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Files
matrix-js-sdk/lib/webrtc/call.js

874 lines
25 KiB
JavaScript

"use strict";
/**
* This is an internal module. See {@link createNewMatrixCall} for the public API.
* @module webrtc/call
*/
var utils = require("../utils");
var EventEmitter = require("events").EventEmitter;
var DEBUG = true; // set true to enable console logging.
// events: onHangup, callPlaced, error
/**
* Construct a new Matrix Call.
* @constructor
* @param {Object} opts Config options.
* @param {string} opts.roomId The room ID for this call.
* @param {Object} opts.webRtc The WebRTC globals from the browser.
* @param {Object} opts.URL The URL global.
* @param {Array<Object>} opts.turnServers Optional. A list of TURN servers.
* @param {MatrixClient} opts.client The Matrix Client instance to send events to.
*/
function MatrixCall(opts) {
this.roomId = opts.roomId;
this.client = opts.client;
this.webRtc = opts.webRtc;
this.URL = opts.URL;
// Array of Objects with urls, username, credential keys
this.turnServers = opts.turnServers || [{
urls: [MatrixCall.FALLBACK_STUN_SERVER]
}];
utils.forEach(this.turnServers, function(server) {
utils.checkObjectHasKeys(server, ["urls"]);
});
this.callId = "c" + new Date().getTime();
this.state = 'fledgling';
this.didConnect = false;
// A queue for candidates waiting to go out.
// We try to amalgamate candidates into a single candidate message where
// possible
this.candidateSendQueue = [];
this.candidateSendTries = 0;
}
/** The length of time a call can be ringing for. */
MatrixCall.CALL_TIMEOUT_MS = 60000;
/** The fallback server to use for STUN. */
MatrixCall.FALLBACK_STUN_SERVER = 'stun:stun.l.google.com:19302';
/** An error code when the local client failed to create an offer. */
MatrixCall.ERR_LOCAL_OFFER_FAILED = "local_offer_failed";
/**
* An error code when there is no local mic/camera to use. This may be because
* the hardware isn't plugged in, or the user has explicitly denied access.
*/
MatrixCall.ERR_NO_USER_MEDIA = "no_user_media";
utils.inherits(MatrixCall, EventEmitter);
/**
* Place a voice call to this room.
* @throws If you have not specified a listener for 'error' events.
*/
MatrixCall.prototype.placeVoiceCall = function() {
checkForErrorListener(this);
_placeCallWithConstraints(this, _getUserMediaVideoContraints('voice'));
this.type = 'voice';
};
/**
* Place a video call to this room.
* @param {Element} localVideoElement a DOM element with the local camera preview.
* @param {Element} remoteVideoElement a DOM element to render video to.
* @throws If you have not specified a listener for 'error' events.
*/
MatrixCall.prototype.placeVideoCall = function(localVideoElement, remoteVideoElement) {
checkForErrorListener(this);
this.localVideoElement = localVideoElement;
this.remoteVideoElement = remoteVideoElement;
_placeCallWithConstraints(this, _getUserMediaVideoContraints('video'));
this.type = 'video';
this.tryPlayRemoteStream();
};
/**
* Retrieve the local video DOM element.
* @return {Element} The dom element
*/
MatrixCall.prototype.getLocalVideoElement = function() {
return this.localVideoElement;
};
/**
* Retrieve the remote video DOM element.
* @return {Element} The dom element
*/
MatrixCall.prototype.getRemoteVideoElement = function() {
return this.remoteVideoElement;
};
/**
* Set the remote video DOM element. If this call is active, video will be
* rendered to it.
* @param {Element} element The DOM element.
*/
MatrixCall.prototype.setRemoteVideoElement = function(element) {
this.remoteVideoElement = element;
this.tryPlayRemoteStream();
};
/**
* Configure this call from an invite event.
* @param {MatrixEvent} event The m.call.invite event
*/
MatrixCall.prototype.initWithInvite = function(event) {
this.msg = event.getContent();
this.peerConn = _createPeerConnection(this);
var self = this;
if (this.peerConn) {
this.peerConn.setRemoteDescription(
new this.webRtc.RtcSessionDescription(this.msg.offer),
hookCallback(self, self.onSetRemoteDescriptionSuccess),
hookCallback(self, self.onSetRemoteDescriptionError)
);
}
this.state = 'ringing';
this.direction = 'inbound';
// firefox and Safari's RTCPeerConnection doesn't add streams until it
// starts getting media on them so we need to figure out whether a video
// channel has been offered by ourselves.
if (this.msg.offer.sdp.indexOf('m=video') > -1) {
this.type = 'video';
}
else {
this.type = 'voice';
}
if (event.getAge()) {
setTimeout(function() {
if (self.state == 'ringing') {
self.state = 'ended';
self.hangupParty = 'remote'; // effectively
stopAllMedia(self);
if (self.peerConn.signalingState != 'closed') {
self.peerConn.close();
}
self.emit("onHangup", self);
}
}, this.msg.lifetime - event.getAge());
}
};
/**
* Configure this call from a hangup event.
* @param {MatrixEvent} event The m.call.hangup event
*/
MatrixCall.prototype.initWithHangup = function(event) {
// perverse as it may seem, sometimes we want to instantiate a call with a
// hangup message (because when getting the state of the room on load, events
// come in reverse order and we want to remember that a call has been hung up)
this.msg = event.getContent();
this.state = 'ended';
};
/**
* Answer a call.
*/
MatrixCall.prototype.answer = function() {
debuglog("Answering call " + this.callId);
var self = this;
if (!this.localAVStream && !this.waitForLocalAVStream) {
this.webRtc.getUserMedia(
_getUserMediaVideoContraints(this.type),
hookCallback(self, self.gotUserMediaForAnswer),
hookCallback(self, self.getUserMediaFailed)
);
this.state = 'wait_local_media';
} else if (this.localAVStream) {
this.gotUserMediaForAnswer(this.localAVStream);
} else if (this.waitForLocalAVStream) {
this.state = 'wait_local_media';
}
};
/**
* Replace this call with a new call, e.g. for glare resolution.
* @param {MatrixCall} newCall The new call.
*/
MatrixCall.prototype.replacedBy = function(newCall) {
debuglog(this.callId + " being replaced by " + newCall.callId);
if (this.state == 'wait_local_media') {
debuglog("Telling new call to wait for local media");
newCall.waitForLocalAVStream = true;
} else if (this.state == 'create_offer') {
debuglog("Handing local stream to new call");
newCall.gotUserMediaForAnswer(this.localAVStream);
delete(this.localAVStream);
} else if (this.state == 'invite_sent') {
debuglog("Handing local stream to new call");
newCall.gotUserMediaForAnswer(this.localAVStream);
delete(this.localAVStream);
}
newCall.localVideoElement = this.localVideoElement;
newCall.remoteVideoElement = this.remoteVideoElement;
this.successor = newCall;
this.hangup(true);
};
/**
* Hangup a call.
* @param {string} reason The reason why the call is being hung up.
* @param {boolean} suppressEvent True to suppress emitting an event.
*/
MatrixCall.prototype.hangup = function(reason, suppressEvent) {
debuglog("Ending call " + this.callId);
terminate(this, "local", reason, !suppressEvent);
var content = {
version: 0,
call_id: this.callId,
reason: reason
};
this.sendEvent('m.call.hangup', content);
};
/**
* Internal
* @param {Object} stream
*/
MatrixCall.prototype.gotUserMediaForInvite = function(stream) {
if (this.successor) {
this.successor.gotUserMediaForAnswer(stream);
return;
}
if (this.state == 'ended') {
return;
}
var self = this;
var videoEl = this.getLocalVideoElement();
if (videoEl && this.type == 'video') {
videoEl.autoplay = true;
videoEl.src = this.URL.createObjectURL(stream);
videoEl.muted = true;
setTimeout(function() {
var vel = self.getLocalVideoElement();
if (vel.play) {
vel.play();
}
}, 0);
}
this.localAVStream = stream;
var audioTracks = stream.getAudioTracks();
for (var i = 0; i < audioTracks.length; i++) {
audioTracks[i].enabled = true;
}
this.peerConn = _createPeerConnection(this);
this.peerConn.addStream(stream);
this.peerConn.createOffer(
hookCallback(self, self.gotLocalOffer),
hookCallback(self, self.getLocalOfferFailed)
);
self.state = 'create_offer';
};
/**
* Internal
* @param {Object} stream
*/
MatrixCall.prototype.gotUserMediaForAnswer = function(stream) {
var self = this;
if (self.state == 'ended') {
return;
}
var localVidEl = self.getLocalVideoElement();
if (localVidEl && self.type == 'video') {
localVidEl.autoplay = true;
localVidEl.src = self.URL.createObjectURL(stream);
localVidEl.muted = self;
setTimeout(function() {
var vel = self.getLocalVideoElement();
if (vel.play) {
vel.play();
}
}, 0);
}
self.localAVStream = stream;
var audioTracks = stream.getAudioTracks();
for (var i = 0; i < audioTracks.length; i++) {
audioTracks[i].enabled = true;
}
self.peerConn.addStream(stream);
var constraints = {
'mandatory': {
'OfferToReceiveAudio': true,
'OfferToReceiveVideo': self.type == 'video'
},
};
self.peerConn.createAnswer(constraints, function(description) {
debuglog("Created answer: " + description);
self.peerConn.setLocalDescription(description, function() {
var content = {
version: 0,
call_id: self.callId,
answer: {
sdp: self.peerConn.localDescription.sdp,
type: self.peerConn.localDescription.type
}
};
self.sendEvent('m.call.answer', content);
self.state = 'connecting';
}, function() {
debuglog("Error setting local description!");
});
});
self.state = 'create_answer';
};
/**
* Internal
* @param {Object} event
*/
MatrixCall.prototype.gotLocalIceCandidate = function(event) {
if (event.candidate) {
debuglog(
"Got local ICE " + event.candidate.sdpMid + " candidate: " +
event.candidate.candidate
);
// As with the offer, note we need to make a copy of this object, not
// pass the original: that broke in Chrome ~m43.
var c = {
candidate: event.candidate.candidate,
sdpMid: event.candidate.sdpMid,
sdpMLineIndex: event.candidate.sdpMLineIndex
};
this.sendCandidate(c);
}
};
/**
* Internal
* @param {Object} cand
*/
MatrixCall.prototype.gotRemoteIceCandidate = function(cand) {
if (this.state == 'ended') {
//debuglog("Ignoring remote ICE candidate because call has ended");
return;
}
debuglog("Got remote ICE " + cand.sdpMid + " candidate: " + cand.candidate);
this.peerConn.addIceCandidate(
new this.webRtc.RtcIceCandidate(cand),
function() {},
function(e) {}
);
};
/**
* Internal
* @param {Object} msg
*/
MatrixCall.prototype.receivedAnswer = function(msg) {
if (this.state == 'ended') {
return;
}
var self = this;
this.peerConn.setRemoteDescription(
new this.webRtc.RtcSessionDescription(msg.answer),
hookCallback(self, self.onSetRemoteDescriptionSuccess),
hookCallback(self, self.onSetRemoteDescriptionError)
);
this.state = 'connecting';
};
/**
* Internal
* @param {Object} description
*/
MatrixCall.prototype.gotLocalOffer = function(description) {
var self = this;
debuglog("Created offer: " + description);
if (self.state == 'ended') {
debuglog("Ignoring newly created offer on call ID " + self.callId +
" because the call has ended");
return;
}
self.peerConn.setLocalDescription(description, function() {
var content = {
version: 0,
call_id: self.callId,
// OpenWebRTC appears to add extra stuff (like the DTLS fingerprint)
// to the description when setting it on the peerconnection.
// According to the spec it should only add ICE
// candidates. Any ICE candidates that have already been generated
// at this point will probably be sent both in the offer and separately.
// Also, note that we have to make a new object here, copying the
// type and sdp properties.
// Passing the RTCSessionDescription object as-is doesn't work in
// Chrome (as of about m43).
offer: {
sdp: self.peerConn.localDescription.sdp,
type: self.peerConn.localDescription.type
},
lifetime: MatrixCall.CALL_TIMEOUT_MS
};
self.sendEvent('m.call.invite', content);
setTimeout(function() {
if (self.state == 'invite_sent') {
self.hangup('invite_timeout');
}
}, MatrixCall.CALL_TIMEOUT_MS);
self.state = 'invite_sent';
}, function() {
debuglog("Error setting local description!");
});
};
/**
* Internal
* @param {Object} error
*/
MatrixCall.prototype.getLocalOfferFailed = function(error) {
this.emit(
"error",
callError(MatrixCall.ERR_LOCAL_OFFER_FAILED, "Failed to start audio for call!")
);
};
/**
* Internal
*/
MatrixCall.prototype.getUserMediaFailed = function() {
this.emit(
"error",
callError(
MatrixCall.ERR_NO_USER_MEDIA,
"Couldn't start capturing media! Is your microphone set up and " +
"does this app have permission?"
)
);
this.hangup();
};
/**
* Internal
*/
MatrixCall.prototype.onIceConnectionStateChanged = function() {
if (this.state == 'ended') {
return; // because ICE can still complete as we're ending the call
}
debuglog(
"Ice connection state changed to: " + this.peerConn.iceConnectionState
);
// ideally we'd consider the call to be connected when we get media but
// chrome doesn't implement any of the 'onstarted' events yet
if (this.peerConn.iceConnectionState == 'completed' ||
this.peerConn.iceConnectionState == 'connected') {
this.state = 'connected';
this.didConnect = true;
} else if (this.peerConn.iceConnectionState == 'failed') {
this.hangup('ice_failed');
}
};
/**
* Internal
*/
MatrixCall.prototype.onSignallingStateChanged = function() {
debuglog(
"call " + this.callId + ": Signalling state changed to: " +
this.peerConn.signalingState
);
};
/**
* Internal
*/
MatrixCall.prototype.onSetRemoteDescriptionSuccess = function() {
debuglog("Set remote description");
};
/**
* Internal
* @param {Object} e
*/
MatrixCall.prototype.onSetRemoteDescriptionError = function(e) {
debuglog("Failed to set remote description" + e);
};
/**
* Internal
* @param {Object} event
*/
MatrixCall.prototype.onAddStream = function(event) {
debuglog("Stream added" + event);
var s = event.stream;
this.remoteAVStream = s;
if (this.direction == 'inbound') {
if (s.getVideoTracks().length > 0) {
this.type = 'video';
} else {
this.type = 'voice';
}
}
var self = this;
forAllTracksOnStream(s, function(t) {
// not currently implemented in chrome
t.onstarted = self.onRemoteStreamTrackStarted;
});
event.stream.onended = function(e) { self.onRemoteStreamEnded(e); };
// not currently implemented in chrome
event.stream.onstarted = function(e) { self.onRemoteStreamStarted(e); };
this.tryPlayRemoteStream();
};
/**
* Internal
* @param {Object} event
*/
MatrixCall.prototype.tryPlayRemoteStream = function(event) {
if (this.getRemoteVideoElement() && this.remoteAVStream) {
var player = this.getRemoteVideoElement();
player.autoplay = true;
player.src = this.URL.createObjectURL(this.remoteAVStream);
var self = this;
setTimeout(function() {
var vel = self.getRemoteVideoElement();
if (vel.play) {
vel.play();
}
// OpenWebRTC does not support oniceconnectionstatechange yet
if (self.webRtc.isOpenWebRTC()) {
self.state = 'connected';
}
}, 0);
}
};
/**
* Internal
* @param {Object} event
*/
MatrixCall.prototype.onRemoteStreamStarted = function(event) {
this.state = 'connected';
};
/**
* Internal
* @param {Object} event
*/
MatrixCall.prototype.onRemoteStreamEnded = function(event) {
debuglog("Remote stream ended");
this.state = 'ended';
this.hangupParty = 'remote';
stopAllMedia(this);
if (this.peerConn.signalingState != 'closed') {
this.peerConn.close();
}
this.emit("onHangup", this);
};
/**
* Internal
* @param {Object} event
*/
MatrixCall.prototype.onRemoteStreamTrackStarted = function(event) {
this.state = 'connected';
};
/**
* Internal
* @param {Object} msg
*/
MatrixCall.prototype.onHangupReceived = function(msg) {
debuglog("Hangup received");
terminate(this, "remote", msg.reason, true);
};
/**
* Internal
* @param {Object} msg
*/
MatrixCall.prototype.onAnsweredElsewhere = function(msg) {
debuglog("Answered elsewhere");
terminate(this, "remote", "answered_elsewhere", true);
};
/**
* Internal
* @param {string} eventType
* @param {Object} content
* @return {Promise}
*/
MatrixCall.prototype.sendEvent = function(eventType, content) {
return this.client.sendEvent(this.roomId, eventType, content);
};
/**
* Internal
* @param {Object} content
*/
MatrixCall.prototype.sendCandidate = function(content) {
// Sends candidates with are sent in a special way because we try to amalgamate
// them into one message
this.candidateSendQueue.push(content);
var self = this;
if (this.candidateSendTries === 0) {
setTimeout(function() {
_sendCandidateQueue(self);
}, 100);
}
};
var terminate = function(self, hangupParty, hangupReason, shouldEmit) {
if (self.getRemoteVideoElement() && self.getRemoteVideoElement().pause) {
self.getRemoteVideoElement().pause();
}
if (self.getLocalVideoElement() && self.getLocalVideoElement().pause) {
self.getLocalVideoElement().pause();
}
self.state = 'ended';
self.hangupParty = hangupParty;
self.hangupReason = hangupReason;
stopAllMedia(self);
if (self.peerConn &&
(hangupParty === "local" || self.peerConn.signalingState != 'closed')) {
self.peerConn.close();
}
if (shouldEmit) {
self.emit("onHangup", self);
}
};
var stopAllMedia = function(self) {
if (self.localAVStream) {
forAllTracksOnStream(self.localAVStream, function(t) {
if (t.stop) {
t.stop();
}
});
// also call stop on the main stream so firefox will stop sharing
// the mic
if (self.localAVStream.stop) {
self.localAVStream.stop();
}
}
if (self.remoteAVStream) {
forAllTracksOnStream(self.remoteAVStream, function(t) {
if (t.stop) {
t.stop();
}
});
}
};
var checkForErrorListener = function(self) {
if (self.listeners("error").length === 0) {
throw new Error(
"You MUST attach an error listener using call.on('error', function() {})"
);
}
};
var callError = function(code, msg) {
var e = new Error(msg);
e.code = code;
return e;
};
var debuglog = function() {
if (DEBUG) {
console.log.apply(console, arguments);
}
};
var _sendCandidateQueue = function(self) {
if (self.candidateSendQueue.length === 0) {
return;
}
var cands = self.candidateSendQueue;
self.candidateSendQueue = [];
++self.candidateSendTries;
var content = {
version: 0,
call_id: self.callId,
candidates: cands
};
debuglog("Attempting to send " + cands.length + " candidates");
self.sendEvent('m.call.candidates', content).then(function() {
self.candidateSendTries = 0;
_sendCandidateQueue(self);
}, function(error) {
for (var i = 0; i < cands.length; i++) {
self.candidateSendQueue.push(cands[i]);
}
if (self.candidateSendTries > 5) {
debuglog(
"Failed to send candidates on attempt %s. Giving up for now.",
self.candidateSendTries
);
self.candidateSendTries = 0;
return;
}
var delayMs = 500 * Math.pow(2, self.candidateSendTries);
++self.candidateSendTries;
debuglog("Failed to send candidates. Retrying in " + delayMs + "ms");
setTimeout(function() {
_sendCandidateQueue(self);
}, delayMs);
});
};
var _placeCallWithConstraints = function(self, constraints) {
self.emit("callPlaced", self);
self.webRtc.getUserMedia(
constraints,
hookCallback(self, self.gotUserMediaForInvite),
hookCallback(self, self.getUserMediaFailed)
);
self.state = 'wait_local_media';
self.direction = 'outbound';
self.config = constraints;
};
var _createPeerConnection = function(self) {
var servers = self.turnServers;
if (self.webRtc.vendor === "mozilla") {
// modify turnServers struct to match what mozilla expects.
servers = [];
for (var i = 0; i < self.turnServers.length; i++) {
for (var j = 0; j < self.turnServers[i].urls.length; j++) {
servers.push({
url: self.turnServers[i].urls[j],
username: self.turnServers[i].username,
credential: self.turnServers[i].credential
});
}
}
}
var pc = new self.webRtc.RtcPeerConnection({
iceServers: servers
});
pc.oniceconnectionstatechange = hookCallback(self, self.onIceConnectionStateChanged);
pc.onsignalingstatechange = hookCallback(self, self.onSignallingStateChanged);
pc.onicecandidate = hookCallback(self, self.gotLocalIceCandidate);
pc.onaddstream = hookCallback(self, self.onAddStream);
return pc;
};
var _getUserMediaVideoContraints = function(callType) {
switch (callType) {
case 'voice':
return ({audio: true, video: false});
case 'video':
return ({audio: true, video: {
mandatory: {
minWidth: 640,
maxWidth: 640,
minHeight: 360,
maxHeight: 360,
}
}});
}
};
var hookCallback = function(call, fn) {
return function() {
return fn.apply(call, arguments);
};
};
var forAllVideoTracksOnStream = function(s, f) {
var tracks = s.getVideoTracks();
for (var i = 0; i < tracks.length; i++) {
f(tracks[i]);
}
};
var forAllAudioTracksOnStream = function(s, f) {
var tracks = s.getAudioTracks();
for (var i = 0; i < tracks.length; i++) {
f(tracks[i]);
}
};
var forAllTracksOnStream = function(s, f) {
forAllVideoTracksOnStream(s, f);
forAllAudioTracksOnStream(s, f);
};
/** The MatrixCall class. */
module.exports.MatrixCall = MatrixCall;
/**
* Create a new Matrix call for the browser.
* @param {MatrixClient} client The client instance to use.
* @param {string} roomId The room the call is in.
* @return {MatrixCall} the call or null if the browser doesn't support calling.
*/
module.exports.createNewMatrixCall = function(client, roomId) {
var w = global.window;
var doc = global.document;
var webRtc = {};
webRtc.isOpenWebRTC = function() {
var scripts = doc.getElementById("script");
if (!scripts || !scripts.length) {
return false;
}
for (var i = 0; i < scripts.length; i++) {
if (scripts[i].src.indexOf("owr.js") > -1) {
return true;
}
}
return false;
};
var getUserMedia = (
w.navigator.getUserMedia || w.navigator.webkitGetUserMedia ||
w.navigator.mozGetUserMedia
);
if (getUserMedia) {
webRtc.getUserMedia = function() {
return getUserMedia.apply(w.navigator, arguments);
};
}
webRtc.RtcPeerConnection = (
w.RTCPeerConnection || w.webkitRTCPeerConnection || w.mozRTCPeerConnection
);
webRtc.RtcSessionDescription = (
w.RTCSessionDescription || w.webkitRTCSessionDescription ||
w.mozRTCSessionDescription
);
webRtc.RtcIceCandidate = (
w.RTCIceCandidate || w.webkitRTCIceCandidate || w.mozRTCIceCandidate
);
webRtc.vendor = null;
if (w.mozRTCPeerConnection) {
webRtc.vendor = "mozilla";
}
else if (w.webkitRTCPeerConnection) {
webRtc.vendor = "webkit";
}
else if (w.RTCPeerConnection) {
webRtc.vendor = "generic";
}
if (!webRtc.RtcIceCandidate || !webRtc.RtcSessionDescription ||
!webRtc.RtcPeerConnection || !webRtc.getUserMedia) {
return null; // Web RTC is not supported.
}
var opts = {
webRtc: webRtc,
client: client,
URL: w.URL,
roomId: roomId
};
return new MatrixCall(opts);
};