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

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import browser from 'browser-detect';
import BaseAudioBridge from './base';
import logger from '/imports/startup/client/logger';
import {
fetchWebRTCMappedStunTurnServers,
getMappedFallbackStun,
} from '/imports/utils/fetchStunTurnServers';
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import {
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isUnifiedPlan,
toUnifiedPlan,
toPlanB,
stripMDnsCandidates,
analyzeSdp,
logSelectedCandidate,
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} 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';
import Settings from '/imports/ui/services/settings';
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const MEDIA = Meteor.settings.public.media;
const MEDIA_TAG = MEDIA.mediaTag;
const CALL_TRANSFER_TIMEOUT = MEDIA.callTransferTimeout;
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const CALL_HANGUP_TIMEOUT = MEDIA.callHangupTimeout;
const CALL_HANGUP_MAX_RETRIES = MEDIA.callHangupMaximumRetries;
const SIPJS_HACK_VIA_WS = MEDIA.sipjsHackViaWs;
const IPV4_FALLBACK_DOMAIN = Meteor.settings.public.app.ipv4FallbackDomain;
const CALL_CONNECT_TIMEOUT = 20000;
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const ICE_NEGOTIATION_TIMEOUT = 20000;
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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 ICE_GATHERING_TIMEOUT = MEDIA.iceGatheringTimeout || 5000;
const BRIDGE_NAME = 'sip';
const WEBSOCKET_KEEP_ALIVE_INTERVAL = MEDIA.websocketKeepAliveInterval || 0;
const WEBSOCKET_KEEP_ALIVE_DEBOUNCE = MEDIA.websocketKeepAliveDebounce || 10;
const TRACE_SIP = MEDIA.traceSip || false;
const AUDIO_MICROPHONE_CONSTRAINTS = Meteor.settings.public.app.defaultSettings
.application.microphoneConstraints;
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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;
};
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/**
* Get error code from SIP.js websocket messages.
*/
const getErrorCode = (error) => {
try {
if (!error) return error;
const match = error.message.match(/code: \d+/g);
const _codeArray = match[0].split(':');
return parseInt(_codeArray[1].trim(), 10);
} catch (e) {
return 0;
}
};
class SIPSession {
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constructor(user, userData, protocol, hostname,
baseCallStates, baseErrorCodes, reconnectAttempt) {
this.user = user;
this.userData = userData;
this.protocol = protocol;
this.hostname = hostname;
this.baseCallStates = baseCallStates;
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this.baseErrorCodes = baseErrorCodes;
this.reconnectAttempt = reconnectAttempt;
this.currentSession = null;
this.remoteStream = null;
this.inputDeviceId = null;
this._hangupFlag = false;
this._reconnecting = false;
this._currentSessionState = null;
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}
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get inputStream() {
if (this.currentSession && this.currentSession.sessionDescriptionHandler) {
return this.currentSession.sessionDescriptionHandler.localMediaStream;
}
return null;
}
joinAudio({ isListenOnly, extension, inputDeviceId }, managerCallback) {
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return new Promise((resolve, reject) => {
const callExtension = extension ? `${extension}${this.userData.voiceBridge}` : this.userData.voiceBridge;
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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;
}
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managerCallback(message).then(resolve);
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};
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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, inputDeviceId })
.catch((reason) => {
reject(reason);
});
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});
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}
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 getMappedFallbackStun();
}
}
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doCall(options) {
const {
isListenOnly,
inputDeviceId,
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} = options;
this.inputDeviceId = inputDeviceId;
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const {
userId,
name,
sessionToken,
} = this.user;
const callerIdName = [
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`${userId}_${getAudioSessionNumber()}`,
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'bbbID',
isListenOnly ? `LISTENONLY-${name}` : name,
].join('-').replace(/"/g, "'");
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this.user.callerIdName = callerIdName;
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this.callOptions = options;
return this.getIceServers(sessionToken)
.then(this.createUserAgent.bind(this))
.then(this.inviteUserAgent.bind(this));
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}
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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');
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this.callback({
status: this.baseCallStates.failed,
error: 1008,
bridgeError: 'Timeout on call transfer',
bridge: BRIDGE_NAME,
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});
this.exitAudio();
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reject(this.baseErrorCodes.REQUEST_TIMEOUT);
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}, CALL_TRANSFER_TIMEOUT);
// This is is the call transfer code ask @chadpilkey
logger.debug({
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logCode: 'sip_js_rtp_payload_send_dtmf',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'Sending DTMF INFO to transfer user');
this.sendDtmf(1);
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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 });
}
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exitAudio() {
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return new Promise((resolve, reject) => {
let hangupRetries = 0;
this._hangupFlag = false;
this.userRequestedHangup = true;
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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.cancel().then(() => {
this._hangupFlag = true;
return resolve();
});
}
if (this.currentSession
&& ((this.currentSession.state === SIP.SessionState.Established))) {
this.currentSession.bye().then(() => {
this._hangupFlag = true;
return resolve();
});
}
if (this.userAgent && this.userAgent.isConnected()) {
this.userAgent.stop();
window.removeEventListener('beforeunload', this.onBeforeUnload);
}
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hangupRetries += 1;
setTimeout(() => {
if (hangupRetries > CALL_HANGUP_MAX_RETRIES) {
this.callback({
status: this.baseCallStates.failed,
error: 1006,
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bridgeError: 'Timeout on call hangup',
bridge: BRIDGE_NAME,
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});
return reject(this.baseErrorCodes.REQUEST_TIMEOUT);
}
if (!this._hangupFlag) return tryHangup();
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return resolve();
}, CALL_HANGUP_TIMEOUT);
};
return tryHangup();
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});
}
stopUserAgent() {
if (this.userAgent && (typeof this.userAgent.stop === 'function')) {
return this.userAgent.stop();
}
return Promise.resolve();
}
onBeforeUnload() {
this.userRequestedHangup = true;
return this.stopUserAgent();
}
createUserAgent(iceServers) {
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return new Promise((resolve, reject) => {
if (this.userRequestedHangup === true) reject();
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const {
hostname,
protocol,
} = this;
const {
callerIdName,
sessionToken,
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} = 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);
};
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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,
keepAliveInterval: WEBSOCKET_KEEP_ALIVE_INTERVAL,
keepAliveDebounce: WEBSOCKET_KEEP_ALIVE_DEBOUNCE,
traceSip: TRACE_SIP,
},
sessionDescriptionHandlerFactoryOptions: {
peerConnectionConfiguration: {
iceServers,
sdpSemantics: 'plan-b',
},
},
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displayName: callerIdName,
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register: false,
userAgentString: 'BigBlueButton',
hackViaWs: SIPJS_HACK_VIA_WS,
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});
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const handleUserAgentConnection = () => {
if (!userAgentConnected) {
userAgentConnected = true;
resolve(this.userAgent);
}
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};
const handleUserAgentDisconnection = () => {
if (this.userAgent) {
if (this.userRequestedHangup) {
userAgentConnected = false;
return;
}
let error;
let bridgeError;
if (!this._reconnecting) {
logger.info({
logCode: 'sip_js_session_ua_disconnected',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'User agent disconnected: trying to reconnect...'
+ ` (userHangup = ${!!this.userRequestedHangup})`);
logger.info({
logCode: 'sip_js_session_ua_reconnecting',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'User agent disconnected, reconnecting');
this.reconnect().then(() => {
logger.info({
logCode: 'sip_js_session_ua_reconnected',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'User agent succesfully reconnected');
}).catch(() => {
if (userAgentConnected) {
error = 1001;
bridgeError = 'Websocket disconnected';
} else {
error = 1002;
bridgeError = 'Websocket failed to connect';
}
this.stopUserAgent();
this.callback({
status: this.baseCallStates.failed,
error,
bridgeError,
bridge: BRIDGE_NAME,
});
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');
window.addEventListener('beforeunload', this.onBeforeUnload.bind(this));
resolve();
}).catch((error) => {
logger.info({
logCode: 'sip_js_session_ua_reconnecting',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'User agent failed to connect, reconnecting');
const code = getErrorCode(error);
// Websocket's 1006 is currently mapped to BBB's 1002
if (code === 1006) {
this.stopUserAgent();
this.callback({
status: this.baseCallStates.failed,
error: 1002,
bridgeError: 'Websocket failed to connect',
bridge: BRIDGE_NAME,
});
return reject({
type: this.baseErrorCodes.CONNECTION_ERROR,
});
}
this.reconnect().then(() => {
logger.info({
logCode: 'sip_js_session_ua_reconnected',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'User agent succesfully reconnected');
resolve();
}).catch(() => {
this.stopUserAgent();
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',
bridge: BRIDGE_NAME,
});
reject({
type: this.baseErrorCodes.CONNECTION_ERROR,
});
});
});
return preturn;
});
}
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reconnect(attempts = 1) {
return new Promise((resolve, reject) => {
if (this._reconnecting) {
return resolve();
}
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if (attempts > USER_AGENT_RECONNECTION_ATTEMPTS) {
return reject({
type: this.baseErrorCodes.CONNECTION_ERROR,
});
}
this._reconnecting = true;
logger.info({
logCode: 'sip_js_session_ua_reconnection_attempt',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, `User agent reconnection attempt ${attempts}`);
this.userAgent.reconnect().then(() => {
this._reconnecting = false;
resolve();
}).catch(() => {
setTimeout(() => {
this._reconnecting = false;
this.reconnect(++attempts).then(() => {
resolve();
}).catch((error) => {
reject(error);
});
}, USER_AGENT_RECONNECTION_DELAY_MS);
});
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});
}
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inviteUserAgent(userAgent) {
return new Promise((resolve, reject) => {
if (this.userRequestedHangup === true) reject();
const {
hostname,
} = this;
const {
callExtension,
isListenOnly,
} = this.callOptions;
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const target = SIP.UserAgent.makeURI(`sip:${callExtension}@${hostname}`);
const userSettingsConstraints = Settings.application.microphoneConstraints;
const audioDeviceConstraints = userSettingsConstraints
|| AUDIO_MICROPHONE_CONSTRAINTS || {};
const matchConstraints = this.filterSupportedConstraints(
audioDeviceConstraints,
);
if (this.inputDeviceId) {
matchConstraints.deviceId = this.inputDeviceId;
}
const inviterOptions = {
sessionDescriptionHandlerOptions: {
constraints: {
audio: isListenOnly
? false
: matchConstraints,
video: false,
},
iceGatheringTimeout: ICE_GATHERING_TIMEOUT,
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},
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));
});
});
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}
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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();
};
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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, bridge: BRIDGE_NAME });
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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 ${CALL_CONNECT_TIMEOUT / 1000}s`,
bridge: BRIDGE_NAME,
});
this.exitAudio();
}, CALL_CONNECT_TIMEOUT);
let iceNegotiationTimeout;
const handleSessionAccepted = () => {
logger.info({ logCode: 'sip_js_session_accepted', extraInfo: { callerIdName: this.user.callerIdName } }, 'Audio call session accepted');
clearTimeout(callTimeout);
// If ICE isn't connected yet then start timeout waiting for ICE to finish
if (!iceCompleted) {
iceNegotiationTimeout = setTimeout(() => {
this.callback({
status: this.baseCallStates.failed,
error: 1010,
bridgeError: 'ICE negotiation timeout after '
+ `${ICE_NEGOTIATION_TIMEOUT / 1000}s`,
bridge: BRIDGE_NAME,
});
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 '
+ `- ${peer.iceConnectionState}`,
bridge: BRIDGE_NAME,
});
};
const handleIceConnectionTerminated = (peer) => {
if (!this.userRequestedHangup) {
logger.error({
logCode: 'sipjs_ice_closed',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'ICE connection closed');
} else return;
this.callback({
status: this.baseCallStates.failed,
error: 1012,
bridgeError: 'ICE connection closed. Current state -'
+ `${peer.iceConnectionState}`,
bridge: BRIDGE_NAME,
});
};
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;
logger.info({
logCode: 'sip_js_connection_state_change',
extraInfo: {
connectionStateChange: peer.connectionState,
callerIdName: this.user.callerIdName,
},
}, 'ICE connection state change - Current connection state - '
+ `${peer.connectionState}`);
switch (peer.connectionState) {
case 'failed':
// Chrome triggers 'failed' for connectionState event, only
handleIceNegotiationFailed(peer);
break;
default:
break;
}
},
oniceconnectionstatechange: (event) => {
const peer = event.target;
switch (peer.iceConnectionState) {
case 'completed':
case 'connected':
if (iceCompleted) {
logger.info({
logCode: 'sip_js_ice_connection_success_after_success',
extraInfo: {
currentState: peer.connectionState,
callerIdName: this.user.callerIdName,
},
}, 'ICE connection success, but user is already connected'
+ 'ignoring it...'
+ `${peer.iceConnectionState}`);
return;
}
logger.info({
logCode: 'sip_js_ice_connection_success',
extraInfo: {
currentState: peer.connectionState,
callerIdName: this.user.callerIdName,
},
}, 'ICE connection success. Current ICE Connection state - '
+ `${peer.iceConnectionState}`);
clearTimeout(callTimeout);
clearTimeout(iceNegotiationTimeout);
iceCompleted = true;
logSelectedCandidate(peer, this.protocolIsIpv6);
checkIfCallReady();
break;
case 'failed':
handleIceNegotiationFailed(peer);
break;
case 'closed':
handleIceConnectionTerminated(peer);
break;
default:
break;
}
},
};
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};
const handleSessionTerminated = (message) => {
clearTimeout(callTimeout);
clearTimeout(iceNegotiationTimeout);
if (!message && !!this.userRequestedHangup) {
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return this.callback({
status: this.baseCallStates.ended,
bridge: BRIDGE_NAME,
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});
}
// if session hasn't even started, we let audio-modal to handle
// any possile errors
if (!this._currentSessionState) return false;
let mappedCause;
let cause;
if (!iceCompleted) {
mappedCause = '1004';
cause = 'ICE error';
} else {
cause = 'Audio Conference Error';
mappedCause = '1005';
}
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logger.error({
logCode: 'sip_js_call_terminated',
extraInfo: { cause, callerIdName: this.user.callerIdName },
}, `Audio call terminated. cause=${cause}`);
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return this.callback({
status: this.baseCallStates.failed,
error: mappedCause,
bridgeError: cause,
bridge: BRIDGE_NAME,
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});
};
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;
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}
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();
});
}
/**
* Filter constraints set in audioDeviceConstraints, based on
* constants supported by browser. This avoids setting a constraint
* unsupported by browser. In currently safari version (13+), for example,
* setting an unsupported constraint crashes the audio.
* @param {Object} audioDeviceConstraints Constraints to be set
* see: https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints
* @return {Object} A new Object of the same type as
* input, containing only the supported constraints.
*/
filterSupportedConstraints(audioDeviceConstraints) {
try {
const matchConstraints = {};
const supportedConstraints = navigator
.mediaDevices.getSupportedConstraints() || {};
Object.entries(audioDeviceConstraints).forEach(
([constraintName, constraintValue]) => {
if (supportedConstraints[constraintName]) {
matchConstraints[constraintName] = constraintValue;
}
}
);
return matchConstraints;
} catch (error) {
logger.error({
logCode: 'sipjs_unsupported_audio_constraint_error',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'SIP.js unsupported constraint error');
return {};
}
}
/**
* Update audio constraints for current local MediaStream (microphone)
* @param {Object} constraints MediaTrackConstraints object. See:
* https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackConstraints
* @return {Promise} A Promise for this process
*/
async updateAudioConstraints(constraints) {
try {
if (typeof constraints !== 'object') return;
logger.info({
logCode: 'sipjs_update_audio_constraint',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'SIP.js updating audio constraint');
const matchConstraints = this.filterSupportedConstraints(constraints);
//Chromium bug - see: https://bugs.chromium.org/p/chromium/issues/detail?id=796964&q=applyConstraints&can=2
if (browser().name === 'chrome') {
matchConstraints.deviceId = this.inputDeviceId;
const stream = await navigator.mediaDevices.getUserMedia(
{ audio: matchConstraints },
);
this.currentSession.sessionDescriptionHandler
.setLocalMediaStream(stream);
} else {
const { localMediaStream } = this.currentSession
.sessionDescriptionHandler;
localMediaStream.getAudioTracks().forEach(
track => track.applyConstraints(matchConstraints),
);
}
} catch (error) {
logger.error({
logCode: 'sipjs_audio_constraint_error',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'SIP.js failed to update audio constraint');
}
}
}
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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;
}
get inputDeviceId() {
return this.media.inputDevice ? this.media.inputDevice.inputDeviceId : null;
}
get inputStream() {
return this.activeSession ? this.activeSession.inputStream : null;
}
joinAudio({ isListenOnly, extension }, 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, bridge: BRIDGE_NAME, });
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);
const { inputDeviceId } = this.media.inputDevice;
this.activeSession.joinAudio({
isListenOnly,
extension: fallbackExtension,
inputDeviceId,
}, callback)
.then((value) => {
resolve(value);
}).catch((reason) => {
reject(reason);
});
}
}
return managerCallback(message);
};
const { inputDeviceId } = this.media.inputDevice;
this.activeSession.joinAudio({
isListenOnly,
extension,
inputDeviceId,
}, callback)
.then((value) => {
resolve(value);
}).catch((reason) => {
reject(reason);
});
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});
}
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() {
this.media.inputDevice.inputDeviceId = DEFAULT_INPUT_DEVICE_ID;
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}
async changeInputDeviceId(inputDeviceId) {
if (!inputDeviceId) {
throw new Error();
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}
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this.media.inputDevice.inputDeviceId = inputDeviceId;
return inputDeviceId;
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}
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liveChangeInputDevice(deviceId) {
const constraints = {
audio: {
deviceId,
},
};
return navigator.mediaDevices.getUserMedia(constraints).then((stream) => {
const peer = this.getPeerConnection();
const senders = peer.getSenders()[0];
const firstTrack = stream.getAudioTracks()[0];
senders.replaceTrack(firstTrack);
});
}
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async changeOutputDevice(value) {
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const audioContext = document.querySelector(MEDIA_TAG);
if (audioContext.setSinkId) {
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try {
await audioContext.setSinkId(value);
audioContext.load();
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this.media.outputDeviceId = value;
} catch (err) {
logger.error({
logCode: 'audio_sip_changeoutputdevice_error',
extraInfo: { error: err, callerIdName: this.user.callerIdName },
}, 'Change Output Device error');
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throw new Error(this.baseErrorCodes.MEDIA_ERROR);
}
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}
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return this.media.outputDeviceId || value;
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}
async updateAudioConstraints(constraints) {
return this.activeSession.updateAudioConstraints(constraints);
}
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}