@textrp/briij-js-sdk 41.0.0

package/src/webrtc/call.ts

@@ -0,0 +1,3061 @@
+/*
+Copyright 2015, 2016 OpenMarket Ltd
+Copyright 2017 New Vector Ltd
+Copyright 2019, 2020 The Matrix.org Foundation C.I.C.
+Copyright 2021 - 2022 Šimon Brandner <simon.bra.ag@gmail.com>
+
+Licensed under the Apache License, Version 2.0 (the "License");
+you may not use this file except in compliance with the License.
+You may obtain a copy of the License at
+
+    http://www.apache.org/licenses/LICENSE-2.0
+
+Unless required by applicable law or agreed to in writing, software
+distributed under the License is distributed on an "AS IS" BASIS,
+WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+See the License for the specific language governing permissions and
+limitations under the License.
+*/
+
+/**
+ * This is an internal module. See {@link createNewBriijCall} for the public API.
+ */
+
+import { v4 as uuidv4 } from "uuid";
+import { parse as parseSdp, write as writeSdp } from "sdp-transform";
+
+import { logger } from "../logger.ts";
+import { checkObjectHasKeys, isNullOrUndefined, recursivelyAssign } from "../utils.ts";
+import { type BriijEvent } from "../models/event.ts";
+import { EventType, type TimelineEvents, ToDeviceMessageId } from "../@types/event.ts";
+import { type RoomMember } from "../models/room-member.ts";
+import { secureRandomString } from "../randomstring.ts";
+import {
+    type MCallReplacesEvent,
+    type MCallAnswer,
+    type MCallInviteNegotiate,
+    type CallCapabilities,
+    SDPStreamMetadataPurpose,
+    type SDPStreamMetadata,
+    SDPStreamMetadataKey,
+    type MCallSDPStreamMetadataChanged,
+    type MCallSelectAnswer,
+    type MCAllAssertedIdentity,
+    type MCallCandidates,
+    type MCallBase,
+    type MCallHangupReject,
+} from "./callEventTypes.ts";
+import { CallFeed } from "./callFeed.ts";
+import { type BriijClient } from "../client.ts";
+import { EventEmitterEvents, TypedEventEmitter } from "../models/typed-event-emitter.ts";
+import { GroupCallUnknownDeviceError } from "./groupCall.ts";
+import { type IScreensharingOpts } from "./mediaHandler.ts";
+import { BriijError } from "../http-api/index.ts";
+import { type 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: BriijClient;
+    /**
+     * 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() + secureRandomString(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.CallSDPStreamMetadataChanged;
+
+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: BriijCall) => void;
+    [CallEvent.FeedsChanged]: (feeds: CallFeed[], call: BriijCall) => void;
+    [CallEvent.Replaced]: (newCall: BriijCall, oldCall: BriijCall) => void;
+    [CallEvent.Error]: (error: CallError, call: BriijCall) => void;
+    [CallEvent.RemoteHoldUnhold]: (onHold: boolean, call: BriijCall) => void;
+    [CallEvent.LocalHoldUnhold]: (onHold: boolean, call: BriijCall) => void;
+    [CallEvent.LengthChanged]: (length: number, call: BriijCall) => void;
+    [CallEvent.State]: (state: CallState, oldState: CallState, call: BriijCall) => void;
+    [CallEvent.Hangup]: (call: BriijCall) => void;
+    [CallEvent.AssertedIdentityChanged]: (call: BriijCall) => void;
+    /* @deprecated */
+    [CallEvent.HoldUnhold]: (onHold: boolean) => void;
+    [CallEvent.SendVoipEvent]: (event: VoipEvent, call: BriijCall) => void;
+    [CallEvent.PeerConnectionCreated]: (peerConn: RTCPeerConnection, call: BriijCall) => 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 BriijCall 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: BriijClient;
+    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?: BriijCall;
+    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 hasOpponentDeviceInfo?: boolean;
+    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 BriijClient 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.getCrypto()) {
+            // All we know is the device ID
+            this.hasOpponentDeviceInfo = true;
+            return;
+        }
+        const userId = this.invitee || this.getOpponentMember()?.userId;
+
+        if (!userId) throw new Error("Couldn't find opponent user ID to init crypto");
+
+        // Here we were calling `BriijClient.crypto.deviceList.downloadKeys` which is not supported by the rust cryptography.
+        this.hasOpponentDeviceInfo = false;
+        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 BriijClient.
+     * @param event - The m.call.invite event
+     */
+    public async initWithInvite(event: BriijEvent): 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 = SDPStreamMetadataKey.findIn(invite);
+        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 BriijClient.
+     * @param event - The m.call.hangup event
+     */
+    public initWithHangup(event: BriijEvent): 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
+     * BriijClient.
+     * @param newCall - The new call.
+     */
+    public replacedBy(newCall: BriijCall): 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.CallSDPStreamMetadataChanged, {
+            [SDPStreamMetadataKey.name]: 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: Omit<MCallAnswer, "version" | "call_id" | "party_id" | "conf_id"> = {
+            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.name]: this.getLocalSDPStreamMetadata(true),
+        };
+
+        answerContent.capabilities = {
+            "m.call.transferee": this.client.supportsCallTransfer,
+            "m.call.dtmf": false,
+        };
+
+        // We have just taken the local description from the peerConn which will
+        // contain all the local candidates added so far, so we can discard any candidates
+        // we had queued up because they'll be in the answer.
+        const discardCount = this.discardDuplicateCandidates();
+        logger.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 BriijError && 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: BriijEvent): 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 BriijClient.
+     */
+    public async onAnswerReceived(event: BriijEvent): 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 = SDPStreamMetadataKey.findIn(content);
+        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: BriijEvent): 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: BriijEvent): 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 = SDPStreamMetadataKey.findIn<SDPStreamMetadata>(content);
+        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.name]: 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: BriijEvent): void {
+        const content = event.getContent<MCallSDPStreamMetadataChanged>();
+        const metadata = SDPStreamMetadataKey.findIn<SDPStreamMetadata>(content);
+        if (metadata) {
+            this.updateRemoteSDPStreamMetadata(metadata);
+        }
+    }
+
+    public async onAssertedIdentityReceived(event: BriijEvent): 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.name] = 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 BriijError && 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.hasOpponentDeviceInfo) {
+                    logger.warn(`Call ${this.callId} sendVoipEvent() failed: we do not have opponentDeviceInfo`);
+                    return;
+                }
+
+                // TODO: Here we were sending the event to the opponent's device as a to-device message with BriijClient.encryptAndSendToDevice.
+                // However due to the switch to Rust cryptography we need to migrate to the new encryptToDeviceMessages API.
+                throw new Error("Unimplemented");
+            } else {
+                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: BriijCall): 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 BriijError && 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: BriijEvent): 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 supportsBriijCall(): 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 createNewBriijCall(
+    client: BriijClient,
+    roomId: string,
+    options?: Pick<CallOpts, "forceTURN" | "invitee" | "opponentDeviceId" | "opponentSessionId" | "groupCallId">,
+): BriijCall | null {
+    if (!supportsBriijCall()) 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 BriijCall(opts);
+
+    client.reEmitter.reEmit(call, Object.values(CallEvent));
+
+    return call;
+}