bigbluebutton-Github/bigbluebutton-html5/imports/ui/services/bbb-webrtc-sfu/audio-broker.js

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import logger from '/imports/startup/client/logger';
import BaseBroker from '/imports/ui/services/bbb-webrtc-sfu/sfu-base-broker';
import WebRtcPeer from '/imports/ui/services/webrtc-base/peer';
const ON_ICE_CANDIDATE_MSG = 'iceCandidate';
const SUBSCRIBER_ANSWER = 'subscriberAnswer';
const DTMF = 'dtmf';
const SFU_COMPONENT_NAME = 'audio';
class AudioBroker extends BaseBroker {
constructor(
wsUrl,
role,
options = {},
) {
super(SFU_COMPONENT_NAME, wsUrl);
this.role = role;
this.offering = true;
// Optional parameters are:
// clientSessionNumber
// iceServers,
// offering,
// mediaServer,
// extension,
// constraints,
// stream,
// signalCandidates
// traceLogs
// networkPriority
// gatheringTimeout
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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// transparentListenOnly
Object.assign(this, options);
}
getLocalStream() {
if (this.webRtcPeer && typeof this.webRtcPeer.getLocalStream === 'function') {
return this.webRtcPeer.getLocalStream();
}
return null;
}
setLocalStream(stream) {
if (this.webRtcPeer == null || this.webRtcPeer.peerConnection == null) {
throw new Error('Missing peer connection');
}
const { peerConnection } = this.webRtcPeer;
const newTracks = stream.getAudioTracks();
const localStream = this.getLocalStream();
const oldTracks = localStream ? localStream.getAudioTracks() : [];
peerConnection.getSenders().forEach((sender, index) => {
if (sender.track && sender.track.kind === 'audio') {
const newTrack = newTracks[index];
if (newTrack == null) return;
// Cleanup old tracks in the local MediaStream
const oldTrack = oldTracks[index];
sender.replaceTrack(newTrack);
if (oldTrack) {
oldTrack.stop();
localStream.removeTrack(oldTrack);
}
localStream.addTrack(newTrack);
}
});
return Promise.resolve();
}
_join() {
return new Promise((resolve, reject) => {
try {
const options = {
audioStream: this.stream,
mediaConstraints: {
audio: this.constraints ? this.constraints : true,
video: false,
},
configuration: this.populatePeerConfiguration(),
onicecandidate: !this.signalCandidates ? null : (candidate) => {
this.onIceCandidate(candidate, this.role);
},
trace: this.traceLogs,
networkPriorities: this.networkPriority ? { audio: this.networkPriority } : undefined,
mediaStreamFactory: this.mediaStreamFactory,
gatheringTimeout: this.gatheringTimeout,
};
const peerRole = this.role === 'sendrecv' ? this.role : 'recvonly';
this.webRtcPeer = new WebRtcPeer(peerRole, options);
this.webRtcPeer.iceQueue = [];
this.webRtcPeer.start();
this.webRtcPeer.peerConnection.onconnectionstatechange = this.handleConnectionStateChange.bind(this);
if (this.offering) {
// We are the offerer
this.webRtcPeer.generateOffer()
.then(this.sendStartReq.bind(this))
.catch(this._handleOfferGenerationFailure.bind(this));
} else if (peerRole === 'recvonly') {
// We are the answerer and we are only listening, so we don't need
// to acquire local media
this.sendStartReq();
} else {
// We are the answerer and we are sending audio, so we need to acquire
// local media before sending the start request
this.webRtcPeer.mediaStreamFactory()
.then(() => { this.sendStartReq(); })
.catch(this._handleOfferGenerationFailure.bind(this));
}
resolve();
} catch (error) {
// 1305: "PEER_NEGOTIATION_FAILED",
const normalizedError = BaseBroker.assembleError(1305);
logger.error({
logCode: `${this.logCodePrefix}_peer_creation_failed`,
extraInfo: {
errorMessage: error.name || error.message || 'Unknown error',
errorCode: normalizedError.errorCode,
sfuComponent: this.sfuComponent,
started: this.started,
},
}, 'Audio peer creation failed');
this.onerror(normalizedError);
reject(normalizedError);
}
});
}
joinAudio() {
return this.openWSConnection()
.then(this._join.bind(this));
}
onWSMessage(message) {
const parsedMessage = JSON.parse(message.data);
switch (parsedMessage.id) {
case 'startResponse':
this.onRemoteDescriptionReceived(parsedMessage);
break;
case 'iceCandidate':
this.handleIceCandidate(parsedMessage.candidate);
break;
case 'webRTCAudioSuccess':
this.onstart(parsedMessage.success);
this.started = true;
break;
feat: add experimental support for ICE restart 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, 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.
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case 'restartIceResponse':
this.handleRestartIceResponse(parsedMessage);
break;
case 'webRTCAudioError':
case 'error':
this.handleSFUError(parsedMessage);
break;
case 'pong':
break;
default:
logger.debug({
logCode: `${this.logCodePrefix}_invalid_req`,
extraInfo: { messageId: parsedMessage.id || 'Unknown', sfuComponent: this.sfuComponent },
}, 'Discarded invalid SFU message');
}
}
handleSFUError(sfuResponse) {
const { code, reason, role } = sfuResponse;
const error = BaseBroker.assembleError(code, reason);
logger.error({
logCode: `${this.logCodePrefix}_sfu_error`,
extraInfo: {
errorCode: code,
errorMessage: error.errorMessage,
role,
sfuComponent: this.sfuComponent,
started: this.started,
},
}, 'Audio failed in SFU');
this.onerror(error);
}
sendLocalDescription(localDescription) {
const message = {
id: SUBSCRIBER_ANSWER,
type: this.sfuComponent,
role: this.role,
sdpOffer: localDescription,
};
this.sendMessage(message);
}
onRemoteDescriptionReceived(sfuResponse) {
if (this.offering) {
return this.processAnswer(sfuResponse);
}
return this.processOffer(sfuResponse);
}
sendStartReq(offer) {
const message = {
id: 'start',
type: this.sfuComponent,
role: this.role,
clientSessionNumber: this.clientSessionNumber,
sdpOffer: offer,
mediaServer: this.mediaServer,
extension: this.extension,
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: this.transparentListenOnly,
};
logger.debug({
logCode: `${this.logCodePrefix}_offer_generated`,
extraInfo: { sfuComponent: this.sfuComponent, role: this.role },
}, 'SFU audio offer generated');
this.sendMessage(message);
}
_handleOfferGenerationFailure(error) {
if (error) {
logger.error({
logCode: `${this.logCodePrefix}_offer_failure`,
extraInfo: {
errorMessage: error.name || error.message || 'Unknown error',
sfuComponent: this.sfuComponent,
},
}, 'Audio offer generation failed');
// 1305: "PEER_NEGOTIATION_FAILED",
this.onerror(error);
}
}
dtmf(tones) {
const message = {
id: DTMF,
type: this.sfuComponent,
tones,
};
this.sendMessage(message);
}
onIceCandidate(candidate, role) {
const message = {
id: ON_ICE_CANDIDATE_MSG,
role,
type: this.sfuComponent,
candidate,
};
this.sendMessage(message);
}
}
export default AudioBroker;