@unwanted/matrix-sdk-mini 34.12.0-1 → 34.12.0-3

package/src/webrtc/call.ts

@@ -1,3074 +0,0 @@
-/*
-Copyright 2015, 2016 OpenMarket Ltd
-Copyright 2017 New Vector Ltd
-Copyright 2019, 2020 The Matrix.org Foundation C.I.C.
-Copyright 2021 - 2022 Šimon Brandner <simon.bra.ag@gmail.com>
-
-Licensed under the Apache License, Version 2.0 (the "License");
-you may not use this file except in compliance with the License.
-You may obtain a copy of the License at
-
-    http://www.apache.org/licenses/LICENSE-2.0
-
-Unless required by applicable law or agreed to in writing, software
-distributed under the License is distributed on an "AS IS" BASIS,
-WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-See the License for the specific language governing permissions and
-limitations under the License.
-*/
-
-/**
- * This is an internal module. See {@link createNewMatrixCall} for the public API.
- */
-
-import { v4 as uuidv4 } from "uuid";
-import { parse as parseSdp, write as writeSdp } from "sdp-transform";
-
-import { logger } from "../logger.ts";
-import { checkObjectHasKeys, isNullOrUndefined, recursivelyAssign } from "../utils.ts";
-import { MatrixEvent } from "../models/event.ts";
-import { EventType, TimelineEvents, ToDeviceMessageId } from "../@types/event.ts";
-import { RoomMember } from "../models/room-member.ts";
-import { randomString } from "../randomstring.ts";
-import {
-    MCallReplacesEvent,
-    MCallAnswer,
-    MCallInviteNegotiate,
-    CallCapabilities,
-    SDPStreamMetadataPurpose,
-    SDPStreamMetadata,
-    SDPStreamMetadataKey,
-    MCallSDPStreamMetadataChanged,
-    MCallSelectAnswer,
-    MCAllAssertedIdentity,
-    MCallCandidates,
-    MCallBase,
-    MCallHangupReject,
-} from "./callEventTypes.ts";
-import { CallFeed } from "./callFeed.ts";
-import { MatrixClient } from "../client.ts";
-import { EventEmitterEvents, TypedEventEmitter } from "../models/typed-event-emitter.ts";
-import { DeviceInfo } from "../crypto/deviceinfo.ts";
-import { GroupCallUnknownDeviceError } from "./groupCall.ts";
-import { IScreensharingOpts } from "./mediaHandler.ts";
-import { MatrixError } from "../http-api/index.ts";
-import { GroupCallStats } from "./stats/groupCallStats.ts";
-
-interface CallOpts {
-    // The room ID for this call.
-    roomId: string;
-    invitee?: string;
-    // The Matrix Client instance to send events to.
-    client: MatrixClient;
-    /**
-     * Whether relay through TURN should be forced.
-     * @deprecated use opts.forceTURN when creating the matrix client
-     * since it's only possible to set this option on outbound calls.
-     */
-    forceTURN?: boolean;
-    // A list of TURN servers.
-    turnServers?: Array<TurnServer>;
-    opponentDeviceId?: string;
-    opponentSessionId?: string;
-    groupCallId?: string;
-}
-
-interface TurnServer {
-    urls: Array<string>;
-    username?: string;
-    password?: string;
-    ttl?: number;
-}
-
-interface AssertedIdentity {
-    id: string;
-    displayName: string;
-}
-
-enum MediaType {
-    AUDIO = "audio",
-    VIDEO = "video",
-}
-
-enum CodecName {
-    OPUS = "opus",
-    // add more as needed
-}
-
-// Used internally to specify modifications to codec parameters in SDP
-interface CodecParamsMod {
-    mediaType: MediaType;
-    codec: CodecName;
-    enableDtx?: boolean; // true to enable discontinuous transmission, false to disable, undefined to leave as-is
-    maxAverageBitrate?: number; // sets the max average bitrate, or undefined to leave as-is
-}
-
-export enum CallState {
-    Fledgling = "fledgling",
-    InviteSent = "invite_sent",
-    WaitLocalMedia = "wait_local_media",
-    CreateOffer = "create_offer",
-    CreateAnswer = "create_answer",
-    Connecting = "connecting",
-    Connected = "connected",
-    Ringing = "ringing",
-    Ended = "ended",
-}
-
-export enum CallType {
-    Voice = "voice",
-    Video = "video",
-}
-
-export enum CallDirection {
-    Inbound = "inbound",
-    Outbound = "outbound",
-}
-
-export enum CallParty {
-    Local = "local",
-    Remote = "remote",
-}
-
-export enum CallEvent {
-    Hangup = "hangup",
-    State = "state",
-    Error = "error",
-    Replaced = "replaced",
-
-    // The value of isLocalOnHold() has changed
-    LocalHoldUnhold = "local_hold_unhold",
-    // The value of isRemoteOnHold() has changed
-    RemoteHoldUnhold = "remote_hold_unhold",
-    // backwards compat alias for LocalHoldUnhold: remove in a major version bump
-    HoldUnhold = "hold_unhold",
-    // Feeds have changed
-    FeedsChanged = "feeds_changed",
-
-    AssertedIdentityChanged = "asserted_identity_changed",
-
-    LengthChanged = "length_changed",
-
-    DataChannel = "datachannel",
-
-    SendVoipEvent = "send_voip_event",
-
-    // When the call instantiates its peer connection
-    // For apps that want to access the underlying peer connection, eg for debugging
-    PeerConnectionCreated = "peer_connection_created",
-}
-
-export enum CallErrorCode {
-    /** The user chose to end the call */
-    UserHangup = "user_hangup",
-
-    /** An error code when the local client failed to create an offer. */
-    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.
-     */
-    NoUserMedia = "no_user_media",
-
-    /**
-     * Error code used when a call event failed to send
-     * because unknown devices were present in the room
-     */
-    UnknownDevices = "unknown_devices",
-
-    /**
-     * Error code used when we fail to send the invite
-     * for some reason other than there being unknown devices
-     */
-    SendInvite = "send_invite",
-
-    /**
-     * An answer could not be created
-     */
-    CreateAnswer = "create_answer",
-
-    /**
-     * An offer could not be created
-     */
-    CreateOffer = "create_offer",
-
-    /**
-     * Error code used when we fail to send the answer
-     * for some reason other than there being unknown devices
-     */
-    SendAnswer = "send_answer",
-
-    /**
-     * The session description from the other side could not be set
-     */
-    SetRemoteDescription = "set_remote_description",
-
-    /**
-     * The session description from this side could not be set
-     */
-    SetLocalDescription = "set_local_description",
-
-    /**
-     * A different device answered the call
-     */
-    AnsweredElsewhere = "answered_elsewhere",
-
-    /**
-     * No media connection could be established to the other party
-     */
-    IceFailed = "ice_failed",
-
-    /**
-     * The invite timed out whilst waiting for an answer
-     */
-    InviteTimeout = "invite_timeout",
-
-    /**
-     * The call was replaced by another call
-     */
-    Replaced = "replaced",
-
-    /**
-     * Signalling for the call could not be sent (other than the initial invite)
-     */
-    SignallingFailed = "signalling_timeout",
-
-    /**
-     * The remote party is busy
-     */
-    UserBusy = "user_busy",
-
-    /**
-     * We transferred the call off to somewhere else
-     */
-    Transferred = "transferred",
-
-    /**
-     * A call from the same user was found with a new session id
-     */
-    NewSession = "new_session",
-}
-
-/**
- * The version field that we set in m.call.* events
- */
-const VOIP_PROTO_VERSION = "1";
-
-/** The fallback ICE server to use for STUN or TURN protocols. */
-export const FALLBACK_ICE_SERVER = "stun:turn.matrix.org";
-
-/** The length of time a call can be ringing for. */
-const CALL_TIMEOUT_MS = 60 * 1000; // ms
-/** The time after which we increment callLength */
-const CALL_LENGTH_INTERVAL = 1000; // ms
-/** The time after which we end the call, if ICE got disconnected */
-const ICE_DISCONNECTED_TIMEOUT = 30 * 1000; // ms
-/** The time after which we try a ICE restart, if ICE got disconnected */
-const ICE_RECONNECTING_TIMEOUT = 2 * 1000; // ms
-export class CallError extends Error {
-    public readonly code: string;
-
-    public constructor(code: CallErrorCode, msg: string, err: Error) {
-        // Still don't think there's any way to have proper nested errors
-        super(msg + ": " + err);
-
-        this.code = code;
-    }
-}
-
-export function genCallID(): string {
-    return Date.now().toString() + randomString(16);
-}
-
-function getCodecParamMods(isPtt: boolean): CodecParamsMod[] {
-    const mods = [
-        {
-            mediaType: "audio",
-            codec: "opus",
-            enableDtx: true,
-            maxAverageBitrate: isPtt ? 12000 : undefined,
-        },
-    ] as CodecParamsMod[];
-
-    return mods;
-}
-
-type CallEventType =
-    | EventType.CallReplaces
-    | EventType.CallAnswer
-    | EventType.CallSelectAnswer
-    | EventType.CallNegotiate
-    | EventType.CallInvite
-    | EventType.CallCandidates
-    | EventType.CallHangup
-    | EventType.CallReject
-    | EventType.CallSDPStreamMetadataChangedPrefix;
-
-export interface VoipEvent {
-    type: "toDevice" | "sendEvent";
-    eventType: string;
-    userId?: string;
-    opponentDeviceId?: string;
-    roomId?: string;
-    content: TimelineEvents[CallEventType];
-}
-
-/**
- * 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.
- */
-export type CallEventHandlerMap = {
-    [CallEvent.DataChannel]: (channel: RTCDataChannel, call: MatrixCall) => void;
-    [CallEvent.FeedsChanged]: (feeds: CallFeed[], call: MatrixCall) => void;
-    [CallEvent.Replaced]: (newCall: MatrixCall, oldCall: MatrixCall) => void;
-    [CallEvent.Error]: (error: CallError, call: MatrixCall) => void;
-    [CallEvent.RemoteHoldUnhold]: (onHold: boolean, call: MatrixCall) => void;
-    [CallEvent.LocalHoldUnhold]: (onHold: boolean, call: MatrixCall) => void;
-    [CallEvent.LengthChanged]: (length: number, call: MatrixCall) => void;
-    [CallEvent.State]: (state: CallState, oldState: CallState, call: MatrixCall) => void;
-    [CallEvent.Hangup]: (call: MatrixCall) => void;
-    [CallEvent.AssertedIdentityChanged]: (call: MatrixCall) => void;
-    /* @deprecated */
-    [CallEvent.HoldUnhold]: (onHold: boolean) => void;
-    [CallEvent.SendVoipEvent]: (event: VoipEvent, call: MatrixCall) => void;
-    [CallEvent.PeerConnectionCreated]: (peerConn: RTCPeerConnection, call: MatrixCall) => void;
-};
-
-// The key of the transceiver map (purpose + media type, separated by ':')
-type TransceiverKey = string;
-
-// 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: SDPStreamMetadataPurpose, kind: TransceiverKey): string {
-    return purpose + ":" + kind;
-}
-
-export class MatrixCall extends TypedEventEmitter<CallEvent, CallEventHandlerMap> {
-    public roomId: string;
-    public callId: string;
-    public invitee?: string;
-    public hangupParty?: CallParty;
-    public hangupReason?: string;
-    public direction?: CallDirection;
-    public ourPartyId: string;
-    public peerConn?: RTCPeerConnection;
-    public toDeviceSeq = 0;
-
-    // whether this call should have push-to-talk semantics
-    // This should be set by the consumer on incoming & outgoing calls.
-    public isPtt = false;
-
-    private _state = CallState.Fledgling;
-    private readonly client: MatrixClient;
-    private readonly forceTURN?: boolean;
-    private readonly turnServers: Array<TurnServer>;
-    // A queue for candidates waiting to go out.
-    // We try to amalgamate candidates into a single candidate message where
-    // possible
-    private candidateSendQueue: Array<RTCIceCandidate> = [];
-    private candidateSendTries = 0;
-    private candidatesEnded = false;
-    private feeds: Array<CallFeed> = [];
-
-    // our transceivers for each purpose and type of media
-    private transceivers = new Map<TransceiverKey, RTCRtpTransceiver>();
-
-    private inviteOrAnswerSent = false;
-    private waitForLocalAVStream = false;
-    private successor?: MatrixCall;
-    private opponentMember?: RoomMember;
-    private opponentVersion?: number | string;
-    // 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.
-    private opponentPartyId: string | null | undefined;
-    private opponentCaps?: CallCapabilities;
-    private iceDisconnectedTimeout?: ReturnType<typeof setTimeout>;
-    private iceReconnectionTimeOut?: ReturnType<typeof setTimeout> | undefined;
-    private inviteTimeout?: ReturnType<typeof setTimeout>;
-    private readonly removeTrackListeners = new Map<MediaStream, () => void>();
-
-    // 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
-    private 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' :(
-    private callStatsAtEnd?: any[];
-
-    // Perfect negotiation state: https://www.w3.org/TR/webrtc/#perfect-negotiation-example
-    private makingOffer = false;
-    private ignoreOffer = false;
-    private isSettingRemoteAnswerPending = false;
-
-    private responsePromiseChain?: Promise<void>;
-
-    // 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
-    private remoteCandidateBuffer = new Map<string, MCallCandidates["candidates"]>();
-
-    private remoteAssertedIdentity?: AssertedIdentity;
-    private remoteSDPStreamMetadata?: SDPStreamMetadata;
-
-    private callLengthInterval?: ReturnType<typeof setInterval>;
-    private callStartTime?: number;
-
-    private opponentDeviceId?: string;
-    private opponentDeviceInfo?: DeviceInfo;
-    private opponentSessionId?: string;
-    public groupCallId?: string;
-
-    // Used to keep the timer for the delay before actually stopping our
-    // video track after muting (see setLocalVideoMuted)
-    private stopVideoTrackTimer?: ReturnType<typeof setTimeout>;
-    // Used to allow connection without Video and Audio. To establish a webrtc connection without media a Data channel is
-    // needed At the moment this property is true if we allow MatrixClient with isVoipWithNoMediaAllowed = true
-    private readonly isOnlyDataChannelAllowed: boolean;
-    private stats: GroupCallStats | undefined;
-
-    /**
-     * Construct a new Matrix Call.
-     * @param opts - Config options.
-     */
-    public constructor(opts: CallOpts) {
-        super();
-
-        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 ?? false;
-        this.ourPartyId = this.client.deviceId;
-        this.opponentDeviceId = opts.opponentDeviceId;
-        this.opponentSessionId = opts.opponentSessionId;
-        this.groupCallId = opts.groupCallId;
-        // Array of Objects with urls, username, credential keys
-        this.turnServers = opts.turnServers || [];
-        if (this.turnServers.length === 0 && this.client.isFallbackICEServerAllowed()) {
-            this.turnServers.push({
-                urls: [FALLBACK_ICE_SERVER],
-            });
-        }
-        for (const server of this.turnServers) {
-            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.
-     */
-    public async placeVoiceCall(): Promise<void> {
-        await this.placeCall(true, false);
-    }
-
-    /**
-     * Place a video call to this room.
-     * @throws If you have not specified a listener for 'error' events.
-     */
-    public async placeVideoCall(): Promise<void> {
-        await this.placeCall(true, true);
-    }
-
-    /**
-     * Create a datachannel using this call's peer connection.
-     * @param label - A human readable label for this datachannel
-     * @param options - An object providing configuration options for the data channel.
-     */
-    public createDataChannel(label: string, options: RTCDataChannelInit | undefined): RTCDataChannel {
-        const dataChannel = this.peerConn!.createDataChannel(label, options);
-        this.emit(CallEvent.DataChannel, dataChannel, this);
-        return dataChannel;
-    }
-
-    public getOpponentMember(): RoomMember | undefined {
-        return this.opponentMember;
-    }
-
-    public getOpponentDeviceId(): string | undefined {
-        return this.opponentDeviceId;
-    }
-
-    public getOpponentSessionId(): string | undefined {
-        return this.opponentSessionId;
-    }
-
-    public opponentCanBeTransferred(): boolean {
-        return Boolean(this.opponentCaps && this.opponentCaps["m.call.transferee"]);
-    }
-
-    public opponentSupportsDTMF(): boolean {
-        return Boolean(this.opponentCaps && this.opponentCaps["m.call.dtmf"]);
-    }
-
-    public getRemoteAssertedIdentity(): AssertedIdentity | undefined {
-        return this.remoteAssertedIdentity;
-    }
-
-    public get state(): CallState {
-        return this._state;
-    }
-
-    private set state(state: CallState) {
-        const oldState = this._state;
-        this._state = state;
-        this.emit(CallEvent.State, state, oldState, this);
-    }
-
-    public get type(): CallType {
-        // 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;
-    }
-
-    public get hasLocalUserMediaVideoTrack(): boolean {
-        return !!this.localUsermediaStream?.getVideoTracks().length;
-    }
-
-    public get hasRemoteUserMediaVideoTrack(): boolean {
-        return this.getRemoteFeeds().some((feed) => {
-            return feed.purpose === SDPStreamMetadataPurpose.Usermedia && feed.stream?.getVideoTracks().length;
-        });
-    }
-
-    public get hasLocalUserMediaAudioTrack(): boolean {
-        return !!this.localUsermediaStream?.getAudioTracks().length;
-    }
-
-    public get hasRemoteUserMediaAudioTrack(): boolean {
-        return this.getRemoteFeeds().some((feed) => {
-            return feed.purpose === SDPStreamMetadataPurpose.Usermedia && !!feed.stream?.getAudioTracks().length;
-        });
-    }
-
-    private get hasUserMediaAudioSender(): boolean {
-        return Boolean(this.transceivers.get(getTransceiverKey(SDPStreamMetadataPurpose.Usermedia, "audio"))?.sender);
-    }
-
-    private get hasUserMediaVideoSender(): boolean {
-        return Boolean(this.transceivers.get(getTransceiverKey(SDPStreamMetadataPurpose.Usermedia, "video"))?.sender);
-    }
-
-    public get localUsermediaFeed(): CallFeed | undefined {
-        return this.getLocalFeeds().find((feed) => feed.purpose === SDPStreamMetadataPurpose.Usermedia);
-    }
-
-    public get localScreensharingFeed(): CallFeed | undefined {
-        return this.getLocalFeeds().find((feed) => feed.purpose === SDPStreamMetadataPurpose.Screenshare);
-    }
-
-    public get localUsermediaStream(): MediaStream | undefined {
-        return this.localUsermediaFeed?.stream;
-    }
-
-    public get localScreensharingStream(): MediaStream | undefined {
-        return this.localScreensharingFeed?.stream;
-    }
-
-    public get remoteUsermediaFeed(): CallFeed | undefined {
-        return this.getRemoteFeeds().find((feed) => feed.purpose === SDPStreamMetadataPurpose.Usermedia);
-    }
-
-    public get remoteScreensharingFeed(): CallFeed | undefined {
-        return this.getRemoteFeeds().find((feed) => feed.purpose === SDPStreamMetadataPurpose.Screenshare);
-    }
-
-    public get remoteUsermediaStream(): MediaStream | undefined {
-        return this.remoteUsermediaFeed?.stream;
-    }
-
-    public get remoteScreensharingStream(): MediaStream | undefined {
-        return this.remoteScreensharingFeed?.stream;
-    }
-
-    private getFeedByStreamId(streamId: string): CallFeed | undefined {
-        return this.getFeeds().find((feed) => feed.stream.id === streamId);
-    }
-
-    /**
-     * Returns an array of all CallFeeds
-     * @returns CallFeeds
-     */
-    public getFeeds(): Array<CallFeed> {
-        return this.feeds;
-    }
-
-    /**
-     * Returns an array of all local CallFeeds
-     * @returns local CallFeeds
-     */
-    public getLocalFeeds(): Array<CallFeed> {
-        return this.feeds.filter((feed) => feed.isLocal());
-    }
-
-    /**
-     * Returns an array of all remote CallFeeds
-     * @returns remote CallFeeds
-     */
-    public getRemoteFeeds(): Array<CallFeed> {
-        return this.feeds.filter((feed) => !feed.isLocal());
-    }
-
-    private async initOpponentCrypto(): Promise<void> {
-        if (!this.opponentDeviceId) return;
-        if (!this.client.getUseE2eForGroupCall()) return;
-        // It's possible to want E2EE and yet not have the means to manage E2EE
-        // ourselves (for example if the client is a RoomWidgetClient)
-        if (!this.client.isCryptoEnabled()) {
-            // All we know is the device ID
-            this.opponentDeviceInfo = new DeviceInfo(this.opponentDeviceId);
-            return;
-        }
-        // if we've got to this point, we do want to init crypto, so throw if we can't
-        if (!this.client.crypto) throw new Error("Crypto is not initialised.");
-
-        const userId = this.invitee || this.getOpponentMember()?.userId;
-
-        if (!userId) throw new Error("Couldn't find opponent user ID to init crypto");
-
-        const deviceInfoMap = await this.client.crypto.deviceList.downloadKeys([userId], false);
-        this.opponentDeviceInfo = deviceInfoMap.get(userId)?.get(this.opponentDeviceId);
-        if (this.opponentDeviceInfo === undefined) {
-            throw new GroupCallUnknownDeviceError(userId);
-        }
-    }
-
-    /**
-     * Generates and returns localSDPStreamMetadata
-     * @returns localSDPStreamMetadata
-     */
-    private getLocalSDPStreamMetadata(updateStreamIds = false): SDPStreamMetadata {
-        const metadata: SDPStreamMetadata = {};
-        for (const localFeed of this.getLocalFeeds()) {
-            if (updateStreamIds) {
-                localFeed.sdpMetadataStreamId = localFeed.stream.id;
-            }
-
-            metadata[localFeed.sdpMetadataStreamId] = {
-                purpose: localFeed.purpose,
-                audio_muted: localFeed.isAudioMuted(),
-                video_muted: localFeed.isVideoMuted(),
-            };
-        }
-        return metadata;
-    }
-
-    /**
-     * Returns true if there are no incoming feeds,
-     * otherwise returns false
-     * @returns no incoming feeds
-     */
-    public noIncomingFeeds(): boolean {
-        return !this.feeds.some((feed) => !feed.isLocal());
-    }
-
-    private pushRemoteFeed(stream: MediaStream): void {
-        // Fallback to old behavior if the other side doesn't support SDPStreamMetadata
-        if (!this.opponentSupportsSDPStreamMetadata()) {
-            this.pushRemoteFeedWithoutMetadata(stream);
-            return;
-        }
-
-        const userId = this.getOpponentMember()!.userId;
-        const purpose = this.remoteSDPStreamMetadata![stream.id].purpose;
-        const audioMuted = this.remoteSDPStreamMetadata![stream.id].audio_muted;
-        const videoMuted = this.remoteSDPStreamMetadata![stream.id].video_muted;
-
-        if (!purpose) {
-            logger.warn(
-                `Call ${this.callId} pushRemoteFeed() ignoring stream because we didn't get any metadata about it (streamId=${stream.id})`,
-            );
-            return;
-        }
-
-        if (this.getFeedByStreamId(stream.id)) {
-            logger.warn(
-                `Call ${this.callId} pushRemoteFeed() ignoring stream because we already have a feed for it (streamId=${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 ${this.callId} pushRemoteFeed() pushed stream (streamId=${stream.id}, active=${stream.active}, purpose=${purpose})`,
-        );
-    }
-
-    /**
-     * This method is used ONLY if the other client doesn't support sending SDPStreamMetadata
-     */
-    private pushRemoteFeedWithoutMetadata(stream: MediaStream): void {
-        const userId = this.getOpponentMember()!.userId;
-        // We can guess the purpose here since the other client can only send one stream
-        const purpose = SDPStreamMetadataPurpose.Usermedia;
-        const oldRemoteStream = this.feeds.find((feed) => !feed.isLocal())?.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 ${this.callId} pushRemoteFeedWithoutMetadata() ignoring new stream because we already have stream (streamId=${stream.id})`,
-            );
-            return;
-        }
-
-        if (this.getFeedByStreamId(stream.id)) {
-            logger.warn(
-                `Call ${this.callId} pushRemoteFeedWithoutMetadata() ignoring stream because we already have a feed for it (streamId=${stream.id})`,
-            );
-            return;
-        }
-
-        this.feeds.push(
-            new CallFeed({
-                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 ${this.callId} pushRemoteFeedWithoutMetadata() pushed stream (streamId=${stream.id}, active=${stream.active})`,
-        );
-    }
-
-    private pushNewLocalFeed(stream: MediaStream, purpose: SDPStreamMetadataPurpose, addToPeerConnection = true): void {
-        const userId = this.client.getUserId()!;
-
-        // Tracks don't always start off enabled, eg. chrome will give a disabled
-        // audio track if you ask for user media audio and already had one that
-        // you'd set to disabled (presumably because it clones them internally).
-        setTracksEnabled(stream.getAudioTracks(), true);
-        setTracksEnabled(stream.getVideoTracks(), true);
-
-        if (this.getFeedByStreamId(stream.id)) {
-            logger.warn(
-                `Call ${this.callId} pushNewLocalFeed() ignoring stream because we already have a feed for it (streamId=${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
-     */
-    public pushLocalFeed(callFeed: CallFeed, addToPeerConnection = true): void {
-        if (this.feeds.some((feed) => callFeed.stream.id === feed.stream.id)) {
-            logger.info(
-                `Call ${this.callId} pushLocalFeed() ignoring duplicate local stream (streamId=${callFeed.stream.id})`,
-            );
-            return;
-        }
-
-        this.feeds.push(callFeed);
-
-        if (addToPeerConnection) {
-            for (const track of callFeed.stream.getTracks()) {
-                logger.info(
-                    `Call ${this.callId} pushLocalFeed() adding track to peer connection (id=${track.id}, kind=${track.kind}, streamId=${callFeed.stream.id}, streamPurpose=${callFeed.purpose}, enabled=${track.enabled})`,
-                );
-
-                const tKey = getTransceiverKey(callFeed.purpose, track.kind);
-                if (this.transceivers.has(tKey)) {
-                    // we already have a sender, so we re-use it. We try to re-use transceivers as much
-                    // as possible because they can't be removed once added, so otherwise they just
-                    // accumulate which makes the SDP very large very quickly: in fact it only takes
-                    // about 6 video tracks to exceed the maximum size of an Olm-encrypted
-                    // Matrix event.
-                    const transceiver = this.transceivers.get(tKey)!;
-
-                    transceiver.sender.replaceTrack(track);
-                    // set the direction to indicate we're going to start sending again
-                    // (this will trigger the re-negotiation)
-                    transceiver.direction = transceiver.direction === "inactive" ? "sendonly" : "sendrecv";
-                } else {
-                    // create a new one. We need to use addTrack rather addTransceiver for this because firefox
-                    // doesn't yet implement RTCRTPSender.setStreams()
-                    // (https://bugzilla.mozilla.org/show_bug.cgi?id=1510802) so we'd have no way to group the
-                    // two tracks together into a stream.
-                    const newSender = this.peerConn!.addTrack(track, callFeed.stream);
-
-                    // now go & fish for the new transceiver
-                    const newTransceiver = this.peerConn!.getTransceivers().find((t) => t.sender === newSender);
-                    if (newTransceiver) {
-                        this.transceivers.set(tKey, newTransceiver);
-                    } else {
-                        logger.warn(
-                            `Call ${this.callId} pushLocalFeed() didn't find a matching transceiver after adding track!`,
-                        );
-                    }
-                }
-            }
-        }
-
-        logger.info(
-            `Call ${this.callId} pushLocalFeed() pushed stream (id=${callFeed.stream.id}, active=${callFeed.stream.active}, purpose=${callFeed.purpose})`,
-        );
-
-        this.emit(CallEvent.FeedsChanged, this.feeds, this);
-    }
-
-    /**
-     * Removes local call feed from the call and its tracks from the peer
-     * connection
-     * @param callFeed - to remove
-     */
-    public removeLocalFeed(callFeed: CallFeed): void {
-        const audioTransceiverKey = getTransceiverKey(callFeed.purpose, "audio");
-        const videoTransceiverKey = getTransceiverKey(callFeed.purpose, "video");
-
-        for (const transceiverKey of [audioTransceiverKey, videoTransceiverKey]) {
-            // this is slightly mixing the track and transceiver API but is basically just shorthand.
-            // There is no way to actually remove a transceiver, so this just sets it to inactive
-            // (or recvonly) and replaces the source with nothing.
-            if (this.transceivers.has(transceiverKey)) {
-                const transceiver = this.transceivers.get(transceiverKey)!;
-                if (transceiver.sender) this.peerConn!.removeTrack(transceiver.sender);
-            }
-        }
-
-        if (callFeed.purpose === SDPStreamMetadataPurpose.Screenshare) {
-            this.client.getMediaHandler().stopScreensharingStream(callFeed.stream);
-        }
-
-        this.deleteFeed(callFeed);
-    }
-
-    private deleteAllFeeds(): void {
-        for (const feed of this.feeds) {
-            if (!feed.isLocal() || !this.groupCallId) {
-                feed.dispose();
-            }
-        }
-
-        this.feeds = [];
-        this.emit(CallEvent.FeedsChanged, this.feeds, this);
-    }
-
-    private deleteFeedByStream(stream: MediaStream): void {
-        const feed = this.getFeedByStreamId(stream.id);
-        if (!feed) {
-            logger.warn(
-                `Call ${this.callId} deleteFeedByStream() didn't find the feed to delete (streamId=${stream.id})`,
-            );
-            return;
-        }
-        this.deleteFeed(feed);
-    }
-
-    private deleteFeed(feed: CallFeed): void {
-        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' :(
-    public async getCurrentCallStats(): Promise<any[] | undefined> {
-        if (this.callHasEnded()) {
-            return this.callStatsAtEnd;
-        }
-
-        return this.collectCallStats();
-    }
-
-    private async collectCallStats(): Promise<any[] | undefined> {
-        // This happens when the call fails before it starts.
-        // For example when we fail to get capture sources
-        if (!this.peerConn) return;
-
-        const statsReport = await this.peerConn.getStats();
-        const stats: any[] = [];
-        statsReport.forEach((item) => {
-            stats.push(item);
-        });
-
-        return stats;
-    }
-
-    /**
-     * Configure this call from an invite event. Used by MatrixClient.
-     * @param event - The m.call.invite event
-     */
-    public async initWithInvite(event: MatrixEvent): Promise<void> {
-        const invite = event.getContent<MCallInviteNegotiate>();
-        this.direction = CallDirection.Inbound;
-
-        // make sure we have valid turn creds. Unless something's gone wrong, it should
-        // poll and keep the credentials valid so this should be instant.
-        const haveTurnCreds = await this.client.checkTurnServers();
-        if (!haveTurnCreds) {
-            logger.warn(
-                `Call ${this.callId} initWithInvite() failed to get TURN credentials! Proceeding with call anyway...`,
-            );
-        }
-
-        const sdpStreamMetadata = invite[SDPStreamMetadataKey];
-        if (sdpStreamMetadata) {
-            this.updateRemoteSDPStreamMetadata(sdpStreamMetadata);
-        } else {
-            logger.debug(
-                `Call ${this.callId} initWithInvite() did not get any SDPStreamMetadata! Can not send/receive multiple streams`,
-            );
-        }
-
-        this.peerConn = this.createPeerConnection();
-        this.emit(CallEvent.PeerConnectionCreated, this.peerConn, this);
-        // we must set the party ID before await-ing on anything: the call event
-        // handler will start giving us more call events (eg. candidates) so if
-        // we haven't set the party ID, we'll ignore them.
-        this.chooseOpponent(event);
-        await this.initOpponentCrypto();
-        try {
-            await this.peerConn.setRemoteDescription(invite.offer);
-            logger.debug(`Call ${this.callId} initWithInvite() set remote description: ${invite.offer.type}`);
-            await this.addBufferedIceCandidates();
-        } catch (e) {
-            logger.debug(`Call ${this.callId} initWithInvite() failed to set remote description`, e);
-            this.terminate(CallParty.Local, CallErrorCode.SetRemoteDescription, false);
-            return;
-        }
-
-        const remoteStream = this.feeds.find((feed) => !feed.isLocal())?.stream;
-
-        // According to previous comments in this file, firefox at some point did not
-        // add streams until media started arriving on them. Testing latest firefox
-        // (81 at time of writing), this is no longer a problem, so let's do it the correct way.
-        //
-        // For example in case of no media webrtc connections like screen share only call we have to allow webrtc
-        // connections without remote media. In this case we always use a data channel. At the moment we allow as well
-        // only data channel as media in the WebRTC connection with this setup here.
-        if (!this.isOnlyDataChannelAllowed && (!remoteStream || remoteStream.getTracks().length === 0)) {
-            logger.error(
-                `Call ${this.callId} initWithInvite() no remote stream or no tracks after setting remote description!`,
-            );
-            this.terminate(CallParty.Local, CallErrorCode.SetRemoteDescription, false);
-            return;
-        }
-
-        this.state = CallState.Ringing;
-
-        if (event.getLocalAge()) {
-            // Time out the call if it's ringing for too long
-            const ringingTimer = setTimeout(() => {
-                if (this.state == CallState.Ringing) {
-                    logger.debug(`Call ${this.callId} initWithInvite() invite has expired. Hanging up.`);
-                    this.hangupParty = CallParty.Remote; // effectively
-                    this.state = CallState.Ended;
-                    this.stopAllMedia();
-                    if (this.peerConn!.signalingState != "closed") {
-                        this.peerConn!.close();
-                    }
-                    this.stats?.removeStatsReportGatherer(this.callId);
-                    this.emit(CallEvent.Hangup, this);
-                }
-            }, invite.lifetime - event.getLocalAge());
-
-            const onState = (state: CallState): void => {
-                if (state !== CallState.Ringing) {
-                    clearTimeout(ringingTimer);
-                    this.off(CallEvent.State, onState);
-                }
-            };
-            this.on(CallEvent.State, onState);
-        }
-    }
-
-    /**
-     * Configure this call from a hangup or reject event. Used by MatrixClient.
-     * @param event - The m.call.hangup event
-     */
-    public initWithHangup(event: MatrixEvent): void {
-        // 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;
-    }
-
-    private shouldAnswerWithMediaType(
-        wantedValue: boolean | undefined,
-        valueOfTheOtherSide: boolean,
-        type: "audio" | "video",
-    ): boolean {
-        if (wantedValue && !valueOfTheOtherSide) {
-            // TODO: Figure out how to do this
-            logger.warn(
-                `Call ${this.callId} shouldAnswerWithMediaType() unable to answer with ${type} because the other side isn't sending it either.`,
-            );
-            return false;
-        } else if (
-            !isNullOrUndefined(wantedValue) &&
-            wantedValue !== valueOfTheOtherSide &&
-            !this.opponentSupportsSDPStreamMetadata()
-        ) {
-            logger.warn(
-                `Call ${this.callId} shouldAnswerWithMediaType() unable to answer with ${type}=${wantedValue} because the other side doesn't support it. Answering with ${type}=${valueOfTheOtherSide}.`,
-            );
-            return valueOfTheOtherSide!;
-        }
-        return wantedValue ?? valueOfTheOtherSide!;
-    }
-
-    /**
-     * Answer a call.
-     */
-    public async answer(audio?: boolean, video?: boolean): Promise<void> {
-        if (this.inviteOrAnswerSent) return;
-        // TODO: Figure out how to do this
-        if (audio === false && video === false) throw new Error("You CANNOT answer a call without media");
-
-        if (!this.localUsermediaStream && !this.waitForLocalAVStream) {
-            const prevState = this.state;
-            const answerWithAudio = this.shouldAnswerWithMediaType(audio, this.hasRemoteUserMediaAudioTrack, "audio");
-            const answerWithVideo = this.shouldAnswerWithMediaType(video, this.hasRemoteUserMediaVideoTrack, "video");
-
-            this.state = CallState.WaitLocalMedia;
-            this.waitForLocalAVStream = true;
-
-            try {
-                const stream = await this.client.getMediaHandler().getUserMediaStream(answerWithAudio, answerWithVideo);
-                this.waitForLocalAVStream = false;
-                const usermediaFeed = new CallFeed({
-                    client: this.client,
-                    roomId: this.roomId,
-                    userId: this.client.getUserId()!,
-                    deviceId: this.client.getDeviceId() ?? undefined,
-                    stream,
-                    purpose: SDPStreamMetadataPurpose.Usermedia,
-                    audioMuted: false,
-                    videoMuted: false,
-                });
-
-                const feeds = [usermediaFeed];
-
-                if (this.localScreensharingFeed) {
-                    feeds.push(this.localScreensharingFeed);
-                }
-
-                this.answerWithCallFeeds(feeds);
-            } catch (e) {
-                if (answerWithVideo) {
-                    // Try to answer without video
-                    logger.warn(
-                        `Call ${this.callId} answer() failed to getUserMedia(), trying to getUserMedia() without video`,
-                    );
-                    this.state = prevState;
-                    this.waitForLocalAVStream = false;
-                    await this.answer(answerWithAudio, false);
-                } else {
-                    this.getUserMediaFailed(<Error>e);
-                    return;
-                }
-            }
-        } else if (this.waitForLocalAVStream) {
-            this.state = CallState.WaitLocalMedia;
-        }
-    }
-
-    public answerWithCallFeeds(callFeeds: CallFeed[]): void {
-        if (this.inviteOrAnswerSent) return;
-
-        this.queueGotCallFeedsForAnswer(callFeeds);
-    }
-
-    /**
-     * Replace this call with a new call, e.g. for glare resolution. Used by
-     * MatrixClient.
-     * @param newCall - The new call.
-     */
-    public replacedBy(newCall: MatrixCall): void {
-        logger.debug(`Call ${this.callId} replacedBy() running (newCallId=${newCall.callId})`);
-        if (this.state === CallState.WaitLocalMedia) {
-            logger.debug(
-                `Call ${this.callId} replacedBy() telling new call to wait for local media (newCallId=${newCall.callId})`,
-            );
-            newCall.waitForLocalAVStream = true;
-        } else if ([CallState.CreateOffer, CallState.InviteSent].includes(this.state)) {
-            if (newCall.direction === CallDirection.Outbound) {
-                newCall.queueGotCallFeedsForAnswer([]);
-            } else {
-                logger.debug(
-                    `Call ${this.callId} replacedBy() handing local stream to new call(newCallId=${newCall.callId})`,
-                );
-                newCall.queueGotCallFeedsForAnswer(this.getLocalFeeds().map((feed) => feed.clone()));
-            }
-        }
-        this.successor = newCall;
-        this.emit(CallEvent.Replaced, newCall, this);
-        this.hangup(CallErrorCode.Replaced, true);
-    }
-
-    /**
-     * Hangup a call.
-     * @param reason - The reason why the call is being hung up.
-     * @param suppressEvent - True to suppress emitting an event.
-     */
-    public hangup(reason: CallErrorCode, suppressEvent: boolean): void {
-        if (this.callHasEnded()) return;
-
-        logger.debug(`Call ${this.callId} hangup() ending call (reason=${reason})`);
-        this.terminate(CallParty.Local, reason, !suppressEvent);
-        // We don't want to send hangup here if we didn't even get to sending an invite
-        if ([CallState.Fledgling, CallState.WaitLocalMedia].includes(this.state)) return;
-        const content: Omit<MCallHangupReject, "version" | "call_id" | "party_id" | "conf_id"> = {};
-        // 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.
-     */
-    public reject(): void {
-        if (this.state !== CallState.Ringing) {
-            throw Error("Call must be in 'ringing' state to reject!");
-        }
-
-        if (this.opponentVersion === 0) {
-            logger.info(
-                `Call ${this.callId} reject() opponent version is less than 1: sending hangup instead of reject (opponentVersion=${this.opponentVersion})`,
-            );
-            this.hangup(CallErrorCode.UserHangup, true);
-            return;
-        }
-
-        logger.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
-     */
-    private async upgradeCall(audio: boolean, video: boolean): Promise<void> {
-        // We don't do call downgrades
-        if (!audio && !video) return;
-        if (!this.opponentSupportsSDPStreamMetadata()) return;
-
-        try {
-            logger.debug(`Call ${this.callId} upgradeCall() upgrading call (audio=${audio}, video=${video})`);
-            const getAudio = audio || this.hasLocalUserMediaAudioTrack;
-            const getVideo = video || this.hasLocalUserMediaVideoTrack;
-
-            // updateLocalUsermediaStream() will take the tracks, use them as
-            // replacement and throw the stream away, so it isn't reusable
-            const stream = await this.client.getMediaHandler().getUserMediaStream(getAudio, getVideo, false);
-            await this.updateLocalUsermediaStream(stream, audio, video);
-        } catch (error) {
-            logger.error(`Call ${this.callId} upgradeCall() failed to upgrade the call`, error);
-            this.emit(
-                CallEvent.Error,
-                new CallError(CallErrorCode.NoUserMedia, "Failed to get camera access: ", <Error>error),
-                this,
-            );
-        }
-    }
-
-    /**
-     * Returns true if this.remoteSDPStreamMetadata is defined, otherwise returns false
-     * @returns can screenshare
-     */
-    public opponentSupportsSDPStreamMetadata(): boolean {
-        return Boolean(this.remoteSDPStreamMetadata);
-    }
-
-    /**
-     * If there is a screensharing stream returns true, otherwise returns false
-     * @returns is screensharing
-     */
-    public isScreensharing(): boolean {
-        return Boolean(this.localScreensharingStream);
-    }
-
-    /**
-     * Starts/stops screensharing
-     * @param enabled - the desired screensharing state
-     * @param opts - screen sharing options
-     * @returns new screensharing state
-     */
-    public async setScreensharingEnabled(enabled: boolean, opts?: IScreensharingOpts): Promise<boolean> {
-        // Skip if there is nothing to do
-        if (enabled && this.isScreensharing()) {
-            logger.warn(
-                `Call ${this.callId} setScreensharingEnabled() there is already a screensharing stream - there is nothing to do!`,
-            );
-            return true;
-        } else if (!enabled && !this.isScreensharing()) {
-            logger.warn(
-                `Call ${this.callId} setScreensharingEnabled() there already isn't a screensharing stream - there is nothing to do!`,
-            );
-            return false;
-        }
-
-        // Fallback to replaceTrack()
-        if (!this.opponentSupportsSDPStreamMetadata()) {
-            return this.setScreensharingEnabledWithoutMetadataSupport(enabled, opts);
-        }
-
-        logger.debug(`Call ${this.callId} setScreensharingEnabled() running (enabled=${enabled})`);
-        if (enabled) {
-            try {
-                const stream = await this.client.getMediaHandler().getScreensharingStream(opts);
-                if (!stream) return false;
-                this.pushNewLocalFeed(stream, SDPStreamMetadataPurpose.Screenshare);
-                return true;
-            } catch (err) {
-                logger.error(`Call ${this.callId} setScreensharingEnabled() failed to get screen-sharing stream:`, err);
-                return false;
-            }
-        } else {
-            const audioTransceiver = this.transceivers.get(
-                getTransceiverKey(SDPStreamMetadataPurpose.Screenshare, "audio"),
-            );
-            const videoTransceiver = this.transceivers.get(
-                getTransceiverKey(SDPStreamMetadataPurpose.Screenshare, "video"),
-            );
-
-            for (const transceiver of [audioTransceiver, videoTransceiver]) {
-                // this is slightly mixing the track and transceiver API but is basically just shorthand
-                // for removing the sender.
-                if (transceiver && transceiver.sender) this.peerConn!.removeTrack(transceiver.sender);
-            }
-
-            this.client.getMediaHandler().stopScreensharingStream(this.localScreensharingStream!);
-            this.deleteFeedByStream(this.localScreensharingStream!);
-            return false;
-        }
-    }
-
-    /**
-     * Starts/stops screensharing
-     * Should be used ONLY if the opponent doesn't support SDPStreamMetadata
-     * @param enabled - the desired screensharing state
-     * @param opts - screen sharing options
-     * @returns new screensharing state
-     */
-    private async setScreensharingEnabledWithoutMetadataSupport(
-        enabled: boolean,
-        opts?: IScreensharingOpts,
-    ): Promise<boolean> {
-        logger.debug(
-            `Call ${this.callId} setScreensharingEnabledWithoutMetadataSupport() running (enabled=${enabled})`,
-        );
-        if (enabled) {
-            try {
-                const stream = await this.client.getMediaHandler().getScreensharingStream(opts);
-                if (!stream) return false;
-
-                const track = stream.getTracks().find((track) => track.kind === "video");
-
-                const sender = this.transceivers.get(
-                    getTransceiverKey(SDPStreamMetadataPurpose.Usermedia, "video"),
-                )?.sender;
-
-                sender?.replaceTrack(track ?? null);
-
-                this.pushNewLocalFeed(stream, SDPStreamMetadataPurpose.Screenshare, false);
-
-                return true;
-            } catch (err) {
-                logger.error(
-                    `Call ${this.callId} setScreensharingEnabledWithoutMetadataSupport() failed to get screen-sharing stream:`,
-                    err,
-                );
-                return false;
-            }
-        } else {
-            const track = this.localUsermediaStream?.getTracks().find((track) => track.kind === "video");
-            const sender = this.transceivers.get(
-                getTransceiverKey(SDPStreamMetadataPurpose.Usermedia, "video"),
-            )?.sender;
-            sender?.replaceTrack(track ?? null);
-
-            this.client.getMediaHandler().stopScreensharingStream(this.localScreensharingStream!);
-            this.deleteFeedByStream(this.localScreensharingStream!);
-
-            return false;
-        }
-    }
-
-    /**
-     * Replaces/adds the tracks from the passed stream to the localUsermediaStream
-     * @param stream - to use a replacement for the local usermedia stream
-     */
-    public async updateLocalUsermediaStream(
-        stream: MediaStream,
-        forceAudio = false,
-        forceVideo = false,
-    ): Promise<void> {
-        const callFeed = this.localUsermediaFeed!;
-        const audioEnabled = forceAudio || (!callFeed.isAudioMuted() && !this.remoteOnHold);
-        const videoEnabled = forceVideo || (!callFeed.isVideoMuted() && !this.remoteOnHold);
-        logger.log(
-            `Call ${this.callId} updateLocalUsermediaStream() running (streamId=${stream.id}, audio=${audioEnabled}, video=${videoEnabled})`,
-        );
-        setTracksEnabled(stream.getAudioTracks(), audioEnabled);
-        setTracksEnabled(stream.getVideoTracks(), videoEnabled);
-
-        // We want to keep the same stream id, so we replace the tracks rather
-        // than the whole stream.
-
-        // Firstly, we replace the tracks in our localUsermediaStream.
-        for (const track of this.localUsermediaStream!.getTracks()) {
-            this.localUsermediaStream!.removeTrack(track);
-            track.stop();
-        }
-        for (const track of stream.getTracks()) {
-            this.localUsermediaStream!.addTrack(track);
-        }
-
-        // Then replace the old tracks, if possible.
-        for (const track of stream.getTracks()) {
-            const tKey = getTransceiverKey(SDPStreamMetadataPurpose.Usermedia, track.kind);
-
-            const transceiver = this.transceivers.get(tKey);
-            const oldSender = transceiver?.sender;
-            let added = false;
-            if (oldSender) {
-                try {
-                    logger.info(
-                        `Call ${this.callId} updateLocalUsermediaStream() replacing track (id=${track.id}, kind=${track.kind}, streamId=${stream.id}, streamPurpose=${callFeed.purpose})`,
-                    );
-                    await oldSender.replaceTrack(track);
-                    // Set the direction to indicate we're going to be sending.
-                    // This is only necessary in the cases where we're upgrading
-                    // the call to video after downgrading it.
-                    transceiver.direction = transceiver.direction === "inactive" ? "sendonly" : "sendrecv";
-                    added = true;
-                } catch (error) {
-                    logger.warn(
-                        `Call ${this.callId} updateLocalUsermediaStream() replaceTrack failed: adding new transceiver instead`,
-                        error,
-                    );
-                }
-            }
-
-            if (!added) {
-                logger.info(
-                    `Call ${this.callId} updateLocalUsermediaStream() adding track to peer connection (id=${track.id}, kind=${track.kind}, streamId=${stream.id}, streamPurpose=${callFeed.purpose})`,
-                );
-
-                const newSender = this.peerConn!.addTrack(track, this.localUsermediaStream!);
-                const newTransceiver = this.peerConn!.getTransceivers().find((t) => t.sender === newSender);
-                if (newTransceiver) {
-                    this.transceivers.set(tKey, newTransceiver);
-                } else {
-                    logger.warn(
-                        `Call ${this.callId} updateLocalUsermediaStream() couldn't find matching transceiver for newly added track!`,
-                    );
-                }
-            }
-        }
-    }
-
-    /**
-     * Set whether our outbound video should be muted or not.
-     * @param muted - True to mute the outbound video.
-     * @returns the new mute state
-     */
-    public async setLocalVideoMuted(muted: boolean): Promise<boolean> {
-        logger.log(`Call ${this.callId} setLocalVideoMuted() running ${muted}`);
-
-        // if we were still thinking about stopping and removing the video
-        // track: don't, because we want it back.
-        if (!muted && this.stopVideoTrackTimer !== undefined) {
-            clearTimeout(this.stopVideoTrackTimer);
-            this.stopVideoTrackTimer = undefined;
-        }
-
-        if (!(await this.client.getMediaHandler().hasVideoDevice())) {
-            return this.isLocalVideoMuted();
-        }
-
-        if (!this.hasUserMediaVideoSender && !muted) {
-            this.localUsermediaFeed?.setAudioVideoMuted(null, muted);
-            await this.upgradeCall(false, true);
-            return this.isLocalVideoMuted();
-        }
-
-        // we may not have a video track - if not, re-request usermedia
-        if (!muted && this.localUsermediaStream!.getVideoTracks().length === 0) {
-            const stream = await this.client.getMediaHandler().getUserMediaStream(true, true);
-            await this.updateLocalUsermediaStream(stream);
-        }
-
-        this.localUsermediaFeed?.setAudioVideoMuted(null, muted);
-
-        this.updateMuteStatus();
-        await this.sendMetadataUpdate();
-
-        // if we're muting video, set a timeout to stop & remove the video track so we release
-        // the camera. We wait a short time to do this because when we disable a track, WebRTC
-        // will send black video for it. If we just stop and remove it straight away, the video
-        // will just freeze which means that when we unmute video, the other side will briefly
-        // get a static frame of us from before we muted. This way, the still frame is just black.
-        // A very small delay is not always enough so the theory here is that it needs to be long
-        // enough for WebRTC to encode a frame: 120ms should be long enough even if we're only
-        // doing 10fps.
-        if (muted) {
-            this.stopVideoTrackTimer = setTimeout(() => {
-                for (const t of this.localUsermediaStream!.getVideoTracks()) {
-                    t.stop();
-                    this.localUsermediaStream!.removeTrack(t);
-                }
-            }, 120);
-        }
-
-        return this.isLocalVideoMuted();
-    }
-
-    /**
-     * Check if local video is muted.
-     *
-     * If there are multiple video tracks, <i>all</i> of the tracks need to be muted
-     * for this to return true. This means if there are no video tracks, this will
-     * return true.
-     * @returns True if the local preview video is muted, else false
-     * (including if the call is not set up yet).
-     */
-    public isLocalVideoMuted(): boolean {
-        return this.localUsermediaFeed?.isVideoMuted() ?? false;
-    }
-
-    /**
-     * Set whether the microphone should be muted or not.
-     * @param muted - True to mute the mic.
-     * @returns the new mute state
-     */
-    public async setMicrophoneMuted(muted: boolean): Promise<boolean> {
-        logger.log(`Call ${this.callId} setMicrophoneMuted() running ${muted}`);
-        if (!(await this.client.getMediaHandler().hasAudioDevice())) {
-            return this.isMicrophoneMuted();
-        }
-
-        if (!muted && (!this.hasUserMediaAudioSender || !this.hasLocalUserMediaAudioTrack)) {
-            await this.upgradeCall(true, false);
-            return this.isMicrophoneMuted();
-        }
-        this.localUsermediaFeed?.setAudioVideoMuted(muted, null);
-        this.updateMuteStatus();
-        await this.sendMetadataUpdate();
-        return this.isMicrophoneMuted();
-    }
-
-    /**
-     * Check if the microphone is muted.
-     *
-     * If there are multiple audio tracks, <i>all</i> of the tracks need to be muted
-     * for this to return true. This means if there are no audio tracks, this will
-     * return true.
-     * @returns True if the mic is muted, else false (including if the call
-     * is not set up yet).
-     */
-    public isMicrophoneMuted(): boolean {
-        return this.localUsermediaFeed?.isAudioMuted() ?? 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)
-     */
-    public isRemoteOnHold(): boolean {
-        return this.remoteOnHold;
-    }
-
-    public setRemoteOnHold(onHold: boolean): void {
-        if (this.isRemoteOnHold() === onHold) return;
-        this.remoteOnHold = onHold;
-
-        for (const transceiver of this.peerConn!.getTransceivers()) {
-            // We don't send hold music or anything so we're not actually
-            // sending anything, but sendrecv is fairly standard for hold and
-            // it makes it a lot easier to figure out who's put who on hold.
-            transceiver.direction = onHold ? "sendonly" : "sendrecv";
-        }
-        this.updateMuteStatus();
-        this.sendMetadataUpdate();
-
-        this.emit(CallEvent.RemoteHoldUnhold, this.remoteOnHold, this);
-    }
-
-    /**
-     * Indicates whether we are 'on hold' to the remote party (ie. if true,
-     * they cannot hear us).
-     * @returns true if the other party has put us on hold
-     */
-    public isLocalOnHold(): boolean {
-        if (this.state !== CallState.Connected) return false;
-
-        let callOnHold = true;
-
-        // We consider a call to be on hold only if *all* the tracks are on hold
-        // (is this the right thing to do?)
-        for (const transceiver of this.peerConn!.getTransceivers()) {
-            const trackOnHold = ["inactive", "recvonly"].includes(transceiver.currentDirection!);
-
-            if (!trackOnHold) callOnHold = false;
-        }
-
-        return callOnHold;
-    }
-
-    /**
-     * Sends a DTMF digit to the other party
-     * @param digit - The digit (nb. string - '#' and '*' are dtmf too)
-     */
-    public sendDtmfDigit(digit: string): void {
-        for (const sender of this.peerConn!.getSenders()) {
-            if (sender.track?.kind === "audio" && sender.dtmf) {
-                sender.dtmf.insertDTMF(digit);
-                return;
-            }
-        }
-
-        throw new Error("Unable to find a track to send DTMF on");
-    }
-
-    private updateMuteStatus(): void {
-        const micShouldBeMuted = this.isMicrophoneMuted() || this.remoteOnHold;
-        const vidShouldBeMuted = this.isLocalVideoMuted() || this.remoteOnHold;
-
-        logger.log(
-            `Call ${this.callId} updateMuteStatus stream ${
-                this.localUsermediaStream!.id
-            } micShouldBeMuted ${micShouldBeMuted} vidShouldBeMuted ${vidShouldBeMuted}`,
-        );
-
-        setTracksEnabled(this.localUsermediaStream!.getAudioTracks(), !micShouldBeMuted);
-        setTracksEnabled(this.localUsermediaStream!.getVideoTracks(), !vidShouldBeMuted);
-    }
-
-    public async sendMetadataUpdate(): Promise<void> {
-        await this.sendVoipEvent(EventType.CallSDPStreamMetadataChangedPrefix, {
-            [SDPStreamMetadataKey]: this.getLocalSDPStreamMetadata(),
-        });
-    }
-
-    private gotCallFeedsForInvite(callFeeds: CallFeed[], requestScreenshareFeed = false): void {
-        if (this.successor) {
-            this.successor.queueGotCallFeedsForAnswer(callFeeds);
-            return;
-        }
-        if (this.callHasEnded()) {
-            this.stopAllMedia();
-            return;
-        }
-
-        for (const feed of callFeeds) {
-            this.pushLocalFeed(feed);
-        }
-
-        if (requestScreenshareFeed) {
-            this.peerConn!.addTransceiver("video", {
-                direction: "recvonly",
-            });
-        }
-
-        this.state = CallState.CreateOffer;
-
-        logger.debug(`Call ${this.callId} gotUserMediaForInvite() run`);
-        // Now we wait for the negotiationneeded event
-    }
-
-    private async sendAnswer(): Promise<void> {
-        const answerContent = {
-            answer: {
-                sdp: this.peerConn!.localDescription!.sdp,
-                // type is now deprecated as of Matrix VoIP v1, but
-                // required to still be sent for backwards compat
-                type: this.peerConn!.localDescription!.type,
-            },
-            [SDPStreamMetadataKey]: this.getLocalSDPStreamMetadata(true),
-        } as MCallAnswer;
-
-        answerContent.capabilities = {
-            "m.call.transferee": this.client.supportsCallTransfer,
-            "m.call.dtmf": false,
-        };
-
-        // We have just taken the local description from the peerConn which will
-        // contain all the local candidates added so far, so we can discard any candidates
-        // we had queued up because they'll be in the answer.
-        const discardCount = this.discardDuplicateCandidates();
-        logger.info(
-            `Call ${this.callId} sendAnswer() discarding ${discardCount} candidates that will be sent in answer`,
-        );
-
-        try {
-            await this.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.
-            this.inviteOrAnswerSent = true;
-        } catch (error) {
-            // We've failed to answer: back to the ringing state
-            this.state = CallState.Ringing;
-            if (error instanceof MatrixError && error.event) this.client.cancelPendingEvent(error.event);
-
-            let code = CallErrorCode.SendAnswer;
-            let message = "Failed to send answer";
-            if ((<Error>error).name == "UnknownDeviceError") {
-                code = CallErrorCode.UnknownDevices;
-                message = "Unknown devices present in the room";
-            }
-            this.emit(CallEvent.Error, new CallError(code, message, <Error>error), this);
-            throw error;
-        }
-
-        // error handler re-throws so this won't happen on error, but
-        // we don't want the same error handling on the candidate queue
-        this.sendCandidateQueue();
-    }
-
-    private queueGotCallFeedsForAnswer(callFeeds: CallFeed[]): void {
-        // 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
-    private mungeSdp(description: RTCSessionDescriptionInit, mods: CodecParamsMod[]): void {
-        // The only way to enable DTX at this time is through SDP munging
-        const sdp = parseSdp(description.sdp!);
-
-        sdp.media.forEach((media) => {
-            const payloadTypeToCodecMap = new Map<number, string>();
-            const codecToPayloadTypeMap = new Map<string, number>();
-            for (const rtp of media.rtp) {
-                payloadTypeToCodecMap.set(rtp.payload, rtp.codec);
-                codecToPayloadTypeMap.set(rtp.codec, rtp.payload);
-            }
-
-            for (const mod of mods) {
-                if (mod.mediaType !== media.type) continue;
-
-                if (!codecToPayloadTypeMap.has(mod.codec)) {
-                    logger.info(
-                        `Call ${this.callId} mungeSdp() ignoring SDP modifications for ${mod.codec} as it's not present.`,
-                    );
-                    continue;
-                }
-
-                const extraConfig: string[] = [];
-                if (mod.enableDtx !== undefined) {
-                    extraConfig.push(`usedtx=${mod.enableDtx ? "1" : "0"}`);
-                }
-                if (mod.maxAverageBitrate !== undefined) {
-                    extraConfig.push(`maxaveragebitrate=${mod.maxAverageBitrate}`);
-                }
-
-                let found = false;
-                for (const fmtp of media.fmtp) {
-                    if (payloadTypeToCodecMap.get(fmtp.payload) === mod.codec) {
-                        found = true;
-                        fmtp.config += ";" + extraConfig.join(";");
-                    }
-                }
-                if (!found) {
-                    media.fmtp.push({
-                        payload: codecToPayloadTypeMap.get(mod.codec)!,
-                        config: extraConfig.join(";"),
-                    });
-                }
-            }
-        });
-        description.sdp = writeSdp(sdp);
-    }
-
-    private async createOffer(): Promise<RTCSessionDescriptionInit> {
-        const offer = await this.peerConn!.createOffer();
-        this.mungeSdp(offer, getCodecParamMods(this.isPtt));
-        return offer;
-    }
-
-    private async createAnswer(): Promise<RTCSessionDescriptionInit> {
-        const answer = await this.peerConn!.createAnswer();
-        this.mungeSdp(answer, getCodecParamMods(this.isPtt));
-        return answer;
-    }
-
-    private async gotCallFeedsForAnswer(callFeeds: CallFeed[]): Promise<void> {
-        if (this.callHasEnded()) return;
-
-        this.waitForLocalAVStream = false;
-
-        for (const feed of callFeeds) {
-            this.pushLocalFeed(feed);
-        }
-
-        this.state = CallState.CreateAnswer;
-
-        let answer: RTCSessionDescriptionInit;
-        try {
-            this.getRidOfRTXCodecs();
-            answer = await this.createAnswer();
-        } catch (err) {
-            logger.debug(`Call ${this.callId} gotCallFeedsForAnswer() failed to create answer: `, err);
-            this.terminate(CallParty.Local, CallErrorCode.CreateAnswer, true);
-            return;
-        }
-
-        try {
-            await this.peerConn!.setLocalDescription(answer);
-
-            // make sure we're still going
-            if (this.callHasEnded()) return;
-
-            this.state = CallState.Connecting;
-
-            // Allow a short time for initial candidates to be gathered
-            await new Promise((resolve) => {
-                setTimeout(resolve, 200);
-            });
-
-            // make sure the call hasn't ended before we continue
-            if (this.callHasEnded()) return;
-
-            this.sendAnswer();
-        } catch (err) {
-            logger.debug(`Call ${this.callId} gotCallFeedsForAnswer() error setting local description!`, err);
-            this.terminate(CallParty.Local, CallErrorCode.SetLocalDescription, true);
-            return;
-        }
-    }
-
-    /**
-     * Internal
-     */
-    private gotLocalIceCandidate = (event: RTCPeerConnectionIceEvent): void => {
-        if (event.candidate) {
-            if (this.candidatesEnded) {
-                logger.warn(`Call ${this.callId} gotLocalIceCandidate() got candidate after candidates have ended!`);
-            }
-
-            logger.debug(`Call ${this.callId} got local ICE ${event.candidate.sdpMid} ${event.candidate.candidate}`);
-
-            if (this.callHasEnded()) return;
-
-            // As with the offer, note we need to make a copy of this object, not
-            // pass the original: that broke in Chrome ~m43.
-            if (event.candidate.candidate === "") {
-                this.queueCandidate(null);
-            } else {
-                this.queueCandidate(event.candidate);
-            }
-        }
-    };
-
-    private onIceGatheringStateChange = (event: Event): void => {
-        logger.debug(
-            `Call ${this.callId} onIceGatheringStateChange() ice gathering state changed to ${
-                this.peerConn!.iceGatheringState
-            }`,
-        );
-        if (this.peerConn?.iceGatheringState === "complete") {
-            this.queueCandidate(null); // We should leave it to WebRTC to announce the end
-            logger.debug(
-                `Call ${this.callId} onIceGatheringStateChange() ice gathering state complete, set candidates have ended`,
-            );
-        }
-    };
-
-    public async onRemoteIceCandidatesReceived(ev: MatrixEvent): Promise<void> {
-        if (this.callHasEnded()) {
-            //debuglog("Ignoring remote ICE candidate because call has ended");
-            return;
-        }
-
-        const content = ev.getContent<MCallCandidates>();
-        const candidates = content.candidates;
-        if (!candidates) {
-            logger.info(
-                `Call ${this.callId} onRemoteIceCandidatesReceived() ignoring candidates event with no candidates!`,
-            );
-            return;
-        }
-
-        const fromPartyId = content.version === 0 ? null : content.party_id || null;
-
-        if (this.opponentPartyId === undefined) {
-            // we haven't picked an opponent yet so save the candidates
-            if (fromPartyId) {
-                logger.info(
-                    `Call ${this.callId} onRemoteIceCandidatesReceived() buffering ${candidates.length} candidates until we pick an opponent`,
-                );
-                const bufferedCandidates = this.remoteCandidateBuffer.get(fromPartyId) || [];
-                bufferedCandidates.push(...candidates);
-                this.remoteCandidateBuffer.set(fromPartyId, bufferedCandidates);
-            }
-            return;
-        }
-
-        if (!this.partyIdMatches(content)) {
-            logger.info(
-                `Call ${this.callId} onRemoteIceCandidatesReceived() ignoring candidates from party ID ${content.party_id}: we have chosen party ID ${this.opponentPartyId}`,
-            );
-
-            return;
-        }
-
-        await this.addIceCandidates(candidates);
-    }
-
-    /**
-     * Used by MatrixClient.
-     */
-    public async onAnswerReceived(event: MatrixEvent): Promise<void> {
-        const content = event.getContent<MCallAnswer>();
-        logger.debug(`Call ${this.callId} onAnswerReceived() running (hangupParty=${content.party_id})`);
-
-        if (this.callHasEnded()) {
-            logger.debug(`Call ${this.callId} onAnswerReceived() ignoring answer because call has ended`);
-            return;
-        }
-
-        if (this.opponentPartyId !== undefined) {
-            logger.info(
-                `Call ${this.callId} onAnswerReceived() ignoring answer from party ID ${content.party_id}: we already have an answer/reject from ${this.opponentPartyId}`,
-            );
-            return;
-        }
-
-        this.chooseOpponent(event);
-        await this.addBufferedIceCandidates();
-
-        this.state = CallState.Connecting;
-
-        const sdpStreamMetadata = content[SDPStreamMetadataKey];
-        if (sdpStreamMetadata) {
-            this.updateRemoteSDPStreamMetadata(sdpStreamMetadata);
-        } else {
-            logger.warn(
-                `Call ${this.callId} onAnswerReceived() did not get any SDPStreamMetadata! Can not send/receive multiple streams`,
-            );
-        }
-
-        try {
-            this.isSettingRemoteAnswerPending = true;
-            await this.peerConn!.setRemoteDescription(content.answer);
-            this.isSettingRemoteAnswerPending = false;
-            logger.debug(`Call ${this.callId} onAnswerReceived() set remote description: ${content.answer.type}`);
-        } catch (e) {
-            this.isSettingRemoteAnswerPending = false;
-            logger.debug(`Call ${this.callId} onAnswerReceived() failed to set remote description`, e);
-            this.terminate(CallParty.Local, CallErrorCode.SetRemoteDescription, false);
-            return;
-        }
-
-        // If the answer we selected has a party_id, send a select_answer event
-        // We do this after setting the remote description since otherwise we'd block
-        // call setup on it
-        if (this.opponentPartyId !== null) {
-            try {
-                await this.sendVoipEvent(EventType.CallSelectAnswer, {
-                    selected_party_id: this.opponentPartyId!,
-                });
-            } catch (err) {
-                // This isn't fatal, and will just mean that if another party has raced to answer
-                // the call, they won't know they got rejected, so we carry on & don't retry.
-                logger.warn(`Call ${this.callId} onAnswerReceived() failed to send select_answer event`, err);
-            }
-        }
-    }
-
-    public async onSelectAnswerReceived(event: MatrixEvent): Promise<void> {
-        if (this.direction !== CallDirection.Inbound) {
-            logger.warn(
-                `Call ${this.callId} onSelectAnswerReceived() got select_answer for an outbound call: ignoring`,
-            );
-            return;
-        }
-
-        const selectedPartyId = event.getContent<MCallSelectAnswer>().selected_party_id;
-
-        if (selectedPartyId === undefined || selectedPartyId === null) {
-            logger.warn(
-                `Call ${this.callId} onSelectAnswerReceived() got nonsensical select_answer with null/undefined selected_party_id: ignoring`,
-            );
-            return;
-        }
-
-        if (selectedPartyId !== this.ourPartyId) {
-            logger.info(
-                `Call ${this.callId} onSelectAnswerReceived() got select_answer for party ID ${selectedPartyId}: we are party ID ${this.ourPartyId}.`,
-            );
-            // The other party has picked somebody else's answer
-            await this.terminate(CallParty.Remote, CallErrorCode.AnsweredElsewhere, true);
-        }
-    }
-
-    public async onNegotiateReceived(event: MatrixEvent): Promise<void> {
-        const content = event.getContent<MCallInviteNegotiate>();
-        const description = content.description;
-        if (!description || !description.sdp || !description.type) {
-            logger.info(`Call ${this.callId} onNegotiateReceived() ignoring invalid m.call.negotiate event`);
-            return;
-        }
-        // Politeness always follows the direction of the call: in a glare situation,
-        // we pick either the inbound or outbound call, so one side will always be
-        // inbound and one outbound
-        const polite = this.direction === CallDirection.Inbound;
-
-        // Here we follow the perfect negotiation logic from
-        // https://w3c.github.io/webrtc-pc/#perfect-negotiation-example
-        const readyForOffer =
-            !this.makingOffer && (this.peerConn!.signalingState === "stable" || this.isSettingRemoteAnswerPending);
-
-        const offerCollision = description.type === "offer" && !readyForOffer;
-
-        this.ignoreOffer = !polite && offerCollision;
-        if (this.ignoreOffer) {
-            logger.info(
-                `Call ${this.callId} onNegotiateReceived() ignoring colliding negotiate event because we're impolite`,
-            );
-            return;
-        }
-
-        const prevLocalOnHold = this.isLocalOnHold();
-
-        const sdpStreamMetadata = content[SDPStreamMetadataKey];
-        if (sdpStreamMetadata) {
-            this.updateRemoteSDPStreamMetadata(sdpStreamMetadata);
-        } else {
-            logger.warn(
-                `Call ${this.callId} onNegotiateReceived() received negotiation event without SDPStreamMetadata!`,
-            );
-        }
-
-        try {
-            this.isSettingRemoteAnswerPending = description.type == "answer";
-            await this.peerConn!.setRemoteDescription(description); // SRD rolls back as needed
-            this.isSettingRemoteAnswerPending = false;
-
-            logger.debug(`Call ${this.callId} onNegotiateReceived() set remote description: ${description.type}`);
-
-            if (description.type === "offer") {
-                let answer: RTCSessionDescriptionInit;
-                try {
-                    this.getRidOfRTXCodecs();
-                    answer = await this.createAnswer();
-                } catch (err) {
-                    logger.debug(`Call ${this.callId} onNegotiateReceived() failed to create answer: `, err);
-                    this.terminate(CallParty.Local, CallErrorCode.CreateAnswer, true);
-                    return;
-                }
-
-                await this.peerConn!.setLocalDescription(answer);
-                logger.debug(`Call ${this.callId} onNegotiateReceived() create an answer`);
-
-                this.sendVoipEvent(EventType.CallNegotiate, {
-                    lifetime: CALL_TIMEOUT_MS,
-                    description: this.peerConn!.localDescription?.toJSON() as RTCSessionDescription,
-                    [SDPStreamMetadataKey]: this.getLocalSDPStreamMetadata(true),
-                });
-            }
-        } catch (err) {
-            this.isSettingRemoteAnswerPending = false;
-            logger.warn(`Call ${this.callId} onNegotiateReceived() failed to complete negotiation`, err);
-        }
-
-        const newLocalOnHold = this.isLocalOnHold();
-        if (prevLocalOnHold !== newLocalOnHold) {
-            this.emit(CallEvent.LocalHoldUnhold, newLocalOnHold, this);
-            // also this one for backwards compat
-            this.emit(CallEvent.HoldUnhold, newLocalOnHold);
-        }
-    }
-
-    private updateRemoteSDPStreamMetadata(metadata: SDPStreamMetadata): void {
-        this.remoteSDPStreamMetadata = recursivelyAssign(this.remoteSDPStreamMetadata || {}, metadata, true);
-        for (const feed of this.getRemoteFeeds()) {
-            const streamId = feed.stream.id;
-            const metadata = this.remoteSDPStreamMetadata![streamId];
-
-            feed.setAudioVideoMuted(metadata?.audio_muted, metadata?.video_muted);
-            feed.purpose = this.remoteSDPStreamMetadata![streamId]?.purpose;
-        }
-    }
-
-    public onSDPStreamMetadataChangedReceived(event: MatrixEvent): void {
-        const content = event.getContent<MCallSDPStreamMetadataChanged>();
-        const metadata = content[SDPStreamMetadataKey];
-        this.updateRemoteSDPStreamMetadata(metadata);
-    }
-
-    public async onAssertedIdentityReceived(event: MatrixEvent): Promise<void> {
-        const content = event.getContent<MCAllAssertedIdentity>();
-        if (!content.asserted_identity) return;
-
-        this.remoteAssertedIdentity = {
-            id: content.asserted_identity.id,
-            displayName: content.asserted_identity.display_name,
-        };
-        this.emit(CallEvent.AssertedIdentityChanged, this);
-    }
-
-    public callHasEnded(): boolean {
-        // 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;
-    }
-
-    private queueGotLocalOffer(): void {
-        // 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();
-        }
-    }
-
-    private async wrappedGotLocalOffer(): Promise<void> {
-        this.makingOffer = true;
-        try {
-            // XXX: in what situations do we believe gotLocalOffer actually throws? It appears
-            // to handle most of its exceptions itself and terminate the call. I'm not entirely
-            // sure it would ever throw, so I can't add a test for these lines.
-            // Also the tense is different between "gotLocalOffer" and "getLocalOfferFailed" so
-            // it's not entirely clear whether getLocalOfferFailed is just misnamed or whether
-            // they've been cross-polinated somehow at some point.
-            await this.gotLocalOffer();
-        } catch (e) {
-            this.getLocalOfferFailed(e as Error);
-            return;
-        } finally {
-            this.makingOffer = false;
-        }
-    }
-
-    private async gotLocalOffer(): Promise<void> {
-        logger.debug(`Call ${this.callId} gotLocalOffer() running`);
-
-        if (this.callHasEnded()) {
-            logger.debug(
-                `Call ${this.callId} gotLocalOffer() ignoring newly created offer because the call has ended"`,
-            );
-            return;
-        }
-
-        let offer: RTCSessionDescriptionInit;
-        try {
-            this.getRidOfRTXCodecs();
-            offer = await this.createOffer();
-        } catch (err) {
-            logger.debug(`Call ${this.callId} gotLocalOffer() failed to create offer: `, err);
-            this.terminate(CallParty.Local, CallErrorCode.CreateOffer, true);
-            return;
-        }
-
-        try {
-            await this.peerConn!.setLocalDescription(offer);
-        } catch (err) {
-            logger.debug(`Call ${this.callId} gotLocalOffer() error setting local description!`, err);
-            this.terminate(CallParty.Local, CallErrorCode.SetLocalDescription, true);
-            return;
-        }
-
-        if (this.peerConn!.iceGatheringState === "gathering") {
-            // Allow a short time for initial candidates to be gathered
-            await new Promise((resolve) => {
-                setTimeout(resolve, 200);
-            });
-        }
-
-        if (this.callHasEnded()) return;
-
-        const eventType = this.state === CallState.CreateOffer ? EventType.CallInvite : EventType.CallNegotiate;
-
-        const content = {
-            lifetime: CALL_TIMEOUT_MS,
-        } as MCallInviteNegotiate;
-
-        if (eventType === EventType.CallInvite && this.invitee) {
-            content.invitee = this.invitee;
-        }
-
-        // clunky because TypeScript can't follow the types through if we use an expression as the key
-        if (this.state === CallState.CreateOffer) {
-            content.offer = this.peerConn!.localDescription?.toJSON() as RTCSessionDescription;
-        } else {
-            content.description = this.peerConn!.localDescription?.toJSON() as RTCSessionDescription;
-        }
-
-        content.capabilities = {
-            "m.call.transferee": this.client.supportsCallTransfer,
-            "m.call.dtmf": false,
-        };
-
-        content[SDPStreamMetadataKey] = this.getLocalSDPStreamMetadata(true);
-
-        // Get rid of any candidates waiting to be sent: they'll be included in the local
-        // description we just got and will send in the offer.
-        const discardCount = this.discardDuplicateCandidates();
-        logger.info(
-            `Call ${this.callId} gotLocalOffer() discarding ${discardCount} candidates that will be sent in offer`,
-        );
-
-        try {
-            await this.sendVoipEvent(eventType, content);
-        } catch (error) {
-            logger.error(`Call ${this.callId} gotLocalOffer() failed to send invite`, error);
-            if (error instanceof MatrixError && error.event) this.client.cancelPendingEvent(error.event);
-
-            let code = CallErrorCode.SignallingFailed;
-            let message = "Signalling failed";
-            if (this.state === CallState.CreateOffer) {
-                code = CallErrorCode.SendInvite;
-                message = "Failed to send invite";
-            }
-            if ((<Error>error).name == "UnknownDeviceError") {
-                code = CallErrorCode.UnknownDevices;
-                message = "Unknown devices present in the room";
-            }
-
-            this.emit(CallEvent.Error, new CallError(code, message, <Error>error), this);
-            this.terminate(CallParty.Local, code, false);
-
-            // no need to carry on & send the candidate queue, but we also
-            // don't want to rethrow the error
-            return;
-        }
-
-        this.sendCandidateQueue();
-        if (this.state === CallState.CreateOffer) {
-            this.inviteOrAnswerSent = true;
-            this.state = CallState.InviteSent;
-            this.inviteTimeout = setTimeout(() => {
-                this.inviteTimeout = undefined;
-                if (this.state === CallState.InviteSent) {
-                    this.hangup(CallErrorCode.InviteTimeout, false);
-                }
-            }, CALL_TIMEOUT_MS);
-        }
-    }
-
-    private getLocalOfferFailed = (err: Error): void => {
-        logger.error(`Call ${this.callId} getLocalOfferFailed() running`, err);
-
-        this.emit(
-            CallEvent.Error,
-            new CallError(CallErrorCode.LocalOfferFailed, "Failed to get local offer!", err),
-            this,
-        );
-        this.terminate(CallParty.Local, CallErrorCode.LocalOfferFailed, false);
-    };
-
-    private getUserMediaFailed = (err: Error): void => {
-        if (this.successor) {
-            this.successor.getUserMediaFailed(err);
-            return;
-        }
-
-        logger.warn(`Call ${this.callId} getUserMediaFailed() failed to get user media - ending call`, err);
-
-        this.emit(
-            CallEvent.Error,
-            new CallError(
-                CallErrorCode.NoUserMedia,
-                "Couldn't start capturing media! Is your microphone set up and does this app have permission?",
-                err,
-            ),
-            this,
-        );
-        this.terminate(CallParty.Local, CallErrorCode.NoUserMedia, false);
-    };
-
-    private placeCallFailed = (err: Error): void => {
-        if (this.successor) {
-            this.successor.placeCallFailed(err);
-            return;
-        }
-
-        logger.warn(`Call ${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);
-    };
-
-    private onIceConnectionStateChanged = (): void => {
-        if (this.callHasEnded()) {
-            return; // because ICE can still complete as we're ending the call
-        }
-        logger.debug(
-            `Call ${this.callId} onIceConnectionStateChanged() running (state=${this.peerConn?.iceConnectionState}, conn=${this.peerConn?.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?.iceConnectionState ?? "")) {
-            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.peerConn?.iceConnectionState == "failed") {
-            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.peerConn?.restartIce as (() => void) | null) {
-                this.candidatesEnded = false;
-                logger.debug(
-                    `Call ${this.callId} onIceConnectionStateChanged() ice restart (state=${this.peerConn?.iceConnectionState})`,
-                );
-                this.peerConn!.restartIce();
-            } else {
-                logger.info(
-                    `Call ${this.callId} onIceConnectionStateChanged() hanging up call (ICE failed and no ICE restart method)`,
-                );
-                this.hangup(CallErrorCode.IceFailed, false);
-            }
-        } else if (this.peerConn?.iceConnectionState == "disconnected") {
-            this.candidatesEnded = false;
-            this.iceReconnectionTimeOut = setTimeout((): void => {
-                logger.info(
-                    `Call ${this.callId} onIceConnectionStateChanged() ICE restarting because of ICE disconnected, (state=${this.peerConn?.iceConnectionState}, conn=${this.peerConn?.connectionState})`,
-                );
-                if (this.peerConn?.restartIce as (() => void) | null) {
-                    this.candidatesEnded = false;
-                    this.peerConn!.restartIce();
-                }
-                this.iceReconnectionTimeOut = undefined;
-            }, ICE_RECONNECTING_TIMEOUT);
-
-            this.iceDisconnectedTimeout = setTimeout((): void => {
-                logger.info(
-                    `Call ${this.callId} onIceConnectionStateChanged() hanging up call (ICE disconnected for too long)`,
-                );
-                this.hangup(CallErrorCode.IceFailed, false);
-            }, ICE_DISCONNECTED_TIMEOUT);
-            this.state = CallState.Connecting;
-        }
-
-        // In PTT mode, override feed status to muted when we lose connection to
-        // the peer, since we don't want to block the line if they're not saying anything.
-        // Experimenting in Chrome, this happens after 5 or 6 seconds, which is probably
-        // fast enough.
-        if (this.isPtt && ["failed", "disconnected"].includes(this.peerConn!.iceConnectionState)) {
-            for (const feed of this.getRemoteFeeds()) {
-                feed.setAudioVideoMuted(true, true);
-            }
-        }
-    };
-
-    private onSignallingStateChanged = (): void => {
-        logger.debug(`Call ${this.callId} onSignallingStateChanged() running (state=${this.peerConn?.signalingState})`);
-    };
-
-    private onTrack = (ev: RTCTrackEvent): void => {
-        if (ev.streams.length === 0) {
-            logger.warn(
-                `Call ${this.callId} onTrack() called with streamless track streamless (kind=${ev.track.kind})`,
-            );
-            return;
-        }
-
-        const stream = ev.streams[0];
-        this.pushRemoteFeed(stream);
-
-        if (!this.removeTrackListeners.has(stream)) {
-            const onRemoveTrack = (): void => {
-                if (stream.getTracks().length === 0) {
-                    logger.info(`Call ${this.callId} onTrack() removing track (streamId=${stream.id})`);
-                    this.deleteFeedByStream(stream);
-                    stream.removeEventListener("removetrack", onRemoveTrack);
-                    this.removeTrackListeners.delete(stream);
-                }
-            };
-            stream.addEventListener("removetrack", onRemoveTrack);
-            this.removeTrackListeners.set(stream, onRemoveTrack);
-        }
-    };
-
-    private onDataChannel = (ev: RTCDataChannelEvent): void => {
-        this.emit(CallEvent.DataChannel, ev.channel, this);
-    };
-
-    /**
-     * 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)
-     */
-    private getRidOfRTXCodecs(): void {
-        // RTCRtpReceiver.getCapabilities and RTCRtpSender.getCapabilities don't seem to be supported on FF before v113
-        if (!RTCRtpReceiver.getCapabilities || !RTCRtpSender.getCapabilities) return;
-
-        const screenshareVideoTransceiver = this.transceivers.get(
-            getTransceiverKey(SDPStreamMetadataPurpose.Screenshare, "video"),
-        );
-
-        // setCodecPreferences isn't supported on FF (as of v113)
-        if (!screenshareVideoTransceiver || !screenshareVideoTransceiver.setCodecPreferences) return;
-
-        const recvCodecs = RTCRtpReceiver.getCapabilities("video")!.codecs;
-        const sendCodecs = RTCRtpSender.getCapabilities("video")!.codecs;
-        const codecs = [];
-
-        for (const 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();
-                }
-            }
-        }
-    }
-
-    private onNegotiationNeeded = async (): Promise<void> => {
-        logger.info(`Call ${this.callId} onNegotiationNeeded() negotiation is needed!`);
-
-        if (this.state !== CallState.CreateOffer && this.opponentVersion === 0) {
-            logger.info(
-                `Call ${this.callId} onNegotiationNeeded() opponent does not support renegotiation: ignoring negotiationneeded event`,
-            );
-            return;
-        }
-
-        this.queueGotLocalOffer();
-    };
-
-    public onHangupReceived = (msg: MCallHangupReject): void => {
-        logger.debug(`Call ${this.callId} onHangupReceived() running`);
-
-        // party ID must match (our chosen partner hanging up the call) or be undefined (we haven't chosen
-        // a partner yet but we're treating the hangup as a reject as per VoIP v0)
-        if (this.partyIdMatches(msg) || this.state === CallState.Ringing) {
-            // default reason is user_hangup
-            this.terminate(CallParty.Remote, msg.reason || CallErrorCode.UserHangup, true);
-        } else {
-            logger.info(
-                `Call ${this.callId} onHangupReceived() ignoring message from party ID ${msg.party_id}: our partner is ${this.opponentPartyId}`,
-            );
-        }
-    };
-
-    public onRejectReceived = (msg: MCallHangupReject): void => {
-        logger.debug(`Call ${this.callId} onRejectReceived() running`);
-
-        // No need to check party_id for reject because if we'd received either
-        // an answer or reject, we wouldn't be in state InviteSent
-
-        const shouldTerminate =
-            // reject events also end the call if it's ringing: it's another of
-            // our devices rejecting the call.
-            [CallState.InviteSent, CallState.Ringing].includes(this.state) ||
-            // also if we're in the init state and it's an inbound call, since
-            // this means we just haven't entered the ringing state yet
-            (this.state === CallState.Fledgling && this.direction === CallDirection.Inbound);
-
-        if (shouldTerminate) {
-            this.terminate(CallParty.Remote, msg.reason || CallErrorCode.UserHangup, true);
-        } else {
-            logger.debug(`Call ${this.callId} onRejectReceived() called in wrong state (state=${this.state})`);
-        }
-    };
-
-    public onAnsweredElsewhere = (msg: MCallAnswer): void => {
-        logger.debug(`Call ${this.callId} onAnsweredElsewhere() running`);
-        this.terminate(CallParty.Remote, CallErrorCode.AnsweredElsewhere, true);
-    };
-
-    /**
-     * @internal
-     */
-    private async sendVoipEvent<K extends keyof Pick<TimelineEvents, CallEventType>>(
-        eventType: K,
-        content: Omit<TimelineEvents[K], "version" | "call_id" | "party_id" | "conf_id">,
-    ): Promise<void> {
-        const realContent = {
-            ...content,
-            version: VOIP_PROTO_VERSION,
-            call_id: this.callId,
-            party_id: this.ourPartyId,
-            conf_id: this.groupCallId,
-        } as TimelineEvents[K];
-
-        if (this.opponentDeviceId) {
-            const toDeviceSeq = this.toDeviceSeq++;
-            const content = {
-                ...realContent,
-                device_id: this.client.deviceId,
-                sender_session_id: this.client.getSessionId(),
-                dest_session_id: this.opponentSessionId,
-                seq: toDeviceSeq,
-                [ToDeviceMessageId]: uuidv4(),
-            };
-
-            this.emit(
-                CallEvent.SendVoipEvent,
-                {
-                    type: "toDevice",
-                    eventType,
-                    userId: this.invitee || this.getOpponentMember()?.userId,
-                    opponentDeviceId: this.opponentDeviceId,
-                    content,
-                },
-                this,
-            );
-
-            const userId = this.invitee || this.getOpponentMember()!.userId;
-            if (this.client.getUseE2eForGroupCall()) {
-                if (!this.opponentDeviceInfo) {
-                    logger.warn(`Call ${this.callId} sendVoipEvent() failed: we do not have opponentDeviceInfo`);
-                    return;
-                }
-
-                await this.client.encryptAndSendToDevices(
-                    [
-                        {
-                            userId,
-                            deviceInfo: this.opponentDeviceInfo,
-                        },
-                    ],
-                    {
-                        type: eventType,
-                        content,
-                    },
-                );
-            } else {
-                await this.client.sendToDevice(
-                    eventType,
-                    new Map<string, any>([[userId, new Map([[this.opponentDeviceId, content]])]]),
-                );
-            }
-        } else {
-            this.emit(
-                CallEvent.SendVoipEvent,
-                {
-                    type: "sendEvent",
-                    eventType,
-                    roomId: this.roomId,
-                    content: realContent,
-                    userId: this.invitee || this.getOpponentMember()?.userId,
-                },
-                this,
-            );
-
-            await this.client.sendEvent(this.roomId!, eventType, realContent);
-        }
-    }
-
-    /**
-     * Queue a candidate to be sent
-     * @param content - The candidate to queue up, or null if candidates have finished being generated
-     *                and end-of-candidates should be signalled
-     */
-    private queueCandidate(content: RTCIceCandidate | null): void {
-        // We partially de-trickle candidates by waiting for `delay` before sending them
-        // amalgamated, in order to avoid sending too many m.call.candidates events and hitting
-        // rate limits in Matrix.
-        // In practice, it'd be better to remove rate limits for m.call.*
-
-        // N.B. this deliberately lets you queue and send blank candidates, which MSC2746
-        // currently proposes as the way to indicate that candidate gathering is complete.
-        // This will hopefully be changed to an explicit rather than implicit notification
-        // shortly.
-        if (content) {
-            this.candidateSendQueue.push(content);
-        } else {
-            this.candidatesEnded = true;
-        }
-
-        // Don't send the ICE candidates yet if the call is in the ringing state: this
-        // means we tried to pick (ie. started generating candidates) and then failed to
-        // send the answer and went back to the ringing state. Queue up the candidates
-        // to send if we successfully send the answer.
-        // Equally don't send if we haven't yet sent the answer because we can send the
-        // first batch of candidates along with the answer
-        if (this.state === CallState.Ringing || !this.inviteOrAnswerSent) return;
-
-        // MSC2746 recommends these values (can be quite long when calling because the
-        // callee will need a while to answer the call)
-        const delay = this.direction === CallDirection.Inbound ? 500 : 2000;
-
-        if (this.candidateSendTries === 0) {
-            setTimeout(() => {
-                this.sendCandidateQueue();
-            }, delay);
-        }
-    }
-
-    // Discard all non-end-of-candidates messages
-    // Return the number of candidate messages that were discarded.
-    // Call this method before sending an invite or answer message
-    private discardDuplicateCandidates(): number {
-        let discardCount = 0;
-        const newQueue: RTCIceCandidate[] = [];
-
-        for (let i = 0; i < this.candidateSendQueue.length; i++) {
-            const candidate = this.candidateSendQueue[i];
-            if (candidate.candidate === "") {
-                newQueue.push(candidate);
-            } else {
-                discardCount++;
-            }
-        }
-
-        this.candidateSendQueue = newQueue;
-
-        return discardCount;
-    }
-
-    /*
-     * Transfers this call to another user
-     */
-    public async transfer(targetUserId: string): Promise<void> {
-        // Fetch the target user's global profile info: their room avatar / displayname
-        // could be different in whatever room we share with them.
-        const profileInfo = await this.client.getProfileInfo(targetUserId);
-
-        const replacementId = genCallID();
-
-        const body = {
-            replacement_id: genCallID(),
-            target_user: {
-                id: targetUserId,
-                display_name: profileInfo.displayname,
-                avatar_url: profileInfo.avatar_url,
-            },
-            create_call: replacementId,
-        } as MCallReplacesEvent;
-
-        await this.sendVoipEvent(EventType.CallReplaces, body);
-
-        await this.terminate(CallParty.Local, CallErrorCode.Transferred, true);
-    }
-
-    /*
-     * Transfers this call to the target call, effectively 'joining' the
-     * two calls (so the remote parties on each call are connected together).
-     */
-    public async transferToCall(transferTargetCall: MatrixCall): Promise<void> {
-        const targetUserId = transferTargetCall.getOpponentMember()?.userId;
-        const targetProfileInfo = targetUserId ? await this.client.getProfileInfo(targetUserId) : undefined;
-        const opponentUserId = this.getOpponentMember()?.userId;
-        const transfereeProfileInfo = opponentUserId ? await this.client.getProfileInfo(opponentUserId) : undefined;
-
-        const newCallId = genCallID();
-
-        const bodyToTransferTarget = {
-            // the replacements on each side have their own ID, and it's distinct from the
-            // ID of the new call (but we can use the same function to generate it)
-            replacement_id: genCallID(),
-            target_user: {
-                id: opponentUserId,
-                display_name: transfereeProfileInfo?.displayname,
-                avatar_url: transfereeProfileInfo?.avatar_url,
-            },
-            await_call: newCallId,
-        } as MCallReplacesEvent;
-
-        await transferTargetCall.sendVoipEvent(EventType.CallReplaces, bodyToTransferTarget);
-
-        const bodyToTransferee = {
-            replacement_id: genCallID(),
-            target_user: {
-                id: targetUserId,
-                display_name: targetProfileInfo?.displayname,
-                avatar_url: targetProfileInfo?.avatar_url,
-            },
-            create_call: newCallId,
-        } as MCallReplacesEvent;
-
-        await this.sendVoipEvent(EventType.CallReplaces, bodyToTransferee);
-
-        await this.terminate(CallParty.Local, CallErrorCode.Transferred, true);
-        await transferTargetCall.terminate(CallParty.Local, CallErrorCode.Transferred, true);
-    }
-
-    private async terminate(hangupParty: CallParty, hangupReason: CallErrorCode, shouldEmit: boolean): Promise<void> {
-        if (this.callHasEnded()) return;
-
-        this.hangupParty = hangupParty;
-        this.hangupReason = hangupReason;
-        this.state = CallState.Ended;
-
-        if (this.inviteTimeout) {
-            clearTimeout(this.inviteTimeout);
-            this.inviteTimeout = undefined;
-        }
-        if (this.iceDisconnectedTimeout !== undefined) {
-            clearTimeout(this.iceDisconnectedTimeout);
-            this.iceDisconnectedTimeout = undefined;
-        }
-        if (this.callLengthInterval) {
-            clearInterval(this.callLengthInterval);
-            this.callLengthInterval = undefined;
-        }
-        if (this.stopVideoTrackTimer !== undefined) {
-            clearTimeout(this.stopVideoTrackTimer);
-            this.stopVideoTrackTimer = undefined;
-        }
-
-        for (const [stream, listener] of this.removeTrackListeners) {
-            stream.removeEventListener("removetrack", listener);
-        }
-        this.removeTrackListeners.clear();
-
-        this.callStatsAtEnd = await this.collectCallStats();
-
-        // Order is important here: first we stopAllMedia() and only then we can deleteAllFeeds()
-        this.stopAllMedia();
-        this.deleteAllFeeds();
-
-        if (this.peerConn && this.peerConn.signalingState !== "closed") {
-            this.peerConn.close();
-        }
-        this.stats?.removeStatsReportGatherer(this.callId);
-
-        if (shouldEmit) {
-            this.emit(CallEvent.Hangup, this);
-        }
-
-        this.client.callEventHandler!.calls.delete(this.callId);
-    }
-
-    private stopAllMedia(): void {
-        logger.debug(`Call ${this.callId} stopAllMedia() running`);
-
-        for (const feed of this.feeds) {
-            // Slightly awkward as local feed need to go via the correct method on
-            // the MediaHandler so they get removed from MediaHandler (remote tracks
-            // don't)
-            // NB. We clone local streams when passing them to individual calls in a group
-            // call, so we can (and should) stop the clones once we no longer need them:
-            // the other clones will continue fine.
-            if (feed.isLocal() && feed.purpose === 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 ${this.callId} stopAllMedia() stopping stream (streamId=${feed.stream.id})`);
-                for (const track of feed.stream.getTracks()) {
-                    track.stop();
-                }
-            }
-        }
-    }
-
-    private checkForErrorListener(): void {
-        if (this.listeners(EventEmitterEvents.Error).length === 0) {
-            throw new Error("You MUST attach an error listener using call.on('error', function() {})");
-        }
-    }
-
-    private async sendCandidateQueue(): Promise<void> {
-        if (this.candidateSendQueue.length === 0 || this.callHasEnded()) {
-            return;
-        }
-
-        const candidates = this.candidateSendQueue;
-        this.candidateSendQueue = [];
-        ++this.candidateSendTries;
-        const content: Pick<MCallCandidates, "candidates"> = {
-            candidates: candidates.map((candidate) => candidate.toJSON()),
-        };
-        if (this.candidatesEnded) {
-            // If there are no more candidates, signal this by adding an empty string candidate
-            content.candidates.push({
-                candidate: "",
-            });
-        }
-        logger.debug(`Call ${this.callId} sendCandidateQueue() attempting to send ${candidates.length} candidates`);
-        try {
-            await this.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
-            this.candidateSendTries = 0;
-
-            // Try to send candidates again just in case we received more candidates while sending.
-            this.sendCandidateQueue();
-        } catch (error) {
-            // don't retry this event: we'll send another one later as we might
-            // have more candidates by then.
-            if (error instanceof MatrixError && error.event) this.client.cancelPendingEvent(error.event);
-
-            // put all the candidates we failed to send back in the queue
-            this.candidateSendQueue.push(...candidates);
-
-            if (this.candidateSendTries > 5) {
-                logger.debug(
-                    `Call ${this.callId} sendCandidateQueue() failed to send candidates on attempt ${this.candidateSendTries}. Giving up on this call.`,
-                    error,
-                );
-
-                const code = CallErrorCode.SignallingFailed;
-                const message = "Signalling failed";
-
-                this.emit(CallEvent.Error, new CallError(code, message, <Error>error), this);
-                this.hangup(code, false);
-
-                return;
-            }
-
-            const delayMs = 500 * Math.pow(2, this.candidateSendTries);
-            ++this.candidateSendTries;
-            logger.debug(
-                `Call ${this.callId} sendCandidateQueue() failed to send candidates. Retrying in ${delayMs}ms`,
-                error,
-            );
-            setTimeout(() => {
-                this.sendCandidateQueue();
-            }, delayMs);
-        }
-    }
-
-    /**
-     * Place a call to this room.
-     * @throws if you have not specified a listener for 'error' events.
-     * @throws if have passed audio=false.
-     */
-    public async placeCall(audio: boolean, video: boolean): Promise<void> {
-        if (!audio) {
-            throw new Error("You CANNOT start a call without audio");
-        }
-        this.state = CallState.WaitLocalMedia;
-
-        let callFeed: CallFeed;
-        try {
-            const stream = await this.client.getMediaHandler().getUserMediaStream(audio, video);
-
-            // make sure all the tracks are enabled (same as pushNewLocalFeed -
-            // we probably ought to just have one code path for adding streams)
-            setTracksEnabled(stream.getAudioTracks(), true);
-            setTracksEnabled(stream.getVideoTracks(), true);
-
-            callFeed = new CallFeed({
-                client: this.client,
-                roomId: this.roomId,
-                userId: this.client.getUserId()!,
-                deviceId: this.client.getDeviceId() ?? undefined,
-                stream,
-                purpose: SDPStreamMetadataPurpose.Usermedia,
-                audioMuted: false,
-                videoMuted: false,
-            });
-        } catch (e) {
-            this.getUserMediaFailed(<Error>e);
-            return;
-        }
-
-        try {
-            await this.placeCallWithCallFeeds([callFeed]);
-        } catch (e) {
-            this.placeCallFailed(<Error>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.
-     */
-    public async placeCallWithCallFeeds(callFeeds: CallFeed[], requestScreenshareFeed = false): Promise<void> {
-        this.checkForErrorListener();
-        this.direction = CallDirection.Outbound;
-
-        await this.initOpponentCrypto();
-
-        // XXX Find a better way to do this
-        this.client.callEventHandler!.calls.set(this.callId, this);
-
-        // make sure we have valid turn creds. Unless something's gone wrong, it should
-        // poll and keep the credentials valid so this should be instant.
-        const haveTurnCreds = await this.client.checkTurnServers();
-        if (!haveTurnCreds) {
-            logger.warn(
-                `Call ${this.callId} placeCallWithCallFeeds() failed to get TURN credentials! Proceeding with call anyway...`,
-            );
-        }
-
-        // create the peer connection now so it can be gathering candidates while we get user
-        // media (assuming a candidate pool size is configured)
-        this.peerConn = this.createPeerConnection();
-        this.emit(CallEvent.PeerConnectionCreated, this.peerConn, this);
-        this.gotCallFeedsForInvite(callFeeds, requestScreenshareFeed);
-    }
-
-    private createPeerConnection(): RTCPeerConnection {
-        const 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);
-
-        const opponentMember: RoomMember | undefined = this.getOpponentMember();
-        const opponentMemberId = opponentMember ? opponentMember.userId : "unknown";
-        this.stats?.addStatsReportGatherer(this.callId, opponentMemberId, pc);
-        return pc;
-    }
-
-    private partyIdMatches(msg: MCallBase): boolean {
-        // They must either match or both be absent (in which case opponentPartyId will be null)
-        // Also we ignore party IDs on the invite/offer if the version is 0, so we must do the same
-        // here and use null if the version is 0 (woe betide any opponent sending messages in the
-        // same call with different versions)
-        const msgPartyId = msg.version === 0 ? null : msg.party_id || null;
-        return msgPartyId === this.opponentPartyId;
-    }
-
-    // Commits to an opponent for the call
-    // ev: An invite or answer event
-    private chooseOpponent(ev: MatrixEvent): void {
-        // I choo-choo-choose you
-        const msg = ev.getContent<MCallInviteNegotiate | MCallAnswer>();
-
-        logger.debug(`Call ${this.callId} chooseOpponent() running (partyId=${msg.party_id})`);
-
-        this.opponentVersion = msg.version;
-        if (this.opponentVersion === 0) {
-            // set to null to indicate that we've chosen an opponent, but because
-            // they're v0 they have no party ID (even if they sent one, we're ignoring it)
-            this.opponentPartyId = null;
-        } else {
-            // set to their party ID, or if they're naughty and didn't send one despite
-            // not being v0, set it to null to indicate we picked an opponent with no
-            // party ID
-            this.opponentPartyId = msg.party_id || null;
-        }
-        this.opponentCaps = msg.capabilities || ({} as CallCapabilities);
-        this.opponentMember = this.client.getRoom(this.roomId)!.getMember(ev.getSender()!) ?? undefined;
-        if (this.opponentMember) {
-            this.stats?.updateOpponentMember(this.callId, this.opponentMember.userId);
-        }
-    }
-
-    private async addBufferedIceCandidates(): Promise<void> {
-        const bufferedCandidates = this.remoteCandidateBuffer.get(this.opponentPartyId!);
-        if (bufferedCandidates) {
-            logger.info(
-                `Call ${this.callId} addBufferedIceCandidates() adding ${bufferedCandidates.length} buffered candidates for opponent ${this.opponentPartyId}`,
-            );
-            await this.addIceCandidates(bufferedCandidates);
-        }
-        this.remoteCandidateBuffer.clear();
-    }
-
-    private async addIceCandidates(candidates: RTCIceCandidate[] | MCallCandidates["candidates"]): Promise<void> {
-        for (const candidate of candidates) {
-            if (
-                (candidate.sdpMid === null || candidate.sdpMid === undefined) &&
-                (candidate.sdpMLineIndex === null || candidate.sdpMLineIndex === undefined)
-            ) {
-                logger.debug(`Call ${this.callId} addIceCandidates() got remote ICE end-of-candidates`);
-            } else {
-                logger.debug(
-                    `Call ${this.callId} addIceCandidates() got remote ICE candidate (sdpMid=${candidate.sdpMid}, candidate=${candidate.candidate})`,
-                );
-            }
-
-            try {
-                await this.peerConn!.addIceCandidate(candidate);
-            } catch (err) {
-                if (!this.ignoreOffer) {
-                    logger.info(`Call ${this.callId} addIceCandidates() failed to add remote ICE candidate`, err);
-                } else {
-                    logger.debug(
-                        `Call ${this.callId} addIceCandidates() failed to add remote ICE candidate because ignoring offer`,
-                        err,
-                    );
-                }
-            }
-        }
-    }
-
-    public get hasPeerConnection(): boolean {
-        return Boolean(this.peerConn);
-    }
-
-    public initStats(stats: GroupCallStats, peerId = "unknown"): void {
-        this.stats = stats;
-        this.stats.start();
-    }
-}
-
-export function setTracksEnabled(tracks: Array<MediaStreamTrack>, enabled: boolean): void {
-    for (const track of tracks) {
-        track.enabled = enabled;
-    }
-}
-
-export function supportsMatrixCall(): boolean {
-    // typeof prevents Node from erroring on an undefined reference
-    if (typeof window === "undefined" || typeof document === "undefined") {
-        // NB. We don't log here as apps try to create a call object as a test for
-        // whether calls are supported, so we shouldn't fill the logs up.
-        return false;
-    }
-
-    // Firefox throws on so little as accessing the RTCPeerConnection when operating in a secure mode.
-    // There's some information at https://bugzilla.mozilla.org/show_bug.cgi?id=1542616 though the concern
-    // is that the browser throwing a SecurityError will brick the client creation process.
-    try {
-        const supported = Boolean(
-            window.RTCPeerConnection ||
-                window.RTCSessionDescription ||
-                window.RTCIceCandidate ||
-                navigator.mediaDevices,
-        );
-        if (!supported) {
-            /* istanbul ignore if */ // Adds a lot of noise to test runs, so disable logging there.
-            if (process.env.NODE_ENV !== "test") {
-                logger.error("WebRTC is not supported in this browser / environment");
-            }
-            return false;
-        }
-    } catch (e) {
-        logger.error("Exception thrown when trying to access WebRTC", e);
-        return false;
-    }
-
-    return true;
-}
-
-/**
- * DEPRECATED
- * Use client.createCall()
- *
- * Create a new Matrix call for the browser.
- * @param client - The client instance to use.
- * @param roomId - The room the call is in.
- * @param options - DEPRECATED optional options map.
- * @returns the call or null if the browser doesn't support calling.
- */
-export function createNewMatrixCall(
-    client: MatrixClient,
-    roomId: string,
-    options?: Pick<CallOpts, "forceTURN" | "invitee" | "opponentDeviceId" | "opponentSessionId" | "groupCallId">,
-): MatrixCall | null {
-    if (!supportsMatrixCall()) return null;
-
-    const optionsForceTURN = options ? options.forceTURN : false;
-
-    const opts: CallOpts = {
-        client: client,
-        roomId: roomId,
-        invitee: options?.invitee,
-        turnServers: client.getTurnServers(),
-        // call level options
-        forceTURN: client.forceTURN || optionsForceTURN,
-        opponentDeviceId: options?.opponentDeviceId,
-        opponentSessionId: options?.opponentSessionId,
-        groupCallId: options?.groupCallId,
-    };
-    const call = new MatrixCall(opts);
-
-    client.reEmitter.reEmit(call, Object.values(CallEvent));
-
-    return call;
-}