bigbluebutton-Github/bigbluebutton-html5/imports/api/audio/client/bridge/sip.js
Mario Jr 49bfe9f48d Set stun/turn server for audio's peer in html5 client
Latest SIP.js version sets this using peerConnectionConfiguration property instead of UserAgent option.
This solves #10569
2020-10-01 10:16:48 -03:00

971 lines
29 KiB
JavaScript
Executable File

import browser from 'browser-detect';
import BaseAudioBridge from './base';
import logger from '/imports/startup/client/logger';
import {
fetchWebRTCMappedStunTurnServers,
getFallbackStun,
} from '/imports/utils/fetchStunTurnServers';
import {
isUnifiedPlan,
toUnifiedPlan,
toPlanB,
stripMDnsCandidates,
analyzeSdp,
logSelectedCandidate,
} from '/imports/utils/sdpUtils';
import { Tracker } from 'meteor/tracker';
import VoiceCallStates from '/imports/api/voice-call-states';
import CallStateOptions from '/imports/api/voice-call-states/utils/callStates';
import Auth from '/imports/ui/services/auth';
const MEDIA = Meteor.settings.public.media;
const MEDIA_TAG = MEDIA.mediaTag;
const CALL_TRANSFER_TIMEOUT = MEDIA.callTransferTimeout;
const CALL_HANGUP_TIMEOUT = MEDIA.callHangupTimeout;
const CALL_HANGUP_MAX_RETRIES = MEDIA.callHangupMaximumRetries;
const IPV4_FALLBACK_DOMAIN = Meteor.settings.public.app.ipv4FallbackDomain;
const CALL_CONNECT_TIMEOUT = 20000;
const ICE_NEGOTIATION_TIMEOUT = 20000;
const AUDIO_SESSION_NUM_KEY = 'AudioSessionNumber';
const USER_AGENT_RECONNECTION_ATTEMPTS = 3;
const USER_AGENT_RECONNECTION_DELAY_MS = 5000;
const USER_AGENT_CONNECTION_TIMEOUT_MS = 5000;
const getAudioSessionNumber = () => {
let currItem = parseInt(sessionStorage.getItem(AUDIO_SESSION_NUM_KEY), 10);
if (!currItem) {
currItem = 0;
}
currItem += 1;
sessionStorage.setItem(AUDIO_SESSION_NUM_KEY, currItem);
return currItem;
};
class SIPSession {
constructor(user, userData, protocol, hostname,
baseCallStates, baseErrorCodes, reconnectAttempt) {
this.user = user;
this.userData = userData;
this.protocol = protocol;
this.hostname = hostname;
this.baseCallStates = baseCallStates;
this.baseErrorCodes = baseErrorCodes;
this.reconnectAttempt = reconnectAttempt;
this.currentSession = null;
this.remoteStream = null;
this._hangupFlag = false;
this._reconnecting = false;
this._currentSessionState = null;
}
joinAudio({ isListenOnly, extension, inputStream }, managerCallback) {
return new Promise((resolve, reject) => {
const callExtension = extension ? `${extension}${this.userData.voiceBridge}` : this.userData.voiceBridge;
const callback = (message) => {
// There will sometimes we erroneous errors put out like timeouts and improper shutdowns,
// but only the first error ever matters
if (this.alreadyErrored) {
logger.info({
logCode: 'sip_js_absorbing_callback_message',
extraInfo: { message },
}, 'Absorbing a redundant callback message.');
return;
}
if (message.status === this.baseCallStates.failed) {
this.alreadyErrored = true;
}
managerCallback(message).then(resolve);
};
this.callback = callback;
// If there's an extension passed it means that we're joining the echo test first
this.inEchoTest = !!extension;
return this.doCall({ callExtension, isListenOnly, inputStream })
.catch((reason) => {
reject(reason);
});
});
}
async getIceServers(sessionToken) {
try {
const iceServers = await fetchWebRTCMappedStunTurnServers(sessionToken);
return iceServers;
} catch (error) {
logger.error({
logCode: 'sip_js_fetchstunturninfo_error',
extraInfo: {
errorCode: error.code,
errorMessage: error.message,
callerIdName: this.user.callerIdName,
},
}, 'Full audio bridge failed to fetch STUN/TURN info');
return getFallbackStun();
}
}
doCall(options) {
const {
isListenOnly,
} = options;
const {
userId,
name,
sessionToken,
} = this.user;
const callerIdName = [
`${userId}_${getAudioSessionNumber()}`,
'bbbID',
isListenOnly ? `LISTENONLY-${name}` : name,
].join('-').replace(/"/g, "'");
this.user.callerIdName = callerIdName;
this.callOptions = options;
return this.getIceServers(sessionToken)
.then(this.createUserAgent.bind(this))
.then(this.inviteUserAgent.bind(this));
}
transferCall(onTransferSuccess) {
return new Promise((resolve, reject) => {
this.inEchoTest = false;
let trackerControl = null;
const timeout = setTimeout(() => {
trackerControl.stop();
logger.error({ logCode: 'sip_js_transfer_timed_out' }, 'Timeout on transferring from echo test to conference');
this.callback({
status: this.baseCallStates.failed,
error: 1008,
bridgeError: 'Timeout on call transfer',
});
this.exitAudio();
reject(this.baseErrorCodes.REQUEST_TIMEOUT);
}, CALL_TRANSFER_TIMEOUT);
// This is is the call transfer code ask @chadpilkey
if (this.sessionSupportRTPPayloadDtmf(this.currentSession)) {
this.currentSession.sessionDescriptionHandler.sendDtmf(1);
} else {
// RFC4733 not supported , sending DTMF through INFO
logger.debug({
logCode: 'sip_js_rtp_payload_dtmf_not_supported',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'Browser do not support payload dtmf, using INFO instead');
this.sendDtmf(1);
}
Tracker.autorun((c) => {
trackerControl = c;
const selector = { meetingId: Auth.meetingID, userId: Auth.userID };
const query = VoiceCallStates.find(selector);
query.observeChanges({
changed: (id, fields) => {
if (fields.callState === CallStateOptions.IN_CONFERENCE) {
clearTimeout(timeout);
onTransferSuccess();
c.stop();
resolve();
}
},
});
});
});
}
/**
*
* sessionSupportRTPPayloadDtmf
* tells if browser support RFC4733 DTMF.
* Safari 13 doens't support it yet
*/
sessionSupportRTPPayloadDtmf(session) {
try {
const sessionDescriptionHandler = session
? session.sessionDescriptionHandler
: this.currentSession.sessionDescriptionHandler;
const senders = sessionDescriptionHandler.peerConnection.getSenders();
return !!(senders[0].dtmf);
} catch (error) {
return false;
}
}
/**
* sendDtmf - send DTMF Tones using INFO message
*
* same as SimpleUser's dtmf
*/
sendDtmf(tone) {
const dtmf = tone;
const duration = 2000;
const body = {
contentDisposition: 'render',
contentType: 'application/dtmf-relay',
content: `Signal=${dtmf}\r\nDuration=${duration}`,
};
const requestOptions = { body };
return this.currentSession.info({ requestOptions });
}
exitAudio() {
return new Promise((resolve, reject) => {
let hangupRetries = 0;
this._hangupFlag = false;
this.userRequestedHangup = true;
const tryHangup = () => {
if (this._hangupFlag) {
resolve();
}
if ((this.currentSession
&& (this.currentSession.state === SIP.SessionState.Terminated))
|| (this.userAgent && (!this.userAgent.isConnected()))) {
this._hangupFlag = true;
return resolve();
}
if (this.currentSession
&& ((this.currentSession.state === SIP.SessionState.Establishing)
|| (this.currentSession.state === SIP.SessionState.Established))) {
this.currentSession.bye().then(() => {
this._hangupFlag = true;
return resolve();
});
}
if (this.userAgent && this.userAgent.isConnected()) {
this.userAgent.stop();
}
hangupRetries += 1;
setTimeout(() => {
if (hangupRetries > CALL_HANGUP_MAX_RETRIES) {
this.callback({
status: this.baseCallStates.failed,
error: 1006,
bridgeError: 'Timeout on call hangup',
});
return reject(this.baseErrorCodes.REQUEST_TIMEOUT);
}
if (!this._hangupFlag) return tryHangup();
return resolve();
}, CALL_HANGUP_TIMEOUT);
};
return tryHangup();
});
}
createUserAgent(iceServers) {
return new Promise((resolve, reject) => {
if (this.userRequestedHangup === true) reject();
const {
hostname,
protocol,
} = this;
const {
callerIdName,
sessionToken,
} = this.user;
logger.debug({ logCode: 'sip_js_creating_user_agent', extraInfo: { callerIdName } }, 'Creating the user agent');
if (this.userAgent && this.userAgent.isConnected()) {
if (this.userAgent.configuration.hostPortParams === this.hostname) {
logger.debug({ logCode: 'sip_js_reusing_user_agent', extraInfo: { callerIdName } }, 'Reusing the user agent');
resolve(this.userAgent);
return;
}
logger.debug({ logCode: 'sip_js_different_host_name', extraInfo: { callerIdName } }, 'Different host name. need to kill');
}
const localSdpCallback = (sdp) => {
// For now we just need to call the utils function to parse and log the different pieces.
// In the future we're going to want to be tracking whether there were TURN candidates
// and IPv4 candidates to make informed decisions about what to do on fallbacks/reconnects.
analyzeSdp(sdp);
};
const remoteSdpCallback = (sdp) => {
// We have have to find the candidate that FS sends back to us to determine if the client
// is connecting with IPv4 or IPv6
const sdpInfo = analyzeSdp(sdp, false);
this.protocolIsIpv6 = sdpInfo.v6Info.found;
};
let userAgentConnected = false;
const token = `sessionToken=${sessionToken}`;
this.userAgent = new SIP.UserAgent({
uri: SIP.UserAgent.makeURI(`sip:${encodeURIComponent(callerIdName)}@${hostname}`),
transportOptions: {
server: `${(protocol === 'https:' ? 'wss://' : 'ws://')}${hostname}/ws?${token}`,
connectionTimeout: USER_AGENT_CONNECTION_TIMEOUT_MS,
},
sessionDescriptionHandlerFactoryOptions: {
peerConnectionConfiguration: {
iceServers,
},
},
displayName: callerIdName,
register: false,
userAgentString: 'BigBlueButton',
});
const handleUserAgentConnection = () => {
userAgentConnected = true;
resolve(this.userAgent);
};
const handleUserAgentDisconnection = () => {
if (this.userAgent) {
if (this.userRequestedHangup) return;
let error;
let bridgeError;
if (!this._reconnecting) {
if (userAgentConnected) {
error = 1001;
bridgeError = 'Websocket disconnected';
} else {
error = 1002;
bridgeError = 'Websocket failed to connect';
}
this.callback({
status: this.baseCallStates.failed,
error,
bridgeError,
});
}
logger.info({
logCode: 'sip_js_session_ua_disconnected',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'User agent disconnected: trying to reconnect...'
+ `${this.userRequestedHangup}`);
logger.info({
logCode: 'sip_js_session_ua_reconnecting',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'User agent failed to connect, reconnecting');
this.userAgent.reconnect().then(() => {
logger.info({
logCode: 'sip_js_session_ua_reconnected',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'User agent succesfully reconnected');
}).catch(() => {
reject(this.baseErrorCodes.CONNECTION_ERROR);
});
}
};
this.userAgent.transport.onConnect = handleUserAgentConnection;
this.userAgent.transport.onDisconnect = handleUserAgentDisconnection;
const preturn = this.userAgent.start().then(() => {
logger.info({
logCode: 'sip_js_session_ua_connected',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'User agent succesfully connected');
resolve();
}).catch(() => {
logger.info({
logCode: 'sip_js_session_ua_reconnecting',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'User agent failed to connect, reconnecting');
this.reconnect().then(() => {
logger.info({
logCode: 'sip_js_session_ua_reconnected',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'User agent succesfully reconnected');
resolve();
}).catch(() => {
logger.info({
logCode: 'sip_js_session_ua_disconnected',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'User agent failed to reconnect after'
+ ` ${USER_AGENT_RECONNECTION_ATTEMPTS} attemps`);
this.callback({
status: this.baseCallStates.failed,
error: 1002,
bridgeError: 'Websocket failed to connect',
});
reject({
type: this.baseErrorCodes.CONNECTION_ERROR,
});
});
});
return preturn;
});
}
reconnect(attempts = 1) {
return new Promise((resolve, reject) => {
if (this._reconnecting) {
return resolve();
}
if (attempts > USER_AGENT_RECONNECTION_ATTEMPTS) {
return reject({
type: this.baseErrorCodes.CONNECTION_ERROR,
});
}
this._reconnecting = true;
setTimeout(() => {
this.userAgent.reconnect().then(() => {
this._reconnecting = false;
resolve();
}).catch(() => {
this._reconnecting = false;
this.reconnect(++attempts).then(() => {
resolve();
}).catch((error) => {
reject(error);
});
});
}, USER_AGENT_RECONNECTION_DELAY_MS);
});
}
inviteUserAgent(userAgent) {
return new Promise((resolve, reject) => {
if (this.userRequestedHangup === true) reject();
const {
hostname,
} = this;
const {
callExtension,
isListenOnly,
} = this.callOptions;
const target = SIP.UserAgent.makeURI(`sip:${callExtension}@${hostname}`);
const inviterOptions = {
sessionDescriptionHandlerOptions: {
constraints: {
audio: !isListenOnly,
video: false,
},
},
sessionDescriptionHandlerModifiersPostICEGathering:
[stripMDnsCandidates],
};
if (isListenOnly) {
inviterOptions.sessionDescriptionHandlerOptions.offerOptions = {
offerToReceiveAudio: true,
};
}
const inviter = new SIP.Inviter(userAgent, target, inviterOptions);
this.currentSession = inviter;
this.setupEventHandlers(inviter).then(() => {
inviter.invite().then(() => {
resolve();
}).catch(e => reject(e));
});
});
}
setupEventHandlers(currentSession) {
return new Promise((resolve, reject) => {
if (this.userRequestedHangup === true) reject();
let iceCompleted = false;
let fsReady = false;
const setupRemoteMedia = () => {
const mediaElement = document.querySelector(MEDIA_TAG);
this.remoteStream = new MediaStream();
this.currentSession.sessionDescriptionHandler
.peerConnection.getReceivers().forEach((receiver) => {
if (receiver.track) {
this.remoteStream.addTrack(receiver.track);
}
});
logger.info({
logCode: 'sip_js_session_playing_remote_media',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'Audio call - playing remote media');
mediaElement.srcObject = this.remoteStream;
mediaElement.play();
};
const checkIfCallReady = () => {
if (this.userRequestedHangup === true) {
this.exitAudio();
resolve();
}
logger.info({
logCode: 'sip_js_session_check_if_call_ready',
extraInfo: {
iceCompleted,
fsReady,
},
}, 'Audio call - check if ICE is finished and FreeSWITCH is ready');
if (iceCompleted && fsReady) {
this.webrtcConnected = true;
setupRemoteMedia();
const { sdp } = this.currentSession.sessionDescriptionHandler
.peerConnection.remoteDescription;
logger.info({
logCode: 'sip_js_session_setup_remote_media',
extraInfo: {
callerIdName: this.user.callerIdName,
sdp,
},
}, 'Audio call - setup remote media');
this.callback({ status: this.baseCallStates.started });
resolve();
}
};
// Sometimes FreeSWITCH just won't respond with anything and hangs. This timeout is to
// avoid that issue
const callTimeout = setTimeout(() => {
this.callback({
status: this.baseCallStates.failed,
error: 1006,
bridgeError: `Call timed out on start after ${CALL_CONNECT_TIMEOUT / 1000}s`,
});
this.exitAudio();
}, CALL_CONNECT_TIMEOUT);
let iceNegotiationTimeout;
const handleSessionAccepted = () => {
logger.info({ logCode: 'sip_js_session_accepted', extraInfo: { callerIdName: this.user.callerIdName } }, 'Audio call session accepted');
clearTimeout(callTimeout);
// If ICE isn't connected yet then start timeout waiting for ICE to finish
if (!iceCompleted) {
iceNegotiationTimeout = setTimeout(() => {
this.callback({
status: this.baseCallStates.failed,
error: 1010,
bridgeError: 'ICE negotiation timeout after '
+ `${ICE_NEGOTIATION_TIMEOUT / 1000}s`,
});
this.exitAudio();
reject({
type: this.baseErrorCodes.CONNECTION_ERROR,
});
}, ICE_NEGOTIATION_TIMEOUT);
}
checkIfCallReady();
};
const handleIceNegotiationFailed = (peer) => {
if (iceCompleted) {
logger.error({
logCode: 'sipjs_ice_failed_after',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'ICE connection failed after success');
} else {
logger.error({
logCode: 'sipjs_ice_failed_before',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'ICE connection failed before success');
}
clearTimeout(callTimeout);
clearTimeout(iceNegotiationTimeout);
this.callback({
status: this.baseCallStates.failed,
error: 1007,
bridgeError: 'ICE negotiation failed. Current state '
+ `- ${peer.iceConnectionState}`,
});
};
const handleIceConnectionTerminated = (peer) => {
if (!this.userRequestedHangup) {
logger.error({
logCode: 'sipjs_ice_closed',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'ICE connection closed');
}
this.callback({
status: this.baseCallStates.failed,
error: 1012,
bridgeError: 'ICE connection closed. Current state -'
+ `${peer.iceConnectionState}`,
});
};
const handleSessionProgress = (update) => {
logger.info({
logCode: 'sip_js_session_progress',
extraInfo: {
callerIdName: this.user.callerIdName,
update,
},
}, 'Audio call session progress update');
this.currentSession.sessionDescriptionHandler.peerConnectionDelegate = {
onconnectionstatechange: (event) => {
const peer = event.target;
switch (peer.connectionState) {
case 'connected':
logger.info({
logCode: 'sip_js_ice_connection_success',
extraInfo: {
currentState: peer.connectionState,
callerIdName: this.user.callerIdName,
},
}, 'ICE connection success. Current state - '
+ `${peer.iceConnectionState}`);
clearTimeout(callTimeout);
clearTimeout(iceNegotiationTimeout);
iceCompleted = true;
logSelectedCandidate(peer, this.protocolIsIpv6);
checkIfCallReady();
break;
case 'failed':
handleIceNegotiationFailed(peer);
break;
case 'closed':
handleIceConnectionTerminated(peer);
break;
default:
break;
}
},
};
};
const handleSessionTerminated = (message, cause) => {
clearTimeout(callTimeout);
clearTimeout(iceNegotiationTimeout);
if (!message && !cause && !!this.userRequestedHangup) {
return this.callback({
status: this.baseCallStates.ended,
});
}
// if session hasn't even started, we let audio-modal to handle
// any possile errors
if (!this._currentSessionState) return false;
logger.error({
logCode: 'sip_js_call_terminated',
extraInfo: { cause, callerIdName: this.user.callerIdName },
}, `Audio call terminated. cause=${cause}`);
let mappedCause;
if (!iceCompleted) {
mappedCause = '1004';
} else {
mappedCause = '1005';
}
return this.callback({
status: this.baseCallStates.failed,
error: mappedCause,
bridgeError: cause,
});
};
currentSession.stateChange.addListener((state) => {
switch (state) {
case SIP.SessionState.Initial:
break;
case SIP.SessionState.Establishing:
handleSessionProgress();
break;
case SIP.SessionState.Established:
handleSessionAccepted();
break;
case SIP.SessionState.Terminating:
break;
case SIP.SessionState.Terminated:
handleSessionTerminated();
break;
default:
logger.error({
logCode: 'sipjs_ice_session_unknown_state',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'SIP.js unknown session state');
break;
}
this._currentSessionState = state;
});
Tracker.autorun((c) => {
const selector = { meetingId: Auth.meetingID, userId: Auth.userID };
const query = VoiceCallStates.find(selector);
query.observeChanges({
changed: (id, fields) => {
if ((this.inEchoTest && fields.callState === CallStateOptions.IN_ECHO_TEST)
|| (!this.inEchoTest && fields.callState === CallStateOptions.IN_CONFERENCE)) {
fsReady = true;
checkIfCallReady();
c.stop();
}
},
});
});
resolve();
});
}
}
export default class SIPBridge extends BaseAudioBridge {
constructor(userData) {
super(userData);
const {
userId,
username,
sessionToken,
} = userData;
this.user = {
userId,
sessionToken,
name: username,
};
this.media = {
inputDevice: {},
};
this.protocol = window.document.location.protocol;
this.hostname = window.document.location.hostname;
// SDP conversion utilitary methods to be used inside SIP.js
window.isUnifiedPlan = isUnifiedPlan;
window.toUnifiedPlan = toUnifiedPlan;
window.toPlanB = toPlanB;
window.stripMDnsCandidates = stripMDnsCandidates;
// No easy way to expose the client logger to sip.js code so we need to attach it globally
window.clientLogger = logger;
}
joinAudio({ isListenOnly, extension, inputStream }, managerCallback) {
const hasFallbackDomain = typeof IPV4_FALLBACK_DOMAIN === 'string' && IPV4_FALLBACK_DOMAIN !== '';
return new Promise((resolve, reject) => {
let { hostname } = this;
this.activeSession = new SIPSession(this.user, this.userData, this.protocol,
hostname, this.baseCallStates, this.baseErrorCodes, false);
const callback = (message) => {
if (message.status === this.baseCallStates.failed) {
let shouldTryReconnect = false;
// Try and get the call to clean up and end on an error
this.activeSession.exitAudio().catch(() => {});
if (this.activeSession.webrtcConnected) {
// webrtc was able to connect so just try again
message.silenceNotifications = true;
callback({ status: this.baseCallStates.reconnecting });
shouldTryReconnect = true;
} else if (hasFallbackDomain === true && hostname !== IPV4_FALLBACK_DOMAIN) {
message.silenceNotifications = true;
logger.info({ logCode: 'sip_js_attempt_ipv4_fallback', extraInfo: { callerIdName: this.user.callerIdName } }, 'Attempting to fallback to IPv4 domain for audio');
hostname = IPV4_FALLBACK_DOMAIN;
shouldTryReconnect = true;
}
if (shouldTryReconnect) {
const fallbackExtension = this.activeSession.inEchoTest ? extension : undefined;
this.activeSession = new SIPSession(this.user, this.userData, this.protocol,
hostname, this.baseCallStates, this.baseErrorCodes, true);
this.activeSession.joinAudio({ isListenOnly, extension: fallbackExtension, inputStream }, callback)
.then((value) => {
resolve(value);
}).catch((reason) => {
reject(reason);
});
}
}
return managerCallback(message);
};
this.activeSession.joinAudio({ isListenOnly, extension, inputStream }, callback)
.then((value) => {
resolve(value);
}).catch((reason) => {
reject(reason);
});
});
}
transferCall(onTransferSuccess) {
return this.activeSession.transferCall(onTransferSuccess);
}
getPeerConnection() {
const { currentSession } = this.activeSession;
if (currentSession && currentSession.sessionDescriptionHandler) {
return currentSession.sessionDescriptionHandler.peerConnection;
}
return null;
}
exitAudio() {
return this.activeSession.exitAudio();
}
setDefaultInputDevice() {
// kept for compatibility
return Promise.resolve();
}
changeInputDevice(deviceId, deviceLabel) {
const {
media,
} = this;
if (media.inputDevice.audioContext) {
const handleAudioContextCloseSuccess = () => {
media.inputDevice.audioContext = null;
media.inputDevice.scriptProcessor = null;
media.inputDevice.source = null;
return this.changeInputDevice(deviceId);
};
return media.inputDevice.audioContext.close().then(handleAudioContextCloseSuccess);
}
if ('AudioContext' in window) {
media.inputDevice.audioContext = new window.AudioContext();
} else {
media.inputDevice.audioContext = new window.webkitAudioContext();
}
media.inputDevice.id = deviceId;
media.inputDevice.label = deviceLabel;
media.inputDevice.scriptProcessor = media.inputDevice.audioContext
.createScriptProcessor(2048, 1, 1);
media.inputDevice.source = null;
const constraints = {
audio: {
deviceId,
},
};
const handleMediaSuccess = (mediaStream) => {
media.inputDevice.stream = mediaStream;
media.inputDevice.source = media.inputDevice.audioContext
.createMediaStreamSource(mediaStream);
media.inputDevice.source.connect(media.inputDevice.scriptProcessor);
media.inputDevice.scriptProcessor.connect(media.inputDevice.audioContext.destination);
return this.media.inputDevice;
};
return navigator.mediaDevices.getUserMedia(constraints).then(handleMediaSuccess);
}
async changeOutputDevice(value) {
const audioContext = document.querySelector(MEDIA_TAG);
if (audioContext.setSinkId) {
try {
audioContext.srcObject = null;
await audioContext.setSinkId(value);
this.media.outputDeviceId = value;
} catch (err) {
logger.error({
logCode: 'audio_sip_changeoutputdevice_error',
extraInfo: { error: err, callerIdName: this.user.callerIdName },
}, 'Change Output Device error');
throw new Error(this.baseErrorCodes.MEDIA_ERROR);
}
}
return this.media.outputDeviceId || value;
}
}