1394 lines
41 KiB
JavaScript
Executable File
1394 lines
41 KiB
JavaScript
Executable File
import BaseAudioBridge from './base';
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import logger from '/imports/startup/client/logger';
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import {
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fetchWebRTCMappedStunTurnServers,
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getMappedFallbackStun,
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} from '/imports/utils/fetchStunTurnServers';
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import {
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isUnifiedPlan,
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toUnifiedPlan,
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toPlanB,
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stripMDnsCandidates,
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filterValidIceCandidates,
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analyzeSdp,
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logSelectedCandidate,
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} from '/imports/utils/sdpUtils';
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import { Tracker } from 'meteor/tracker';
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import VoiceCallStates from '/imports/api/voice-call-states';
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import CallStateOptions from '/imports/api/voice-call-states/utils/callStates';
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import Auth from '/imports/ui/services/auth';
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import Storage from '/imports/ui/services/storage/session';
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import browserInfo from '/imports/utils/browserInfo';
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import {
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getCurrentAudioSessionNumber,
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getAudioSessionNumber,
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getAudioConstraints,
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filterSupportedConstraints,
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} from '/imports/api/audio/client/bridge/service';
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const MEDIA = Meteor.settings.public.media;
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const MEDIA_TAG = MEDIA.mediaTag;
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const CALL_HANGUP_TIMEOUT = MEDIA.callHangupTimeout;
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const CALL_HANGUP_MAX_RETRIES = MEDIA.callHangupMaximumRetries;
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const SIPJS_HACK_VIA_WS = MEDIA.sipjsHackViaWs;
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const IPV4_FALLBACK_DOMAIN = Meteor.settings.public.app.ipv4FallbackDomain;
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const CALL_CONNECT_TIMEOUT = 20000;
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const ICE_NEGOTIATION_TIMEOUT = 20000;
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const USER_AGENT_RECONNECTION_ATTEMPTS = MEDIA.audioReconnectionAttempts || 3;
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const USER_AGENT_RECONNECTION_DELAY_MS = MEDIA.audioReconnectionDelay || 5000;
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const USER_AGENT_CONNECTION_TIMEOUT_MS = MEDIA.audioConnectionTimeout || 5000;
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const ICE_GATHERING_TIMEOUT = MEDIA.iceGatheringTimeout || 5000;
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const BRIDGE_NAME = 'sip';
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const WEBSOCKET_KEEP_ALIVE_INTERVAL = MEDIA.websocketKeepAliveInterval || 0;
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const WEBSOCKET_KEEP_ALIVE_DEBOUNCE = MEDIA.websocketKeepAliveDebounce || 10;
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const TRACE_SIP = MEDIA.traceSip || false;
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const SDP_SEMANTICS = MEDIA.sdpSemantics;
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const FORCE_RELAY = MEDIA.forceRelay;
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const DEFAULT_INPUT_DEVICE_ID = 'default';
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const DEFAULT_OUTPUT_DEVICE_ID = 'default';
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const INPUT_DEVICE_ID_KEY = 'audioInputDeviceId';
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const OUTPUT_DEVICE_ID_KEY = 'audioOutputDeviceId';
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const UA_SERVER_VERSION = Meteor.settings.public.app.bbbServerVersion;
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const UA_CLIENT_VERSION = Meteor.settings.public.app.html5ClientBuild;
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/**
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* Get error code from SIP.js websocket messages.
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*/
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const getErrorCode = (error) => {
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try {
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if (!error) return error;
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const match = error.message.match(/code: \d+/g);
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const _codeArray = match[0].split(':');
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return parseInt(_codeArray[1].trim(), 10);
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} catch (e) {
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return 0;
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}
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};
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class SIPSession {
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constructor(user, userData, protocol, hostname,
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baseCallStates, baseErrorCodes, reconnectAttempt) {
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this.user = user;
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this.userData = userData;
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this.protocol = protocol;
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this.hostname = hostname;
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this.baseCallStates = baseCallStates;
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this.baseErrorCodes = baseErrorCodes;
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this.reconnectAttempt = reconnectAttempt;
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this.currentSession = null;
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this.remoteStream = null;
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this.bridgeName = BRIDGE_NAME;
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this._inputDeviceId = null;
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this._outputDeviceId = null;
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this._hangupFlag = false;
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this._reconnecting = false;
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this._currentSessionState = null;
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this._ignoreCallState = false;
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}
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get inputStream() {
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if (this.currentSession && this.currentSession.sessionDescriptionHandler) {
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return this.currentSession.sessionDescriptionHandler.localMediaStream;
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}
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return null;
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}
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/**
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* Set the input stream for the peer that represents the current session.
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* Internally, this will call the sender's replaceTrack function.
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* @param {MediaStream} stream The MediaStream object to be used as input
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* stream
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* @return {Promise} A Promise that is resolved with the
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* MediaStream object that was set.
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*/
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async setInputStream(stream) {
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try {
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if (!this.currentSession
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|| !this.currentSession.sessionDescriptionHandler
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) return null;
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await this.currentSession.sessionDescriptionHandler
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.setLocalMediaStream(stream);
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return stream;
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} catch (error) {
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logger.warn({
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logCode: 'sip_js_setinputstream_error',
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extraInfo: {
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errorCode: error.code,
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errorMessage: error.message,
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callerIdName: this.user.callerIdName,
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},
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}, 'Failed to set input stream (mic)');
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return null;
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}
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}
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/**
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* Change the input device with the given deviceId, without renegotiating
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* peer.
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* A new MediaStream object is created for the given deviceId. This object
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* is returned by the resolved promise.
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* @param {String} deviceId The id of the device to be set as input
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* @return {Promise} A promise that is resolved with the MediaStream
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* object after changing the input device.
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*/
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async liveChangeInputDevice(deviceId) {
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try {
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this.inputDeviceId = deviceId;
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const constraints = {
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audio: getAudioConstraints({ deviceId: this.inputDeviceId }),
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};
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this.inputStream.getAudioTracks().forEach((t) => t.stop());
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return await navigator.mediaDevices.getUserMedia(constraints)
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.then(this.setInputStream.bind(this));
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} catch (error) {
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logger.warn({
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logCode: 'sip_js_livechangeinputdevice_error',
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extraInfo: {
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errorCode: error.code,
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errorMessage: error.message,
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callerIdName: this.user.callerIdName,
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},
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}, 'Failed to change input device (mic)');
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return null;
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}
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}
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get inputDeviceId() {
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if (!this._inputDeviceId) {
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const stream = this.inputStream;
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if (stream) {
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const track = stream.getAudioTracks().find(
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(t) => t.getSettings().deviceId,
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);
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if (track && (typeof track.getSettings === 'function')) {
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const { deviceId } = track.getSettings();
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this._inputDeviceId = deviceId;
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}
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}
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}
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return this._inputDeviceId;
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}
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set inputDeviceId(deviceId) {
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this._inputDeviceId = deviceId;
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}
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get outputDeviceId() {
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if (!this._outputDeviceId) {
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const audioElement = document.querySelector(MEDIA_TAG);
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if (audioElement) {
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this._outputDeviceId = audioElement.sinkId;
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}
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}
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return this._outputDeviceId;
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}
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set outputDeviceId(deviceId) {
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this._outputDeviceId = deviceId;
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}
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/**
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* This _ignoreCallState flag is set to true when we want to ignore SIP's
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* call state retrieved directly from FreeSWITCH ESL, when doing some checks
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* (for example , when checking if call stopped).
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* We need to ignore this , for example, when moderator is in
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* breakout audio transfer ("Join Audio" button in breakout panel): in this
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* case , we will monitor moderator's lifecycle in audio conference by
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* using the SIP state taken from SIP.js only (ignoring the ESL's call state).
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* @param {boolean} value true to ignore call state, false otherwise.
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*/
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set ignoreCallState(value) {
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this._ignoreCallState = value;
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}
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get ignoreCallState() {
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return this._ignoreCallState;
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}
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joinAudio({
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isListenOnly,
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extension,
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inputDeviceId,
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outputDeviceId,
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validIceCandidates,
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}, managerCallback) {
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return new Promise((resolve, reject) => {
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const callExtension = extension ? `${extension}${this.userData.voiceBridge}` : this.userData.voiceBridge;
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this.ignoreCallState = false;
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const callback = (message) => {
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// There will sometimes we erroneous errors put out like timeouts and improper shutdowns,
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// but only the first error ever matters
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if (this.alreadyErrored) {
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logger.info({
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logCode: 'sip_js_absorbing_callback_message',
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extraInfo: { message },
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}, 'Absorbing a redundant callback message.');
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return;
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}
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if (message.status === this.baseCallStates.failed) {
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this.alreadyErrored = true;
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}
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managerCallback(message).then(resolve);
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};
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this.callback = callback;
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// If there's an extension passed it means that we're joining the echo test first
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this.inEchoTest = !!extension;
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this.validIceCandidates = validIceCandidates;
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return this.doCall({
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callExtension,
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isListenOnly,
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inputDeviceId,
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outputDeviceId,
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}).catch((reason) => {
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reject(reason);
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});
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});
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}
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async getIceServers(sessionToken) {
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try {
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const iceServers = await fetchWebRTCMappedStunTurnServers(sessionToken);
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return iceServers;
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} catch (error) {
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logger.error({
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logCode: 'sip_js_fetchstunturninfo_error',
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extraInfo: {
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errorCode: error.code,
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errorMessage: error.message,
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callerIdName: this.user.callerIdName,
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},
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}, 'Full audio bridge failed to fetch STUN/TURN info');
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return getMappedFallbackStun();
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}
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}
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doCall(options) {
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const {
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isListenOnly,
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inputDeviceId,
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outputDeviceId,
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} = options;
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this.inputDeviceId = inputDeviceId;
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this.outputDeviceId = outputDeviceId;
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const {
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userId,
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name,
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sessionToken,
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} = this.user;
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const callerIdName = [
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`${userId}_${getAudioSessionNumber()}`,
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'bbbID',
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isListenOnly ? `LISTENONLY-${name}` : name,
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].join('-').replace(/"/g, "'");
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this.user.callerIdName = callerIdName;
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this.callOptions = options;
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return this.getIceServers(sessionToken)
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.then(this.createUserAgent.bind(this))
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.then(this.inviteUserAgent.bind(this));
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}
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/**
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*
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* sessionSupportRTPPayloadDtmf
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* tells if browser support RFC4733 DTMF.
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* Safari 13 doens't support it yet
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*/
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sessionSupportRTPPayloadDtmf(session) {
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try {
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const sessionDescriptionHandler = session
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? session.sessionDescriptionHandler
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: this.currentSession.sessionDescriptionHandler;
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const senders = sessionDescriptionHandler.peerConnection.getSenders();
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return !!(senders[0].dtmf);
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} catch (error) {
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return false;
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}
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}
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/**
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* sendDtmf - send DTMF Tones using INFO message
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*
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* same as SimpleUser's dtmf
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*/
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sendDtmf(tone) {
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const dtmf = tone;
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const duration = 2000;
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const body = {
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contentDisposition: 'render',
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contentType: 'application/dtmf-relay',
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content: `Signal=${dtmf}\r\nDuration=${duration}`,
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};
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const requestOptions = { body };
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return this.currentSession.info({ requestOptions });
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}
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exitAudio() {
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return new Promise((resolve, reject) => {
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let hangupRetries = 0;
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this._hangupFlag = false;
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this.userRequestedHangup = true;
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const tryHangup = () => {
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if (this._hangupFlag) {
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resolve();
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}
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if ((this.currentSession
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&& (this.currentSession.state === SIP.SessionState.Terminated))
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|| (this.userAgent && (!this.userAgent.isConnected()))) {
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this._hangupFlag = true;
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return resolve();
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}
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if (this.currentSession
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&& ((this.currentSession.state === SIP.SessionState.Establishing))) {
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this.currentSession.cancel().then(() => {
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this._hangupFlag = true;
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return resolve();
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});
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}
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if (this.currentSession
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&& ((this.currentSession.state === SIP.SessionState.Established))) {
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this.currentSession.bye().then(() => {
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this._hangupFlag = true;
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return resolve();
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});
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}
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if (this.userAgent && this.userAgent.isConnected()) {
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this.userAgent.stop();
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window.removeEventListener('beforeunload', this.onBeforeUnload);
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}
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hangupRetries += 1;
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setTimeout(() => {
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if (hangupRetries > CALL_HANGUP_MAX_RETRIES) {
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this.callback({
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status: this.baseCallStates.failed,
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error: 1006,
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bridgeError: 'Timeout on call hangup',
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bridge: this.bridgeName,
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});
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return reject(this.baseErrorCodes.REQUEST_TIMEOUT);
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}
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if (!this._hangupFlag) return tryHangup();
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return resolve();
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}, CALL_HANGUP_TIMEOUT);
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};
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return tryHangup();
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});
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}
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stopUserAgent() {
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if (this.userAgent && (typeof this.userAgent.stop === 'function')) {
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return this.userAgent.stop();
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}
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return Promise.resolve();
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}
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onBeforeUnload() {
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this.userRequestedHangup = true;
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return this.stopUserAgent();
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}
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createUserAgent(iceServers) {
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return new Promise((resolve, reject) => {
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if (this.userRequestedHangup === true) reject();
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const {
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hostname,
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protocol,
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} = this;
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const {
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callerIdName,
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sessionToken,
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} = this.user;
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logger.debug({ logCode: 'sip_js_creating_user_agent', extraInfo: { callerIdName } }, 'Creating the user agent');
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if (this.userAgent && this.userAgent.isConnected()) {
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if (this.userAgent.configuration.hostPortParams === this.hostname) {
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logger.debug({ logCode: 'sip_js_reusing_user_agent', extraInfo: { callerIdName } }, 'Reusing the user agent');
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resolve(this.userAgent);
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return;
|
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}
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logger.debug({ logCode: 'sip_js_different_host_name', extraInfo: { callerIdName } }, 'Different host name. need to kill');
|
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}
|
|
|
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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
|
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// and IPv4 candidates to make informed decisions about what to do on fallbacks/reconnects.
|
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analyzeSdp(sdp);
|
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};
|
|
|
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const remoteSdpCallback = (sdp) => {
|
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// We have have to find the candidate that FS sends back to us to determine if the client
|
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// is connecting with IPv4 or IPv6
|
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const sdpInfo = analyzeSdp(sdp, false);
|
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this.protocolIsIpv6 = sdpInfo.v6Info.found;
|
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};
|
|
|
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let userAgentConnected = false;
|
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const token = `sessionToken=${sessionToken}`;
|
|
|
|
this.userAgent = new SIP.UserAgent({
|
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uri: SIP.UserAgent.makeURI(`sip:${encodeURIComponent(callerIdName)}@${hostname}`),
|
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transportOptions: {
|
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server: `${(protocol === 'https:' ? 'wss://' : 'ws://')}${hostname}/ws?${token}`,
|
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connectionTimeout: USER_AGENT_CONNECTION_TIMEOUT_MS,
|
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keepAliveInterval: WEBSOCKET_KEEP_ALIVE_INTERVAL,
|
|
keepAliveDebounce: WEBSOCKET_KEEP_ALIVE_DEBOUNCE,
|
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traceSip: TRACE_SIP,
|
|
},
|
|
sessionDescriptionHandlerFactoryOptions: {
|
|
peerConnectionConfiguration: {
|
|
iceServers,
|
|
sdpSemantics: SDP_SEMANTICS,
|
|
iceTransportPolicy: FORCE_RELAY ? 'relay' : undefined,
|
|
},
|
|
},
|
|
displayName: callerIdName,
|
|
register: false,
|
|
userAgentString: `BigBlueButton/${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 inviterOptions = {
|
|
sessionDescriptionHandlerOptions: {
|
|
constraints: {
|
|
audio: isListenOnly
|
|
? false
|
|
: matchConstraints,
|
|
video: false,
|
|
},
|
|
iceGatheringTimeout: ICE_GATHERING_TIMEOUT,
|
|
},
|
|
sessionDescriptionHandlerModifiersPostICEGathering: [
|
|
stripMDnsCandidates,
|
|
filterValidIceCandidates.bind(this, this.validIceCandidates),
|
|
],
|
|
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);
|
|
|
|
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, 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);
|
|
|
|
query.observeChanges({
|
|
changed: (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();
|
|
}
|
|
},
|
|
});
|
|
});
|
|
|
|
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 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');
|
|
}
|
|
}
|
|
}
|
|
|
|
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;
|
|
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 inputDeviceId() {
|
|
const sessionInputDeviceId = Storage.getItem(INPUT_DEVICE_ID_KEY);
|
|
|
|
if (sessionInputDeviceId) {
|
|
return sessionInputDeviceId;
|
|
}
|
|
|
|
if (this.media.inputDeviceId) {
|
|
return this.media.inputDeviceId;
|
|
}
|
|
|
|
if (this.activeSession) {
|
|
return this.activeSession.inputDeviceId;
|
|
}
|
|
|
|
return DEFAULT_INPUT_DEVICE_ID;
|
|
}
|
|
|
|
set inputDeviceId(deviceId) {
|
|
Storage.setItem(INPUT_DEVICE_ID_KEY, deviceId);
|
|
this.media.inputDeviceId = deviceId;
|
|
|
|
if (this.activeSession) {
|
|
this.activeSession.inputDeviceId = deviceId;
|
|
}
|
|
}
|
|
|
|
get outputDeviceId() {
|
|
const sessionOutputDeviceId = Storage.getItem(OUTPUT_DEVICE_ID_KEY);
|
|
if (sessionOutputDeviceId) {
|
|
return sessionOutputDeviceId;
|
|
}
|
|
|
|
if (this.media.outputDeviceId) {
|
|
return this.media.outputDeviceId;
|
|
}
|
|
|
|
if (this.activeSession) {
|
|
return this.activeSession.outputDeviceId;
|
|
}
|
|
|
|
return DEFAULT_OUTPUT_DEVICE_ID;
|
|
}
|
|
|
|
set outputDeviceId(deviceId) {
|
|
Storage.setItem(OUTPUT_DEVICE_ID_KEY, deviceId);
|
|
this.media.outputDeviceId = deviceId;
|
|
|
|
if (this.activeSession) {
|
|
this.activeSession.outputDeviceId = deviceId;
|
|
}
|
|
}
|
|
|
|
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 }, 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,
|
|
}, callback)
|
|
.then((value) => {
|
|
this.changeOutputDevice(outputDeviceId, true);
|
|
resolve(value);
|
|
}).catch((reason) => {
|
|
reject(reason);
|
|
});
|
|
}
|
|
}
|
|
|
|
return managerCallback(message);
|
|
};
|
|
|
|
const { inputDeviceId, outputDeviceId } = this;
|
|
this.activeSession.joinAudio({
|
|
isListenOnly,
|
|
extension,
|
|
inputDeviceId,
|
|
outputDeviceId,
|
|
validIceCandidates,
|
|
}, callback)
|
|
.then((value) => {
|
|
this.changeOutputDevice(outputDeviceId, true);
|
|
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() {
|
|
return this.activeSession.exitAudio();
|
|
}
|
|
|
|
liveChangeInputDevice(deviceId) {
|
|
this.inputDeviceId = deviceId;
|
|
return this.activeSession.liveChangeInputDevice(deviceId);
|
|
}
|
|
|
|
async updateAudioConstraints(constraints) {
|
|
return this.activeSession.updateAudioConstraints(constraints);
|
|
}
|
|
}
|
|
|
|
module.exports = SIPBridge;
|