@unwanted/matrix-sdk-mini 34.12.0-1 → 34.12.0-2

package/lib/webrtc/call.js

@@ -1,2541 +0,0 @@
-import _asyncToGenerator from "@babel/runtime/helpers/asyncToGenerator";
-import _defineProperty from "@babel/runtime/helpers/defineProperty";
-function ownKeys(e, r) { var t = Object.keys(e); if (Object.getOwnPropertySymbols) { var o = Object.getOwnPropertySymbols(e); r && (o = o.filter(function (r) { return Object.getOwnPropertyDescriptor(e, r).enumerable; })), t.push.apply(t, o); } return t; }
-function _objectSpread(e) { for (var r = 1; r < arguments.length; r++) { var t = null != arguments[r] ? arguments[r] : {}; r % 2 ? ownKeys(Object(t), !0).forEach(function (r) { _defineProperty(e, r, t[r]); }) : Object.getOwnPropertyDescriptors ? Object.defineProperties(e, Object.getOwnPropertyDescriptors(t)) : ownKeys(Object(t)).forEach(function (r) { Object.defineProperty(e, r, Object.getOwnPropertyDescriptor(t, r)); }); } return e; }
-/*
-Copyright 2015, 2016 OpenMarket Ltd
-Copyright 2017 New Vector Ltd
-Copyright 2019, 2020 The Matrix.org Foundation C.I.C.
-Copyright 2021 - 2022 Šimon Brandner <simon.bra.ag@gmail.com>
-
-Licensed under the Apache License, Version 2.0 (the "License");
-you may not use this file except in compliance with the License.
-You may obtain a copy of the License at
-
-    http://www.apache.org/licenses/LICENSE-2.0
-
-Unless required by applicable law or agreed to in writing, software
-distributed under the License is distributed on an "AS IS" BASIS,
-WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-See the License for the specific language governing permissions and
-limitations under the License.
-*/
-
-/**
- * This is an internal module. See {@link createNewMatrixCall} for the public API.
- */
-
-import { v4 as uuidv4 } from "uuid";
-import { parse as parseSdp, write as writeSdp } from "sdp-transform";
-import { logger } from "../logger.js";
-import { checkObjectHasKeys, isNullOrUndefined, recursivelyAssign } from "../utils.js";
-import { EventType, ToDeviceMessageId } from "../@types/event.js";
-import { randomString } from "../randomstring.js";
-import { SDPStreamMetadataPurpose, SDPStreamMetadataKey } from "./callEventTypes.js";
-import { CallFeed } from "./callFeed.js";
-import { EventEmitterEvents, TypedEventEmitter } from "../models/typed-event-emitter.js";
-import { DeviceInfo } from "../crypto/deviceinfo.js";
-import { GroupCallUnknownDeviceError } from "./groupCall.js";
-import { MatrixError } from "../http-api/index.js";
-var MediaType = /*#__PURE__*/function (MediaType) {
-  MediaType["AUDIO"] = "audio";
-  MediaType["VIDEO"] = "video";
-  return MediaType;
-}(MediaType || {});
-var CodecName = /*#__PURE__*/function (CodecName) {
-  CodecName["OPUS"] = "opus";
-  return CodecName;
-}(CodecName || {}); // add more as needed
-// Used internally to specify modifications to codec parameters in SDP
-export var CallState = /*#__PURE__*/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";
-  return CallState;
-}({});
-export var CallType = /*#__PURE__*/function (CallType) {
-  CallType["Voice"] = "voice";
-  CallType["Video"] = "video";
-  return CallType;
-}({});
-export var CallDirection = /*#__PURE__*/function (CallDirection) {
-  CallDirection["Inbound"] = "inbound";
-  CallDirection["Outbound"] = "outbound";
-  return CallDirection;
-}({});
-export var CallParty = /*#__PURE__*/function (CallParty) {
-  CallParty["Local"] = "local";
-  CallParty["Remote"] = "remote";
-  return CallParty;
-}({});
-export var CallEvent = /*#__PURE__*/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["PeerConnectionCreated"] = "peer_connection_created";
-  return CallEvent;
-}({});
-export var CallErrorCode = /*#__PURE__*/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";
-  return CallErrorCode;
-}({});
-
-/**
- * The version field that we set in m.call.* events
- */
-var VOIP_PROTO_VERSION = "1";
-
-/** The fallback ICE server to use for STUN or TURN protocols. */
-export var FALLBACK_ICE_SERVER = "stun:turn.matrix.org";
-
-/** The length of time a call can be ringing for. */
-var CALL_TIMEOUT_MS = 60 * 1000; // ms
-/** The time after which we increment callLength */
-var CALL_LENGTH_INTERVAL = 1000; // ms
-/** The time after which we end the call, if ICE got disconnected */
-var ICE_DISCONNECTED_TIMEOUT = 30 * 1000; // ms
-/** The time after which we try a ICE restart, if ICE got disconnected */
-var ICE_RECONNECTING_TIMEOUT = 2 * 1000; // ms
-export class CallError extends Error {
-  constructor(code, msg, err) {
-    // Still don't think there's any way to have proper nested errors
-    super(msg + ": " + err);
-    _defineProperty(this, "code", void 0);
-    this.code = code;
-  }
-}
-export function genCallID() {
-  return Date.now().toString() + randomString(16);
-}
-function getCodecParamMods(isPtt) {
-  var mods = [{
-    mediaType: "audio",
-    codec: "opus",
-    enableDtx: true,
-    maxAverageBitrate: isPtt ? 12000 : undefined
-  }];
-  return mods;
-}
-
-/**
- * These now all have the call object as an argument. Why? Well, to know which call a given event is
- * about you have three options:
- *  1. Use a closure as the callback that remembers what call it's listening to. This can be
- *     a pain because you need to pass the listener function again when you remove the listener,
- *     which might be somewhere else.
- *  2. Use not-very-well-known fact that EventEmitter sets 'this' to the emitter object in the
- *     callback. This doesn't really play well with modern Typescript and eslint and doesn't work
- *     with our pattern of re-emitting events.
- *  3. Pass the object in question as an argument to the callback.
- *
- * Now that we have group calls which have to deal with multiple call objects, this will
- * become more important, and I think methods 1 and 2 are just going to cause issues.
- */
-
-// The key of the transceiver map (purpose + media type, separated by ':')
-
-// 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;
-}
-export class MatrixCall extends TypedEventEmitter {
-  /**
-   * Construct a new Matrix Call.
-   * @param opts - Config options.
-   */
-  constructor(opts) {
-    var _this, _opts$forceTURN;
-    super();
-    _this = this;
-    _defineProperty(this, "roomId", void 0);
-    _defineProperty(this, "callId", void 0);
-    _defineProperty(this, "invitee", void 0);
-    _defineProperty(this, "hangupParty", void 0);
-    _defineProperty(this, "hangupReason", void 0);
-    _defineProperty(this, "direction", void 0);
-    _defineProperty(this, "ourPartyId", void 0);
-    _defineProperty(this, "peerConn", void 0);
-    _defineProperty(this, "toDeviceSeq", 0);
-    // whether this call should have push-to-talk semantics
-    // This should be set by the consumer on incoming & outgoing calls.
-    _defineProperty(this, "isPtt", false);
-    _defineProperty(this, "_state", CallState.Fledgling);
-    _defineProperty(this, "client", void 0);
-    _defineProperty(this, "forceTURN", void 0);
-    _defineProperty(this, "turnServers", void 0);
-    // A queue for candidates waiting to go out.
-    // We try to amalgamate candidates into a single candidate message where
-    // possible
-    _defineProperty(this, "candidateSendQueue", []);
-    _defineProperty(this, "candidateSendTries", 0);
-    _defineProperty(this, "candidatesEnded", false);
-    _defineProperty(this, "feeds", []);
-    // our transceivers for each purpose and type of media
-    _defineProperty(this, "transceivers", new Map());
-    _defineProperty(this, "inviteOrAnswerSent", false);
-    _defineProperty(this, "waitForLocalAVStream", false);
-    _defineProperty(this, "successor", void 0);
-    _defineProperty(this, "opponentMember", void 0);
-    _defineProperty(this, "opponentVersion", void 0);
-    // 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.
-    _defineProperty(this, "opponentPartyId", void 0);
-    _defineProperty(this, "opponentCaps", void 0);
-    _defineProperty(this, "iceDisconnectedTimeout", void 0);
-    _defineProperty(this, "iceReconnectionTimeOut", void 0);
-    _defineProperty(this, "inviteTimeout", void 0);
-    _defineProperty(this, "removeTrackListeners", new Map());
-    // 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
-    _defineProperty(this, "remoteOnHold", false);
-    // 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' :(
-    _defineProperty(this, "callStatsAtEnd", void 0);
-    // Perfect negotiation state: https://www.w3.org/TR/webrtc/#perfect-negotiation-example
-    _defineProperty(this, "makingOffer", false);
-    _defineProperty(this, "ignoreOffer", false);
-    _defineProperty(this, "isSettingRemoteAnswerPending", false);
-    _defineProperty(this, "responsePromiseChain", void 0);
-    // 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
-    _defineProperty(this, "remoteCandidateBuffer", new Map());
-    _defineProperty(this, "remoteAssertedIdentity", void 0);
-    _defineProperty(this, "remoteSDPStreamMetadata", void 0);
-    _defineProperty(this, "callLengthInterval", void 0);
-    _defineProperty(this, "callStartTime", void 0);
-    _defineProperty(this, "opponentDeviceId", void 0);
-    _defineProperty(this, "opponentDeviceInfo", void 0);
-    _defineProperty(this, "opponentSessionId", void 0);
-    _defineProperty(this, "groupCallId", void 0);
-    // Used to keep the timer for the delay before actually stopping our
-    // video track after muting (see setLocalVideoMuted)
-    _defineProperty(this, "stopVideoTrackTimer", void 0);
-    // 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
-    _defineProperty(this, "isOnlyDataChannelAllowed", void 0);
-    _defineProperty(this, "stats", void 0);
-    /**
-     * Internal
-     */
-    _defineProperty(this, "gotLocalIceCandidate", event => {
-      if (event.candidate) {
-        if (this.candidatesEnded) {
-          logger.warn("Call ".concat(this.callId, " gotLocalIceCandidate() got candidate after candidates have ended!"));
-        }
-        logger.debug("Call ".concat(this.callId, " got local ICE ").concat(event.candidate.sdpMid, " ").concat(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);
-        }
-      }
-    });
-    _defineProperty(this, "onIceGatheringStateChange", event => {
-      var _this$peerConn;
-      logger.debug("Call ".concat(this.callId, " onIceGatheringStateChange() ice gathering state changed to ").concat(this.peerConn.iceGatheringState));
-      if (((_this$peerConn = this.peerConn) === null || _this$peerConn === void 0 ? void 0 : _this$peerConn.iceGatheringState) === "complete") {
-        this.queueCandidate(null); // We should leave it to WebRTC to announce the end
-        logger.debug("Call ".concat(this.callId, " onIceGatheringStateChange() ice gathering state complete, set candidates have ended"));
-      }
-    });
-    _defineProperty(this, "getLocalOfferFailed", err => {
-      logger.error("Call ".concat(this.callId, " getLocalOfferFailed() running"), err);
-      this.emit(CallEvent.Error, new CallError(CallErrorCode.LocalOfferFailed, "Failed to get local offer!", err), this);
-      this.terminate(CallParty.Local, CallErrorCode.LocalOfferFailed, false);
-    });
-    _defineProperty(this, "getUserMediaFailed", err => {
-      if (this.successor) {
-        this.successor.getUserMediaFailed(err);
-        return;
-      }
-      logger.warn("Call ".concat(this.callId, " getUserMediaFailed() failed to get user media - ending call"), err);
-      this.emit(CallEvent.Error, new CallError(CallErrorCode.NoUserMedia, "Couldn't start capturing media! Is your microphone set up and does this app have permission?", err), this);
-      this.terminate(CallParty.Local, CallErrorCode.NoUserMedia, false);
-    });
-    _defineProperty(this, "placeCallFailed", err => {
-      if (this.successor) {
-        this.successor.placeCallFailed(err);
-        return;
-      }
-      logger.warn("Call ".concat(this.callId, " placeCallWithCallFeeds() failed - ending call"), err);
-      this.emit(CallEvent.Error, new CallError(CallErrorCode.IceFailed, "Couldn't start call! Invalid ICE server configuration.", err), this);
-      this.terminate(CallParty.Local, CallErrorCode.IceFailed, false);
-    });
-    _defineProperty(this, "onIceConnectionStateChanged", () => {
-      var _this$peerConn2, _this$peerConn3, _this$peerConn$iceCon, _this$peerConn4, _this$peerConn5, _this$peerConn8;
-      if (this.callHasEnded()) {
-        return; // because ICE can still complete as we're ending the call
-      }
-      logger.debug("Call ".concat(this.callId, " onIceConnectionStateChanged() running (state=").concat((_this$peerConn2 = this.peerConn) === null || _this$peerConn2 === void 0 ? void 0 : _this$peerConn2.iceConnectionState, ", conn=").concat((_this$peerConn3 = this.peerConn) === null || _this$peerConn3 === void 0 ? void 0 : _this$peerConn3.connectionState, ")"));
-
-      // 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$peerConn4 = this.peerConn) === null || _this$peerConn4 === void 0 ? void 0 : _this$peerConn4.iceConnectionState) !== null && _this$peerConn$iceCon !== void 0 ? _this$peerConn$iceCon : "")) {
-        clearTimeout(this.iceDisconnectedTimeout);
-        this.iceDisconnectedTimeout = undefined;
-        if (this.iceReconnectionTimeOut) {
-          clearTimeout(this.iceReconnectionTimeOut);
-        }
-        this.state = CallState.Connected;
-        if (!this.callLengthInterval && !this.callStartTime) {
-          this.callStartTime = Date.now();
-          this.callLengthInterval = setInterval(() => {
-            this.emit(CallEvent.LengthChanged, Math.round((Date.now() - this.callStartTime) / 1000), this);
-          }, CALL_LENGTH_INTERVAL);
-        }
-      } else if (((_this$peerConn5 = this.peerConn) === null || _this$peerConn5 === void 0 ? void 0 : _this$peerConn5.iceConnectionState) == "failed") {
-        var _this$peerConn6;
-        this.candidatesEnded = false;
-        // Firefox for Android does not yet have support for restartIce()
-        // (the types say it's always defined though, so we have to cast
-        // to prevent typescript from warning).
-        if ((_this$peerConn6 = this.peerConn) !== null && _this$peerConn6 !== void 0 && _this$peerConn6.restartIce) {
-          var _this$peerConn7;
-          this.candidatesEnded = false;
-          logger.debug("Call ".concat(this.callId, " onIceConnectionStateChanged() ice restart (state=").concat((_this$peerConn7 = this.peerConn) === null || _this$peerConn7 === void 0 ? void 0 : _this$peerConn7.iceConnectionState, ")"));
-          this.peerConn.restartIce();
-        } else {
-          logger.info("Call ".concat(this.callId, " onIceConnectionStateChanged() hanging up call (ICE failed and no ICE restart method)"));
-          this.hangup(CallErrorCode.IceFailed, false);
-        }
-      } else if (((_this$peerConn8 = this.peerConn) === null || _this$peerConn8 === void 0 ? void 0 : _this$peerConn8.iceConnectionState) == "disconnected") {
-        this.candidatesEnded = false;
-        this.iceReconnectionTimeOut = setTimeout(() => {
-          var _this$peerConn9, _this$peerConn10, _this$peerConn11;
-          logger.info("Call ".concat(this.callId, " onIceConnectionStateChanged() ICE restarting because of ICE disconnected, (state=").concat((_this$peerConn9 = this.peerConn) === null || _this$peerConn9 === void 0 ? void 0 : _this$peerConn9.iceConnectionState, ", conn=").concat((_this$peerConn10 = this.peerConn) === null || _this$peerConn10 === void 0 ? void 0 : _this$peerConn10.connectionState, ")"));
-          if ((_this$peerConn11 = this.peerConn) !== null && _this$peerConn11 !== void 0 && _this$peerConn11.restartIce) {
-            this.candidatesEnded = false;
-            this.peerConn.restartIce();
-          }
-          this.iceReconnectionTimeOut = undefined;
-        }, ICE_RECONNECTING_TIMEOUT);
-        this.iceDisconnectedTimeout = setTimeout(() => {
-          logger.info("Call ".concat(this.callId, " onIceConnectionStateChanged() hanging up call (ICE disconnected for too long)"));
-          this.hangup(CallErrorCode.IceFailed, false);
-        }, ICE_DISCONNECTED_TIMEOUT);
-        this.state = CallState.Connecting;
-      }
-
-      // In PTT mode, override feed status to muted when we lose connection to
-      // the peer, since we don't want to block the line if they're not saying anything.
-      // Experimenting in Chrome, this happens after 5 or 6 seconds, which is probably
-      // fast enough.
-      if (this.isPtt && ["failed", "disconnected"].includes(this.peerConn.iceConnectionState)) {
-        for (var feed of this.getRemoteFeeds()) {
-          feed.setAudioVideoMuted(true, true);
-        }
-      }
-    });
-    _defineProperty(this, "onSignallingStateChanged", () => {
-      var _this$peerConn12;
-      logger.debug("Call ".concat(this.callId, " onSignallingStateChanged() running (state=").concat((_this$peerConn12 = this.peerConn) === null || _this$peerConn12 === void 0 ? void 0 : _this$peerConn12.signalingState, ")"));
-    });
-    _defineProperty(this, "onTrack", ev => {
-      if (ev.streams.length === 0) {
-        logger.warn("Call ".concat(this.callId, " onTrack() called with streamless track streamless (kind=").concat(ev.track.kind, ")"));
-        return;
-      }
-      var stream = ev.streams[0];
-      this.pushRemoteFeed(stream);
-      if (!this.removeTrackListeners.has(stream)) {
-        var onRemoveTrack = () => {
-          if (stream.getTracks().length === 0) {
-            logger.info("Call ".concat(this.callId, " onTrack() removing track (streamId=").concat(stream.id, ")"));
-            this.deleteFeedByStream(stream);
-            stream.removeEventListener("removetrack", onRemoveTrack);
-            this.removeTrackListeners.delete(stream);
-          }
-        };
-        stream.addEventListener("removetrack", onRemoveTrack);
-        this.removeTrackListeners.set(stream, onRemoveTrack);
-      }
-    });
-    _defineProperty(this, "onDataChannel", ev => {
-      this.emit(CallEvent.DataChannel, ev.channel, this);
-    });
-    _defineProperty(this, "onNegotiationNeeded", /*#__PURE__*/_asyncToGenerator(function* () {
-      logger.info("Call ".concat(_this.callId, " onNegotiationNeeded() negotiation is needed!"));
-      if (_this.state !== CallState.CreateOffer && _this.opponentVersion === 0) {
-        logger.info("Call ".concat(_this.callId, " onNegotiationNeeded() opponent does not support renegotiation: ignoring negotiationneeded event"));
-        return;
-      }
-      _this.queueGotLocalOffer();
-    }));
-    _defineProperty(this, "onHangupReceived", msg => {
-      logger.debug("Call ".concat(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.info("Call ".concat(this.callId, " onHangupReceived() ignoring message from party ID ").concat(msg.party_id, ": our partner is ").concat(this.opponentPartyId));
-      }
-    });
-    _defineProperty(this, "onRejectReceived", msg => {
-      logger.debug("Call ".concat(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
-
-      var 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.debug("Call ".concat(this.callId, " onRejectReceived() called in wrong state (state=").concat(this.state, ")"));
-      }
-    });
-    _defineProperty(this, "onAnsweredElsewhere", msg => {
-      logger.debug("Call ".concat(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 (var server of this.turnServers) {
-      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.
-   */
-  placeVoiceCall() {
-    var _this2 = this;
-    return _asyncToGenerator(function* () {
-      yield _this2.placeCall(true, false);
-    })();
-  }
-
-  /**
-   * Place a video call to this room.
-   * @throws If you have not specified a listener for 'error' events.
-   */
-  placeVideoCall() {
-    var _this3 = this;
-    return _asyncToGenerator(function* () {
-      yield _this3.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) {
-    var 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) {
-    var 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$stream;
-      return feed.purpose === SDPStreamMetadataPurpose.Usermedia && ((_feed$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$stream2;
-      return feed.purpose === SDPStreamMetadataPurpose.Usermedia && !!((_feed$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(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(SDPStreamMetadataPurpose.Usermedia, "video"))) === null || _this$transceivers$ge2 === void 0 ? void 0 : _this$transceivers$ge2.sender);
-  }
-  get localUsermediaFeed() {
-    return this.getLocalFeeds().find(feed => feed.purpose === SDPStreamMetadataPurpose.Usermedia);
-  }
-  get localScreensharingFeed() {
-    return this.getLocalFeeds().find(feed => feed.purpose === 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 === SDPStreamMetadataPurpose.Usermedia);
-  }
-  get remoteScreensharingFeed() {
-    return this.getRemoteFeeds().find(feed => feed.purpose === 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());
-  }
-  initOpponentCrypto() {
-    var _this4 = this;
-    return _asyncToGenerator(function* () {
-      var _this4$getOpponentMem, _deviceInfoMap$get;
-      if (!_this4.opponentDeviceId) return;
-      if (!_this4.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 (!_this4.client.isCryptoEnabled()) {
-        // All we know is the device ID
-        _this4.opponentDeviceInfo = new DeviceInfo(_this4.opponentDeviceId);
-        return;
-      }
-      // if we've got to this point, we do want to init crypto, so throw if we can't
-      if (!_this4.client.crypto) throw new Error("Crypto is not initialised.");
-      var userId = _this4.invitee || ((_this4$getOpponentMem = _this4.getOpponentMember()) === null || _this4$getOpponentMem === void 0 ? void 0 : _this4$getOpponentMem.userId);
-      if (!userId) throw new Error("Couldn't find opponent user ID to init crypto");
-      var deviceInfoMap = yield _this4.client.crypto.deviceList.downloadKeys([userId], false);
-      _this4.opponentDeviceInfo = (_deviceInfoMap$get = deviceInfoMap.get(userId)) === null || _deviceInfoMap$get === void 0 ? void 0 : _deviceInfoMap$get.get(_this4.opponentDeviceId);
-      if (_this4.opponentDeviceInfo === undefined) {
-        throw new GroupCallUnknownDeviceError(userId);
-      }
-    })();
-  }
-
-  /**
-   * Generates and returns localSDPStreamMetadata
-   * @returns localSDPStreamMetadata
-   */
-  getLocalSDPStreamMetadata() {
-    var updateStreamIds = arguments.length > 0 && arguments[0] !== undefined ? arguments[0] : false;
-    var metadata = {};
-    for (var 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;
-    }
-    var userId = this.getOpponentMember().userId;
-    var purpose = this.remoteSDPStreamMetadata[stream.id].purpose;
-    var audioMuted = this.remoteSDPStreamMetadata[stream.id].audio_muted;
-    var videoMuted = this.remoteSDPStreamMetadata[stream.id].video_muted;
-    if (!purpose) {
-      logger.warn("Call ".concat(this.callId, " pushRemoteFeed() ignoring stream because we didn't get any metadata about it (streamId=").concat(stream.id, ")"));
-      return;
-    }
-    if (this.getFeedByStreamId(stream.id)) {
-      logger.warn("Call ".concat(this.callId, " pushRemoteFeed() ignoring stream because we already have a feed for it (streamId=").concat(stream.id, ")"));
-      return;
-    }
-    this.feeds.push(new 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.info("Call ".concat(this.callId, " pushRemoteFeed() pushed stream (streamId=").concat(stream.id, ", active=").concat(stream.active, ", purpose=").concat(purpose, ")"));
-  }
-
-  /**
-   * This method is used ONLY if the other client doesn't support sending SDPStreamMetadata
-   */
-  pushRemoteFeedWithoutMetadata(stream) {
-    var _this$feeds$find;
-    var userId = this.getOpponentMember().userId;
-    // We can guess the purpose here since the other client can only send one stream
-    var purpose = SDPStreamMetadataPurpose.Usermedia;
-    var 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.warn("Call ".concat(this.callId, " pushRemoteFeedWithoutMetadata() ignoring new stream because we already have stream (streamId=").concat(stream.id, ")"));
-      return;
-    }
-    if (this.getFeedByStreamId(stream.id)) {
-      logger.warn("Call ".concat(this.callId, " pushRemoteFeedWithoutMetadata() ignoring stream because we already have a feed for it (streamId=").concat(stream.id, ")"));
-      return;
-    }
-    this.feeds.push(new 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.info("Call ".concat(this.callId, " pushRemoteFeedWithoutMetadata() pushed stream (streamId=").concat(stream.id, ", active=").concat(stream.active, ")"));
-  }
-  pushNewLocalFeed(stream, purpose) {
-    var addToPeerConnection = arguments.length > 2 && arguments[2] !== undefined ? arguments[2] : true;
-    var 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.warn("Call ".concat(this.callId, " pushNewLocalFeed() ignoring stream because we already have a feed for it (streamId=").concat(stream.id, ")"));
-      return;
-    }
-    this.pushLocalFeed(new 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) {
-    var _this5 = this;
-    var addToPeerConnection = arguments.length > 1 && arguments[1] !== undefined ? arguments[1] : true;
-    if (this.feeds.some(feed => callFeed.stream.id === feed.stream.id)) {
-      logger.info("Call ".concat(this.callId, " pushLocalFeed() ignoring duplicate local stream (streamId=").concat(callFeed.stream.id, ")"));
-      return;
-    }
-    this.feeds.push(callFeed);
-    if (addToPeerConnection) {
-      var _loop = function _loop() {
-        logger.info("Call ".concat(_this5.callId, " pushLocalFeed() adding track to peer connection (id=").concat(track.id, ", kind=").concat(track.kind, ", streamId=").concat(callFeed.stream.id, ", streamPurpose=").concat(callFeed.purpose, ", enabled=").concat(track.enabled, ")"));
-        var tKey = getTransceiverKey(callFeed.purpose, track.kind);
-        if (_this5.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.
-          var transceiver = _this5.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.
-          var newSender = _this5.peerConn.addTrack(track, callFeed.stream);
-
-          // now go & fish for the new transceiver
-          var newTransceiver = _this5.peerConn.getTransceivers().find(t => t.sender === newSender);
-          if (newTransceiver) {
-            _this5.transceivers.set(tKey, newTransceiver);
-          } else {
-            logger.warn("Call ".concat(_this5.callId, " pushLocalFeed() didn't find a matching transceiver after adding track!"));
-          }
-        }
-      };
-      for (var track of callFeed.stream.getTracks()) {
-        _loop();
-      }
-    }
-    logger.info("Call ".concat(this.callId, " pushLocalFeed() pushed stream (id=").concat(callFeed.stream.id, ", active=").concat(callFeed.stream.active, ", purpose=").concat(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) {
-    var audioTransceiverKey = getTransceiverKey(callFeed.purpose, "audio");
-    var videoTransceiverKey = getTransceiverKey(callFeed.purpose, "video");
-    for (var 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)) {
-        var transceiver = this.transceivers.get(transceiverKey);
-        if (transceiver.sender) this.peerConn.removeTrack(transceiver.sender);
-      }
-    }
-    if (callFeed.purpose === SDPStreamMetadataPurpose.Screenshare) {
-      this.client.getMediaHandler().stopScreensharingStream(callFeed.stream);
-    }
-    this.deleteFeed(callFeed);
-  }
-  deleteAllFeeds() {
-    for (var feed of this.feeds) {
-      if (!feed.isLocal() || !this.groupCallId) {
-        feed.dispose();
-      }
-    }
-    this.feeds = [];
-    this.emit(CallEvent.FeedsChanged, this.feeds, this);
-  }
-  deleteFeedByStream(stream) {
-    var feed = this.getFeedByStreamId(stream.id);
-    if (!feed) {
-      logger.warn("Call ".concat(this.callId, " deleteFeedByStream() didn't find the feed to delete (streamId=").concat(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' :(
-  getCurrentCallStats() {
-    var _this6 = this;
-    return _asyncToGenerator(function* () {
-      if (_this6.callHasEnded()) {
-        return _this6.callStatsAtEnd;
-      }
-      return _this6.collectCallStats();
-    })();
-  }
-  collectCallStats() {
-    var _this7 = this;
-    return _asyncToGenerator(function* () {
-      // This happens when the call fails before it starts.
-      // For example when we fail to get capture sources
-      if (!_this7.peerConn) return;
-      var statsReport = yield _this7.peerConn.getStats();
-      var 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
-   */
-  initWithInvite(event) {
-    var _this8 = this;
-    return _asyncToGenerator(function* () {
-      var _this8$feeds$find;
-      var invite = event.getContent();
-      _this8.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.
-      var haveTurnCreds = yield _this8.client.checkTurnServers();
-      if (!haveTurnCreds) {
-        logger.warn("Call ".concat(_this8.callId, " initWithInvite() failed to get TURN credentials! Proceeding with call anyway..."));
-      }
-      var sdpStreamMetadata = invite[SDPStreamMetadataKey];
-      if (sdpStreamMetadata) {
-        _this8.updateRemoteSDPStreamMetadata(sdpStreamMetadata);
-      } else {
-        logger.debug("Call ".concat(_this8.callId, " initWithInvite() did not get any SDPStreamMetadata! Can not send/receive multiple streams"));
-      }
-      _this8.peerConn = _this8.createPeerConnection();
-      _this8.emit(CallEvent.PeerConnectionCreated, _this8.peerConn, _this8);
-      // 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.
-      _this8.chooseOpponent(event);
-      yield _this8.initOpponentCrypto();
-      try {
-        yield _this8.peerConn.setRemoteDescription(invite.offer);
-        logger.debug("Call ".concat(_this8.callId, " initWithInvite() set remote description: ").concat(invite.offer.type));
-        yield _this8.addBufferedIceCandidates();
-      } catch (e) {
-        logger.debug("Call ".concat(_this8.callId, " initWithInvite() failed to set remote description"), e);
-        _this8.terminate(CallParty.Local, CallErrorCode.SetRemoteDescription, false);
-        return;
-      }
-      var remoteStream = (_this8$feeds$find = _this8.feeds.find(feed => !feed.isLocal())) === null || _this8$feeds$find === void 0 ? void 0 : _this8$feeds$find.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 (!_this8.isOnlyDataChannelAllowed && (!remoteStream || remoteStream.getTracks().length === 0)) {
-        logger.error("Call ".concat(_this8.callId, " initWithInvite() no remote stream or no tracks after setting remote description!"));
-        _this8.terminate(CallParty.Local, CallErrorCode.SetRemoteDescription, false);
-        return;
-      }
-      _this8.state = CallState.Ringing;
-      if (event.getLocalAge()) {
-        // Time out the call if it's ringing for too long
-        var ringingTimer = setTimeout(() => {
-          if (_this8.state == CallState.Ringing) {
-            var _this8$stats;
-            logger.debug("Call ".concat(_this8.callId, " initWithInvite() invite has expired. Hanging up."));
-            _this8.hangupParty = CallParty.Remote; // effectively
-            _this8.state = CallState.Ended;
-            _this8.stopAllMedia();
-            if (_this8.peerConn.signalingState != "closed") {
-              _this8.peerConn.close();
-            }
-            (_this8$stats = _this8.stats) === null || _this8$stats === void 0 || _this8$stats.removeStatsReportGatherer(_this8.callId);
-            _this8.emit(CallEvent.Hangup, _this8);
-          }
-        }, invite.lifetime - event.getLocalAge());
-        var onState = state => {
-          if (state !== CallState.Ringing) {
-            clearTimeout(ringingTimer);
-            _this8.off(CallEvent.State, onState);
-          }
-        };
-        _this8.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.warn("Call ".concat(this.callId, " shouldAnswerWithMediaType() unable to answer with ").concat(type, " because the other side isn't sending it either."));
-      return false;
-    } else if (!isNullOrUndefined(wantedValue) && wantedValue !== valueOfTheOtherSide && !this.opponentSupportsSDPStreamMetadata()) {
-      logger.warn("Call ".concat(this.callId, " shouldAnswerWithMediaType() unable to answer with ").concat(type, "=").concat(wantedValue, " because the other side doesn't support it. Answering with ").concat(type, "=").concat(valueOfTheOtherSide, "."));
-      return valueOfTheOtherSide;
-    }
-    return wantedValue !== null && wantedValue !== void 0 ? wantedValue : valueOfTheOtherSide;
-  }
-
-  /**
-   * Answer a call.
-   */
-  answer(audio, video) {
-    var _this9 = this;
-    return _asyncToGenerator(function* () {
-      if (_this9.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 (!_this9.localUsermediaStream && !_this9.waitForLocalAVStream) {
-        var prevState = _this9.state;
-        var answerWithAudio = _this9.shouldAnswerWithMediaType(audio, _this9.hasRemoteUserMediaAudioTrack, "audio");
-        var answerWithVideo = _this9.shouldAnswerWithMediaType(video, _this9.hasRemoteUserMediaVideoTrack, "video");
-        _this9.state = CallState.WaitLocalMedia;
-        _this9.waitForLocalAVStream = true;
-        try {
-          var _this9$client$getDevi;
-          var stream = yield _this9.client.getMediaHandler().getUserMediaStream(answerWithAudio, answerWithVideo);
-          _this9.waitForLocalAVStream = false;
-          var usermediaFeed = new CallFeed({
-            client: _this9.client,
-            roomId: _this9.roomId,
-            userId: _this9.client.getUserId(),
-            deviceId: (_this9$client$getDevi = _this9.client.getDeviceId()) !== null && _this9$client$getDevi !== void 0 ? _this9$client$getDevi : undefined,
-            stream,
-            purpose: SDPStreamMetadataPurpose.Usermedia,
-            audioMuted: false,
-            videoMuted: false
-          });
-          var feeds = [usermediaFeed];
-          if (_this9.localScreensharingFeed) {
-            feeds.push(_this9.localScreensharingFeed);
-          }
-          _this9.answerWithCallFeeds(feeds);
-        } catch (e) {
-          if (answerWithVideo) {
-            // Try to answer without video
-            logger.warn("Call ".concat(_this9.callId, " answer() failed to getUserMedia(), trying to getUserMedia() without video"));
-            _this9.state = prevState;
-            _this9.waitForLocalAVStream = false;
-            yield _this9.answer(answerWithAudio, false);
-          } else {
-            _this9.getUserMediaFailed(e);
-            return;
-          }
-        }
-      } else if (_this9.waitForLocalAVStream) {
-        _this9.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.debug("Call ".concat(this.callId, " replacedBy() running (newCallId=").concat(newCall.callId, ")"));
-    if (this.state === CallState.WaitLocalMedia) {
-      logger.debug("Call ".concat(this.callId, " replacedBy() telling new call to wait for local media (newCallId=").concat(newCall.callId, ")"));
-      newCall.waitForLocalAVStream = true;
-    } else if ([CallState.CreateOffer, CallState.InviteSent].includes(this.state)) {
-      if (newCall.direction === CallDirection.Outbound) {
-        newCall.queueGotCallFeedsForAnswer([]);
-      } else {
-        logger.debug("Call ".concat(this.callId, " replacedBy() handing local stream to new call(newCallId=").concat(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.debug("Call ".concat(this.callId, " hangup() ending call (reason=").concat(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;
-    var content = {};
-    // Don't send UserHangup reason to older clients
-    if (this.opponentVersion && this.opponentVersion !== 0 || reason !== CallErrorCode.UserHangup) {
-      content["reason"] = reason;
-    }
-    this.sendVoipEvent(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.info("Call ".concat(this.callId, " reject() opponent version is less than 1: sending hangup instead of reject (opponentVersion=").concat(this.opponentVersion, ")"));
-      this.hangup(CallErrorCode.UserHangup, true);
-      return;
-    }
-    logger.debug("Rejecting call: " + this.callId);
-    this.terminate(CallParty.Local, CallErrorCode.UserHangup, true);
-    this.sendVoipEvent(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
-   */
-  upgradeCall(audio, video) {
-    var _this10 = this;
-    return _asyncToGenerator(function* () {
-      // We don't do call downgrades
-      if (!audio && !video) return;
-      if (!_this10.opponentSupportsSDPStreamMetadata()) return;
-      try {
-        logger.debug("Call ".concat(_this10.callId, " upgradeCall() upgrading call (audio=").concat(audio, ", video=").concat(video, ")"));
-        var getAudio = audio || _this10.hasLocalUserMediaAudioTrack;
-        var getVideo = video || _this10.hasLocalUserMediaVideoTrack;
-
-        // updateLocalUsermediaStream() will take the tracks, use them as
-        // replacement and throw the stream away, so it isn't reusable
-        var stream = yield _this10.client.getMediaHandler().getUserMediaStream(getAudio, getVideo, false);
-        yield _this10.updateLocalUsermediaStream(stream, audio, video);
-      } catch (error) {
-        logger.error("Call ".concat(_this10.callId, " upgradeCall() failed to upgrade the call"), error);
-        _this10.emit(CallEvent.Error, new CallError(CallErrorCode.NoUserMedia, "Failed to get camera access: ", error), _this10);
-      }
-    })();
-  }
-
-  /**
-   * 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 opts - screen sharing options
-   * @returns new screensharing state
-   */
-  setScreensharingEnabled(enabled, opts) {
-    var _this11 = this;
-    return _asyncToGenerator(function* () {
-      // Skip if there is nothing to do
-      if (enabled && _this11.isScreensharing()) {
-        logger.warn("Call ".concat(_this11.callId, " setScreensharingEnabled() there is already a screensharing stream - there is nothing to do!"));
-        return true;
-      } else if (!enabled && !_this11.isScreensharing()) {
-        logger.warn("Call ".concat(_this11.callId, " setScreensharingEnabled() there already isn't a screensharing stream - there is nothing to do!"));
-        return false;
-      }
-
-      // Fallback to replaceTrack()
-      if (!_this11.opponentSupportsSDPStreamMetadata()) {
-        return _this11.setScreensharingEnabledWithoutMetadataSupport(enabled, opts);
-      }
-      logger.debug("Call ".concat(_this11.callId, " setScreensharingEnabled() running (enabled=").concat(enabled, ")"));
-      if (enabled) {
-        try {
-          var stream = yield _this11.client.getMediaHandler().getScreensharingStream(opts);
-          if (!stream) return false;
-          _this11.pushNewLocalFeed(stream, SDPStreamMetadataPurpose.Screenshare);
-          return true;
-        } catch (err) {
-          logger.error("Call ".concat(_this11.callId, " setScreensharingEnabled() failed to get screen-sharing stream:"), err);
-          return false;
-        }
-      } else {
-        var audioTransceiver = _this11.transceivers.get(getTransceiverKey(SDPStreamMetadataPurpose.Screenshare, "audio"));
-        var videoTransceiver = _this11.transceivers.get(getTransceiverKey(SDPStreamMetadataPurpose.Screenshare, "video"));
-        for (var 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) _this11.peerConn.removeTrack(transceiver.sender);
-        }
-        _this11.client.getMediaHandler().stopScreensharingStream(_this11.localScreensharingStream);
-        _this11.deleteFeedByStream(_this11.localScreensharingStream);
-        return false;
-      }
-    })();
-  }
-
-  /**
-   * Starts/stops screensharing
-   * Should be used ONLY if the opponent doesn't support SDPStreamMetadata
-   * @param enabled - the desired screensharing state
-   * @param opts - screen sharing options
-   * @returns new screensharing state
-   */
-  setScreensharingEnabledWithoutMetadataSupport(enabled, opts) {
-    var _this12 = this;
-    return _asyncToGenerator(function* () {
-      logger.debug("Call ".concat(_this12.callId, " setScreensharingEnabledWithoutMetadataSupport() running (enabled=").concat(enabled, ")"));
-      if (enabled) {
-        try {
-          var _this12$transceivers$;
-          var stream = yield _this12.client.getMediaHandler().getScreensharingStream(opts);
-          if (!stream) return false;
-          var track = stream.getTracks().find(track => track.kind === "video");
-          var sender = (_this12$transceivers$ = _this12.transceivers.get(getTransceiverKey(SDPStreamMetadataPurpose.Usermedia, "video"))) === null || _this12$transceivers$ === void 0 ? void 0 : _this12$transceivers$.sender;
-          sender === null || sender === void 0 || sender.replaceTrack(track !== null && track !== void 0 ? track : null);
-          _this12.pushNewLocalFeed(stream, SDPStreamMetadataPurpose.Screenshare, false);
-          return true;
-        } catch (err) {
-          logger.error("Call ".concat(_this12.callId, " setScreensharingEnabledWithoutMetadataSupport() failed to get screen-sharing stream:"), err);
-          return false;
-        }
-      } else {
-        var _this12$localUsermedi, _this12$transceivers$2;
-        var _track = (_this12$localUsermedi = _this12.localUsermediaStream) === null || _this12$localUsermedi === void 0 ? void 0 : _this12$localUsermedi.getTracks().find(track => track.kind === "video");
-        var _sender = (_this12$transceivers$2 = _this12.transceivers.get(getTransceiverKey(SDPStreamMetadataPurpose.Usermedia, "video"))) === null || _this12$transceivers$2 === void 0 ? void 0 : _this12$transceivers$2.sender;
-        _sender === null || _sender === void 0 || _sender.replaceTrack(_track !== null && _track !== void 0 ? _track : null);
-        _this12.client.getMediaHandler().stopScreensharingStream(_this12.localScreensharingStream);
-        _this12.deleteFeedByStream(_this12.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
-   */
-  updateLocalUsermediaStream(stream) {
-    var _arguments = arguments,
-      _this13 = this;
-    return _asyncToGenerator(function* () {
-      var forceAudio = _arguments.length > 1 && _arguments[1] !== undefined ? _arguments[1] : false;
-      var forceVideo = _arguments.length > 2 && _arguments[2] !== undefined ? _arguments[2] : false;
-      var callFeed = _this13.localUsermediaFeed;
-      var audioEnabled = forceAudio || !callFeed.isAudioMuted() && !_this13.remoteOnHold;
-      var videoEnabled = forceVideo || !callFeed.isVideoMuted() && !_this13.remoteOnHold;
-      logger.log("Call ".concat(_this13.callId, " updateLocalUsermediaStream() running (streamId=").concat(stream.id, ", audio=").concat(audioEnabled, ", video=").concat(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 (var track of _this13.localUsermediaStream.getTracks()) {
-        _this13.localUsermediaStream.removeTrack(track);
-        track.stop();
-      }
-      for (var _track2 of stream.getTracks()) {
-        _this13.localUsermediaStream.addTrack(_track2);
-      }
-
-      // Then replace the old tracks, if possible.
-      var _loop2 = function* _loop2() {
-        var tKey = getTransceiverKey(SDPStreamMetadataPurpose.Usermedia, _track3.kind);
-        var transceiver = _this13.transceivers.get(tKey);
-        var oldSender = transceiver === null || transceiver === void 0 ? void 0 : transceiver.sender;
-        var added = false;
-        if (oldSender) {
-          try {
-            logger.info("Call ".concat(_this13.callId, " updateLocalUsermediaStream() replacing track (id=").concat(_track3.id, ", kind=").concat(_track3.kind, ", streamId=").concat(stream.id, ", streamPurpose=").concat(callFeed.purpose, ")"));
-            yield oldSender.replaceTrack(_track3);
-            // 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.warn("Call ".concat(_this13.callId, " updateLocalUsermediaStream() replaceTrack failed: adding new transceiver instead"), error);
-          }
-        }
-        if (!added) {
-          logger.info("Call ".concat(_this13.callId, " updateLocalUsermediaStream() adding track to peer connection (id=").concat(_track3.id, ", kind=").concat(_track3.kind, ", streamId=").concat(stream.id, ", streamPurpose=").concat(callFeed.purpose, ")"));
-          var newSender = _this13.peerConn.addTrack(_track3, _this13.localUsermediaStream);
-          var newTransceiver = _this13.peerConn.getTransceivers().find(t => t.sender === newSender);
-          if (newTransceiver) {
-            _this13.transceivers.set(tKey, newTransceiver);
-          } else {
-            logger.warn("Call ".concat(_this13.callId, " updateLocalUsermediaStream() couldn't find matching transceiver for newly added track!"));
-          }
-        }
-      };
-      for (var _track3 of stream.getTracks()) {
-        yield* _loop2();
-      }
-    })();
-  }
-
-  /**
-   * Set whether our outbound video should be muted or not.
-   * @param muted - True to mute the outbound video.
-   * @returns the new mute state
-   */
-  setLocalVideoMuted(muted) {
-    var _this14 = this;
-    return _asyncToGenerator(function* () {
-      var _this14$localUsermedi2;
-      logger.log("Call ".concat(_this14.callId, " setLocalVideoMuted() running ").concat(muted));
-
-      // if we were still thinking about stopping and removing the video
-      // track: don't, because we want it back.
-      if (!muted && _this14.stopVideoTrackTimer !== undefined) {
-        clearTimeout(_this14.stopVideoTrackTimer);
-        _this14.stopVideoTrackTimer = undefined;
-      }
-      if (!(yield _this14.client.getMediaHandler().hasVideoDevice())) {
-        return _this14.isLocalVideoMuted();
-      }
-      if (!_this14.hasUserMediaVideoSender && !muted) {
-        var _this14$localUsermedi;
-        (_this14$localUsermedi = _this14.localUsermediaFeed) === null || _this14$localUsermedi === void 0 || _this14$localUsermedi.setAudioVideoMuted(null, muted);
-        yield _this14.upgradeCall(false, true);
-        return _this14.isLocalVideoMuted();
-      }
-
-      // we may not have a video track - if not, re-request usermedia
-      if (!muted && _this14.localUsermediaStream.getVideoTracks().length === 0) {
-        var stream = yield _this14.client.getMediaHandler().getUserMediaStream(true, true);
-        yield _this14.updateLocalUsermediaStream(stream);
-      }
-      (_this14$localUsermedi2 = _this14.localUsermediaFeed) === null || _this14$localUsermedi2 === void 0 || _this14$localUsermedi2.setAudioVideoMuted(null, muted);
-      _this14.updateMuteStatus();
-      yield _this14.sendMetadataUpdate();
-
-      // if we're muting video, set a timeout to stop & remove the video track so we release
-      // the camera. We wait a short time to do this because when we disable a track, WebRTC
-      // will send black video for it. If we just stop and remove it straight away, the video
-      // will just freeze which means that when we unmute video, the other side will briefly
-      // get a static frame of us from before we muted. This way, the still frame is just black.
-      // A very small delay is not always enough so the theory here is that it needs to be long
-      // enough for WebRTC to encode a frame: 120ms should be long enough even if we're only
-      // doing 10fps.
-      if (muted) {
-        _this14.stopVideoTrackTimer = setTimeout(() => {
-          for (var t of _this14.localUsermediaStream.getVideoTracks()) {
-            t.stop();
-            _this14.localUsermediaStream.removeTrack(t);
-          }
-        }, 120);
-      }
-      return _this14.isLocalVideoMuted();
-    })();
-  }
-
-  /**
-   * Check if local video is muted.
-   *
-   * If there are multiple video tracks, <i>all</i> of the tracks need to be muted
-   * for this to return true. This means if there are no video tracks, this will
-   * return true.
-   * @returns True if the local preview video is muted, else false
-   * (including if the call is not set up yet).
-   */
-  isLocalVideoMuted() {
-    var _this$localUsermediaF2, _this$localUsermediaF3;
-    return (_this$localUsermediaF2 = (_this$localUsermediaF3 = this.localUsermediaFeed) === null || _this$localUsermediaF3 === void 0 ? void 0 : _this$localUsermediaF3.isVideoMuted()) !== null && _this$localUsermediaF2 !== void 0 ? _this$localUsermediaF2 : false;
-  }
-
-  /**
-   * Set whether the microphone should be muted or not.
-   * @param muted - True to mute the mic.
-   * @returns the new mute state
-   */
-  setMicrophoneMuted(muted) {
-    var _this15 = this;
-    return _asyncToGenerator(function* () {
-      var _this15$localUsermedi;
-      logger.log("Call ".concat(_this15.callId, " setMicrophoneMuted() running ").concat(muted));
-      if (!(yield _this15.client.getMediaHandler().hasAudioDevice())) {
-        return _this15.isMicrophoneMuted();
-      }
-      if (!muted && (!_this15.hasUserMediaAudioSender || !_this15.hasLocalUserMediaAudioTrack)) {
-        yield _this15.upgradeCall(true, false);
-        return _this15.isMicrophoneMuted();
-      }
-      (_this15$localUsermedi = _this15.localUsermediaFeed) === null || _this15$localUsermedi === void 0 || _this15$localUsermedi.setAudioVideoMuted(muted, null);
-      _this15.updateMuteStatus();
-      yield _this15.sendMetadataUpdate();
-      return _this15.isMicrophoneMuted();
-    })();
-  }
-
-  /**
-   * Check if the microphone is muted.
-   *
-   * If there are multiple audio tracks, <i>all</i> of the tracks need to be muted
-   * for this to return true. This means if there are no audio tracks, this will
-   * return true.
-   * @returns True if the mic is muted, else false (including if the call
-   * is not set up yet).
-   */
-  isMicrophoneMuted() {
-    var _this$localUsermediaF4, _this$localUsermediaF5;
-    return (_this$localUsermediaF4 = (_this$localUsermediaF5 = this.localUsermediaFeed) === null || _this$localUsermediaF5 === void 0 ? void 0 : _this$localUsermediaF5.isAudioMuted()) !== null && _this$localUsermediaF4 !== void 0 ? _this$localUsermediaF4 : false;
-  }
-
-  /**
-   * @returns true if we have put the party on the other side of the call on hold
-   * (that is, we are signalling to them that we are not listening)
-   */
-  isRemoteOnHold() {
-    return this.remoteOnHold;
-  }
-  setRemoteOnHold(onHold) {
-    if (this.isRemoteOnHold() === onHold) return;
-    this.remoteOnHold = onHold;
-    for (var transceiver of this.peerConn.getTransceivers()) {
-      // We don't send hold music or anything so we're not actually
-      // sending anything, but sendrecv is fairly standard for hold and
-      // it makes it a lot easier to figure out who's put who on hold.
-      transceiver.direction = onHold ? "sendonly" : "sendrecv";
-    }
-    this.updateMuteStatus();
-    this.sendMetadataUpdate();
-    this.emit(CallEvent.RemoteHoldUnhold, this.remoteOnHold, this);
-  }
-
-  /**
-   * Indicates whether we are 'on hold' to the remote party (ie. if true,
-   * they cannot hear us).
-   * @returns true if the other party has put us on hold
-   */
-  isLocalOnHold() {
-    if (this.state !== CallState.Connected) return false;
-    var callOnHold = true;
-
-    // We consider a call to be on hold only if *all* the tracks are on hold
-    // (is this the right thing to do?)
-    for (var transceiver of this.peerConn.getTransceivers()) {
-      var trackOnHold = ["inactive", "recvonly"].includes(transceiver.currentDirection);
-      if (!trackOnHold) callOnHold = false;
-    }
-    return callOnHold;
-  }
-
-  /**
-   * Sends a DTMF digit to the other party
-   * @param digit - The digit (nb. string - '#' and '*' are dtmf too)
-   */
-  sendDtmfDigit(digit) {
-    for (var sender of this.peerConn.getSenders()) {
-      var _sender$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() {
-    var micShouldBeMuted = this.isMicrophoneMuted() || this.remoteOnHold;
-    var vidShouldBeMuted = this.isLocalVideoMuted() || this.remoteOnHold;
-    logger.log("Call ".concat(this.callId, " updateMuteStatus stream ").concat(this.localUsermediaStream.id, " micShouldBeMuted ").concat(micShouldBeMuted, " vidShouldBeMuted ").concat(vidShouldBeMuted));
-    setTracksEnabled(this.localUsermediaStream.getAudioTracks(), !micShouldBeMuted);
-    setTracksEnabled(this.localUsermediaStream.getVideoTracks(), !vidShouldBeMuted);
-  }
-  sendMetadataUpdate() {
-    var _this16 = this;
-    return _asyncToGenerator(function* () {
-      yield _this16.sendVoipEvent(EventType.CallSDPStreamMetadataChangedPrefix, {
-        [SDPStreamMetadataKey]: _this16.getLocalSDPStreamMetadata()
-      });
-    })();
-  }
-  gotCallFeedsForInvite(callFeeds) {
-    var requestScreenshareFeed = arguments.length > 1 && arguments[1] !== undefined ? arguments[1] : false;
-    if (this.successor) {
-      this.successor.queueGotCallFeedsForAnswer(callFeeds);
-      return;
-    }
-    if (this.callHasEnded()) {
-      this.stopAllMedia();
-      return;
-    }
-    for (var feed of callFeeds) {
-      this.pushLocalFeed(feed);
-    }
-    if (requestScreenshareFeed) {
-      this.peerConn.addTransceiver("video", {
-        direction: "recvonly"
-      });
-    }
-    this.state = CallState.CreateOffer;
-    logger.debug("Call ".concat(this.callId, " gotUserMediaForInvite() run"));
-    // Now we wait for the negotiationneeded event
-  }
-  sendAnswer() {
-    var _this17 = this;
-    return _asyncToGenerator(function* () {
-      var answerContent = {
-        answer: {
-          sdp: _this17.peerConn.localDescription.sdp,
-          // type is now deprecated as of Matrix VoIP v1, but
-          // required to still be sent for backwards compat
-          type: _this17.peerConn.localDescription.type
-        },
-        [SDPStreamMetadataKey]: _this17.getLocalSDPStreamMetadata(true)
-      };
-      answerContent.capabilities = {
-        "m.call.transferee": _this17.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.
-      var discardCount = _this17.discardDuplicateCandidates();
-      logger.info("Call ".concat(_this17.callId, " sendAnswer() discarding ").concat(discardCount, " candidates that will be sent in answer"));
-      try {
-        yield _this17.sendVoipEvent(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.
-        _this17.inviteOrAnswerSent = true;
-      } catch (error) {
-        // We've failed to answer: back to the ringing state
-        _this17.state = CallState.Ringing;
-        if (error instanceof MatrixError && error.event) _this17.client.cancelPendingEvent(error.event);
-        var code = CallErrorCode.SendAnswer;
-        var message = "Failed to send answer";
-        if (error.name == "UnknownDeviceError") {
-          code = CallErrorCode.UnknownDevices;
-          message = "Unknown devices present in the room";
-        }
-        _this17.emit(CallEvent.Error, new CallError(code, message, error), _this17);
-        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
-      _this17.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
-    var sdp = parseSdp(description.sdp);
-    sdp.media.forEach(media => {
-      var payloadTypeToCodecMap = new Map();
-      var codecToPayloadTypeMap = new Map();
-      for (var rtp of media.rtp) {
-        payloadTypeToCodecMap.set(rtp.payload, rtp.codec);
-        codecToPayloadTypeMap.set(rtp.codec, rtp.payload);
-      }
-      for (var mod of mods) {
-        if (mod.mediaType !== media.type) continue;
-        if (!codecToPayloadTypeMap.has(mod.codec)) {
-          logger.info("Call ".concat(this.callId, " mungeSdp() ignoring SDP modifications for ").concat(mod.codec, " as it's not present."));
-          continue;
-        }
-        var extraConfig = [];
-        if (mod.enableDtx !== undefined) {
-          extraConfig.push("usedtx=".concat(mod.enableDtx ? "1" : "0"));
-        }
-        if (mod.maxAverageBitrate !== undefined) {
-          extraConfig.push("maxaveragebitrate=".concat(mod.maxAverageBitrate));
-        }
-        var found = false;
-        for (var 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 = writeSdp(sdp);
-  }
-  createOffer() {
-    var _this18 = this;
-    return _asyncToGenerator(function* () {
-      var offer = yield _this18.peerConn.createOffer();
-      _this18.mungeSdp(offer, getCodecParamMods(_this18.isPtt));
-      return offer;
-    })();
-  }
-  createAnswer() {
-    var _this19 = this;
-    return _asyncToGenerator(function* () {
-      var answer = yield _this19.peerConn.createAnswer();
-      _this19.mungeSdp(answer, getCodecParamMods(_this19.isPtt));
-      return answer;
-    })();
-  }
-  gotCallFeedsForAnswer(callFeeds) {
-    var _this20 = this;
-    return _asyncToGenerator(function* () {
-      if (_this20.callHasEnded()) return;
-      _this20.waitForLocalAVStream = false;
-      for (var feed of callFeeds) {
-        _this20.pushLocalFeed(feed);
-      }
-      _this20.state = CallState.CreateAnswer;
-      var answer;
-      try {
-        _this20.getRidOfRTXCodecs();
-        answer = yield _this20.createAnswer();
-      } catch (err) {
-        logger.debug("Call ".concat(_this20.callId, " gotCallFeedsForAnswer() failed to create answer: "), err);
-        _this20.terminate(CallParty.Local, CallErrorCode.CreateAnswer, true);
-        return;
-      }
-      try {
-        yield _this20.peerConn.setLocalDescription(answer);
-
-        // make sure we're still going
-        if (_this20.callHasEnded()) return;
-        _this20.state = CallState.Connecting;
-
-        // Allow a short time for initial candidates to be gathered
-        yield new Promise(resolve => {
-          setTimeout(resolve, 200);
-        });
-
-        // make sure the call hasn't ended before we continue
-        if (_this20.callHasEnded()) return;
-        _this20.sendAnswer();
-      } catch (err) {
-        logger.debug("Call ".concat(_this20.callId, " gotCallFeedsForAnswer() error setting local description!"), err);
-        _this20.terminate(CallParty.Local, CallErrorCode.SetLocalDescription, true);
-        return;
-      }
-    })();
-  }
-  onRemoteIceCandidatesReceived(ev) {
-    var _this21 = this;
-    return _asyncToGenerator(function* () {
-      if (_this21.callHasEnded()) {
-        //debuglog("Ignoring remote ICE candidate because call has ended");
-        return;
-      }
-      var content = ev.getContent();
-      var candidates = content.candidates;
-      if (!candidates) {
-        logger.info("Call ".concat(_this21.callId, " onRemoteIceCandidatesReceived() ignoring candidates event with no candidates!"));
-        return;
-      }
-      var fromPartyId = content.version === 0 ? null : content.party_id || null;
-      if (_this21.opponentPartyId === undefined) {
-        // we haven't picked an opponent yet so save the candidates
-        if (fromPartyId) {
-          logger.info("Call ".concat(_this21.callId, " onRemoteIceCandidatesReceived() buffering ").concat(candidates.length, " candidates until we pick an opponent"));
-          var bufferedCandidates = _this21.remoteCandidateBuffer.get(fromPartyId) || [];
-          bufferedCandidates.push(...candidates);
-          _this21.remoteCandidateBuffer.set(fromPartyId, bufferedCandidates);
-        }
-        return;
-      }
-      if (!_this21.partyIdMatches(content)) {
-        logger.info("Call ".concat(_this21.callId, " onRemoteIceCandidatesReceived() ignoring candidates from party ID ").concat(content.party_id, ": we have chosen party ID ").concat(_this21.opponentPartyId));
-        return;
-      }
-      yield _this21.addIceCandidates(candidates);
-    })();
-  }
-
-  /**
-   * Used by MatrixClient.
-   */
-  onAnswerReceived(event) {
-    var _this22 = this;
-    return _asyncToGenerator(function* () {
-      var content = event.getContent();
-      logger.debug("Call ".concat(_this22.callId, " onAnswerReceived() running (hangupParty=").concat(content.party_id, ")"));
-      if (_this22.callHasEnded()) {
-        logger.debug("Call ".concat(_this22.callId, " onAnswerReceived() ignoring answer because call has ended"));
-        return;
-      }
-      if (_this22.opponentPartyId !== undefined) {
-        logger.info("Call ".concat(_this22.callId, " onAnswerReceived() ignoring answer from party ID ").concat(content.party_id, ": we already have an answer/reject from ").concat(_this22.opponentPartyId));
-        return;
-      }
-      _this22.chooseOpponent(event);
-      yield _this22.addBufferedIceCandidates();
-      _this22.state = CallState.Connecting;
-      var sdpStreamMetadata = content[SDPStreamMetadataKey];
-      if (sdpStreamMetadata) {
-        _this22.updateRemoteSDPStreamMetadata(sdpStreamMetadata);
-      } else {
-        logger.warn("Call ".concat(_this22.callId, " onAnswerReceived() did not get any SDPStreamMetadata! Can not send/receive multiple streams"));
-      }
-      try {
-        _this22.isSettingRemoteAnswerPending = true;
-        yield _this22.peerConn.setRemoteDescription(content.answer);
-        _this22.isSettingRemoteAnswerPending = false;
-        logger.debug("Call ".concat(_this22.callId, " onAnswerReceived() set remote description: ").concat(content.answer.type));
-      } catch (e) {
-        _this22.isSettingRemoteAnswerPending = false;
-        logger.debug("Call ".concat(_this22.callId, " onAnswerReceived() failed to set remote description"), e);
-        _this22.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 (_this22.opponentPartyId !== null) {
-        try {
-          yield _this22.sendVoipEvent(EventType.CallSelectAnswer, {
-            selected_party_id: _this22.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.warn("Call ".concat(_this22.callId, " onAnswerReceived() failed to send select_answer event"), err);
-        }
-      }
-    })();
-  }
-  onSelectAnswerReceived(event) {
-    var _this23 = this;
-    return _asyncToGenerator(function* () {
-      if (_this23.direction !== CallDirection.Inbound) {
-        logger.warn("Call ".concat(_this23.callId, " onSelectAnswerReceived() got select_answer for an outbound call: ignoring"));
-        return;
-      }
-      var selectedPartyId = event.getContent().selected_party_id;
-      if (selectedPartyId === undefined || selectedPartyId === null) {
-        logger.warn("Call ".concat(_this23.callId, " onSelectAnswerReceived() got nonsensical select_answer with null/undefined selected_party_id: ignoring"));
-        return;
-      }
-      if (selectedPartyId !== _this23.ourPartyId) {
-        logger.info("Call ".concat(_this23.callId, " onSelectAnswerReceived() got select_answer for party ID ").concat(selectedPartyId, ": we are party ID ").concat(_this23.ourPartyId, "."));
-        // The other party has picked somebody else's answer
-        yield _this23.terminate(CallParty.Remote, CallErrorCode.AnsweredElsewhere, true);
-      }
-    })();
-  }
-  onNegotiateReceived(event) {
-    var _this24 = this;
-    return _asyncToGenerator(function* () {
-      var content = event.getContent();
-      var description = content.description;
-      if (!description || !description.sdp || !description.type) {
-        logger.info("Call ".concat(_this24.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
-      var polite = _this24.direction === CallDirection.Inbound;
-
-      // Here we follow the perfect negotiation logic from
-      // https://w3c.github.io/webrtc-pc/#perfect-negotiation-example
-      var readyForOffer = !_this24.makingOffer && (_this24.peerConn.signalingState === "stable" || _this24.isSettingRemoteAnswerPending);
-      var offerCollision = description.type === "offer" && !readyForOffer;
-      _this24.ignoreOffer = !polite && offerCollision;
-      if (_this24.ignoreOffer) {
-        logger.info("Call ".concat(_this24.callId, " onNegotiateReceived() ignoring colliding negotiate event because we're impolite"));
-        return;
-      }
-      var prevLocalOnHold = _this24.isLocalOnHold();
-      var sdpStreamMetadata = content[SDPStreamMetadataKey];
-      if (sdpStreamMetadata) {
-        _this24.updateRemoteSDPStreamMetadata(sdpStreamMetadata);
-      } else {
-        logger.warn("Call ".concat(_this24.callId, " onNegotiateReceived() received negotiation event without SDPStreamMetadata!"));
-      }
-      try {
-        _this24.isSettingRemoteAnswerPending = description.type == "answer";
-        yield _this24.peerConn.setRemoteDescription(description); // SRD rolls back as needed
-        _this24.isSettingRemoteAnswerPending = false;
-        logger.debug("Call ".concat(_this24.callId, " onNegotiateReceived() set remote description: ").concat(description.type));
-        if (description.type === "offer") {
-          var _localDescription;
-          var answer;
-          try {
-            _this24.getRidOfRTXCodecs();
-            answer = yield _this24.createAnswer();
-          } catch (err) {
-            logger.debug("Call ".concat(_this24.callId, " onNegotiateReceived() failed to create answer: "), err);
-            _this24.terminate(CallParty.Local, CallErrorCode.CreateAnswer, true);
-            return;
-          }
-          yield _this24.peerConn.setLocalDescription(answer);
-          logger.debug("Call ".concat(_this24.callId, " onNegotiateReceived() create an answer"));
-          _this24.sendVoipEvent(EventType.CallNegotiate, {
-            lifetime: CALL_TIMEOUT_MS,
-            description: (_localDescription = _this24.peerConn.localDescription) === null || _localDescription === void 0 ? void 0 : _localDescription.toJSON(),
-            [SDPStreamMetadataKey]: _this24.getLocalSDPStreamMetadata(true)
-          });
-        }
-      } catch (err) {
-        _this24.isSettingRemoteAnswerPending = false;
-        logger.warn("Call ".concat(_this24.callId, " onNegotiateReceived() failed to complete negotiation"), err);
-      }
-      var newLocalOnHold = _this24.isLocalOnHold();
-      if (prevLocalOnHold !== newLocalOnHold) {
-        _this24.emit(CallEvent.LocalHoldUnhold, newLocalOnHold, _this24);
-        // also this one for backwards compat
-        _this24.emit(CallEvent.HoldUnhold, newLocalOnHold);
-      }
-    })();
-  }
-  updateRemoteSDPStreamMetadata(metadata) {
-    this.remoteSDPStreamMetadata = recursivelyAssign(this.remoteSDPStreamMetadata || {}, metadata, true);
-    for (var feed of this.getRemoteFeeds()) {
-      var _streamId;
-      var streamId = feed.stream.id;
-      var _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) {
-    var content = event.getContent();
-    var metadata = content[SDPStreamMetadataKey];
-    this.updateRemoteSDPStreamMetadata(metadata);
-  }
-  onAssertedIdentityReceived(event) {
-    var _this25 = this;
-    return _asyncToGenerator(function* () {
-      var content = event.getContent();
-      if (!content.asserted_identity) return;
-      _this25.remoteAssertedIdentity = {
-        id: content.asserted_identity.id,
-        displayName: content.asserted_identity.display_name
-      };
-      _this25.emit(CallEvent.AssertedIdentityChanged, _this25);
-    })();
-  }
-  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();
-    }
-  }
-  wrappedGotLocalOffer() {
-    var _this26 = this;
-    return _asyncToGenerator(function* () {
-      _this26.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.
-        yield _this26.gotLocalOffer();
-      } catch (e) {
-        _this26.getLocalOfferFailed(e);
-        return;
-      } finally {
-        _this26.makingOffer = false;
-      }
-    })();
-  }
-  gotLocalOffer() {
-    var _this27 = this;
-    return _asyncToGenerator(function* () {
-      logger.debug("Call ".concat(_this27.callId, " gotLocalOffer() running"));
-      if (_this27.callHasEnded()) {
-        logger.debug("Call ".concat(_this27.callId, " gotLocalOffer() ignoring newly created offer because the call has ended\""));
-        return;
-      }
-      var offer;
-      try {
-        _this27.getRidOfRTXCodecs();
-        offer = yield _this27.createOffer();
-      } catch (err) {
-        logger.debug("Call ".concat(_this27.callId, " gotLocalOffer() failed to create offer: "), err);
-        _this27.terminate(CallParty.Local, CallErrorCode.CreateOffer, true);
-        return;
-      }
-      try {
-        yield _this27.peerConn.setLocalDescription(offer);
-      } catch (err) {
-        logger.debug("Call ".concat(_this27.callId, " gotLocalOffer() error setting local description!"), err);
-        _this27.terminate(CallParty.Local, CallErrorCode.SetLocalDescription, true);
-        return;
-      }
-      if (_this27.peerConn.iceGatheringState === "gathering") {
-        // Allow a short time for initial candidates to be gathered
-        yield new Promise(resolve => {
-          setTimeout(resolve, 200);
-        });
-      }
-      if (_this27.callHasEnded()) return;
-      var eventType = _this27.state === CallState.CreateOffer ? EventType.CallInvite : EventType.CallNegotiate;
-      var content = {
-        lifetime: CALL_TIMEOUT_MS
-      };
-      if (eventType === EventType.CallInvite && _this27.invitee) {
-        content.invitee = _this27.invitee;
-      }
-
-      // clunky because TypeScript can't follow the types through if we use an expression as the key
-      if (_this27.state === CallState.CreateOffer) {
-        var _localDescription2;
-        content.offer = (_localDescription2 = _this27.peerConn.localDescription) === null || _localDescription2 === void 0 ? void 0 : _localDescription2.toJSON();
-      } else {
-        var _localDescription3;
-        content.description = (_localDescription3 = _this27.peerConn.localDescription) === null || _localDescription3 === void 0 ? void 0 : _localDescription3.toJSON();
-      }
-      content.capabilities = {
-        "m.call.transferee": _this27.client.supportsCallTransfer,
-        "m.call.dtmf": false
-      };
-      content[SDPStreamMetadataKey] = _this27.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.
-      var discardCount = _this27.discardDuplicateCandidates();
-      logger.info("Call ".concat(_this27.callId, " gotLocalOffer() discarding ").concat(discardCount, " candidates that will be sent in offer"));
-      try {
-        yield _this27.sendVoipEvent(eventType, content);
-      } catch (error) {
-        logger.error("Call ".concat(_this27.callId, " gotLocalOffer() failed to send invite"), error);
-        if (error instanceof MatrixError && error.event) _this27.client.cancelPendingEvent(error.event);
-        var code = CallErrorCode.SignallingFailed;
-        var message = "Signalling failed";
-        if (_this27.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";
-        }
-        _this27.emit(CallEvent.Error, new CallError(code, message, error), _this27);
-        _this27.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;
-      }
-      _this27.sendCandidateQueue();
-      if (_this27.state === CallState.CreateOffer) {
-        _this27.inviteOrAnswerSent = true;
-        _this27.state = CallState.InviteSent;
-        _this27.inviteTimeout = setTimeout(() => {
-          _this27.inviteTimeout = undefined;
-          if (_this27.state === CallState.InviteSent) {
-            _this27.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 before v113
-    if (!RTCRtpReceiver.getCapabilities || !RTCRtpSender.getCapabilities) return;
-    var screenshareVideoTransceiver = this.transceivers.get(getTransceiverKey(SDPStreamMetadataPurpose.Screenshare, "video"));
-
-    // setCodecPreferences isn't supported on FF (as of v113)
-    if (!screenshareVideoTransceiver || !screenshareVideoTransceiver.setCodecPreferences) return;
-    var recvCodecs = RTCRtpReceiver.getCapabilities("video").codecs;
-    var sendCodecs = RTCRtpSender.getCapabilities("video").codecs;
-    var codecs = [];
-    for (var codec of [...recvCodecs, ...sendCodecs]) {
-      if (codec.mimeType !== "video/rtx") {
-        codecs.push(codec);
-        try {
-          screenshareVideoTransceiver.setCodecPreferences(codecs);
-        } catch (e) {
-          // Specifically, Chrome around version 125 and Electron 30 (which is Chromium 124) return an H.264 codec in
-          // the sender's capabilities but throw when you try to set it. Hence... this mess.
-          // Specifically, that codec is:
-          // {
-          //   clockRate: 90000,
-          //   mimeType: "video/H264",
-          //   sdpFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640034",
-          // }
-          logger.info("Working around buggy WebRTC impl: claimed to support codec but threw when setting codec preferences", codec, e);
-          codecs.pop();
-        }
-      }
-    }
-  }
-  /**
-   * @internal
-   */
-  sendVoipEvent(eventType, content) {
-    var _this28 = this;
-    return _asyncToGenerator(function* () {
-      var realContent = _objectSpread(_objectSpread({}, content), {}, {
-        version: VOIP_PROTO_VERSION,
-        call_id: _this28.callId,
-        party_id: _this28.ourPartyId,
-        conf_id: _this28.groupCallId
-      });
-      if (_this28.opponentDeviceId) {
-        var _this28$getOpponentMe;
-        var toDeviceSeq = _this28.toDeviceSeq++;
-        var _content = _objectSpread(_objectSpread({}, realContent), {}, {
-          device_id: _this28.client.deviceId,
-          sender_session_id: _this28.client.getSessionId(),
-          dest_session_id: _this28.opponentSessionId,
-          seq: toDeviceSeq,
-          [ToDeviceMessageId]: uuidv4()
-        });
-        _this28.emit(CallEvent.SendVoipEvent, {
-          type: "toDevice",
-          eventType,
-          userId: _this28.invitee || ((_this28$getOpponentMe = _this28.getOpponentMember()) === null || _this28$getOpponentMe === void 0 ? void 0 : _this28$getOpponentMe.userId),
-          opponentDeviceId: _this28.opponentDeviceId,
-          content: _content
-        }, _this28);
-        var userId = _this28.invitee || _this28.getOpponentMember().userId;
-        if (_this28.client.getUseE2eForGroupCall()) {
-          if (!_this28.opponentDeviceInfo) {
-            logger.warn("Call ".concat(_this28.callId, " sendVoipEvent() failed: we do not have opponentDeviceInfo"));
-            return;
-          }
-          yield _this28.client.encryptAndSendToDevices([{
-            userId,
-            deviceInfo: _this28.opponentDeviceInfo
-          }], {
-            type: eventType,
-            content: _content
-          });
-        } else {
-          yield _this28.client.sendToDevice(eventType, new Map([[userId, new Map([[_this28.opponentDeviceId, _content]])]]));
-        }
-      } else {
-        var _this28$getOpponentMe2;
-        _this28.emit(CallEvent.SendVoipEvent, {
-          type: "sendEvent",
-          eventType,
-          roomId: _this28.roomId,
-          content: realContent,
-          userId: _this28.invitee || ((_this28$getOpponentMe2 = _this28.getOpponentMember()) === null || _this28$getOpponentMe2 === void 0 ? void 0 : _this28$getOpponentMe2.userId)
-        }, _this28);
-        yield _this28.client.sendEvent(_this28.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)
-    var 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() {
-    var discardCount = 0;
-    var newQueue = [];
-    for (var i = 0; i < this.candidateSendQueue.length; i++) {
-      var candidate = this.candidateSendQueue[i];
-      if (candidate.candidate === "") {
-        newQueue.push(candidate);
-      } else {
-        discardCount++;
-      }
-    }
-    this.candidateSendQueue = newQueue;
-    return discardCount;
-  }
-
-  /*
-   * Transfers this call to another user
-   */
-  transfer(targetUserId) {
-    var _this29 = this;
-    return _asyncToGenerator(function* () {
-      // Fetch the target user's global profile info: their room avatar / displayname
-      // could be different in whatever room we share with them.
-      var profileInfo = yield _this29.client.getProfileInfo(targetUserId);
-      var replacementId = genCallID();
-      var body = {
-        replacement_id: genCallID(),
-        target_user: {
-          id: targetUserId,
-          display_name: profileInfo.displayname,
-          avatar_url: profileInfo.avatar_url
-        },
-        create_call: replacementId
-      };
-      yield _this29.sendVoipEvent(EventType.CallReplaces, body);
-      yield _this29.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).
-   */
-  transferToCall(transferTargetCall) {
-    var _this30 = this;
-    return _asyncToGenerator(function* () {
-      var _transferTargetCall$g, _this30$getOpponentMe;
-      var targetUserId = (_transferTargetCall$g = transferTargetCall.getOpponentMember()) === null || _transferTargetCall$g === void 0 ? void 0 : _transferTargetCall$g.userId;
-      var targetProfileInfo = targetUserId ? yield _this30.client.getProfileInfo(targetUserId) : undefined;
-      var opponentUserId = (_this30$getOpponentMe = _this30.getOpponentMember()) === null || _this30$getOpponentMe === void 0 ? void 0 : _this30$getOpponentMe.userId;
-      var transfereeProfileInfo = opponentUserId ? yield _this30.client.getProfileInfo(opponentUserId) : undefined;
-      var newCallId = genCallID();
-      var bodyToTransferTarget = {
-        // the replacements on each side have their own ID, and it's distinct from the
-        // ID of the new call (but we can use the same function to generate it)
-        replacement_id: genCallID(),
-        target_user: {
-          id: opponentUserId,
-          display_name: transfereeProfileInfo === null || transfereeProfileInfo === void 0 ? void 0 : transfereeProfileInfo.displayname,
-          avatar_url: transfereeProfileInfo === null || transfereeProfileInfo === void 0 ? void 0 : transfereeProfileInfo.avatar_url
-        },
-        await_call: newCallId
-      };
-      yield transferTargetCall.sendVoipEvent(EventType.CallReplaces, bodyToTransferTarget);
-      var bodyToTransferee = {
-        replacement_id: genCallID(),
-        target_user: {
-          id: targetUserId,
-          display_name: targetProfileInfo === null || targetProfileInfo === void 0 ? void 0 : targetProfileInfo.displayname,
-          avatar_url: targetProfileInfo === null || targetProfileInfo === void 0 ? void 0 : targetProfileInfo.avatar_url
-        },
-        create_call: newCallId
-      };
-      yield _this30.sendVoipEvent(EventType.CallReplaces, bodyToTransferee);
-      yield _this30.terminate(CallParty.Local, CallErrorCode.Transferred, true);
-      yield transferTargetCall.terminate(CallParty.Local, CallErrorCode.Transferred, true);
-    })();
-  }
-  terminate(hangupParty, hangupReason, shouldEmit) {
-    var _this31 = this;
-    return _asyncToGenerator(function* () {
-      var _this31$stats;
-      if (_this31.callHasEnded()) return;
-      _this31.hangupParty = hangupParty;
-      _this31.hangupReason = hangupReason;
-      _this31.state = CallState.Ended;
-      if (_this31.inviteTimeout) {
-        clearTimeout(_this31.inviteTimeout);
-        _this31.inviteTimeout = undefined;
-      }
-      if (_this31.iceDisconnectedTimeout !== undefined) {
-        clearTimeout(_this31.iceDisconnectedTimeout);
-        _this31.iceDisconnectedTimeout = undefined;
-      }
-      if (_this31.callLengthInterval) {
-        clearInterval(_this31.callLengthInterval);
-        _this31.callLengthInterval = undefined;
-      }
-      if (_this31.stopVideoTrackTimer !== undefined) {
-        clearTimeout(_this31.stopVideoTrackTimer);
-        _this31.stopVideoTrackTimer = undefined;
-      }
-      for (var [stream, listener] of _this31.removeTrackListeners) {
-        stream.removeEventListener("removetrack", listener);
-      }
-      _this31.removeTrackListeners.clear();
-      _this31.callStatsAtEnd = yield _this31.collectCallStats();
-
-      // Order is important here: first we stopAllMedia() and only then we can deleteAllFeeds()
-      _this31.stopAllMedia();
-      _this31.deleteAllFeeds();
-      if (_this31.peerConn && _this31.peerConn.signalingState !== "closed") {
-        _this31.peerConn.close();
-      }
-      (_this31$stats = _this31.stats) === null || _this31$stats === void 0 || _this31$stats.removeStatsReportGatherer(_this31.callId);
-      if (shouldEmit) {
-        _this31.emit(CallEvent.Hangup, _this31);
-      }
-      _this31.client.callEventHandler.calls.delete(_this31.callId);
-    })();
-  }
-  stopAllMedia() {
-    logger.debug("Call ".concat(this.callId, " stopAllMedia() running"));
-    for (var 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 === SDPStreamMetadataPurpose.Usermedia) {
-        this.client.getMediaHandler().stopUserMediaStream(feed.stream);
-      } else if (feed.isLocal() && feed.purpose === SDPStreamMetadataPurpose.Screenshare) {
-        this.client.getMediaHandler().stopScreensharingStream(feed.stream);
-      } else if (!feed.isLocal()) {
-        logger.debug("Call ".concat(this.callId, " stopAllMedia() stopping stream (streamId=").concat(feed.stream.id, ")"));
-        for (var track of feed.stream.getTracks()) {
-          track.stop();
-        }
-      }
-    }
-  }
-  checkForErrorListener() {
-    if (this.listeners(EventEmitterEvents.Error).length === 0) {
-      throw new Error("You MUST attach an error listener using call.on('error', function() {})");
-    }
-  }
-  sendCandidateQueue() {
-    var _this32 = this;
-    return _asyncToGenerator(function* () {
-      if (_this32.candidateSendQueue.length === 0 || _this32.callHasEnded()) {
-        return;
-      }
-      var candidates = _this32.candidateSendQueue;
-      _this32.candidateSendQueue = [];
-      ++_this32.candidateSendTries;
-      var content = {
-        candidates: candidates.map(candidate => candidate.toJSON())
-      };
-      if (_this32.candidatesEnded) {
-        // If there are no more candidates, signal this by adding an empty string candidate
-        content.candidates.push({
-          candidate: ""
-        });
-      }
-      logger.debug("Call ".concat(_this32.callId, " sendCandidateQueue() attempting to send ").concat(candidates.length, " candidates"));
-      try {
-        yield _this32.sendVoipEvent(EventType.CallCandidates, content);
-        // reset our retry count if we have successfully sent our candidates
-        // otherwise queueCandidate() will refuse to try to flush the queue
-        _this32.candidateSendTries = 0;
-
-        // Try to send candidates again just in case we received more candidates while sending.
-        _this32.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 MatrixError && error.event) _this32.client.cancelPendingEvent(error.event);
-
-        // put all the candidates we failed to send back in the queue
-        _this32.candidateSendQueue.push(...candidates);
-        if (_this32.candidateSendTries > 5) {
-          logger.debug("Call ".concat(_this32.callId, " sendCandidateQueue() failed to send candidates on attempt ").concat(_this32.candidateSendTries, ". Giving up on this call."), error);
-          var code = CallErrorCode.SignallingFailed;
-          var message = "Signalling failed";
-          _this32.emit(CallEvent.Error, new CallError(code, message, error), _this32);
-          _this32.hangup(code, false);
-          return;
-        }
-        var delayMs = 500 * Math.pow(2, _this32.candidateSendTries);
-        ++_this32.candidateSendTries;
-        logger.debug("Call ".concat(_this32.callId, " sendCandidateQueue() failed to send candidates. Retrying in ").concat(delayMs, "ms"), error);
-        setTimeout(() => {
-          _this32.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.
-   */
-  placeCall(audio, video) {
-    var _this33 = this;
-    return _asyncToGenerator(function* () {
-      if (!audio) {
-        throw new Error("You CANNOT start a call without audio");
-      }
-      _this33.state = CallState.WaitLocalMedia;
-      var callFeed;
-      try {
-        var _this33$client$getDev;
-        var stream = yield _this33.client.getMediaHandler().getUserMediaStream(audio, video);
-
-        // make sure all the tracks are enabled (same as pushNewLocalFeed -
-        // we probably ought to just have one code path for adding streams)
-        setTracksEnabled(stream.getAudioTracks(), true);
-        setTracksEnabled(stream.getVideoTracks(), true);
-        callFeed = new CallFeed({
-          client: _this33.client,
-          roomId: _this33.roomId,
-          userId: _this33.client.getUserId(),
-          deviceId: (_this33$client$getDev = _this33.client.getDeviceId()) !== null && _this33$client$getDev !== void 0 ? _this33$client$getDev : undefined,
-          stream,
-          purpose: SDPStreamMetadataPurpose.Usermedia,
-          audioMuted: false,
-          videoMuted: false
-        });
-      } catch (e) {
-        _this33.getUserMediaFailed(e);
-        return;
-      }
-      try {
-        yield _this33.placeCallWithCallFeeds([callFeed]);
-      } catch (e) {
-        _this33.placeCallFailed(e);
-        return;
-      }
-    })();
-  }
-
-  /**
-   * Place a call to this room with call feed.
-   * @param callFeeds - to use
-   * @throws if you have not specified a listener for 'error' events.
-   * @throws if have passed audio=false.
-   */
-  placeCallWithCallFeeds(callFeeds) {
-    var _arguments2 = arguments,
-      _this34 = this;
-    return _asyncToGenerator(function* () {
-      var requestScreenshareFeed = _arguments2.length > 1 && _arguments2[1] !== undefined ? _arguments2[1] : false;
-      _this34.checkForErrorListener();
-      _this34.direction = CallDirection.Outbound;
-      yield _this34.initOpponentCrypto();
-
-      // XXX Find a better way to do this
-      _this34.client.callEventHandler.calls.set(_this34.callId, _this34);
-
-      // make sure we have valid turn creds. Unless something's gone wrong, it should
-      // poll and keep the credentials valid so this should be instant.
-      var haveTurnCreds = yield _this34.client.checkTurnServers();
-      if (!haveTurnCreds) {
-        logger.warn("Call ".concat(_this34.callId, " placeCallWithCallFeeds() failed to get TURN credentials! Proceeding with call anyway..."));
-      }
-
-      // create the peer connection now so it can be gathering candidates while we get user
-      // media (assuming a candidate pool size is configured)
-      _this34.peerConn = _this34.createPeerConnection();
-      _this34.emit(CallEvent.PeerConnectionCreated, _this34.peerConn, _this34);
-      _this34.gotCallFeedsForInvite(callFeeds, requestScreenshareFeed);
-    })();
-  }
-  createPeerConnection() {
-    var _this$stats;
-    var pc = new window.RTCPeerConnection({
-      iceTransportPolicy: this.forceTURN ? "relay" : undefined,
-      iceServers: this.turnServers.length ? this.turnServers : undefined,
-      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);
-    var opponentMember = this.getOpponentMember();
-    var opponentMemberId = opponentMember ? opponentMember.userId : "unknown";
-    (_this$stats = this.stats) === null || _this$stats === void 0 || _this$stats.addStatsReportGatherer(this.callId, opponentMemberId, pc);
-    return pc;
-  }
-  partyIdMatches(msg) {
-    // They must either match or both be absent (in which case opponentPartyId will be null)
-    // Also we ignore party IDs on the invite/offer if the version is 0, so we must do the same
-    // here and use null if the version is 0 (woe betide any opponent sending messages in the
-    // same call with different versions)
-    var msgPartyId = msg.version === 0 ? null : msg.party_id || null;
-    return msgPartyId === this.opponentPartyId;
-  }
-
-  // Commits to an opponent for the call
-  // ev: An invite or answer event
-  chooseOpponent(ev) {
-    var _getMember;
-    // I choo-choo-choose you
-    var msg = ev.getContent();
-    logger.debug("Call ".concat(this.callId, " chooseOpponent() running (partyId=").concat(msg.party_id, ")"));
-    this.opponentVersion = msg.version;
-    if (this.opponentVersion === 0) {
-      // set to null to indicate that we've chosen an opponent, but because
-      // they're v0 they have no party ID (even if they sent one, we're ignoring it)
-      this.opponentPartyId = null;
-    } else {
-      // set to their party ID, or if they're naughty and didn't send one despite
-      // not being v0, set it to null to indicate we picked an opponent with no
-      // party ID
-      this.opponentPartyId = msg.party_id || null;
-    }
-    this.opponentCaps = msg.capabilities || {};
-    this.opponentMember = (_getMember = this.client.getRoom(this.roomId).getMember(ev.getSender())) !== null && _getMember !== void 0 ? _getMember : undefined;
-    if (this.opponentMember) {
-      var _this$stats2;
-      (_this$stats2 = this.stats) === null || _this$stats2 === void 0 || _this$stats2.updateOpponentMember(this.callId, this.opponentMember.userId);
-    }
-  }
-  addBufferedIceCandidates() {
-    var _this35 = this;
-    return _asyncToGenerator(function* () {
-      var bufferedCandidates = _this35.remoteCandidateBuffer.get(_this35.opponentPartyId);
-      if (bufferedCandidates) {
-        logger.info("Call ".concat(_this35.callId, " addBufferedIceCandidates() adding ").concat(bufferedCandidates.length, " buffered candidates for opponent ").concat(_this35.opponentPartyId));
-        yield _this35.addIceCandidates(bufferedCandidates);
-      }
-      _this35.remoteCandidateBuffer.clear();
-    })();
-  }
-  addIceCandidates(candidates) {
-    var _this36 = this;
-    return _asyncToGenerator(function* () {
-      for (var candidate of candidates) {
-        if ((candidate.sdpMid === null || candidate.sdpMid === undefined) && (candidate.sdpMLineIndex === null || candidate.sdpMLineIndex === undefined)) {
-          logger.debug("Call ".concat(_this36.callId, " addIceCandidates() got remote ICE end-of-candidates"));
-        } else {
-          logger.debug("Call ".concat(_this36.callId, " addIceCandidates() got remote ICE candidate (sdpMid=").concat(candidate.sdpMid, ", candidate=").concat(candidate.candidate, ")"));
-        }
-        try {
-          yield _this36.peerConn.addIceCandidate(candidate);
-        } catch (err) {
-          if (!_this36.ignoreOffer) {
-            logger.info("Call ".concat(_this36.callId, " addIceCandidates() failed to add remote ICE candidate"), err);
-          } else {
-            logger.debug("Call ".concat(_this36.callId, " addIceCandidates() failed to add remote ICE candidate because ignoring offer"), err);
-          }
-        }
-      }
-    })();
-  }
-  get hasPeerConnection() {
-    return Boolean(this.peerConn);
-  }
-  initStats(stats) {
-    var peerId = arguments.length > 1 && arguments[1] !== undefined ? arguments[1] : "unknown";
-    this.stats = stats;
-    this.stats.start();
-  }
-}
-export function setTracksEnabled(tracks, enabled) {
-  for (var track of tracks) {
-    track.enabled = enabled;
-  }
-}
-export 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 {
-    var 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.error("WebRTC is not supported in this browser / environment");
-      }
-      return false;
-    }
-  } catch (e) {
-    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.
- */
-export function createNewMatrixCall(client, roomId, options) {
-  if (!supportsMatrixCall()) return null;
-  var optionsForceTURN = options ? options.forceTURN : false;
-  var 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
-  };
-  var call = new MatrixCall(opts);
-  client.reEmitter.reEmit(call, Object.values(CallEvent));
-  return call;
-}
-//# sourceMappingURL=call.js.map