@textrp/briij-js-sdk 41.0.0

package/lib/webrtc/call.js

@@ -0,0 +1,2596 @@
+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 createNewBriijCall} 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 { secureRandomString } from "../randomstring.js";
+import { SDPStreamMetadataPurpose, SDPStreamMetadataKey } from "./callEventTypes.js";
+import { CallFeed } from "./callFeed.js";
+import { EventEmitterEvents, TypedEventEmitter } from "../models/typed-event-emitter.js";
+import { GroupCallUnknownDeviceError } from "./groupCall.js";
+import { BriijError } 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"; // add more as needed
+  return CodecName;
+}(CodecName || {}); // 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";
+  // The value of isLocalOnHold() has changed
+  CallEvent["LocalHoldUnhold"] = "local_hold_unhold";
+  // The value of isRemoteOnHold() has changed
+  CallEvent["RemoteHoldUnhold"] = "remote_hold_unhold";
+  // backwards compat alias for LocalHoldUnhold: remove in a major version bump
+  CallEvent["HoldUnhold"] = "hold_unhold";
+  // Feeds have changed
+  CallEvent["FeedsChanged"] = "feeds_changed";
+  CallEvent["AssertedIdentityChanged"] = "asserted_identity_changed";
+  CallEvent["LengthChanged"] = "length_changed";
+  CallEvent["DataChannel"] = "datachannel";
+  CallEvent["SendVoipEvent"] = "send_voip_event";
+  // When the call instantiates its peer connection
+  // For apps that want to access the underlying peer connection, eg for debugging
+  CallEvent["PeerConnectionCreated"] = "peer_connection_created";
+  return CallEvent;
+}({});
+export var CallErrorCode = /*#__PURE__*/function (CallErrorCode) {
+  /** The user chose to end the call */
+  CallErrorCode["UserHangup"] = "user_hangup";
+  /** An error code when the local client failed to create an offer. */
+  CallErrorCode["LocalOfferFailed"] = "local_offer_failed";
+  /**
+   * An error code when there is no local mic/camera to use. This may be because
+   * the hardware isn't plugged in, or the user has explicitly denied access.
+   */
+  CallErrorCode["NoUserMedia"] = "no_user_media";
+  /**
+   * Error code used when a call event failed to send
+   * because unknown devices were present in the room
+   */
+  CallErrorCode["UnknownDevices"] = "unknown_devices";
+  /**
+   * Error code used when we fail to send the invite
+   * for some reason other than there being unknown devices
+   */
+  CallErrorCode["SendInvite"] = "send_invite";
+  /**
+   * An answer could not be created
+   */
+  CallErrorCode["CreateAnswer"] = "create_answer";
+  /**
+   * An offer could not be created
+   */
+  CallErrorCode["CreateOffer"] = "create_offer";
+  /**
+   * Error code used when we fail to send the answer
+   * for some reason other than there being unknown devices
+   */
+  CallErrorCode["SendAnswer"] = "send_answer";
+  /**
+   * The session description from the other side could not be set
+   */
+  CallErrorCode["SetRemoteDescription"] = "set_remote_description";
+  /**
+   * The session description from this side could not be set
+   */
+  CallErrorCode["SetLocalDescription"] = "set_local_description";
+  /**
+   * A different device answered the call
+   */
+  CallErrorCode["AnsweredElsewhere"] = "answered_elsewhere";
+  /**
+   * No media connection could be established to the other party
+   */
+  CallErrorCode["IceFailed"] = "ice_failed";
+  /**
+   * The invite timed out whilst waiting for an answer
+   */
+  CallErrorCode["InviteTimeout"] = "invite_timeout";
+  /**
+   * The call was replaced by another call
+   */
+  CallErrorCode["Replaced"] = "replaced";
+  /**
+   * Signalling for the call could not be sent (other than the initial invite)
+   */
+  CallErrorCode["SignallingFailed"] = "signalling_timeout";
+  /**
+   * The remote party is busy
+   */
+  CallErrorCode["UserBusy"] = "user_busy";
+  /**
+   * We transferred the call off to somewhere else
+   */
+  CallErrorCode["Transferred"] = "transferred";
+  /**
+   * A call from the same user was found with a new session id
+   */
+  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() + secureRandomString(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 BriijCall 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, "hasOpponentDeviceInfo", 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 BriijClient 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$peerConn0, _this$peerConn1;
+          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$peerConn9.iceConnectionState, ", conn=").concat((_this$peerConn0 = this.peerConn) === null || _this$peerConn0 === void 0 ? void 0 : _this$peerConn0.connectionState, ")"));
+          if ((_this$peerConn1 = this.peerConn) !== null && _this$peerConn1 !== void 0 && _this$peerConn1.restartIce) {
+            this.candidatesEnded = false;
+            this.peerConn.restartIce();
+          }
+          this.iceReconnectionTimeOut = undefined;
+        }, ICE_RECONNECTING_TIMEOUT);
+        this.iceDisconnectedTimeout = setTimeout(() => {
+          logger.info("Call ".concat(this.callId, " onIceConnectionStateChanged() hanging up call (ICE disconnected for too long)"));
+          this.hangup(CallErrorCode.IceFailed, false);
+        }, ICE_DISCONNECTED_TIMEOUT);
+        this.state = CallState.Connecting;
+      }
+
+      // In PTT mode, override feed status to muted when we lose connection to
+      // the peer, since we don't want to block the line if they're not saying anything.
+      // Experimenting in Chrome, this happens after 5 or 6 seconds, which is probably
+      // fast enough.
+      if (this.isPtt && ["failed", "disconnected"].includes(this.peerConn.iceConnectionState)) {
+        for (var feed of this.getRemoteFeeds()) {
+          feed.setAudioVideoMuted(true, true);
+        }
+      }
+    });
+    _defineProperty(this, "onSignallingStateChanged", () => {
+      var _this$peerConn10;
+      logger.debug("Call ".concat(this.callId, " onSignallingStateChanged() running (state=").concat((_this$peerConn10 = this.peerConn) === null || _this$peerConn10 === void 0 ? void 0 : _this$peerConn10.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$getOpponentMem;
+      if (!_this4.opponentDeviceId) return;
+      if (!_this4.client.getUseE2eForGroupCall()) return;
+      // It's possible to want E2EE and yet not have the means to manage E2EE
+      // ourselves (for example if the client is a RoomWidgetClient)
+      if (!_this4.client.getCrypto()) {
+        // All we know is the device ID
+        _this4.hasOpponentDeviceInfo = true;
+        return;
+      }
+      var userId = _this4.invitee || ((_this4$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");
+
+      // Here we were calling `BriijClient.crypto.deviceList.downloadKeys` which is not supported by the rust cryptography.
+      _this4.hasOpponentDeviceInfo = false;
+      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 BriijClient.
+   * @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 = SDPStreamMetadataKey.findIn(invite);
+      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 BriijClient.
+   * @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
+   * BriijClient.
+   * @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 _this0 = this;
+    return _asyncToGenerator(function* () {
+      // We don't do call downgrades
+      if (!audio && !video) return;
+      if (!_this0.opponentSupportsSDPStreamMetadata()) return;
+      try {
+        logger.debug("Call ".concat(_this0.callId, " upgradeCall() upgrading call (audio=").concat(audio, ", video=").concat(video, ")"));
+        var getAudio = audio || _this0.hasLocalUserMediaAudioTrack;
+        var getVideo = video || _this0.hasLocalUserMediaVideoTrack;
+
+        // updateLocalUsermediaStream() will take the tracks, use them as
+        // replacement and throw the stream away, so it isn't reusable
+        var stream = yield _this0.client.getMediaHandler().getUserMediaStream(getAudio, getVideo, false);
+        yield _this0.updateLocalUsermediaStream(stream, audio, video);
+      } catch (error) {
+        logger.error("Call ".concat(_this0.callId, " upgradeCall() failed to upgrade the call"), error);
+        _this0.emit(CallEvent.Error, new CallError(CallErrorCode.NoUserMedia, "Failed to get camera access: ", error), _this0);
+      }
+    })();
+  }
+
+  /**
+   * 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 _this1 = this;
+    return _asyncToGenerator(function* () {
+      // Skip if there is nothing to do
+      if (enabled && _this1.isScreensharing()) {
+        logger.warn("Call ".concat(_this1.callId, " setScreensharingEnabled() there is already a screensharing stream - there is nothing to do!"));
+        return true;
+      } else if (!enabled && !_this1.isScreensharing()) {
+        logger.warn("Call ".concat(_this1.callId, " setScreensharingEnabled() there already isn't a screensharing stream - there is nothing to do!"));
+        return false;
+      }
+
+      // Fallback to replaceTrack()
+      if (!_this1.opponentSupportsSDPStreamMetadata()) {
+        return _this1.setScreensharingEnabledWithoutMetadataSupport(enabled, opts);
+      }
+      logger.debug("Call ".concat(_this1.callId, " setScreensharingEnabled() running (enabled=").concat(enabled, ")"));
+      if (enabled) {
+        try {
+          var stream = yield _this1.client.getMediaHandler().getScreensharingStream(opts);
+          if (!stream) return false;
+          _this1.pushNewLocalFeed(stream, SDPStreamMetadataPurpose.Screenshare);
+          return true;
+        } catch (err) {
+          logger.error("Call ".concat(_this1.callId, " setScreensharingEnabled() failed to get screen-sharing stream:"), err);
+          return false;
+        }
+      } else {
+        var audioTransceiver = _this1.transceivers.get(getTransceiverKey(SDPStreamMetadataPurpose.Screenshare, "audio"));
+        var videoTransceiver = _this1.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) _this1.peerConn.removeTrack(transceiver.sender);
+        }
+        _this1.client.getMediaHandler().stopScreensharingStream(_this1.localScreensharingStream);
+        _this1.deleteFeedByStream(_this1.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 _this10 = this;
+    return _asyncToGenerator(function* () {
+      logger.debug("Call ".concat(_this10.callId, " setScreensharingEnabledWithoutMetadataSupport() running (enabled=").concat(enabled, ")"));
+      if (enabled) {
+        try {
+          var _this10$transceivers$;
+          var stream = yield _this10.client.getMediaHandler().getScreensharingStream(opts);
+          if (!stream) return false;
+          var track = stream.getTracks().find(track => track.kind === "video");
+          var sender = (_this10$transceivers$ = _this10.transceivers.get(getTransceiverKey(SDPStreamMetadataPurpose.Usermedia, "video"))) === null || _this10$transceivers$ === void 0 ? void 0 : _this10$transceivers$.sender;
+          sender === null || sender === void 0 || sender.replaceTrack(track !== null && track !== void 0 ? track : null);
+          _this10.pushNewLocalFeed(stream, SDPStreamMetadataPurpose.Screenshare, false);
+          return true;
+        } catch (err) {
+          logger.error("Call ".concat(_this10.callId, " setScreensharingEnabledWithoutMetadataSupport() failed to get screen-sharing stream:"), err);
+          return false;
+        }
+      } else {
+        var _this10$localUsermedi, _this10$transceivers$2;
+        var _track = (_this10$localUsermedi = _this10.localUsermediaStream) === null || _this10$localUsermedi === void 0 ? void 0 : _this10$localUsermedi.getTracks().find(track => track.kind === "video");
+        var _sender = (_this10$transceivers$2 = _this10.transceivers.get(getTransceiverKey(SDPStreamMetadataPurpose.Usermedia, "video"))) === null || _this10$transceivers$2 === void 0 ? void 0 : _this10$transceivers$2.sender;
+        _sender === null || _sender === void 0 || _sender.replaceTrack(_track !== null && _track !== void 0 ? _track : null);
+        _this10.client.getMediaHandler().stopScreensharingStream(_this10.localScreensharingStream);
+        _this10.deleteFeedByStream(_this10.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,
+      _this11 = 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 = _this11.localUsermediaFeed;
+      var audioEnabled = forceAudio || !callFeed.isAudioMuted() && !_this11.remoteOnHold;
+      var videoEnabled = forceVideo || !callFeed.isVideoMuted() && !_this11.remoteOnHold;
+      logger.log("Call ".concat(_this11.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 _this11.localUsermediaStream.getTracks()) {
+        _this11.localUsermediaStream.removeTrack(track);
+        track.stop();
+      }
+      for (var _track2 of stream.getTracks()) {
+        _this11.localUsermediaStream.addTrack(_track2);
+      }
+
+      // Then replace the old tracks, if possible.
+      var _loop2 = function* _loop2() {
+        var tKey = getTransceiverKey(SDPStreamMetadataPurpose.Usermedia, _track3.kind);
+        var transceiver = _this11.transceivers.get(tKey);
+        var oldSender = transceiver === null || transceiver === void 0 ? void 0 : transceiver.sender;
+        var added = false;
+        if (oldSender) {
+          try {
+            logger.info("Call ".concat(_this11.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(_this11.callId, " updateLocalUsermediaStream() replaceTrack failed: adding new transceiver instead"), error);
+          }
+        }
+        if (!added) {
+          logger.info("Call ".concat(_this11.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 = _this11.peerConn.addTrack(_track3, _this11.localUsermediaStream);
+          var newTransceiver = _this11.peerConn.getTransceivers().find(t => t.sender === newSender);
+          if (newTransceiver) {
+            _this11.transceivers.set(tKey, newTransceiver);
+          } else {
+            logger.warn("Call ".concat(_this11.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 _this12 = this;
+    return _asyncToGenerator(function* () {
+      var _this12$localUsermedi2;
+      logger.log("Call ".concat(_this12.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 && _this12.stopVideoTrackTimer !== undefined) {
+        clearTimeout(_this12.stopVideoTrackTimer);
+        _this12.stopVideoTrackTimer = undefined;
+      }
+      if (!(yield _this12.client.getMediaHandler().hasVideoDevice())) {
+        return _this12.isLocalVideoMuted();
+      }
+      if (!_this12.hasUserMediaVideoSender && !muted) {
+        var _this12$localUsermedi;
+        (_this12$localUsermedi = _this12.localUsermediaFeed) === null || _this12$localUsermedi === void 0 || _this12$localUsermedi.setAudioVideoMuted(null, muted);
+        yield _this12.upgradeCall(false, true);
+        return _this12.isLocalVideoMuted();
+      }
+
+      // we may not have a video track - if not, re-request usermedia
+      if (!muted && _this12.localUsermediaStream.getVideoTracks().length === 0) {
+        var stream = yield _this12.client.getMediaHandler().getUserMediaStream(true, true);
+        yield _this12.updateLocalUsermediaStream(stream);
+      }
+      (_this12$localUsermedi2 = _this12.localUsermediaFeed) === null || _this12$localUsermedi2 === void 0 || _this12$localUsermedi2.setAudioVideoMuted(null, muted);
+      _this12.updateMuteStatus();
+      yield _this12.sendMetadataUpdate();
+
+      // if we're muting video, set a timeout to stop & remove the video track so we release
+      // the camera. We wait a short time to do this because when we disable a track, WebRTC
+      // will send black video for it. If we just stop and remove it straight away, the video
+      // will just freeze which means that when we unmute video, the other side will briefly
+      // get a static frame of us from before we muted. This way, the still frame is just black.
+      // A very small delay is not always enough so the theory here is that it needs to be long
+      // enough for WebRTC to encode a frame: 120ms should be long enough even if we're only
+      // doing 10fps.
+      if (muted) {
+        _this12.stopVideoTrackTimer = setTimeout(() => {
+          for (var t of _this12.localUsermediaStream.getVideoTracks()) {
+            t.stop();
+            _this12.localUsermediaStream.removeTrack(t);
+          }
+        }, 120);
+      }
+      return _this12.isLocalVideoMuted();
+    })();
+  }
+
+  /**
+   * Check if local video is muted.
+   *
+   * If there are multiple video tracks, <i>all</i> of the tracks need to be muted
+   * for this to return true. This means if there are no video tracks, this will
+   * return true.
+   * @returns True if the local preview video is muted, else false
+   * (including if the call is not set up yet).
+   */
+  isLocalVideoMuted() {
+    var _this$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$localUsermediaF2 : false;
+  }
+
+  /**
+   * Set whether the microphone should be muted or not.
+   * @param muted - True to mute the mic.
+   * @returns the new mute state
+   */
+  setMicrophoneMuted(muted) {
+    var _this13 = this;
+    return _asyncToGenerator(function* () {
+      var _this13$localUsermedi;
+      logger.log("Call ".concat(_this13.callId, " setMicrophoneMuted() running ").concat(muted));
+      if (!(yield _this13.client.getMediaHandler().hasAudioDevice())) {
+        return _this13.isMicrophoneMuted();
+      }
+      if (!muted && (!_this13.hasUserMediaAudioSender || !_this13.hasLocalUserMediaAudioTrack)) {
+        yield _this13.upgradeCall(true, false);
+        return _this13.isMicrophoneMuted();
+      }
+      (_this13$localUsermedi = _this13.localUsermediaFeed) === null || _this13$localUsermedi === void 0 || _this13$localUsermedi.setAudioVideoMuted(muted, null);
+      _this13.updateMuteStatus();
+      yield _this13.sendMetadataUpdate();
+      return _this13.isMicrophoneMuted();
+    })();
+  }
+
+  /**
+   * Check if the microphone is muted.
+   *
+   * If there are multiple audio tracks, <i>all</i> of the tracks need to be muted
+   * for this to return true. This means if there are no audio tracks, this will
+   * return true.
+   * @returns True if the mic is muted, else false (including if the call
+   * is not set up yet).
+   */
+  isMicrophoneMuted() {
+    var _this$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 _this14 = this;
+    return _asyncToGenerator(function* () {
+      yield _this14.sendVoipEvent(EventType.CallSDPStreamMetadataChanged, {
+        [SDPStreamMetadataKey.name]: _this14.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 _this15 = this;
+    return _asyncToGenerator(function* () {
+      var answerContent = {
+        answer: {
+          sdp: _this15.peerConn.localDescription.sdp,
+          // type is now deprecated as of Matrix VoIP v1, but
+          // required to still be sent for backwards compat
+          type: _this15.peerConn.localDescription.type
+        },
+        [SDPStreamMetadataKey.name]: _this15.getLocalSDPStreamMetadata(true)
+      };
+      answerContent.capabilities = {
+        "m.call.transferee": _this15.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 = _this15.discardDuplicateCandidates();
+      logger.info("Call ".concat(_this15.callId, " sendAnswer() discarding ").concat(discardCount, " candidates that will be sent in answer"));
+      try {
+        yield _this15.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.
+        _this15.inviteOrAnswerSent = true;
+      } catch (error) {
+        // We've failed to answer: back to the ringing state
+        _this15.state = CallState.Ringing;
+        if (error instanceof BriijError && error.event) _this15.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";
+        }
+        _this15.emit(CallEvent.Error, new CallError(code, message, error), _this15);
+        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
+      _this15.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 _this16 = this;
+    return _asyncToGenerator(function* () {
+      var offer = yield _this16.peerConn.createOffer();
+      _this16.mungeSdp(offer, getCodecParamMods(_this16.isPtt));
+      return offer;
+    })();
+  }
+  createAnswer() {
+    var _this17 = this;
+    return _asyncToGenerator(function* () {
+      var answer = yield _this17.peerConn.createAnswer();
+      _this17.mungeSdp(answer, getCodecParamMods(_this17.isPtt));
+      return answer;
+    })();
+  }
+  gotCallFeedsForAnswer(callFeeds) {
+    var _this18 = this;
+    return _asyncToGenerator(function* () {
+      if (_this18.callHasEnded()) return;
+      _this18.waitForLocalAVStream = false;
+      for (var feed of callFeeds) {
+        _this18.pushLocalFeed(feed);
+      }
+      _this18.state = CallState.CreateAnswer;
+      var answer;
+      try {
+        _this18.getRidOfRTXCodecs();
+        answer = yield _this18.createAnswer();
+      } catch (err) {
+        logger.debug("Call ".concat(_this18.callId, " gotCallFeedsForAnswer() failed to create answer: "), err);
+        _this18.terminate(CallParty.Local, CallErrorCode.CreateAnswer, true);
+        return;
+      }
+      try {
+        yield _this18.peerConn.setLocalDescription(answer);
+
+        // make sure we're still going
+        if (_this18.callHasEnded()) return;
+        _this18.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 (_this18.callHasEnded()) return;
+        _this18.sendAnswer();
+      } catch (err) {
+        logger.debug("Call ".concat(_this18.callId, " gotCallFeedsForAnswer() error setting local description!"), err);
+        _this18.terminate(CallParty.Local, CallErrorCode.SetLocalDescription, true);
+        return;
+      }
+    })();
+  }
+  onRemoteIceCandidatesReceived(ev) {
+    var _this19 = this;
+    return _asyncToGenerator(function* () {
+      if (_this19.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(_this19.callId, " onRemoteIceCandidatesReceived() ignoring candidates event with no candidates!"));
+        return;
+      }
+      var fromPartyId = content.version === 0 ? null : content.party_id || null;
+      if (_this19.opponentPartyId === undefined) {
+        // we haven't picked an opponent yet so save the candidates
+        if (fromPartyId) {
+          logger.info("Call ".concat(_this19.callId, " onRemoteIceCandidatesReceived() buffering ").concat(candidates.length, " candidates until we pick an opponent"));
+          var bufferedCandidates = _this19.remoteCandidateBuffer.get(fromPartyId) || [];
+          bufferedCandidates.push(...candidates);
+          _this19.remoteCandidateBuffer.set(fromPartyId, bufferedCandidates);
+        }
+        return;
+      }
+      if (!_this19.partyIdMatches(content)) {
+        logger.info("Call ".concat(_this19.callId, " onRemoteIceCandidatesReceived() ignoring candidates from party ID ").concat(content.party_id, ": we have chosen party ID ").concat(_this19.opponentPartyId));
+        return;
+      }
+      yield _this19.addIceCandidates(candidates);
+    })();
+  }
+
+  /**
+   * Used by BriijClient.
+   */
+  onAnswerReceived(event) {
+    var _this20 = this;
+    return _asyncToGenerator(function* () {
+      var content = event.getContent();
+      logger.debug("Call ".concat(_this20.callId, " onAnswerReceived() running (hangupParty=").concat(content.party_id, ")"));
+      if (_this20.callHasEnded()) {
+        logger.debug("Call ".concat(_this20.callId, " onAnswerReceived() ignoring answer because call has ended"));
+        return;
+      }
+      if (_this20.opponentPartyId !== undefined) {
+        logger.info("Call ".concat(_this20.callId, " onAnswerReceived() ignoring answer from party ID ").concat(content.party_id, ": we already have an answer/reject from ").concat(_this20.opponentPartyId));
+        return;
+      }
+      _this20.chooseOpponent(event);
+      yield _this20.addBufferedIceCandidates();
+      _this20.state = CallState.Connecting;
+      var sdpStreamMetadata = SDPStreamMetadataKey.findIn(content);
+      if (sdpStreamMetadata) {
+        _this20.updateRemoteSDPStreamMetadata(sdpStreamMetadata);
+      } else {
+        logger.warn("Call ".concat(_this20.callId, " onAnswerReceived() did not get any SDPStreamMetadata! Can not send/receive multiple streams"));
+      }
+      try {
+        _this20.isSettingRemoteAnswerPending = true;
+        yield _this20.peerConn.setRemoteDescription(content.answer);
+        _this20.isSettingRemoteAnswerPending = false;
+        logger.debug("Call ".concat(_this20.callId, " onAnswerReceived() set remote description: ").concat(content.answer.type));
+      } catch (e) {
+        _this20.isSettingRemoteAnswerPending = false;
+        logger.debug("Call ".concat(_this20.callId, " onAnswerReceived() failed to set remote description"), e);
+        _this20.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 (_this20.opponentPartyId !== null) {
+        try {
+          yield _this20.sendVoipEvent(EventType.CallSelectAnswer, {
+            selected_party_id: _this20.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(_this20.callId, " onAnswerReceived() failed to send select_answer event"), err);
+        }
+      }
+    })();
+  }
+  onSelectAnswerReceived(event) {
+    var _this21 = this;
+    return _asyncToGenerator(function* () {
+      if (_this21.direction !== CallDirection.Inbound) {
+        logger.warn("Call ".concat(_this21.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(_this21.callId, " onSelectAnswerReceived() got nonsensical select_answer with null/undefined selected_party_id: ignoring"));
+        return;
+      }
+      if (selectedPartyId !== _this21.ourPartyId) {
+        logger.info("Call ".concat(_this21.callId, " onSelectAnswerReceived() got select_answer for party ID ").concat(selectedPartyId, ": we are party ID ").concat(_this21.ourPartyId, "."));
+        // The other party has picked somebody else's answer
+        yield _this21.terminate(CallParty.Remote, CallErrorCode.AnsweredElsewhere, true);
+      }
+    })();
+  }
+  onNegotiateReceived(event) {
+    var _this22 = this;
+    return _asyncToGenerator(function* () {
+      var content = event.getContent();
+      var description = content.description;
+      if (!description || !description.sdp || !description.type) {
+        logger.info("Call ".concat(_this22.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 = _this22.direction === CallDirection.Inbound;
+
+      // Here we follow the perfect negotiation logic from
+      // https://w3c.github.io/webrtc-pc/#perfect-negotiation-example
+      var readyForOffer = !_this22.makingOffer && (_this22.peerConn.signalingState === "stable" || _this22.isSettingRemoteAnswerPending);
+      var offerCollision = description.type === "offer" && !readyForOffer;
+      _this22.ignoreOffer = !polite && offerCollision;
+      if (_this22.ignoreOffer) {
+        logger.info("Call ".concat(_this22.callId, " onNegotiateReceived() ignoring colliding negotiate event because we're impolite"));
+        return;
+      }
+      var prevLocalOnHold = _this22.isLocalOnHold();
+      var sdpStreamMetadata = SDPStreamMetadataKey.findIn(content);
+      if (sdpStreamMetadata) {
+        _this22.updateRemoteSDPStreamMetadata(sdpStreamMetadata);
+      } else {
+        logger.warn("Call ".concat(_this22.callId, " onNegotiateReceived() received negotiation event without SDPStreamMetadata!"));
+      }
+      try {
+        _this22.isSettingRemoteAnswerPending = description.type == "answer";
+        yield _this22.peerConn.setRemoteDescription(description); // SRD rolls back as needed
+        _this22.isSettingRemoteAnswerPending = false;
+        logger.debug("Call ".concat(_this22.callId, " onNegotiateReceived() set remote description: ").concat(description.type));
+        if (description.type === "offer") {
+          var _localDescription;
+          var answer;
+          try {
+            _this22.getRidOfRTXCodecs();
+            answer = yield _this22.createAnswer();
+          } catch (err) {
+            logger.debug("Call ".concat(_this22.callId, " onNegotiateReceived() failed to create answer: "), err);
+            _this22.terminate(CallParty.Local, CallErrorCode.CreateAnswer, true);
+            return;
+          }
+          yield _this22.peerConn.setLocalDescription(answer);
+          logger.debug("Call ".concat(_this22.callId, " onNegotiateReceived() create an answer"));
+          _this22.sendVoipEvent(EventType.CallNegotiate, {
+            lifetime: CALL_TIMEOUT_MS,
+            description: (_localDescription = _this22.peerConn.localDescription) === null || _localDescription === void 0 ? void 0 : _localDescription.toJSON(),
+            [SDPStreamMetadataKey.name]: _this22.getLocalSDPStreamMetadata(true)
+          });
+        }
+      } catch (err) {
+        _this22.isSettingRemoteAnswerPending = false;
+        logger.warn("Call ".concat(_this22.callId, " onNegotiateReceived() failed to complete negotiation"), err);
+      }
+      var newLocalOnHold = _this22.isLocalOnHold();
+      if (prevLocalOnHold !== newLocalOnHold) {
+        _this22.emit(CallEvent.LocalHoldUnhold, newLocalOnHold, _this22);
+        // also this one for backwards compat
+        _this22.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 = SDPStreamMetadataKey.findIn(content);
+    if (metadata) {
+      this.updateRemoteSDPStreamMetadata(metadata);
+    }
+  }
+  onAssertedIdentityReceived(event) {
+    var _this23 = this;
+    return _asyncToGenerator(function* () {
+      var content = event.getContent();
+      if (!content.asserted_identity) return;
+      _this23.remoteAssertedIdentity = {
+        id: content.asserted_identity.id,
+        displayName: content.asserted_identity.display_name
+      };
+      _this23.emit(CallEvent.AssertedIdentityChanged, _this23);
+    })();
+  }
+  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 _this24 = this;
+    return _asyncToGenerator(function* () {
+      _this24.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 _this24.gotLocalOffer();
+      } catch (e) {
+        _this24.getLocalOfferFailed(e);
+        return;
+      } finally {
+        _this24.makingOffer = false;
+      }
+    })();
+  }
+  gotLocalOffer() {
+    var _this25 = this;
+    return _asyncToGenerator(function* () {
+      logger.debug("Call ".concat(_this25.callId, " gotLocalOffer() running"));
+      if (_this25.callHasEnded()) {
+        logger.debug("Call ".concat(_this25.callId, " gotLocalOffer() ignoring newly created offer because the call has ended\""));
+        return;
+      }
+      var offer;
+      try {
+        _this25.getRidOfRTXCodecs();
+        offer = yield _this25.createOffer();
+      } catch (err) {
+        logger.debug("Call ".concat(_this25.callId, " gotLocalOffer() failed to create offer: "), err);
+        _this25.terminate(CallParty.Local, CallErrorCode.CreateOffer, true);
+        return;
+      }
+      try {
+        yield _this25.peerConn.setLocalDescription(offer);
+      } catch (err) {
+        logger.debug("Call ".concat(_this25.callId, " gotLocalOffer() error setting local description!"), err);
+        _this25.terminate(CallParty.Local, CallErrorCode.SetLocalDescription, true);
+        return;
+      }
+      if (_this25.peerConn.iceGatheringState === "gathering") {
+        // Allow a short time for initial candidates to be gathered
+        yield new Promise(resolve => {
+          setTimeout(resolve, 200);
+        });
+      }
+      if (_this25.callHasEnded()) return;
+      var eventType = _this25.state === CallState.CreateOffer ? EventType.CallInvite : EventType.CallNegotiate;
+      var content = {
+        lifetime: CALL_TIMEOUT_MS
+      };
+      if (eventType === EventType.CallInvite && _this25.invitee) {
+        content.invitee = _this25.invitee;
+      }
+
+      // clunky because TypeScript can't follow the types through if we use an expression as the key
+      if (_this25.state === CallState.CreateOffer) {
+        var _localDescription2;
+        content.offer = (_localDescription2 = _this25.peerConn.localDescription) === null || _localDescription2 === void 0 ? void 0 : _localDescription2.toJSON();
+      } else {
+        var _localDescription3;
+        content.description = (_localDescription3 = _this25.peerConn.localDescription) === null || _localDescription3 === void 0 ? void 0 : _localDescription3.toJSON();
+      }
+      content.capabilities = {
+        "m.call.transferee": _this25.client.supportsCallTransfer,
+        "m.call.dtmf": false
+      };
+      content[SDPStreamMetadataKey.name] = _this25.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 = _this25.discardDuplicateCandidates();
+      logger.info("Call ".concat(_this25.callId, " gotLocalOffer() discarding ").concat(discardCount, " candidates that will be sent in offer"));
+      try {
+        yield _this25.sendVoipEvent(eventType, content);
+      } catch (error) {
+        logger.error("Call ".concat(_this25.callId, " gotLocalOffer() failed to send invite"), error);
+        if (error instanceof BriijError && error.event) _this25.client.cancelPendingEvent(error.event);
+        var code = CallErrorCode.SignallingFailed;
+        var message = "Signalling failed";
+        if (_this25.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";
+        }
+        _this25.emit(CallEvent.Error, new CallError(code, message, error), _this25);
+        _this25.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;
+      }
+      _this25.sendCandidateQueue();
+      if (_this25.state === CallState.CreateOffer) {
+        _this25.inviteOrAnswerSent = true;
+        _this25.state = CallState.InviteSent;
+        _this25.inviteTimeout = setTimeout(() => {
+          _this25.inviteTimeout = undefined;
+          if (_this25.state === CallState.InviteSent) {
+            _this25.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 _this26 = this;
+    return _asyncToGenerator(function* () {
+      var realContent = _objectSpread(_objectSpread({}, content), {}, {
+        version: VOIP_PROTO_VERSION,
+        call_id: _this26.callId,
+        party_id: _this26.ourPartyId,
+        conf_id: _this26.groupCallId
+      });
+      if (_this26.opponentDeviceId) {
+        var _this26$getOpponentMe;
+        var toDeviceSeq = _this26.toDeviceSeq++;
+        var _content = _objectSpread(_objectSpread({}, realContent), {}, {
+          device_id: _this26.client.deviceId,
+          sender_session_id: _this26.client.getSessionId(),
+          dest_session_id: _this26.opponentSessionId,
+          seq: toDeviceSeq,
+          [ToDeviceMessageId]: uuidv4()
+        });
+        _this26.emit(CallEvent.SendVoipEvent, {
+          type: "toDevice",
+          eventType,
+          userId: _this26.invitee || ((_this26$getOpponentMe = _this26.getOpponentMember()) === null || _this26$getOpponentMe === void 0 ? void 0 : _this26$getOpponentMe.userId),
+          opponentDeviceId: _this26.opponentDeviceId,
+          content: _content
+        }, _this26);
+        var userId = _this26.invitee || _this26.getOpponentMember().userId;
+        if (_this26.client.getUseE2eForGroupCall()) {
+          if (!_this26.hasOpponentDeviceInfo) {
+            logger.warn("Call ".concat(_this26.callId, " sendVoipEvent() failed: we do not have opponentDeviceInfo"));
+            return;
+          }
+
+          // TODO: Here we were sending the event to the opponent's device as a to-device message with BriijClient.encryptAndSendToDevice.
+          // However due to the switch to Rust cryptography we need to migrate to the new encryptToDeviceMessages API.
+          throw new Error("Unimplemented");
+        } else {
+          yield _this26.client.sendToDevice(eventType, new Map([[userId, new Map([[_this26.opponentDeviceId, _content]])]]));
+        }
+      } else {
+        var _this26$getOpponentMe2;
+        _this26.emit(CallEvent.SendVoipEvent, {
+          type: "sendEvent",
+          eventType,
+          roomId: _this26.roomId,
+          content: realContent,
+          userId: _this26.invitee || ((_this26$getOpponentMe2 = _this26.getOpponentMember()) === null || _this26$getOpponentMe2 === void 0 ? void 0 : _this26$getOpponentMe2.userId)
+        }, _this26);
+        yield _this26.client.sendEvent(_this26.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 _this27 = 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 _this27.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 _this27.sendVoipEvent(EventType.CallReplaces, body);
+      yield _this27.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 _this28 = this;
+    return _asyncToGenerator(function* () {
+      var _transferTargetCall$g, _this28$getOpponentMe;
+      var targetUserId = (_transferTargetCall$g = transferTargetCall.getOpponentMember()) === null || _transferTargetCall$g === void 0 ? void 0 : _transferTargetCall$g.userId;
+      var targetProfileInfo = targetUserId ? yield _this28.client.getProfileInfo(targetUserId) : undefined;
+      var opponentUserId = (_this28$getOpponentMe = _this28.getOpponentMember()) === null || _this28$getOpponentMe === void 0 ? void 0 : _this28$getOpponentMe.userId;
+      var transfereeProfileInfo = opponentUserId ? yield _this28.client.getProfileInfo(opponentUserId) : undefined;
+      var newCallId = genCallID();
+      var bodyToTransferTarget = {
+        // the replacements on each side have their own ID, and it's distinct from the
+        // ID of the new call (but we can use the same function to generate it)
+        replacement_id: genCallID(),
+        target_user: {
+          id: opponentUserId,
+          display_name: transfereeProfileInfo === null || transfereeProfileInfo === void 0 ? void 0 : transfereeProfileInfo.displayname,
+          avatar_url: transfereeProfileInfo === null || transfereeProfileInfo === void 0 ? void 0 : transfereeProfileInfo.avatar_url
+        },
+        await_call: newCallId
+      };
+      yield transferTargetCall.sendVoipEvent(EventType.CallReplaces, bodyToTransferTarget);
+      var bodyToTransferee = {
+        replacement_id: genCallID(),
+        target_user: {
+          id: targetUserId,
+          display_name: targetProfileInfo === null || targetProfileInfo === void 0 ? void 0 : targetProfileInfo.displayname,
+          avatar_url: targetProfileInfo === null || targetProfileInfo === void 0 ? void 0 : targetProfileInfo.avatar_url
+        },
+        create_call: newCallId
+      };
+      yield _this28.sendVoipEvent(EventType.CallReplaces, bodyToTransferee);
+      yield _this28.terminate(CallParty.Local, CallErrorCode.Transferred, true);
+      yield transferTargetCall.terminate(CallParty.Local, CallErrorCode.Transferred, true);
+    })();
+  }
+  terminate(hangupParty, hangupReason, shouldEmit) {
+    var _this29 = this;
+    return _asyncToGenerator(function* () {
+      var _this29$stats;
+      if (_this29.callHasEnded()) return;
+      _this29.hangupParty = hangupParty;
+      _this29.hangupReason = hangupReason;
+      _this29.state = CallState.Ended;
+      if (_this29.inviteTimeout) {
+        clearTimeout(_this29.inviteTimeout);
+        _this29.inviteTimeout = undefined;
+      }
+      if (_this29.iceDisconnectedTimeout !== undefined) {
+        clearTimeout(_this29.iceDisconnectedTimeout);
+        _this29.iceDisconnectedTimeout = undefined;
+      }
+      if (_this29.callLengthInterval) {
+        clearInterval(_this29.callLengthInterval);
+        _this29.callLengthInterval = undefined;
+      }
+      if (_this29.stopVideoTrackTimer !== undefined) {
+        clearTimeout(_this29.stopVideoTrackTimer);
+        _this29.stopVideoTrackTimer = undefined;
+      }
+      for (var [stream, listener] of _this29.removeTrackListeners) {
+        stream.removeEventListener("removetrack", listener);
+      }
+      _this29.removeTrackListeners.clear();
+      _this29.callStatsAtEnd = yield _this29.collectCallStats();
+
+      // Order is important here: first we stopAllMedia() and only then we can deleteAllFeeds()
+      _this29.stopAllMedia();
+      _this29.deleteAllFeeds();
+      if (_this29.peerConn && _this29.peerConn.signalingState !== "closed") {
+        _this29.peerConn.close();
+      }
+      (_this29$stats = _this29.stats) === null || _this29$stats === void 0 || _this29$stats.removeStatsReportGatherer(_this29.callId);
+      if (shouldEmit) {
+        _this29.emit(CallEvent.Hangup, _this29);
+      }
+      _this29.client.callEventHandler.calls.delete(_this29.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 _this30 = this;
+    return _asyncToGenerator(function* () {
+      if (_this30.candidateSendQueue.length === 0 || _this30.callHasEnded()) {
+        return;
+      }
+      var candidates = _this30.candidateSendQueue;
+      _this30.candidateSendQueue = [];
+      ++_this30.candidateSendTries;
+      var content = {
+        candidates: candidates.map(candidate => candidate.toJSON())
+      };
+      if (_this30.candidatesEnded) {
+        // If there are no more candidates, signal this by adding an empty string candidate
+        content.candidates.push({
+          candidate: ""
+        });
+      }
+      logger.debug("Call ".concat(_this30.callId, " sendCandidateQueue() attempting to send ").concat(candidates.length, " candidates"));
+      try {
+        yield _this30.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
+        _this30.candidateSendTries = 0;
+
+        // Try to send candidates again just in case we received more candidates while sending.
+        _this30.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 BriijError && error.event) _this30.client.cancelPendingEvent(error.event);
+
+        // put all the candidates we failed to send back in the queue
+        _this30.candidateSendQueue.push(...candidates);
+        if (_this30.candidateSendTries > 5) {
+          logger.debug("Call ".concat(_this30.callId, " sendCandidateQueue() failed to send candidates on attempt ").concat(_this30.candidateSendTries, ". Giving up on this call."), error);
+          var code = CallErrorCode.SignallingFailed;
+          var message = "Signalling failed";
+          _this30.emit(CallEvent.Error, new CallError(code, message, error), _this30);
+          _this30.hangup(code, false);
+          return;
+        }
+        var delayMs = 500 * Math.pow(2, _this30.candidateSendTries);
+        ++_this30.candidateSendTries;
+        logger.debug("Call ".concat(_this30.callId, " sendCandidateQueue() failed to send candidates. Retrying in ").concat(delayMs, "ms"), error);
+        setTimeout(() => {
+          _this30.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 _this31 = this;
+    return _asyncToGenerator(function* () {
+      if (!audio) {
+        throw new Error("You CANNOT start a call without audio");
+      }
+      _this31.state = CallState.WaitLocalMedia;
+      var callFeed;
+      try {
+        var _this31$client$getDev;
+        var stream = yield _this31.client.getMediaHandler().getUserMediaStream(audio, video);
+
+        // make sure all the tracks are enabled (same as pushNewLocalFeed -
+        // we probably ought to just have one code path for adding streams)
+        setTracksEnabled(stream.getAudioTracks(), true);
+        setTracksEnabled(stream.getVideoTracks(), true);
+        callFeed = new CallFeed({
+          client: _this31.client,
+          roomId: _this31.roomId,
+          userId: _this31.client.getUserId(),
+          deviceId: (_this31$client$getDev = _this31.client.getDeviceId()) !== null && _this31$client$getDev !== void 0 ? _this31$client$getDev : undefined,
+          stream,
+          purpose: SDPStreamMetadataPurpose.Usermedia,
+          audioMuted: false,
+          videoMuted: false
+        });
+      } catch (e) {
+        _this31.getUserMediaFailed(e);
+        return;
+      }
+      try {
+        yield _this31.placeCallWithCallFeeds([callFeed]);
+      } catch (e) {
+        _this31.placeCallFailed(e);
+        return;
+      }
+    })();
+  }
+
+  /**
+   * Place a call to this room with call feed.
+   * @param callFeeds - to use
+   * @throws if you have not specified a listener for 'error' events.
+   * @throws if have passed audio=false.
+   */
+  placeCallWithCallFeeds(callFeeds) {
+    var _arguments2 = arguments,
+      _this32 = this;
+    return _asyncToGenerator(function* () {
+      var requestScreenshareFeed = _arguments2.length > 1 && _arguments2[1] !== undefined ? _arguments2[1] : false;
+      _this32.checkForErrorListener();
+      _this32.direction = CallDirection.Outbound;
+      yield _this32.initOpponentCrypto();
+
+      // XXX Find a better way to do this
+      _this32.client.callEventHandler.calls.set(_this32.callId, _this32);
+
+      // make sure we have valid turn creds. Unless something's gone wrong, it should
+      // poll and keep the credentials valid so this should be instant.
+      var haveTurnCreds = yield _this32.client.checkTurnServers();
+      if (!haveTurnCreds) {
+        logger.warn("Call ".concat(_this32.callId, " placeCallWithCallFeeds() failed to get TURN credentials! Proceeding with call anyway..."));
+      }
+
+      // create the peer connection now so it can be gathering candidates while we get user
+      // media (assuming a candidate pool size is configured)
+      _this32.peerConn = _this32.createPeerConnection();
+      _this32.emit(CallEvent.PeerConnectionCreated, _this32.peerConn, _this32);
+      _this32.gotCallFeedsForInvite(callFeeds, requestScreenshareFeed);
+    })();
+  }
+  createPeerConnection() {
+    var _this$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 _this33 = this;
+    return _asyncToGenerator(function* () {
+      var bufferedCandidates = _this33.remoteCandidateBuffer.get(_this33.opponentPartyId);
+      if (bufferedCandidates) {
+        logger.info("Call ".concat(_this33.callId, " addBufferedIceCandidates() adding ").concat(bufferedCandidates.length, " buffered candidates for opponent ").concat(_this33.opponentPartyId));
+        yield _this33.addIceCandidates(bufferedCandidates);
+      }
+      _this33.remoteCandidateBuffer.clear();
+    })();
+  }
+  addIceCandidates(candidates) {
+    var _this34 = 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(_this34.callId, " addIceCandidates() got remote ICE end-of-candidates"));
+        } else {
+          logger.debug("Call ".concat(_this34.callId, " addIceCandidates() got remote ICE candidate (sdpMid=").concat(candidate.sdpMid, ", candidate=").concat(candidate.candidate, ")"));
+        }
+        try {
+          yield _this34.peerConn.addIceCandidate(candidate);
+        } catch (err) {
+          if (!_this34.ignoreOffer) {
+            logger.info("Call ".concat(_this34.callId, " addIceCandidates() failed to add remote ICE candidate"), err);
+          } else {
+            logger.debug("Call ".concat(_this34.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 supportsBriijCall() {
+  // 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 _ref2, _ref3, _window$RTCPeerConnec;
+    var supported = Boolean((_ref2 = (_ref3 = (_window$RTCPeerConnec = window.RTCPeerConnection) !== null && _window$RTCPeerConnec !== void 0 ? _window$RTCPeerConnec : window.RTCSessionDescription) !== null && _ref3 !== void 0 ? _ref3 : window.RTCIceCandidate) !== null && _ref2 !== void 0 ? _ref2 : 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 createNewBriijCall(client, roomId, options) {
+  if (!supportsBriijCall()) 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 BriijCall(opts);
+  client.reEmitter.reEmit(call, Object.values(CallEvent));
+  return call;
+}
+//# sourceMappingURL=call.js.map