@twilio/webrtc - npm Package Compare versions

Comparing version 2.1.4 to 2.2.0

CHANGELOG.md

@@ -0,1 +1,21 @@
+2.2.0 (January 10, 2019)
+========================
+
+New Features
+------------
+
+- `getStats` on Firefox will now consume the spec-compliant `RTCIceCandidateStats`
+  available in [versions 65 and above](https://www.fxsitecompat.com/en-CA/docs/2018/rtcicecandidatestats-has-been-updated-to-the-latest-spec/). (JSDK-2235)
+- Added support for Unified Plan SDPs on Safari 12.1. (JSDK-2231)
+- Removed workaround for this [Safari bug](https://bugs.webkit.org/show_bug.cgi?id=174323).
+- Removed workaround for this Chrome [bug](https://bugs.chromium.org/p/chromium/issues/detail?id=774303).
+  Now, we no longer suppress the RTCPeerConnection's native `RTCTrackEvent`. 
+- Worked around the [deprecation](https://blog.mozilla.org/webrtc/getstats-isremote-65/) of the
+  `isRemote` property in `RTCInboundRTPStreamStats` and `RTCOutboundRTPStreamStats` in Firefox.
+
+Bug Fixes
+---------
+
+- Worked around Chrome [Bug 894231](https://bugs.chromium.org/p/chromium/issues/detail?id=894231).
+
 2.1.4 (December 5, 2018)
@@ -2,0 +22,0 @@ =========================

lib/getstats.js

@@ -1,6 +0,12 @@
-/* global webkitRTCPeerConnection, mozRTCPeerConnection */
+/* global RTCRtpTransceiver */
 'use strict';
 
 var flatMap = require('./util').flatMap;
+var guessBrowser = require('./util').guessBrowser;
 
+var guess = guessBrowser();
+var isChrome = guess === 'chrome';
+var isFirefox = guess === 'firefox';
+var isSafari = guess === 'safari';
+
 /**
@@ -40,11 +46,14 @@ * Get the standardized {@link RTCPeerConnection} statistics.
   var trackStatsPromises = flatMap([
-    [localAudioTracks, 'localAudioTrackStats'],
-    [localVideoTracks, 'localVideoTrackStats'],
-    [remoteAudioTracks, 'remoteAudioTrackStats'],
-    [remoteVideoTracks, 'remoteVideoTrackStats']
-  ], function(pair) {
-    var tracks = pair[0];
-    var statsArrayName = pair[1];
+    [localAudioTracks, 'localAudioTrackStats', false],
+    [localVideoTracks, 'localVideoTrackStats', false],
+    [remoteAudioTracks, 'remoteAudioTrackStats', true],
+    [remoteVideoTracks, 'remoteVideoTrackStats', true]
+  ], function(triple) {
+    var tracks = triple[0];
+    var statsArrayName = triple[1];
+    var isRemote = triple[2];
     return tracks.map(function(track) {
-      return getTrackStats(peerConnection, track, options).then(function(stats) {
+      return getTrackStats(peerConnection, track, Object.assign({
+        isRemote: isRemote
+      }, options)).then(function(stats) {
         stats.trackId = track.id;
@@ -74,8 +83,8 @@ statsResponse[statsArrayName].push(stats);
 
-  if (typeof options.testForChrome !== 'undefined'
-    || typeof webkitRTCPeerConnection !== 'undefined') {
-    return peerConnection.getStats().then(standardizeChromeActiveIceCandidatePairStats);
+  if (typeof options.testForChrome !== 'undefined' || isChrome
+    || typeof options.testForSafari  !== 'undefined' || isSafari) {
+    return peerConnection.getStats().then(
+      standardizeChromeOrSafariActiveIceCandidatePairStats);
   }
-  if (typeof options.testForFirefox !== 'undefined'
-    || typeof mozRTCPeerConnection !== 'undefined') {
+  if (typeof options.testForFirefox !== 'undefined' || isFirefox) {
     return peerConnection.getStats().then(standardizeFirefoxActiveIceCandidatePairStats);
@@ -87,7 +96,7 @@ }
 /**
- * Standardize the active RTCIceCandidate pair's statistics in Chrome.
+ * Standardize the active RTCIceCandidate pair's statistics in Chrome or Safari.
  * @param {RTCStatsReport} stats
  * @returns {?StandardizedActiveIceCandidatePairStatsReport}
  */
-function standardizeChromeActiveIceCandidatePairStats(stats) {
+function standardizeChromeOrSafariActiveIceCandidatePairStats(stats) {
   var activeCandidatePairStats = Array.from(stats.values()).find(function(stat) {
@@ -188,6 +197,6 @@ return stat.type === 'candidate-pair' && stat.nominated;
     { key: 'candidateType', type: 'string' },
-    { key: 'ip', ffKey: 'ipAddress', type: 'string' },
-    { key: 'port', ffKey: 'portNumber', type: 'number' },
+    { key: 'ip', ffKeys: ['address', 'ipAddress'], type: 'string' },
+    { key: 'port', ffKeys: ['portNumber'], type: 'number' },
     { key: 'priority', type: 'number' },
-    { key: 'protocol', ffKey: 'transport', type: 'string' },
+    { key: 'protocol', ffKeys: ['transport'], type: 'string' },
     { key: 'url', type: 'string' }
@@ -210,3 +219,5 @@ ];
     ? standardizedLocalCandidateStatsKeys.reduce(function(report, keyInfo) {
-      var key = keyInfo.ffKey || keyInfo.key;
+      var key = keyInfo.ffKeys && keyInfo.ffKeys.find(function(key) {
+        return key in activeLocalCandidateStats;
+      }) || keyInfo.key;
       report[keyInfo.key] = typeof activeLocalCandidateStats[key] === keyInfo.type
@@ -223,3 +234,5 @@ ? key === 'candidateType'
     ? standardizedCandidateStatsKeys.reduce(function(report, keyInfo) {
-      var key = keyInfo.ffKey || keyInfo.key;
+      var key = keyInfo.ffKeys && keyInfo.ffKeys.find(function(key) {
+        return key in activeRemoteCandidateStats;
+      }) || keyInfo.key;
       report[keyInfo.key] = typeof activeRemoteCandidateStats[key] === keyInfo.type
@@ -300,10 +313,21 @@ ? key === 'candidateType'
 
-  if (typeof options.testForChrome !== 'undefined' ||
-    typeof webkitRTCPeerConnection !== 'undefined') {
-    return chromeGetTrackStats(peerConnection, track);
+  if (typeof options.testForChrome !== 'undefined' || isChrome) {
+    return chromeOrSafariGetTrackStats(peerConnection, track);
   }
-  if (typeof options.testForFirefox  !== 'undefined' ||
-    typeof mozRTCPeerConnection !== 'undefined') {
-    return firefoxGetTrackStats(peerConnection, track);
+  if (typeof options.testForFirefox  !== 'undefined' || isFirefox) {
+    return firefoxGetTrackStats(peerConnection, track, options.isRemote);
   }
+  if (typeof options.testForSafari  !== 'undefined' || isSafari) {
+    if (typeof options.testForSafari  !== 'undefined'
+      || 'currentDirection' in RTCRtpTransceiver.prototype) {
+      return chromeOrSafariGetTrackStats(peerConnection, track);
+    }
+    // NOTE(syerrapragada): getStats() is not supported on
+    // Safari versions where plan-b is the SDP format
+    // due to this bug: https://bugs.webkit.org/show_bug.cgi?id=192601
+    return Promise.reject(new Error([
+      'getStats() is not supported on this version of Safari',
+      'due to this bug: https://bugs.webkit.org/show_bug.cgi?id=192601'
+    ].join(' ')));
+  }
   return Promise.reject(new Error('RTCPeerConnection#getStats() not supported'));
@@ -313,3 +337,3 @@ }
 /**
- * Get the standardized statistics for a particular MediaStreamTrack in Chrome.
+ * Get the standardized statistics for a particular MediaStreamTrack in Chrome or Safari.
  * @param {RTCPeerConnection} peerConnection
@@ -319,7 +343,7 @@ * @param {MediaStreamTrack} track
  */
-function chromeGetTrackStats(peerConnection, track) {
+function chromeOrSafariGetTrackStats(peerConnection, track) {
   return new Promise(function(resolve, reject) {
-    peerConnection.getStats(function(response) {
-      resolve(standardizeChromeStats(response, track));
-    }, null, reject);
+    peerConnection.getStats(track).then(function(response) {
+      resolve(standardizeChromeOrSafariStats(response));
+    }, reject);
   });
@@ -332,8 +356,9 @@ }
  * @param {MediaStreamTrack} track
+ * @param {boolean} isRemote
  * @returns {Promise.<StandardizedTrackStatsReport>}
  */
-function firefoxGetTrackStats(peerConnection, track) {
+function firefoxGetTrackStats(peerConnection, track, isRemote) {
   return new Promise(function(resolve, reject) {
     peerConnection.getStats(track).then(function(response) {
-      resolve(standardizeFirefoxStats(response));
+      resolve(standardizeFirefoxStats(response, isRemote));
     }, reject);
@@ -344,72 +369,133 @@ });
 /**
- * Standardize the MediaStreamTrack's statistics in Chrome.
+ * Standardize the MediaStreamTrack's statistics in Chrome or Safari.
  * @param {RTCStatsResponse} response
- * @param {MediaStreamTrack} track
  * @returns {StandardizedTrackStatsReport}
  */
-function standardizeChromeStats(response, track) {
-  var ssrcReport = response.result().find(function(report) {
-    return report.type === 'ssrc' && report.stat('googTrackId') === track.id;
+function standardizeChromeOrSafariStats(response) {
+  var inbound = null;
+  var outbound = null;
+  var track = null;
+  var codec = null;
+
+  response.forEach(function(stat) {
+    switch (stat.type) {
+      case 'inbound-rtp':
+        inbound = stat;
+        break;
+      case 'outbound-rtp':
+        outbound = stat;
+        break;
+      case 'track':
+        track = stat;
+        break;
+      case 'codec':
+        codec = stat;
+        break;
+    }
   });
 
+  var isRemote = track && track.remoteSource;
   var standardizedStats = {};
 
-  if (ssrcReport) {
-    standardizedStats.timestamp = Math.round(Number(ssrcReport.timestamp));
-    standardizedStats = ssrcReport.names().reduce(function(stats, name) {
-      switch (name) {
-        case 'googCodecName':
-          stats.codecName = ssrcReport.stat(name);
-          break;
-        case 'googRtt':
-          stats.roundTripTime = Number(ssrcReport.stat(name));
-          break;
-        case 'googJitterReceived':
-          stats.jitter = Number(ssrcReport.stat(name));
-          break;
-        case 'googFrameWidthInput':
-          stats.frameWidthInput = Number(ssrcReport.stat(name));
-          break;
-        case 'googFrameHeightInput':
-          stats.frameHeightInput = Number(ssrcReport.stat(name));
-          break;
-        case 'googFrameWidthSent':
-          stats.frameWidthSent = Number(ssrcReport.stat(name));
-          break;
-        case 'googFrameHeightSent':
-          stats.frameHeightSent = Number(ssrcReport.stat(name));
-          break;
-        case 'googFrameWidthReceived':
-          stats.frameWidthReceived = Number(ssrcReport.stat(name));
-          break;
-        case 'googFrameHeightReceived':
-          stats.frameHeightReceived = Number(ssrcReport.stat(name));
-          break;
-        case 'googFrameRateInput':
-          stats.frameRateInput = Number(ssrcReport.stat(name));
-          break;
-        case 'googFrameRateSent':
-          stats.frameRateSent = Number(ssrcReport.stat(name));
-          break;
-        case 'googFrameRateReceived':
-          stats.frameRateReceived = Number(ssrcReport.stat(name));
-          break;
-        case 'ssrc':
-          stats[name] = ssrcReport.stat(name);
-          break;
-        case 'bytesReceived':
-        case 'bytesSent':
-        case 'packetsLost':
-        case 'packetsReceived':
-        case 'packetsSent':
-        case 'audioInputLevel':
-        case 'audioOutputLevel':
-          stats[name] = Number(ssrcReport.stat(name));
-          break;
-      }
+  var first = isRemote ? inbound : outbound;
+  var second = track;
+  var third = codec;
 
-      return stats;
-    }, standardizedStats);
+  function getStatValue(name) {
+    if (first && typeof first[name] !== 'undefined') {
+      return first[name];
+    }
+
+    if (second && typeof second[name] !== 'undefined') {
+      return second[name];
+    }
+
+    if (third && typeof third[name] !== 'undefined') {
+      return third[name];
+    }
+
+    return null;
   }
 
+  var ssrc = getStatValue('ssrc');
+  if (typeof ssrc === 'number') {
+    standardizedStats.ssrc = String(ssrc);
+  }
+
+  var timestamp = getStatValue('timestamp');
+  standardizedStats.timestamp = Math.round(timestamp);
+
+  var mimeType = getStatValue('mimeType');
+  if (typeof mimeType === 'string') {
+    mimeType = mimeType.split('/');
+    standardizedStats.codecName = mimeType[mimeType.length - 1];
+  }
+
+  var roundTripTime = getStatValue('roundTripTime');
+  if (typeof roundTripTime === 'number') {
+    standardizedStats.roundTripTime = roundTripTime;
+  }
+
+  var jitter = getStatValue('jitter');
+  if (typeof jitter === 'number') {
+    standardizedStats.jitter = Math.round(jitter * 1000);
+  }
+
+  var frameWidth = getStatValue('frameWidth');
+  if (typeof frameWidth === 'number') {
+    if (isRemote) {
+      standardizedStats.frameWidthReceived = frameWidth;
+    } else {
+      standardizedStats.frameWidthSent = frameWidth;
+    }
+  }
+
+  var frameHeight = getStatValue('frameHeight');
+  if (typeof frameHeight === 'number') {
+    if (isRemote) {
+      standardizedStats.frameHeightReceived = frameHeight;
+    } else {
+      standardizedStats.frameHeightSent = frameHeight;
+    }
+  }
+
+  var framesPerSecond = getStatValue('framesPerSecond');
+  if (typeof framesPerSecond === 'number') {
+    standardizedStats.frameRateSent = framesPerSecond;
+  }
+
+  var bytesReceived = getStatValue('bytesReceived');
+  if (typeof bytesReceived === 'number') {
+    standardizedStats.bytesReceived = bytesReceived;
+  }
+
+  var bytesSent = getStatValue('bytesSent');
+  if (typeof bytesSent === 'number') {
+    standardizedStats.bytesSent = bytesSent;
+  }
+
+  var packetsLost = getStatValue('packetsLost');
+  if (typeof packetsLost === 'number') {
+    standardizedStats.packetsLost = packetsLost;
+  }
+
+  var packetsReceived = getStatValue('packetsReceived');
+  if (typeof packetsReceived === 'number') {
+    standardizedStats.packetsReceived = packetsReceived;
+  }
+
+  var packetsSent = getStatValue('packetsSent');
+  if (typeof packetsSent === 'number') {
+    standardizedStats.packetsSent = packetsSent;
+  }
+
+  var audioLevel = getStatValue('audioLevel');
+  if (typeof audioLevel === 'number') {
+    if (isRemote) {
+      standardizedStats.audioOutputLevel = audioLevel;
+    } else {
+      standardizedStats.audioInputLevel = audioLevel;
+    }
+  }
+
   return standardizedStats;
@@ -421,5 +507,6 @@ }
  * @param {RTCStatsReport} response
+ * @param {boolean} isRemote
  * @returns {StandardizedTrackStatsReport}
  */
-function standardizeFirefoxStats(response) {
+function standardizeFirefoxStats(response, isRemote) {
   // NOTE(mroberts): If getStats is called on a closed RTCPeerConnection,
@@ -433,32 +520,44 @@ // Firefox returns undefined instead of an RTCStatsReport. We workaround this
 
-  var inbound = Array.from(response.values()).find(function(stat) {
-    return stat.type === 'inbound-rtp';
-  });
+  var inbound = null;
+  var outbound = null;
 
-  var outbound = Array.from(response.values()).find(function(stat) {
-    return stat.type === 'outbound-rtp';
+  // NOTE(mmalavalli): Starting from Firefox 63, RTC{Inbound, Outbound}RTPStreamStats.isRemote
+  // will be deprecated, followed by its removal in Firefox 66. Also, trying to
+  // access members of the remote RTC{Inbound, Outbound}RTPStreamStats without
+  // using RTCStatsReport.get(remoteId) will trigger console warnings. So, we
+  // no longer depend on "isRemote", and we call RTCStatsReport.get(remoteId)
+  // to access the remote RTC{Inbound, Outbound}RTPStreamStats.
+  //
+  // Source: https://blog.mozilla.org/webrtc/getstats-isremote-65/
+  //
+  response.forEach(function(stat) {
+    if (stat.isRemote) {
+      return;
+    }
+    switch (stat.type) {
+      case 'inbound-rtp':
+        inbound = stat;
+        outbound = response.get(stat.remoteId);
+        break;
+      case 'outbound-rtp':
+        outbound = stat;
+        inbound = response.get(stat.remoteId);
+        break;
+    }
   });
 
-  var standardizedStats = {};
+  var first = isRemote ? inbound : outbound;
+  var second = isRemote ? outbound : inbound;
 
   function getStatValue(name) {
-    var first = outbound;
-    var second = inbound;
-
-    if (outbound && outbound.isRemote) {
-      first = inbound;
-      second = outbound;
-    }
-
     if (first && typeof first[name] !== 'undefined') {
       return first[name];
     }
-
     if (second && typeof second[name] !== 'undefined') {
       return second[name];
     }
-
     return null;
   }
 
+  var standardizedStats = {};
   var timestamp = getStatValue('timestamp');
@@ -468,4 +567,4 @@ standardizedStats.timestamp = Math.round(timestamp);
   var ssrc = getStatValue('ssrc');
-  if (typeof ssrc === 'string') {
-    standardizedStats.ssrc = ssrc;
+  if (typeof ssrc === 'number') {
+    standardizedStats.ssrc = String(ssrc);
   }
@@ -488,3 +587,3 @@
 
-  var roundTripTime = getStatValue('mozRtt');
+  var roundTripTime = getStatValue('roundTripTime');
   if (typeof roundTripTime === 'number') {
@@ -522,3 +621,2 @@ standardizedStats.roundTripTime = roundTripTime;
 
-
 /**
@@ -525,0 +623,0 @@ * Standardized RTCIceCandidate statistics.

lib/rtcpeerconnection/chrome.js

@@ -46,9 +46,2 @@ /* globals RTCDataChannel, RTCSessionDescription, webkitRTCPeerConnection */
 
-  // NOTE(mmalavalli): Because of a bug related to "ontrack", we prevent it
-  // from being delegated to ChromeRTCPeerConnection. For now, this bug
-  // manifests when we run Chrome with the flag: --enable-blink-features=RTCRtpSender
-  // Existing bug: https://bugs.chromium.org/p/chromium/issues/detail?id=774303
-  // Bug filed by us: https://bugs.chromium.org/p/chromium/issues/detail?id=783433
-  util.interceptEvent(this, 'track');
-
   /* eslint new-cap:0 */
@@ -138,9 +131,13 @@ var peerConnection = new PeerConnection(newConfiguration, constraints);
 // supported natively in Chrome.
+// NOTE(mmalavalli): This shim also works around a Chrome bug in "unified-plan"
+// SDPs where adding a MediaStreamTrack that was previously added and removed
+// generates an SDP where the MSID does not match the MediaStreamTrack ID.
+//
+// Chrome bug: https://bugs.chromium.org/p/chromium/issues/detail?id=894231
+//
 ChromeRTCPeerConnection.prototype.addTrack = function addTrack() {
   var args = [].slice.call(arguments);
-  if (this._peerConnection.addTrack) {
-    return this._peerConnection.addTrack.apply(this._peerConnection, args);
-  }
+  var track = args[0];
+  var sender = this._senders.get(track);
 
-  var track = args[0];
   if (this._peerConnection.signalingState === 'closed') {
@@ -150,4 +147,2 @@ throw new Error('Cannot add MediaStreamTrack [' + track.id + ', '
   }
-
-  var sender = this._senders.get(track);
   if (sender && sender.track) {
@@ -157,2 +152,12 @@ throw new Error('Cannot add MediaStreamTrack [' + track.id + ', '
   }
+  if (this._peerConnection.addTrack) {
+    if (this._sdpSemantics === 'unified-plan') {
+      sender = getActiveSenders(this._peerConnection).get(track)
+        || this._peerConnection.addTrack.apply(this._peerConnection, args);
+      this._senders.set(track, sender);
+      return sender;
+    }
+    return this._peerConnection.addTrack.apply(this._peerConnection, args);
+  }
+
   this._peerConnection.removeStream(this._localStream);
@@ -167,2 +172,18 @@ this._localStream.addTrack(track);
 
+// NOTE(mmalavalli): This shim works around a Chrome bug in "unified-plan"
+// SDPs where adding a MediaStreamTrack that was previously added and removed
+// generates an SDP where the MSID does not match the MediaStreamTrack ID.
+//
+// Chrome bug: https://bugs.chromium.org/p/chromium/issues/detail?id=894231
+//
+if (PeerConnection.prototype.addTransceiver) {
+  ChromeRTCPeerConnection.prototype.addTransceiver = function addTransceiver() {
+    var transceiver = this._peerConnection.addTransceiver.apply(this._peerConnection, arguments);
+    var sender = transceiver.sender;
+    var track = sender.track;
+    this._senders.set(track, sender);
+    return transceiver;
+  };
+}
+
 // NOTE(mmalavalli): This shim supports our limited case of removing
@@ -172,2 +193,8 @@ // MediaStreamTracks from one MediaStream. It has been implemented this
 // supported natively in Chrome.
+// NOTE(mmalavalli): This shim also works around a Chrome bug in "unified-plan"
+// SDPs where adding a MediaStreamTrack that was previously added and removed
+// generates an SDP where the MSID does not match the MediaStreamTrack ID.
+//
+// Chrome bug: https://bugs.chromium.org/p/chromium/issues/detail?id=894231
+//
 ChromeRTCPeerConnection.prototype.removeTrack = function removeTrack(sender) {
@@ -180,3 +207,9 @@ if (this._peerConnection.signalingState === 'closed') {
     try {
-      this._peerConnection.removeTrack(sender);
+      if (this._sdpSemantics === 'unified-plan') {
+        if (sender.track) {
+          this._senders.set(sender.track, new RTCRtpSenderShim(null));
+        }
+      } else {
+        this._peerConnection.removeTrack(sender);
+      }
     }
@@ -206,6 +239,5 @@ catch (error) {
 ChromeRTCPeerConnection.prototype.getSenders = function getSenders() {
-  if (this._peerConnection.getSenders) {
-    return this._peerConnection.getSenders();
-  }
-  return Array.from(this._senders.values());
+  return this._peerConnection.getSenders && this._sdpSemantics === 'plan-b'
+    ? this._peerConnection.getSenders()
+    : Array.from(this._senders.values());
 };
@@ -257,8 +289,3 @@
   if (this._pendingRemoteOffer) {
-    var mediaStreamTracks = util.flatMap(this.getRemoteStreams(), function(mediaStream) {
-      return mediaStream.getTracks();
-    });
-
     promise = this._peerConnection.setRemoteDescription(this._pendingRemoteOffer).then(function setRemoteDescriptionSucceeded() {
-      maybeDispatchTrackEvents(self, mediaStreamTracks);
       // NOTE(mroberts): The signalingStates between the ChromeRTCPeerConnection
@@ -340,19 +367,2 @@ // and the underlying RTCPeerConnection implementation have converged. We
 
-// Dispatch 'track' events to ChromeRTCPeerConnection if new
-// MediaStreamTracks have been added. This is a temporary workaround
-// for the unreliable MediaStreamTrack#addtrack event. Do this only if
-// the native RTCPeerConnection has not implemented 'ontrack'.
-function maybeDispatchTrackEvents(peerConnection, mediaStreamTracks) {
-  var currentMediaStreamTracks = util.flatMap(peerConnection.getRemoteStreams(), function(mediaStream) {
-    return mediaStream.getTracks();
-  });
-  var mediaStreamTracksAdded = util.difference(currentMediaStreamTracks, mediaStreamTracks);
-
-  mediaStreamTracksAdded.forEach(function(mediaStreamTrack) {
-    var newEvent = new Event('track');
-    newEvent.track = mediaStreamTrack;
-    peerConnection.dispatchEvent(newEvent);
-  });
-}
-
 // NOTE(mroberts): We workaround Chrome's lack of rollback support, per the
@@ -381,6 +391,2 @@ // workaround suggested here: https://bugs.chromium.org/p/webrtc/issues/detail?id=5738#c3
 
-  var mediaStreamTracks = util.flatMap(peerConnection.getRemoteStreams(), function(mediaStream) {
-    return mediaStream.getTracks();
-  });
-
   var pendingLocalOffer = local ? peerConnection._pendingLocalOffer : peerConnection._pendingRemoteOffer;
@@ -435,14 +441,6 @@ var pendingRemoteOffer = local ? peerConnection._pendingRemoteOffer : peerConnection._pendingLocalOffer;
 
-  return promise || peerConnection._peerConnection[setLocalDescription](unwrap(description)).then(function() {
-    if (!local) {
-      maybeDispatchTrackEvents(peerConnection, mediaStreamTracks);
-    }
-  });
+  return promise || peerConnection._peerConnection[setLocalDescription](unwrap(description));
 }
 
 function setRemoteAnswer(peerConnection, answer) {
-  var mediaStreamTracks = util.flatMap(peerConnection.getRemoteStreams(), function(mediaStream) {
-    return mediaStream.getTracks();
-  });
-
   // Apply the pending local offer.
@@ -454,3 +452,2 @@ var pendingLocalOffer = peerConnection._pendingLocalOffer;
   }).then(function setRemoteAnswerSucceeded() {
-    maybeDispatchTrackEvents(peerConnection, mediaStreamTracks);
     // NOTE(mroberts): The signalingStates between the ChromeRTCPeerConnection
@@ -526,2 +523,15 @@ // and the underlying RTCPeerConnection implementation have converged. We
 /**
+ * Gets the active RTCRtpSenders of the RTCPeerConnection.
+ * @param peerConnection
+ * @returns {Map<MediaStreamTrack, RTCRtpSender>}
+ */
+function getActiveSenders(peerConnection) {
+  return new Map(peerConnection.getSenders().filter(function(sender) {
+    return sender.track;
+  }).map(function(sender) {
+    return [sender.track, sender];
+  }));
+}
+
+/**
  * Get the actual `sdpSemantics`.
@@ -532,5 +542,10 @@ * @param {SdpSemantics} sdpSemantics
 function getSdpSemantics(sdpSemantics) {
+  // NOTE(mmalavalli): Once Chrome stops supporting "plan-b" SDPs, it will
+  // ignore the "sdpSemantics" flag. So, in order to differentiate between
+  // versions of Chrome which support only "plan-b" SDPs and versions of Chrome
+  // which support only "unified-plan" SDPs, which check whether PeerConnection's
+  // addStream() method is present.
   return sdpUtils.checkIfSdpSemanticsIsSupported()
     ? sdpSemantics
-    : 'plan-b';
+    : PeerConnection.prototype.addStream ? 'plan-b' : 'unified-plan';
 }
@@ -537,0 +552,0 @@

lib/rtcpeerconnection/safari.js

@@ -1,2 +0,2 @@
-/* globals RTCPeerConnection, RTCSessionDescription */
+/* globals RTCPeerConnection, RTCRtpTransceiver, RTCSessionDescription */
 'use strict';
@@ -8,5 +8,11 @@
 var RTCRtpSenderShim = require('../rtcrtpsender');
-var updateTracksToSSRCs = require('../util/sdp').updatePlanBTrackIdsToSSRCs;
+var sdpUtils = require('../util/sdp');
 var util = require('../util');
 
+var isUnifiedPlan = 'currentDirection' in RTCRtpTransceiver.prototype;
+
+var updateTrackIdsToSSRCs = isUnifiedPlan
+  ? sdpUtils.updateUnifiedPlanTrackIdsToSSRCs
+  : sdpUtils.updatePlanBTrackIdsToSSRCs;
+
 function SafariRTCPeerConnection(configuration) {
@@ -59,12 +65,6 @@ if (!(this instanceof SafariRTCPeerConnection)) {
     },
-    // NOTE(mroberts): Keep this here until the following is fixed.
-    //
-    //   https://bugs.webkit.org/show_bug.cgi?id=174323
-    //
     localDescription: {
       enumerable: true,
       get: function() {
-        return this._isClosed
-          ? null
-          : this._pendingLocalOffer || this._peerConnection.localDescription;
+        return this._pendingLocalOffer || this._peerConnection.localDescription;
       }
@@ -91,5 +91,3 @@ },
       get: function() {
-        return this._isClosed
-          ? null
-          : this._pendingRemoteOffer || this._peerConnection.remoteDescription;
+        return this._pendingRemoteOffer || this._peerConnection.remoteDescription;
       }
@@ -165,10 +163,22 @@ },
   // good enough for our application.
-  if (options.offerToReceiveAudio && !this._audioTransceiver) {
+  if (options.offerToReceiveAudio && !this._audioTransceiver && !(isUnifiedPlan && hasSendersForTracksOfKind(this, 'audio'))) {
     delete options.offerToReceiveAudio;
-    this._audioTransceiver = this.addTransceiver('audio');
+    try {
+      this._audioTransceiver = isUnifiedPlan
+        ? this.addTransceiver('audio', { direction: 'recvonly' })
+        : this.addTransceiver('audio');
+    } catch (e) {
+      return Promise.reject(e);
+    }
   }
 
-  if (options.offerToReceiveVideo && !this._videoTransceiver) {
+  if (options.offerToReceiveVideo && !this._videoTransceiver && !(isUnifiedPlan && hasSendersForTracksOfKind(this, 'video'))) {
     delete options.offerToReceiveVideo;
-    this._videoTransceiver = this.addTransceiver('video');
+    try {
+      this._videoTransceiver = isUnifiedPlan
+        ? this.addTransceiver('video', { direction: 'recvonly' })
+        : this.addTransceiver('video');
+    } catch (e) {
+      return Promise.reject(e);
+    }
   }
@@ -179,3 +189,3 @@
       type: offer.type,
-      sdp: updateTracksToSSRCs(self._tracksToSSRCs, offer.sdp)
+      sdp: updateTrackIdsToSSRCs(self._tracksToSSRCs, offer.sdp)
     });
@@ -194,3 +204,6 @@ });
       self._pendingRemoteOffer = null;
-      return answer;
+      return isUnifiedPlan ? new RTCSessionDescription({
+        type: answer.type,
+        sdp: updateTrackIdsToSSRCs(self._tracksToSSRCs, answer.sdp)
+      }) : answer;
     }, function setRemoteDescriptionOrCreateAnswerFailed(error) {
@@ -202,3 +215,8 @@ self._pendingRemoteOffer = null;
 
-  return this._peerConnection.createAnswer(options);
+  return this._peerConnection.createAnswer(options).then(function(answer) {
+    return isUnifiedPlan ? new RTCSessionDescription({
+      type: answer.type,
+      sdp: updateTrackIdsToSSRCs(self._tracksToSSRCs, answer.sdp)
+    }) : answer;
+  });
 };
@@ -259,2 +277,18 @@
 
+// NOTE(mmalavalli): This shim works around a Safari bug in "unified-plan"
+// SDPs where adding a MediaStreamTrack that was previously added and removed
+// generates an SDP where the MSID does not match the MediaStreamTrack ID.
+//
+// Safari bug: https://bugs.webkit.org/show_bug.cgi?id=192101
+//
+if (RTCPeerConnection.prototype.addTransceiver) {
+  SafariRTCPeerConnection.prototype.addTransceiver = function addTransceiver() {
+    var transceiver = this._peerConnection.addTransceiver.apply(this._peerConnection, arguments);
+    var sender = transceiver.sender;
+    var track = sender.track;
+    this._senders.set(track, sender);
+    return transceiver;
+  };
+}
+
 // NOTE(mroberts): We can't really remove tracks right now, at least if we
@@ -265,2 +299,8 @@ // ever want to add them back...
 //
+// NOTE(mmalavalli): This shim also works around a Safari bug in "unified-plan"
+// SDPs where adding a MediaStreamTrack that was previously added and removed
+// generates an SDP where the MSID does not match the MediaStreamTrack ID.
+//
+// Safari bug: https://bugs.webkit.org/show_bug.cgi?id=192101
+//
 SafariRTCPeerConnection.prototype.removeTrack = function removeTrack(sender) {
@@ -362,2 +402,15 @@ if (this._isClosed) {
 /**
+ * Whether a SafariRTCPeerConnection has any RTCRtpSender(s) for the given
+ * MediaStreamTrack kind.
+ * @param {SafariRTCPeerConnection} peerConnection
+ * @param {'audio' | 'video'} kind
+ * @returns {boolean}
+ */
+function hasSendersForTracksOfKind(peerConnection, kind) {
+  return !!peerConnection.getTransceivers().find(function(transceiver) {
+    return transceiver.sender && transceiver.sender.track && transceiver.sender.track.kind === kind;
+  });
+}
+
+/**
  * Shim an RTCDataChannel. This function mutates the RTCDataChannel.
@@ -364,0 +417,0 @@ * @param {RTCDataChannel} dataChannel

package.json

 {
   "name": "@twilio/webrtc",
-  "version": "2.1.4",
+  "version": "2.2.0",
   "description": "WebRTC-related APIs and shims used by twilio-video.js",
@@ -46,3 +46,3 @@ "scripts": {
     "karma-mocha": "^1.3.0",
-    "karma-safaritechpreview-launcher": "0.0.6",
+    "karma-safari-launcher": "^1.0.0",
     "karma-spec-reporter": "0.0.31",
@@ -55,4 +55,4 @@ "mocha": "^3.5.0",
     "watchify": "^3.9.0",
-    "webrtc-adapter": "^6.0.1"
+    "webrtc-adapter": "^6.4.8"
   }
 }

README.md

@@ -80,5 +80,5 @@ twilio-webrtc.js
   more information.
-* Does not depend on the native "track" event (currently behind the flag: `--enable-blink-features=RTCRtpSender`) because
-  of [this bug](https://bugs.chromium.org/p/chromium/issues/detail?id=774303), which partly refers to "ontrack" not firing when expected.
-  We have filed [this bug](https://bugs.chromium.org/p/chromium/issues/detail?id=783433) specifically for this.
+* Works around a [this bug](https://bugs.chromium.org/p/chromium/issues/detail?id=894231)
+  in "unified-plan" SDPs where adding a `MediaStreamTrack` that was previously added and
+  removed generates an SDP where the MSID does not match the `MediaStreamTrack` ID.
 
@@ -106,4 +106,2 @@ #### Firefox
   `MediaStreamTrack`.
-* Provides a workaround for [this bug](https://bugs.webkit.org/show_bug.cgi?id=174323), where
-  trying to access the `localDescription` or `remoteDescription` throws an exception.
 * Provides a shim for the `removeTrack` method in order to work around [this bug](https://bugs.webkit.org/show_bug.cgi?id=174327).
@@ -110,0 +108,0 @@ * Provides a shim for the `addTrack` method. Since `removeTrack` is shimmed, there is a necessity to

No alert changes

Improved metrics

Total package byte prevSize

116025

increased by5.83%

Lines of code

2693

increased by5.98%

Worsened metrics

Number of lines in readme file

124

decreased by-1.59%

No dependency changes