bigbluebutton-Github/bigbluebutton-html5/imports/ui/services/webrtc-base/peer.js

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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();
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.
2024-06-05 19:26:27 +08:00
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;
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.
2024-06-05 19:26:27 +08:00
};
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;
}
}
}
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.
2024-06-05 19:26:27 +08:00
_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', {
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.
2024-06-05 19:26:27 +08:00
direction: this._getTransceiverDirection(),
});
}
if (useVideo) {
this.peerConnection.addTransceiver('video', {
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.
2024-06-05 19:26:27 +08:00
direction: this._getTransceiverDirection(),
});
}
break;
}
case 'sendonly':
case 'sendrecv': {
await this.mediaStreamFactory();
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.
2024-06-05 19:26:27 +08:00
this._processMediaStreams();
break;
}
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.
2024-06-05 19:26:27 +08:00
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;
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.
2024-06-05 19:26:27 +08:00
})
.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)
feat(audio): add experimental transparent listen only mode This is an initial, experimental implementation of the feature proposed in https://github.com/bigbluebutton/bigbluebutton/issues/14021. The intention is to phase out the explicit listen only mode with two overarching goals: - Reduce UX friction and increase familiarity: the existence of a separate listen only mode is a source of confusion for the majority of users Reduce average server-side CPU usage while also making it possible for having full audio-only meetings. The proof-of-concept works based on the assumption that a "many concurrent active talkers" scenario is both rare and not useful. With that in mind, this including two server-side triggers: - On microphone inactivity (currently mute action that is sustained for 4 seconds, configurable): FreeSWITCH channels are held (which translates to much lower CPU usage, virtually 0%). Receiving channels are switched, server side, to a listening mode (SFU, mediasoup). * This required an extension to mediasoup two allow re-assigning producers to already established consumers. No re-negotiation is done. - On microphone activity (currently unmute action, immediate): FreeSWITCH channels are unheld, listening mode is deactivated and the mute state is updated accordingly (in this order). This is *off by default*. It needs to be enabled in two places: - `/etc/bigbluebutton/bbb-webrtc-sfu/production.yml` -> `transparentListenOnly: true` - End users: * Server wide: `/etc/bigbluebutton/bbb-html5.yml` -> `public.media.transparentListenOnly: true` * Per user: `userdata-bbb_transparent_listen_only=true`
2023-08-08 02:28:17 +08:00
.then(async () => {
if (this.mode === 'sendonly' || this.mode === 'sendrecv') {
await this.mediaStreamFactory();
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.
2024-06-05 19:26:27 +08:00
this._processMediaStreams();
} else if (this.mode === 'passive-sendrecv') {
this._processMediaStreams();
feat(audio): add experimental transparent listen only mode This is an initial, experimental implementation of the feature proposed in https://github.com/bigbluebutton/bigbluebutton/issues/14021. The intention is to phase out the explicit listen only mode with two overarching goals: - Reduce UX friction and increase familiarity: the existence of a separate listen only mode is a source of confusion for the majority of users Reduce average server-side CPU usage while also making it possible for having full audio-only meetings. The proof-of-concept works based on the assumption that a "many concurrent active talkers" scenario is both rare and not useful. With that in mind, this including two server-side triggers: - On microphone inactivity (currently mute action that is sustained for 4 seconds, configurable): FreeSWITCH channels are held (which translates to much lower CPU usage, virtually 0%). Receiving channels are switched, server side, to a listening mode (SFU, mediasoup). * This required an extension to mediasoup two allow re-assigning producers to already established consumers. No re-negotiation is done. - On microphone activity (currently unmute action, immediate): FreeSWITCH channels are unheld, listening mode is deactivated and the mute state is updated accordingly (in this order). This is *off by default*. It needs to be enabled in two places: - `/etc/bigbluebutton/bbb-webrtc-sfu/production.yml` -> `transparentListenOnly: true` - End users: * Server wide: `/etc/bigbluebutton/bbb-html5.yml` -> `public.media.transparentListenOnly: true` * Per user: `userdata-bbb_transparent_listen_only=true`
2023-08-08 02:28:17 +08:00
}
})
.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;
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.
2024-06-05 19:26:27 +08:00
})
.catch((error) => {
this.logger.error('BBB::WebRtcPeer::processOffer - failed', error);
throw error;
});
}
feat: add experimental support for ICE restart (#21208) We currently use full renegotiation for audio, video, and screen sharing reconnections, which involves re-creating transports and signaling channels from scratch. While effective in some scenarios, this approach is slow and, especially with outbound cameras and screen sharing, prone to failures. To counter that, WebRTC provides a mechanism to restart ICE without needing to re-create the peer connection. This allows us to avoid full renegotiation and bypass some server-side signaling limitations. Implementing ICE restart should make outbound camera/screen sharing reconnections more reliable and faster. This commit implements the ICE restart procedure for all WebRTC components' *outbound* peers. It is based on bbb-webrtc-sfu >= v2.15.0-beta.0, which added support for ICE restart requests. This feature is *off by default*. To enable it, adjust the following flags: - `/etc/bigbluebutton/bbb-webrtc-sfu/production.yml`: `allowIceRestart: true` - `/etc/bigbluebutton/bbb-html5.yml`: `public.kurento.restartIce` * Refer to the inline documentation; this can be enabled on the client side per media type. * Note: The default max retries for audio is lower than for cameras/screen sharing (1 vs 3). This is because the full renegotiation process for audio is more reliable, so ICE restart is attempted first, followed by full renegotiation if necessary. This approach is less suitable for cameras/ screen sharing, where longer retry periods for ICE restart make sense since full renegotation there is... iffy. Endpoints that are inbound/`recvonly` only (client's perspective) do *not* support ICE restart yet. There are two main reasons: - Server-side changes are required to support `recvonly` endpoints, particularly the proper handling of the server’s `setup` role in the its SDPs during an ICE restart. These changes are too broad for now, so they are deferred to future releases (SFU@v2.16). - Full reconnections for `recvonly` endpoints are currently reliable, unlike for `send*` endpoints. ICE restarts could still provide benefits for `recvonly` endpoints, but we need the server updates first.
2024-09-20 18:35:32 +08:00
restartIce(remoteSdp, initiator) {
if (this.isPeerConnectionClosed()) {
this.logger.error('BBB::WebRtcPeer::restartIce - peer connection closed');
throw new Error('Peer connection is closed');
}
const sdp = new RTCSessionDescription({
type: initiator ? 'offer' : 'answer',
sdp: remoteSdp,
});
this.logger.debug('BBB::WebRtcPeer::restartIce - setting remote description', sdp);
// If this peer was the original initiator, process remote first
if (initiator) {
return this.peerConnection.setRemoteDescription(sdp)
.then(() => this.peerConnection.createAnswer())
.then((answer) => this.peerConnection.setLocalDescription(answer))
.then(() => {
const localDescription = this.getLocalSessionDescriptor();
this.logger.debug('BBB::WebRtcPeer::restartIce - local description set', localDescription.sdp);
return localDescription.sdp;
});
}
// not the initiator - need to create offer first
return this.peerConnection.createOffer({ iceRestart: true })
.then((newOffer) => this.peerConnection.setLocalDescription(newOffer))
.then(() => {
const localDescription = this.getLocalSessionDescriptor();
this.logger.debug('BBB::WebRtcPeer::restartIce - local description set', localDescription.sdp);
return localDescription.sdp;
})
.then(() => this.peerConnection.setRemoteDescription(sdp));
}
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();
}
}