bigbluebutton-Github/bigbluebutton-html5/imports/api/audio/client/bridge/sip.js
prlanzarin b17ba35238 fix(audio): decouple remote media setup (play) from state callback
Audio state callback and remote media setup both depend on FS's state
(comes through Meteor) and the ICE state (local, peer connection). The
caveat: FS's state can come delayed on reconnection scenarios because
Meteor's websocket generally takes significantly longer to re-connect than
the peer connection, which means the ICE state gets completed way before FS
is flagged as ready.
The practical issue: while outbound audio (client -> FS) will work, inbound
audio (FS -> client) won't _just because it wasn't played_ (even though
data is  coming through).

This commit decouples the remote media setup step from the state
through:
  - Setup remote media when ICE state is completed
  - Run the state callback only after FS is flagged as ready. This
    should maintain the UI states consistent across client-server.
    Keep in mind the assumption that if FS is ready, ICE is completed by
    consequence.
2023-04-11 16:02:20 -03:00

1316 lines
40 KiB
JavaScript
Executable File

import BaseAudioBridge from './base';
import logger from '/imports/startup/client/logger';
import {
fetchWebRTCMappedStunTurnServers,
getMappedFallbackStun,
} from '/imports/utils/fetchStunTurnServers';
import {
isUnifiedPlan,
toUnifiedPlan,
toPlanB,
stripMDnsCandidates,
filterValidIceCandidates,
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';
import browserInfo from '/imports/utils/browserInfo';
import {
getCurrentAudioSessionNumber,
getAudioSessionNumber,
getAudioConstraints,
filterSupportedConstraints,
doGUM,
} from '/imports/api/audio/client/bridge/service';
const MEDIA = Meteor.settings.public.media;
const MEDIA_TAG = MEDIA.mediaTag;
const CALL_HANGUP_TIMEOUT = MEDIA.callHangupTimeout;
const CALL_HANGUP_MAX_RETRIES = MEDIA.callHangupMaximumRetries;
const SIPJS_HACK_VIA_WS = MEDIA.sipjsHackViaWs;
const SIPJS_ALLOW_MDNS = MEDIA.sipjsAllowMdns || false;
const IPV4_FALLBACK_DOMAIN = Meteor.settings.public.app.ipv4FallbackDomain;
const CALL_CONNECT_TIMEOUT = 20000;
const ICE_NEGOTIATION_TIMEOUT = 20000;
const USER_AGENT_RECONNECTION_ATTEMPTS = MEDIA.audioReconnectionAttempts || 3;
const USER_AGENT_RECONNECTION_DELAY_MS = MEDIA.audioReconnectionDelay || 5000;
const USER_AGENT_CONNECTION_TIMEOUT_MS = MEDIA.audioConnectionTimeout || 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 SDP_SEMANTICS = MEDIA.sdpSemantics;
const FORCE_RELAY = MEDIA.forceRelay;
const UA_SERVER_VERSION = Meteor.settings.public.app.bbbServerVersion;
const UA_CLIENT_VERSION = Meteor.settings.public.app.html5ClientBuild;
/**
* 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 {
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.bridgeName = BRIDGE_NAME;
this._inputDeviceId = null;
this._outputDeviceId = null;
this._hangupFlag = false;
this._reconnecting = false;
this._currentSessionState = null;
this._ignoreCallState = false;
this.mediaStreamFactory = this.mediaStreamFactory.bind(this)
}
get inputStream() {
if (this.currentSession && this.currentSession.sessionDescriptionHandler) {
return this.currentSession.sessionDescriptionHandler.localMediaStream;
}
return null;
}
/**
* Set the input stream for the peer that represents the current session.
* Internally, this will call the sender's replaceTrack function.
* @param {MediaStream} stream The MediaStream object to be used as input
* stream
* @return {Promise} A Promise that is resolved with the
* MediaStream object that was set.
*/
setInputStream(stream) {
if (!this.currentSession?.sessionDescriptionHandler) return null;
return this.currentSession.sessionDescriptionHandler.setLocalMediaStream(stream);
}
get inputDeviceId() {
if (!this._inputDeviceId) {
const stream = this.inputStream;
if (stream) {
const track = stream.getAudioTracks().find(
(t) => t.getSettings().deviceId,
);
if (track && (typeof track.getSettings === 'function')) {
const { deviceId } = track.getSettings();
this._inputDeviceId = deviceId;
}
}
}
return this._inputDeviceId;
}
set inputDeviceId(deviceId) {
this._inputDeviceId = deviceId;
}
get outputDeviceId() {
if (!this._outputDeviceId) {
const audioElement = document.querySelector(MEDIA_TAG);
if (audioElement) {
this._outputDeviceId = audioElement.sinkId;
}
}
return this._outputDeviceId;
}
set outputDeviceId(deviceId) {
this._outputDeviceId = deviceId;
}
/**
* This _ignoreCallState flag is set to true when we want to ignore SIP's
* call state retrieved directly from FreeSWITCH ESL, when doing some checks
* (for example , when checking if call stopped).
* We need to ignore this , for example, when moderator is in
* breakout audio transfer ("Join Audio" button in breakout panel): in this
* case , we will monitor moderator's lifecycle in audio conference by
* using the SIP state taken from SIP.js only (ignoring the ESL's call state).
* @param {boolean} value true to ignore call state, false otherwise.
*/
set ignoreCallState(value) {
this._ignoreCallState = value;
}
get ignoreCallState() {
return this._ignoreCallState;
}
joinAudio({
isListenOnly,
extension,
inputDeviceId,
outputDeviceId,
validIceCandidates,
inputStream,
}, managerCallback) {
return new Promise((resolve, reject) => {
const callExtension = extension ? `${extension}${this.userData.voiceBridge}` : this.userData.voiceBridge;
this.ignoreCallState = false;
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;
this.validIceCandidates = validIceCandidates;
return this.doCall({
callExtension,
isListenOnly,
inputDeviceId,
outputDeviceId,
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 getMappedFallbackStun();
}
}
doCall(options) {
const {
isListenOnly,
inputDeviceId,
outputDeviceId,
inputStream,
} = options;
this.inputDeviceId = inputDeviceId;
this.outputDeviceId = outputDeviceId;
// If a valid MediaStream was provided it means it was preloaded somewhere
// else - let's use it so we don't call gUM needlessly
if (inputStream && inputStream.active) this.preloadedInputStream = inputStream;
const {
userId,
name,
sessionToken,
} = this.user;
const callerIdName = [
`${userId}_${getAudioSessionNumber()}`,
'bbbID',
isListenOnly ? `LISTENONLY-${name}` : name,
].join('-').replace(/"/g, "'");
this.user.callerIdName = callerIdName;
this.callOptions = options;
return this.getIceServers(sessionToken)
.then(this.createUserAgent.bind(this))
.then(this.inviteUserAgent.bind(this));
}
/**
*
* sessionSupportRTPPayloadDtmf
* tells if browser support RFC4733 DTMF.
* Safari 13 doens't support it yet
*/
sessionSupportRTPPayloadDtmf(session) {
try {
const sessionDescriptionHandler = session
? session.sessionDescriptionHandler
: this.currentSession.sessionDescriptionHandler;
const senders = sessionDescriptionHandler.peerConnection.getSenders();
return !!(senders[0].dtmf);
} catch (error) {
return false;
}
}
/**
* sendDtmf - send DTMF Tones using INFO message
*
* same as SimpleUser's dtmf
*/
sendDtmf(tone) {
const dtmf = tone;
const duration = 2000;
const body = {
contentDisposition: 'render',
contentType: 'application/dtmf-relay',
content: `Signal=${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.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);
}
hangupRetries += 1;
setTimeout(() => {
if (hangupRetries > CALL_HANGUP_MAX_RETRIES) {
this.callback({
status: this.baseCallStates.failed,
error: 1006,
bridgeError: 'Timeout on call hangup',
bridge: this.bridgeName,
});
return reject(this.baseErrorCodes.REQUEST_TIMEOUT);
}
if (!this._hangupFlag) return tryHangup();
return resolve();
}, CALL_HANGUP_TIMEOUT);
};
return tryHangup();
});
}
stopUserAgent() {
if (this.userAgent && (typeof this.userAgent.stop === 'function')) {
return this.userAgent.stop();
}
return Promise.resolve();
}
onBeforeUnload() {
this.userRequestedHangup = true;
return this.stopUserAgent();
}
mediaStreamFactory(constraints) {
if (this.preloadedInputStream && this.preloadedInputStream.active) {
return Promise.resolve(this.preloadedInputStream);
}
// The rest of this mimicks the default factory behavior.
if (!constraints.audio && !constraints.video) {
return Promise.resolve(new MediaStream());
}
return doGUM(constraints, true);
}
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}`;
// Create session description handler factory
const customSDHFactory = SIP.Web.defaultSessionDescriptionHandlerFactory(this.mediaStreamFactory);
this.userAgent = new SIP.UserAgent({
uri: SIP.UserAgent.makeURI(`sip:${encodeURIComponent(callerIdName)}@${hostname}`),
transportOptions: {
server: `${(protocol === 'https:' ? 'wss://' : 'ws://')}${hostname}/ws?${token}`,
connectionTimeout: USER_AGENT_CONNECTION_TIMEOUT_MS,
keepAliveInterval: WEBSOCKET_KEEP_ALIVE_INTERVAL,
keepAliveDebounce: WEBSOCKET_KEEP_ALIVE_DEBOUNCE,
traceSip: TRACE_SIP,
},
sessionDescriptionHandlerFactory: customSDHFactory,
sessionDescriptionHandlerFactoryOptions: {
peerConnectionConfiguration: {
iceServers,
sdpSemantics: SDP_SEMANTICS,
iceTransportPolicy: FORCE_RELAY ? 'relay' : undefined,
},
},
displayName: callerIdName,
register: false,
userAgentString: `BigBlueButton/${UA_SERVER_VERSION} (HTML5, rv:${UA_CLIENT_VERSION}) ${window.navigator.userAgent}`,
hackViaWs: SIPJS_HACK_VIA_WS,
});
const handleUserAgentConnection = () => {
if (!userAgentConnected) {
userAgentConnected = true;
resolve(this.userAgent);
}
};
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: this.bridgeName,
});
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: this.bridgeName,
});
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: this.bridgeName,
});
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;
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);
});
});
}
isValidIceCandidate(event) {
return event.candidate
&& this.validIceCandidates
&& this.validIceCandidates.find((validCandidate) => (
(validCandidate.address === event.candidate.address)
|| (validCandidate.relatedAddress === event.candidate.address))
&& (validCandidate.protocol === event.candidate.protocol));
}
onIceGatheringStateChange(event) {
const iceGatheringState = event.target
? event.target.iceGatheringState
: null;
if ((iceGatheringState === 'gathering') && (!this._iceGatheringStartTime)) {
this._iceGatheringStartTime = new Date();
}
if (iceGatheringState === 'complete') {
const secondsToGatherIce = (new Date()
- (this._iceGatheringStartTime || this._sessionStartTime)) / 1000;
logger.info({
logCode: 'sip_js_ice_gathering_time',
extraInfo: {
callerIdName: this.user.callerIdName,
secondsToGatherIce,
},
}, `ICE gathering candidates took (s): ${secondsToGatherIce}`);
}
}
onIceCandidate(sessionDescriptionHandler, event) {
if (this.isValidIceCandidate(event)) {
logger.info({
logCode: 'sip_js_found_valid_candidate_from_trickle_ice',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'Found a valid candidate from trickle ICE, finishing gathering');
if (sessionDescriptionHandler.iceGatheringCompleteResolve) {
sessionDescriptionHandler.iceGatheringCompleteResolve();
}
}
}
initSessionDescriptionHandler(sessionDescriptionHandler) {
/* eslint-disable no-param-reassign */
sessionDescriptionHandler.peerConnectionDelegate = {
onicecandidate:
this.onIceCandidate.bind(this, sessionDescriptionHandler),
onicegatheringstatechange:
this.onIceGatheringStateChange.bind(this),
};
/* eslint-enable no-param-reassign */
}
inviteUserAgent(userAgent) {
return new Promise((resolve, reject) => {
if (this.userRequestedHangup === true) reject();
const {
hostname,
} = this;
const {
callExtension,
isListenOnly,
} = this.callOptions;
this._sessionStartTime = new Date();
const target = SIP.UserAgent.makeURI(`sip:${callExtension}@${hostname}`);
const matchConstraints = getAudioConstraints({ deviceId: this.inputDeviceId });
const iceModifiers = [
filterValidIceCandidates.bind(this, this.validIceCandidates),
];
if (!SIPJS_ALLOW_MDNS) iceModifiers.push(stripMDnsCandidates);
const inviterOptions = {
sessionDescriptionHandlerOptions: {
constraints: {
audio: isListenOnly
? false
: matchConstraints,
video: false,
},
iceGatheringTimeout: ICE_GATHERING_TIMEOUT,
},
sessionDescriptionHandlerModifiersPostICEGathering: iceModifiers,
delegate: {
onSessionDescriptionHandler:
this.initSessionDescriptionHandler.bind(this),
},
};
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;
let sessionTerminated = false;
const setupRemoteMedia = () => {
const mediaElement = document.querySelector(MEDIA_TAG);
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.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) {
this.webrtcConnected = true;
setupRemoteMedia();
}
if (fsReady) {
this.callback({ status: this.baseCallStates.started, bridge: this.bridgeName });
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 ${CALL_CONNECT_TIMEOUT / 1000}s`,
bridge: this.bridgeName,
});
this.exitAudio();
}, CALL_CONNECT_TIMEOUT);
let iceNegotiationTimeout;
const handleSessionAccepted = () => {
logger.info({ logCode: 'sip_js_session_accepted', extraInfo: { callerIdName: this.user.callerIdName } }, 'Audio call session accepted');
clearTimeout(callTimeout);
// If ICE isn't connected yet then start timeout waiting for ICE to finish
if (!iceCompleted) {
iceNegotiationTimeout = setTimeout(() => {
this.callback({
status: this.baseCallStates.failed,
error: 1010,
bridgeError: 'ICE negotiation timeout after '
+ `${ICE_NEGOTIATION_TIMEOUT / 1000}s`,
bridge: this.bridgeName,
});
this.exitAudio();
reject({
type: this.baseErrorCodes.CONNECTION_ERROR,
});
}, ICE_NEGOTIATION_TIMEOUT);
}
checkIfCallReady();
};
const handleIceNegotiationFailed = (peer) => {
if (iceCompleted) {
logger.warn({
logCode: 'sipjs_ice_failed_after',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'ICE connection failed after success');
} else {
logger.warn({
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 '
+ `- ${peer.iceConnectionState}`,
bridge: this.bridgeName,
});
};
const handleIceConnectionTerminated = (peer) => {
if (!this.userRequestedHangup) {
logger.warn({
logCode: 'sipjs_ice_closed',
extraInfo: {
callerIdName: this.user.callerIdName,
},
}, 'ICE connection closed');
} else return;
this.callback({
status: this.baseCallStates.failed,
error: 1012,
bridgeError: 'ICE connection closed. Current state -'
+ `${peer.iceConnectionState}`,
bridge: this.bridgeName,
});
};
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 - '
+ `${peer.connectionState}`);
switch (peer.connectionState) {
case 'failed':
// Chrome triggers 'failed' for connectionState event, only
handleIceNegotiationFailed(peer);
break;
default:
break;
}
};
this.currentSession.sessionDescriptionHandler.peerConnectionDelegate
.oniceconnectionstatechange = (event) => {
const peer = event.target;
switch (peer.iceConnectionState) {
case 'completed':
case 'connected':
if (iceCompleted) {
logger.info({
logCode: 'sip_js_ice_connection_success_after_success',
extraInfo: {
currentState: peer.connectionState,
callerIdName: this.user.callerIdName,
},
}, 'ICE connection success, but user is already connected, '
+ 'ignoring it...'
+ `${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 - '
+ `${peer.iceConnectionState}`);
clearTimeout(callTimeout);
clearTimeout(iceNegotiationTimeout);
iceCompleted = true;
logSelectedCandidate(peer, this.protocolIsIpv6);
checkIfCallReady();
break;
case 'failed':
handleIceNegotiationFailed(peer);
break;
case 'closed':
handleIceConnectionTerminated(peer);
break;
default:
break;
}
};
};
const checkIfCallStopped = (message) => {
if ((!this.ignoreCallState && fsReady) || !sessionTerminated) {
return null;
}
if (!message && !!this.userRequestedHangup) {
return this.callback({
status: this.baseCallStates.ended,
bridge: this.bridgeName,
});
}
// if session hasn't even started, we let audio-modal to handle
// any possile errors
if (!this._currentSessionState) return false;
let mappedCause;
let cause;
if (!iceCompleted) {
mappedCause = '1004';
cause = 'ICE error';
} else {
cause = 'Audio Conference Error';
mappedCause = '1005';
}
logger.warn({
logCode: 'sip_js_call_terminated',
extraInfo: { cause, callerIdName: this.user.callerIdName },
}, `Audio call terminated. cause=${cause}`);
return this.callback({
status: this.baseCallStates.failed,
error: mappedCause,
bridgeError: cause,
bridge: this.bridgeName,
});
}
const handleSessionTerminated = (message) => {
logger.info({
logCode: 'sip_js_session_terminated',
extraInfo: { callerIdName: this.user.callerIdName },
}, 'SIP.js session terminated');
clearTimeout(callTimeout);
clearTimeout(iceNegotiationTimeout);
sessionTerminated = true;
checkIfCallStopped();
};
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.warn({
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,
clientSession: getCurrentAudioSessionNumber(),
};
const query = VoiceCallStates.find(selector);
const callback = (id, fields) => {
if (!fsReady && ((this.inEchoTest && fields.callState === CallStateOptions.IN_ECHO_TEST)
|| (!this.inEchoTest && fields.callState === CallStateOptions.IN_CONFERENCE))) {
fsReady = true;
checkIfCallReady();
}
if (fields.callState === CallStateOptions.CALL_ENDED) {
fsReady = false;
c.stop();
checkIfCallStopped();
}
};
query.observeChanges({
added: (id, fields) => callback(id, fields),
changed: (id, fields) => callback(id, fields),
});
});
resolve();
});
}
/**
* 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 = filterSupportedConstraints(constraints);
//Chromium bug - see: https://bugs.chromium.org/p/chromium/issues/detail?id=796964&q=applyConstraints&can=2
const { isChrome } = browserInfo;
if (isChrome) {
matchConstraints.deviceId = this.inputDeviceId;
const stream = await doGUM({ 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');
}
}
}
export default class SIPBridge extends BaseAudioBridge {
constructor(userData) {
super(userData);
const {
userId,
username,
sessionToken,
} = userData;
this.user = {
userId,
sessionToken,
name: username,
};
this.protocol = window.document.location.protocol;
if (MEDIA['sip_ws_host'] != null && MEDIA['sip_ws_host'] != '') {
this.hostname = MEDIA.sip_ws_host;
} else {
this.hostname = window.document.location.hostname;
}
this.bridgeName = BRIDGE_NAME;
// 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 inputStream() {
return this.activeSession ? this.activeSession.inputStream : null;
}
/**
* Wrapper for SIPSession's ignoreCallState flag
* @param {boolean} value
*/
set ignoreCallState(value) {
if (this.activeSession) {
this.activeSession.ignoreCallState = value;
}
}
get ignoreCallState() {
return this.activeSession ? this.activeSession.ignoreCallState : false;
}
joinAudio({
isListenOnly,
extension,
validIceCandidates,
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, bridge: this.bridgeName, });
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, outputDeviceId } = this;
this.activeSession.joinAudio({
isListenOnly,
extension: fallbackExtension,
inputDeviceId,
outputDeviceId,
validIceCandidates,
inputStream,
}, callback)
.then((value) => {
resolve(value);
}).catch((reason) => {
reject(reason);
});
}
}
return managerCallback(message);
};
const { inputDeviceId, outputDeviceId } = this;
this.activeSession.joinAudio({
isListenOnly,
extension,
inputDeviceId,
outputDeviceId,
validIceCandidates,
inputStream,
}, callback)
.then((value) => {
resolve(value);
}).catch((reason) => {
reject(reason);
});
});
}
transferCall(onTransferSuccess) {
this.activeSession.inEchoTest = false;
logger.debug({
logCode: 'sip_js_rtp_payload_send_dtmf',
extraInfo: {
callerIdName: this.activeSession.user.callerIdName,
},
}, 'Sending DTMF INFO to transfer user');
return this.trackTransferState(onTransferSuccess);
}
sendDtmf(tones) {
this.activeSession.sendDtmf(tones);
}
getPeerConnection() {
if (!this.activeSession) return null;
const { currentSession } = this.activeSession;
if (currentSession && currentSession.sessionDescriptionHandler) {
return currentSession.sessionDescriptionHandler.peerConnection;
}
return null;
}
exitAudio() {
if (this.activeSession == null) return Promise.resolve();
return this.activeSession.exitAudio();
}
setInputStream(stream) {
return this.activeSession.setInputStream(stream);
}
async updateAudioConstraints(constraints) {
return this.activeSession.updateAudioConstraints(constraints);
}
}
module.exports = SIPBridge;