/* global __filename, RTCSessionDescription */
import { getLogger } from 'jitsi-meet-logger';
import transform from 'sdp-transform';
import * as GlobalOnErrorHandler from '../util/GlobalOnErrorHandler';
import JitsiRemoteTrack from './JitsiRemoteTrack';
import * as MediaType from '../../service/RTC/MediaType';
import LocalSdpMunger from './LocalSdpMunger';
import RTC from './RTC';
import RTCUtils from './RTCUtils';
import RTCBrowserType from './RTCBrowserType';
import RTCEvents from '../../service/RTC/RTCEvents';
import RtxModifier from '../xmpp/RtxModifier';
// FIXME SDP tools should end up in some kind of util module
import SDP from '../xmpp/SDP';
import SdpConsistency from '../xmpp/SdpConsistency';
import { SdpTransformWrap } from '../xmpp/SdpTransformUtil';
import SDPUtil from '../xmpp/SDPUtil';
import * as SignalingEvents from '../../service/RTC/SignalingEvents';
const logger = getLogger(__filename);
const SIMULCAST_LAYERS = 3;
/* eslint-disable max-params */
/**
* Creates new instance of 'TraceablePeerConnection'.
*
* @param {RTC} rtc the instance of RTC service
* @param {number} id the peer connection id assigned by the parent RTC module.
* @param {SignalingLayer} signalingLayer the signaling layer instance
* @param {object} iceConfig WebRTC 'PeerConnection' ICE config
* @param {object} constraints WebRTC 'PeerConnection' constraints
* @param {boolean} isP2P indicates whether or not the new instance will be used
* in a peer to peer connection
* @param {object} options TracablePeerConnection config options.
* @param {boolean} options.disableSimulcast if set to 'true' will disable
* the simulcast
* @param {boolean} options.disableRtx if set to 'true' will disable the RTX
* @param {boolean} options.preferH264 if set to 'true' H264 will be preferred
* over other video codecs.
*
* FIXME: initially the purpose of TraceablePeerConnection was to be able to
* debug the peer connection. Since many other responsibilities have been added
* it would make sense to extract a separate class from it and come up with
* a more suitable name.
*
* @constructor
*/
export default function TraceablePeerConnection(
rtc,
id,
signalingLayer,
iceConfig,
constraints,
isP2P,
options) {
/**
* Indicates whether or not this peer connection instance is actively
* sending/receiving media. When set to false the SDP media
* direction will be adjusted to 'inactive' in order to suspend media
* transmission.
* @type {boolean}
* @private
*/
this.mediaTransferActive = true;
/**
* The parent instance of RTC service which created this
* TracablePeerConnection.
* @type {RTC}
*/
this.rtc = rtc;
/**
* The peer connection identifier assigned by the RTC module.
* @type {number}
*/
this.id = id;
/**
* Indicates whether or not this instance is used in a peer to peer
* connection.
* @type {boolean}
*/
this.isP2P = isP2P;
// FIXME: We should support multiple streams per jid.
/**
* The map holds remote tracks associated with this peer connection.
* It maps user's JID to media type and remote track
* (one track per media type per user's JID).
* @type {Map>}
*/
this.remoteTracks = new Map();
/**
* A map which stores local tracks mapped by {@link JitsiLocalTrack.rtcId}
* @type {Map}
*/
this.localTracks = new Map();
/**
* @typedef {Object} TPCGroupInfo
* @property {string} semantics the SSRC groups semantics
* @property {Array} ssrcs group's SSRCs in order where the first
* one is group's primary SSRC, the second one is secondary (RTX) and so
* on...
*/
/**
* @typedef {Object} TPCSSRCInfo
* @property {Array} ssrcs an array which holds all track's SSRCs
* @property {Array} groups an array stores all track's SSRC
* groups
*/
/**
* Holds the info about local track's SSRCs mapped per their
* {@link JitsiLocalTrack.rtcId}
* @type {Map}
*/
this.localSSRCs = new Map();
/**
* The local ICE username fragment for this session.
*/
this.localUfrag = null;
/**
* The remote ICE username fragment for this session.
*/
this.remoteUfrag = null;
/**
* The signaling layer which operates this peer connection.
* @type {SignalingLayer}
*/
this.signalingLayer = signalingLayer;
// SignalingLayer listeners
this._peerVideoTypeChanged = this._peerVideoTypeChanged.bind(this);
this.signalingLayer.on(
SignalingEvents.PEER_VIDEO_TYPE_CHANGED,
this._peerVideoTypeChanged);
this._peerMutedChanged = this._peerMutedChanged.bind(this);
this.signalingLayer.on(
SignalingEvents.PEER_MUTED_CHANGED,
this._peerMutedChanged);
this.options = options;
this.peerconnection
= new RTCUtils.RTCPeerConnectionType(iceConfig, constraints);
this.updateLog = [];
this.stats = {};
this.statsinterval = null;
/**
* @type {number}
*/
this.maxstats = 0;
const Interop = require('sdp-interop').Interop;
this.interop = new Interop();
const Simulcast = require('sdp-simulcast');
this.simulcast = new Simulcast({ numOfLayers: SIMULCAST_LAYERS,
explodeRemoteSimulcast: false });
this.sdpConsistency = new SdpConsistency(this.toString());
/**
* Munges local SDP provided to the Jingle Session in order to prevent from
* sending SSRC updates on attach/detach and mute/unmute (for video).
* @type {LocalSdpMunger}
*/
this.localSdpMunger = new LocalSdpMunger(this);
/**
* TracablePeerConnection uses RTC's eventEmitter
* @type {EventEmitter}
*/
this.eventEmitter = rtc.eventEmitter;
this.rtxModifier = new RtxModifier();
// override as desired
this.trace = (what, info) => {
/* logger.warn('WTRACE', what, info);
if (info && RTCBrowserType.isIExplorer()) {
if (info.length > 1024) {
logger.warn('WTRACE', what, info.substr(1024));
}
if (info.length > 2048) {
logger.warn('WTRACE', what, info.substr(2048));
}
}*/
this.updateLog.push({
time: new Date(),
type: what,
value: info || ''
});
};
this.onicecandidate = null;
this.peerconnection.onicecandidate = event => {
// FIXME: this causes stack overflow with Temasys Plugin
if (!RTCBrowserType.isTemasysPluginUsed()) {
this.trace(
'onicecandidate',
JSON.stringify(event.candidate, null, ' '));
}
if (this.onicecandidate !== null) {
this.onicecandidate(event);
}
};
this.peerconnection.onaddstream
= event => this._remoteStreamAdded(event.stream);
this.peerconnection.onremovestream
= event => this._remoteStreamRemoved(event.stream);
this.onsignalingstatechange = null;
this.peerconnection.onsignalingstatechange = event => {
this.trace('onsignalingstatechange', this.signalingState);
if (this.onsignalingstatechange !== null) {
this.onsignalingstatechange(event);
}
};
this.oniceconnectionstatechange = null;
this.peerconnection.oniceconnectionstatechange = event => {
this.trace('oniceconnectionstatechange', this.iceConnectionState);
if (this.oniceconnectionstatechange !== null) {
this.oniceconnectionstatechange(event);
}
};
this.onnegotiationneeded = null;
this.peerconnection.onnegotiationneeded = event => {
this.trace('onnegotiationneeded');
if (this.onnegotiationneeded !== null) {
this.onnegotiationneeded(event);
}
};
this.ondatachannel = null;
this.peerconnection.ondatachannel = event => {
this.trace('ondatachannel', event);
if (this.ondatachannel !== null) {
this.ondatachannel(event);
}
};
// XXX: do all non-firefox browsers which we support also support this?
if (!RTCBrowserType.isFirefox() && this.maxstats) {
this.statsinterval = window.setInterval(() => {
this.peerconnection.getStats(stats => {
const results = stats.result();
const now = new Date();
for (let i = 0; i < results.length; ++i) {
results[i].names().forEach(name => {
// eslint-disable-next-line no-shadow
const id = `${results[i].id}-${name}`;
let s = this.stats[id];
if (!s) {
this.stats[id] = s = {
startTime: now,
endTime: now,
values: [],
times: []
};
}
s.values.push(results[i].stat(name));
s.times.push(now.getTime());
if (s.values.length > this.maxstats) {
s.values.shift();
s.times.shift();
}
s.endTime = now;
});
}
});
}, 1000);
}
logger.info(`Create new ${this}`);
}
/* eslint-enable max-params */
/**
* Returns a string representation of a SessionDescription object.
*/
const dumpSDP = function(description) {
if (typeof description === 'undefined' || description === null) {
return '';
}
return `type: ${description.type}\r\n${description.sdp}`;
};
/**
* Forwards the {@link peerconnection.iceConnectionState} state except that it
* will convert "completed" into "connected" where both mean that the ICE has
* succeeded and is up and running. We never see "completed" state for
* the JVB connection, but it started appearing for the P2P one. This method
* allows to adapt old logic to this new situation.
* @return {string}
*/
TraceablePeerConnection.prototype.getConnectionState = function() {
const state = this.peerconnection.iceConnectionState;
if (state === 'completed') {
return 'connected';
}
return state;
};
/**
* Obtains the media direction for given {@link MediaType}. The method takes
* into account whether or not there are any local tracks for media and
* the {@link mediaTransferActive} flag.
* @param {MediaType} mediaType
* @return {string} one of the SDP direction constants ('sendrecv, 'recvonly'
* etc.) which should be used when setting local description on the peer
* connection.
* @private
*/
TraceablePeerConnection.prototype._getDesiredMediaDirection
= function(mediaType) {
if (this.mediaTransferActive) {
return this.hasAnyTracksOfType(mediaType) ? 'sendrecv' : 'recvonly';
}
return 'inactive';
};
/**
* Tells whether or not this TPC instance is using Simulcast.
* @return {boolean} true if simulcast is enabled and active or
* false if it's turned off.
*/
TraceablePeerConnection.prototype.isSimulcastOn = function() {
return !this.options.disableSimulcast
&& RTCBrowserType.supportsSimulcast()
&& !this.isP2P;
};
/**
* Handles {@link SignalingEvents.PEER_VIDEO_TYPE_CHANGED}
* @param {string} endpointId the video owner's ID (MUC nickname)
* @param {VideoType} videoType the new value
* @private
*/
TraceablePeerConnection.prototype._peerVideoTypeChanged
= function(endpointId, videoType) {
// Check if endpointId has a value to avoid action on random track
if (!endpointId) {
logger.error(`No endpointID on peerVideoTypeChanged ${this}`);
return;
}
const videoTrack = this.getRemoteTracks(endpointId, MediaType.VIDEO);
if (videoTrack.length) {
// NOTE 1 track per media type is assumed
videoTrack[0]._setVideoType(videoType);
}
};
/**
* Handles remote track mute / unmute events.
* @param {string} endpointId the track owner's identifier (MUC nickname)
* @param {MediaType} mediaType "audio" or "video"
* @param {boolean} isMuted the new mute state
* @private
*/
TraceablePeerConnection.prototype._peerMutedChanged
= function(endpointId, mediaType, isMuted) {
// Check if endpointId is a value to avoid doing action on all remote tracks
if (!endpointId) {
logger.error('On peerMuteChanged - no endpoint ID');
return;
}
const track = this.getRemoteTracks(endpointId, mediaType);
if (track.length) {
// NOTE 1 track per media type is assumed
track[0].setMute(isMuted);
}
};
/**
* Obtains local tracks for given {@link MediaType}. If the mediaType
* argument is omitted the list of all local tracks will be returned.
* @param {MediaType} [mediaType]
* @return {Array}
*/
TraceablePeerConnection.prototype.getLocalTracks = function(mediaType) {
let tracks = Array.from(this.localTracks.values());
if (mediaType !== undefined) {
tracks = tracks.filter(track => track.getType() === mediaType);
}
return tracks;
};
/**
* Checks whether or not this {@link TraceablePeerConnection} instance contains
* any local tracks for given mediaType.
* @param {MediaType} mediaType
* @return {boolean}
*/
TraceablePeerConnection.prototype.hasAnyTracksOfType = function(mediaType) {
if (!mediaType) {
throw new Error('"mediaType" is required');
}
return this.getLocalTracks(mediaType).length > 0;
};
/**
* Obtains all remote tracks currently known to this PeerConnection instance.
* @param {string} [endpointId] the track owner's identifier (MUC nickname)
* @param {MediaType} [mediaType] the remote tracks will be filtered
* by their media type if this argument is specified.
* @return {Array}
*/
TraceablePeerConnection.prototype.getRemoteTracks
= function(endpointId, mediaType) {
const remoteTracks = [];
const endpoints
= endpointId ? [ endpointId ] : this.remoteTracks.keys();
for (const endpoint of endpoints) {
const endpointTrackMap = this.remoteTracks.get(endpoint);
if (!endpointTrackMap) {
// Otherwise an empty Map() would have to be allocated above
// eslint-disable-next-line no-continue
continue;
}
for (const trackMediaType of endpointTrackMap.keys()) {
// per media type filtering
if (!mediaType || mediaType === trackMediaType) {
const mediaTrack = endpointTrackMap.get(trackMediaType);
if (mediaTrack) {
remoteTracks.push(mediaTrack);
}
}
}
}
return remoteTracks;
};
/**
* Tries to find {@link JitsiTrack} for given SSRC number. It will search both
* local and remote tracks bound to this instance.
* @param {number} ssrc
* @return {JitsiTrack|null}
*/
TraceablePeerConnection.prototype.getTrackBySSRC = function(ssrc) {
if (typeof ssrc !== 'number') {
throw new Error(`SSRC ${ssrc} is not a number`);
}
for (const localTrack of this.localTracks.values()) {
if (this.getLocalSSRC(localTrack) === ssrc) {
return localTrack;
}
}
for (const remoteTrack of this.getRemoteTracks()) {
if (remoteTrack.getSSRC() === ssrc) {
return remoteTrack;
}
}
return null;
};
/**
* Called when new remote MediaStream is added to the PeerConnection.
* @param {MediaStream} stream the WebRTC MediaStream for remote participant
*/
TraceablePeerConnection.prototype._remoteStreamAdded = function(stream) {
const streamId = RTC.getStreamID(stream);
if (!RTC.isUserStreamById(streamId)) {
logger.info(
`${this} ignored remote 'stream added' event for non-user stream`
+ `id: ${streamId}`);
return;
}
// Bind 'addtrack'/'removetrack' event handlers
if (RTCBrowserType.isChrome() || RTCBrowserType.isNWJS()
|| RTCBrowserType.isElectron()) {
stream.onaddtrack = event => {
this._remoteTrackAdded(event.target, event.track);
};
stream.onremovetrack = event => {
this._remoteTrackRemoved(event.target, event.track);
};
}
// Call remoteTrackAdded for each track in the stream
const streamAudioTracks = stream.getAudioTracks();
for (const audioTrack of streamAudioTracks) {
this._remoteTrackAdded(stream, audioTrack);
}
const streamVideoTracks = stream.getVideoTracks();
for (const videoTrack of streamVideoTracks) {
this._remoteTrackAdded(stream, videoTrack);
}
};
/**
* Called on "track added" and "stream added" PeerConnection events (because we
* handle streams on per track basis). Finds the owner and the SSRC for
* the track and passes that to ChatRoom for further processing.
* @param {MediaStream} stream the WebRTC MediaStream instance which is
* the parent of the track
* @param {MediaStreamTrack} track the WebRTC MediaStreamTrack added for remote
* participant
*/
TraceablePeerConnection.prototype._remoteTrackAdded = function(stream, track) {
const streamId = RTC.getStreamID(stream);
const mediaType = track.kind;
logger.info(`${this} remote track added:`, streamId, mediaType);
// look up an associated JID for a stream id
if (!mediaType) {
GlobalOnErrorHandler.callErrorHandler(
new Error(
`MediaType undefined for remote track, stream id: ${streamId}`
));
// Abort
return;
}
const remoteSDP = new SDP(this.remoteDescription.sdp);
const mediaLines
= remoteSDP.media.filter(mls => mls.startsWith(`m=${mediaType}`));
if (!mediaLines.length) {
GlobalOnErrorHandler.callErrorHandler(
new Error(
`No media lines for type ${mediaType
} found in remote SDP for remote track: ${streamId}`));
// Abort
return;
}
let ssrcLines = SDPUtil.findLines(mediaLines[0], 'a=ssrc:');
ssrcLines = ssrcLines.filter(
line => {
const msid
= RTCBrowserType.isTemasysPluginUsed() ? 'mslabel' : 'msid';
return line.indexOf(`${msid}:${streamId}`) !== -1;
});
if (!ssrcLines.length) {
GlobalOnErrorHandler.callErrorHandler(
new Error(
`No SSRC lines for streamId ${streamId
} for remote track, media type: ${mediaType}`));
// Abort
return;
}
// FIXME the length of ssrcLines[0] not verified, but it will fail
// with global error handler anyway
const ssrcStr = ssrcLines[0].substring(7).split(' ')[0];
const trackSsrc = Number(ssrcStr);
const ownerEndpointId = this.signalingLayer.getSSRCOwner(trackSsrc);
if (isNaN(trackSsrc) || trackSsrc < 0) {
GlobalOnErrorHandler.callErrorHandler(
new Error(
`Invalid SSRC: ${ssrcStr
} for remote track, msid: ${streamId
} media type: ${mediaType}`));
// Abort
return;
} else if (!ownerEndpointId) {
GlobalOnErrorHandler.callErrorHandler(
new Error(
`No SSRC owner known for: ${trackSsrc
} for remote track, msid: ${streamId
} media type: ${mediaType}`));
// Abort
return;
}
logger.log(`${this} associated ssrc`, ownerEndpointId, trackSsrc);
const peerMediaInfo
= this.signalingLayer.getPeerMediaInfo(ownerEndpointId, mediaType);
if (!peerMediaInfo) {
GlobalOnErrorHandler.callErrorHandler(
new Error(
`${this}: no peer media info available for ${
ownerEndpointId}`));
return;
}
const muted = peerMediaInfo.muted;
const videoType = peerMediaInfo.videoType; // can be undefined
this._createRemoteTrack(
ownerEndpointId, stream, track, mediaType, videoType, trackSsrc, muted);
};
// FIXME cleanup params
/* eslint-disable max-params */
/**
* Initializes a new JitsiRemoteTrack instance with the data provided by
* the signaling layer and SDP.
*
* @param {string} ownerEndpointId the owner's endpoint ID (MUC nickname)
* @param {MediaStream} stream the WebRTC stream instance
* @param {MediaStreamTrack} track the WebRTC track instance
* @param {MediaType} mediaType the track's type of the media
* @param {VideoType} [videoType] the track's type of the video (if applicable)
* @param {number} ssrc the track's main SSRC number
* @param {boolean} muted the initial muted status
*/
TraceablePeerConnection.prototype._createRemoteTrack
= function(ownerEndpointId, stream, track, mediaType, videoType, ssrc, muted) {
const remoteTrack
= new JitsiRemoteTrack(
this.rtc, this.rtc.conference,
ownerEndpointId,
stream, track, mediaType, videoType, ssrc, muted, this.isP2P);
let remoteTracksMap = this.remoteTracks.get(ownerEndpointId);
if (!remoteTracksMap) {
remoteTracksMap = new Map();
this.remoteTracks.set(ownerEndpointId, remoteTracksMap);
}
if (remoteTracksMap.has(mediaType)) {
logger.error(
`${this} overwriting remote track! ${remoteTrack}`,
ownerEndpointId, mediaType);
}
remoteTracksMap.set(mediaType, remoteTrack);
this.eventEmitter.emit(RTCEvents.REMOTE_TRACK_ADDED, remoteTrack);
};
/* eslint-enable max-params */
/**
* Handles remote stream removal.
* @param stream the WebRTC MediaStream object which is being removed from the
* PeerConnection
*/
TraceablePeerConnection.prototype._remoteStreamRemoved = function(stream) {
if (!RTC.isUserStream(stream)) {
const id = RTC.getStreamID(stream);
logger.info(
`Ignored remote 'stream removed' event for non-user stream ${id}`);
return;
}
// Call remoteTrackRemoved for each track in the stream
const streamVideoTracks = stream.getVideoTracks();
for (const videoTrack of streamVideoTracks) {
this._remoteTrackRemoved(stream, videoTrack);
}
const streamAudioTracks = stream.getAudioTracks();
for (const audioTrack of streamAudioTracks) {
this._remoteTrackRemoved(stream, audioTrack);
}
};
/**
* Handles remote media track removal.
* @param {MediaStream} stream WebRTC MediaStream instance which is the parent
* of the track.
* @param {MediaStreamTrack} track the WebRTC MediaStreamTrack which has been
* removed from the PeerConnection.
*/
TraceablePeerConnection.prototype._remoteTrackRemoved
= function(stream, track) {
const streamId = RTC.getStreamID(stream);
const trackId = track && track.id;
logger.info(`${this} - remote track removed: ${streamId}, ${trackId}`);
if (!streamId) {
GlobalOnErrorHandler.callErrorHandler(
new Error(`${this} remote track removal failed - no stream ID`));
return;
}
if (!trackId) {
GlobalOnErrorHandler.callErrorHandler(
new Error(`${this} remote track removal failed - no track ID`));
return;
}
if (!this._removeRemoteTrackById(streamId, trackId)) {
// NOTE this warning is always printed when user leaves the room,
// because we remove remote tracks manually on MUC member left event,
// before the SSRCs are removed by Jicofo. In most cases it is fine to
// ignore this warning, but still it's better to keep it printed for
// debugging purposes.
//
// We could change the behaviour to emit track removed only from here,
// but the order of the events will change and consuming apps could
// behave unexpectedly (the "user left" event would come before "track
// removed" events).
logger.warn(
`${this} Removed track not found for msid: ${streamId},
track id: ${trackId}`);
}
};
/**
* Finds remote track by it's stream and track ids.
* @param {string} streamId the media stream id as defined by the WebRTC
* @param {string} trackId the media track id as defined by the WebRTC
* @return {JitsiRemoteTrack|undefined} the track's instance or
* undefined if not found.
* @private
*/
TraceablePeerConnection.prototype._getRemoteTrackById
= function(streamId, trackId) {
// .find will break the loop once the first match is found
for (const endpointTrackMap of this.remoteTracks.values()) {
for (const mediaTrack of endpointTrackMap.values()) {
// FIXME verify and try to use ===
/* eslint-disable eqeqeq */
if (mediaTrack.getStreamId() == streamId
&& mediaTrack.getTrackId() == trackId) {
return mediaTrack;
}
/* eslint-enable eqeqeq */
}
}
return undefined;
};
/**
* Removes all JitsiRemoteTracks associated with given MUC nickname
* (resource part of the JID). Returns array of removed tracks.
*
* @param {string} owner - The resource part of the MUC JID.
* @returns {JitsiRemoteTrack[]}
*/
TraceablePeerConnection.prototype.removeRemoteTracks = function(owner) {
const removedTracks = [];
const remoteTracksMap = this.remoteTracks.get(owner);
if (remoteTracksMap) {
const removedAudioTrack = remoteTracksMap.get(MediaType.AUDIO);
const removedVideoTrack = remoteTracksMap.get(MediaType.VIDEO);
removedAudioTrack && removedTracks.push(removedAudioTrack);
removedVideoTrack && removedTracks.push(removedVideoTrack);
this.remoteTracks.delete(owner);
}
logger.debug(
`${this} removed remote tracks for ${owner
} count: ${removedTracks.length}`);
return removedTracks;
};
/**
* Removes and disposes given JitsiRemoteTrack instance. Emits
* {@link RTCEvents.REMOTE_TRACK_REMOVED}.
* @param {JitsiRemoteTrack} toBeRemoved
*/
TraceablePeerConnection.prototype._removeRemoteTrack = function(toBeRemoved) {
toBeRemoved.dispose();
const participantId = toBeRemoved.getParticipantId();
const remoteTracksMap = this.remoteTracks.get(participantId);
if (!remoteTracksMap) {
logger.error(
`removeRemoteTrack: no remote tracks map for ${participantId}`);
} else if (!remoteTracksMap.delete(toBeRemoved.getType())) {
logger.error(
`Failed to remove ${toBeRemoved} - type mapping messed up ?`);
}
this.eventEmitter.emit(RTCEvents.REMOTE_TRACK_REMOVED, toBeRemoved);
};
/**
* Removes and disposes JitsiRemoteTrack identified by given stream and
* track ids.
*
* @param {string} streamId the media stream id as defined by the WebRTC
* @param {string} trackId the media track id as defined by the WebRTC
* @returns {JitsiRemoteTrack|undefined} the track which has been removed or
* undefined if no track matching given stream and track ids was
* found.
*/
TraceablePeerConnection.prototype._removeRemoteTrackById
= function(streamId, trackId) {
const toBeRemoved = this._getRemoteTrackById(streamId, trackId);
if (toBeRemoved) {
this._removeRemoteTrack(toBeRemoved);
}
return toBeRemoved;
};
/**
* @typedef {Object} SSRCGroupInfo
* @property {Array} ssrcs group's SSRCs
* @property {string} semantics
*/
/**
* @typedef {Object} TrackSSRCInfo
* @property {Array} ssrcs track's SSRCs
* @property {Array} groups track's SSRC groups
*/
/**
* Returns map with keys msid and TrackSSRCInfo values.
* @param {Object} desc the WebRTC SDP instance.
* @return {Map}
*/
function extractSSRCMap(desc) {
/**
* Track SSRC infos mapped by stream ID (msid)
* @type {Map}
*/
const ssrcMap = new Map();
/**
* Groups mapped by primary SSRC number
* @type {Map>}
*/
const groupsMap = new Map();
if (typeof desc !== 'object' || desc === null
|| typeof desc.sdp !== 'string') {
logger.warn('An empty description was passed as an argument.');
return ssrcMap;
}
const session = transform.parse(desc.sdp);
if (!Array.isArray(session.media)) {
return ssrcMap;
}
for (const mLine of session.media) {
if (!Array.isArray(mLine.ssrcs)) {
continue; // eslint-disable-line no-continue
}
if (Array.isArray(mLine.ssrcGroups)) {
for (const group of mLine.ssrcGroups) {
if (typeof group.semantics !== 'undefined'
&& typeof group.ssrcs !== 'undefined') {
// Parse SSRCs and store as numbers
const groupSSRCs
= group.ssrcs.split(' ')
.map(ssrcStr => parseInt(ssrcStr, 10));
const primarySSRC = groupSSRCs[0];
// Note that group.semantics is already present
group.ssrcs = groupSSRCs;
// eslint-disable-next-line max-depth
if (!groupsMap.has(primarySSRC)) {
groupsMap.set(primarySSRC, []);
}
groupsMap.get(primarySSRC).push(group);
}
}
}
for (const ssrc of mLine.ssrcs) {
if (ssrc.attribute !== 'msid') {
continue; // eslint-disable-line no-continue
}
const msid = ssrc.value;
let ssrcInfo = ssrcMap.get(msid);
if (!ssrcInfo) {
ssrcInfo = {
ssrcs: [],
groups: [],
msid
};
ssrcMap.set(msid, ssrcInfo);
}
const ssrcNumber = ssrc.id;
ssrcInfo.ssrcs.push(ssrcNumber);
if (groupsMap.has(ssrcNumber)) {
const ssrcGroups = groupsMap.get(ssrcNumber);
for (const group of ssrcGroups) {
ssrcInfo.groups.push(group);
}
}
}
}
return ssrcMap;
}
/**
* Takes a SessionDescription object and returns a "normalized" version.
* Currently it only takes care of ordering the a=ssrc lines.
*/
const normalizePlanB = function(desc) {
if (typeof desc !== 'object' || desc === null
|| typeof desc.sdp !== 'string') {
logger.warn('An empty description was passed as an argument.');
return desc;
}
// eslint-disable-next-line no-shadow
const transform = require('sdp-transform');
const session = transform.parse(desc.sdp);
if (typeof session !== 'undefined'
&& typeof session.media !== 'undefined'
&& Array.isArray(session.media)) {
session.media.forEach(mLine => {
// Chrome appears to be picky about the order in which a=ssrc lines
// are listed in an m-line when rtx is enabled (and thus there are
// a=ssrc-group lines with FID semantics). Specifically if we have
// "a=ssrc-group:FID S1 S2" and the "a=ssrc:S2" lines appear before
// the "a=ssrc:S1" lines, SRD fails.
// So, put SSRC which appear as the first SSRC in an FID ssrc-group
// first.
const firstSsrcs = [];
const newSsrcLines = [];
if (typeof mLine.ssrcGroups !== 'undefined'
&& Array.isArray(mLine.ssrcGroups)) {
mLine.ssrcGroups.forEach(group => {
if (typeof group.semantics !== 'undefined'
&& group.semantics === 'FID') {
if (typeof group.ssrcs !== 'undefined') {
firstSsrcs.push(Number(group.ssrcs.split(' ')[0]));
}
}
});
}
if (Array.isArray(mLine.ssrcs)) {
let i;
for (i = 0; i < mLine.ssrcs.length; i++) {
if (typeof mLine.ssrcs[i] === 'object'
&& typeof mLine.ssrcs[i].id !== 'undefined'
&& firstSsrcs.indexOf(mLine.ssrcs[i].id) >= 0) {
newSsrcLines.push(mLine.ssrcs[i]);
delete mLine.ssrcs[i];
}
}
for (i = 0; i < mLine.ssrcs.length; i++) {
if (typeof mLine.ssrcs[i] !== 'undefined') {
newSsrcLines.push(mLine.ssrcs[i]);
}
}
mLine.ssrcs = newSsrcLines;
}
});
}
const resStr = transform.write(session);
return new RTCSessionDescription({
type: desc.type,
sdp: resStr
});
};
/**
*
* @param {JitsiLocalTrack} localTrack
*/
TraceablePeerConnection.prototype.getLocalSSRC = function(localTrack) {
const ssrcInfo = this._getSSRC(localTrack.rtcId);
return ssrcInfo && ssrcInfo.ssrcs[0];
};
/* eslint-disable-next-line vars-on-top */
const getters = {
signalingState() {
return this.peerconnection.signalingState;
},
iceConnectionState() {
return this.peerconnection.iceConnectionState;
},
localDescription() {
let desc = this.peerconnection.localDescription;
this.trace('getLocalDescription::preTransform', dumpSDP(desc));
// if we're running on FF, transform to Plan B first.
if (RTCBrowserType.usesUnifiedPlan()) {
desc = this.interop.toPlanB(desc);
this.trace('getLocalDescription::postTransform (Plan B)',
dumpSDP(desc));
}
if (RTCBrowserType.doesVideoMuteByStreamRemove()) {
this.localSdpMunger.maybeMungeLocalSdp(desc);
logger.debug(
'getLocalDescription::postTransform (munge local SDP)', desc);
}
return desc || {};
},
remoteDescription() {
let desc = this.peerconnection.remoteDescription;
this.trace('getRemoteDescription::preTransform', dumpSDP(desc));
// if we're running on FF, transform to Plan B first.
if (RTCBrowserType.usesUnifiedPlan()) {
desc = this.interop.toPlanB(desc);
this.trace(
'getRemoteDescription::postTransform (Plan B)', dumpSDP(desc));
}
return desc || {};
}
};
Object.keys(getters).forEach(prop => {
Object.defineProperty(
TraceablePeerConnection.prototype,
prop, {
get: getters[prop]
}
);
});
TraceablePeerConnection.prototype._getSSRC = function(rtcId) {
return this.localSSRCs.get(rtcId);
};
/**
* Add {@link JitsiLocalTrack} to this TPC.
* @param {JitsiLocalTrack} track
*/
TraceablePeerConnection.prototype.addTrack = function(track) {
const rtcId = track.rtcId;
logger.info(`add ${track} to: ${this}`);
if (this.localTracks.has(rtcId)) {
logger.error(`${track} is already in ${this}`);
return;
}
this.localTracks.set(rtcId, track);
const webrtcStream = track.getOriginalStream();
if (webrtcStream) {
this._addStream(webrtcStream);
// It's not ok for a track to not have a WebRTC stream if:
} else if (!RTCBrowserType.doesVideoMuteByStreamRemove()
|| track.isAudioTrack()
|| (track.isVideoTrack() && !track.isMuted())) {
logger.error(`${this} no WebRTC stream for: ${track}`);
}
// Muted video tracks do not have WebRTC stream
if (RTCBrowserType.doesVideoMuteByStreamRemove()
&& track.isVideoTrack() && track.isMuted()) {
const ssrcInfo = this.generateNewStreamSSRCInfo(track);
this.sdpConsistency.setPrimarySsrc(ssrcInfo.ssrcs[0]);
const simGroup
= ssrcInfo.groups.find(groupInfo => groupInfo.semantics === 'SIM');
if (simGroup) {
this.simulcast.setSsrcCache(simGroup.ssrcs);
}
const fidGroups
= ssrcInfo.groups.filter(
groupInfo => groupInfo.semantics === 'FID');
if (fidGroups) {
const rtxSsrcMapping = new Map();
fidGroups.forEach(fidGroup => {
const primarySsrc = fidGroup.ssrcs[0];
const rtxSsrc = fidGroup.ssrcs[1];
rtxSsrcMapping.set(primarySsrc, rtxSsrc);
});
this.rtxModifier.setSsrcCache(rtxSsrcMapping);
}
}
};
/**
* Adds local track as part of the unmute operation.
* @param {JitsiLocalTrack} track the track to be added as part of the unmute
* operation
* @return {boolean} true if the state of underlying PC has changed and
* the renegotiation is required or false otherwise.
*/
TraceablePeerConnection.prototype.addTrackUnmute = function(track) {
if (!this._assertTrackBelongs('addTrackUnmute', track)) {
// Abort
return false;
}
logger.info(`Adding ${track} as unmute to ${this}`);
const webRtcStream = track.getOriginalStream();
if (!webRtcStream) {
logger.error(
`Unable to add ${track} as unmute to ${this} - no WebRTC stream`);
return false;
}
this._addStream(webRtcStream);
return true;
};
/**
* Adds WebRTC media stream to the underlying PeerConnection
* @param {MediaStream} mediaStream
* @private
*/
TraceablePeerConnection.prototype._addStream = function(mediaStream) {
this.peerconnection.addStream(mediaStream);
};
/**
* Removes WebRTC media stream from the underlying PeerConection
* @param {MediaStream} mediaStream
*/
TraceablePeerConnection.prototype._removeStream = function(mediaStream) {
if (RTCBrowserType.isFirefox()) {
this._handleFirefoxRemoveStream(mediaStream);
} else {
this.peerconnection.removeStream(mediaStream);
}
};
/**
* This method when called will check if given localTrack belongs to
* this TPC (that it has been previously added using {@link addTrack}). If the
* track does not belong an error message will be logged.
* @param {string} methodName the method name that will be logged in an error
* message
* @param {JitsiLocalTrack} localTrack
* @return {boolean} true if given local track belongs to this TPC or
* false otherwise.
* @private
*/
TraceablePeerConnection.prototype._assertTrackBelongs
= function(methodName, localTrack) {
const doesBelong = this.localTracks.has(localTrack.rtcId);
if (!doesBelong) {
logger.error(
`${methodName}: ${localTrack} does not belong to ${this}`);
}
return doesBelong;
};
/**
* Remove local track from this TPC.
* @param {JitsiLocalTrack} localTrack the track to be removed from this TPC.
*
* FIXME It should probably remove a boolean just like {@link removeTrackMute}
* The same applies to addTrack.
*/
TraceablePeerConnection.prototype.removeTrack = function(localTrack) {
const webRtcStream = localTrack.getOriginalStream();
this.trace(
'removeStream',
localTrack.rtcId, webRtcStream ? webRtcStream.id : undefined);
if (!this._assertTrackBelongs('removeStream', localTrack)) {
// Abort - nothing to be done here
return;
}
this.localTracks.delete(localTrack.rtcId);
this.localSSRCs.delete(localTrack.rtcId);
if (webRtcStream) {
if (RTCBrowserType.isFirefox()) {
this._handleFirefoxRemoveStream(webRtcStream);
} else {
this.peerconnection.removeStream(webRtcStream);
}
}
};
/**
* Removes local track as part of the mute operation.
* @param {JitsiLocalTrack} localTrack the local track to be remove as part of
* the mute operation.
* @return {boolean} true if the underlying PeerConnection's state has
* changed and the renegotiation is required or false otherwise.
*/
TraceablePeerConnection.prototype.removeTrackMute = function(localTrack) {
const webRtcStream = localTrack.getOriginalStream();
this.trace(
'removeStreamMute',
localTrack.rtcId, webRtcStream ? webRtcStream.id : null);
if (!this._assertTrackBelongs('removeStreamMute', localTrack)) {
// Abort - nothing to be done here
return false;
}
if (webRtcStream) {
logger.info(
`Removing ${localTrack} as mute from ${this}`);
this._removeStream(webRtcStream);
return true;
}
logger.error(`removeStreamMute - no WebRTC stream for ${localTrack}`);
return false;
};
/**
* Remove stream handling for firefox
* @param stream: webrtc media stream
*/
TraceablePeerConnection.prototype._handleFirefoxRemoveStream
= function(stream) {
if (!stream) {
// There is nothing to be changed
return;
}
let 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.
let 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.error('Cannot remove tracks: no tracks.');
return;
}
// Find the right sender (for audio or video)
this.peerconnection.getSenders().some(s => {
if (s.track === track) {
sender = s;
return true;
}
return false;
});
if (sender) {
this.peerconnection.removeTrack(sender);
} else {
logger.log('Cannot remove tracks: no RTPSender.');
}
};
TraceablePeerConnection.prototype.createDataChannel = function(label, opts) {
this.trace('createDataChannel', label, opts);
return this.peerconnection.createDataChannel(label, opts);
};
/**
* Ensures that the simulcast ssrc-group appears after any other ssrc-groups
* in the SDP so that simulcast is properly activated.
*
* @param {Object} localSdp the WebRTC session description instance for
* the local description.
* @private
*/
TraceablePeerConnection.prototype._ensureSimulcastGroupIsLast
= function(localSdp) {
let sdpStr = localSdp.sdp;
const videoStartIndex = sdpStr.indexOf('m=video');
const simStartIndex = sdpStr.indexOf('a=ssrc-group:SIM', videoStartIndex);
let otherStartIndex = sdpStr.lastIndexOf('a=ssrc-group');
if (simStartIndex === -1
|| otherStartIndex === -1
|| otherStartIndex === simStartIndex) {
return;
}
const simEndIndex = sdpStr.indexOf('\r\n', simStartIndex);
const simStr = sdpStr.substring(simStartIndex, simEndIndex + 2);
sdpStr = sdpStr.replace(simStr, '');
otherStartIndex = sdpStr.lastIndexOf('a=ssrc-group');
const otherEndIndex = sdpStr.indexOf('\r\n', otherStartIndex);
const sdpHead = sdpStr.slice(0, otherEndIndex);
const simStrTrimmed = simStr.trim();
const sdpTail = sdpStr.slice(otherEndIndex);
sdpStr = `${sdpHead}\r\n${simStrTrimmed}${sdpTail}`;
localSdp.sdp = sdpStr;
};
/**
* Will adjust audio and video media direction in the given SDP object to
* reflect the current status of the {@link mediaTransferActive} flag.
* @param {Object} localDescription the WebRTC session description instance for
* the local description.
* @private
*/
TraceablePeerConnection.prototype._adjustLocalMediaDirection
= function(localDescription) {
const transformer = new SdpTransformWrap(localDescription.sdp);
let modifiedDirection = false;
const audioMedia = transformer.selectMedia('audio');
if (audioMedia) {
const desiredAudioDirection
= this._getDesiredMediaDirection(MediaType.AUDIO);
if (audioMedia.direction !== desiredAudioDirection) {
audioMedia.direction = desiredAudioDirection;
logger.info(
`Adjusted local audio direction to ${desiredAudioDirection}`);
modifiedDirection = true;
}
} else {
logger.warn('No "audio" media found int the local description');
}
const videoMedia = transformer.selectMedia('video');
if (videoMedia) {
const desiredVideoDirection
= this._getDesiredMediaDirection(MediaType.VIDEO);
if (videoMedia.direction !== desiredVideoDirection) {
videoMedia.direction = desiredVideoDirection;
logger.info(
`Adjusted local video direction to ${desiredVideoDirection}`);
modifiedDirection = true;
}
} else {
logger.warn('No "video" media found in the local description');
}
if (modifiedDirection) {
localDescription.sdp = transformer.toRawSDP();
}
};
TraceablePeerConnection.prototype.setLocalDescription
= function(description, successCallback, failureCallback) {
let localSdp = description;
this.trace('setLocalDescription::preTransform', dumpSDP(localSdp));
this._adjustLocalMediaDirection(localSdp);
this._ensureSimulcastGroupIsLast(localSdp);
// if we're using unified plan, transform to it first.
if (RTCBrowserType.usesUnifiedPlan()) {
localSdp = this.interop.toUnifiedPlan(localSdp);
this.trace(
'setLocalDescription::postTransform (Unified Plan)',
dumpSDP(localSdp));
}
this.peerconnection.setLocalDescription(localSdp,
() => {
this.trace('setLocalDescriptionOnSuccess');
const localUfrag = SDPUtil.getUfrag(localSdp.sdp);
if (localUfrag !== this.localUfrag) {
this.localUfrag = localUfrag;
this.eventEmitter.emit(
RTCEvents.LOCAL_UFRAG_CHANGED, this, localUfrag);
}
successCallback();
},
err => {
this.trace('setLocalDescriptionOnFailure', err);
this.eventEmitter.emit(
RTCEvents.SET_LOCAL_DESCRIPTION_FAILED,
err, this);
failureCallback(err);
}
);
};
/**
* Enables/disables media transmission on this peer connection. When disabled
* the SDP media direction in the local SDP will be adjusted to 'inactive' which
* means that no data will be received or sent, but the connection should be
* kept alive.
* @param {boolean} active true to enable the media transmission or
* false to disable.
* @public
*/
TraceablePeerConnection.prototype.setMediaTransferActive = function(active) {
this.mediaTransferActive = active;
};
TraceablePeerConnection.prototype.setRemoteDescription
= function(description, successCallback, failureCallback) {
this.trace('setRemoteDescription::preTransform', dumpSDP(description));
// TODO the focus should squeze or explode the remote simulcast
// eslint-disable-next-line no-param-reassign
description = this.simulcast.mungeRemoteDescription(description);
this.trace(
'setRemoteDescription::postTransform (simulcast)',
dumpSDP(description));
if (this.options.preferH264) {
const parsedSdp = transform.parse(description.sdp);
const videoMLine = parsedSdp.media.find(m => m.type === 'video');
SDPUtil.preferVideoCodec(videoMLine, 'h264');
description.sdp = transform.write(parsedSdp);
}
// If the browser uses unified plan, transform to it first
if (RTCBrowserType.usesUnifiedPlan()) {
description.sdp = this.rtxModifier.stripRtx(description.sdp);
this.trace(
'setRemoteDescription::postTransform (stripRtx)',
dumpSDP(description));
// eslint-disable-next-line no-param-reassign
description = this.interop.toUnifiedPlan(description);
this.trace(
'setRemoteDescription::postTransform (Plan A)',
dumpSDP(description));
} else {
// Plan B
// eslint-disable-next-line no-param-reassign
description = normalizePlanB(description);
}
this.peerconnection.setRemoteDescription(
description,
() => {
this.trace('setRemoteDescriptionOnSuccess');
const remoteUfrag = SDPUtil.getUfrag(description.sdp);
if (remoteUfrag !== this.remoteUfrag) {
this.remoteUfrag = remoteUfrag;
this.eventEmitter.emit(
RTCEvents.REMOTE_UFRAG_CHANGED, this, remoteUfrag);
}
successCallback();
},
err => {
this.trace('setRemoteDescriptionOnFailure', err);
this.eventEmitter.emit(
RTCEvents.SET_REMOTE_DESCRIPTION_FAILED,
err,
this);
failureCallback(err);
});
};
/**
* Makes the underlying TraceablePeerConnection generate new SSRC for
* the recvonly video stream.
*/
TraceablePeerConnection.prototype.generateRecvonlySsrc = function() {
const newSSRC = SDPUtil.generateSsrc();
logger.info(`${this} generated new recvonly SSRC: ${newSSRC}`);
this.sdpConsistency.setPrimarySsrc(newSSRC);
};
/**
* Makes the underlying TraceablePeerConnection forget the current primary video
* SSRC.
*/
TraceablePeerConnection.prototype.clearRecvonlySsrc = function() {
logger.info('Clearing primary video SSRC!');
this.sdpConsistency.clearVideoSsrcCache();
};
/**
* Closes underlying WebRTC PeerConnection instance and removes all remote
* tracks by emitting {@link RTCEvents.REMOTE_TRACK_REMOVED} for each one of
* them.
*/
TraceablePeerConnection.prototype.close = function() {
this.trace('stop');
// Off SignalingEvents
this.signalingLayer.off(
SignalingEvents.PEER_MUTED_CHANGED, this._peerMutedChanged);
this.signalingLayer.off(
SignalingEvents.PEER_VIDEO_TYPE_CHANGED, this._peerVideoTypeChanged);
for (const peerTracks of this.remoteTracks.values()) {
for (const remoteTrack of peerTracks.values()) {
this._removeRemoteTrack(remoteTrack);
}
}
this.remoteTracks.clear();
if (!this.rtc._removePeerConnection(this)) {
logger.error('RTC._removePeerConnection returned false');
}
if (this.statsinterval !== null) {
window.clearInterval(this.statsinterval);
this.statsinterval = null;
}
this.peerconnection.close();
};
/**
* 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
*/
const _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((a, i) => {
if (SDPUtil.findLine(
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('');
}
};
TraceablePeerConnection.prototype.createAnswer
= function(successCallback, failureCallback, constraints) {
this._createOfferOrAnswer(
false /* answer */, successCallback, failureCallback, constraints);
};
TraceablePeerConnection.prototype.createOffer
= function(successCallback, failureCallback, constraints) {
this._createOfferOrAnswer(
true /* offer */, successCallback, failureCallback, constraints);
};
/* eslint-disable max-params */
TraceablePeerConnection.prototype._createOfferOrAnswer
= function(isOffer, successCallback, failureCallback, constraints) {
const logName = isOffer ? 'Offer' : 'Answer';
this.trace(`create${logName}`, JSON.stringify(constraints, null, ' '));
const _successCallback = resultSdp => {
try {
this.trace(
`create${logName}OnSuccess::preTransform`, dumpSDP(resultSdp));
// if we're using unified plan, transform to Plan B.
if (RTCBrowserType.usesUnifiedPlan()) {
// eslint-disable-next-line no-param-reassign
resultSdp = this.interop.toPlanB(resultSdp);
this.trace(
`create${logName}OnSuccess::postTransform (Plan B)`,
dumpSDP(resultSdp));
}
/**
* We don't keep ssrcs consitent for Firefox because rewriting
* the ssrcs between createAnswer and setLocalDescription breaks
* the caching in sdp-interop (sdp-interop must know about all
* ssrcs, and it updates its cache in toPlanB so if we rewrite them
* after that, when we try and go back to unified plan it will
* complain about unmapped ssrcs)
*/
if (!RTCBrowserType.isFirefox()) {
// If there are no local video tracks, then a "recvonly"
// SSRC needs to be generated
if (!this.hasAnyTracksOfType(MediaType.VIDEO)
&& !this.sdpConsistency.hasPrimarySsrcCached()) {
this.generateRecvonlySsrc();
}
resultSdp.sdp
= this.sdpConsistency.makeVideoPrimarySsrcsConsistent(
resultSdp.sdp);
this.trace(
`create${logName}OnSuccess::postTransform `
+ '(make primary audio/video ssrcs consistent)',
dumpSDP(resultSdp));
}
// Add simulcast streams if simulcast is enabled
if (this.isSimulcastOn()) {
// eslint-disable-next-line no-param-reassign
resultSdp = this.simulcast.mungeLocalDescription(resultSdp);
this.trace(
`create${logName}`
+ 'OnSuccess::postTransform (simulcast)',
dumpSDP(resultSdp));
}
if (!this.options.disableRtx && RTCBrowserType.supportsRtx()) {
resultSdp.sdp
= this.rtxModifier.modifyRtxSsrcs(resultSdp.sdp);
this.trace(
`create${logName}`
+ 'OnSuccess::postTransform (rtx modifier)',
dumpSDP(resultSdp));
}
// Fix the setup attribute (see _fixAnswerRFC4145Setup for
// details)
if (!isOffer) {
const remoteDescription
= new SDP(this.remoteDescription.sdp);
const localDescription = new SDP(resultSdp.sdp);
_fixAnswerRFC4145Setup(remoteDescription, localDescription);
resultSdp.sdp = localDescription.raw;
}
const ssrcMap = extractSSRCMap(resultSdp);
logger.debug('Got local SSRCs MAP: ', ssrcMap);
this._processLocalSSRCsMap(ssrcMap);
successCallback(resultSdp);
} catch (e) {
this.trace(`create${logName}OnError`, e);
this.trace(`create${logName}OnError`, dumpSDP(resultSdp));
logger.error(`create${logName}OnError`, e, dumpSDP(resultSdp));
failureCallback(e);
}
};
const _errorCallback = err => {
this.trace(`create${logName}OnFailure`, err);
const eventType
= isOffer
? RTCEvents.CREATE_OFFER_FAILED
: RTCEvents.CREATE_ANSWER_FAILED;
this.eventEmitter.emit(eventType, err, this);
failureCallback(err);
};
// NOTE Temasys plugin does not support "bind" on peerconnection methods
if (isOffer) {
this.peerconnection.createOffer(
_successCallback, _errorCallback, constraints);
} else {
this.peerconnection.createAnswer(
_successCallback, _errorCallback, constraints);
}
};
/* eslint-enable max-params */
/**
* Extract primary SSRC from given {@link TrackSSRCInfo} object.
* @param {TrackSSRCInfo} ssrcObj
* @return {number|null} the primary SSRC or null
*/
function extractPrimarySSRC(ssrcObj) {
if (ssrcObj && ssrcObj.groups && ssrcObj.groups.length) {
return ssrcObj.groups[0].ssrcs[0];
} else if (ssrcObj && ssrcObj.ssrcs && ssrcObj.ssrcs.length) {
return ssrcObj.ssrcs[0];
}
return null;
}
/**
* Goes over the SSRC map extracted from the latest local description and tries
* to match them with the local tracks (by MSID). Will update the values
* currently stored in the {@link TraceablePeerConnection.localSSRCs} map.
* @param {Map} ssrcMap
* @private
*/
TraceablePeerConnection.prototype._processLocalSSRCsMap = function(ssrcMap) {
for (const track of this.localTracks.values()) {
const trackMSID = track.getMSID();
if (ssrcMap.has(trackMSID)) {
const newSSRC = ssrcMap.get(trackMSID);
if (!newSSRC) {
logger.error(`No SSRC found for: ${trackMSID} in ${this}`);
return;
}
const oldSSRC = this.localSSRCs.get(track.rtcId);
const newSSRCNum = extractPrimarySSRC(newSSRC);
const oldSSRCNum = extractPrimarySSRC(oldSSRC);
// eslint-disable-next-line no-negated-condition
if (newSSRCNum !== oldSSRCNum) {
if (oldSSRCNum === null) {
logger.info(
`Storing new local SSRC for ${track} in ${this}`,
newSSRC);
} else {
logger.error(
`Overwriting SSRC for ${track} ${trackMSID} in ${this
} with: `, newSSRC);
}
this.localSSRCs.set(track.rtcId, newSSRC);
} else {
logger.debug(
`The local SSRC(${newSSRCNum}) for ${track} ${trackMSID}`
+ `is still up to date in ${this}`);
}
} else {
logger.warn(`No local track matched with: ${trackMSID} in ${this}`);
}
}
};
TraceablePeerConnection.prototype.addIceCandidate
= function(candidate, successCallback, failureCallback) {
// var self = this;
this.trace('addIceCandidate', JSON.stringify(candidate, null, ' '));
this.peerconnection.addIceCandidate(
candidate, successCallback, failureCallback);
/* maybe later
this.peerconnection.addIceCandidate(candidate,
function () {
self.trace('addIceCandidateOnSuccess');
successCallback();
},
function (err) {
self.trace('addIceCandidateOnFailure', err);
failureCallback(err);
}
);
*/
};
TraceablePeerConnection.prototype.getStats = function(callback, errback) {
// TODO: Is this the correct way to handle Opera, Temasys?
if (RTCBrowserType.isFirefox()
|| RTCBrowserType.isTemasysPluginUsed()
|| RTCBrowserType.isReactNative()) {
this.peerconnection.getStats(
null,
callback,
errback || (() => {
// Making sure that getStats won't fail if error callback is
// not passed.
}));
} else {
this.peerconnection.getStats(callback);
}
};
/**
* Generates and stores new SSRC info object for given local track.
* The method should be called only for a video track being added to this TPC
* in the muted state (given that the current browser uses this strategy).
* @param {JitsiLocalTrack} track
* @return {TPCSSRCInfo}
*/
TraceablePeerConnection.prototype.generateNewStreamSSRCInfo = function(track) {
const rtcId = track.rtcId;
let ssrcInfo = this._getSSRC(rtcId);
if (ssrcInfo) {
logger.error(`Will overwrite local SSRCs for track ID: ${rtcId}`);
}
if (this.isSimulcastOn()) {
ssrcInfo = {
ssrcs: [],
groups: []
};
for (let i = 0; i < SIMULCAST_LAYERS; i++) {
ssrcInfo.ssrcs.push(SDPUtil.generateSsrc());
}
ssrcInfo.groups.push({
ssrcs: ssrcInfo.ssrcs.slice(),
semantics: 'SIM'
});
} else {
ssrcInfo = {
ssrcs: [ SDPUtil.generateSsrc() ],
groups: []
};
}
if (!this.options.disableRtx && RTCBrowserType.supportsRtx()) {
// Specifically use a for loop here because we'll
// be adding to the list we're iterating over, so we
// only want to iterate through the items originally
// on the list
const currNumSsrcs = ssrcInfo.ssrcs.length;
for (let i = 0; i < currNumSsrcs; ++i) {
const primarySsrc = ssrcInfo.ssrcs[i];
const rtxSsrc = SDPUtil.generateSsrc();
ssrcInfo.ssrcs.push(rtxSsrc);
ssrcInfo.groups.push({
ssrcs: [ primarySsrc, rtxSsrc ],
semantics: 'FID'
});
}
}
ssrcInfo.msid = track.storedMSID;
this.localSSRCs.set(rtcId, ssrcInfo);
return ssrcInfo;
};
/**
* Creates a text representation of this TraceablePeerConnection
* instance.
* @return {string}
*/
TraceablePeerConnection.prototype.toString = function() {
return `TPC[${this.id},p2p:${this.isP2P}]`;
};