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Diffstat (limited to 'includes/external/matrix/node_modules/matrix-js-sdk/src/webrtc/call.ts')
| -rw-r--r-- | includes/external/matrix/node_modules/matrix-js-sdk/src/webrtc/call.ts | 2962 |
1 files changed, 0 insertions, 2962 deletions
diff --git a/includes/external/matrix/node_modules/matrix-js-sdk/src/webrtc/call.ts b/includes/external/matrix/node_modules/matrix-js-sdk/src/webrtc/call.ts deleted file mode 100644 index cd75c10..0000000 --- a/includes/external/matrix/node_modules/matrix-js-sdk/src/webrtc/call.ts +++ /dev/null @@ -1,2962 +0,0 @@ -/* -Copyright 2015, 2016 OpenMarket Ltd -Copyright 2017 New Vector Ltd -Copyright 2019, 2020 The Matrix.org Foundation C.I.C. -Copyright 2021 - 2022 Šimon Brandner <simon.bra.ag@gmail.com> - -Licensed under the Apache License, Version 2.0 (the "License"); -you may not use this file except in compliance with the License. -You may obtain a copy of the License at - - http://www.apache.org/licenses/LICENSE-2.0 - -Unless required by applicable law or agreed to in writing, software -distributed under the License is distributed on an "AS IS" BASIS, -WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. -See the License for the specific language governing permissions and -limitations under the License. -*/ - -/** - * This is an internal module. See {@link createNewMatrixCall} for the public API. - */ - -import { v4 as uuidv4 } from "uuid"; -import { parse as parseSdp, write as writeSdp } from "sdp-transform"; - -import { logger } from "../logger"; -import * as utils from "../utils"; -import { IContent, MatrixEvent } from "../models/event"; -import { EventType, ToDeviceMessageId } from "../@types/event"; -import { RoomMember } from "../models/room-member"; -import { randomString } from "../randomstring"; -import { - MCallReplacesEvent, - MCallAnswer, - MCallInviteNegotiate, - CallCapabilities, - SDPStreamMetadataPurpose, - SDPStreamMetadata, - SDPStreamMetadataKey, - MCallSDPStreamMetadataChanged, - MCallSelectAnswer, - MCAllAssertedIdentity, - MCallCandidates, - MCallBase, - MCallHangupReject, -} from "./callEventTypes"; -import { CallFeed } from "./callFeed"; -import { MatrixClient } from "../client"; -import { EventEmitterEvents, TypedEventEmitter } from "../models/typed-event-emitter"; -import { DeviceInfo } from "../crypto/deviceinfo"; -import { GroupCallUnknownDeviceError } from "./groupCall"; -import { IScreensharingOpts } from "./mediaHandler"; -import { MatrixError } from "../http-api"; -import { GroupCallStats } from "./stats/groupCallStats"; - -interface CallOpts { - // The room ID for this call. - roomId: string; - invitee?: string; - // The Matrix Client instance to send events to. - client: MatrixClient; - /** - * Whether relay through TURN should be forced. - * @deprecated use opts.forceTURN when creating the matrix client - * since it's only possible to set this option on outbound calls. - */ - forceTURN?: boolean; - // A list of TURN servers. - turnServers?: Array<TurnServer>; - opponentDeviceId?: string; - opponentSessionId?: string; - groupCallId?: string; -} - -interface TurnServer { - urls: Array<string>; - username?: string; - password?: string; - ttl?: number; -} - -interface AssertedIdentity { - id: string; - displayName: string; -} - -enum MediaType { - AUDIO = "audio", - VIDEO = "video", -} - -enum CodecName { - OPUS = "opus", - // add more as needed -} - -// Used internally to specify modifications to codec parameters in SDP -interface CodecParamsMod { - mediaType: MediaType; - codec: CodecName; - enableDtx?: boolean; // true to enable discontinuous transmission, false to disable, undefined to leave as-is - maxAverageBitrate?: number; // sets the max average bitrate, or undefined to leave as-is -} - -export enum CallState { - Fledgling = "fledgling", - InviteSent = "invite_sent", - WaitLocalMedia = "wait_local_media", - CreateOffer = "create_offer", - CreateAnswer = "create_answer", - Connecting = "connecting", - Connected = "connected", - Ringing = "ringing", - Ended = "ended", -} - -export enum CallType { - Voice = "voice", - Video = "video", -} - -export enum CallDirection { - Inbound = "inbound", - Outbound = "outbound", -} - -export enum CallParty { - Local = "local", - Remote = "remote", -} - -export enum CallEvent { - Hangup = "hangup", - State = "state", - Error = "error", - Replaced = "replaced", - - // The value of isLocalOnHold() has changed - LocalHoldUnhold = "local_hold_unhold", - // The value of isRemoteOnHold() has changed - RemoteHoldUnhold = "remote_hold_unhold", - // backwards compat alias for LocalHoldUnhold: remove in a major version bump - HoldUnhold = "hold_unhold", - // Feeds have changed - FeedsChanged = "feeds_changed", - - AssertedIdentityChanged = "asserted_identity_changed", - - LengthChanged = "length_changed", - - DataChannel = "datachannel", - - SendVoipEvent = "send_voip_event", -} - -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. */ -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 - -export class CallError extends Error { - public readonly code: string; - - public constructor(code: CallErrorCode, msg: string, err: Error) { - // Still don't think there's any way to have proper nested errors - super(msg + ": " + err); - - this.code = code; - } -} - -export function genCallID(): string { - return Date.now().toString() + randomString(16); -} - -function getCodecParamMods(isPtt: boolean): CodecParamsMod[] { - const mods = [ - { - mediaType: "audio", - codec: "opus", - enableDtx: true, - maxAverageBitrate: isPtt ? 12000 : undefined, - }, - ] as CodecParamsMod[]; - - return mods; -} - -export interface VoipEvent { - type: "toDevice" | "sendEvent"; - eventType: string; - userId?: string; - opponentDeviceId?: string; - roomId?: string; - content: Record<string, unknown>; -} - -/** - * These now all have the call object as an argument. Why? Well, to know which call a given event is - * about you have three options: - * 1. Use a closure as the callback that remembers what call it's listening to. This can be - * a pain because you need to pass the listener function again when you remove the listener, - * which might be somewhere else. - * 2. Use not-very-well-known fact that EventEmitter sets 'this' to the emitter object in the - * callback. This doesn't really play well with modern Typescript and eslint and doesn't work - * with our pattern of re-emitting events. - * 3. Pass the object in question as an argument to the callback. - * - * Now that we have group calls which have to deal with multiple call objects, this will - * become more important, and I think methods 1 and 2 are just going to cause issues. - */ -export type CallEventHandlerMap = { - [CallEvent.DataChannel]: (channel: RTCDataChannel, call: MatrixCall) => void; - [CallEvent.FeedsChanged]: (feeds: CallFeed[], call: MatrixCall) => void; - [CallEvent.Replaced]: (newCall: MatrixCall, oldCall: MatrixCall) => void; - [CallEvent.Error]: (error: CallError, call: MatrixCall) => void; - [CallEvent.RemoteHoldUnhold]: (onHold: boolean, call: MatrixCall) => void; - [CallEvent.LocalHoldUnhold]: (onHold: boolean, call: MatrixCall) => void; - [CallEvent.LengthChanged]: (length: number, call: MatrixCall) => void; - [CallEvent.State]: (state: CallState, oldState: CallState, call: MatrixCall) => void; - [CallEvent.Hangup]: (call: MatrixCall) => void; - [CallEvent.AssertedIdentityChanged]: (call: MatrixCall) => void; - /* @deprecated */ - [CallEvent.HoldUnhold]: (onHold: boolean) => void; - [CallEvent.SendVoipEvent]: (event: VoipEvent, call: MatrixCall) => void; -}; - -// The key of the transceiver map (purpose + media type, separated by ':') -type TransceiverKey = string; - -// generates keys for the map of transceivers -// kind is unfortunately a string rather than MediaType as this is the type of -// track.kind -function getTransceiverKey(purpose: SDPStreamMetadataPurpose, kind: TransceiverKey): string { - return purpose + ":" + kind; -} - -export class MatrixCall extends TypedEventEmitter<CallEvent, CallEventHandlerMap> { - public roomId: string; - public callId: string; - public invitee?: string; - public hangupParty?: CallParty; - public hangupReason?: string; - public direction?: CallDirection; - public ourPartyId: string; - public peerConn?: RTCPeerConnection; - public toDeviceSeq = 0; - - // whether this call should have push-to-talk semantics - // This should be set by the consumer on incoming & outgoing calls. - public isPtt = false; - - private _state = CallState.Fledgling; - private readonly client: MatrixClient; - private readonly forceTURN?: boolean; - private readonly turnServers: Array<TurnServer>; - // A queue for candidates waiting to go out. - // We try to amalgamate candidates into a single candidate message where - // possible - private candidateSendQueue: Array<RTCIceCandidate> = []; - private candidateSendTries = 0; - private candidatesEnded = false; - private feeds: Array<CallFeed> = []; - - // our transceivers for each purpose and type of media - private transceivers = new Map<TransceiverKey, RTCRtpTransceiver>(); - - private inviteOrAnswerSent = false; - private waitForLocalAVStream = false; - private successor?: MatrixCall; - private opponentMember?: RoomMember; - private opponentVersion?: number | string; - // The party ID of the other side: undefined if we haven't chosen a partner - // yet, null if we have but they didn't send a party ID. - private opponentPartyId: string | null | undefined; - private opponentCaps?: CallCapabilities; - private iceDisconnectedTimeout?: ReturnType<typeof setTimeout>; - private 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 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, RTCIceCandidate[]>(); - - private remoteAssertedIdentity?: AssertedIdentity; - private remoteSDPStreamMetadata?: SDPStreamMetadata; - - private callLengthInterval?: ReturnType<typeof setInterval>; - private callStartTime?: number; - - private opponentDeviceId?: string; - private opponentDeviceInfo?: DeviceInfo; - private opponentSessionId?: string; - public groupCallId?: string; - - // Used to keep the timer for the delay before actually stopping our - // video track after muting (see setLocalVideoMuted) - private stopVideoTrackTimer?: ReturnType<typeof setTimeout>; - // Used to allow connection without Video and Audio. To establish a webrtc connection without media a Data channel is - // needed At the moment this property is true if we allow MatrixClient with isVoipWithNoMediaAllowed = true - private readonly isOnlyDataChannelAllowed: boolean; - private stats: GroupCallStats | undefined; - - /** - * Construct a new Matrix Call. - * @param opts - Config options. - */ - public constructor(opts: CallOpts) { - super(); - - this.roomId = opts.roomId; - this.invitee = opts.invitee; - this.client = opts.client; - - if (!this.client.deviceId) throw new Error("Client must have a device ID to start calls"); - - this.forceTURN = opts.forceTURN ?? false; - this.ourPartyId = this.client.deviceId; - this.opponentDeviceId = opts.opponentDeviceId; - this.opponentSessionId = opts.opponentSessionId; - this.groupCallId = opts.groupCallId; - // Array of Objects with urls, username, credential keys - this.turnServers = opts.turnServers || []; - if (this.turnServers.length === 0 && this.client.isFallbackICEServerAllowed()) { - this.turnServers.push({ - urls: [FALLBACK_ICE_SERVER], - }); - } - for (const server of this.turnServers) { - utils.checkObjectHasKeys(server, ["urls"]); - } - this.callId = genCallID(); - // If the Client provides calls without audio and video we need a datachannel for a webrtc connection - this.isOnlyDataChannelAllowed = this.client.isVoipWithNoMediaAllowed; - } - - /** - * Place a voice call to this room. - * @throws If you have not specified a listener for 'error' events. - */ - public async placeVoiceCall(): Promise<void> { - await this.placeCall(true, false); - } - - /** - * Place a video call to this room. - * @throws If you have not specified a listener for 'error' events. - */ - public async placeVideoCall(): Promise<void> { - await this.placeCall(true, true); - } - - /** - * Create a datachannel using this call's peer connection. - * @param label - A human readable label for this datachannel - * @param options - An object providing configuration options for the data channel. - */ - public createDataChannel(label: string, options: RTCDataChannelInit | undefined): RTCDataChannel { - const dataChannel = this.peerConn!.createDataChannel(label, options); - this.emit(CallEvent.DataChannel, dataChannel, this); - return dataChannel; - } - - public getOpponentMember(): RoomMember | undefined { - return this.opponentMember; - } - - public getOpponentDeviceId(): string | undefined { - return this.opponentDeviceId; - } - - public getOpponentSessionId(): string | undefined { - return this.opponentSessionId; - } - - public opponentCanBeTransferred(): boolean { - return Boolean(this.opponentCaps && this.opponentCaps["m.call.transferee"]); - } - - public opponentSupportsDTMF(): boolean { - return Boolean(this.opponentCaps && this.opponentCaps["m.call.dtmf"]); - } - - public getRemoteAssertedIdentity(): AssertedIdentity | undefined { - return this.remoteAssertedIdentity; - } - - public get state(): CallState { - return this._state; - } - - private set state(state: CallState) { - const oldState = this._state; - this._state = state; - this.emit(CallEvent.State, state, oldState, this); - } - - public get type(): CallType { - // we may want to look for a video receiver here rather than a track to match the - // sender behaviour, although in practice they should be the same thing - return this.hasUserMediaVideoSender || this.hasRemoteUserMediaVideoTrack ? CallType.Video : CallType.Voice; - } - - public get hasLocalUserMediaVideoTrack(): boolean { - return !!this.localUsermediaStream?.getVideoTracks().length; - } - - public get hasRemoteUserMediaVideoTrack(): boolean { - return this.getRemoteFeeds().some((feed) => { - return feed.purpose === SDPStreamMetadataPurpose.Usermedia && feed.stream?.getVideoTracks().length; - }); - } - - public get hasLocalUserMediaAudioTrack(): boolean { - return !!this.localUsermediaStream?.getAudioTracks().length; - } - - public get hasRemoteUserMediaAudioTrack(): boolean { - return this.getRemoteFeeds().some((feed) => { - return feed.purpose === SDPStreamMetadataPurpose.Usermedia && !!feed.stream?.getAudioTracks().length; - }); - } - - private get hasUserMediaAudioSender(): boolean { - return Boolean(this.transceivers.get(getTransceiverKey(SDPStreamMetadataPurpose.Usermedia, "audio"))?.sender); - } - - private get hasUserMediaVideoSender(): boolean { - return Boolean(this.transceivers.get(getTransceiverKey(SDPStreamMetadataPurpose.Usermedia, "video"))?.sender); - } - - public get localUsermediaFeed(): CallFeed | undefined { - return this.getLocalFeeds().find((feed) => feed.purpose === SDPStreamMetadataPurpose.Usermedia); - } - - public get localScreensharingFeed(): CallFeed | undefined { - return this.getLocalFeeds().find((feed) => feed.purpose === SDPStreamMetadataPurpose.Screenshare); - } - - public get localUsermediaStream(): MediaStream | undefined { - return this.localUsermediaFeed?.stream; - } - - public get localScreensharingStream(): MediaStream | undefined { - return this.localScreensharingFeed?.stream; - } - - public get remoteUsermediaFeed(): CallFeed | undefined { - return this.getRemoteFeeds().find((feed) => feed.purpose === SDPStreamMetadataPurpose.Usermedia); - } - - public get remoteScreensharingFeed(): CallFeed | undefined { - return this.getRemoteFeeds().find((feed) => feed.purpose === SDPStreamMetadataPurpose.Screenshare); - } - - public get remoteUsermediaStream(): MediaStream | undefined { - return this.remoteUsermediaFeed?.stream; - } - - public get remoteScreensharingStream(): MediaStream | undefined { - return this.remoteScreensharingFeed?.stream; - } - - private getFeedByStreamId(streamId: string): CallFeed | undefined { - return this.getFeeds().find((feed) => feed.stream.id === streamId); - } - - /** - * Returns an array of all CallFeeds - * @returns CallFeeds - */ - public getFeeds(): Array<CallFeed> { - return this.feeds; - } - - /** - * Returns an array of all local CallFeeds - * @returns local CallFeeds - */ - public getLocalFeeds(): Array<CallFeed> { - return this.feeds.filter((feed) => feed.isLocal()); - } - - /** - * Returns an array of all remote CallFeeds - * @returns remote CallFeeds - */ - public getRemoteFeeds(): Array<CallFeed> { - return this.feeds.filter((feed) => !feed.isLocal()); - } - - private async initOpponentCrypto(): Promise<void> { - if (!this.opponentDeviceId) return; - if (!this.client.getUseE2eForGroupCall()) return; - // It's possible to want E2EE and yet not have the means to manage E2EE - // ourselves (for example if the client is a RoomWidgetClient) - if (!this.client.isCryptoEnabled()) { - // All we know is the device ID - this.opponentDeviceInfo = new DeviceInfo(this.opponentDeviceId); - return; - } - // if we've got to this point, we do want to init crypto, so throw if we can't - if (!this.client.crypto) throw new Error("Crypto is not initialised."); - - const userId = this.invitee || this.getOpponentMember()?.userId; - - if (!userId) throw new Error("Couldn't find opponent user ID to init crypto"); - - const deviceInfoMap = await this.client.crypto.deviceList.downloadKeys([userId], false); - this.opponentDeviceInfo = deviceInfoMap.get(userId)?.get(this.opponentDeviceId); - if (this.opponentDeviceInfo === undefined) { - throw new GroupCallUnknownDeviceError(userId); - } - } - - /** - * Generates and returns localSDPStreamMetadata - * @returns localSDPStreamMetadata - */ - private getLocalSDPStreamMetadata(updateStreamIds = false): SDPStreamMetadata { - const metadata: SDPStreamMetadata = {}; - for (const localFeed of this.getLocalFeeds()) { - if (updateStreamIds) { - localFeed.sdpMetadataStreamId = localFeed.stream.id; - } - - metadata[localFeed.sdpMetadataStreamId] = { - purpose: localFeed.purpose, - audio_muted: localFeed.isAudioMuted(), - video_muted: localFeed.isVideoMuted(), - }; - } - return metadata; - } - - /** - * Returns true if there are no incoming feeds, - * otherwise returns false - * @returns no incoming feeds - */ - public noIncomingFeeds(): boolean { - return !this.feeds.some((feed) => !feed.isLocal()); - } - - private pushRemoteFeed(stream: MediaStream): void { - // Fallback to old behavior if the other side doesn't support SDPStreamMetadata - if (!this.opponentSupportsSDPStreamMetadata()) { - this.pushRemoteFeedWithoutMetadata(stream); - return; - } - - const userId = this.getOpponentMember()!.userId; - const purpose = this.remoteSDPStreamMetadata![stream.id].purpose; - const audioMuted = this.remoteSDPStreamMetadata![stream.id].audio_muted; - const videoMuted = this.remoteSDPStreamMetadata![stream.id].video_muted; - - if (!purpose) { - logger.warn( - `Call ${this.callId} pushRemoteFeed() ignoring stream because we didn't get any metadata about it (streamId=${stream.id})`, - ); - return; - } - - if (this.getFeedByStreamId(stream.id)) { - logger.warn( - `Call ${this.callId} pushRemoteFeed() ignoring stream because we already have a feed for it (streamId=${stream.id})`, - ); - return; - } - - this.feeds.push( - new CallFeed({ - client: this.client, - call: this, - roomId: this.roomId, - userId, - deviceId: this.getOpponentDeviceId(), - stream, - purpose, - audioMuted, - videoMuted, - }), - ); - - this.emit(CallEvent.FeedsChanged, this.feeds, this); - - logger.info( - `Call ${this.callId} pushRemoteFeed() pushed stream (streamId=${stream.id}, active=${stream.active}, purpose=${purpose})`, - ); - } - - /** - * This method is used ONLY if the other client doesn't support sending SDPStreamMetadata - */ - private pushRemoteFeedWithoutMetadata(stream: MediaStream): void { - const userId = this.getOpponentMember()!.userId; - // We can guess the purpose here since the other client can only send one stream - const purpose = SDPStreamMetadataPurpose.Usermedia; - const oldRemoteStream = this.feeds.find((feed) => !feed.isLocal())?.stream; - - // Note that we check by ID and always set the remote stream: Chrome appears - // to make new stream objects when transceiver directionality is changed and the 'active' - // status of streams change - Dave - // If we already have a stream, check this stream has the same id - if (oldRemoteStream && stream.id !== oldRemoteStream.id) { - logger.warn( - `Call ${this.callId} pushRemoteFeedWithoutMetadata() ignoring new stream because we already have stream (streamId=${stream.id})`, - ); - return; - } - - if (this.getFeedByStreamId(stream.id)) { - logger.warn( - `Call ${this.callId} pushRemoteFeedWithoutMetadata() ignoring stream because we already have a feed for it (streamId=${stream.id})`, - ); - return; - } - - this.feeds.push( - new CallFeed({ - client: this.client, - call: this, - roomId: this.roomId, - audioMuted: false, - videoMuted: false, - userId, - deviceId: this.getOpponentDeviceId(), - stream, - purpose, - }), - ); - - this.emit(CallEvent.FeedsChanged, this.feeds, this); - - logger.info( - `Call ${this.callId} pushRemoteFeedWithoutMetadata() pushed stream (streamId=${stream.id}, active=${stream.active})`, - ); - } - - private pushNewLocalFeed(stream: MediaStream, purpose: SDPStreamMetadataPurpose, addToPeerConnection = true): void { - const userId = this.client.getUserId()!; - - // Tracks don't always start off enabled, eg. chrome will give a disabled - // audio track if you ask for user media audio and already had one that - // you'd set to disabled (presumably because it clones them internally). - setTracksEnabled(stream.getAudioTracks(), true); - setTracksEnabled(stream.getVideoTracks(), true); - - if (this.getFeedByStreamId(stream.id)) { - logger.warn( - `Call ${this.callId} pushNewLocalFeed() ignoring stream because we already have a feed for it (streamId=${stream.id})`, - ); - return; - } - - this.pushLocalFeed( - new CallFeed({ - client: this.client, - roomId: this.roomId, - audioMuted: false, - videoMuted: false, - userId, - deviceId: this.getOpponentDeviceId(), - stream, - purpose, - }), - addToPeerConnection, - ); - } - - /** - * Pushes supplied feed to the call - * @param callFeed - to push - * @param addToPeerConnection - whether to add the tracks to the peer connection - */ - public pushLocalFeed(callFeed: CallFeed, addToPeerConnection = true): void { - if (this.feeds.some((feed) => callFeed.stream.id === feed.stream.id)) { - logger.info( - `Call ${this.callId} pushLocalFeed() ignoring duplicate local stream (streamId=${callFeed.stream.id})`, - ); - return; - } - - this.feeds.push(callFeed); - - if (addToPeerConnection) { - for (const track of callFeed.stream.getTracks()) { - logger.info( - `Call ${this.callId} pushLocalFeed() adding track to peer connection (id=${track.id}, kind=${track.kind}, streamId=${callFeed.stream.id}, streamPurpose=${callFeed.purpose}, enabled=${track.enabled})`, - ); - - const tKey = getTransceiverKey(callFeed.purpose, track.kind); - if (this.transceivers.has(tKey)) { - // we already have a sender, so we re-use it. We try to re-use transceivers as much - // as possible because they can't be removed once added, so otherwise they just - // accumulate which makes the SDP very large very quickly: in fact it only takes - // about 6 video tracks to exceed the maximum size of an Olm-encrypted - // Matrix event. - const transceiver = this.transceivers.get(tKey)!; - - transceiver.sender.replaceTrack(track); - // set the direction to indicate we're going to start sending again - // (this will trigger the re-negotiation) - transceiver.direction = transceiver.direction === "inactive" ? "sendonly" : "sendrecv"; - } else { - // create a new one. We need to use addTrack rather addTransceiver for this because firefox - // doesn't yet implement RTCRTPSender.setStreams() - // (https://bugzilla.mozilla.org/show_bug.cgi?id=1510802) so we'd have no way to group the - // two tracks together into a stream. - const newSender = this.peerConn!.addTrack(track, callFeed.stream); - - // now go & fish for the new transceiver - const newTransceiver = this.peerConn!.getTransceivers().find((t) => t.sender === newSender); - if (newTransceiver) { - this.transceivers.set(tKey, newTransceiver); - } else { - logger.warn( - `Call ${this.callId} pushLocalFeed() didn't find a matching transceiver after adding track!`, - ); - } - } - } - } - - logger.info( - `Call ${this.callId} pushLocalFeed() pushed stream (id=${callFeed.stream.id}, active=${callFeed.stream.active}, purpose=${callFeed.purpose})`, - ); - - this.emit(CallEvent.FeedsChanged, this.feeds, this); - } - - /** - * Removes local call feed from the call and its tracks from the peer - * connection - * @param callFeed - to remove - */ - public removeLocalFeed(callFeed: CallFeed): void { - const audioTransceiverKey = getTransceiverKey(callFeed.purpose, "audio"); - const videoTransceiverKey = getTransceiverKey(callFeed.purpose, "video"); - - for (const transceiverKey of [audioTransceiverKey, videoTransceiverKey]) { - // this is slightly mixing the track and transceiver API but is basically just shorthand. - // There is no way to actually remove a transceiver, so this just sets it to inactive - // (or recvonly) and replaces the source with nothing. - if (this.transceivers.has(transceiverKey)) { - const transceiver = this.transceivers.get(transceiverKey)!; - if (transceiver.sender) this.peerConn!.removeTrack(transceiver.sender); - } - } - - if (callFeed.purpose === SDPStreamMetadataPurpose.Screenshare) { - this.client.getMediaHandler().stopScreensharingStream(callFeed.stream); - } - - this.deleteFeed(callFeed); - } - - private deleteAllFeeds(): void { - for (const feed of this.feeds) { - if (!feed.isLocal() || !this.groupCallId) { - feed.dispose(); - } - } - - this.feeds = []; - this.emit(CallEvent.FeedsChanged, this.feeds, this); - } - - private deleteFeedByStream(stream: MediaStream): void { - const feed = this.getFeedByStreamId(stream.id); - if (!feed) { - logger.warn( - `Call ${this.callId} deleteFeedByStream() didn't find the feed to delete (streamId=${stream.id})`, - ); - return; - } - this.deleteFeed(feed); - } - - private deleteFeed(feed: CallFeed): void { - feed.dispose(); - this.feeds.splice(this.feeds.indexOf(feed), 1); - this.emit(CallEvent.FeedsChanged, this.feeds, this); - } - - // The typescript definitions have this type as 'any' :( - public async getCurrentCallStats(): Promise<any[] | undefined> { - if (this.callHasEnded()) { - return this.callStatsAtEnd; - } - - return this.collectCallStats(); - } - - private async collectCallStats(): Promise<any[] | undefined> { - // This happens when the call fails before it starts. - // For example when we fail to get capture sources - if (!this.peerConn) return; - - const statsReport = await this.peerConn.getStats(); - const stats: any[] = []; - statsReport.forEach((item) => { - stats.push(item); - }); - - return stats; - } - - /** - * Configure this call from an invite event. Used by MatrixClient. - * @param event - The m.call.invite event - */ - public async initWithInvite(event: MatrixEvent): Promise<void> { - const invite = event.getContent<MCallInviteNegotiate>(); - this.direction = CallDirection.Inbound; - - // make sure we have valid turn creds. Unless something's gone wrong, it should - // poll and keep the credentials valid so this should be instant. - const haveTurnCreds = await this.client.checkTurnServers(); - if (!haveTurnCreds) { - logger.warn( - `Call ${this.callId} initWithInvite() failed to get TURN credentials! Proceeding with call anyway...`, - ); - } - - const sdpStreamMetadata = invite[SDPStreamMetadataKey]; - if (sdpStreamMetadata) { - this.updateRemoteSDPStreamMetadata(sdpStreamMetadata); - } else { - logger.debug( - `Call ${this.callId} initWithInvite() did not get any SDPStreamMetadata! Can not send/receive multiple streams`, - ); - } - - this.peerConn = this.createPeerConnection(); - // we must set the party ID before await-ing on anything: the call event - // handler will start giving us more call events (eg. candidates) so if - // we haven't set the party ID, we'll ignore them. - this.chooseOpponent(event); - await this.initOpponentCrypto(); - try { - await this.peerConn.setRemoteDescription(invite.offer); - await this.addBufferedIceCandidates(); - } catch (e) { - logger.debug(`Call ${this.callId} initWithInvite() failed to set remote description`, e); - this.terminate(CallParty.Local, CallErrorCode.SetRemoteDescription, false); - return; - } - - const remoteStream = this.feeds.find((feed) => !feed.isLocal())?.stream; - - // According to previous comments in this file, firefox at some point did not - // add streams until media started arriving on them. Testing latest firefox - // (81 at time of writing), this is no longer a problem, so let's do it the correct way. - // - // For example in case of no media webrtc connections like screen share only call we have to allow webrtc - // connections without remote media. In this case we always use a data channel. At the moment we allow as well - // only data channel as media in the WebRTC connection with this setup here. - if (!this.isOnlyDataChannelAllowed && (!remoteStream || remoteStream.getTracks().length === 0)) { - logger.error( - `Call ${this.callId} initWithInvite() no remote stream or no tracks after setting remote description!`, - ); - this.terminate(CallParty.Local, CallErrorCode.SetRemoteDescription, false); - return; - } - - this.state = CallState.Ringing; - - if (event.getLocalAge()) { - // Time out the call if it's ringing for too long - const ringingTimer = setTimeout(() => { - if (this.state == CallState.Ringing) { - logger.debug(`Call ${this.callId} initWithInvite() invite has expired. Hanging up.`); - this.hangupParty = CallParty.Remote; // effectively - this.state = CallState.Ended; - this.stopAllMedia(); - if (this.peerConn!.signalingState != "closed") { - this.peerConn!.close(); - } - this.stats?.removeStatsReportGatherer(this.callId); - this.emit(CallEvent.Hangup, this); - } - }, invite.lifetime - event.getLocalAge()); - - const onState = (state: CallState): void => { - if (state !== CallState.Ringing) { - clearTimeout(ringingTimer); - this.off(CallEvent.State, onState); - } - }; - this.on(CallEvent.State, onState); - } - } - - /** - * Configure this call from a hangup or reject event. Used by MatrixClient. - * @param event - The m.call.hangup event - */ - public initWithHangup(event: MatrixEvent): void { - // perverse as it may seem, sometimes we want to instantiate a call with a - // hangup message (because when getting the state of the room on load, events - // come in reverse order and we want to remember that a call has been hung up) - this.state = CallState.Ended; - } - - private shouldAnswerWithMediaType( - wantedValue: boolean | undefined, - valueOfTheOtherSide: boolean, - type: "audio" | "video", - ): boolean { - if (wantedValue && !valueOfTheOtherSide) { - // TODO: Figure out how to do this - logger.warn( - `Call ${this.callId} shouldAnswerWithMediaType() unable to answer with ${type} because the other side isn't sending it either.`, - ); - return false; - } else if ( - !utils.isNullOrUndefined(wantedValue) && - wantedValue !== valueOfTheOtherSide && - !this.opponentSupportsSDPStreamMetadata() - ) { - logger.warn( - `Call ${this.callId} shouldAnswerWithMediaType() unable to answer with ${type}=${wantedValue} because the other side doesn't support it. Answering with ${type}=${valueOfTheOtherSide}.`, - ); - return valueOfTheOtherSide!; - } - return wantedValue ?? valueOfTheOtherSide!; - } - - /** - * Answer a call. - */ - public async answer(audio?: boolean, video?: boolean): Promise<void> { - if (this.inviteOrAnswerSent) return; - // TODO: Figure out how to do this - if (audio === false && video === false) throw new Error("You CANNOT answer a call without media"); - - if (!this.localUsermediaStream && !this.waitForLocalAVStream) { - const prevState = this.state; - const answerWithAudio = this.shouldAnswerWithMediaType(audio, this.hasRemoteUserMediaAudioTrack, "audio"); - const answerWithVideo = this.shouldAnswerWithMediaType(video, this.hasRemoteUserMediaVideoTrack, "video"); - - this.state = CallState.WaitLocalMedia; - this.waitForLocalAVStream = true; - - try { - const stream = await this.client.getMediaHandler().getUserMediaStream(answerWithAudio, answerWithVideo); - this.waitForLocalAVStream = false; - const usermediaFeed = new CallFeed({ - client: this.client, - roomId: this.roomId, - userId: this.client.getUserId()!, - deviceId: this.client.getDeviceId() ?? undefined, - stream, - purpose: SDPStreamMetadataPurpose.Usermedia, - audioMuted: false, - videoMuted: false, - }); - - const feeds = [usermediaFeed]; - - if (this.localScreensharingFeed) { - feeds.push(this.localScreensharingFeed); - } - - this.answerWithCallFeeds(feeds); - } catch (e) { - if (answerWithVideo) { - // Try to answer without video - logger.warn( - `Call ${this.callId} answer() failed to getUserMedia(), trying to getUserMedia() without video`, - ); - this.state = prevState; - this.waitForLocalAVStream = false; - await this.answer(answerWithAudio, false); - } else { - this.getUserMediaFailed(<Error>e); - return; - } - } - } else if (this.waitForLocalAVStream) { - this.state = CallState.WaitLocalMedia; - } - } - - public answerWithCallFeeds(callFeeds: CallFeed[]): void { - if (this.inviteOrAnswerSent) return; - - this.queueGotCallFeedsForAnswer(callFeeds); - } - - /** - * Replace this call with a new call, e.g. for glare resolution. Used by - * MatrixClient. - * @param newCall - The new call. - */ - public replacedBy(newCall: MatrixCall): void { - logger.debug(`Call ${this.callId} replacedBy() running (newCallId=${newCall.callId})`); - if (this.state === CallState.WaitLocalMedia) { - logger.debug( - `Call ${this.callId} replacedBy() telling new call to wait for local media (newCallId=${newCall.callId})`, - ); - newCall.waitForLocalAVStream = true; - } else if ([CallState.CreateOffer, CallState.InviteSent].includes(this.state)) { - if (newCall.direction === CallDirection.Outbound) { - newCall.queueGotCallFeedsForAnswer([]); - } else { - logger.debug( - `Call ${this.callId} replacedBy() handing local stream to new call(newCallId=${newCall.callId})`, - ); - newCall.queueGotCallFeedsForAnswer(this.getLocalFeeds().map((feed) => feed.clone())); - } - } - this.successor = newCall; - this.emit(CallEvent.Replaced, newCall, this); - this.hangup(CallErrorCode.Replaced, true); - } - - /** - * Hangup a call. - * @param reason - The reason why the call is being hung up. - * @param suppressEvent - True to suppress emitting an event. - */ - public hangup(reason: CallErrorCode, suppressEvent: boolean): void { - if (this.callHasEnded()) return; - - logger.debug(`Call ${this.callId} hangup() ending call (reason=${reason})`); - this.terminate(CallParty.Local, reason, !suppressEvent); - // We don't want to send hangup here if we didn't even get to sending an invite - if ([CallState.Fledgling, CallState.WaitLocalMedia].includes(this.state)) return; - const content: IContent = {}; - // 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 desktopCapturerSourceId - optional id of the desktop capturer source to use - * @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 desktopCapturerSourceId - optional id of the desktop capturer source to use - * @returns new screensharing state - */ - private async setScreensharingEnabledWithoutMetadataSupport( - enabled: boolean, - opts?: IScreensharingOpts, - ): Promise<boolean> { - logger.debug( - `Call ${this.callId} setScreensharingEnabledWithoutMetadataSupport() running (enabled=${enabled})`, - ); - if (enabled) { - try { - const stream = await this.client.getMediaHandler().getScreensharingStream(opts); - if (!stream) return false; - - const track = stream.getTracks().find((track) => track.kind === "video"); - - const sender = this.transceivers.get( - getTransceiverKey(SDPStreamMetadataPurpose.Usermedia, "video"), - )?.sender; - - sender?.replaceTrack(track ?? null); - - this.pushNewLocalFeed(stream, SDPStreamMetadataPurpose.Screenshare, false); - - return true; - } catch (err) { - logger.error( - `Call ${this.callId} setScreensharingEnabledWithoutMetadataSupport() failed to get screen-sharing stream:`, - err, - ); - return false; - } - } else { - const track = this.localUsermediaStream?.getTracks().find((track) => track.kind === "video"); - const sender = this.transceivers.get( - getTransceiverKey(SDPStreamMetadataPurpose.Usermedia, "video"), - )?.sender; - sender?.replaceTrack(track ?? null); - - this.client.getMediaHandler().stopScreensharingStream(this.localScreensharingStream!); - this.deleteFeedByStream(this.localScreensharingStream!); - - return false; - } - } - - /** - * Replaces/adds the tracks from the passed stream to the localUsermediaStream - * @param stream - to use a replacement for the local usermedia stream - */ - public async updateLocalUsermediaStream( - stream: MediaStream, - forceAudio = false, - forceVideo = false, - ): Promise<void> { - const callFeed = this.localUsermediaFeed!; - const audioEnabled = forceAudio || (!callFeed.isAudioMuted() && !this.remoteOnHold); - const videoEnabled = forceVideo || (!callFeed.isVideoMuted() && !this.remoteOnHold); - logger.log( - `Call ${this.callId} updateLocalUsermediaStream() running (streamId=${stream.id}, audio=${audioEnabled}, video=${videoEnabled})`, - ); - setTracksEnabled(stream.getAudioTracks(), audioEnabled); - setTracksEnabled(stream.getVideoTracks(), videoEnabled); - - // We want to keep the same stream id, so we replace the tracks rather - // than the whole stream. - - // Firstly, we replace the tracks in our localUsermediaStream. - for (const track of this.localUsermediaStream!.getTracks()) { - this.localUsermediaStream!.removeTrack(track); - track.stop(); - } - for (const track of stream.getTracks()) { - this.localUsermediaStream!.addTrack(track); - } - - // Then replace the old tracks, if possible. - for (const track of stream.getTracks()) { - const tKey = getTransceiverKey(SDPStreamMetadataPurpose.Usermedia, track.kind); - - const transceiver = this.transceivers.get(tKey); - const oldSender = transceiver?.sender; - let added = false; - if (oldSender) { - try { - logger.info( - `Call ${this.callId} updateLocalUsermediaStream() replacing track (id=${track.id}, kind=${track.kind}, streamId=${stream.id}, streamPurpose=${callFeed.purpose})`, - ); - await oldSender.replaceTrack(track); - // Set the direction to indicate we're going to be sending. - // This is only necessary in the cases where we're upgrading - // the call to video after downgrading it. - transceiver.direction = transceiver.direction === "inactive" ? "sendonly" : "sendrecv"; - added = true; - } catch (error) { - logger.warn( - `Call ${this.callId} updateLocalUsermediaStream() replaceTrack failed: adding new transceiver instead`, - error, - ); - } - } - - if (!added) { - logger.info( - `Call ${this.callId} updateLocalUsermediaStream() adding track to peer connection (id=${track.id}, kind=${track.kind}, streamId=${stream.id}, streamPurpose=${callFeed.purpose})`, - ); - - const newSender = this.peerConn!.addTrack(track, this.localUsermediaStream!); - const newTransceiver = this.peerConn!.getTransceivers().find((t) => t.sender === newSender); - if (newTransceiver) { - this.transceivers.set(tKey, newTransceiver); - } else { - logger.warn( - `Call ${this.callId} updateLocalUsermediaStream() couldn't find matching transceiver for newly added track!`, - ); - } - } - } - } - - /** - * Set whether our outbound video should be muted or not. - * @param muted - True to mute the outbound video. - * @returns the new mute state - */ - public async setLocalVideoMuted(muted: boolean): Promise<boolean> { - logger.log(`Call ${this.callId} setLocalVideoMuted() running ${muted}`); - - // if we were still thinking about stopping and removing the video - // track: don't, because we want it back. - if (!muted && this.stopVideoTrackTimer !== undefined) { - clearTimeout(this.stopVideoTrackTimer); - this.stopVideoTrackTimer = undefined; - } - - if (!(await this.client.getMediaHandler().hasVideoDevice())) { - return this.isLocalVideoMuted(); - } - - if (!this.hasUserMediaVideoSender && !muted) { - this.localUsermediaFeed?.setAudioVideoMuted(null, muted); - await this.upgradeCall(false, true); - return this.isLocalVideoMuted(); - } - - // we may not have a video track - if not, re-request usermedia - if (!muted && this.localUsermediaStream!.getVideoTracks().length === 0) { - const stream = await this.client.getMediaHandler().getUserMediaStream(true, true); - await this.updateLocalUsermediaStream(stream); - } - - this.localUsermediaFeed?.setAudioVideoMuted(null, muted); - - this.updateMuteStatus(); - await this.sendMetadataUpdate(); - - // if we're muting video, set a timeout to stop & remove the video track so we release - // the camera. We wait a short time to do this because when we disable a track, WebRTC - // will send black video for it. If we just stop and remove it straight away, the video - // will just freeze which means that when we unmute video, the other side will briefly - // get a static frame of us from before we muted. This way, the still frame is just black. - // A very small delay is not always enough so the theory here is that it needs to be long - // enough for WebRTC to encode a frame: 120ms should be long enough even if we're only - // doing 10fps. - if (muted) { - this.stopVideoTrackTimer = setTimeout(() => { - for (const t of this.localUsermediaStream!.getVideoTracks()) { - t.stop(); - this.localUsermediaStream!.removeTrack(t); - } - }, 120); - } - - return this.isLocalVideoMuted(); - } - - /** - * Check if local video is muted. - * - * If there are multiple video tracks, <i>all</i> of the tracks need to be muted - * for this to return true. This means if there are no video tracks, this will - * return true. - * @returns True if the local preview video is muted, else false - * (including if the call is not set up yet). - */ - public isLocalVideoMuted(): boolean { - return this.localUsermediaFeed?.isVideoMuted() ?? false; - } - - /** - * Set whether the microphone should be muted or not. - * @param muted - True to mute the mic. - * @returns the new mute state - */ - public async setMicrophoneMuted(muted: boolean): Promise<boolean> { - logger.log(`Call ${this.callId} setMicrophoneMuted() running ${muted}`); - if (!(await this.client.getMediaHandler().hasAudioDevice())) { - return this.isMicrophoneMuted(); - } - - if (!muted && (!this.hasUserMediaAudioSender || !this.hasLocalUserMediaAudioTrack)) { - await this.upgradeCall(true, false); - return this.isMicrophoneMuted(); - } - this.localUsermediaFeed?.setAudioVideoMuted(muted, null); - this.updateMuteStatus(); - await this.sendMetadataUpdate(); - return this.isMicrophoneMuted(); - } - - /** - * Check if the microphone is muted. - * - * If there are multiple audio tracks, <i>all</i> of the tracks need to be muted - * for this to return true. This means if there are no audio tracks, this will - * return true. - * @returns True if the mic is muted, else false (including if the call - * is not set up yet). - */ - public isMicrophoneMuted(): boolean { - return this.localUsermediaFeed?.isAudioMuted() ?? false; - } - - /** - * @returns true if we have put the party on the other side of the call on hold - * (that is, we are signalling to them that we are not listening) - */ - public isRemoteOnHold(): boolean { - return this.remoteOnHold; - } - - public setRemoteOnHold(onHold: boolean): void { - if (this.isRemoteOnHold() === onHold) return; - this.remoteOnHold = onHold; - - for (const transceiver of this.peerConn!.getTransceivers()) { - // We don't send hold music or anything so we're not actually - // sending anything, but sendrecv is fairly standard for hold and - // it makes it a lot easier to figure out who's put who on hold. - transceiver.direction = onHold ? "sendonly" : "sendrecv"; - } - this.updateMuteStatus(); - this.sendMetadataUpdate(); - - this.emit(CallEvent.RemoteHoldUnhold, this.remoteOnHold, this); - } - - /** - * Indicates whether we are 'on hold' to the remote party (ie. if true, - * they cannot hear us). - * @returns true if the other party has put us on hold - */ - public isLocalOnHold(): boolean { - if (this.state !== CallState.Connected) return false; - - let callOnHold = true; - - // We consider a call to be on hold only if *all* the tracks are on hold - // (is this the right thing to do?) - for (const transceiver of this.peerConn!.getTransceivers()) { - const trackOnHold = ["inactive", "recvonly"].includes(transceiver.currentDirection!); - - if (!trackOnHold) callOnHold = false; - } - - return callOnHold; - } - - /** - * Sends a DTMF digit to the other party - * @param digit - The digit (nb. string - '#' and '*' are dtmf too) - */ - public sendDtmfDigit(digit: string): void { - for (const sender of this.peerConn!.getSenders()) { - if (sender.track?.kind === "audio" && sender.dtmf) { - sender.dtmf.insertDTMF(digit); - return; - } - } - - throw new Error("Unable to find a track to send DTMF on"); - } - - private updateMuteStatus(): void { - const micShouldBeMuted = this.isMicrophoneMuted() || this.remoteOnHold; - const vidShouldBeMuted = this.isLocalVideoMuted() || this.remoteOnHold; - - logger.log( - `Call ${this.callId} updateMuteStatus stream ${ - this.localUsermediaStream!.id - } micShouldBeMuted ${micShouldBeMuted} vidShouldBeMuted ${vidShouldBeMuted}`, - ); - - setTracksEnabled(this.localUsermediaStream!.getAudioTracks(), !micShouldBeMuted); - setTracksEnabled(this.localUsermediaStream!.getVideoTracks(), !vidShouldBeMuted); - } - - public async sendMetadataUpdate(): Promise<void> { - await this.sendVoipEvent(EventType.CallSDPStreamMetadataChangedPrefix, { - [SDPStreamMetadataKey]: this.getLocalSDPStreamMetadata(), - }); - } - - private gotCallFeedsForInvite(callFeeds: CallFeed[], requestScreenshareFeed = false): void { - if (this.successor) { - this.successor.queueGotCallFeedsForAnswer(callFeeds); - return; - } - if (this.callHasEnded()) { - this.stopAllMedia(); - return; - } - - for (const feed of callFeeds) { - this.pushLocalFeed(feed); - } - - if (requestScreenshareFeed) { - this.peerConn!.addTransceiver("video", { - direction: "recvonly", - }); - } - - this.state = CallState.CreateOffer; - - logger.debug(`Call ${this.callId} gotUserMediaForInvite() run`); - // Now we wait for the negotiationneeded event - } - - private async sendAnswer(): Promise<void> { - const answerContent = { - answer: { - sdp: this.peerConn!.localDescription!.sdp, - // type is now deprecated as of Matrix VoIP v1, but - // required to still be sent for backwards compat - type: this.peerConn!.localDescription!.type, - }, - [SDPStreamMetadataKey]: this.getLocalSDPStreamMetadata(true), - } as MCallAnswer; - - answerContent.capabilities = { - "m.call.transferee": this.client.supportsCallTransfer, - "m.call.dtmf": false, - }; - - // We have just taken the local description from the peerConn which will - // contain all the local candidates added so far, so we can discard any candidates - // we had queued up because they'll be in the answer. - const discardCount = this.discardDuplicateCandidates(); - logger.info( - `Call ${this.callId} sendAnswer() discarding ${discardCount} candidates that will be sent in answer`, - ); - - try { - await this.sendVoipEvent(EventType.CallAnswer, answerContent); - // If this isn't the first time we've tried to send the answer, - // we may have candidates queued up, so send them now. - this.inviteOrAnswerSent = true; - } catch (error) { - // We've failed to answer: back to the ringing state - this.state = CallState.Ringing; - if (error instanceof MatrixError && error.event) this.client.cancelPendingEvent(error.event); - - let code = CallErrorCode.SendAnswer; - let message = "Failed to send answer"; - if ((<Error>error).name == "UnknownDeviceError") { - code = CallErrorCode.UnknownDevices; - message = "Unknown devices present in the room"; - } - this.emit(CallEvent.Error, new CallError(code, message, <Error>error), this); - throw error; - } - - // error handler re-throws so this won't happen on error, but - // we don't want the same error handling on the candidate queue - this.sendCandidateQueue(); - } - - private queueGotCallFeedsForAnswer(callFeeds: CallFeed[]): void { - // Ensure only one negotiate/answer event is being processed at a time. - if (this.responsePromiseChain) { - this.responsePromiseChain = this.responsePromiseChain.then(() => this.gotCallFeedsForAnswer(callFeeds)); - } else { - this.responsePromiseChain = this.gotCallFeedsForAnswer(callFeeds); - } - } - - // Enables DTX (discontinuous transmission) on the given session to reduce - // bandwidth when transmitting silence - private mungeSdp(description: RTCSessionDescriptionInit, mods: CodecParamsMod[]): void { - // The only way to enable DTX at this time is through SDP munging - const sdp = parseSdp(description.sdp!); - - sdp.media.forEach((media) => { - const payloadTypeToCodecMap = new Map<number, string>(); - const codecToPayloadTypeMap = new Map<string, number>(); - for (const rtp of media.rtp) { - payloadTypeToCodecMap.set(rtp.payload, rtp.codec); - codecToPayloadTypeMap.set(rtp.codec, rtp.payload); - } - - for (const mod of mods) { - if (mod.mediaType !== media.type) continue; - - if (!codecToPayloadTypeMap.has(mod.codec)) { - logger.info( - `Call ${this.callId} mungeSdp() ignoring SDP modifications for ${mod.codec} as it's not present.`, - ); - continue; - } - - const extraConfig: string[] = []; - if (mod.enableDtx !== undefined) { - extraConfig.push(`usedtx=${mod.enableDtx ? "1" : "0"}`); - } - if (mod.maxAverageBitrate !== undefined) { - extraConfig.push(`maxaveragebitrate=${mod.maxAverageBitrate}`); - } - - let found = false; - for (const fmtp of media.fmtp) { - if (payloadTypeToCodecMap.get(fmtp.payload) === mod.codec) { - found = true; - fmtp.config += ";" + extraConfig.join(";"); - } - } - if (!found) { - media.fmtp.push({ - payload: codecToPayloadTypeMap.get(mod.codec)!, - config: extraConfig.join(";"), - }); - } - } - }); - description.sdp = writeSdp(sdp); - } - - private async createOffer(): Promise<RTCSessionDescriptionInit> { - const offer = await this.peerConn!.createOffer(); - this.mungeSdp(offer, getCodecParamMods(this.isPtt)); - return offer; - } - - private async createAnswer(): Promise<RTCSessionDescriptionInit> { - const answer = await this.peerConn!.createAnswer(); - this.mungeSdp(answer, getCodecParamMods(this.isPtt)); - return answer; - } - - private async gotCallFeedsForAnswer(callFeeds: CallFeed[]): Promise<void> { - if (this.callHasEnded()) return; - - this.waitForLocalAVStream = false; - - for (const feed of callFeeds) { - this.pushLocalFeed(feed); - } - - this.state = CallState.CreateAnswer; - - let answer: RTCSessionDescriptionInit; - try { - this.getRidOfRTXCodecs(); - answer = await this.createAnswer(); - } catch (err) { - logger.debug(`Call ${this.callId} gotCallFeedsForAnswer() failed to create answer: `, err); - this.terminate(CallParty.Local, CallErrorCode.CreateAnswer, true); - return; - } - - try { - await this.peerConn!.setLocalDescription(answer); - - // make sure we're still going - if (this.callHasEnded()) return; - - this.state = CallState.Connecting; - - // Allow a short time for initial candidates to be gathered - await new Promise((resolve) => { - setTimeout(resolve, 200); - }); - - // make sure the call hasn't ended before we continue - if (this.callHasEnded()) return; - - this.sendAnswer(); - } catch (err) { - logger.debug(`Call ${this.callId} gotCallFeedsForAnswer() error setting local description!`, err); - this.terminate(CallParty.Local, CallErrorCode.SetLocalDescription, true); - return; - } - } - - /** - * Internal - */ - private gotLocalIceCandidate = (event: RTCPeerConnectionIceEvent): void => { - if (event.candidate) { - if (this.candidatesEnded) { - logger.warn( - `Call ${this.callId} gotLocalIceCandidate() got candidate after candidates have ended - ignoring!`, - ); - return; - } - - 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); - } - }; - - public async onRemoteIceCandidatesReceived(ev: MatrixEvent): Promise<void> { - if (this.callHasEnded()) { - //debuglog("Ignoring remote ICE candidate because call has ended"); - return; - } - - const content = ev.getContent<MCallCandidates>(); - const candidates = content.candidates; - if (!candidates) { - logger.info( - `Call ${this.callId} onRemoteIceCandidatesReceived() ignoring candidates event with no candidates!`, - ); - return; - } - - const fromPartyId = content.version === 0 ? null : content.party_id || null; - - if (this.opponentPartyId === undefined) { - // we haven't picked an opponent yet so save the candidates - if (fromPartyId) { - logger.info( - `Call ${this.callId} onRemoteIceCandidatesReceived() buffering ${candidates.length} candidates until we pick an opponent`, - ); - const bufferedCandidates = this.remoteCandidateBuffer.get(fromPartyId) || []; - bufferedCandidates.push(...candidates); - this.remoteCandidateBuffer.set(fromPartyId, bufferedCandidates); - } - return; - } - - if (!this.partyIdMatches(content)) { - logger.info( - `Call ${this.callId} onRemoteIceCandidatesReceived() ignoring candidates from party ID ${content.party_id}: we have chosen party ID ${this.opponentPartyId}`, - ); - - return; - } - - await this.addIceCandidates(candidates); - } - - /** - * Used by MatrixClient. - */ - public async onAnswerReceived(event: MatrixEvent): Promise<void> { - const content = event.getContent<MCallAnswer>(); - logger.debug(`Call ${this.callId} onAnswerReceived() running (hangupParty=${content.party_id})`); - - if (this.callHasEnded()) { - logger.debug(`Call ${this.callId} onAnswerReceived() ignoring answer because call has ended`); - return; - } - - if (this.opponentPartyId !== undefined) { - logger.info( - `Call ${this.callId} onAnswerReceived() ignoring answer from party ID ${content.party_id}: we already have an answer/reject from ${this.opponentPartyId}`, - ); - return; - } - - this.chooseOpponent(event); - await this.addBufferedIceCandidates(); - - this.state = CallState.Connecting; - - const sdpStreamMetadata = content[SDPStreamMetadataKey]; - if (sdpStreamMetadata) { - this.updateRemoteSDPStreamMetadata(sdpStreamMetadata); - } else { - logger.warn( - `Call ${this.callId} onAnswerReceived() did not get any SDPStreamMetadata! Can not send/receive multiple streams`, - ); - } - - try { - await this.peerConn!.setRemoteDescription(content.answer); - } catch (e) { - logger.debug(`Call ${this.callId} onAnswerReceived() failed to set remote description`, e); - this.terminate(CallParty.Local, CallErrorCode.SetRemoteDescription, false); - return; - } - - // If the answer we selected has a party_id, send a select_answer event - // We do this after setting the remote description since otherwise we'd block - // call setup on it - if (this.opponentPartyId !== null) { - try { - await this.sendVoipEvent(EventType.CallSelectAnswer, { - selected_party_id: this.opponentPartyId, - }); - } catch (err) { - // This isn't fatal, and will just mean that if another party has raced to answer - // the call, they won't know they got rejected, so we carry on & don't retry. - logger.warn(`Call ${this.callId} onAnswerReceived() failed to send select_answer event`, err); - } - } - } - - public async onSelectAnswerReceived(event: MatrixEvent): Promise<void> { - if (this.direction !== CallDirection.Inbound) { - logger.warn( - `Call ${this.callId} onSelectAnswerReceived() got select_answer for an outbound call: ignoring`, - ); - return; - } - - const selectedPartyId = event.getContent<MCallSelectAnswer>().selected_party_id; - - if (selectedPartyId === undefined || selectedPartyId === null) { - logger.warn( - `Call ${this.callId} onSelectAnswerReceived() got nonsensical select_answer with null/undefined selected_party_id: ignoring`, - ); - return; - } - - if (selectedPartyId !== this.ourPartyId) { - logger.info( - `Call ${this.callId} onSelectAnswerReceived() got select_answer for party ID ${selectedPartyId}: we are party ID ${this.ourPartyId}.`, - ); - // The other party has picked somebody else's answer - await this.terminate(CallParty.Remote, CallErrorCode.AnsweredElsewhere, true); - } - } - - public async onNegotiateReceived(event: MatrixEvent): Promise<void> { - const content = event.getContent<MCallInviteNegotiate>(); - const description = content.description; - if (!description || !description.sdp || !description.type) { - logger.info(`Call ${this.callId} onNegotiateReceived() ignoring invalid m.call.negotiate event`); - return; - } - // Politeness always follows the direction of the call: in a glare situation, - // we pick either the inbound or outbound call, so one side will always be - // inbound and one outbound - const polite = this.direction === CallDirection.Inbound; - - // Here we follow the perfect negotiation logic from - // https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/Perfect_negotiation - const offerCollision = - description.type === "offer" && (this.makingOffer || this.peerConn!.signalingState !== "stable"); - - this.ignoreOffer = !polite && offerCollision; - if (this.ignoreOffer) { - logger.info( - `Call ${this.callId} onNegotiateReceived() ignoring colliding negotiate event because we're impolite`, - ); - return; - } - - const prevLocalOnHold = this.isLocalOnHold(); - - const sdpStreamMetadata = content[SDPStreamMetadataKey]; - if (sdpStreamMetadata) { - this.updateRemoteSDPStreamMetadata(sdpStreamMetadata); - } else { - logger.warn( - `Call ${this.callId} onNegotiateReceived() received negotiation event without SDPStreamMetadata!`, - ); - } - - try { - await this.peerConn!.setRemoteDescription(description); - - 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); - - this.sendVoipEvent(EventType.CallNegotiate, { - description: this.peerConn!.localDescription?.toJSON(), - [SDPStreamMetadataKey]: this.getLocalSDPStreamMetadata(true), - }); - } - } catch (err) { - 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 = utils.recursivelyAssign(this.remoteSDPStreamMetadata || {}, metadata, true); - for (const feed of this.getRemoteFeeds()) { - const streamId = feed.stream.id; - const metadata = this.remoteSDPStreamMetadata![streamId]; - - feed.setAudioVideoMuted(metadata?.audio_muted, metadata?.video_muted); - feed.purpose = this.remoteSDPStreamMetadata![streamId]?.purpose; - } - } - - public onSDPStreamMetadataChangedReceived(event: MatrixEvent): void { - const content = event.getContent<MCallSDPStreamMetadataChanged>(); - const metadata = content[SDPStreamMetadataKey]; - this.updateRemoteSDPStreamMetadata(metadata); - } - - public async onAssertedIdentityReceived(event: MatrixEvent): Promise<void> { - const content = event.getContent<MCAllAssertedIdentity>(); - if (!content.asserted_identity) return; - - this.remoteAssertedIdentity = { - id: content.asserted_identity.id, - displayName: content.asserted_identity.display_name, - }; - this.emit(CallEvent.AssertedIdentityChanged, this); - } - - public callHasEnded(): boolean { - // This exists as workaround to typescript trying to be clever and erroring - // when putting if (this.state === CallState.Ended) return; twice in the same - // function, even though that function is async. - return this.state === CallState.Ended; - } - - private queueGotLocalOffer(): void { - // Ensure only one negotiate/answer event is being processed at a time. - if (this.responsePromiseChain) { - this.responsePromiseChain = this.responsePromiseChain.then(() => this.wrappedGotLocalOffer()); - } else { - this.responsePromiseChain = this.wrappedGotLocalOffer(); - } - } - - private async wrappedGotLocalOffer(): Promise<void> { - this.makingOffer = true; - try { - // XXX: in what situations do we believe gotLocalOffer actually throws? It appears - // to handle most of its exceptions itself and terminate the call. I'm not entirely - // sure it would ever throw, so I can't add a test for these lines. - // Also the tense is different between "gotLocalOffer" and "getLocalOfferFailed" so - // it's not entirely clear whether getLocalOfferFailed is just misnamed or whether - // they've been cross-polinated somehow at some point. - await this.gotLocalOffer(); - } catch (e) { - this.getLocalOfferFailed(e as Error); - return; - } finally { - this.makingOffer = false; - } - } - - private async gotLocalOffer(): Promise<void> { - logger.debug(`Call ${this.callId} gotLocalOffer() running`); - - if (this.callHasEnded()) { - logger.debug( - `Call ${this.callId} gotLocalOffer() ignoring newly created offer because the call has ended"`, - ); - return; - } - - let offer: RTCSessionDescriptionInit; - try { - this.getRidOfRTXCodecs(); - offer = await this.createOffer(); - } catch (err) { - logger.debug(`Call ${this.callId} gotLocalOffer() failed to create offer: `, err); - this.terminate(CallParty.Local, CallErrorCode.CreateOffer, true); - return; - } - - try { - await this.peerConn!.setLocalDescription(offer); - } catch (err) { - logger.debug(`Call ${this.callId} gotLocalOffer() error setting local description!`, err); - this.terminate(CallParty.Local, CallErrorCode.SetLocalDescription, true); - return; - } - - if (this.peerConn!.iceGatheringState === "gathering") { - // Allow a short time for initial candidates to be gathered - await new Promise((resolve) => { - setTimeout(resolve, 200); - }); - } - - if (this.callHasEnded()) return; - - const eventType = this.state === CallState.CreateOffer ? EventType.CallInvite : EventType.CallNegotiate; - - const content = { - lifetime: CALL_TIMEOUT_MS, - } as MCallInviteNegotiate; - - if (eventType === EventType.CallInvite && this.invitee) { - content.invitee = this.invitee; - } - - // clunky because TypeScript can't follow the types through if we use an expression as the key - if (this.state === CallState.CreateOffer) { - content.offer = this.peerConn!.localDescription?.toJSON(); - } else { - content.description = this.peerConn!.localDescription?.toJSON(); - } - - content.capabilities = { - "m.call.transferee": this.client.supportsCallTransfer, - "m.call.dtmf": false, - }; - - content[SDPStreamMetadataKey] = this.getLocalSDPStreamMetadata(true); - - // Get rid of any candidates waiting to be sent: they'll be included in the local - // description we just got and will send in the offer. - const discardCount = this.discardDuplicateCandidates(); - logger.info( - `Call ${this.callId} gotLocalOffer() discarding ${discardCount} candidates that will be sent in offer`, - ); - - try { - await this.sendVoipEvent(eventType, content); - } catch (error) { - logger.error(`Call ${this.callId} gotLocalOffer() failed to send invite`, error); - if (error instanceof MatrixError && error.event) this.client.cancelPendingEvent(error.event); - - let code = CallErrorCode.SignallingFailed; - let message = "Signalling failed"; - if (this.state === CallState.CreateOffer) { - code = CallErrorCode.SendInvite; - message = "Failed to send invite"; - } - if ((<Error>error).name == "UnknownDeviceError") { - code = CallErrorCode.UnknownDevices; - message = "Unknown devices present in the room"; - } - - this.emit(CallEvent.Error, new CallError(code, message, <Error>error), this); - this.terminate(CallParty.Local, code, false); - - // no need to carry on & send the candidate queue, but we also - // don't want to rethrow the error - return; - } - - this.sendCandidateQueue(); - if (this.state === CallState.CreateOffer) { - this.inviteOrAnswerSent = true; - this.state = CallState.InviteSent; - this.inviteTimeout = setTimeout(() => { - this.inviteTimeout = undefined; - if (this.state === CallState.InviteSent) { - this.hangup(CallErrorCode.InviteTimeout, false); - } - }, CALL_TIMEOUT_MS); - } - } - - private getLocalOfferFailed = (err: Error): void => { - logger.error(`Call ${this.callId} getLocalOfferFailed() running`, err); - - this.emit( - CallEvent.Error, - new CallError(CallErrorCode.LocalOfferFailed, "Failed to get local offer!", err), - this, - ); - this.terminate(CallParty.Local, CallErrorCode.LocalOfferFailed, false); - }; - - private getUserMediaFailed = (err: Error): void => { - if (this.successor) { - this.successor.getUserMediaFailed(err); - return; - } - - logger.warn(`Call ${this.callId} getUserMediaFailed() failed to get user media - ending call`, err); - - this.emit( - CallEvent.Error, - new CallError( - CallErrorCode.NoUserMedia, - "Couldn't start capturing media! Is your microphone set up and " + "does this app have permission?", - err, - ), - this, - ); - this.terminate(CallParty.Local, CallErrorCode.NoUserMedia, false); - }; - - private 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})`, - ); - - // 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; - 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") { - // 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; - 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.iceDisconnectedTimeout = setTimeout(() => { - 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 - if (!RTCRtpReceiver.getCapabilities || !RTCRtpSender.getCapabilities) return; - - const recvCodecs = RTCRtpReceiver.getCapabilities("video")!.codecs; - const sendCodecs = RTCRtpSender.getCapabilities("video")!.codecs; - const codecs = [...sendCodecs, ...recvCodecs]; - - for (const codec of codecs) { - if (codec.mimeType === "video/rtx") { - const rtxCodecIndex = codecs.indexOf(codec); - codecs.splice(rtxCodecIndex, 1); - } - } - - const screenshareVideoTransceiver = this.transceivers.get( - getTransceiverKey(SDPStreamMetadataPurpose.Screenshare, "video"), - ); - if (screenshareVideoTransceiver) screenshareVideoTransceiver.setCodecPreferences(codecs); - } - - 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(eventType: string, content: object): Promise<void> { - const realContent = Object.assign({}, content, { - version: VOIP_PROTO_VERSION, - call_id: this.callId, - party_id: this.ourPartyId, - conf_id: this.groupCallId, - }); - - if (this.opponentDeviceId) { - const toDeviceSeq = this.toDeviceSeq++; - const content = { - ...realContent, - device_id: this.client.deviceId, - sender_session_id: this.client.getSessionId(), - dest_session_id: this.opponentSessionId, - seq: toDeviceSeq, - [ToDeviceMessageId]: uuidv4(), - }; - - this.emit( - CallEvent.SendVoipEvent, - { - type: "toDevice", - eventType, - userId: this.invitee || this.getOpponentMember()?.userId, - opponentDeviceId: this.opponentDeviceId, - content, - }, - this, - ); - - const userId = this.invitee || this.getOpponentMember()!.userId; - if (this.client.getUseE2eForGroupCall()) { - if (!this.opponentDeviceInfo) { - logger.warn(`Call ${this.callId} sendVoipEvent() failed: we do not have opponentDeviceInfo`); - return; - } - - await this.client.encryptAndSendToDevices( - [ - { - userId, - deviceInfo: this.opponentDeviceInfo, - }, - ], - { - type: eventType, - content, - }, - ); - } else { - await this.client.sendToDevice( - eventType, - new Map<string, any>([[userId, new Map([[this.opponentDeviceId, content]])]]), - ); - } - } else { - this.emit( - CallEvent.SendVoipEvent, - { - type: "sendEvent", - eventType, - roomId: this.roomId, - content: realContent, - userId: this.invitee || this.getOpponentMember()?.userId, - }, - this, - ); - - await this.client.sendEvent(this.roomId!, eventType, realContent); - } - } - - /** - * Queue a candidate to be sent - * @param content - The candidate to queue up, or null if candidates have finished being generated - * and end-of-candidates should be signalled - */ - private queueCandidate(content: RTCIceCandidate | null): void { - // We partially de-trickle candidates by waiting for `delay` before sending them - // amalgamated, in order to avoid sending too many m.call.candidates events and hitting - // rate limits in Matrix. - // In practice, it'd be better to remove rate limits for m.call.* - - // N.B. this deliberately lets you queue and send blank candidates, which MSC2746 - // currently proposes as the way to indicate that candidate gathering is complete. - // This will hopefully be changed to an explicit rather than implicit notification - // shortly. - if (content) { - this.candidateSendQueue.push(content); - } else { - this.candidatesEnded = true; - } - - // Don't send the ICE candidates yet if the call is in the ringing state: this - // means we tried to pick (ie. started generating candidates) and then failed to - // send the answer and went back to the ringing state. Queue up the candidates - // to send if we successfully send the answer. - // Equally don't send if we haven't yet sent the answer because we can send the - // first batch of candidates along with the answer - if (this.state === CallState.Ringing || !this.inviteOrAnswerSent) return; - - // MSC2746 recommends these values (can be quite long when calling because the - // callee will need a while to answer the call) - const delay = this.direction === CallDirection.Inbound ? 500 : 2000; - - if (this.candidateSendTries === 0) { - setTimeout(() => { - this.sendCandidateQueue(); - }, delay); - } - } - - // Discard all non-end-of-candidates messages - // Return the number of candidate messages that were discarded. - // Call this method before sending an invite or answer message - private discardDuplicateCandidates(): number { - let discardCount = 0; - const newQueue: RTCIceCandidate[] = []; - - for (let i = 0; i < this.candidateSendQueue.length; i++) { - const candidate = this.candidateSendQueue[i]; - if (candidate.candidate === "") { - newQueue.push(candidate); - } else { - discardCount++; - } - } - - this.candidateSendQueue = newQueue; - - return discardCount; - } - - /* - * Transfers this call to another user - */ - public async transfer(targetUserId: string): Promise<void> { - // Fetch the target user's global profile info: their room avatar / displayname - // could be different in whatever room we share with them. - const profileInfo = await this.client.getProfileInfo(targetUserId); - - const replacementId = genCallID(); - - const body = { - replacement_id: genCallID(), - target_user: { - id: targetUserId, - display_name: profileInfo.displayname, - avatar_url: profileInfo.avatar_url, - }, - create_call: replacementId, - } as MCallReplacesEvent; - - await this.sendVoipEvent(EventType.CallReplaces, body); - - await this.terminate(CallParty.Local, CallErrorCode.Transferred, true); - } - - /* - * Transfers this call to the target call, effectively 'joining' the - * two calls (so the remote parties on each call are connected together). - */ - public async transferToCall(transferTargetCall: MatrixCall): Promise<void> { - const targetUserId = transferTargetCall.getOpponentMember()?.userId; - const targetProfileInfo = targetUserId ? await this.client.getProfileInfo(targetUserId) : undefined; - const opponentUserId = this.getOpponentMember()?.userId; - const transfereeProfileInfo = opponentUserId ? await this.client.getProfileInfo(opponentUserId) : undefined; - - const newCallId = genCallID(); - - const bodyToTransferTarget = { - // the replacements on each side have their own ID, and it's distinct from the - // ID of the new call (but we can use the same function to generate it) - replacement_id: genCallID(), - target_user: { - id: opponentUserId, - display_name: transfereeProfileInfo?.displayname, - avatar_url: transfereeProfileInfo?.avatar_url, - }, - await_call: newCallId, - } as MCallReplacesEvent; - - await transferTargetCall.sendVoipEvent(EventType.CallReplaces, bodyToTransferTarget); - - const bodyToTransferee = { - replacement_id: genCallID(), - target_user: { - id: targetUserId, - display_name: targetProfileInfo?.displayname, - avatar_url: targetProfileInfo?.avatar_url, - }, - create_call: newCallId, - } as MCallReplacesEvent; - - await this.sendVoipEvent(EventType.CallReplaces, bodyToTransferee); - - await this.terminate(CallParty.Local, CallErrorCode.Transferred, true); - await transferTargetCall.terminate(CallParty.Local, CallErrorCode.Transferred, true); - } - - private async terminate(hangupParty: CallParty, hangupReason: CallErrorCode, shouldEmit: boolean): Promise<void> { - if (this.callHasEnded()) return; - - this.hangupParty = hangupParty; - this.hangupReason = hangupReason; - this.state = CallState.Ended; - - if (this.inviteTimeout) { - clearTimeout(this.inviteTimeout); - this.inviteTimeout = undefined; - } - if (this.iceDisconnectedTimeout !== undefined) { - clearTimeout(this.iceDisconnectedTimeout); - this.iceDisconnectedTimeout = undefined; - } - if (this.callLengthInterval) { - clearInterval(this.callLengthInterval); - this.callLengthInterval = undefined; - } - if (this.stopVideoTrackTimer !== undefined) { - clearTimeout(this.stopVideoTrackTimer); - this.stopVideoTrackTimer = undefined; - } - - for (const [stream, listener] of this.removeTrackListeners) { - stream.removeEventListener("removetrack", listener); - } - this.removeTrackListeners.clear(); - - this.callStatsAtEnd = await this.collectCallStats(); - - // Order is important here: first we stopAllMedia() and only then we can deleteAllFeeds() - this.stopAllMedia(); - this.deleteAllFeeds(); - - if (this.peerConn && this.peerConn.signalingState !== "closed") { - this.peerConn.close(); - } - this.stats?.removeStatsReportGatherer(this.callId); - - if (shouldEmit) { - this.emit(CallEvent.Hangup, this); - } - - this.client.callEventHandler!.calls.delete(this.callId); - } - - private stopAllMedia(): void { - logger.debug(`Call ${this.callId} stopAllMedia() running`); - - for (const feed of this.feeds) { - // Slightly awkward as local feed need to go via the correct method on - // the MediaHandler so they get removed from MediaHandler (remote tracks - // don't) - // NB. We clone local streams when passing them to individual calls in a group - // call, so we can (and should) stop the clones once we no longer need them: - // the other clones will continue fine. - if (feed.isLocal() && feed.purpose === SDPStreamMetadataPurpose.Usermedia) { - this.client.getMediaHandler().stopUserMediaStream(feed.stream); - } else if (feed.isLocal() && feed.purpose === SDPStreamMetadataPurpose.Screenshare) { - this.client.getMediaHandler().stopScreensharingStream(feed.stream); - } else if (!feed.isLocal()) { - logger.debug(`Call ${this.callId} stopAllMedia() stopping stream (streamId=${feed.stream.id})`); - for (const track of feed.stream.getTracks()) { - track.stop(); - } - } - } - } - - private checkForErrorListener(): void { - if (this.listeners(EventEmitterEvents.Error).length === 0) { - throw new Error("You MUST attach an error listener using call.on('error', function() {})"); - } - } - - private async sendCandidateQueue(): Promise<void> { - if (this.candidateSendQueue.length === 0 || this.callHasEnded()) { - return; - } - - const candidates = this.candidateSendQueue; - this.candidateSendQueue = []; - ++this.candidateSendTries; - const content = { candidates: candidates.map((candidate) => candidate.toJSON()) }; - if (this.candidatesEnded) { - // If there are no more candidates, signal this by adding an empty string candidate - content.candidates.push({ - candidate: "", - }); - } - logger.debug(`Call ${this.callId} sendCandidateQueue() attempting to send ${candidates.length} candidates`); - try { - await this.sendVoipEvent(EventType.CallCandidates, content); - // reset our retry count if we have successfully sent our candidates - // otherwise queueCandidate() will refuse to try to flush the queue - this.candidateSendTries = 0; - - // Try to send candidates again just in case we received more candidates while sending. - this.sendCandidateQueue(); - } catch (error) { - // don't retry this event: we'll send another one later as we might - // have more candidates by then. - if (error instanceof MatrixError && error.event) this.client.cancelPendingEvent(error.event); - - // put all the candidates we failed to send back in the queue - this.candidateSendQueue.push(...candidates); - - if (this.candidateSendTries > 5) { - logger.debug( - `Call ${this.callId} sendCandidateQueue() failed to send candidates on attempt ${this.candidateSendTries}. Giving up on this call.`, - error, - ); - - const code = CallErrorCode.SignallingFailed; - const message = "Signalling failed"; - - this.emit(CallEvent.Error, new CallError(code, message, <Error>error), this); - this.hangup(code, false); - - return; - } - - const delayMs = 500 * Math.pow(2, this.candidateSendTries); - ++this.candidateSendTries; - logger.debug( - `Call ${this.callId} sendCandidateQueue() failed to send candidates. Retrying in ${delayMs}ms`, - error, - ); - setTimeout(() => { - this.sendCandidateQueue(); - }, delayMs); - } - } - - /** - * Place a call to this room. - * @throws if you have not specified a listener for 'error' events. - * @throws if have passed audio=false. - */ - public async placeCall(audio: boolean, video: boolean): Promise<void> { - if (!audio) { - throw new Error("You CANNOT start a call without audio"); - } - this.state = CallState.WaitLocalMedia; - - 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); - - const 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, - }); - await this.placeCallWithCallFeeds([callFeed]); - } catch (e) { - this.getUserMediaFailed(<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.gotCallFeedsForInvite(callFeeds, requestScreenshareFeed); - } - - private createPeerConnection(): RTCPeerConnection { - const pc = new window.RTCPeerConnection({ - iceTransportPolicy: this.forceTURN ? "relay" : undefined, - iceServers: this.turnServers, - iceCandidatePoolSize: this.client.iceCandidatePoolSize, - bundlePolicy: "max-bundle", - }); - - // 'connectionstatechange' would be better, but firefox doesn't implement that. - pc.addEventListener("iceconnectionstatechange", this.onIceConnectionStateChanged); - pc.addEventListener("signalingstatechange", this.onSignallingStateChanged); - pc.addEventListener("icecandidate", this.gotLocalIceCandidate); - pc.addEventListener("icegatheringstatechange", this.onIceGatheringStateChange); - pc.addEventListener("track", this.onTrack); - pc.addEventListener("negotiationneeded", this.onNegotiationNeeded); - pc.addEventListener("datachannel", this.onDataChannel); - - this.stats?.addStatsReportGatherer(this.callId, "unknown", pc); - return pc; - } - - private partyIdMatches(msg: MCallBase): boolean { - // They must either match or both be absent (in which case opponentPartyId will be null) - // Also we ignore party IDs on the invite/offer if the version is 0, so we must do the same - // here and use null if the version is 0 (woe betide any opponent sending messages in the - // same call with different versions) - const msgPartyId = msg.version === 0 ? null : msg.party_id || null; - return msgPartyId === this.opponentPartyId; - } - - // Commits to an opponent for the call - // ev: An invite or answer event - private chooseOpponent(ev: MatrixEvent): void { - // I choo-choo-choose you - const msg = ev.getContent<MCallInviteNegotiate | MCallAnswer>(); - - logger.debug(`Call ${this.callId} chooseOpponent() running (partyId=${msg.party_id})`); - - this.opponentVersion = msg.version; - if (this.opponentVersion === 0) { - // set to null to indicate that we've chosen an opponent, but because - // they're v0 they have no party ID (even if they sent one, we're ignoring it) - this.opponentPartyId = null; - } else { - // set to their party ID, or if they're naughty and didn't send one despite - // not being v0, set it to null to indicate we picked an opponent with no - // party ID - this.opponentPartyId = msg.party_id || null; - } - this.opponentCaps = msg.capabilities || ({} as CallCapabilities); - this.opponentMember = this.client.getRoom(this.roomId)!.getMember(ev.getSender()!) ?? undefined; - } - - 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[]): 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); - } - } - } - } - - public get hasPeerConnection(): boolean { - return Boolean(this.peerConn); - } - - public initStats(stats: GroupCallStats, peerId = "unknown"): void { - this.stats = stats; - this.stats.start(); - } -} - -export function setTracksEnabled(tracks: Array<MediaStreamTrack>, enabled: boolean): void { - for (const track of tracks) { - track.enabled = enabled; - } -} - -export function supportsMatrixCall(): boolean { - // typeof prevents Node from erroring on an undefined reference - if (typeof window === "undefined" || typeof document === "undefined") { - // NB. We don't log here as apps try to create a call object as a test for - // whether calls are supported, so we shouldn't fill the logs up. - return false; - } - - // Firefox throws on so little as accessing the RTCPeerConnection when operating in a secure mode. - // There's some information at https://bugzilla.mozilla.org/show_bug.cgi?id=1542616 though the concern - // is that the browser throwing a SecurityError will brick the client creation process. - try { - const supported = Boolean( - window.RTCPeerConnection || - window.RTCSessionDescription || - window.RTCIceCandidate || - navigator.mediaDevices, - ); - if (!supported) { - /* istanbul ignore if */ // Adds a lot of noise to test runs, so disable logging there. - if (process.env.NODE_ENV !== "test") { - logger.error("WebRTC is not supported in this browser / environment"); - } - return false; - } - } catch (e) { - logger.error("Exception thrown when trying to access WebRTC", e); - return false; - } - - return true; -} - -/** - * DEPRECATED - * Use client.createCall() - * - * Create a new Matrix call for the browser. - * @param client - The client instance to use. - * @param roomId - The room the call is in. - * @param options - DEPRECATED optional options map. - * @returns the call or null if the browser doesn't support calling. - */ -export function createNewMatrixCall( - client: MatrixClient, - roomId: string, - options?: Pick<CallOpts, "forceTURN" | "invitee" | "opponentDeviceId" | "opponentSessionId" | "groupCallId">, -): MatrixCall | null { - if (!supportsMatrixCall()) return null; - - const optionsForceTURN = options ? options.forceTURN : false; - - const opts: CallOpts = { - client: client, - roomId: roomId, - invitee: options?.invitee, - turnServers: client.getTurnServers(), - // call level options - forceTURN: client.forceTURN || optionsForceTURN, - opponentDeviceId: options?.opponentDeviceId, - opponentSessionId: options?.opponentSessionId, - groupCallId: options?.groupCallId, - }; - const call = new MatrixCall(opts); - - client.reEmitter.reEmit(call, Object.values(CallEvent)); - - return call; -} |