/* jshint -W117 */ var logger = require("jitsi-meet-logger").getLogger(__filename); var JingleSession = require("./JingleSession"); var TraceablePeerConnection = require("./TraceablePeerConnection"); var MediaType = require("../../service/RTC/MediaType"); var SDPDiffer = require("./SDPDiffer"); var SDPUtil = require("./SDPUtil"); var SDP = require("./SDP"); var async = require("async"); var XMPPEvents = require("../../service/xmpp/XMPPEvents"); var RTCBrowserType = require("../RTC/RTCBrowserType"); var RTC = require("../RTC/RTC"); /** * Constant tells how long we're going to wait for IQ response, before timeout * error is triggered. * @type {number} */ var IQ_TIMEOUT = 10000; // Jingle stuff function JingleSessionPC(me, sid, peerjid, connection, media_constraints, ice_config, service, eventEmitter) { JingleSession.call(this, me, sid, peerjid, connection, media_constraints, ice_config, service, eventEmitter); this.localSDP = null; this.remoteSDP = null; this.hadstuncandidate = false; this.hadturncandidate = false; this.lasticecandidate = false; this.closed = false; this.addssrc = []; this.removessrc = []; this.pendingop = null; this.modifyingLocalStreams = false; this.modifiedSSRCs = {}; /** * A map that stores SSRCs of remote streams. And is used only locally * We store the mapping when jingle is received, and later is used * onaddstream webrtc event where we have only the ssrc * FIXME: This map got filled and never cleaned and can grow durring long * conference * @type {{}} maps SSRC number to jid */ this.ssrcOwners = {}; this.webrtcIceUdpDisable = !!this.service.options.webrtcIceUdpDisable; this.webrtcIceTcpDisable = !!this.service.options.webrtcIceTcpDisable; this.modifySourcesQueue = async.queue(this._modifySources.bind(this), 1); // We start with the queue paused. We resume it when the signaling state is // stable and the ice connection state is connected. this.modifySourcesQueue.pause(); } //XXX this is badly broken... JingleSessionPC.prototype = JingleSession.prototype; JingleSessionPC.prototype.constructor = JingleSessionPC; JingleSessionPC.prototype.updateModifySourcesQueue = function() { var signalingState = this.peerconnection.signalingState; var iceConnectionState = this.peerconnection.iceConnectionState; if (signalingState === 'stable' && iceConnectionState === 'connected') { this.modifySourcesQueue.resume(); } else { this.modifySourcesQueue.pause(); } }; JingleSessionPC.prototype.doInitialize = function () { var self = this; this.hadstuncandidate = false; this.hadturncandidate = false; this.lasticecandidate = false; // True if reconnect is in progress this.isreconnect = false; // Set to true if the connection was ever stable this.wasstable = false; this.peerconnection = new TraceablePeerConnection( this.connection.jingle.ice_config, RTC.getPCConstraints(), this); this.peerconnection.onicecandidate = function (ev) { if (!ev) { // There was an incomplete check for ev before which left the last // line of the function unprotected from a potential throw of an // exception. Consequently, it may be argued that the check is // unnecessary. Anyway, I'm leaving it and making the check // complete. return; } var candidate = ev.candidate; if (candidate) { // Discard candidates of disabled protocols. var protocol = candidate.protocol; if (typeof protocol === 'string') { protocol = protocol.toLowerCase(); if (protocol == 'tcp') { if (self.webrtcIceTcpDisable) return; } else if (protocol == 'udp') { if (self.webrtcIceUdpDisable) return; } } } self.sendIceCandidate(candidate); }; this.peerconnection.onaddstream = function (event) { self.remoteStreamAdded(event.stream); }; this.peerconnection.onremovestream = function (event) { self.remoteStreamRemoved(event.stream); }; this.peerconnection.onsignalingstatechange = function (event) { if (!(self && self.peerconnection)) return; if (self.peerconnection.signalingState === 'stable') { self.wasstable = true; } self.updateModifySourcesQueue(); }; /** * The oniceconnectionstatechange event handler contains the code to execute when the iceconnectionstatechange event, * of type Event, is received by this RTCPeerConnection. Such an event is sent when the value of * RTCPeerConnection.iceConnectionState changes. * * @param event the event containing information about the change */ this.peerconnection.oniceconnectionstatechange = function (event) { if (!(self && self.peerconnection)) return; var now = window.performance.now(); self.room.connectionTimes["ice.state." + self.peerconnection.iceConnectionState] = now; logger.log("(TIME) ICE " + self.peerconnection.iceConnectionState + ":\t", now); self.updateModifySourcesQueue(); switch (self.peerconnection.iceConnectionState) { case 'connected': // Informs interested parties that the connection has been restored. if (self.peerconnection.signalingState === 'stable' && self.isreconnect) self.room.eventEmitter.emit(XMPPEvents.CONNECTION_RESTORED); self.isreconnect = false; break; case 'disconnected': if(self.closed) break; self.isreconnect = true; // Informs interested parties that the connection has been interrupted. if (self.wasstable) self.room.eventEmitter.emit(XMPPEvents.CONNECTION_INTERRUPTED); break; case 'failed': self.room.eventEmitter.emit(XMPPEvents.CONFERENCE_SETUP_FAILED); break; } }; this.peerconnection.onnegotiationneeded = function (event) { self.room.eventEmitter.emit(XMPPEvents.PEERCONNECTION_READY, self); }; }; JingleSessionPC.prototype.sendIceCandidate = function (candidate) { var self = this; if (candidate && !this.lasticecandidate) { var ice = SDPUtil.iceparams(this.localSDP.media[candidate.sdpMLineIndex], this.localSDP.session); var jcand = SDPUtil.candidateToJingle(candidate.candidate); if (!(ice && jcand)) { logger.error('failed to get ice && jcand'); return; } ice.xmlns = 'urn:xmpp:jingle:transports:ice-udp:1'; if (jcand.type === 'srflx') { this.hadstuncandidate = true; } else if (jcand.type === 'relay') { this.hadturncandidate = true; } if (this.usedrip) { if (this.drip_container.length === 0) { // start 20ms callout window.setTimeout(function () { if (self.drip_container.length === 0) return; self.sendIceCandidates(self.drip_container); self.drip_container = []; }, 20); } this.drip_container.push(candidate); } else { self.sendIceCandidates([candidate]); } } else { logger.log('sendIceCandidate: last candidate.'); // FIXME: remember to re-think in ICE-restart this.lasticecandidate = true; logger.log('Have we encountered any srflx candidates? ' + this.hadstuncandidate); logger.log('Have we encountered any relay candidates? ' + this.hadturncandidate); } }; JingleSessionPC.prototype.sendIceCandidates = function (candidates) { logger.log('sendIceCandidates', candidates); var cand = $iq({to: this.peerjid, type: 'set'}) .c('jingle', {xmlns: 'urn:xmpp:jingle:1', action: 'transport-info', initiator: this.initiator, sid: this.sid}); for (var mid = 0; mid < this.localSDP.media.length; mid++) { var cands = candidates.filter(function (el) { return el.sdpMLineIndex == mid; }); var mline = SDPUtil.parse_mline(this.localSDP.media[mid].split('\r\n')[0]); if (cands.length > 0) { var ice = SDPUtil.iceparams(this.localSDP.media[mid], this.localSDP.session); ice.xmlns = 'urn:xmpp:jingle:transports:ice-udp:1'; cand.c('content', {creator: this.initiator == this.me ? 'initiator' : 'responder', name: (cands[0].sdpMid? cands[0].sdpMid : mline.media) }).c('transport', ice); for (var i = 0; i < cands.length; i++) { cand.c('candidate', SDPUtil.candidateToJingle(cands[i].candidate)).up(); } // add fingerprint var fingerprint_line = SDPUtil.find_line(this.localSDP.media[mid], 'a=fingerprint:', this.localSDP.session); if (fingerprint_line) { var tmp = SDPUtil.parse_fingerprint(fingerprint_line); tmp.required = true; cand.c( 'fingerprint', {xmlns: 'urn:xmpp:jingle:apps:dtls:0'}) .t(tmp.fingerprint); delete tmp.fingerprint; cand.attrs(tmp); cand.up(); } cand.up(); // transport cand.up(); // content } } // might merge last-candidate notification into this, but it is called alot later. See webrtc issue #2340 //logger.log('was this the last candidate', this.lasticecandidate); this.connection.sendIQ( cand, null, this.newJingleErrorHandler(cand), IQ_TIMEOUT); }; JingleSessionPC.prototype.readSsrcInfo = function (contents) { var self = this; $(contents).each(function (idx, content) { var name = $(content).attr('name'); var mediaType = this.getAttribute('name'); var ssrcs = $(content).find('description>source[xmlns="urn:xmpp:jingle:apps:rtp:ssma:0"]'); ssrcs.each(function () { var ssrc = this.getAttribute('ssrc'); $(this).find('>ssrc-info[xmlns="http://jitsi.org/jitmeet"]').each( function () { var owner = this.getAttribute('owner'); self.ssrcOwners[ssrc] = owner; } ); }); }); }; /** * Does accept incoming Jingle 'session-initiate' and should send * 'session-accept' in result. * @param jingleOffer jQuery selector pointing to the jingle element of * the offer IQ * @param success callback called when we accept incoming session successfully * and receive RESULT packet to 'session-accept' sent. * @param failure function(error) called if for any reason we fail to accept * the incoming offer. 'error' argument can be used to log some details * about the error. */ JingleSessionPC.prototype.acceptOffer = function(jingleOffer, success, failure) { this.state = 'active'; this.setOfferCycle(jingleOffer, function() { // setOfferCycle succeeded, now we have self.localSDP up to date // Let's send an answer ! // FIXME we may not care about RESULT packet for session-accept // then we should either call 'success' here immediately or // modify sendSessionAccept method to do that this.sendSessionAccept(this.localSDP, success, failure); }.bind(this), failure); }; /** * This is a setRemoteDescription/setLocalDescription cycle which starts at * converting Strophe Jingle IQ into remote offer SDP. Once converted * setRemoteDescription, createAnswer and setLocalDescription calls follow. * @param jingleOfferIq jQuery selector pointing to the jingle element of * the offer IQ * @param success callback called when sRD/sLD cycle finishes successfully. * @param failure callback called with an error object as an argument if we fail * at any point during setRD, createAnswer, setLD. */ JingleSessionPC.prototype.setOfferCycle = function (jingleOfferIq, success, failure) { // Set Jingle offer as RD this.setOffer(jingleOfferIq, function() { // Set offer OK, now let's try create an answer this.createAnswer(function(answer) { // Create answer OK, set it as local SDP this.setLocalDescription(answer, success, failure); }.bind(this), failure); }.bind(this), failure); }; /** * Sets remote offer on PeerConnection by converting given Jingle offer IQ into * SDP and setting it as remote description. * @param jingleOfferIq jQuery selector pointing to the jingle element of * the offer IQ * @param success callback called when setRemoteDescription on PeerConnection * succeeds * @param failure callback called with an error argument when * setRemoteDescription fails. */ JingleSessionPC.prototype.setOffer = function (jingleOfferIq, success, failure) { this.remoteSDP = new SDP(''); if (this.webrtcIceTcpDisable) { this.remoteSDP.removeTcpCandidates = true; } if (this.webrtcIceUdpDisable) { this.remoteSDP.removeUdpCandidates = true; } this.remoteSDP.fromJingle(jingleOfferIq); this.readSsrcInfo($(jingleOfferIq).find(">content")); var remotedesc = new RTCSessionDescription({type: 'offer', sdp: this.remoteSDP.raw}); this.peerconnection.setRemoteDescription(remotedesc, function () { //logger.log('setRemoteDescription success'); if (success) { success(); } }, function (e) { logger.error('setRemoteDescription error', e); if (failure) failure(e); JingleSessionPC.onJingleFatalError(this, e); }.bind(this) ); }; /** * This is a wrapper to PeerConnection.createAnswer in order to generate failure * event when error occurs. It also includes "media_constraints" if any are set * on this JingleSessionPC instance. * @param success callback called when PC.createAnswer succeeds, SDP will be * the first argument * @param failure callback called with error argument when setAnswer fails */ JingleSessionPC.prototype.createAnswer = function (success, failure) { //logger.log('createAnswer'); var self = this; this.peerconnection.createAnswer( function (answer) { var modifiedAnswer = new SDP(answer.sdp); JingleSessionPC._fixAnswerRFC4145Setup( /* offer */ self.remoteSDP, /* answer */ modifiedAnswer); answer.sdp = modifiedAnswer.raw; success(answer); }, function (error) { logger.error("createAnswer failed", error); if (failure) failure(error); self.room.eventEmitter.emit( XMPPEvents.CONFERENCE_SETUP_FAILED, error); }, this.media_constraints ); }; JingleSessionPC.prototype.setLocalDescription = function (sdp, success, failure) { var self = this; this.localSDP = new SDP(sdp.sdp); sdp.sdp = this.localSDP.raw; this.peerconnection.setLocalDescription(sdp, function () { if (success) success(); }, function (error) { logger.error('setLocalDescription failed', error); if (failure) failure(error); self.room.eventEmitter.emit(XMPPEvents.CONFERENCE_SETUP_FAILED); } ); // Some checks for STUN and TURN candiates present in local SDP // Eventually could be removed as we don't really care var cands = SDPUtil.find_lines(this.localSDP.raw, 'a=candidate:'); for (var j = 0; j < cands.length; j++) { var cand = SDPUtil.parse_icecandidate(cands[j]); if (cand.type == 'srflx') { this.hadstuncandidate = true; } else if (cand.type == 'relay') { this.hadturncandidate = true; } } }; /** * Modifies the values of the setup attributes (defined by * {@link http://tools.ietf.org/html/rfc4145#section-4}) of a specific SDP * answer in order to overcome a delay of 1 second in the connection * establishment between Chrome and Videobridge. * * @param {SDP} offer - the SDP offer to which the specified SDP answer is * being prepared to respond * @param {SDP} answer - the SDP to modify * @private */ JingleSessionPC._fixAnswerRFC4145Setup = function (offer, answer) { if (!RTCBrowserType.isChrome()) { // It looks like Firefox doesn't agree with the fix (at least in its // current implementation) because it effectively remains active even // after we tell it to become passive. Apart from Firefox which I tested // after the fix was deployed, I tested Chrome only. In order to prevent // issues with other browsers, limit the fix to Chrome for the time // being. return; } // XXX Videobridge is the (SDP) offerer and WebRTC (e.g. Chrome) is the // answerer (as orchestrated by Jicofo). In accord with // http://tools.ietf.org/html/rfc5245#section-5.2 and because both peers // are ICE FULL agents, Videobridge will take on the controlling role and // WebRTC will take on the controlled role. In accord with // https://tools.ietf.org/html/rfc5763#section-5, Videobridge will use the // setup attribute value of setup:actpass and WebRTC will be allowed to // choose either the setup attribute value of setup:active or // setup:passive. Chrome will by default choose setup:active because it is // RECOMMENDED by the respective RFC since setup:passive adds additional // latency. The case of setup:active allows WebRTC to send a DTLS // ClientHello as soon as an ICE connectivity check of its succeeds. // Unfortunately, Videobridge will be unable to respond immediately because // may not have WebRTC's answer or may have not completed the ICE // connectivity establishment. Even more unfortunate is that in the // described scenario Chrome's DTLS implementation will insist on // retransmitting its ClientHello after a second (the time is in accord // with the respective RFC) and will thus cause the whole connection // establishment to exceed at least 1 second. To work around Chrome's // idiosyncracy, don't allow it to send a ClientHello i.e. change its // default choice of setup:active to setup:passive. if (offer && answer && offer.media && answer.media && offer.media.length == answer.media.length) { answer.media.forEach(function (a, i) { if (SDPUtil.find_line( offer.media[i], 'a=setup:actpass', offer.session)) { answer.media[i] = a.replace(/a=setup:active/g, 'a=setup:passive'); } }); answer.raw = answer.session + answer.media.join(''); } }; /** * Although it states "replace transport" it does accept full Jingle offer * which should contain new ICE transport details. * @param jingleOfferElem an element Jingle IQ that contains new offer and * transport info. * @param success callback called when we succeed to accept new offer. * @param failure function(error) called when we fail to accept new offer. */ JingleSessionPC.prototype.replaceTransport = function (jingleOfferElem, success, failure) { // Set offer as RD this.setOfferCycle(jingleOfferElem, function () { // Set local description OK, now localSDP up to date this.sendTransportAccept(this.localSDP, success, failure); }.bind(this), failure); }; /** * Sends Jingle 'session-accept' message. * @param localSDP the 'SDP' object with local session description * @param success callback called when we recive 'RESULT' packet for * 'session-accept' * @param failure function(error) called when we receive an error response or * when the request has timed out. */ JingleSessionPC.prototype.sendSessionAccept = function (localSDP, success, failure) { var accept = $iq({to: this.peerjid, type: 'set'}) .c('jingle', {xmlns: 'urn:xmpp:jingle:1', action: 'session-accept', initiator: this.initiator, responder: this.responder, sid: this.sid }); if (this.webrtcIceTcpDisable) { localSDP.removeTcpCandidates = true; } if (this.webrtcIceUdpDisable) { localSDP.removeUdpCandidates = true; } localSDP.toJingle( accept, this.initiator == this.me ? 'initiator' : 'responder', null); this.fixJingle(accept); // Calling tree() to print something useful accept = accept.tree(); logger.info("Sending session-accept", accept); this.connection.sendIQ(accept, success, this.newJingleErrorHandler(accept, failure), IQ_TIMEOUT); // XXX Videobridge needs WebRTC's answer (ICE ufrag and pwd, DTLS // fingerprint and setup) ASAP in order to start the connection // establishment. this.connection.flush(); }; /** * Sends Jingle 'transport-accept' message which is a response to * 'transport-replace'. * @param localSDP the 'SDP' object with local session description * @param success callback called when we receive 'RESULT' packet for * 'transport-replace' * @param failure function(error) called when we receive an error response or * when the request has timed out. */ JingleSessionPC.prototype.sendTransportAccept = function(localSDP, success, failure) { var self = this; var tAccept = $iq({to: this.peerjid, type: 'set'}) .c('jingle', {xmlns: 'urn:xmpp:jingle:1', action: 'transport-accept', initiator: this.initiator, sid: this.sid}); localSDP.media.forEach(function(medialines, idx){ var mline = SDPUtil.parse_mline(medialines.split('\r\n')[0]); tAccept.c('content', { creator: self.initiator == self.me ? 'initiator' : 'responder', name: mline.media } ); localSDP.transportToJingle(idx, tAccept); tAccept.up(); }); // Calling tree() to print something useful to the logger tAccept = tAccept.tree(); console.info("Sending transport-accept: ", tAccept); self.connection.sendIQ(tAccept, success, self.newJingleErrorHandler(tAccept, failure), IQ_TIMEOUT); }; /** * Sends Jingle 'transport-reject' message which is a response to * 'transport-replace'. * @param success callback called when we receive 'RESULT' packet for * 'transport-replace' * @param failure function(error) called when we receive an error response or * when the request has timed out. */ JingleSessionPC.prototype.sendTransportReject = function(success, failure) { // Send 'transport-reject', so that the focus will // know that we've failed var tReject = $iq({to: this.peerjid, type: 'set'}) .c('jingle', {xmlns: 'urn:xmpp:jingle:1', action: 'transport-reject', initiator: this.initiator, sid: this.sid}); tReject = tReject.tree(); logger.info("Sending 'transport-reject", tReject); this.connection.sendIQ(tReject, success, this.newJingleErrorHandler(tReject, failure), IQ_TIMEOUT); }; JingleSessionPC.prototype.terminate = function (reason, text, success, failure) { var term = $iq({to: this.peerjid, type: 'set'}) .c('jingle', {xmlns: 'urn:xmpp:jingle:1', action: 'session-terminate', initiator: this.initiator, sid: this.sid}) .c('reason') .c(reason || 'success'); if (text) { term.up().c('text').t(text); } // Calling tree() to print something useful term = term.tree(); logger.info("Sending session-terminate", term); this.connection.sendIQ( term, success, this.newJingleErrorHandler(term, failure), IQ_TIMEOUT); // this should result in 'onTerminated' being called by strope.jingle.js this.connection.jingle.terminate(this.sid); }; JingleSessionPC.prototype.onTerminated = function (reasonCondition, reasonText) { this.state = 'ended'; // Do something with reason and reasonCondition when we start to care //this.reasonCondition = reasonCondition; //this.reasonText = reasonText; logger.info("Session terminated", this, reasonCondition, reasonText); this.close(); }; /** * Handles a Jingle source-add message for this Jingle session. * @param elem An array of Jingle "content" elements. */ JingleSessionPC.prototype.addSource = function (elem) { var self = this; // FIXME: dirty waiting if (!this.peerconnection.localDescription) { logger.warn("addSource - localDescription not ready yet") setTimeout(function() { self.addSource(elem); }, 200 ); return; } logger.log('addssrc', new Date().getTime()); logger.log('ice', this.peerconnection.iceConnectionState); this.readSsrcInfo(elem); var sdp = new SDP(this.peerconnection.remoteDescription.sdp); var mySdp = new SDP(this.peerconnection.localDescription.sdp); $(elem).each(function (idx, content) { var name = $(content).attr('name'); var lines = ''; $(content).find('ssrc-group[xmlns="urn:xmpp:jingle:apps:rtp:ssma:0"]').each(function() { var semantics = this.getAttribute('semantics'); var ssrcs = $(this).find('>source').map(function () { return this.getAttribute('ssrc'); }).get(); if (ssrcs.length) { lines += 'a=ssrc-group:' + semantics + ' ' + ssrcs.join(' ') + '\r\n'; } }); var tmp = $(content).find('source[xmlns="urn:xmpp:jingle:apps:rtp:ssma:0"]'); // can handle both >source and >description>source tmp.each(function () { var ssrc = $(this).attr('ssrc'); if(mySdp.containsSSRC(ssrc)){ /** * This happens when multiple participants change their streams at the same time and * ColibriFocus.modifySources have to wait for stable state. In the meantime multiple * addssrc are scheduled for update IQ. See */ logger.warn("Got add stream request for my own ssrc: "+ssrc); return; } if (sdp.containsSSRC(ssrc)) { logger.warn("Source-add request for existing SSRC: " + ssrc); return; } $(this).find('>parameter').each(function () { lines += 'a=ssrc:' + ssrc + ' ' + $(this).attr('name'); if ($(this).attr('value') && $(this).attr('value').length) lines += ':' + $(this).attr('value'); lines += '\r\n'; }); }); sdp.media.forEach(function(media, idx) { if (!SDPUtil.find_line(media, 'a=mid:' + name)) return; sdp.media[idx] += lines; if (!self.addssrc[idx]) self.addssrc[idx] = ''; self.addssrc[idx] += lines; }); sdp.raw = sdp.session + sdp.media.join(''); }); this.modifySourcesQueue.push(function() { // When a source is added and if this is FF, a new channel is allocated // for receiving the added source. We need to diffuse the SSRC of this // new recvonly channel to the rest of the peers. logger.log('modify sources done'); var newSdp = new SDP(self.peerconnection.localDescription.sdp); logger.log("SDPs", mySdp, newSdp); self.notifyMySSRCUpdate(mySdp, newSdp); }); }; /** * Handles a Jingle source-remove message for this Jingle session. * @param elem An array of Jingle "content" elements. */ JingleSessionPC.prototype.removeSource = function (elem) { var self = this; // FIXME: dirty waiting if (!this.peerconnection.localDescription) { logger.warn("removeSource - localDescription not ready yet"); setTimeout(function() { self.removeSource(elem); }, 200 ); return; } logger.log('removessrc', new Date().getTime()); logger.log('ice', this.peerconnection.iceConnectionState); var sdp = new SDP(this.peerconnection.remoteDescription.sdp); var mySdp = new SDP(this.peerconnection.localDescription.sdp); $(elem).each(function (idx, content) { var name = $(content).attr('name'); var lines = ''; $(content).find('ssrc-group[xmlns="urn:xmpp:jingle:apps:rtp:ssma:0"]').each(function() { var semantics = this.getAttribute('semantics'); var ssrcs = $(this).find('>source').map(function () { return this.getAttribute('ssrc'); }).get(); if (ssrcs.length) { lines += 'a=ssrc-group:' + semantics + ' ' + ssrcs.join(' ') + '\r\n'; } }); var ssrcs = []; var tmp = $(content).find('source[xmlns="urn:xmpp:jingle:apps:rtp:ssma:0"]'); // can handle both >source and >description>source tmp.each(function () { var ssrc = $(this).attr('ssrc'); // This should never happen, but can be useful for bug detection if(mySdp.containsSSRC(ssrc)){ logger.error("Got remove stream request for my own ssrc: "+ssrc); return; } ssrcs.push(ssrc); }); sdp.media.forEach(function(media, idx) { if (!SDPUtil.find_line(media, 'a=mid:' + name)) return; if (!self.removessrc[idx]) self.removessrc[idx] = ''; ssrcs.forEach(function(ssrc) { var ssrcLines = SDPUtil.find_lines(media, 'a=ssrc:' + ssrc); if (ssrcLines.length) self.removessrc[idx] += ssrcLines.join("\r\n")+"\r\n"; // Clear any pending 'source-add' for this SSRC if (self.addssrc[idx]) { self.addssrc[idx] = self.addssrc[idx].replace( new RegExp('^a=ssrc:'+ssrc+' .*\r\n', 'gm'), ''); } }); self.removessrc[idx] += lines; }); sdp.raw = sdp.session + sdp.media.join(''); }); this.modifySourcesQueue.push(function() { // When a source is removed and if this is FF, the recvonly channel that // receives the remote stream is deactivated . We need to diffuse the // recvonly SSRC removal to the rest of the peers. logger.log('modify sources done'); var newSdp = new SDP(self.peerconnection.localDescription.sdp); logger.log("SDPs", mySdp, newSdp); self.notifyMySSRCUpdate(mySdp, newSdp); }); }; JingleSessionPC.prototype._modifySources = function (successCallback, queueCallback) { var self = this; if (this.peerconnection.signalingState == 'closed') return; if (!(this.addssrc.length || this.removessrc.length || this.pendingop !== null || this.modifyingLocalStreams)){ // There is nothing to do since scheduled job might have been // executed by another succeeding call if(successCallback){ successCallback(); } queueCallback(); return; } // Reset switch streams flags this.modifyingLocalStreams = false; var sdp = new SDP(this.peerconnection.remoteDescription.sdp); // add sources this.addssrc.forEach(function(lines, idx) { sdp.media[idx] += lines; }); this.addssrc = []; // remove sources this.removessrc.forEach(function(lines, idx) { lines = lines.split('\r\n'); lines.pop(); // remove empty last element; lines.forEach(function(line) { sdp.media[idx] = sdp.media[idx].replace(line + '\r\n', ''); }); }); this.removessrc = []; sdp.raw = sdp.session + sdp.media.join(''); this.peerconnection.setRemoteDescription(new RTCSessionDescription({type: 'offer', sdp: sdp.raw}), function() { if(self.signalingState == 'closed') { logger.error("createAnswer attempt on closed state"); queueCallback("createAnswer attempt on closed state"); return; } self.peerconnection.createAnswer( function(modifiedAnswer) { // change video direction, see https://github.com/jitsi/jitmeet/issues/41 if (self.pendingop !== null) { var sdp = new SDP(modifiedAnswer.sdp); if (sdp.media.length > 1) { switch(self.pendingop) { case 'mute': sdp.media[1] = sdp.media[1].replace('a=sendrecv', 'a=recvonly'); break; case 'unmute': sdp.media[1] = sdp.media[1].replace('a=recvonly', 'a=sendrecv'); break; } sdp.raw = sdp.session + sdp.media.join(''); modifiedAnswer.sdp = sdp.raw; } self.pendingop = null; } // FIXME: pushing down an answer while ice connection state // is still checking is bad... //logger.log(self.peerconnection.iceConnectionState); // trying to work around another chrome bug //modifiedAnswer.sdp = modifiedAnswer.sdp.replace(/a=setup:active/g, 'a=setup:actpass'); self.peerconnection.setLocalDescription(modifiedAnswer, function() { if(successCallback){ successCallback(); } queueCallback(); }, function(error) { logger.error('modified setLocalDescription failed', error); queueCallback(error); } ); }, function(error) { logger.error('modified answer failed', error); queueCallback(error); } ); }, function(error) { logger.error('modify failed', error); queueCallback(error); } ); }; /** * Adds stream. * @param stream new stream that will be added. * @param success_callback callback executed after successful stream addition. * @param ssrcInfo object with information about the SSRCs associated with the * stream. * @param dontModifySources {boolean} if true _modifySources won't be called. * Used for streams added before the call start. */ JingleSessionPC.prototype.addStream = function (stream, callback, ssrcInfo, dontModifySources) { // Remember SDP to figure out added/removed SSRCs var oldSdp = null; if(this.peerconnection) { if(this.peerconnection.localDescription) { oldSdp = new SDP(this.peerconnection.localDescription.sdp); } //when adding muted stream we have to pass the ssrcInfo but we don't //have a stream if(stream || ssrcInfo) this.peerconnection.addStream(stream, ssrcInfo); } // Conference is not active if(!oldSdp || !this.peerconnection || dontModifySources) { if(ssrcInfo) { //available only on video unmute or when adding muted stream this.modifiedSSRCs[ssrcInfo.type] = this.modifiedSSRCs[ssrcInfo.type] || []; this.modifiedSSRCs[ssrcInfo.type].push(ssrcInfo); } callback(); return; } this.modifyingLocalStreams = true; var self = this; this.modifySourcesQueue.push(function() { logger.log('modify sources done'); if(ssrcInfo) { //available only on video unmute or when adding muted stream self.modifiedSSRCs[ssrcInfo.type] = self.modifiedSSRCs[ssrcInfo.type] || []; self.modifiedSSRCs[ssrcInfo.type].push(ssrcInfo); } callback(); var newSdp = new SDP(self.peerconnection.localDescription.sdp); logger.log("SDPs", oldSdp, newSdp); self.notifyMySSRCUpdate(oldSdp, newSdp); }); } /** * Generate ssrc info object for a stream with the following properties: * - ssrcs - Array of the ssrcs associated with the stream. * - groups - Array of the groups associated with the stream. */ JingleSessionPC.prototype.generateNewStreamSSRCInfo = function () { return this.peerconnection.generateNewStreamSSRCInfo(); }; /** * Remove streams. * @param stream stream that will be removed. * @param success_callback callback executed after successful stream addition. * @param ssrcInfo object with information about the SSRCs associated with the * stream. */ JingleSessionPC.prototype.removeStream = function (stream, callback, ssrcInfo) { // Remember SDP to figure out added/removed SSRCs var oldSdp = null; if(this.peerconnection) { if(this.peerconnection.localDescription) { oldSdp = new SDP(this.peerconnection.localDescription.sdp); } if (RTCBrowserType.getBrowserType() === RTCBrowserType.RTC_BROWSER_FIREFOX) { if(!stream)//There is nothing to be changed return; var sender = null; // On Firefox we don't replace MediaStreams as this messes up the // m-lines (which can't be removed in Plan Unified) and brings a lot // of complications. Instead, we use the RTPSender and remove just // the track. var track = null; if(stream.getAudioTracks() && stream.getAudioTracks().length) { track = stream.getAudioTracks()[0]; } else if(stream.getVideoTracks() && stream.getVideoTracks().length) { track = stream.getVideoTracks()[0]; } if(!track) { logger.log("Cannot remove tracks: no tracks."); return; } // Find the right sender (for audio or video) this.peerconnection.peerconnection.getSenders().some(function (s) { if (s.track === track) { sender = s; return true; } }); if (sender) { this.peerconnection.peerconnection.removeTrack(sender); } else { logger.log("Cannot remove tracks: no RTPSender."); } } else if(stream) this.peerconnection.removeStream(stream, false, ssrcInfo); // else // NOTE: If there is no stream and the browser is not FF we still need to do // some transformation in order to send remove-source for the muted // streams. That's why we aren't calling return here. } // Conference is not active if(!oldSdp || !this.peerconnection) { callback(); return; } this.modifyingLocalStreams = true; var self = this; this.modifySourcesQueue.push(function() { logger.log('modify sources done'); callback(); var newSdp = new SDP(self.peerconnection.localDescription.sdp); if(ssrcInfo) { self.modifiedSSRCs[ssrcInfo.type] = self.modifiedSSRCs[ssrcInfo.type] || []; self.modifiedSSRCs[ssrcInfo.type].push(ssrcInfo); } logger.log("SDPs", oldSdp, newSdp); self.notifyMySSRCUpdate(oldSdp, newSdp); }); } /** * Figures out added/removed ssrcs and send update IQs. * @param old_sdp SDP object for old description. * @param new_sdp SDP object for new description. */ JingleSessionPC.prototype.notifyMySSRCUpdate = function (old_sdp, new_sdp) { if (!(this.peerconnection.signalingState == 'stable' && this.peerconnection.iceConnectionState == 'connected')){ logger.log("Too early to send updates"); return; } // send source-remove IQ. sdpDiffer = new SDPDiffer(new_sdp, old_sdp); var remove = $iq({to: this.peerjid, type: 'set'}) .c('jingle', { xmlns: 'urn:xmpp:jingle:1', action: 'source-remove', initiator: this.initiator, sid: this.sid } ); sdpDiffer.toJingle(remove); var removed = this.fixJingle(remove); if (removed && remove) { logger.info("Sending source-remove", remove.tree()); this.connection.sendIQ( remove, null, this.newJingleErrorHandler(remove), IQ_TIMEOUT); } else { logger.log('removal not necessary'); } // send source-add IQ. var sdpDiffer = new SDPDiffer(old_sdp, new_sdp); var add = $iq({to: this.peerjid, type: 'set'}) .c('jingle', { xmlns: 'urn:xmpp:jingle:1', action: 'source-add', initiator: this.initiator, sid: this.sid } ); sdpDiffer.toJingle(add); var added = this.fixJingle(add); if (added && add) { logger.info("Sending source-add", add.tree()); this.connection.sendIQ( add, null, this.newJingleErrorHandler(add), IQ_TIMEOUT); } else { logger.log('addition not necessary'); } }; /** * Method returns function(errorResponse) which is a callback to be passed to * Strophe connection.sendIQ method. An 'error' structure is created that is * passed as 1st argument to given failureCb. The format of this * structure is as follows: * { * code: {XMPP error response code} * reason: {the name of XMPP error reason element or 'timeout' if the request * has timed out within IQ_TIMEOUT milliseconds} * source: {request.tree() that provides original request} * session: {JingleSessionPC instance on which the error occurred} * } * @param request Strophe IQ instance which is the request to be dumped into * the error structure * @param failureCb function(error) called when error response was returned or * when a timeout has occurred. * @returns {function(this:JingleSessionPC)} */ JingleSessionPC.prototype.newJingleErrorHandler = function(request, failureCb) { return function (errResponse) { var error = { }; // Get XMPP error code and condition(reason) var errorElSel = $(errResponse).find('error'); if (errorElSel.length) { error.code = errorElSel.attr('code'); var errorReasonSel = $(errResponse).find('error :first'); if (errorReasonSel.length) error.reason = errorReasonSel[0].tagName; } if (!errResponse) { error.reason = 'timeout'; } error.source = null; if (request && "function" == typeof request.tree) { error.source = request.tree(); } error.session = this; logger.error("Jingle error", error); if (failureCb) { failureCb(error); } this.room.eventEmitter.emit(XMPPEvents.JINGLE_ERROR, error); }.bind(this); }; JingleSessionPC.onJingleFatalError = function (session, error) { this.room.eventEmitter.emit(XMPPEvents.CONFERENCE_SETUP_FAILED); this.room.eventEmitter.emit(XMPPEvents.JINGLE_FATAL_ERROR, session, error); }; /** * Called when new remote MediaStream is added to the PeerConnection. * @param stream the WebRTC MediaStream for remote participant */ JingleSessionPC.prototype.remoteStreamAdded = function (stream) { var self = this; if (!RTC.isUserStream(stream)) { logger.info( "Ignored remote 'stream added' event for non-user stream", stream); return; } // Bind 'addtrack'/'removetrack' event handlers if (RTCBrowserType.isChrome()) { stream.onaddtrack = function (event) { self.remoteTrackAdded(event.target, event.track); }; stream.onremovetrack = function (event) { self.remoteTrackRemoved(event.target, event.track); }; } // Call remoteTrackAdded for each track in the stream stream.getAudioTracks().forEach(function (track) { self.remoteTrackAdded(stream, track); }); stream.getVideoTracks().forEach(function (track) { self.remoteTrackAdded(stream, track); }); }; /** * Called on "track added" and "stream added" PeerConnection events(cause we * handle streams on per track basis). Does find the owner and the SSRC for * the track and passes that to ChatRoom for further processing. * @param stream WebRTC MediaStream instance which is the parent of the track * @param track the WebRTC MediaStreamTrack added for remote participant */ JingleSessionPC.prototype.remoteTrackAdded = function (stream, track) { logger.info("Remote track added", stream, track); var streamId = RTC.getStreamID(stream); var mediaType = track.kind; // This is our event structure which will be passed by the ChatRoom as // XMPPEvents.REMOTE_TRACK_ADDED data var jitsiTrackAddedEvent = { stream: stream, track: track, mediaType: track.kind, /* 'audio' or 'video' */ owner: undefined, /* to be determined below */ muted: null /* will be set in the ChatRoom */ }; // look up an associated JID for a stream id if (!mediaType) { logger.error("MediaType undefined", track); return; } var remoteSDP = new SDP(this.peerconnection.remoteDescription.sdp); var medialines = remoteSDP.media.filter(function (mediaLines){ return mediaLines.startsWith("m=" + mediaType); }); if (!medialines.length) { logger.error("No media for type " + mediaType + " found in remote SDP"); return; } var ssrclines = SDPUtil.find_lines(medialines[0], 'a=ssrc:'); ssrclines = ssrclines.filter(function (line) { if (RTCBrowserType.isTemasysPluginUsed()) { return ((line.indexOf('mslabel:' + streamId) !== -1)); } else { return ((line.indexOf('msid:' + streamId) !== -1)); } }); var thessrc; if (ssrclines.length) { thessrc = ssrclines[0].substring(7).split(' ')[0]; if (!this.ssrcOwners[thessrc]) { logger.error("No SSRC owner known for: " + thessrc); return; } jitsiTrackAddedEvent.owner = this.ssrcOwners[thessrc]; logger.log('associated jid', this.ssrcOwners[thessrc], thessrc); } else { logger.error("No SSRC lines for ", streamId); return; } jitsiTrackAddedEvent.ssrc = thessrc; this.room.remoteTrackAdded(jitsiTrackAddedEvent); }; /** * Handles remote stream removal. * @param stream the WebRTC MediaStream object which is being removed from the * PeerConnection */ JingleSessionPC.prototype.remoteStreamRemoved = function (stream) { var self = this; if (!RTC.isUserStream(stream)) { logger.info( "Ignored remote 'stream removed' event for non-user stream", stream); return; } // Call remoteTrackRemoved for each track in the stream stream.getVideoTracks().forEach(function(track){ self.remoteTrackRemoved(stream, track); }); stream.getAudioTracks().forEach(function(track) { self.remoteTrackRemoved(stream, track); }); }; /** * Handles remote media track removal. * @param stream WebRTC MediaStream instance which is the parent of the track * @param track the WebRTC MediaStreamTrack which has been removed from * the PeerConnection. */ JingleSessionPC.prototype.remoteTrackRemoved = function (stream, track) { logger.info("Remote track removed", stream, track); var streamId = RTC.getStreamID(stream); var trackId = track && track.id; if (!streamId) { logger.error("No stream ID for", stream); } else if (!trackId) { logger.error("No track ID for", track); } else { this.room.eventEmitter.emit( XMPPEvents.REMOTE_TRACK_REMOVED, streamId, trackId); } }; /** * Returns the ice connection state for the peer connection. * @returns the ice connection state for the peer connection. */ JingleSessionPC.prototype.getIceConnectionState = function () { return this.peerconnection.iceConnectionState; }; /** * Closes the peerconnection. */ JingleSessionPC.prototype.close = function () { this.closed = true; this.peerconnection && this.peerconnection.close(); }; /** * Fixes the outgoing jingle packets by removing the nodes related to the * muted/unmuted streams, handles removing of muted stream, etc. * @param jingle the jingle packet that is going to be sent * @returns {boolean} true if the jingle has to be sent and false otherwise. */ JingleSessionPC.prototype.fixJingle = function(jingle) { var action = $(jingle.nodeTree).find("jingle").attr("action"); switch (action) { case "source-add": case "session-accept": this.fixSourceAddJingle(jingle); break; case "source-remove": this.fixSourceRemoveJingle(jingle); break; default: logger.error("Unknown jingle action!"); return false; } var sources = $(jingle.tree()).find(">jingle>content>description>source"); return sources && sources.length > 0; }; /** * Fixes the outgoing jingle packets with action source-add by removing the * nodes related to the unmuted streams * @param jingle the jingle packet that is going to be sent * @returns {boolean} true if the jingle has to be sent and false otherwise. */ JingleSessionPC.prototype.fixSourceAddJingle = function (jingle) { var ssrcs = this.modifiedSSRCs["unmute"]; this.modifiedSSRCs["unmute"] = []; if(ssrcs && ssrcs.length) { ssrcs.forEach(function (ssrcObj) { var desc = $(jingle.tree()).find(">jingle>content[name=\"" + ssrcObj.mtype + "\"]>description"); if(!desc || !desc.length) return; ssrcObj.ssrc.ssrcs.forEach(function (ssrc) { var sourceNode = desc.find(">source[ssrc=\"" + ssrc + "\"]"); sourceNode.remove(); }); ssrcObj.ssrc.groups.forEach(function (group) { var groupNode = desc.find(">ssrc-group[semantics=\"" + group.group.semantics + "\"]:has(source[ssrc=\"" + group.primarySSRC + "\"])"); groupNode.remove(); }); }); } ssrcs = this.modifiedSSRCs["addMuted"]; this.modifiedSSRCs["addMuted"] = []; if(ssrcs && ssrcs.length) { ssrcs.forEach(function (ssrcObj) { var desc = createDescriptionNode(jingle, ssrcObj.mtype); var cname = Math.random().toString(36).substring(2); ssrcObj.ssrc.ssrcs.forEach(function (ssrc) { var sourceNode = desc.find(">source[ssrc=\"" +ssrc + "\"]"); sourceNode.remove(); var sourceXML = "" + "" + "" + ""; desc.append(sourceXML); }); ssrcObj.ssrc.groups.forEach(function (group) { var groupNode = desc.find(">ssrc-group[semantics=\"" + group.group.semantics + "\"]:has(source[ssrc=\"" + group.primarySSRC + "\"])"); groupNode.remove(); desc.append("" + ""); }); }); } }; /** * Fixes the outgoing jingle packets with action source-remove by removing the * nodes related to the muted streams, handles removing of muted stream * @param jingle the jingle packet that is going to be sent * @returns {boolean} true if the jingle has to be sent and false otherwise. */ JingleSessionPC.prototype.fixSourceRemoveJingle = function(jingle) { var ssrcs = this.modifiedSSRCs["mute"]; this.modifiedSSRCs["mute"] = []; if(ssrcs && ssrcs.length) ssrcs.forEach(function (ssrcObj) { ssrcObj.ssrc.ssrcs.forEach(function (ssrc) { var sourceNode = $(jingle.tree()).find(">jingle>content[name=\"" + ssrcObj.mtype + "\"]>description>source[ssrc=\"" + ssrc + "\"]"); sourceNode.remove(); }); ssrcObj.ssrc.groups.forEach(function (group) { var groupNode = $(jingle.tree()).find(">jingle>content[name=\"" + ssrcObj.mtype + "\"]>description>ssrc-group[semantics=\"" + group.group.semantics + "\"]:has(source[ssrc=\"" + group.primarySSRC + "\"])"); groupNode.remove(); }); }); ssrcs = this.modifiedSSRCs["remove"]; this.modifiedSSRCs["remove"] = []; if(ssrcs && ssrcs.length) ssrcs.forEach(function (ssrcObj) { var desc = createDescriptionNode(jingle, ssrcObj.mtype); ssrcObj.ssrc.ssrcs.forEach(function (ssrc) { var sourceNode = desc.find(">source[ssrc=\"" +ssrc + "\"]"); if(!sourceNode || !sourceNode.length) { //Maybe we have to include cname, msid, etc here? desc.append(""); } }); ssrcObj.ssrc.groups.forEach(function (group) { var groupNode = desc.find(">ssrc-group[semantics=\"" + group.group.semantics + "\"]:has(source[ssrc=\"" + group.primarySSRC + "\"])"); if(!groupNode || !groupNode.length) { desc.append("" + ""); } }); }); }; /** * Returns the description node related to the passed content type. If the node * doesn't exists it will be created. * @param jingle - the jingle packet * @param mtype - the content type(audio, video, etc.) */ function createDescriptionNode(jingle, mtype) { var content = $(jingle.tree()).find(">jingle>content[name=\"" + mtype + "\"]"); if(!content || !content.length) { $(jingle.tree()).find(">jingle").append( ""); content = $(jingle.tree()).find(">jingle>content[name=\"" + mtype + "\"]"); } var desc = content.find(">description"); if(!desc || !desc.length) { content.append(""); desc = content.find(">description"); } return desc; } module.exports = JingleSessionPC;