diff options
Diffstat (limited to 'includes/external/matrix/node_modules/matrix-js-sdk/lib/webrtc/call.js')
| -rw-r--r-- | includes/external/matrix/node_modules/matrix-js-sdk/lib/webrtc/call.js | 2341 |
1 files changed, 0 insertions, 2341 deletions
diff --git a/includes/external/matrix/node_modules/matrix-js-sdk/lib/webrtc/call.js b/includes/external/matrix/node_modules/matrix-js-sdk/lib/webrtc/call.js deleted file mode 100644 index 45e23ad..0000000 --- a/includes/external/matrix/node_modules/matrix-js-sdk/lib/webrtc/call.js +++ /dev/null @@ -1,2341 +0,0 @@ -"use strict"; - -var _interopRequireDefault = require("@babel/runtime/helpers/interopRequireDefault"); -Object.defineProperty(exports, "__esModule", { - value: true -}); -exports.MatrixCall = exports.CallType = exports.CallState = exports.CallParty = exports.CallEvent = exports.CallErrorCode = exports.CallError = exports.CallDirection = void 0; -exports.createNewMatrixCall = createNewMatrixCall; -exports.genCallID = genCallID; -exports.setTracksEnabled = setTracksEnabled; -exports.supportsMatrixCall = supportsMatrixCall; -var _defineProperty2 = _interopRequireDefault(require("@babel/runtime/helpers/defineProperty")); -var _uuid = require("uuid"); -var _sdpTransform = require("sdp-transform"); -var _logger = require("../logger"); -var utils = _interopRequireWildcard(require("../utils")); -var _event = require("../@types/event"); -var _randomstring = require("../randomstring"); -var _callEventTypes = require("./callEventTypes"); -var _callFeed = require("./callFeed"); -var _typedEventEmitter = require("../models/typed-event-emitter"); -var _deviceinfo = require("../crypto/deviceinfo"); -var _groupCall = require("./groupCall"); -var _httpApi = require("../http-api"); -function _getRequireWildcardCache(nodeInterop) { if (typeof WeakMap !== "function") return null; var cacheBabelInterop = new WeakMap(); var cacheNodeInterop = new WeakMap(); return (_getRequireWildcardCache = function (nodeInterop) { return nodeInterop ? cacheNodeInterop : cacheBabelInterop; })(nodeInterop); } -function _interopRequireWildcard(obj, nodeInterop) { if (!nodeInterop && obj && obj.__esModule) { return obj; } if (obj === null || typeof obj !== "object" && typeof obj !== "function") { return { default: obj }; } var cache = _getRequireWildcardCache(nodeInterop); if (cache && cache.has(obj)) { return cache.get(obj); } var newObj = {}; var hasPropertyDescriptor = Object.defineProperty && Object.getOwnPropertyDescriptor; for (var key in obj) { if (key !== "default" && Object.prototype.hasOwnProperty.call(obj, key)) { var desc = hasPropertyDescriptor ? Object.getOwnPropertyDescriptor(obj, key) : null; if (desc && (desc.get || desc.set)) { Object.defineProperty(newObj, key, desc); } else { newObj[key] = obj[key]; } } } newObj.default = obj; if (cache) { cache.set(obj, newObj); } return newObj; } -function ownKeys(object, enumerableOnly) { var keys = Object.keys(object); if (Object.getOwnPropertySymbols) { var symbols = Object.getOwnPropertySymbols(object); enumerableOnly && (symbols = symbols.filter(function (sym) { return Object.getOwnPropertyDescriptor(object, sym).enumerable; })), keys.push.apply(keys, symbols); } return keys; } -function _objectSpread(target) { for (var i = 1; i < arguments.length; i++) { var source = null != arguments[i] ? arguments[i] : {}; i % 2 ? ownKeys(Object(source), !0).forEach(function (key) { (0, _defineProperty2.default)(target, key, source[key]); }) : Object.getOwnPropertyDescriptors ? Object.defineProperties(target, Object.getOwnPropertyDescriptors(source)) : ownKeys(Object(source)).forEach(function (key) { Object.defineProperty(target, key, Object.getOwnPropertyDescriptor(source, key)); }); } return target; } -var MediaType; -(function (MediaType) { - MediaType["AUDIO"] = "audio"; - MediaType["VIDEO"] = "video"; -})(MediaType || (MediaType = {})); -var CodecName; // add more as needed -// Used internally to specify modifications to codec parameters in SDP -(function (CodecName) { - CodecName["OPUS"] = "opus"; -})(CodecName || (CodecName = {})); -let CallState; -exports.CallState = CallState; -(function (CallState) { - CallState["Fledgling"] = "fledgling"; - CallState["InviteSent"] = "invite_sent"; - CallState["WaitLocalMedia"] = "wait_local_media"; - CallState["CreateOffer"] = "create_offer"; - CallState["CreateAnswer"] = "create_answer"; - CallState["Connecting"] = "connecting"; - CallState["Connected"] = "connected"; - CallState["Ringing"] = "ringing"; - CallState["Ended"] = "ended"; -})(CallState || (exports.CallState = CallState = {})); -let CallType; -exports.CallType = CallType; -(function (CallType) { - CallType["Voice"] = "voice"; - CallType["Video"] = "video"; -})(CallType || (exports.CallType = CallType = {})); -let CallDirection; -exports.CallDirection = CallDirection; -(function (CallDirection) { - CallDirection["Inbound"] = "inbound"; - CallDirection["Outbound"] = "outbound"; -})(CallDirection || (exports.CallDirection = CallDirection = {})); -let CallParty; -exports.CallParty = CallParty; -(function (CallParty) { - CallParty["Local"] = "local"; - CallParty["Remote"] = "remote"; -})(CallParty || (exports.CallParty = CallParty = {})); -let CallEvent; -exports.CallEvent = CallEvent; -(function (CallEvent) { - CallEvent["Hangup"] = "hangup"; - CallEvent["State"] = "state"; - CallEvent["Error"] = "error"; - CallEvent["Replaced"] = "replaced"; - CallEvent["LocalHoldUnhold"] = "local_hold_unhold"; - CallEvent["RemoteHoldUnhold"] = "remote_hold_unhold"; - CallEvent["HoldUnhold"] = "hold_unhold"; - CallEvent["FeedsChanged"] = "feeds_changed"; - CallEvent["AssertedIdentityChanged"] = "asserted_identity_changed"; - CallEvent["LengthChanged"] = "length_changed"; - CallEvent["DataChannel"] = "datachannel"; - CallEvent["SendVoipEvent"] = "send_voip_event"; -})(CallEvent || (exports.CallEvent = CallEvent = {})); -let CallErrorCode; -/** - * The version field that we set in m.call.* events - */ -exports.CallErrorCode = CallErrorCode; -(function (CallErrorCode) { - CallErrorCode["UserHangup"] = "user_hangup"; - CallErrorCode["LocalOfferFailed"] = "local_offer_failed"; - CallErrorCode["NoUserMedia"] = "no_user_media"; - CallErrorCode["UnknownDevices"] = "unknown_devices"; - CallErrorCode["SendInvite"] = "send_invite"; - CallErrorCode["CreateAnswer"] = "create_answer"; - CallErrorCode["CreateOffer"] = "create_offer"; - CallErrorCode["SendAnswer"] = "send_answer"; - CallErrorCode["SetRemoteDescription"] = "set_remote_description"; - CallErrorCode["SetLocalDescription"] = "set_local_description"; - CallErrorCode["AnsweredElsewhere"] = "answered_elsewhere"; - CallErrorCode["IceFailed"] = "ice_failed"; - CallErrorCode["InviteTimeout"] = "invite_timeout"; - CallErrorCode["Replaced"] = "replaced"; - CallErrorCode["SignallingFailed"] = "signalling_timeout"; - CallErrorCode["UserBusy"] = "user_busy"; - CallErrorCode["Transferred"] = "transferred"; - CallErrorCode["NewSession"] = "new_session"; -})(CallErrorCode || (exports.CallErrorCode = CallErrorCode = {})); -const VOIP_PROTO_VERSION = "1"; - -/** The fallback ICE server to use for STUN or TURN protocols. */ -const FALLBACK_ICE_SERVER = "stun:turn.matrix.org"; - -/** The length of time a call can be ringing for. */ -const CALL_TIMEOUT_MS = 60 * 1000; // ms -/** The time after which we increment callLength */ -const CALL_LENGTH_INTERVAL = 1000; // ms -/** The time after which we end the call, if ICE got disconnected */ -const ICE_DISCONNECTED_TIMEOUT = 30 * 1000; // ms - -class CallError extends Error { - constructor(code, msg, err) { - // Still don't think there's any way to have proper nested errors - super(msg + ": " + err); - (0, _defineProperty2.default)(this, "code", void 0); - this.code = code; - } -} -exports.CallError = CallError; -function genCallID() { - return Date.now().toString() + (0, _randomstring.randomString)(16); -} -function getCodecParamMods(isPtt) { - const mods = [{ - mediaType: "audio", - codec: "opus", - enableDtx: true, - maxAverageBitrate: isPtt ? 12000 : undefined - }]; - return mods; -} -// generates keys for the map of transceivers -// kind is unfortunately a string rather than MediaType as this is the type of -// track.kind -function getTransceiverKey(purpose, kind) { - return purpose + ":" + kind; -} -class MatrixCall extends _typedEventEmitter.TypedEventEmitter { - // whether this call should have push-to-talk semantics - // This should be set by the consumer on incoming & outgoing calls. - - // A queue for candidates waiting to go out. - // We try to amalgamate candidates into a single candidate message where - // possible - - // our transceivers for each purpose and type of media - - // The party ID of the other side: undefined if we haven't chosen a partner - // yet, null if we have but they didn't send a party ID. - - // The logic of when & if a call is on hold is nontrivial and explained in is*OnHold - // This flag represents whether we want the other party to be on hold - - // the stats for the call at the point it ended. We can't get these after we - // tear the call down, so we just grab a snapshot before we stop the call. - // The typescript definitions have this type as 'any' :( - - // Perfect negotiation state: https://www.w3.org/TR/webrtc/#perfect-negotiation-example - - // If candidates arrive before we've picked an opponent (which, in particular, - // will happen if the opponent sends candidates eagerly before the user answers - // the call) we buffer them up here so we can then add the ones from the party we pick - - // Used to keep the timer for the delay before actually stopping our - // video track after muting (see setLocalVideoMuted) - - // Used to allow connection without Video and Audio. To establish a webrtc connection without media a Data channel is - // needed At the moment this property is true if we allow MatrixClient with isVoipWithNoMediaAllowed = true - - /** - * Construct a new Matrix Call. - * @param opts - Config options. - */ - constructor(opts) { - var _opts$forceTURN; - super(); - (0, _defineProperty2.default)(this, "roomId", void 0); - (0, _defineProperty2.default)(this, "callId", void 0); - (0, _defineProperty2.default)(this, "invitee", void 0); - (0, _defineProperty2.default)(this, "hangupParty", void 0); - (0, _defineProperty2.default)(this, "hangupReason", void 0); - (0, _defineProperty2.default)(this, "direction", void 0); - (0, _defineProperty2.default)(this, "ourPartyId", void 0); - (0, _defineProperty2.default)(this, "peerConn", void 0); - (0, _defineProperty2.default)(this, "toDeviceSeq", 0); - (0, _defineProperty2.default)(this, "isPtt", false); - (0, _defineProperty2.default)(this, "_state", CallState.Fledgling); - (0, _defineProperty2.default)(this, "client", void 0); - (0, _defineProperty2.default)(this, "forceTURN", void 0); - (0, _defineProperty2.default)(this, "turnServers", void 0); - (0, _defineProperty2.default)(this, "candidateSendQueue", []); - (0, _defineProperty2.default)(this, "candidateSendTries", 0); - (0, _defineProperty2.default)(this, "candidatesEnded", false); - (0, _defineProperty2.default)(this, "feeds", []); - (0, _defineProperty2.default)(this, "transceivers", new Map()); - (0, _defineProperty2.default)(this, "inviteOrAnswerSent", false); - (0, _defineProperty2.default)(this, "waitForLocalAVStream", false); - (0, _defineProperty2.default)(this, "successor", void 0); - (0, _defineProperty2.default)(this, "opponentMember", void 0); - (0, _defineProperty2.default)(this, "opponentVersion", void 0); - (0, _defineProperty2.default)(this, "opponentPartyId", void 0); - (0, _defineProperty2.default)(this, "opponentCaps", void 0); - (0, _defineProperty2.default)(this, "iceDisconnectedTimeout", void 0); - (0, _defineProperty2.default)(this, "inviteTimeout", void 0); - (0, _defineProperty2.default)(this, "removeTrackListeners", new Map()); - (0, _defineProperty2.default)(this, "remoteOnHold", false); - (0, _defineProperty2.default)(this, "callStatsAtEnd", void 0); - (0, _defineProperty2.default)(this, "makingOffer", false); - (0, _defineProperty2.default)(this, "ignoreOffer", false); - (0, _defineProperty2.default)(this, "responsePromiseChain", void 0); - (0, _defineProperty2.default)(this, "remoteCandidateBuffer", new Map()); - (0, _defineProperty2.default)(this, "remoteAssertedIdentity", void 0); - (0, _defineProperty2.default)(this, "remoteSDPStreamMetadata", void 0); - (0, _defineProperty2.default)(this, "callLengthInterval", void 0); - (0, _defineProperty2.default)(this, "callStartTime", void 0); - (0, _defineProperty2.default)(this, "opponentDeviceId", void 0); - (0, _defineProperty2.default)(this, "opponentDeviceInfo", void 0); - (0, _defineProperty2.default)(this, "opponentSessionId", void 0); - (0, _defineProperty2.default)(this, "groupCallId", void 0); - (0, _defineProperty2.default)(this, "stopVideoTrackTimer", void 0); - (0, _defineProperty2.default)(this, "isOnlyDataChannelAllowed", void 0); - (0, _defineProperty2.default)(this, "stats", void 0); - (0, _defineProperty2.default)(this, "gotLocalIceCandidate", event => { - if (event.candidate) { - if (this.candidatesEnded) { - _logger.logger.warn(`Call ${this.callId} gotLocalIceCandidate() got candidate after candidates have ended - ignoring!`); - return; - } - _logger.logger.debug(`Call ${this.callId} got local ICE ${event.candidate.sdpMid} ${event.candidate.candidate}`); - if (this.callHasEnded()) return; - - // As with the offer, note we need to make a copy of this object, not - // pass the original: that broke in Chrome ~m43. - if (event.candidate.candidate === "") { - this.queueCandidate(null); - } else { - this.queueCandidate(event.candidate); - } - } - }); - (0, _defineProperty2.default)(this, "onIceGatheringStateChange", event => { - var _this$peerConn; - _logger.logger.debug(`Call ${this.callId} onIceGatheringStateChange() ice gathering state changed to ${this.peerConn.iceGatheringState}`); - if (((_this$peerConn = this.peerConn) === null || _this$peerConn === void 0 ? void 0 : _this$peerConn.iceGatheringState) === "complete") { - this.queueCandidate(null); - } - }); - (0, _defineProperty2.default)(this, "getLocalOfferFailed", err => { - _logger.logger.error(`Call ${this.callId} getLocalOfferFailed() running`, err); - this.emit(CallEvent.Error, new CallError(CallErrorCode.LocalOfferFailed, "Failed to get local offer!", err), this); - this.terminate(CallParty.Local, CallErrorCode.LocalOfferFailed, false); - }); - (0, _defineProperty2.default)(this, "getUserMediaFailed", err => { - if (this.successor) { - this.successor.getUserMediaFailed(err); - return; - } - _logger.logger.warn(`Call ${this.callId} getUserMediaFailed() failed to get user media - ending call`, err); - this.emit(CallEvent.Error, new CallError(CallErrorCode.NoUserMedia, "Couldn't start capturing media! Is your microphone set up and " + "does this app have permission?", err), this); - this.terminate(CallParty.Local, CallErrorCode.NoUserMedia, false); - }); - (0, _defineProperty2.default)(this, "onIceConnectionStateChanged", () => { - var _this$peerConn2, _this$peerConn$iceCon, _this$peerConn3, _this$peerConn4, _this$peerConn6; - if (this.callHasEnded()) { - return; // because ICE can still complete as we're ending the call - } - - _logger.logger.debug(`Call ${this.callId} onIceConnectionStateChanged() running (state=${(_this$peerConn2 = this.peerConn) === null || _this$peerConn2 === void 0 ? void 0 : _this$peerConn2.iceConnectionState})`); - - // ideally we'd consider the call to be connected when we get media but - // chrome doesn't implement any of the 'onstarted' events yet - if (["connected", "completed"].includes((_this$peerConn$iceCon = (_this$peerConn3 = this.peerConn) === null || _this$peerConn3 === void 0 ? void 0 : _this$peerConn3.iceConnectionState) !== null && _this$peerConn$iceCon !== void 0 ? _this$peerConn$iceCon : "")) { - clearTimeout(this.iceDisconnectedTimeout); - this.iceDisconnectedTimeout = undefined; - this.state = CallState.Connected; - if (!this.callLengthInterval && !this.callStartTime) { - this.callStartTime = Date.now(); - this.callLengthInterval = setInterval(() => { - this.emit(CallEvent.LengthChanged, Math.round((Date.now() - this.callStartTime) / 1000), this); - }, CALL_LENGTH_INTERVAL); - } - } else if (((_this$peerConn4 = this.peerConn) === null || _this$peerConn4 === void 0 ? void 0 : _this$peerConn4.iceConnectionState) == "failed") { - var _this$peerConn5; - // Firefox for Android does not yet have support for restartIce() - // (the types say it's always defined though, so we have to cast - // to prevent typescript from warning). - if ((_this$peerConn5 = this.peerConn) !== null && _this$peerConn5 !== void 0 && _this$peerConn5.restartIce) { - this.candidatesEnded = false; - this.peerConn.restartIce(); - } else { - _logger.logger.info(`Call ${this.callId} onIceConnectionStateChanged() hanging up call (ICE failed and no ICE restart method)`); - this.hangup(CallErrorCode.IceFailed, false); - } - } else if (((_this$peerConn6 = this.peerConn) === null || _this$peerConn6 === void 0 ? void 0 : _this$peerConn6.iceConnectionState) == "disconnected") { - this.iceDisconnectedTimeout = setTimeout(() => { - _logger.logger.info(`Call ${this.callId} onIceConnectionStateChanged() hanging up call (ICE disconnected for too long)`); - this.hangup(CallErrorCode.IceFailed, false); - }, ICE_DISCONNECTED_TIMEOUT); - this.state = CallState.Connecting; - } - - // In PTT mode, override feed status to muted when we lose connection to - // the peer, since we don't want to block the line if they're not saying anything. - // Experimenting in Chrome, this happens after 5 or 6 seconds, which is probably - // fast enough. - if (this.isPtt && ["failed", "disconnected"].includes(this.peerConn.iceConnectionState)) { - for (const feed of this.getRemoteFeeds()) { - feed.setAudioVideoMuted(true, true); - } - } - }); - (0, _defineProperty2.default)(this, "onSignallingStateChanged", () => { - var _this$peerConn7; - _logger.logger.debug(`Call ${this.callId} onSignallingStateChanged() running (state=${(_this$peerConn7 = this.peerConn) === null || _this$peerConn7 === void 0 ? void 0 : _this$peerConn7.signalingState})`); - }); - (0, _defineProperty2.default)(this, "onTrack", ev => { - if (ev.streams.length === 0) { - _logger.logger.warn(`Call ${this.callId} onTrack() called with streamless track streamless (kind=${ev.track.kind})`); - return; - } - const stream = ev.streams[0]; - this.pushRemoteFeed(stream); - if (!this.removeTrackListeners.has(stream)) { - const onRemoveTrack = () => { - if (stream.getTracks().length === 0) { - _logger.logger.info(`Call ${this.callId} onTrack() removing track (streamId=${stream.id})`); - this.deleteFeedByStream(stream); - stream.removeEventListener("removetrack", onRemoveTrack); - this.removeTrackListeners.delete(stream); - } - }; - stream.addEventListener("removetrack", onRemoveTrack); - this.removeTrackListeners.set(stream, onRemoveTrack); - } - }); - (0, _defineProperty2.default)(this, "onDataChannel", ev => { - this.emit(CallEvent.DataChannel, ev.channel, this); - }); - (0, _defineProperty2.default)(this, "onNegotiationNeeded", async () => { - _logger.logger.info(`Call ${this.callId} onNegotiationNeeded() negotiation is needed!`); - if (this.state !== CallState.CreateOffer && this.opponentVersion === 0) { - _logger.logger.info(`Call ${this.callId} onNegotiationNeeded() opponent does not support renegotiation: ignoring negotiationneeded event`); - return; - } - this.queueGotLocalOffer(); - }); - (0, _defineProperty2.default)(this, "onHangupReceived", msg => { - _logger.logger.debug(`Call ${this.callId} onHangupReceived() running`); - - // party ID must match (our chosen partner hanging up the call) or be undefined (we haven't chosen - // a partner yet but we're treating the hangup as a reject as per VoIP v0) - if (this.partyIdMatches(msg) || this.state === CallState.Ringing) { - // default reason is user_hangup - this.terminate(CallParty.Remote, msg.reason || CallErrorCode.UserHangup, true); - } else { - _logger.logger.info(`Call ${this.callId} onHangupReceived() ignoring message from party ID ${msg.party_id}: our partner is ${this.opponentPartyId}`); - } - }); - (0, _defineProperty2.default)(this, "onRejectReceived", msg => { - _logger.logger.debug(`Call ${this.callId} onRejectReceived() running`); - - // No need to check party_id for reject because if we'd received either - // an answer or reject, we wouldn't be in state InviteSent - - const shouldTerminate = - // reject events also end the call if it's ringing: it's another of - // our devices rejecting the call. - [CallState.InviteSent, CallState.Ringing].includes(this.state) || - // also if we're in the init state and it's an inbound call, since - // this means we just haven't entered the ringing state yet - this.state === CallState.Fledgling && this.direction === CallDirection.Inbound; - if (shouldTerminate) { - this.terminate(CallParty.Remote, msg.reason || CallErrorCode.UserHangup, true); - } else { - _logger.logger.debug(`Call ${this.callId} onRejectReceived() called in wrong state (state=${this.state})`); - } - }); - (0, _defineProperty2.default)(this, "onAnsweredElsewhere", msg => { - _logger.logger.debug(`Call ${this.callId} onAnsweredElsewhere() running`); - this.terminate(CallParty.Remote, CallErrorCode.AnsweredElsewhere, true); - }); - this.roomId = opts.roomId; - this.invitee = opts.invitee; - this.client = opts.client; - if (!this.client.deviceId) throw new Error("Client must have a device ID to start calls"); - this.forceTURN = (_opts$forceTURN = opts.forceTURN) !== null && _opts$forceTURN !== void 0 ? _opts$forceTURN : false; - this.ourPartyId = this.client.deviceId; - this.opponentDeviceId = opts.opponentDeviceId; - this.opponentSessionId = opts.opponentSessionId; - this.groupCallId = opts.groupCallId; - // Array of Objects with urls, username, credential keys - this.turnServers = opts.turnServers || []; - if (this.turnServers.length === 0 && this.client.isFallbackICEServerAllowed()) { - this.turnServers.push({ - urls: [FALLBACK_ICE_SERVER] - }); - } - for (const server of this.turnServers) { - utils.checkObjectHasKeys(server, ["urls"]); - } - this.callId = genCallID(); - // If the Client provides calls without audio and video we need a datachannel for a webrtc connection - this.isOnlyDataChannelAllowed = this.client.isVoipWithNoMediaAllowed; - } - - /** - * Place a voice call to this room. - * @throws If you have not specified a listener for 'error' events. - */ - async placeVoiceCall() { - await this.placeCall(true, false); - } - - /** - * Place a video call to this room. - * @throws If you have not specified a listener for 'error' events. - */ - async placeVideoCall() { - await this.placeCall(true, true); - } - - /** - * Create a datachannel using this call's peer connection. - * @param label - A human readable label for this datachannel - * @param options - An object providing configuration options for the data channel. - */ - createDataChannel(label, options) { - const dataChannel = this.peerConn.createDataChannel(label, options); - this.emit(CallEvent.DataChannel, dataChannel, this); - return dataChannel; - } - getOpponentMember() { - return this.opponentMember; - } - getOpponentDeviceId() { - return this.opponentDeviceId; - } - getOpponentSessionId() { - return this.opponentSessionId; - } - opponentCanBeTransferred() { - return Boolean(this.opponentCaps && this.opponentCaps["m.call.transferee"]); - } - opponentSupportsDTMF() { - return Boolean(this.opponentCaps && this.opponentCaps["m.call.dtmf"]); - } - getRemoteAssertedIdentity() { - return this.remoteAssertedIdentity; - } - get state() { - return this._state; - } - set state(state) { - const oldState = this._state; - this._state = state; - this.emit(CallEvent.State, state, oldState, this); - } - get type() { - // we may want to look for a video receiver here rather than a track to match the - // sender behaviour, although in practice they should be the same thing - return this.hasUserMediaVideoSender || this.hasRemoteUserMediaVideoTrack ? CallType.Video : CallType.Voice; - } - get hasLocalUserMediaVideoTrack() { - var _this$localUsermediaS; - return !!((_this$localUsermediaS = this.localUsermediaStream) !== null && _this$localUsermediaS !== void 0 && _this$localUsermediaS.getVideoTracks().length); - } - get hasRemoteUserMediaVideoTrack() { - return this.getRemoteFeeds().some(feed => { - var _feed$stream; - return feed.purpose === _callEventTypes.SDPStreamMetadataPurpose.Usermedia && ((_feed$stream = feed.stream) === null || _feed$stream === void 0 ? void 0 : _feed$stream.getVideoTracks().length); - }); - } - get hasLocalUserMediaAudioTrack() { - var _this$localUsermediaS2; - return !!((_this$localUsermediaS2 = this.localUsermediaStream) !== null && _this$localUsermediaS2 !== void 0 && _this$localUsermediaS2.getAudioTracks().length); - } - get hasRemoteUserMediaAudioTrack() { - return this.getRemoteFeeds().some(feed => { - var _feed$stream2; - return feed.purpose === _callEventTypes.SDPStreamMetadataPurpose.Usermedia && !!((_feed$stream2 = feed.stream) !== null && _feed$stream2 !== void 0 && _feed$stream2.getAudioTracks().length); - }); - } - get hasUserMediaAudioSender() { - var _this$transceivers$ge; - return Boolean((_this$transceivers$ge = this.transceivers.get(getTransceiverKey(_callEventTypes.SDPStreamMetadataPurpose.Usermedia, "audio"))) === null || _this$transceivers$ge === void 0 ? void 0 : _this$transceivers$ge.sender); - } - get hasUserMediaVideoSender() { - var _this$transceivers$ge2; - return Boolean((_this$transceivers$ge2 = this.transceivers.get(getTransceiverKey(_callEventTypes.SDPStreamMetadataPurpose.Usermedia, "video"))) === null || _this$transceivers$ge2 === void 0 ? void 0 : _this$transceivers$ge2.sender); - } - get localUsermediaFeed() { - return this.getLocalFeeds().find(feed => feed.purpose === _callEventTypes.SDPStreamMetadataPurpose.Usermedia); - } - get localScreensharingFeed() { - return this.getLocalFeeds().find(feed => feed.purpose === _callEventTypes.SDPStreamMetadataPurpose.Screenshare); - } - get localUsermediaStream() { - var _this$localUsermediaF; - return (_this$localUsermediaF = this.localUsermediaFeed) === null || _this$localUsermediaF === void 0 ? void 0 : _this$localUsermediaF.stream; - } - get localScreensharingStream() { - var _this$localScreenshar; - return (_this$localScreenshar = this.localScreensharingFeed) === null || _this$localScreenshar === void 0 ? void 0 : _this$localScreenshar.stream; - } - get remoteUsermediaFeed() { - return this.getRemoteFeeds().find(feed => feed.purpose === _callEventTypes.SDPStreamMetadataPurpose.Usermedia); - } - get remoteScreensharingFeed() { - return this.getRemoteFeeds().find(feed => feed.purpose === _callEventTypes.SDPStreamMetadataPurpose.Screenshare); - } - get remoteUsermediaStream() { - var _this$remoteUsermedia; - return (_this$remoteUsermedia = this.remoteUsermediaFeed) === null || _this$remoteUsermedia === void 0 ? void 0 : _this$remoteUsermedia.stream; - } - get remoteScreensharingStream() { - var _this$remoteScreensha; - return (_this$remoteScreensha = this.remoteScreensharingFeed) === null || _this$remoteScreensha === void 0 ? void 0 : _this$remoteScreensha.stream; - } - getFeedByStreamId(streamId) { - return this.getFeeds().find(feed => feed.stream.id === streamId); - } - - /** - * Returns an array of all CallFeeds - * @returns CallFeeds - */ - getFeeds() { - return this.feeds; - } - - /** - * Returns an array of all local CallFeeds - * @returns local CallFeeds - */ - getLocalFeeds() { - return this.feeds.filter(feed => feed.isLocal()); - } - - /** - * Returns an array of all remote CallFeeds - * @returns remote CallFeeds - */ - getRemoteFeeds() { - return this.feeds.filter(feed => !feed.isLocal()); - } - async initOpponentCrypto() { - var _this$getOpponentMemb, _deviceInfoMap$get; - if (!this.opponentDeviceId) return; - if (!this.client.getUseE2eForGroupCall()) return; - // It's possible to want E2EE and yet not have the means to manage E2EE - // ourselves (for example if the client is a RoomWidgetClient) - if (!this.client.isCryptoEnabled()) { - // All we know is the device ID - this.opponentDeviceInfo = new _deviceinfo.DeviceInfo(this.opponentDeviceId); - return; - } - // if we've got to this point, we do want to init crypto, so throw if we can't - if (!this.client.crypto) throw new Error("Crypto is not initialised."); - const userId = this.invitee || ((_this$getOpponentMemb = this.getOpponentMember()) === null || _this$getOpponentMemb === void 0 ? void 0 : _this$getOpponentMemb.userId); - if (!userId) throw new Error("Couldn't find opponent user ID to init crypto"); - const deviceInfoMap = await this.client.crypto.deviceList.downloadKeys([userId], false); - this.opponentDeviceInfo = (_deviceInfoMap$get = deviceInfoMap.get(userId)) === null || _deviceInfoMap$get === void 0 ? void 0 : _deviceInfoMap$get.get(this.opponentDeviceId); - if (this.opponentDeviceInfo === undefined) { - throw new _groupCall.GroupCallUnknownDeviceError(userId); - } - } - - /** - * Generates and returns localSDPStreamMetadata - * @returns localSDPStreamMetadata - */ - getLocalSDPStreamMetadata(updateStreamIds = false) { - const metadata = {}; - for (const localFeed of this.getLocalFeeds()) { - if (updateStreamIds) { - localFeed.sdpMetadataStreamId = localFeed.stream.id; - } - metadata[localFeed.sdpMetadataStreamId] = { - purpose: localFeed.purpose, - audio_muted: localFeed.isAudioMuted(), - video_muted: localFeed.isVideoMuted() - }; - } - return metadata; - } - - /** - * Returns true if there are no incoming feeds, - * otherwise returns false - * @returns no incoming feeds - */ - noIncomingFeeds() { - return !this.feeds.some(feed => !feed.isLocal()); - } - pushRemoteFeed(stream) { - // Fallback to old behavior if the other side doesn't support SDPStreamMetadata - if (!this.opponentSupportsSDPStreamMetadata()) { - this.pushRemoteFeedWithoutMetadata(stream); - return; - } - const userId = this.getOpponentMember().userId; - const purpose = this.remoteSDPStreamMetadata[stream.id].purpose; - const audioMuted = this.remoteSDPStreamMetadata[stream.id].audio_muted; - const videoMuted = this.remoteSDPStreamMetadata[stream.id].video_muted; - if (!purpose) { - _logger.logger.warn(`Call ${this.callId} pushRemoteFeed() ignoring stream because we didn't get any metadata about it (streamId=${stream.id})`); - return; - } - if (this.getFeedByStreamId(stream.id)) { - _logger.logger.warn(`Call ${this.callId} pushRemoteFeed() ignoring stream because we already have a feed for it (streamId=${stream.id})`); - return; - } - this.feeds.push(new _callFeed.CallFeed({ - client: this.client, - call: this, - roomId: this.roomId, - userId, - deviceId: this.getOpponentDeviceId(), - stream, - purpose, - audioMuted, - videoMuted - })); - this.emit(CallEvent.FeedsChanged, this.feeds, this); - _logger.logger.info(`Call ${this.callId} pushRemoteFeed() pushed stream (streamId=${stream.id}, active=${stream.active}, purpose=${purpose})`); - } - - /** - * This method is used ONLY if the other client doesn't support sending SDPStreamMetadata - */ - pushRemoteFeedWithoutMetadata(stream) { - var _this$feeds$find; - const userId = this.getOpponentMember().userId; - // We can guess the purpose here since the other client can only send one stream - const purpose = _callEventTypes.SDPStreamMetadataPurpose.Usermedia; - const oldRemoteStream = (_this$feeds$find = this.feeds.find(feed => !feed.isLocal())) === null || _this$feeds$find === void 0 ? void 0 : _this$feeds$find.stream; - - // Note that we check by ID and always set the remote stream: Chrome appears - // to make new stream objects when transceiver directionality is changed and the 'active' - // status of streams change - Dave - // If we already have a stream, check this stream has the same id - if (oldRemoteStream && stream.id !== oldRemoteStream.id) { - _logger.logger.warn(`Call ${this.callId} pushRemoteFeedWithoutMetadata() ignoring new stream because we already have stream (streamId=${stream.id})`); - return; - } - if (this.getFeedByStreamId(stream.id)) { - _logger.logger.warn(`Call ${this.callId} pushRemoteFeedWithoutMetadata() ignoring stream because we already have a feed for it (streamId=${stream.id})`); - return; - } - this.feeds.push(new _callFeed.CallFeed({ - client: this.client, - call: this, - roomId: this.roomId, - audioMuted: false, - videoMuted: false, - userId, - deviceId: this.getOpponentDeviceId(), - stream, - purpose - })); - this.emit(CallEvent.FeedsChanged, this.feeds, this); - _logger.logger.info(`Call ${this.callId} pushRemoteFeedWithoutMetadata() pushed stream (streamId=${stream.id}, active=${stream.active})`); - } - pushNewLocalFeed(stream, purpose, addToPeerConnection = true) { - const userId = this.client.getUserId(); - - // Tracks don't always start off enabled, eg. chrome will give a disabled - // audio track if you ask for user media audio and already had one that - // you'd set to disabled (presumably because it clones them internally). - setTracksEnabled(stream.getAudioTracks(), true); - setTracksEnabled(stream.getVideoTracks(), true); - if (this.getFeedByStreamId(stream.id)) { - _logger.logger.warn(`Call ${this.callId} pushNewLocalFeed() ignoring stream because we already have a feed for it (streamId=${stream.id})`); - return; - } - this.pushLocalFeed(new _callFeed.CallFeed({ - client: this.client, - roomId: this.roomId, - audioMuted: false, - videoMuted: false, - userId, - deviceId: this.getOpponentDeviceId(), - stream, - purpose - }), addToPeerConnection); - } - - /** - * Pushes supplied feed to the call - * @param callFeed - to push - * @param addToPeerConnection - whether to add the tracks to the peer connection - */ - pushLocalFeed(callFeed, addToPeerConnection = true) { - if (this.feeds.some(feed => callFeed.stream.id === feed.stream.id)) { - _logger.logger.info(`Call ${this.callId} pushLocalFeed() ignoring duplicate local stream (streamId=${callFeed.stream.id})`); - return; - } - this.feeds.push(callFeed); - if (addToPeerConnection) { - for (const track of callFeed.stream.getTracks()) { - _logger.logger.info(`Call ${this.callId} pushLocalFeed() adding track to peer connection (id=${track.id}, kind=${track.kind}, streamId=${callFeed.stream.id}, streamPurpose=${callFeed.purpose}, enabled=${track.enabled})`); - const tKey = getTransceiverKey(callFeed.purpose, track.kind); - if (this.transceivers.has(tKey)) { - // we already have a sender, so we re-use it. We try to re-use transceivers as much - // as possible because they can't be removed once added, so otherwise they just - // accumulate which makes the SDP very large very quickly: in fact it only takes - // about 6 video tracks to exceed the maximum size of an Olm-encrypted - // Matrix event. - const transceiver = this.transceivers.get(tKey); - transceiver.sender.replaceTrack(track); - // set the direction to indicate we're going to start sending again - // (this will trigger the re-negotiation) - transceiver.direction = transceiver.direction === "inactive" ? "sendonly" : "sendrecv"; - } else { - // create a new one. We need to use addTrack rather addTransceiver for this because firefox - // doesn't yet implement RTCRTPSender.setStreams() - // (https://bugzilla.mozilla.org/show_bug.cgi?id=1510802) so we'd have no way to group the - // two tracks together into a stream. - const newSender = this.peerConn.addTrack(track, callFeed.stream); - - // now go & fish for the new transceiver - const newTransceiver = this.peerConn.getTransceivers().find(t => t.sender === newSender); - if (newTransceiver) { - this.transceivers.set(tKey, newTransceiver); - } else { - _logger.logger.warn(`Call ${this.callId} pushLocalFeed() didn't find a matching transceiver after adding track!`); - } - } - } - } - _logger.logger.info(`Call ${this.callId} pushLocalFeed() pushed stream (id=${callFeed.stream.id}, active=${callFeed.stream.active}, purpose=${callFeed.purpose})`); - this.emit(CallEvent.FeedsChanged, this.feeds, this); - } - - /** - * Removes local call feed from the call and its tracks from the peer - * connection - * @param callFeed - to remove - */ - removeLocalFeed(callFeed) { - const audioTransceiverKey = getTransceiverKey(callFeed.purpose, "audio"); - const videoTransceiverKey = getTransceiverKey(callFeed.purpose, "video"); - for (const transceiverKey of [audioTransceiverKey, videoTransceiverKey]) { - // this is slightly mixing the track and transceiver API but is basically just shorthand. - // There is no way to actually remove a transceiver, so this just sets it to inactive - // (or recvonly) and replaces the source with nothing. - if (this.transceivers.has(transceiverKey)) { - const transceiver = this.transceivers.get(transceiverKey); - if (transceiver.sender) this.peerConn.removeTrack(transceiver.sender); - } - } - if (callFeed.purpose === _callEventTypes.SDPStreamMetadataPurpose.Screenshare) { - this.client.getMediaHandler().stopScreensharingStream(callFeed.stream); - } - this.deleteFeed(callFeed); - } - deleteAllFeeds() { - for (const feed of this.feeds) { - if (!feed.isLocal() || !this.groupCallId) { - feed.dispose(); - } - } - this.feeds = []; - this.emit(CallEvent.FeedsChanged, this.feeds, this); - } - deleteFeedByStream(stream) { - const feed = this.getFeedByStreamId(stream.id); - if (!feed) { - _logger.logger.warn(`Call ${this.callId} deleteFeedByStream() didn't find the feed to delete (streamId=${stream.id})`); - return; - } - this.deleteFeed(feed); - } - deleteFeed(feed) { - feed.dispose(); - this.feeds.splice(this.feeds.indexOf(feed), 1); - this.emit(CallEvent.FeedsChanged, this.feeds, this); - } - - // The typescript definitions have this type as 'any' :( - async getCurrentCallStats() { - if (this.callHasEnded()) { - return this.callStatsAtEnd; - } - return this.collectCallStats(); - } - async collectCallStats() { - // This happens when the call fails before it starts. - // For example when we fail to get capture sources - if (!this.peerConn) return; - const statsReport = await this.peerConn.getStats(); - const stats = []; - statsReport.forEach(item => { - stats.push(item); - }); - return stats; - } - - /** - * Configure this call from an invite event. Used by MatrixClient. - * @param event - The m.call.invite event - */ - async initWithInvite(event) { - var _this$feeds$find2; - const invite = event.getContent(); - this.direction = CallDirection.Inbound; - - // make sure we have valid turn creds. Unless something's gone wrong, it should - // poll and keep the credentials valid so this should be instant. - const haveTurnCreds = await this.client.checkTurnServers(); - if (!haveTurnCreds) { - _logger.logger.warn(`Call ${this.callId} initWithInvite() failed to get TURN credentials! Proceeding with call anyway...`); - } - const sdpStreamMetadata = invite[_callEventTypes.SDPStreamMetadataKey]; - if (sdpStreamMetadata) { - this.updateRemoteSDPStreamMetadata(sdpStreamMetadata); - } else { - _logger.logger.debug(`Call ${this.callId} initWithInvite() did not get any SDPStreamMetadata! Can not send/receive multiple streams`); - } - this.peerConn = this.createPeerConnection(); - // we must set the party ID before await-ing on anything: the call event - // handler will start giving us more call events (eg. candidates) so if - // we haven't set the party ID, we'll ignore them. - this.chooseOpponent(event); - await this.initOpponentCrypto(); - try { - await this.peerConn.setRemoteDescription(invite.offer); - await this.addBufferedIceCandidates(); - } catch (e) { - _logger.logger.debug(`Call ${this.callId} initWithInvite() failed to set remote description`, e); - this.terminate(CallParty.Local, CallErrorCode.SetRemoteDescription, false); - return; - } - const remoteStream = (_this$feeds$find2 = this.feeds.find(feed => !feed.isLocal())) === null || _this$feeds$find2 === void 0 ? void 0 : _this$feeds$find2.stream; - - // According to previous comments in this file, firefox at some point did not - // add streams until media started arriving on them. Testing latest firefox - // (81 at time of writing), this is no longer a problem, so let's do it the correct way. - // - // For example in case of no media webrtc connections like screen share only call we have to allow webrtc - // connections without remote media. In this case we always use a data channel. At the moment we allow as well - // only data channel as media in the WebRTC connection with this setup here. - if (!this.isOnlyDataChannelAllowed && (!remoteStream || remoteStream.getTracks().length === 0)) { - _logger.logger.error(`Call ${this.callId} initWithInvite() no remote stream or no tracks after setting remote description!`); - this.terminate(CallParty.Local, CallErrorCode.SetRemoteDescription, false); - return; - } - this.state = CallState.Ringing; - if (event.getLocalAge()) { - // Time out the call if it's ringing for too long - const ringingTimer = setTimeout(() => { - if (this.state == CallState.Ringing) { - var _this$stats; - _logger.logger.debug(`Call ${this.callId} initWithInvite() invite has expired. Hanging up.`); - this.hangupParty = CallParty.Remote; // effectively - this.state = CallState.Ended; - this.stopAllMedia(); - if (this.peerConn.signalingState != "closed") { - this.peerConn.close(); - } - (_this$stats = this.stats) === null || _this$stats === void 0 ? void 0 : _this$stats.removeStatsReportGatherer(this.callId); - this.emit(CallEvent.Hangup, this); - } - }, invite.lifetime - event.getLocalAge()); - const onState = state => { - if (state !== CallState.Ringing) { - clearTimeout(ringingTimer); - this.off(CallEvent.State, onState); - } - }; - this.on(CallEvent.State, onState); - } - } - - /** - * Configure this call from a hangup or reject event. Used by MatrixClient. - * @param event - The m.call.hangup event - */ - initWithHangup(event) { - // perverse as it may seem, sometimes we want to instantiate a call with a - // hangup message (because when getting the state of the room on load, events - // come in reverse order and we want to remember that a call has been hung up) - this.state = CallState.Ended; - } - shouldAnswerWithMediaType(wantedValue, valueOfTheOtherSide, type) { - if (wantedValue && !valueOfTheOtherSide) { - // TODO: Figure out how to do this - _logger.logger.warn(`Call ${this.callId} shouldAnswerWithMediaType() unable to answer with ${type} because the other side isn't sending it either.`); - return false; - } else if (!utils.isNullOrUndefined(wantedValue) && wantedValue !== valueOfTheOtherSide && !this.opponentSupportsSDPStreamMetadata()) { - _logger.logger.warn(`Call ${this.callId} shouldAnswerWithMediaType() unable to answer with ${type}=${wantedValue} because the other side doesn't support it. Answering with ${type}=${valueOfTheOtherSide}.`); - return valueOfTheOtherSide; - } - return wantedValue !== null && wantedValue !== void 0 ? wantedValue : valueOfTheOtherSide; - } - - /** - * Answer a call. - */ - async answer(audio, video) { - if (this.inviteOrAnswerSent) return; - // TODO: Figure out how to do this - if (audio === false && video === false) throw new Error("You CANNOT answer a call without media"); - if (!this.localUsermediaStream && !this.waitForLocalAVStream) { - const prevState = this.state; - const answerWithAudio = this.shouldAnswerWithMediaType(audio, this.hasRemoteUserMediaAudioTrack, "audio"); - const answerWithVideo = this.shouldAnswerWithMediaType(video, this.hasRemoteUserMediaVideoTrack, "video"); - this.state = CallState.WaitLocalMedia; - this.waitForLocalAVStream = true; - try { - var _this$client$getDevic; - const stream = await this.client.getMediaHandler().getUserMediaStream(answerWithAudio, answerWithVideo); - this.waitForLocalAVStream = false; - const usermediaFeed = new _callFeed.CallFeed({ - client: this.client, - roomId: this.roomId, - userId: this.client.getUserId(), - deviceId: (_this$client$getDevic = this.client.getDeviceId()) !== null && _this$client$getDevic !== void 0 ? _this$client$getDevic : undefined, - stream, - purpose: _callEventTypes.SDPStreamMetadataPurpose.Usermedia, - audioMuted: false, - videoMuted: false - }); - const feeds = [usermediaFeed]; - if (this.localScreensharingFeed) { - feeds.push(this.localScreensharingFeed); - } - this.answerWithCallFeeds(feeds); - } catch (e) { - if (answerWithVideo) { - // Try to answer without video - _logger.logger.warn(`Call ${this.callId} answer() failed to getUserMedia(), trying to getUserMedia() without video`); - this.state = prevState; - this.waitForLocalAVStream = false; - await this.answer(answerWithAudio, false); - } else { - this.getUserMediaFailed(e); - return; - } - } - } else if (this.waitForLocalAVStream) { - this.state = CallState.WaitLocalMedia; - } - } - answerWithCallFeeds(callFeeds) { - if (this.inviteOrAnswerSent) return; - this.queueGotCallFeedsForAnswer(callFeeds); - } - - /** - * Replace this call with a new call, e.g. for glare resolution. Used by - * MatrixClient. - * @param newCall - The new call. - */ - replacedBy(newCall) { - _logger.logger.debug(`Call ${this.callId} replacedBy() running (newCallId=${newCall.callId})`); - if (this.state === CallState.WaitLocalMedia) { - _logger.logger.debug(`Call ${this.callId} replacedBy() telling new call to wait for local media (newCallId=${newCall.callId})`); - newCall.waitForLocalAVStream = true; - } else if ([CallState.CreateOffer, CallState.InviteSent].includes(this.state)) { - if (newCall.direction === CallDirection.Outbound) { - newCall.queueGotCallFeedsForAnswer([]); - } else { - _logger.logger.debug(`Call ${this.callId} replacedBy() handing local stream to new call(newCallId=${newCall.callId})`); - newCall.queueGotCallFeedsForAnswer(this.getLocalFeeds().map(feed => feed.clone())); - } - } - this.successor = newCall; - this.emit(CallEvent.Replaced, newCall, this); - this.hangup(CallErrorCode.Replaced, true); - } - - /** - * Hangup a call. - * @param reason - The reason why the call is being hung up. - * @param suppressEvent - True to suppress emitting an event. - */ - hangup(reason, suppressEvent) { - if (this.callHasEnded()) return; - _logger.logger.debug(`Call ${this.callId} hangup() ending call (reason=${reason})`); - this.terminate(CallParty.Local, reason, !suppressEvent); - // We don't want to send hangup here if we didn't even get to sending an invite - if ([CallState.Fledgling, CallState.WaitLocalMedia].includes(this.state)) return; - const content = {}; - // Don't send UserHangup reason to older clients - if (this.opponentVersion && this.opponentVersion !== 0 || reason !== CallErrorCode.UserHangup) { - content["reason"] = reason; - } - this.sendVoipEvent(_event.EventType.CallHangup, content); - } - - /** - * Reject a call - * This used to be done by calling hangup, but is a separate method and protocol - * event as of MSC2746. - */ - reject() { - if (this.state !== CallState.Ringing) { - throw Error("Call must be in 'ringing' state to reject!"); - } - if (this.opponentVersion === 0) { - _logger.logger.info(`Call ${this.callId} reject() opponent version is less than 1: sending hangup instead of reject (opponentVersion=${this.opponentVersion})`); - this.hangup(CallErrorCode.UserHangup, true); - return; - } - _logger.logger.debug("Rejecting call: " + this.callId); - this.terminate(CallParty.Local, CallErrorCode.UserHangup, true); - this.sendVoipEvent(_event.EventType.CallReject, {}); - } - - /** - * Adds an audio and/or video track - upgrades the call - * @param audio - should add an audio track - * @param video - should add an video track - */ - async upgradeCall(audio, video) { - // We don't do call downgrades - if (!audio && !video) return; - if (!this.opponentSupportsSDPStreamMetadata()) return; - try { - _logger.logger.debug(`Call ${this.callId} upgradeCall() upgrading call (audio=${audio}, video=${video})`); - const getAudio = audio || this.hasLocalUserMediaAudioTrack; - const getVideo = video || this.hasLocalUserMediaVideoTrack; - - // updateLocalUsermediaStream() will take the tracks, use them as - // replacement and throw the stream away, so it isn't reusable - const stream = await this.client.getMediaHandler().getUserMediaStream(getAudio, getVideo, false); - await this.updateLocalUsermediaStream(stream, audio, video); - } catch (error) { - _logger.logger.error(`Call ${this.callId} upgradeCall() failed to upgrade the call`, error); - this.emit(CallEvent.Error, new CallError(CallErrorCode.NoUserMedia, "Failed to get camera access: ", error), this); - } - } - - /** - * Returns true if this.remoteSDPStreamMetadata is defined, otherwise returns false - * @returns can screenshare - */ - opponentSupportsSDPStreamMetadata() { - return Boolean(this.remoteSDPStreamMetadata); - } - - /** - * If there is a screensharing stream returns true, otherwise returns false - * @returns is screensharing - */ - isScreensharing() { - return Boolean(this.localScreensharingStream); - } - - /** - * Starts/stops screensharing - * @param enabled - the desired screensharing state - * @param desktopCapturerSourceId - optional id of the desktop capturer source to use - * @returns new screensharing state - */ - async setScreensharingEnabled(enabled, opts) { - // Skip if there is nothing to do - if (enabled && this.isScreensharing()) { - _logger.logger.warn(`Call ${this.callId} setScreensharingEnabled() there is already a screensharing stream - there is nothing to do!`); - return true; - } else if (!enabled && !this.isScreensharing()) { - _logger.logger.warn(`Call ${this.callId} setScreensharingEnabled() there already isn't a screensharing stream - there is nothing to do!`); - return false; - } - - // Fallback to replaceTrack() - if (!this.opponentSupportsSDPStreamMetadata()) { - return this.setScreensharingEnabledWithoutMetadataSupport(enabled, opts); - } - _logger.logger.debug(`Call ${this.callId} setScreensharingEnabled() running (enabled=${enabled})`); - if (enabled) { - try { - const stream = await this.client.getMediaHandler().getScreensharingStream(opts); - if (!stream) return false; - this.pushNewLocalFeed(stream, _callEventTypes.SDPStreamMetadataPurpose.Screenshare); - return true; - } catch (err) { - _logger.logger.error(`Call ${this.callId} setScreensharingEnabled() failed to get screen-sharing stream:`, err); - return false; - } - } else { - const audioTransceiver = this.transceivers.get(getTransceiverKey(_callEventTypes.SDPStreamMetadataPurpose.Screenshare, "audio")); - const videoTransceiver = this.transceivers.get(getTransceiverKey(_callEventTypes.SDPStreamMetadataPurpose.Screenshare, "video")); - for (const transceiver of [audioTransceiver, videoTransceiver]) { - // this is slightly mixing the track and transceiver API but is basically just shorthand - // for removing the sender. - if (transceiver && transceiver.sender) this.peerConn.removeTrack(transceiver.sender); - } - this.client.getMediaHandler().stopScreensharingStream(this.localScreensharingStream); - this.deleteFeedByStream(this.localScreensharingStream); - return false; - } - } - - /** - * Starts/stops screensharing - * Should be used ONLY if the opponent doesn't support SDPStreamMetadata - * @param enabled - the desired screensharing state - * @param desktopCapturerSourceId - optional id of the desktop capturer source to use - * @returns new screensharing state - */ - async setScreensharingEnabledWithoutMetadataSupport(enabled, opts) { - _logger.logger.debug(`Call ${this.callId} setScreensharingEnabledWithoutMetadataSupport() running (enabled=${enabled})`); - if (enabled) { - try { - var _this$transceivers$ge3; - const stream = await this.client.getMediaHandler().getScreensharingStream(opts); - if (!stream) return false; - const track = stream.getTracks().find(track => track.kind === "video"); - const sender = (_this$transceivers$ge3 = this.transceivers.get(getTransceiverKey(_callEventTypes.SDPStreamMetadataPurpose.Usermedia, "video"))) === null || _this$transceivers$ge3 === void 0 ? void 0 : _this$transceivers$ge3.sender; - sender === null || sender === void 0 ? void 0 : sender.replaceTrack(track !== null && track !== void 0 ? track : null); - this.pushNewLocalFeed(stream, _callEventTypes.SDPStreamMetadataPurpose.Screenshare, false); - return true; - } catch (err) { - _logger.logger.error(`Call ${this.callId} setScreensharingEnabledWithoutMetadataSupport() failed to get screen-sharing stream:`, err); - return false; - } - } else { - var _this$localUsermediaS3, _this$transceivers$ge4; - const track = (_this$localUsermediaS3 = this.localUsermediaStream) === null || _this$localUsermediaS3 === void 0 ? void 0 : _this$localUsermediaS3.getTracks().find(track => track.kind === "video"); - const sender = (_this$transceivers$ge4 = this.transceivers.get(getTransceiverKey(_callEventTypes.SDPStreamMetadataPurpose.Usermedia, "video"))) === null || _this$transceivers$ge4 === void 0 ? void 0 : _this$transceivers$ge4.sender; - sender === null || sender === void 0 ? void 0 : sender.replaceTrack(track !== null && track !== void 0 ? track : null); - this.client.getMediaHandler().stopScreensharingStream(this.localScreensharingStream); - this.deleteFeedByStream(this.localScreensharingStream); - return false; - } - } - - /** - * Replaces/adds the tracks from the passed stream to the localUsermediaStream - * @param stream - to use a replacement for the local usermedia stream - */ - async updateLocalUsermediaStream(stream, forceAudio = false, forceVideo = false) { - const callFeed = this.localUsermediaFeed; - const audioEnabled = forceAudio || !callFeed.isAudioMuted() && !this.remoteOnHold; - const videoEnabled = forceVideo || !callFeed.isVideoMuted() && !this.remoteOnHold; - _logger.logger.log(`Call ${this.callId} updateLocalUsermediaStream() running (streamId=${stream.id}, audio=${audioEnabled}, video=${videoEnabled})`); - setTracksEnabled(stream.getAudioTracks(), audioEnabled); - setTracksEnabled(stream.getVideoTracks(), videoEnabled); - - // We want to keep the same stream id, so we replace the tracks rather - // than the whole stream. - - // Firstly, we replace the tracks in our localUsermediaStream. - for (const track of this.localUsermediaStream.getTracks()) { - this.localUsermediaStream.removeTrack(track); - track.stop(); - } - for (const track of stream.getTracks()) { - this.localUsermediaStream.addTrack(track); - } - - // Then replace the old tracks, if possible. - for (const track of stream.getTracks()) { - const tKey = getTransceiverKey(_callEventTypes.SDPStreamMetadataPurpose.Usermedia, track.kind); - const transceiver = this.transceivers.get(tKey); - const oldSender = transceiver === null || transceiver === void 0 ? void 0 : transceiver.sender; - let added = false; - if (oldSender) { - try { - _logger.logger.info(`Call ${this.callId} updateLocalUsermediaStream() replacing track (id=${track.id}, kind=${track.kind}, streamId=${stream.id}, streamPurpose=${callFeed.purpose})`); - await oldSender.replaceTrack(track); - // Set the direction to indicate we're going to be sending. - // This is only necessary in the cases where we're upgrading - // the call to video after downgrading it. - transceiver.direction = transceiver.direction === "inactive" ? "sendonly" : "sendrecv"; - added = true; - } catch (error) { - _logger.logger.warn(`Call ${this.callId} updateLocalUsermediaStream() replaceTrack failed: adding new transceiver instead`, error); - } - } - if (!added) { - _logger.logger.info(`Call ${this.callId} updateLocalUsermediaStream() adding track to peer connection (id=${track.id}, kind=${track.kind}, streamId=${stream.id}, streamPurpose=${callFeed.purpose})`); - const newSender = this.peerConn.addTrack(track, this.localUsermediaStream); - const newTransceiver = this.peerConn.getTransceivers().find(t => t.sender === newSender); - if (newTransceiver) { - this.transceivers.set(tKey, newTransceiver); - } else { - _logger.logger.warn(`Call ${this.callId} updateLocalUsermediaStream() couldn't find matching transceiver for newly added track!`); - } - } - } - } - - /** - * Set whether our outbound video should be muted or not. - * @param muted - True to mute the outbound video. - * @returns the new mute state - */ - async setLocalVideoMuted(muted) { - var _this$localUsermediaF3; - _logger.logger.log(`Call ${this.callId} setLocalVideoMuted() running ${muted}`); - - // if we were still thinking about stopping and removing the video - // track: don't, because we want it back. - if (!muted && this.stopVideoTrackTimer !== undefined) { - clearTimeout(this.stopVideoTrackTimer); - this.stopVideoTrackTimer = undefined; - } - if (!(await this.client.getMediaHandler().hasVideoDevice())) { - return this.isLocalVideoMuted(); - } - if (!this.hasUserMediaVideoSender && !muted) { - var _this$localUsermediaF2; - (_this$localUsermediaF2 = this.localUsermediaFeed) === null || _this$localUsermediaF2 === void 0 ? void 0 : _this$localUsermediaF2.setAudioVideoMuted(null, muted); - await this.upgradeCall(false, true); - return this.isLocalVideoMuted(); - } - - // we may not have a video track - if not, re-request usermedia - if (!muted && this.localUsermediaStream.getVideoTracks().length === 0) { - const stream = await this.client.getMediaHandler().getUserMediaStream(true, true); - await this.updateLocalUsermediaStream(stream); - } - (_this$localUsermediaF3 = this.localUsermediaFeed) === null || _this$localUsermediaF3 === void 0 ? void 0 : _this$localUsermediaF3.setAudioVideoMuted(null, muted); - this.updateMuteStatus(); - await this.sendMetadataUpdate(); - - // if we're muting video, set a timeout to stop & remove the video track so we release - // the camera. We wait a short time to do this because when we disable a track, WebRTC - // will send black video for it. If we just stop and remove it straight away, the video - // will just freeze which means that when we unmute video, the other side will briefly - // get a static frame of us from before we muted. This way, the still frame is just black. - // A very small delay is not always enough so the theory here is that it needs to be long - // enough for WebRTC to encode a frame: 120ms should be long enough even if we're only - // doing 10fps. - if (muted) { - this.stopVideoTrackTimer = setTimeout(() => { - for (const t of this.localUsermediaStream.getVideoTracks()) { - t.stop(); - this.localUsermediaStream.removeTrack(t); - } - }, 120); - } - return this.isLocalVideoMuted(); - } - - /** - * Check if local video is muted. - * - * If there are multiple video tracks, <i>all</i> of the tracks need to be muted - * for this to return true. This means if there are no video tracks, this will - * return true. - * @returns True if the local preview video is muted, else false - * (including if the call is not set up yet). - */ - isLocalVideoMuted() { - var _this$localUsermediaF4, _this$localUsermediaF5; - return (_this$localUsermediaF4 = (_this$localUsermediaF5 = this.localUsermediaFeed) === null || _this$localUsermediaF5 === void 0 ? void 0 : _this$localUsermediaF5.isVideoMuted()) !== null && _this$localUsermediaF4 !== void 0 ? _this$localUsermediaF4 : false; - } - - /** - * Set whether the microphone should be muted or not. - * @param muted - True to mute the mic. - * @returns the new mute state - */ - async setMicrophoneMuted(muted) { - var _this$localUsermediaF6; - _logger.logger.log(`Call ${this.callId} setMicrophoneMuted() running ${muted}`); - if (!(await this.client.getMediaHandler().hasAudioDevice())) { - return this.isMicrophoneMuted(); - } - if (!muted && (!this.hasUserMediaAudioSender || !this.hasLocalUserMediaAudioTrack)) { - await this.upgradeCall(true, false); - return this.isMicrophoneMuted(); - } - (_this$localUsermediaF6 = this.localUsermediaFeed) === null || _this$localUsermediaF6 === void 0 ? void 0 : _this$localUsermediaF6.setAudioVideoMuted(muted, null); - this.updateMuteStatus(); - await this.sendMetadataUpdate(); - return this.isMicrophoneMuted(); - } - - /** - * Check if the microphone is muted. - * - * If there are multiple audio tracks, <i>all</i> of the tracks need to be muted - * for this to return true. This means if there are no audio tracks, this will - * return true. - * @returns True if the mic is muted, else false (including if the call - * is not set up yet). - */ - isMicrophoneMuted() { - var _this$localUsermediaF7, _this$localUsermediaF8; - return (_this$localUsermediaF7 = (_this$localUsermediaF8 = this.localUsermediaFeed) === null || _this$localUsermediaF8 === void 0 ? void 0 : _this$localUsermediaF8.isAudioMuted()) !== null && _this$localUsermediaF7 !== void 0 ? _this$localUsermediaF7 : false; - } - - /** - * @returns true if we have put the party on the other side of the call on hold - * (that is, we are signalling to them that we are not listening) - */ - isRemoteOnHold() { - return this.remoteOnHold; - } - setRemoteOnHold(onHold) { - if (this.isRemoteOnHold() === onHold) return; - this.remoteOnHold = onHold; - for (const transceiver of this.peerConn.getTransceivers()) { - // We don't send hold music or anything so we're not actually - // sending anything, but sendrecv is fairly standard for hold and - // it makes it a lot easier to figure out who's put who on hold. - transceiver.direction = onHold ? "sendonly" : "sendrecv"; - } - this.updateMuteStatus(); - this.sendMetadataUpdate(); - this.emit(CallEvent.RemoteHoldUnhold, this.remoteOnHold, this); - } - - /** - * Indicates whether we are 'on hold' to the remote party (ie. if true, - * they cannot hear us). - * @returns true if the other party has put us on hold - */ - isLocalOnHold() { - if (this.state !== CallState.Connected) return false; - let callOnHold = true; - - // We consider a call to be on hold only if *all* the tracks are on hold - // (is this the right thing to do?) - for (const transceiver of this.peerConn.getTransceivers()) { - const trackOnHold = ["inactive", "recvonly"].includes(transceiver.currentDirection); - if (!trackOnHold) callOnHold = false; - } - return callOnHold; - } - - /** - * Sends a DTMF digit to the other party - * @param digit - The digit (nb. string - '#' and '*' are dtmf too) - */ - sendDtmfDigit(digit) { - for (const sender of this.peerConn.getSenders()) { - var _sender$track; - if (((_sender$track = sender.track) === null || _sender$track === void 0 ? void 0 : _sender$track.kind) === "audio" && sender.dtmf) { - sender.dtmf.insertDTMF(digit); - return; - } - } - throw new Error("Unable to find a track to send DTMF on"); - } - updateMuteStatus() { - const micShouldBeMuted = this.isMicrophoneMuted() || this.remoteOnHold; - const vidShouldBeMuted = this.isLocalVideoMuted() || this.remoteOnHold; - _logger.logger.log(`Call ${this.callId} updateMuteStatus stream ${this.localUsermediaStream.id} micShouldBeMuted ${micShouldBeMuted} vidShouldBeMuted ${vidShouldBeMuted}`); - setTracksEnabled(this.localUsermediaStream.getAudioTracks(), !micShouldBeMuted); - setTracksEnabled(this.localUsermediaStream.getVideoTracks(), !vidShouldBeMuted); - } - async sendMetadataUpdate() { - await this.sendVoipEvent(_event.EventType.CallSDPStreamMetadataChangedPrefix, { - [_callEventTypes.SDPStreamMetadataKey]: this.getLocalSDPStreamMetadata() - }); - } - gotCallFeedsForInvite(callFeeds, requestScreenshareFeed = false) { - if (this.successor) { - this.successor.queueGotCallFeedsForAnswer(callFeeds); - return; - } - if (this.callHasEnded()) { - this.stopAllMedia(); - return; - } - for (const feed of callFeeds) { - this.pushLocalFeed(feed); - } - if (requestScreenshareFeed) { - this.peerConn.addTransceiver("video", { - direction: "recvonly" - }); - } - this.state = CallState.CreateOffer; - _logger.logger.debug(`Call ${this.callId} gotUserMediaForInvite() run`); - // Now we wait for the negotiationneeded event - } - - async sendAnswer() { - const answerContent = { - answer: { - sdp: this.peerConn.localDescription.sdp, - // type is now deprecated as of Matrix VoIP v1, but - // required to still be sent for backwards compat - type: this.peerConn.localDescription.type - }, - [_callEventTypes.SDPStreamMetadataKey]: this.getLocalSDPStreamMetadata(true) - }; - answerContent.capabilities = { - "m.call.transferee": this.client.supportsCallTransfer, - "m.call.dtmf": false - }; - - // We have just taken the local description from the peerConn which will - // contain all the local candidates added so far, so we can discard any candidates - // we had queued up because they'll be in the answer. - const discardCount = this.discardDuplicateCandidates(); - _logger.logger.info(`Call ${this.callId} sendAnswer() discarding ${discardCount} candidates that will be sent in answer`); - try { - await this.sendVoipEvent(_event.EventType.CallAnswer, answerContent); - // If this isn't the first time we've tried to send the answer, - // we may have candidates queued up, so send them now. - this.inviteOrAnswerSent = true; - } catch (error) { - // We've failed to answer: back to the ringing state - this.state = CallState.Ringing; - if (error instanceof _httpApi.MatrixError && error.event) this.client.cancelPendingEvent(error.event); - let code = CallErrorCode.SendAnswer; - let message = "Failed to send answer"; - if (error.name == "UnknownDeviceError") { - code = CallErrorCode.UnknownDevices; - message = "Unknown devices present in the room"; - } - this.emit(CallEvent.Error, new CallError(code, message, error), this); - throw error; - } - - // error handler re-throws so this won't happen on error, but - // we don't want the same error handling on the candidate queue - this.sendCandidateQueue(); - } - queueGotCallFeedsForAnswer(callFeeds) { - // Ensure only one negotiate/answer event is being processed at a time. - if (this.responsePromiseChain) { - this.responsePromiseChain = this.responsePromiseChain.then(() => this.gotCallFeedsForAnswer(callFeeds)); - } else { - this.responsePromiseChain = this.gotCallFeedsForAnswer(callFeeds); - } - } - - // Enables DTX (discontinuous transmission) on the given session to reduce - // bandwidth when transmitting silence - mungeSdp(description, mods) { - // The only way to enable DTX at this time is through SDP munging - const sdp = (0, _sdpTransform.parse)(description.sdp); - sdp.media.forEach(media => { - const payloadTypeToCodecMap = new Map(); - const codecToPayloadTypeMap = new Map(); - for (const rtp of media.rtp) { - payloadTypeToCodecMap.set(rtp.payload, rtp.codec); - codecToPayloadTypeMap.set(rtp.codec, rtp.payload); - } - for (const mod of mods) { - if (mod.mediaType !== media.type) continue; - if (!codecToPayloadTypeMap.has(mod.codec)) { - _logger.logger.info(`Call ${this.callId} mungeSdp() ignoring SDP modifications for ${mod.codec} as it's not present.`); - continue; - } - const extraConfig = []; - if (mod.enableDtx !== undefined) { - extraConfig.push(`usedtx=${mod.enableDtx ? "1" : "0"}`); - } - if (mod.maxAverageBitrate !== undefined) { - extraConfig.push(`maxaveragebitrate=${mod.maxAverageBitrate}`); - } - let found = false; - for (const fmtp of media.fmtp) { - if (payloadTypeToCodecMap.get(fmtp.payload) === mod.codec) { - found = true; - fmtp.config += ";" + extraConfig.join(";"); - } - } - if (!found) { - media.fmtp.push({ - payload: codecToPayloadTypeMap.get(mod.codec), - config: extraConfig.join(";") - }); - } - } - }); - description.sdp = (0, _sdpTransform.write)(sdp); - } - async createOffer() { - const offer = await this.peerConn.createOffer(); - this.mungeSdp(offer, getCodecParamMods(this.isPtt)); - return offer; - } - async createAnswer() { - const answer = await this.peerConn.createAnswer(); - this.mungeSdp(answer, getCodecParamMods(this.isPtt)); - return answer; - } - async gotCallFeedsForAnswer(callFeeds) { - if (this.callHasEnded()) return; - this.waitForLocalAVStream = false; - for (const feed of callFeeds) { - this.pushLocalFeed(feed); - } - this.state = CallState.CreateAnswer; - let answer; - try { - this.getRidOfRTXCodecs(); - answer = await this.createAnswer(); - } catch (err) { - _logger.logger.debug(`Call ${this.callId} gotCallFeedsForAnswer() failed to create answer: `, err); - this.terminate(CallParty.Local, CallErrorCode.CreateAnswer, true); - return; - } - try { - await this.peerConn.setLocalDescription(answer); - - // make sure we're still going - if (this.callHasEnded()) return; - this.state = CallState.Connecting; - - // Allow a short time for initial candidates to be gathered - await new Promise(resolve => { - setTimeout(resolve, 200); - }); - - // make sure the call hasn't ended before we continue - if (this.callHasEnded()) return; - this.sendAnswer(); - } catch (err) { - _logger.logger.debug(`Call ${this.callId} gotCallFeedsForAnswer() error setting local description!`, err); - this.terminate(CallParty.Local, CallErrorCode.SetLocalDescription, true); - return; - } - } - - /** - * Internal - */ - - async onRemoteIceCandidatesReceived(ev) { - if (this.callHasEnded()) { - //debuglog("Ignoring remote ICE candidate because call has ended"); - return; - } - const content = ev.getContent(); - const candidates = content.candidates; - if (!candidates) { - _logger.logger.info(`Call ${this.callId} onRemoteIceCandidatesReceived() ignoring candidates event with no candidates!`); - return; - } - const fromPartyId = content.version === 0 ? null : content.party_id || null; - if (this.opponentPartyId === undefined) { - // we haven't picked an opponent yet so save the candidates - if (fromPartyId) { - _logger.logger.info(`Call ${this.callId} onRemoteIceCandidatesReceived() buffering ${candidates.length} candidates until we pick an opponent`); - const bufferedCandidates = this.remoteCandidateBuffer.get(fromPartyId) || []; - bufferedCandidates.push(...candidates); - this.remoteCandidateBuffer.set(fromPartyId, bufferedCandidates); - } - return; - } - if (!this.partyIdMatches(content)) { - _logger.logger.info(`Call ${this.callId} onRemoteIceCandidatesReceived() ignoring candidates from party ID ${content.party_id}: we have chosen party ID ${this.opponentPartyId}`); - return; - } - await this.addIceCandidates(candidates); - } - - /** - * Used by MatrixClient. - */ - async onAnswerReceived(event) { - const content = event.getContent(); - _logger.logger.debug(`Call ${this.callId} onAnswerReceived() running (hangupParty=${content.party_id})`); - if (this.callHasEnded()) { - _logger.logger.debug(`Call ${this.callId} onAnswerReceived() ignoring answer because call has ended`); - return; - } - if (this.opponentPartyId !== undefined) { - _logger.logger.info(`Call ${this.callId} onAnswerReceived() ignoring answer from party ID ${content.party_id}: we already have an answer/reject from ${this.opponentPartyId}`); - return; - } - this.chooseOpponent(event); - await this.addBufferedIceCandidates(); - this.state = CallState.Connecting; - const sdpStreamMetadata = content[_callEventTypes.SDPStreamMetadataKey]; - if (sdpStreamMetadata) { - this.updateRemoteSDPStreamMetadata(sdpStreamMetadata); - } else { - _logger.logger.warn(`Call ${this.callId} onAnswerReceived() did not get any SDPStreamMetadata! Can not send/receive multiple streams`); - } - try { - await this.peerConn.setRemoteDescription(content.answer); - } catch (e) { - _logger.logger.debug(`Call ${this.callId} onAnswerReceived() failed to set remote description`, e); - this.terminate(CallParty.Local, CallErrorCode.SetRemoteDescription, false); - return; - } - - // If the answer we selected has a party_id, send a select_answer event - // We do this after setting the remote description since otherwise we'd block - // call setup on it - if (this.opponentPartyId !== null) { - try { - await this.sendVoipEvent(_event.EventType.CallSelectAnswer, { - selected_party_id: this.opponentPartyId - }); - } catch (err) { - // This isn't fatal, and will just mean that if another party has raced to answer - // the call, they won't know they got rejected, so we carry on & don't retry. - _logger.logger.warn(`Call ${this.callId} onAnswerReceived() failed to send select_answer event`, err); - } - } - } - async onSelectAnswerReceived(event) { - if (this.direction !== CallDirection.Inbound) { - _logger.logger.warn(`Call ${this.callId} onSelectAnswerReceived() got select_answer for an outbound call: ignoring`); - return; - } - const selectedPartyId = event.getContent().selected_party_id; - if (selectedPartyId === undefined || selectedPartyId === null) { - _logger.logger.warn(`Call ${this.callId} onSelectAnswerReceived() got nonsensical select_answer with null/undefined selected_party_id: ignoring`); - return; - } - if (selectedPartyId !== this.ourPartyId) { - _logger.logger.info(`Call ${this.callId} onSelectAnswerReceived() got select_answer for party ID ${selectedPartyId}: we are party ID ${this.ourPartyId}.`); - // The other party has picked somebody else's answer - await this.terminate(CallParty.Remote, CallErrorCode.AnsweredElsewhere, true); - } - } - async onNegotiateReceived(event) { - const content = event.getContent(); - const description = content.description; - if (!description || !description.sdp || !description.type) { - _logger.logger.info(`Call ${this.callId} onNegotiateReceived() ignoring invalid m.call.negotiate event`); - return; - } - // Politeness always follows the direction of the call: in a glare situation, - // we pick either the inbound or outbound call, so one side will always be - // inbound and one outbound - const polite = this.direction === CallDirection.Inbound; - - // Here we follow the perfect negotiation logic from - // https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/Perfect_negotiation - const offerCollision = description.type === "offer" && (this.makingOffer || this.peerConn.signalingState !== "stable"); - this.ignoreOffer = !polite && offerCollision; - if (this.ignoreOffer) { - _logger.logger.info(`Call ${this.callId} onNegotiateReceived() ignoring colliding negotiate event because we're impolite`); - return; - } - const prevLocalOnHold = this.isLocalOnHold(); - const sdpStreamMetadata = content[_callEventTypes.SDPStreamMetadataKey]; - if (sdpStreamMetadata) { - this.updateRemoteSDPStreamMetadata(sdpStreamMetadata); - } else { - _logger.logger.warn(`Call ${this.callId} onNegotiateReceived() received negotiation event without SDPStreamMetadata!`); - } - try { - await this.peerConn.setRemoteDescription(description); - if (description.type === "offer") { - var _localDescription; - let answer; - try { - this.getRidOfRTXCodecs(); - answer = await this.createAnswer(); - } catch (err) { - _logger.logger.debug(`Call ${this.callId} onNegotiateReceived() failed to create answer: `, err); - this.terminate(CallParty.Local, CallErrorCode.CreateAnswer, true); - return; - } - await this.peerConn.setLocalDescription(answer); - this.sendVoipEvent(_event.EventType.CallNegotiate, { - description: (_localDescription = this.peerConn.localDescription) === null || _localDescription === void 0 ? void 0 : _localDescription.toJSON(), - [_callEventTypes.SDPStreamMetadataKey]: this.getLocalSDPStreamMetadata(true) - }); - } - } catch (err) { - _logger.logger.warn(`Call ${this.callId} onNegotiateReceived() failed to complete negotiation`, err); - } - const newLocalOnHold = this.isLocalOnHold(); - if (prevLocalOnHold !== newLocalOnHold) { - this.emit(CallEvent.LocalHoldUnhold, newLocalOnHold, this); - // also this one for backwards compat - this.emit(CallEvent.HoldUnhold, newLocalOnHold); - } - } - updateRemoteSDPStreamMetadata(metadata) { - this.remoteSDPStreamMetadata = utils.recursivelyAssign(this.remoteSDPStreamMetadata || {}, metadata, true); - for (const feed of this.getRemoteFeeds()) { - var _streamId; - const streamId = feed.stream.id; - const metadata = this.remoteSDPStreamMetadata[streamId]; - feed.setAudioVideoMuted(metadata === null || metadata === void 0 ? void 0 : metadata.audio_muted, metadata === null || metadata === void 0 ? void 0 : metadata.video_muted); - feed.purpose = (_streamId = this.remoteSDPStreamMetadata[streamId]) === null || _streamId === void 0 ? void 0 : _streamId.purpose; - } - } - onSDPStreamMetadataChangedReceived(event) { - const content = event.getContent(); - const metadata = content[_callEventTypes.SDPStreamMetadataKey]; - this.updateRemoteSDPStreamMetadata(metadata); - } - async onAssertedIdentityReceived(event) { - const content = event.getContent(); - if (!content.asserted_identity) return; - this.remoteAssertedIdentity = { - id: content.asserted_identity.id, - displayName: content.asserted_identity.display_name - }; - this.emit(CallEvent.AssertedIdentityChanged, this); - } - callHasEnded() { - // This exists as workaround to typescript trying to be clever and erroring - // when putting if (this.state === CallState.Ended) return; twice in the same - // function, even though that function is async. - return this.state === CallState.Ended; - } - queueGotLocalOffer() { - // Ensure only one negotiate/answer event is being processed at a time. - if (this.responsePromiseChain) { - this.responsePromiseChain = this.responsePromiseChain.then(() => this.wrappedGotLocalOffer()); - } else { - this.responsePromiseChain = this.wrappedGotLocalOffer(); - } - } - async wrappedGotLocalOffer() { - this.makingOffer = true; - try { - // XXX: in what situations do we believe gotLocalOffer actually throws? It appears - // to handle most of its exceptions itself and terminate the call. I'm not entirely - // sure it would ever throw, so I can't add a test for these lines. - // Also the tense is different between "gotLocalOffer" and "getLocalOfferFailed" so - // it's not entirely clear whether getLocalOfferFailed is just misnamed or whether - // they've been cross-polinated somehow at some point. - await this.gotLocalOffer(); - } catch (e) { - this.getLocalOfferFailed(e); - return; - } finally { - this.makingOffer = false; - } - } - async gotLocalOffer() { - _logger.logger.debug(`Call ${this.callId} gotLocalOffer() running`); - if (this.callHasEnded()) { - _logger.logger.debug(`Call ${this.callId} gotLocalOffer() ignoring newly created offer because the call has ended"`); - return; - } - let offer; - try { - this.getRidOfRTXCodecs(); - offer = await this.createOffer(); - } catch (err) { - _logger.logger.debug(`Call ${this.callId} gotLocalOffer() failed to create offer: `, err); - this.terminate(CallParty.Local, CallErrorCode.CreateOffer, true); - return; - } - try { - await this.peerConn.setLocalDescription(offer); - } catch (err) { - _logger.logger.debug(`Call ${this.callId} gotLocalOffer() error setting local description!`, err); - this.terminate(CallParty.Local, CallErrorCode.SetLocalDescription, true); - return; - } - if (this.peerConn.iceGatheringState === "gathering") { - // Allow a short time for initial candidates to be gathered - await new Promise(resolve => { - setTimeout(resolve, 200); - }); - } - if (this.callHasEnded()) return; - const eventType = this.state === CallState.CreateOffer ? _event.EventType.CallInvite : _event.EventType.CallNegotiate; - const content = { - lifetime: CALL_TIMEOUT_MS - }; - if (eventType === _event.EventType.CallInvite && this.invitee) { - content.invitee = this.invitee; - } - - // clunky because TypeScript can't follow the types through if we use an expression as the key - if (this.state === CallState.CreateOffer) { - var _localDescription2; - content.offer = (_localDescription2 = this.peerConn.localDescription) === null || _localDescription2 === void 0 ? void 0 : _localDescription2.toJSON(); - } else { - var _localDescription3; - content.description = (_localDescription3 = this.peerConn.localDescription) === null || _localDescription3 === void 0 ? void 0 : _localDescription3.toJSON(); - } - content.capabilities = { - "m.call.transferee": this.client.supportsCallTransfer, - "m.call.dtmf": false - }; - content[_callEventTypes.SDPStreamMetadataKey] = this.getLocalSDPStreamMetadata(true); - - // Get rid of any candidates waiting to be sent: they'll be included in the local - // description we just got and will send in the offer. - const discardCount = this.discardDuplicateCandidates(); - _logger.logger.info(`Call ${this.callId} gotLocalOffer() discarding ${discardCount} candidates that will be sent in offer`); - try { - await this.sendVoipEvent(eventType, content); - } catch (error) { - _logger.logger.error(`Call ${this.callId} gotLocalOffer() failed to send invite`, error); - if (error instanceof _httpApi.MatrixError && error.event) this.client.cancelPendingEvent(error.event); - let code = CallErrorCode.SignallingFailed; - let message = "Signalling failed"; - if (this.state === CallState.CreateOffer) { - code = CallErrorCode.SendInvite; - message = "Failed to send invite"; - } - if (error.name == "UnknownDeviceError") { - code = CallErrorCode.UnknownDevices; - message = "Unknown devices present in the room"; - } - this.emit(CallEvent.Error, new CallError(code, message, error), this); - this.terminate(CallParty.Local, code, false); - - // no need to carry on & send the candidate queue, but we also - // don't want to rethrow the error - return; - } - this.sendCandidateQueue(); - if (this.state === CallState.CreateOffer) { - this.inviteOrAnswerSent = true; - this.state = CallState.InviteSent; - this.inviteTimeout = setTimeout(() => { - this.inviteTimeout = undefined; - if (this.state === CallState.InviteSent) { - this.hangup(CallErrorCode.InviteTimeout, false); - } - }, CALL_TIMEOUT_MS); - } - } - /** - * This method removes all video/rtx codecs from screensharing video - * transceivers. This is necessary since they can cause problems. Without - * this the following steps should produce an error: - * Chromium calls Firefox - * Firefox answers - * Firefox starts screen-sharing - * Chromium starts screen-sharing - * Call crashes for Chromium with: - * [96685:23:0518/162603.933321:ERROR:webrtc_video_engine.cc(3296)] RTX codec (PT=97) mapped to PT=96 which is not in the codec list. - * [96685:23:0518/162603.933377:ERROR:webrtc_video_engine.cc(1171)] GetChangedRecvParameters called without any video codecs. - * [96685:23:0518/162603.933430:ERROR:sdp_offer_answer.cc(4302)] Failed to set local video description recv parameters for m-section with mid='2'. (INVALID_PARAMETER) - */ - getRidOfRTXCodecs() { - // RTCRtpReceiver.getCapabilities and RTCRtpSender.getCapabilities don't seem to be supported on FF - if (!RTCRtpReceiver.getCapabilities || !RTCRtpSender.getCapabilities) return; - const recvCodecs = RTCRtpReceiver.getCapabilities("video").codecs; - const sendCodecs = RTCRtpSender.getCapabilities("video").codecs; - const codecs = [...sendCodecs, ...recvCodecs]; - for (const codec of codecs) { - if (codec.mimeType === "video/rtx") { - const rtxCodecIndex = codecs.indexOf(codec); - codecs.splice(rtxCodecIndex, 1); - } - } - const screenshareVideoTransceiver = this.transceivers.get(getTransceiverKey(_callEventTypes.SDPStreamMetadataPurpose.Screenshare, "video")); - if (screenshareVideoTransceiver) screenshareVideoTransceiver.setCodecPreferences(codecs); - } - /** - * @internal - */ - async sendVoipEvent(eventType, content) { - const realContent = Object.assign({}, content, { - version: VOIP_PROTO_VERSION, - call_id: this.callId, - party_id: this.ourPartyId, - conf_id: this.groupCallId - }); - if (this.opponentDeviceId) { - var _this$getOpponentMemb2; - const toDeviceSeq = this.toDeviceSeq++; - const content = _objectSpread(_objectSpread({}, realContent), {}, { - device_id: this.client.deviceId, - sender_session_id: this.client.getSessionId(), - dest_session_id: this.opponentSessionId, - seq: toDeviceSeq, - [_event.ToDeviceMessageId]: (0, _uuid.v4)() - }); - this.emit(CallEvent.SendVoipEvent, { - type: "toDevice", - eventType, - userId: this.invitee || ((_this$getOpponentMemb2 = this.getOpponentMember()) === null || _this$getOpponentMemb2 === void 0 ? void 0 : _this$getOpponentMemb2.userId), - opponentDeviceId: this.opponentDeviceId, - content - }, this); - const userId = this.invitee || this.getOpponentMember().userId; - if (this.client.getUseE2eForGroupCall()) { - if (!this.opponentDeviceInfo) { - _logger.logger.warn(`Call ${this.callId} sendVoipEvent() failed: we do not have opponentDeviceInfo`); - return; - } - await this.client.encryptAndSendToDevices([{ - userId, - deviceInfo: this.opponentDeviceInfo - }], { - type: eventType, - content - }); - } else { - await this.client.sendToDevice(eventType, new Map([[userId, new Map([[this.opponentDeviceId, content]])]])); - } - } else { - var _this$getOpponentMemb3; - this.emit(CallEvent.SendVoipEvent, { - type: "sendEvent", - eventType, - roomId: this.roomId, - content: realContent, - userId: this.invitee || ((_this$getOpponentMemb3 = this.getOpponentMember()) === null || _this$getOpponentMemb3 === void 0 ? void 0 : _this$getOpponentMemb3.userId) - }, this); - await this.client.sendEvent(this.roomId, eventType, realContent); - } - } - - /** - * Queue a candidate to be sent - * @param content - The candidate to queue up, or null if candidates have finished being generated - * and end-of-candidates should be signalled - */ - queueCandidate(content) { - // We partially de-trickle candidates by waiting for `delay` before sending them - // amalgamated, in order to avoid sending too many m.call.candidates events and hitting - // rate limits in Matrix. - // In practice, it'd be better to remove rate limits for m.call.* - - // N.B. this deliberately lets you queue and send blank candidates, which MSC2746 - // currently proposes as the way to indicate that candidate gathering is complete. - // This will hopefully be changed to an explicit rather than implicit notification - // shortly. - if (content) { - this.candidateSendQueue.push(content); - } else { - this.candidatesEnded = true; - } - - // Don't send the ICE candidates yet if the call is in the ringing state: this - // means we tried to pick (ie. started generating candidates) and then failed to - // send the answer and went back to the ringing state. Queue up the candidates - // to send if we successfully send the answer. - // Equally don't send if we haven't yet sent the answer because we can send the - // first batch of candidates along with the answer - if (this.state === CallState.Ringing || !this.inviteOrAnswerSent) return; - - // MSC2746 recommends these values (can be quite long when calling because the - // callee will need a while to answer the call) - const delay = this.direction === CallDirection.Inbound ? 500 : 2000; - if (this.candidateSendTries === 0) { - setTimeout(() => { - this.sendCandidateQueue(); - }, delay); - } - } - - // Discard all non-end-of-candidates messages - // Return the number of candidate messages that were discarded. - // Call this method before sending an invite or answer message - discardDuplicateCandidates() { - let discardCount = 0; - const newQueue = []; - for (let i = 0; i < this.candidateSendQueue.length; i++) { - const candidate = this.candidateSendQueue[i]; - if (candidate.candidate === "") { - newQueue.push(candidate); - } else { - discardCount++; - } - } - this.candidateSendQueue = newQueue; - return discardCount; - } - - /* - * Transfers this call to another user - */ - async transfer(targetUserId) { - // Fetch the target user's global profile info: their room avatar / displayname - // could be different in whatever room we share with them. - const profileInfo = await this.client.getProfileInfo(targetUserId); - const replacementId = genCallID(); - const body = { - replacement_id: genCallID(), - target_user: { - id: targetUserId, - display_name: profileInfo.displayname, - avatar_url: profileInfo.avatar_url - }, - create_call: replacementId - }; - await this.sendVoipEvent(_event.EventType.CallReplaces, body); - await this.terminate(CallParty.Local, CallErrorCode.Transferred, true); - } - - /* - * Transfers this call to the target call, effectively 'joining' the - * two calls (so the remote parties on each call are connected together). - */ - async transferToCall(transferTargetCall) { - var _transferTargetCall$g, _this$getOpponentMemb4; - const targetUserId = (_transferTargetCall$g = transferTargetCall.getOpponentMember()) === null || _transferTargetCall$g === void 0 ? void 0 : _transferTargetCall$g.userId; - const targetProfileInfo = targetUserId ? await this.client.getProfileInfo(targetUserId) : undefined; - const opponentUserId = (_this$getOpponentMemb4 = this.getOpponentMember()) === null || _this$getOpponentMemb4 === void 0 ? void 0 : _this$getOpponentMemb4.userId; - const transfereeProfileInfo = opponentUserId ? await this.client.getProfileInfo(opponentUserId) : undefined; - const newCallId = genCallID(); - const bodyToTransferTarget = { - // the replacements on each side have their own ID, and it's distinct from the - // ID of the new call (but we can use the same function to generate it) - replacement_id: genCallID(), - target_user: { - id: opponentUserId, - display_name: transfereeProfileInfo === null || transfereeProfileInfo === void 0 ? void 0 : transfereeProfileInfo.displayname, - avatar_url: transfereeProfileInfo === null || transfereeProfileInfo === void 0 ? void 0 : transfereeProfileInfo.avatar_url - }, - await_call: newCallId - }; - await transferTargetCall.sendVoipEvent(_event.EventType.CallReplaces, bodyToTransferTarget); - const bodyToTransferee = { - replacement_id: genCallID(), - target_user: { - id: targetUserId, - display_name: targetProfileInfo === null || targetProfileInfo === void 0 ? void 0 : targetProfileInfo.displayname, - avatar_url: targetProfileInfo === null || targetProfileInfo === void 0 ? void 0 : targetProfileInfo.avatar_url - }, - create_call: newCallId - }; - await this.sendVoipEvent(_event.EventType.CallReplaces, bodyToTransferee); - await this.terminate(CallParty.Local, CallErrorCode.Transferred, true); - await transferTargetCall.terminate(CallParty.Local, CallErrorCode.Transferred, true); - } - async terminate(hangupParty, hangupReason, shouldEmit) { - var _this$stats2; - if (this.callHasEnded()) return; - this.hangupParty = hangupParty; - this.hangupReason = hangupReason; - this.state = CallState.Ended; - if (this.inviteTimeout) { - clearTimeout(this.inviteTimeout); - this.inviteTimeout = undefined; - } - if (this.iceDisconnectedTimeout !== undefined) { - clearTimeout(this.iceDisconnectedTimeout); - this.iceDisconnectedTimeout = undefined; - } - if (this.callLengthInterval) { - clearInterval(this.callLengthInterval); - this.callLengthInterval = undefined; - } - if (this.stopVideoTrackTimer !== undefined) { - clearTimeout(this.stopVideoTrackTimer); - this.stopVideoTrackTimer = undefined; - } - for (const [stream, listener] of this.removeTrackListeners) { - stream.removeEventListener("removetrack", listener); - } - this.removeTrackListeners.clear(); - this.callStatsAtEnd = await this.collectCallStats(); - - // Order is important here: first we stopAllMedia() and only then we can deleteAllFeeds() - this.stopAllMedia(); - this.deleteAllFeeds(); - if (this.peerConn && this.peerConn.signalingState !== "closed") { - this.peerConn.close(); - } - (_this$stats2 = this.stats) === null || _this$stats2 === void 0 ? void 0 : _this$stats2.removeStatsReportGatherer(this.callId); - if (shouldEmit) { - this.emit(CallEvent.Hangup, this); - } - this.client.callEventHandler.calls.delete(this.callId); - } - stopAllMedia() { - _logger.logger.debug(`Call ${this.callId} stopAllMedia() running`); - for (const feed of this.feeds) { - // Slightly awkward as local feed need to go via the correct method on - // the MediaHandler so they get removed from MediaHandler (remote tracks - // don't) - // NB. We clone local streams when passing them to individual calls in a group - // call, so we can (and should) stop the clones once we no longer need them: - // the other clones will continue fine. - if (feed.isLocal() && feed.purpose === _callEventTypes.SDPStreamMetadataPurpose.Usermedia) { - this.client.getMediaHandler().stopUserMediaStream(feed.stream); - } else if (feed.isLocal() && feed.purpose === _callEventTypes.SDPStreamMetadataPurpose.Screenshare) { - this.client.getMediaHandler().stopScreensharingStream(feed.stream); - } else if (!feed.isLocal()) { - _logger.logger.debug(`Call ${this.callId} stopAllMedia() stopping stream (streamId=${feed.stream.id})`); - for (const track of feed.stream.getTracks()) { - track.stop(); - } - } - } - } - checkForErrorListener() { - if (this.listeners(_typedEventEmitter.EventEmitterEvents.Error).length === 0) { - throw new Error("You MUST attach an error listener using call.on('error', function() {})"); - } - } - async sendCandidateQueue() { - if (this.candidateSendQueue.length === 0 || this.callHasEnded()) { - return; - } - const candidates = this.candidateSendQueue; - this.candidateSendQueue = []; - ++this.candidateSendTries; - const content = { - candidates: candidates.map(candidate => candidate.toJSON()) - }; - if (this.candidatesEnded) { - // If there are no more candidates, signal this by adding an empty string candidate - content.candidates.push({ - candidate: "" - }); - } - _logger.logger.debug(`Call ${this.callId} sendCandidateQueue() attempting to send ${candidates.length} candidates`); - try { - await this.sendVoipEvent(_event.EventType.CallCandidates, content); - // reset our retry count if we have successfully sent our candidates - // otherwise queueCandidate() will refuse to try to flush the queue - this.candidateSendTries = 0; - - // Try to send candidates again just in case we received more candidates while sending. - this.sendCandidateQueue(); - } catch (error) { - // don't retry this event: we'll send another one later as we might - // have more candidates by then. - if (error instanceof _httpApi.MatrixError && error.event) this.client.cancelPendingEvent(error.event); - - // put all the candidates we failed to send back in the queue - this.candidateSendQueue.push(...candidates); - if (this.candidateSendTries > 5) { - _logger.logger.debug(`Call ${this.callId} sendCandidateQueue() failed to send candidates on attempt ${this.candidateSendTries}. Giving up on this call.`, error); - const code = CallErrorCode.SignallingFailed; - const message = "Signalling failed"; - this.emit(CallEvent.Error, new CallError(code, message, error), this); - this.hangup(code, false); - return; - } - const delayMs = 500 * Math.pow(2, this.candidateSendTries); - ++this.candidateSendTries; - _logger.logger.debug(`Call ${this.callId} sendCandidateQueue() failed to send candidates. Retrying in ${delayMs}ms`, error); - setTimeout(() => { - this.sendCandidateQueue(); - }, delayMs); - } - } - - /** - * Place a call to this room. - * @throws if you have not specified a listener for 'error' events. - * @throws if have passed audio=false. - */ - async placeCall(audio, video) { - if (!audio) { - throw new Error("You CANNOT start a call without audio"); - } - this.state = CallState.WaitLocalMedia; - try { - var _this$client$getDevic2; - const stream = await this.client.getMediaHandler().getUserMediaStream(audio, video); - - // make sure all the tracks are enabled (same as pushNewLocalFeed - - // we probably ought to just have one code path for adding streams) - setTracksEnabled(stream.getAudioTracks(), true); - setTracksEnabled(stream.getVideoTracks(), true); - const callFeed = new _callFeed.CallFeed({ - client: this.client, - roomId: this.roomId, - userId: this.client.getUserId(), - deviceId: (_this$client$getDevic2 = this.client.getDeviceId()) !== null && _this$client$getDevic2 !== void 0 ? _this$client$getDevic2 : undefined, - stream, - purpose: _callEventTypes.SDPStreamMetadataPurpose.Usermedia, - audioMuted: false, - videoMuted: false - }); - await this.placeCallWithCallFeeds([callFeed]); - } catch (e) { - this.getUserMediaFailed(e); - return; - } - } - - /** - * Place a call to this room with call feed. - * @param callFeeds - to use - * @throws if you have not specified a listener for 'error' events. - * @throws if have passed audio=false. - */ - async placeCallWithCallFeeds(callFeeds, requestScreenshareFeed = false) { - this.checkForErrorListener(); - this.direction = CallDirection.Outbound; - await this.initOpponentCrypto(); - - // XXX Find a better way to do this - this.client.callEventHandler.calls.set(this.callId, this); - - // make sure we have valid turn creds. Unless something's gone wrong, it should - // poll and keep the credentials valid so this should be instant. - const haveTurnCreds = await this.client.checkTurnServers(); - if (!haveTurnCreds) { - _logger.logger.warn(`Call ${this.callId} placeCallWithCallFeeds() failed to get TURN credentials! Proceeding with call anyway...`); - } - - // create the peer connection now so it can be gathering candidates while we get user - // media (assuming a candidate pool size is configured) - this.peerConn = this.createPeerConnection(); - this.gotCallFeedsForInvite(callFeeds, requestScreenshareFeed); - } - createPeerConnection() { - var _this$stats3; - const pc = new window.RTCPeerConnection({ - iceTransportPolicy: this.forceTURN ? "relay" : undefined, - iceServers: this.turnServers, - iceCandidatePoolSize: this.client.iceCandidatePoolSize, - bundlePolicy: "max-bundle" - }); - - // 'connectionstatechange' would be better, but firefox doesn't implement that. - pc.addEventListener("iceconnectionstatechange", this.onIceConnectionStateChanged); - pc.addEventListener("signalingstatechange", this.onSignallingStateChanged); - pc.addEventListener("icecandidate", this.gotLocalIceCandidate); - pc.addEventListener("icegatheringstatechange", this.onIceGatheringStateChange); - pc.addEventListener("track", this.onTrack); - pc.addEventListener("negotiationneeded", this.onNegotiationNeeded); - pc.addEventListener("datachannel", this.onDataChannel); - (_this$stats3 = this.stats) === null || _this$stats3 === void 0 ? void 0 : _this$stats3.addStatsReportGatherer(this.callId, "unknown", pc); - return pc; - } - partyIdMatches(msg) { - // They must either match or both be absent (in which case opponentPartyId will be null) - // Also we ignore party IDs on the invite/offer if the version is 0, so we must do the same - // here and use null if the version is 0 (woe betide any opponent sending messages in the - // same call with different versions) - const msgPartyId = msg.version === 0 ? null : msg.party_id || null; - return msgPartyId === this.opponentPartyId; - } - - // Commits to an opponent for the call - // ev: An invite or answer event - chooseOpponent(ev) { - var _getMember; - // I choo-choo-choose you - const msg = ev.getContent(); - _logger.logger.debug(`Call ${this.callId} chooseOpponent() running (partyId=${msg.party_id})`); - this.opponentVersion = msg.version; - if (this.opponentVersion === 0) { - // set to null to indicate that we've chosen an opponent, but because - // they're v0 they have no party ID (even if they sent one, we're ignoring it) - this.opponentPartyId = null; - } else { - // set to their party ID, or if they're naughty and didn't send one despite - // not being v0, set it to null to indicate we picked an opponent with no - // party ID - this.opponentPartyId = msg.party_id || null; - } - this.opponentCaps = msg.capabilities || {}; - this.opponentMember = (_getMember = this.client.getRoom(this.roomId).getMember(ev.getSender())) !== null && _getMember !== void 0 ? _getMember : undefined; - } - async addBufferedIceCandidates() { - const bufferedCandidates = this.remoteCandidateBuffer.get(this.opponentPartyId); - if (bufferedCandidates) { - _logger.logger.info(`Call ${this.callId} addBufferedIceCandidates() adding ${bufferedCandidates.length} buffered candidates for opponent ${this.opponentPartyId}`); - await this.addIceCandidates(bufferedCandidates); - } - this.remoteCandidateBuffer.clear(); - } - async addIceCandidates(candidates) { - for (const candidate of candidates) { - if ((candidate.sdpMid === null || candidate.sdpMid === undefined) && (candidate.sdpMLineIndex === null || candidate.sdpMLineIndex === undefined)) { - _logger.logger.debug(`Call ${this.callId} addIceCandidates() got remote ICE end-of-candidates`); - } else { - _logger.logger.debug(`Call ${this.callId} addIceCandidates() got remote ICE candidate (sdpMid=${candidate.sdpMid}, candidate=${candidate.candidate})`); - } - try { - await this.peerConn.addIceCandidate(candidate); - } catch (err) { - if (!this.ignoreOffer) { - _logger.logger.info(`Call ${this.callId} addIceCandidates() failed to add remote ICE candidate`, err); - } - } - } - } - get hasPeerConnection() { - return Boolean(this.peerConn); - } - initStats(stats, peerId = "unknown") { - this.stats = stats; - this.stats.start(); - } -} -exports.MatrixCall = MatrixCall; -function setTracksEnabled(tracks, enabled) { - for (const track of tracks) { - track.enabled = enabled; - } -} -function supportsMatrixCall() { - // typeof prevents Node from erroring on an undefined reference - if (typeof window === "undefined" || typeof document === "undefined") { - // NB. We don't log here as apps try to create a call object as a test for - // whether calls are supported, so we shouldn't fill the logs up. - return false; - } - - // Firefox throws on so little as accessing the RTCPeerConnection when operating in a secure mode. - // There's some information at https://bugzilla.mozilla.org/show_bug.cgi?id=1542616 though the concern - // is that the browser throwing a SecurityError will brick the client creation process. - try { - const supported = Boolean(window.RTCPeerConnection || window.RTCSessionDescription || window.RTCIceCandidate || navigator.mediaDevices); - if (!supported) { - /* istanbul ignore if */ // Adds a lot of noise to test runs, so disable logging there. - if (process.env.NODE_ENV !== "test") { - _logger.logger.error("WebRTC is not supported in this browser / environment"); - } - return false; - } - } catch (e) { - _logger.logger.error("Exception thrown when trying to access WebRTC", e); - return false; - } - return true; -} - -/** - * DEPRECATED - * Use client.createCall() - * - * Create a new Matrix call for the browser. - * @param client - The client instance to use. - * @param roomId - The room the call is in. - * @param options - DEPRECATED optional options map. - * @returns the call or null if the browser doesn't support calling. - */ -function createNewMatrixCall(client, roomId, options) { - if (!supportsMatrixCall()) return null; - const optionsForceTURN = options ? options.forceTURN : false; - const opts = { - client: client, - roomId: roomId, - invitee: options === null || options === void 0 ? void 0 : options.invitee, - turnServers: client.getTurnServers(), - // call level options - forceTURN: client.forceTURN || optionsForceTURN, - opponentDeviceId: options === null || options === void 0 ? void 0 : options.opponentDeviceId, - opponentSessionId: options === null || options === void 0 ? void 0 : options.opponentSessionId, - groupCallId: options === null || options === void 0 ? void 0 : options.groupCallId - }; - const call = new MatrixCall(opts); - client.reEmitter.reEmit(call, Object.values(CallEvent)); - return call; -} -//# sourceMappingURL=call.js.map
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