/ringrtc",
|
|
3
|
-
"version": "2.
|
|
4
|
-
"repository":
|
|
3
|
+
"version": "2.60.1",
|
|
4
|
+
"repository": {
|
|
5
|
+
"type": "git",
|
|
6
|
+
"url": "https://github.com/signalapp/ringrtc.git",
|
|
7
|
+
"directory": "src/node"
|
|
8
|
+
},
|
|
5
9
|
"description": "Signal Messenger voice and video calling library.",
|
|
6
10
|
"main": "dist/index.js",
|
|
7
11
|
"types": "dist/index.d.ts",
|
|
@@ -29,7 +33,7 @@
|
|
|
29
33
|
},
|
|
30
34
|
"config": {
|
|
31
35
|
"prebuildUrl": "https://build-artifacts.signal.org/libraries/ringrtc-desktop-build-v${npm_package_version}.tar.gz",
|
|
32
|
-
"prebuildChecksum": "
|
|
36
|
+
"prebuildChecksum": "818cf135f9f1e8b209223bdc2e51241844368b60be069055baed5a57b329f455"
|
|
33
37
|
},
|
|
34
38
|
"author": "",
|
|
35
39
|
"license": "AGPL-3.0-only",
|