bigbluebutton-Github/bigbluebutton-html5/imports/ui/services/webrtc-base/peer.js
prlanzarin 325887e325 feat(audio): rework audio join without listen only
This is a rework of the audio join procedure whithout the explict listen
only separation in mind. It's supposed to be used in conjunction with
the transparent listen only feature so that the distinction between
modes is seamless with minimal server-side impact. An abridged list of
changes:
  - Let the user pick no input device when joining microphone while
    allowing them to set an input device on the fly later on
  - Give the user the option to join audio with no input device whenever
    we fail to obtain input devices, with the option to try re-enabling
    them on the fly later on
  - Add the option to open the audio settings modal (echo test et al)
    via the in-call device selection chevron
  - Rework the SFU audio bridge and its services to support
    adding/removing tracks on the fly without renegotiation
  - Rework the SFU audio bridge and its services to support a new peer
    role called "passive-sendrecv". That role is used by dupled peers
    that have no active input source on start, but might have one later
    on.
  - Remove stale PermissionsOverlay component from the audio modal
  - Rework how permission errors are detected using the Permissions API
  - Rework the local echo test so that it uses a separate media tag
    rather than the remote
  - Add new, separate dialplans that mute/hold FreeSWITCH channels on
    hold based on UA strings. This is orchestrated server-side via
    webrtc-sfu and akka-apps. The basic difference here is that channels
    now join in their desired state rather than waiting for client side
    observers to sync the state up. It also mitigates transparent listen
    only performance edge cases on multiple audio channels joining at
    the same time.

The old, decoupled listen only mode is still present in code while we
validate this new approach. To test this, transparentListenOnly
must be enabled and listen only mode must be disable on audio join so
that the user skips straight through microphone join.
2024-08-15 00:43:28 +00:00

501 lines
16 KiB
JavaScript

import { EventEmitter2 } from 'eventemitter2';
import {
stopStream,
stopTrack,
silentConsole,
} from '/imports/ui/services/webrtc-base/utils';
export default class WebRtcPeer extends EventEmitter2 {
constructor(mode, options = {}) {
super({ newListener: true });
this.mode = mode;
this.options = options;
this.peerConnection = this.options.peerConnection;
this.videoStream = this.options.videoStream;
this.audioStream = this.options.audioStream;
this.mediaConstraints = this.options.mediaConstraints;
this.trace = this.options.trace;
this.configuration = this.options.configuration;
this.onicecandidate = this.options.onicecandidate;
this.oncandidategatheringdone = this.options.oncandidategatheringdone;
// this.networkPriorities: <{
// audio: <'very-low' | 'low' | 'medium' | 'high' | undefined>
// video: <'very-low' | 'low' | 'medium' | 'high' | undefined>
// } | undefined >
this.networkPriorities = this.options.networkPriorities;
this.candidateGatheringDone = false;
this._outboundCandidateQueue = [];
this._inboundCandidateQueue = [];
this._waitForGatheringPromise = null;
this._waitForGatheringTimeout = null;
this._handleIceCandidate = this._handleIceCandidate.bind(this);
this._handleSignalingStateChange = this._handleSignalingStateChange.bind(this);
this._gatheringTimeout = this.options.gatheringTimeout;
this._assignOverrides();
this.logger.debug('BBB::WebRtcPeer::constructor - created', {
mode: this.mode,
options: this.options,
});
}
_getTransceiverDirection() {
switch (this.mode) {
case 'sendonly':
case 'recvonly':
case 'sendrecv':
return this.mode;
case 'recv':
return 'recvonly';
case 'send':
return 'sendonly';
case 'passive-sendrecv':
return 'sendrecv';
default:
return 'inactive';
}
}
_assignOverrides() {
if (typeof this.onicecandidate === 'function') {
this.on('icecandidate', this.onicecandidate);
}
if (typeof this.oncandidategatheringdone === 'function') {
this.on('candidategatheringdone', this.oncandidategatheringdone);
}
if (typeof this.options.mediaStreamFactory === 'function') {
this._mediaStreamFactory = this.options.mediaStreamFactory.bind(this);
}
}
_processEncodingOptions() {
this.peerConnection?.getSenders().forEach((sender) => {
const { track } = sender;
if (track) {
// TODO: this is not ideal and a bit anti-spec. The correct thing to do
// would be to set this in the transceiver creation via sendEncodings in
// addTransceiver, but FF doesn't support that. So we should split this
// between Chromium/WebKit (addTransceiver) and FF (this way) later - prlanzarin
const parameters = sender.getParameters();
// The encoder parameters might not be up yet; if that's the case,
// add a filler object so we can alter the parameters anyways
if (parameters.encodings == null || parameters.encodings.length === 0) {
parameters.encodings = [{}];
}
parameters.encodings.forEach((encoding) => {
// networkPriority
if (this.networkPriorities && this.networkPriorities[track.kind]) {
// eslint-disable-next-line no-param-reassign
encoding.networkPriority = this.networkPriorities[track.kind];
}
// Add further custom encoding parameters here
});
try {
sender.setParameters(parameters);
} catch (error) {
this.logger.error('BBB::WebRtcPeer::_processEncodingOptions - setParameters failed', error);
}
}
});
}
_flushInboundCandidateQueue() {
while (this._inboundCandidateQueue.length) {
const entry = this._inboundCandidateQueue.shift();
if (entry.candidate && entry.promise) {
try {
if (this.isPeerConnectionClosed()) {
entry.promise.resolve();
} else {
this.peerConnection.addIceCandidate(entry.candidate)
.then(entry.promise.resolve)
.catch(entry.promise.reject);
}
} catch (error) {
entry.promise.reject(error);
}
}
}
}
_trackQueueFlushEvents() {
this.on('newListener', (event) => {
if (event === 'icecandidate' || event === 'candidategatheringdone') {
while (this._outboundCandidateQueue.length) {
const candidate = this._outboundCandidateQueue.shift();
if (!candidate) this._emitCandidateGatheringDone();
}
}
});
this.peerConnection?.addEventListener('signalingstatechange', this._handleSignalingStateChange);
}
_emitCandidateGatheringDone() {
if (!this.candidateGatheringDone) {
this.emit('candidategatheringdone');
this.candidateGatheringDone = true;
}
}
_handleIceCandidate({ candidate }) {
if (this.hasListeners('icecandidate') || this.hasListeners('candidategatheringdone')) {
if (candidate) {
this.emit('icecandidate', candidate);
this.candidateGatheringDone = false;
} else this._emitCandidateGatheringDone();
} else if (!this.candidateGatheringDone) {
this._outboundCandidateQueue.push(candidate);
if (!candidate) this.candidateGatheringDone = true;
}
}
_handleSignalingStateChange() {
if (this.peerConnection?.signalingState === 'stable') {
this._flushInboundCandidateQueue();
}
}
waitForGathering(timeout = 0) {
if (timeout <= 0) return Promise.resolve();
if (this.isPeerConnectionClosed()) throw new Error('PeerConnection is closed');
if (this.peerConnection.iceGatheringState === 'complete') return Promise.resolve();
if (this._waitForGatheringPromise) return this._waitForGatheringPromise;
this._waitForGatheringPromise = new Promise((resolve) => {
this.once('candidategatheringdone', resolve);
this._waitForGatheringTimeout = setTimeout(() => {
this._emitCandidateGatheringDone();
}, timeout);
});
return this._waitForGatheringPromise;
}
_setRemoteDescription(rtcSessionDescription) {
if (this.isPeerConnectionClosed()) {
this.logger.error('BBB::WebRtcPeer::_setRemoteDescription - peer connection closed');
throw new Error('Peer connection is closed');
}
this.logger.debug('BBB::WebRtcPeer::_setRemoteDescription - setting remote description', rtcSessionDescription);
return this.peerConnection.setRemoteDescription(rtcSessionDescription);
}
_setLocalDescription(rtcSessionDescription) {
if (this.isPeerConnectionClosed()) {
this.logger.error('BBB::WebRtcPeer::_setLocalDescription - peer connection closed');
throw new Error('Peer connection is closed');
}
if (typeof this._gatheringTimeout === 'number' && this._gatheringTimeout > 0) {
this.logger.debug('BBB::WebRtcPeer::_setLocalDescription - setting description with gathering timer', rtcSessionDescription, this._gatheringTimeout);
return this.peerConnection.setLocalDescription(rtcSessionDescription)
.then(() => this.waitForGathering(this._gatheringTimeout));
}
this.logger.debug('BBB::WebRtcPeer::_setLocalDescription- setting description', rtcSessionDescription);
return this.peerConnection.setLocalDescription(rtcSessionDescription);
}
// Public method can be overridden via options
mediaStreamFactory() {
if (this.videoStream || this.audioStream) {
return Promise.resolve();
}
const handleGUMResolution = (stream) => {
if (stream.getAudioTracks().length > 0) {
this.audioStream = stream;
this.logger.debug('BBB::WebRtcPeer::mediaStreamFactory - generated audio', this.audioStream);
}
if (stream.getVideoTracks().length > 0) {
this.videoStream = stream;
this.logger.debug('BBB::WebRtcPeer::mediaStreamFactory - generated video', this.videoStream);
}
return stream;
};
if (typeof this._mediaStreamFactory === 'function') {
return this._mediaStreamFactory(this.mediaConstraints).then(handleGUMResolution);
}
this.logger.info('BBB::WebRtcPeer::mediaStreamFactory - running default factory', this.mediaConstraints);
return navigator.mediaDevices.getUserMedia(this.mediaConstraints)
.then(handleGUMResolution)
.catch((error) => {
this.logger.error('BBB::WebRtcPeer::mediaStreamFactory - gUM failed', error);
throw error;
});
}
set peerConnection(pc) {
this._pc = pc;
}
get peerConnection() {
return this._pc;
}
get logger() {
if (this.trace) return console;
return silentConsole;
}
getLocalSessionDescriptor() {
return this.peerConnection?.localDescription;
}
getRemoteSessionDescriptor() {
return this.peerConnection?.remoteDescription;
}
getLocalStream() {
if (this.peerConnection) {
if (this.localStream == null) this.localStream = new MediaStream();
const senders = this.peerConnection.getSenders();
const oldTracks = this.localStream.getTracks();
senders.forEach(({ track }) => {
if (track && !oldTracks.includes(track)) {
this.localStream.addTrack(track);
}
});
oldTracks.forEach((oldTrack) => {
if (!senders.some(({ track }) => track && track.id === oldTrack.id)) {
this.localStream.removeTrack(oldTrack);
}
});
return this.localStream;
}
return null;
}
getRemoteStream() {
if (this.remoteStream) {
return this.remoteStream;
}
if (this.peerConnection) {
this.remoteStream = new MediaStream();
this.peerConnection.getReceivers().forEach(({ track }) => {
if (track) {
this.remoteStream.addTrack(track);
}
});
return this.remoteStream;
}
return null;
}
isPeerConnectionClosed() {
return !this.peerConnection || this.peerConnection.signalingState === 'closed';
}
start() {
// Init PeerConnection
if (!this.peerConnection) {
this.peerConnection = new RTCPeerConnection(this.configuration);
}
if (this.isPeerConnectionClosed()) {
this.logger.trace('BBB::WebRtcPeer::start - peer connection closed');
throw new Error('Invalid peer state: closed');
}
this.peerConnection.addEventListener('icecandidate', this._handleIceCandidate);
this._trackQueueFlushEvents();
}
addIceCandidate(iceCandidate) {
const candidate = new RTCIceCandidate(iceCandidate);
switch (this.peerConnection?.signalingState) {
case 'closed':
this.logger.trace('BBB::WebRtcPeer::addIceCandidate - peer connection closed');
throw new Error('PeerConnection object is closed');
case 'stable': {
if (this.peerConnection.remoteDescription) {
this.logger.debug('BBB::WebRtcPeer::addIceCandidate - adding candidate', candidate);
return this.peerConnection.addIceCandidate(candidate);
}
}
// eslint-ignore-next-line no-fallthrough
default: {
this.logger.debug('BBB::WebRtcPeer::addIceCandidate - buffering inbound candidate', candidate);
const promise = new Promise();
this._inboundCandidateQueue.push({
candidate,
promise,
});
return promise;
}
}
}
_processMediaStreams() {
if (this.videoStream) {
this.videoStream.getTracks().forEach((track) => {
this.peerConnection.addTrack(track, this.videoStream);
});
}
if (this.audioStream) {
this.audioStream.getTracks().forEach((track) => {
this.peerConnection.addTrack(track, this.audioStream);
});
}
this.peerConnection.getTransceivers().forEach((transceiver) => {
// eslint-disable-next-line no-param-reassign
transceiver.direction = this._getTransceiverDirection();
});
}
async generateOffer() {
switch (this.mode) {
case 'recvonly': {
const useAudio = this.mediaConstraints
&& ((typeof this.mediaConstraints.audio === 'boolean' && this.mediaConstraints.audio)
|| (typeof this.mediaConstraints.audio === 'object'));
const useVideo = this.mediaConstraints
&& ((typeof this.mediaConstraints.video === 'boolean' && this.mediaConstraints.video)
|| (typeof this.mediaConstraints.video === 'object'));
if (useAudio) {
this.peerConnection.addTransceiver('audio', {
direction: this._getTransceiverDirection(),
});
}
if (useVideo) {
this.peerConnection.addTransceiver('video', {
direction: this._getTransceiverDirection(),
});
}
break;
}
case 'sendonly':
case 'sendrecv': {
await this.mediaStreamFactory();
this._processMediaStreams();
break;
}
case 'passive-sendrecv':
this._processMediaStreams();
break;
default:
break;
}
return this.peerConnection.createOffer()
.then((offer) => {
this.logger.debug('BBB::WebRtcPeer::generateOffer - created offer', offer);
return this._setLocalDescription(offer);
})
.then(() => {
this._processEncodingOptions();
const localDescription = this.getLocalSessionDescriptor();
this.logger.debug('BBB::WebRtcPeer::generateOffer - local description set', localDescription);
return localDescription.sdp;
})
.catch((error) => {
this.logger.error('BBB::WebRtcPeer::generateOffer - failed', error);
throw error;
});
}
processAnswer(sdp) {
const answer = new RTCSessionDescription({
type: 'answer',
sdp,
});
return this._setRemoteDescription(answer);
}
processOffer(sdp) {
const offer = new RTCSessionDescription({
type: 'offer',
sdp,
});
return this._setRemoteDescription(offer)
.then(async () => {
if (this.mode === 'sendonly' || this.mode === 'sendrecv') {
await this.mediaStreamFactory();
this._processMediaStreams();
} else if (this.mode === 'passive-sendrecv') {
this._processMediaStreams();
}
})
.then(() => this.peerConnection.createAnswer())
.then((answer) => {
this.logger.debug('BBB::WebRtcPeer::processOffer - created answer', answer);
return this._setLocalDescription(answer);
})
.then(() => {
const localDescription = this.getLocalSessionDescriptor();
this.logger.debug('BBB::WebRtcPeer::processOffer - local description set', localDescription.sdp);
return localDescription.sdp;
})
.catch((error) => {
this.logger.error('BBB::WebRtcPeer::processOffer - failed', error);
throw error;
});
}
dispose() {
this.logger.debug('BBB::WebRtcPeer::dispose');
try {
if (this.peerConnection) {
this.peerConnection.getSenders().forEach(({ track }) => stopTrack(track));
if (!this.isPeerConnectionClosed()) this.peerConnection.close();
this.peerConnection = null;
}
if (this.localStream) {
stopStream(this.localStream);
this.localStream = null;
}
if (this.remoteStream) {
stopStream(this.remoteStream);
this.remoteStream = null;
}
this._outboundCandidateQueue = [];
this.candidateGatheringDone = false;
if (this._waitForGatheringPromise) this._waitForGatheringPromise = null;
if (this._waitForGatheringTimeout) {
clearTimeout(this._waitForGatheringTimeout);
this._waitForGatheringTimeout = null;
}
} catch (error) {
this.logger.trace('BBB::WebRtcPeer::dispose - failed', error);
}
this.removeAllListeners();
}
}