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webrtc.js
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webrtc.js
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/**
* @copyright Copyright (c) 2019 Daniel Calviño Sánchez <danxuliu@gmail.com>
* @copyright Copyright (c) 2019 Ivan Sein <ivan@nextcloud.com>
* @copyright Copyright (c) 2019 Joachim Bauch <bauch@struktur.de>
* @copyright Copyright (c) 2019 Joas Schilling <coding@schilljs.com>
*
* @author Daniel Calviño Sánchez <danxuliu@gmail.com>
* @author Ivan Sein <ivan@nextcloud.com>
* @author Joachim Bauch <bauch@struktur.de>
* @author Joas Schilling <coding@schilljs.com>
*
* @license GNU AGPL version 3 or any later version
*
* This program is free software: you can redistribute it and/or modify
* it under the terms of the GNU Affero General Public License as
* published by the Free Software Foundation, either version 3 of the
* License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Affero General Public License for more details.
*
* You should have received a copy of the GNU Affero General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*
*/
import SimpleWebRTC from './simplewebrtc/simplewebrtc'
import { PARTICIPANT } from '../../constants.js'
import store from '../../store/index.js'
import {
showError,
TOAST_PERMANENT_TIMEOUT,
TOAST_DEFAULT_TIMEOUT,
} from '@nextcloud/dialogs'
let webrtc
const spreedPeerConnectionTable = []
let previousUsersInRoom = []
let usersInCallMapping = {}
let ownPeer = null
let ownScreenPeer = null
let selfInCall = PARTICIPANT.CALL_FLAG.DISCONNECTED
const delayedConnectionToPeer = []
let callParticipantCollection = null
let localCallParticipantModel = null
let showedTURNWarning = false
let sendCurrentStateWithRepetitionTimeout = null
function arrayDiff(a, b) {
return a.filter(function(i) {
return b.indexOf(i) < 0
})
}
function createScreensharingPeer(signaling, sessionId) {
const currentSessionId = signaling.getSessionId()
const useMcu = signaling.hasFeature('mcu')
if (useMcu && !webrtc.webrtc.getPeers(currentSessionId, 'screen').length) {
if (ownScreenPeer) {
ownScreenPeer.end()
}
// Create own publishing stream.
ownScreenPeer = webrtc.webrtc.createPeer({
id: currentSessionId,
type: 'screen',
sharemyscreen: true,
enableDataChannels: false,
receiveMedia: {
offerToReceiveAudio: 0,
offerToReceiveVideo: 0,
},
broadcaster: currentSessionId,
})
webrtc.emit('createdPeer', ownScreenPeer)
ownScreenPeer.start()
localCallParticipantModel.setScreenPeer(ownScreenPeer)
}
if (sessionId === currentSessionId) {
return
}
if (useMcu) {
// TODO(jojo): Already create peer object to avoid duplicate offers.
// TODO(jojo): We should use "requestOffer" as with regular
// audio/video peers. Not possible right now as there is no way
// for clients to know that screensharing is active and an offer
// from the MCU should be requested.
signaling.sendOffer(sessionId, 'screen')
} else if (!useMcu) {
const screenPeers = webrtc.webrtc.getPeers(sessionId, 'screen')
const screenPeerSharedTo = screenPeers.find(function(screenPeer) {
return screenPeer.sharemyscreen === true
})
if (!screenPeerSharedTo) {
const peer = webrtc.webrtc.createPeer({
id: sessionId,
type: 'screen',
sharemyscreen: true,
enableDataChannels: false,
receiveMedia: {
offerToReceiveAudio: 0,
offerToReceiveVideo: 0,
},
broadcaster: currentSessionId,
})
webrtc.emit('createdPeer', peer)
peer.start()
}
}
}
function checkStartPublishOwnPeer(signaling) {
'use strict'
const currentSessionId = signaling.getSessionId()
if (!webrtc.webrtc.localStreams.length || webrtc.webrtc.getPeers(currentSessionId, 'video').length) {
// No media yet or already publishing.
return
}
if (ownPeer) {
webrtc.removePeers(ownPeer.id)
ownPeer.end()
}
// Create own publishing stream.
ownPeer = webrtc.webrtc.createPeer({
id: currentSessionId,
type: 'video',
enableDataChannels: true,
receiveMedia: {
offerToReceiveAudio: 0,
offerToReceiveVideo: 0,
},
sendVideoIfAvailable: signaling.getSendVideoIfAvailable(),
})
webrtc.emit('createdPeer', ownPeer)
ownPeer.start()
localCallParticipantModel.setPeer(ownPeer)
}
function sendCurrentMediaState() {
if (!webrtc.webrtc.isVideoEnabled()) {
webrtc.webrtc.emit('videoOff')
} else {
webrtc.webrtc.emit('videoOn')
}
if (!webrtc.webrtc.isAudioEnabled()) {
webrtc.webrtc.emit('audioOff')
} else {
webrtc.webrtc.emit('audioOn')
}
}
// TODO The participant name should be got from the participant list, but it is
// not currently possible to associate a Nextcloud ID with a standalone
// signaling ID for guests.
function sendCurrentNick() {
webrtc.webrtc.emit('nickChanged', store.getters.getDisplayName())
}
function sendCurrentStateWithRepetition(timeout) {
if (!timeout) {
timeout = 0
clearTimeout(sendCurrentStateWithRepetitionTimeout)
}
sendCurrentStateWithRepetitionTimeout = setTimeout(function() {
sendCurrentMediaState()
sendCurrentNick()
if (!timeout) {
timeout = 1000
} else {
timeout *= 2
}
if (timeout > 16000) {
sendCurrentStateWithRepetitionTimeout = null
return
}
sendCurrentStateWithRepetition(timeout)
}, timeout)
}
function userHasStreams(user) {
let flags = user
if (flags.hasOwnProperty('inCall')) {
flags = flags.inCall
}
flags = flags || PARTICIPANT.CALL_FLAG.DISCONNECTED
const REQUIRED_FLAGS = PARTICIPANT.CALL_FLAG.WITH_AUDIO | PARTICIPANT.CALL_FLAG.WITH_VIDEO
return (flags & REQUIRED_FLAGS) !== 0
}
function usersChanged(signaling, newUsers, disconnectedSessionIds) {
'use strict'
const currentSessionId = signaling.getSessionId()
const useMcu = signaling.hasFeature('mcu')
if (useMcu && newUsers.length) {
checkStartPublishOwnPeer(signaling)
}
newUsers.forEach(function(user) {
if (!user.inCall) {
return
}
// TODO(fancycode): Adjust property name of internal PHP backend to be all lowercase.
const sessionId = user.sessionId || user.sessionid
if (!sessionId || sessionId === currentSessionId || previousUsersInRoom.indexOf(sessionId) !== -1) {
return
}
previousUsersInRoom.push(sessionId)
// Use null to differentiate between guest (null) and not known yet
// (undefined).
// TODO(fancycode): Adjust property name of internal PHP backend to be all lowercase.
const userId = user.userId || user.userid || null
// When the external signaling server is used the Nextcloud session id
// will be provided in its own property. When the internal signaling
// server is used the Nextcloud session id and the signaling session id
// are the same and thus set from the signaling session id.
const nextcloudSessionId = user.nextcloudSessionId || user.nextcloudsessionid || sessionId
let callParticipantModel = callParticipantCollection.get(sessionId)
if (!callParticipantModel) {
callParticipantModel = callParticipantCollection.add({
peerId: sessionId,
webRtc: webrtc,
})
}
callParticipantModel.setUserId(userId)
callParticipantModel.setNextcloudSessionId(nextcloudSessionId)
if (user.internal) {
callParticipantModel.set('internal', true)
}
// When the MCU is used and the other participant has no streams or
// when no MCU is used and neither the local participant nor the
// other one has no streams there will be no Peer for that other
// participant, so a null Peer needs to be explicitly set now.
if ((signaling.hasFeature('mcu') && user && !userHasStreams(user))
|| (!signaling.hasFeature('mcu') && user && !userHasStreams(user) && !webrtc.webrtc.localStreams.length)) {
callParticipantModel.setPeer(null)
// As there is no Peer for the other participant the current state
// will not be sent once it is connected, so it needs to be sent
// now.
// When there is no MCU this is only needed for the nick; as the
// local participant has no streams it will be automatically marked
// with audio and video not available on the other end, so there is
// no need to send the media state.
if (signaling.hasFeature('mcu')) {
sendCurrentStateWithRepetition()
} else {
sendCurrentNick()
}
}
const createPeer = function() {
const peer = webrtc.webrtc.createPeer({
id: sessionId,
type: 'video',
enableDataChannels: true,
receiveMedia: {
offerToReceiveAudio: 1,
offerToReceiveVideo: 1,
},
sendVideoIfAvailable: signaling.getSendVideoIfAvailable(),
})
webrtc.emit('createdPeer', peer)
peer.start()
}
if (!webrtc.webrtc.getPeers(sessionId, 'video').length) {
if (useMcu && userHasStreams(user)) {
// TODO(jojo): Already create peer object to avoid duplicate offers.
signaling.requestOffer(user, 'video')
delayedConnectionToPeer[user.sessionId] = setInterval(function() {
console.debug('No offer received for new peer, request offer again')
signaling.requestOffer(user, 'video')
}, 10000)
} else if (!useMcu && userHasStreams(selfInCall) && (!userHasStreams(user) || sessionId < currentSessionId)) {
// To avoid overloading the user joining a room (who previously called
// all the other participants), we decide who calls who by comparing
// the session ids of the users: "larger" ids call "smaller" ones.
console.debug('Starting call with', user)
createPeer()
} else if (!useMcu && userHasStreams(selfInCall) && userHasStreams(user) && sessionId > currentSessionId) {
// If the remote peer is not aware that it was disconnected
// from the current peer the remote peer will not send a new
// offer; thus, if the current peer does not receive a new
// offer in a reasonable time, the current peer calls the
// remote peer instead of waiting to be called to
// reestablish the connection.
delayedConnectionToPeer[sessionId] = setInterval(function() {
// New offers are periodically sent until a connection
// is established. As an offer can not be sent again
// from an existing peer it must be removed and a new
// one must be created from scratch.
webrtc.webrtc.getPeers(sessionId, 'video').forEach(function(peer) {
peer.end()
})
console.debug('No offer nor answer received, sending offer again')
createPeer()
}, 10000)
} else {
console.debug('User has no streams, not sending another offer')
}
}
// Send shared screen to new participants
if (webrtc.getLocalScreen()) {
createScreensharingPeer(signaling, sessionId)
}
})
disconnectedSessionIds.forEach(function(sessionId) {
console.debug('Remove disconnected peer', sessionId)
webrtc.removePeers(sessionId)
callParticipantCollection.remove(sessionId)
if (delayedConnectionToPeer[sessionId]) {
clearInterval(delayedConnectionToPeer[sessionId])
delete delayedConnectionToPeer[sessionId]
}
})
previousUsersInRoom = arrayDiff(previousUsersInRoom, disconnectedSessionIds)
}
function usersInCallChanged(signaling, users) {
// The passed list are the users that are currently in the room,
// i.e. that are in the call and should call each other.
const currentSessionId = signaling.getSessionId()
const currentUsersInRoom = []
const userMapping = {}
selfInCall = PARTICIPANT.CALL_FLAG.DISCONNECTED
let sessionId
for (sessionId in users) {
if (!users.hasOwnProperty(sessionId)) {
continue
}
const user = users[sessionId]
if (!user.inCall) {
continue
}
if (sessionId === currentSessionId) {
selfInCall = user.inCall
continue
}
currentUsersInRoom.push(sessionId)
userMapping[sessionId] = user
}
if (!selfInCall) {
// Own session is no longer in the call, disconnect from all others.
usersChanged(signaling, [], previousUsersInRoom)
return
}
const newSessionIds = arrayDiff(currentUsersInRoom, previousUsersInRoom)
const disconnectedSessionIds = arrayDiff(previousUsersInRoom, currentUsersInRoom)
const newUsers = []
newSessionIds.forEach(function(sessionId) {
newUsers.push(userMapping[sessionId])
})
if (newUsers.length || disconnectedSessionIds.length) {
usersChanged(signaling, newUsers, disconnectedSessionIds)
}
}
export default function initWebRTC(signaling, _callParticipantCollection, _localCallParticipantModel) {
callParticipantCollection = _callParticipantCollection
localCallParticipantModel = _localCallParticipantModel
signaling.on('usersLeft', function(users) {
users.forEach(function(user) {
delete usersInCallMapping[user]
})
usersChanged(signaling, [], users)
})
signaling.on('usersChanged', function(users) {
users.forEach(function(user) {
const sessionId = user.sessionId || user.sessionid
usersInCallMapping[sessionId] = user
})
usersInCallChanged(signaling, usersInCallMapping)
})
signaling.on('participantFlagsChanged', function(event) {
/**
* event {
* roomid: "1609407087",
* sessionid: "…",
* flags: 1
* }
*/
const callParticipantModel = callParticipantCollection.get(event.sessionid)
if (callParticipantModel) {
callParticipantModel.set('speaking', (event.flags & PARTICIPANT.SIP_FLAG.SPEAKING) > 0)
callParticipantModel.set('audioAvailable', (event.flags & PARTICIPANT.SIP_FLAG.MUTE_MICROPHONE) === 0)
callParticipantModel.set('raisedHand', {
state: (event.flags & PARTICIPANT.SIP_FLAG.RAISE_HAND) !== 0,
timestamp: Date.now(),
})
}
})
signaling.on('usersInRoom', function(users) {
usersInCallMapping = {}
users.forEach(function(user) {
const sessionId = user.sessionId || user.sessionid
usersInCallMapping[sessionId] = user
})
usersInCallChanged(signaling, usersInCallMapping)
})
signaling.on('leaveCall', function(token, reconnect) {
// When the MCU is used and there is a connection error the call is
// left and then joined again to perform the reconnection. In those
// cases the call should be kept active from the point of view of
// WebRTC.
if (reconnect) {
return
}
clearErrorNotification()
webrtc.leaveCall()
})
signaling.on('message', function(message) {
if (message.type === 'answer' && message.roomType === 'video' && delayedConnectionToPeer[message.from]) {
clearInterval(delayedConnectionToPeer[message.from])
delete delayedConnectionToPeer[message.from]
return
}
if (message.type !== 'offer') {
return
}
const peers = webrtc.webrtc.peers
const stalePeer = peers.find(function(peer) {
if (peer.sharemyscreen) {
return false
}
return peer.id === message.from && peer.type === message.roomType && peer.sid !== message.sid
})
if (stalePeer) {
stalePeer.end()
}
if (message.roomType === 'video' && delayedConnectionToPeer[message.from]) {
clearInterval(delayedConnectionToPeer[message.from])
delete delayedConnectionToPeer[message.from]
}
if (!selfInCall) {
console.debug('Offer received when not in the call, ignore')
message.type = 'offer-to-ignore'
}
// MCU screen offers do not include the "broadcaster" property,
// which is expected by SimpleWebRTC in screen offers from a remote
// peer, so it needs to be explicitly added.
if (signaling.hasFeature('mcu') && message.roomType === 'screen') {
message.broadcaster = message.from
}
})
webrtc = new SimpleWebRTC({
remoteVideosEl: '',
autoRequestMedia: true,
debug: false,
media: {
audio: true,
video: true,
},
autoAdjustMic: false,
audioFallback: true,
detectSpeakingEvents: true,
connection: signaling,
enableDataChannels: true,
nick: store.getters.getDisplayName(),
})
if (signaling.hasFeature('mcu')) {
// Force "Plan-B" semantics if the MCU is used, which doesn't support
// "Unified Plan" with SimpleWebRTC yet.
webrtc.webrtc.config.peerConnectionConfig.sdpSemantics = 'plan-b'
}
if (!window.OCA.Talk) {
window.OCA.Talk = {}
}
window.OCA.Talk.SimpleWebRTC = webrtc
signaling.on('pullMessagesStoppedOnFail', function() {
// Force leaving the call in WebRTC; when pulling messages stops due
// to failures the room is left, and leaving the room indirectly
// runs signaling.leaveCurrentCall(), but if the signaling fails to
// leave the call (which is likely due to the messages failing to be
// received) no event will be triggered and the call will not be
// left from WebRTC point of view.
webrtc.leaveCall()
})
webrtc.startMedia = function(token) {
webrtc.joinCall(token)
}
const sendDataChannelToAll = function(channel, message, payload) {
// If running with MCU, the message must be sent through the
// publishing peer and will be distributed by the MCU to subscribers.
if (ownPeer && signaling.hasFeature && signaling.hasFeature('mcu')) {
ownPeer.sendDirectly(channel, message, payload)
return
}
webrtc.sendDirectlyToAll(channel, message, payload)
}
function handleIceConnectionStateConnected(peer) {
// Send the current information about the state.
if (!signaling.hasFeature('mcu')) {
// Only the media state needs to be sent, the nick was already sent
// in the offer/answer.
sendCurrentMediaState()
} else {
sendCurrentStateWithRepetition()
}
// Reset ice restart counter for peer
if (spreedPeerConnectionTable[peer.id] > 0) {
spreedPeerConnectionTable[peer.id] = 0
}
}
function handleIceConnectionStateDisconnected(peer) {
setTimeout(function() {
if (peer.pc.iceConnectionState !== 'disconnected') {
return
}
peer.emit('extendedIceConnectionStateChange', 'disconnected-long')
if (!signaling.hasFeature('mcu')) {
// Disconnections are not handled with the MCU, only
// failures.
// If the peer is still disconnected after 5 seconds we try
// ICE restart.
if (spreedPeerConnectionTable[peer.id] < 5) {
if (peer.pc.localDescription.type === 'offer'
&& peer.pc.signalingState === 'stable') {
spreedPeerConnectionTable[peer.id]++
console.debug('ICE restart after disconnect.', peer)
peer.icerestart()
}
}
}
}, 5000)
}
function handleIceConnectionStateFailed(peer) {
if (!showedTURNWarning && !signaling.settings.turnservers.length) {
showError(
t('spreed', 'Could not establish a connection with at least one participant. A TURN server might be needed for your scenario. Please ask your administrator to set one up following {linkstart}this documentation{linkend}.')
.replace('{linkstart}', '<a target="_blank" rel="noreferrer nofollow" class="external" href="https://nextcloud-talk.readthedocs.io/en/latest/TURN/">')
.replace('{linkend}', ' ↗</a>'),
{
timeout: TOAST_PERMANENT_TIMEOUT,
isHTML: true,
}
)
showedTURNWarning = true
}
if (!signaling.hasFeature('mcu')) {
if (spreedPeerConnectionTable[peer.id] < 5) {
if (peer.pc.localDescription.type === 'offer'
&& peer.pc.signalingState === 'stable') {
spreedPeerConnectionTable[peer.id]++
console.debug('ICE restart after failure.', peer)
peer.icerestart()
}
} else {
console.error('ICE failed after 5 tries.', peer)
peer.emit('extendedIceConnectionStateChange', 'failed-no-restart')
}
} else {
// This handles ICE failures of a receiver peer; ICE failures of
// the sender peer are handled in the "iceFailed" event.
console.debug('Request offer again', peer)
signaling.requestOffer(peer.id, 'video')
delayedConnectionToPeer[peer.id] = setInterval(function() {
console.debug('No offer received, request offer again', peer)
signaling.requestOffer(peer.id, 'video')
}, 10000)
}
}
function setHandlerForIceConnectionStateChange(peer) {
// Initialize ice restart counter for peer
spreedPeerConnectionTable[peer.id] = 0
peer.pc.addEventListener('iceconnectionstatechange', function() {
peer.emit('extendedIceConnectionStateChange', peer.pc.iceConnectionState)
switch (peer.pc.iceConnectionState) {
case 'checking':
console.debug('Connecting to peer...', peer)
break
case 'connected':
case 'completed': // on caller side
console.debug('Connection established.', peer)
handleIceConnectionStateConnected(peer)
break
case 'disconnected':
console.debug('Disconnected.', peer)
handleIceConnectionStateDisconnected(peer)
break
case 'failed':
console.debug('Connection failed.', peer)
handleIceConnectionStateFailed(peer)
break
case 'closed':
console.debug('Connection closed.', peer)
break
}
})
}
const forceReconnect = function(signaling, flags) {
if (ownPeer) {
webrtc.removePeers(ownPeer.id)
ownPeer.end()
ownPeer = null
localCallParticipantModel.setPeer(ownPeer)
}
usersChanged(signaling, [], previousUsersInRoom)
usersInCallMapping = {}
previousUsersInRoom = []
// Reconnects with a new session id will trigger "usersChanged"
// with the users in the room and that will re-establish the
// peerconnection streams.
// If flags are undefined the current call flags are used.
signaling.forceReconnect(true, flags)
}
function setHandlerForNegotiationNeeded(peer) {
peer.pc.addEventListener('negotiationneeded', function() {
// Negotiation needed will be first triggered before the connection
// is established, but forcing a reconnection should be done only
// once the connection was established.
if (peer.pc.iceConnectionState !== 'new' && peer.pc.iceConnectionState !== 'checking') {
forceReconnect(signaling)
}
})
}
webrtc.on('createdPeer', function(peer) {
console.debug('Peer created', peer)
if (peer.id !== signaling.getSessionId() && !peer.sharemyscreen) {
// In some strange cases a Peer can be added before its
// participant is found in the list of participants.
let callParticipantModel = callParticipantCollection.get(peer.id)
if (!callParticipantModel) {
callParticipantModel = callParticipantCollection.add({
peerId: peer.id,
webRtc: webrtc,
})
}
if (peer.type === 'video') {
callParticipantModel.setPeer(peer)
} else {
callParticipantModel.setScreenPeer(peer)
}
}
if (peer.type === 'video') {
if (peer.id === signaling.getSessionId()) {
console.debug('Not adding ICE connection state handler for own peer', peer)
} else {
setHandlerForIceConnectionStateChange(peer)
}
setHandlerForNegotiationNeeded(peer)
// Make sure required data channels exist for all peers. This
// is required for peers that get created by SimpleWebRTC from
// received "Offer" messages. Otherwise the "channelMessage"
// will not be called.
peer.getDataChannel('status')
}
})
function checkPeerMedia(peer, track, mediaType) {
return new Promise((resolve, reject) => {
peer.pc.getStats(track).then(function(stats) {
let result = false
stats.forEach(function(statsReport) {
if (result || statsReport.mediaType !== mediaType || !statsReport.hasOwnProperty('bytesReceived')) {
return
}
if (statsReport.bytesReceived > 0) {
if (mediaType === 'video' && statsReport.bytesReceived < 2000) {
// A video with less than 2000 bytes is an empty single frame of the MCU
// console.debug('Participant is registered with with video but didn\'t send a lot of data, so we assume the video is disabled for now.')
result = true
return
}
webrtc.emit('unmute', {
id: peer.id,
name: mediaType,
})
result = true
}
})
if (result) {
resolve()
} else {
reject(new Error('No bytes received'))
}
})
})
}
function stopPeerCheckAudioMedia(peer) {
clearInterval(peer.check_audio_interval)
peer.check_audio_interval = null
}
function stopPeerCheckVideoMedia(peer) {
clearInterval(peer.check_video_interval)
peer.check_video_interval = null
}
function stopPeerIdCheckMediaType(peerId, mediaType) {
// There should be just one video peer with that id, but iterating is
// safer.
const peers = webrtc.getPeers(peerId, 'video')
peers.forEach(function(peer) {
if (mediaType === 'audio') {
stopPeerCheckAudioMedia(peer)
} else if (mediaType === 'video') {
stopPeerCheckVideoMedia(peer)
}
})
}
if (signaling.hasFeature('mcu')) {
webrtc.on('mute', function(data) {
stopPeerIdCheckMediaType(data.id, data.name)
})
webrtc.on('unmute', function(data) {
stopPeerIdCheckMediaType(data.id, data.name)
})
}
function stopPeerCheckMedia(peer) {
stopPeerCheckAudioMedia(peer)
stopPeerCheckVideoMedia(peer)
}
function startPeerCheckMedia(peer, stream) {
stopPeerCheckMedia(peer)
peer.check_video_interval = setInterval(function() {
stream.getVideoTracks().forEach(function(video) {
checkPeerMedia(peer, video, 'video').then(function() {
stopPeerCheckVideoMedia(peer)
}).catch(() => {
})
})
}, 1000)
peer.check_audio_interval = setInterval(function() {
stream.getAudioTracks().forEach(function(audio) {
checkPeerMedia(peer, audio, 'audio').then(function() {
stopPeerCheckAudioMedia(peer)
}).catch(() => {
})
})
}, 1000)
}
webrtc.on('peerStreamAdded', function(peer) {
// With the MCU, a newly subscribed stream might not get the
// "audioOn"/"videoOn" messages as they are only sent when
// a user starts publishing. Instead wait for initial data
// and trigger events locally.
if (!signaling.hasFeature('mcu')) {
return
}
if (peer.type === 'screen') {
return
}
startPeerCheckMedia(peer, peer.stream)
})
webrtc.on('peerStreamRemoved', function(peer) {
stopPeerCheckMedia(peer)
})
webrtc.webrtc.on('videoOn', function() {
if (signaling.getSendVideoIfAvailable()) {
return
}
// When enabling the local video if the video is not being sent a
// reconnection is forced to start sending it.
signaling.setSendVideoIfAvailable(true)
let flags = signaling.getCurrentCallFlags()
flags |= PARTICIPANT.CALL_FLAG.WITH_VIDEO
forceReconnect(signaling, flags)
})
webrtc.webrtc.on('iceFailed', function(/* peer */) {
if (!signaling.hasFeature('mcu')) {
// ICE restarts will be handled by "iceConnectionStateChange"
// above.
return
}
// For now assume the connection to the MCU is interrupted on ICE
// failures and force a reconnection of all streams.
forceReconnect(signaling)
})
let localStreamRequestedTimeout = null
let localStreamRequestedTimeoutNotification = null
let errorNotificationHandle = null
const clearLocalStreamRequestedTimeoutAndHideNotification = function() {
clearTimeout(localStreamRequestedTimeout)
localStreamRequestedTimeout = null
if (localStreamRequestedTimeoutNotification) {
localStreamRequestedTimeoutNotification.hideToast()
localStreamRequestedTimeoutNotification = null
}
}
const clearErrorNotification = function() {
if (errorNotificationHandle) {
errorNotificationHandle.hideToast()
errorNotificationHandle = null
}
}
// In some cases the browser may enter in a faulty state in which
// "getUserMedia" does not return neither successfully nor with an
// error. It is not possible to detect this except by guessing when some
// time passes and the user has not granted nor rejected the media
// permissions.
webrtc.on('localStreamRequested', function() {
clearLocalStreamRequestedTimeoutAndHideNotification()
localStreamRequestedTimeout = setTimeout(function() {
// FIXME emit an event and handle it as needed instead of
// calling UI code from here.
localStreamRequestedTimeoutNotification = showError(t('spreed', 'This is taking longer than expected. Are the media permissions already granted (or rejected)? If yes please restart your browser, as audio and video are failing'), {
timeout: TOAST_PERMANENT_TIMEOUT,
})
}, 10000)
})
signaling.on('leaveRoom', function(token) {
if (signaling.currentRoomToken === token) {
clearLocalStreamRequestedTimeoutAndHideNotification()
clearErrorNotification()
}
})
webrtc.on('localMediaStarted', function(/* configuration */) {
console.info('localMediaStarted')
clearLocalStreamRequestedTimeoutAndHideNotification()
if (signaling.hasFeature('mcu')) {
checkStartPublishOwnPeer(signaling)
}
})
webrtc.on('localMediaError', function(error) {
console.warn('Access to microphone & camera failed', error)
clearLocalStreamRequestedTimeoutAndHideNotification()
let message
let timeout = TOAST_PERMANENT_TIMEOUT
if ((error.name === 'NotSupportedError'
&& webrtc.capabilities.supportRTCPeerConnection)
|| (error.name === 'NotAllowedError'
&& error.message && error.message.indexOf('Only secure origins') !== -1)) {
message = t('spreed', 'Access to microphone & camera is only possible with HTTPS')
message += ': ' + t('spreed', 'Please move your setup to HTTPS')
} else if (error.name === 'NotAllowedError') {
message = t('spreed', 'Access to microphone & camera was denied')
timeout = TOAST_DEFAULT_TIMEOUT
} else if (!webrtc.capabilities.support) {
console.error('WebRTC not supported')
message = t('spreed', 'WebRTC is not supported in your browser')
message += ': ' + t('spreed', 'Please use a different browser like Firefox or Chrome')
} else {
// Mostly happens in Chrome (NotFoundError): when no audio device available
// not sure what else can cause this
message = t('spreed', 'Error while accessing microphone & camera')
console.error('Error while accessing microphone & camera: ', error.message, error.name)
}
errorNotificationHandle = showError(message, {
timeout: timeout,
})
})
webrtc.on('channelOpen', function(channel) {
console.debug('%s datachannel is open', channel.label)
})
webrtc.on('channelMessage', function(peer, label, data) {
if (label === 'status' || label === 'JanusDataChannel') {
if (data.type === 'audioOn') {
webrtc.emit('unmute', { id: peer.id, name: 'audio' })
} else if (data.type === 'audioOff') {
webrtc.emit('mute', { id: peer.id, name: 'audio' })
} else if (data.type === 'videoOn') {
webrtc.emit('unmute', { id: peer.id, name: 'video' })
} else if (data.type === 'videoOff') {
webrtc.emit('mute', { id: peer.id, name: 'video' })
} else if (data.type === 'nickChanged') {
const name = typeof (data.payload) === 'string' ? data.payload : data.payload.name
webrtc.emit('nick', { id: peer.id, name: name })
} else if (data.type === 'speaking' || data.type === 'stoppedSpeaking') {
// Valid known messages, but handled elsewhere
} else {
console.debug('Unknown message type %s from %s datachannel', data.type, label, data)
}
} else {
console.debug('Unknown message from %s datachannel', label, data)
}
})
webrtc.on('speaking', function() {
sendDataChannelToAll('status', 'speaking')
})
webrtc.on('stoppedSpeaking', function() {
sendDataChannelToAll('status', 'stoppedSpeaking')
})
// Send the audio on and off events via data channel
webrtc.on('audioOn', function() {
sendDataChannelToAll('status', 'audioOn')
})
webrtc.on('audioOff', function() {
sendDataChannelToAll('status', 'audioOff')
})
webrtc.on('videoOn', function() {
sendDataChannelToAll('status', 'videoOn')
})
webrtc.on('videoOff', function() {