bigbluebutton-Github/bigbluebutton-html5/imports/api/audio/client/bridge/sfu-audio-bridge.js

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import BaseAudioBridge from './base';
import Auth from '/imports/ui/services/auth';
import logger from '/imports/startup/client/logger';
import AudioBroker from '/imports/ui/services/bbb-webrtc-sfu/audio-broker';
import loadAndPlayMediaStream from '/imports/ui/services/bbb-webrtc-sfu/load-play';
import {
fetchWebRTCMappedStunTurnServers,
getMappedFallbackStun,
} from '/imports/utils/fetchStunTurnServers';
import getFromMeetingSettings from '/imports/ui/services/meeting-settings';
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`
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import getFromUserSettings from '/imports/ui/services/users-settings';
import browserInfo from '/imports/utils/browserInfo';
import {
getAudioSessionNumber,
getAudioConstraints,
filterSupportedConstraints,
doGUM,
} from '/imports/api/audio/client/bridge/service';
import { shouldForceRelay } from '/imports/ui/services/bbb-webrtc-sfu/utils';
const SENDRECV_ROLE = 'sendrecv';
const RECV_ROLE = 'recv';
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.
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const PASSIVE_SENDRECV_ROLE = 'passive-sendrecv';
const BRIDGE_NAME = 'fullaudio';
const IS_CHROME = browserInfo.isChrome;
// SFU's base broker has distinct error codes so that it can be reused by different
// modules. Errors that have a valid, localized counterpart in audio manager are
// mapped so that the user gets a localized error message.
// The ones that haven't (ie SFU's servers-side errors), aren't mapped.
const errorCodeMap = {
1301: 1001,
1302: 1002,
1305: 1005,
1307: 1007,
};
// Error codes that are prone to a retry according to RETRY_THROUGH_RELAY
const RETRYABLE_ERRORS = [1007, 1010];
const mapErrorCode = (error) => {
const { errorCode } = error;
const mappedErrorCode = errorCodeMap[errorCode];
if (errorCode == null || mappedErrorCode == null) return error;
// eslint-disable-next-line no-param-reassign
error.errorCode = mappedErrorCode;
return error;
};
const getMediaServerAdapter = (listenOnly = false) => {
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const SETTINGS = window.meetingClientSettings;
const MEDIA = SETTINGS.public.media;
const DEFAULT_LISTENONLY_MEDIA_SERVER = SETTINGS.public.kurento.listenOnlyMediaServer;
const DEFAULT_FULLAUDIO_MEDIA_SERVER = MEDIA.audio.fullAudioMediaServer;
if (listenOnly) {
return getFromMeetingSettings(
'media-server-listenonly',
DEFAULT_LISTENONLY_MEDIA_SERVER,
);
}
return getFromMeetingSettings(
'media-server-fullaudio',
DEFAULT_FULLAUDIO_MEDIA_SERVER,
);
};
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const isTransparentListenOnlyEnabled = () => {
const SETTINGS = window.meetingClientSettings;
const MEDIA = SETTINGS.public.media;
const TRANSPARENT_LISTEN_ONLY = MEDIA.transparentListenOnly;
return getFromUserSettings(
'bbb_transparent_listen_only',
TRANSPARENT_LISTEN_ONLY,
);
};
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`
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export default class SFUAudioBridge extends BaseAudioBridge {
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`
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static getOfferingRole(isListenOnly) {
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const SETTINGS = window.meetingClientSettings;
const MEDIA = SETTINGS.public.media;
const LISTEN_ONLY_OFFERING = MEDIA.listenOnlyOffering;
const FULLAUDIO_OFFERING = MEDIA.fullAudioOffering;
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.
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return isListenOnly && !isTransparentListenOnlyEnabled()
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`
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? LISTEN_ONLY_OFFERING
: (!isTransparentListenOnlyEnabled() && FULLAUDIO_OFFERING);
}
constructor(userData) {
super();
this.userId = userData.userId;
this.name = userData.username;
this.sessionToken = userData.sessionToken;
this.broker = null;
this.reconnecting = false;
this.iceServers = [];
this.bridgeName = BRIDGE_NAME;
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.
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this.isListenOnly = false;
this.bypassGUM = false;
this.supportsTransparentListenOnly = isTransparentListenOnlyEnabled;
this.handleTermination = this.handleTermination.bind(this);
}
get inputStream() {
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.
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// Only return the stream if the broker is active and the role isn't recvonly
// Input stream == actual input-capturing stream, not the one that's being played
if (this.broker && this.role !== RECV_ROLE) {
return this.broker.getLocalStream();
}
return null;
}
get role() {
return this.broker?.role;
}
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
getBrokerRole({ hasInputStream }) {
if (this.isListenOnly) {
return isTransparentListenOnlyEnabled()
? PASSIVE_SENDRECV_ROLE
: RECV_ROLE;
}
if (this.bypassGUM && !hasInputStream) return PASSIVE_SENDRECV_ROLE;
return SENDRECV_ROLE;
}
setInputStream(stream) {
if (this.broker == null) return null;
return this.broker.setLocalStream(stream);
}
getPeerConnection() {
if (!this.broker) return null;
const { webRtcPeer } = this.broker;
if (webRtcPeer) return webRtcPeer.peerConnection;
return null;
}
// eslint-disable-next-line class-methods-use-this
mediaStreamFactory(constraints) {
return doGUM(constraints, true);
}
setConnectionTimeout() {
if (this.connectionTimeout) this.clearConnectionTimeout();
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const SETTINGS = window.meetingClientSettings;
const MEDIA = SETTINGS.public.media;
const CONNECTION_TIMEOUT_MS = MEDIA.listenOnlyCallTimeout || 15000;
this.connectionTimeout = setTimeout(() => {
const error = new Error(`ICE negotiation timeout after ${CONNECTION_TIMEOUT_MS / 1000}s`);
error.errorCode = 1010;
// Duplicating key-vals because I can'decide settle on an error pattern - prlanzarin again
error.errorCause = error.message;
error.errorMessage = error.message;
this.handleBrokerFailure(error);
}, CONNECTION_TIMEOUT_MS);
}
clearConnectionTimeout() {
if (this.connectionTimeout) {
clearTimeout(this.connectionTimeout);
this.connectionTimeout = null;
}
}
dispatchAutoplayHandlingEvent(mediaElement) {
const tagFailedEvent = new CustomEvent('audioPlayFailed', {
detail: { mediaElement },
});
window.dispatchEvent(tagFailedEvent);
this.callback({ status: this.baseCallStates.autoplayBlocked, bridge: this.bridgeName });
}
reconnect(options = {}) {
// If broker has already started, fire the reconnecting callback so the user
// knows what's going on
if (this.broker.started) {
this.callback({ status: this.baseCallStates.reconnecting, bridge: this.bridgeName });
} else {
// Otherwise: override termination handler so the ended callback doesn't get
// triggered - this is a retry attempt and the user shouldn't be notified
// yet.
this.broker.onended = () => {};
}
this.broker.stop();
this.reconnecting = true;
this._startBroker({ isListenOnly: this.isListenOnly, ...options })
.catch((error) => {
// Error handling is a no-op because it will be "handled" in handleBrokerFailure
logger.debug({
logCode: 'sfuaudio_reconnect_failed',
extraInfo: {
errorMessage: error.errorMessage,
reconnecting: this.reconnecting,
bridge: this.bridgeName,
role: this.role,
},
}, 'SFU audio reconnect failed');
});
}
handleBrokerFailure(error) {
return new Promise((resolve, reject) => {
this.clearConnectionTimeout();
mapErrorCode(error);
const { errorMessage, errorCause, errorCode } = error;
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const SETTINGS = window.meetingClientSettings;
const MEDIA = SETTINGS.public.media;
const RETRY_THROUGH_RELAY = MEDIA.audio.retryThroughRelay || false;
if (!this.reconnecting) {
if (this.broker.started) {
logger.error({
logCode: 'sfuaudio_error_try_to_reconnect',
extraInfo: {
errorMessage,
errorCode,
errorCause,
bridge: this.bridgeName,
role: this.role,
},
}, 'SFU audio failed, try to reconnect');
this.reconnect();
return resolve();
}
if (RETRYABLE_ERRORS.includes(errorCode) && RETRY_THROUGH_RELAY) {
logger.error({
logCode: 'sfuaudio_error_retry_through_relay',
extraInfo: {
errorMessage,
errorCode,
errorCause,
bridge: this.bridgeName,
role: this.role,
},
}, 'SFU audio failed to connect, retry through relay');
this.reconnect({ forceRelay: true });
return resolve();
}
}
// Already tried reconnecting once OR the user handn't successfully
// connected firsthand and retrying isn't an option. Finish the session
// and reject with the error
logger.error({
logCode: 'sfuaudio_error',
extraInfo: {
errorMessage,
errorCode,
errorCause,
reconnecting: this.reconnecting,
bridge: this.bridgeName,
role: this.role,
},
}, 'SFU audio failed');
this.clearConnectionTimeout();
this.broker.stop();
this.callback({
status: this.baseCallStates.failed,
error: errorCode,
bridgeError: errorMessage,
bridge: this.bridgeName,
});
return reject(error);
});
}
handleTermination() {
this.clearConnectionTimeout();
return this.callback({ status: this.baseCallStates.ended, bridge: this.bridgeName });
}
handleStart() {
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const SETTINGS = window.meetingClientSettings;
const MEDIA = SETTINGS.public.media;
const MEDIA_TAG = MEDIA.mediaTag.replace(/#/g, '');
const stream = this.broker.webRtcPeer.getRemoteStream();
const mediaElement = document.getElementById(MEDIA_TAG);
return loadAndPlayMediaStream(stream, mediaElement, false).then(() => {
this.callback({
status: this.baseCallStates.started,
bridge: this.bridgeName,
});
this.clearConnectionTimeout();
this.reconnecting = false;
}).catch((error) => {
// NotAllowedError equals autoplay issues, fire autoplay handling event.
// This will be handled in audio-manager.
if (error.name === 'NotAllowedError') {
logger.error({
logCode: 'sfuaudio_error_autoplay',
extraInfo: {
errorName: error.name,
bridge: this.bridgeName,
role: this.role,
},
}, 'SFU audio media play failed due to autoplay error');
this.dispatchAutoplayHandlingEvent(mediaElement);
// For connection purposes, this worked - the autoplay thing is a client
// side soft issue to be handled at the UI/UX level, not WebRTC/negotiation
// So: clear the connection timer
this.clearConnectionTimeout();
this.reconnecting = false;
} else {
const normalizedError = {
errorCode: 1004,
errorMessage: error.message || 'AUDIO_PLAY_FAILED',
};
this.callback({
status: this.baseCallStates.failed,
error: normalizedError.errorCode,
bridgeError: normalizedError.errorMessage,
bridge: this.bridgeName,
});
throw normalizedError;
}
});
}
async _startBroker(options) {
try {
this.iceServers = await fetchWebRTCMappedStunTurnServers(this.sessionToken);
} catch (error) {
logger.error({ logCode: 'sfuaudio_stun-turn_fetch_failed' },
'SFU audio bridge failed to fetch STUN/TURN info, using default servers');
this.iceServers = getMappedFallbackStun();
}
return new Promise((resolve, reject) => {
const {
isListenOnly,
extension,
inputStream,
forceRelay: _forceRelay = false,
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.
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bypassGUM = false,
} = options;
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const SETTINGS = window.meetingClientSettings;
const MEDIA = SETTINGS.public.media;
const SIGNAL_CANDIDATES = SETTINGS.public.kurento.signalCandidates;
const SFU_URL = SETTINGS.public.kurento.wsUrl;
const TRACE_LOGS = SETTINGS.public.kurento.traceLogs;
const GATHERING_TIMEOUT = SETTINGS.public.kurento.gatheringTimeout;
const RETRY_THROUGH_RELAY = MEDIA.audio.retryThroughRelay || false;
const { audio: NETWORK_PRIORITY } = MEDIA.networkPriorities || {};
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.
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const {
enabled: RESTART_ICE = false,
retries: RESTART_ICE_RETRIES = 1,
} = SETTINGS.public.kurento?.restartIce?.audio || {};
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const handleInitError = (_error) => {
mapErrorCode(_error);
if (!RETRYABLE_ERRORS.includes(_error?.errorCode)
|| !RETRY_THROUGH_RELAY
|| this.reconnecting) {
reject(_error);
}
};
try {
this.inEchoTest = !!extension;
this.isListenOnly = isListenOnly;
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.bypassGUM = bypassGUM;
const role = this.getBrokerRole({
hasInputStream: !!inputStream,
});
const brokerOptions = {
clientSessionNumber: getAudioSessionNumber(),
extension,
iceServers: this.iceServers,
mediaServer: getMediaServerAdapter(isListenOnly),
constraints: getAudioConstraints({ deviceId: this.inputDeviceId }),
forceRelay: _forceRelay || shouldForceRelay(),
stream: (inputStream && inputStream.active) ? inputStream : undefined,
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`
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offering: SFUAudioBridge.getOfferingRole(this.isListenOnly),
signalCandidates: SIGNAL_CANDIDATES,
traceLogs: TRACE_LOGS,
networkPriority: NETWORK_PRIORITY,
mediaStreamFactory: this.mediaStreamFactory,
gatheringTimeout: GATHERING_TIMEOUT,
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
transparentListenOnly: isTransparentListenOnlyEnabled(),
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
bypassGUM,
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
// ICE restart only works for publishers right now - recvonly full
// reconnection works ok without it.
restartIce: RESTART_ICE && !isListenOnly,
restartIceMaxRetries: RESTART_ICE_RETRIES,
};
this.broker = new AudioBroker(
Auth.authenticateURL(SFU_URL),
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
role,
brokerOptions,
);
this.broker.onended = this.handleTermination.bind(this);
this.broker.onerror = (error) => {
this.handleBrokerFailure(error).catch(reject);
};
this.broker.onstart = () => {
this.handleStart().then(resolve).catch(reject);
};
// Set up a connectionTimeout in case the server or network are botching
// negotiation or conn checks.
this.setConnectionTimeout();
this.broker.joinAudio().catch(handleInitError);
} catch (error) {
handleInitError(error);
}
});
}
async joinAudio(options, callback) {
this.callback = callback;
this.reconnecting = false;
return this._startBroker(options);
}
sendDtmf(tones) {
if (this.broker) {
this.broker.dtmf(tones);
}
}
transferCall(onTransferSuccess) {
this.inEchoTest = false;
return this.trackTransferState(onTransferSuccess);
}
async updateAudioConstraints(constraints) {
try {
if (typeof constraints !== 'object') return;
const matchConstraints = filterSupportedConstraints(constraints);
if (IS_CHROME) {
matchConstraints.deviceId = this.inputDeviceId;
const stream = await doGUM({ audio: matchConstraints });
await this.setInputStream(stream);
} else {
this.inputStream.getAudioTracks()
.forEach((track) => track.applyConstraints(matchConstraints));
}
} catch (error) {
logger.error({
logCode: 'sfuaudio_audio_constraint_error',
extraInfo: {
errorCode: error.code,
errorMessage: error.message,
bridgeName: this.bridgeName,
role: this.role,
},
}, 'Failed to update audio constraint');
}
}
trickleIce() {
return new Promise((resolve, reject) => {
try {
fetchWebRTCMappedStunTurnServers(this.sessionToken)
.then((iceServers) => {
2024-07-03 01:58:58 +08:00
const SETTINGS = window.meetingClientSettings;
const MEDIA = SETTINGS.public.media;
const SFU_URL = SETTINGS.public.kurento.wsUrl;
const TRACE_LOGS = SETTINGS.public.kurento.traceLogs;
const GATHERING_TIMEOUT = SETTINGS.public.kurento.gatheringTimeout;
const LISTEN_ONLY_OFFERING = MEDIA.listenOnlyOffering;
const options = {
clientSessionNumber: getAudioSessionNumber(),
iceServers,
offering: LISTEN_ONLY_OFFERING,
traceLogs: TRACE_LOGS,
gatheringTimeout: GATHERING_TIMEOUT,
};
this.broker = new AudioBroker(
Auth.authenticateURL(SFU_URL),
RECV_ROLE,
options,
);
this.broker.onstart = () => {
const { peerConnection } = this.broker.webRtcPeer;
if (!peerConnection) return resolve(null);
const selectedCandidatePair = peerConnection.getReceivers()[0]
.transport.iceTransport.getSelectedCandidatePair();
const validIceCandidate = [selectedCandidatePair.local];
this.broker.stop();
return resolve(validIceCandidate);
};
this.broker.joinAudio().catch(reject);
});
} catch (error) {
// Rollback
this.exitAudio();
reject(error);
}
});
}
exitAudio() {
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const SETTINGS = window.meetingClientSettings;
const MEDIA = SETTINGS.public.media;
const MEDIA_TAG = MEDIA.mediaTag.replace(/#/g, '');
const mediaElement = document.getElementById(MEDIA_TAG);
this.clearConnectionTimeout();
this.reconnecting = false;
if (this.broker) {
this.broker.stop();
this.broker = null;
}
if (mediaElement && typeof mediaElement.pause === 'function') {
mediaElement.pause();
mediaElement.srcObject = null;
}
return Promise.resolve();
}
}